| Entries |
| Document | Title | Date |
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| 20080198839 | SYSTEM AND METHOD OF COMMUNICATION IN AN IP MULTIMEDIA SUBSYSTEM NETWORK - A system and method of communication in an IMS network is disclosed. An apparatus that incorporates teachings of the present disclosure may include, for example, a call processing server having a controller element that receives from a terminal device a calling ID for establishing communications with a called party, submits to a telephone number mapping (ENUM) server a query corresponding to the calling ID, receives from the ENUM server a plurality of communication identifiers retrieved from a Naming Authority Pointer record according to a grade of service (GoS) of the called party, and selects according to the GoS of the called party a communication identifier from the plurality of communication identifiers to establish communications with the called party. Additional embodiments are disclosed. | 08-21-2008 |
| 20080198840 | IP EXTENSION PHONE SYSTEM AND SERVER SYSTEM - There is provided an IP extension phone system including: a first server; a second server; and a portable IP phone, wherein when the second server detects connection of the portable IP phone, the second server assigns an IP address corresponding to the connected position to the portable IP phone, the portable IP phone registers the IP address in the first server, and the first server records an extension number and the IP address of the portable IP phone in an extension number management table and further records the extension number in a group management table while associating the extension number with a group corresponding to a segment of the IP address. | 08-21-2008 |
| 20080198841 | Uma Cs/Ps Split Architecture and Interface - An UMA network controller (SGW,D | 08-21-2008 |
| 20080198842 | PHONE APPLIANCE WITH DISPLAY SCREEN AND METHODS OF USING THE SAME - A phone appliance and method of use are provided where the phone appliance can be used to make VoIP communications calls. In a preferred embodiment, the phone appliance includes an RF connection for connecting to a computer or other computing device for facilitating the placement of the VoIP communications calls. The phone appliance further includes a display or portal for depicting advertisements provided by various advertisers. The advertisements provided can be used to defray all or part of the cost associated with making VoIP communications calls. The portal can also be used to communicate with businesses for ordering products, such as ordering a pizza, and to perform various services, such as purchasing stocks. In an exemplary system, the phone appliance is used to transmit to a control center information related to the user of the phone appliance, such as interests and buying habits, and queries for receiving additional information for various advertised products and services. The control center transmits the queries to the appropriate vendors for providing the user with additional information. Other functions and features are provided to the phone appliance, such as being able to download e-mail messages stored within or received by the computer. | 08-21-2008 |
| 20080205378 | SYSTEM AND METHOD FOR RECORDING AND MONITORING COMMUNICATIONS USING A MEDIA SERVER - A communication system including a media server through which communication packets are exchanged for recording and monitoring purposes is disclosed. A tap is associated with each communication endpoint allowing for cradle to grave recording of communications despite their subsequent routing or branching. An incoming communication is routed to a first tap and upon selection of a receiving party; the first tap is routed to a second tap which forwards communication packets on to the receiving party. The taps may be used to forward communication packets to any number of other taps or destinations, such as a recording device, monitoring user, or other user in the form of a conference. | 08-28-2008 |
| 20080205379 | Systems and methods for enabling IP signaling in wireless networks - Under one aspect, a system for transmitting an IP-based message from an initiator to a receiver lacking an IP address via a packet-switched (PS) network capable of communicating IP-based messages and a circuit-switched (CS) network capable of communicating non-IP-based messages, includes: a serving node in communication with the PS network and the CS network, the serving node including logic to: receive the IP-based message from the initiator over the PS network; generate a trigger message responsive to the IP-based message, the trigger message including a non-IP-based message including instructions for the receiver to initiate a connection to the PS network; and transmit the trigger message to the receiver via the CS network, and the receiver including logic to initiate a connection to the PS network and to receive the IP-based message responsive to the trigger message. Methods are also disclosed. | 08-28-2008 |
| 20080205380 | SWITCHBOARD FOR MULTIPLE DATA RATE COMMUNICATION SYSTEM - A switchboard device and methods of operation of same are disclosed. Embodiments of the invention may provide a flexible means of interconnecting wideband and narrowband communications interfaces, where wideband communications interfaces may transfer low-band data and high-band data, and narrowband communication interfaces may transfer low-band data. Low-band data may be combined and sent to a narrowband communications interface or a wideband communications interface. High-band data may be combined and sent to a wideband communications interface. The low-band data may represent audio signals below a predetermined frequency, while the high-band data may represent audio signals above the predetermined frequency. The predetermined frequency may be, for example, approximately 4 kHz. The spectral mask of the low-band data may meet the spectral mask of G.712. Methods of operating embodiments of the present invention are included. An additional aspect of the present invention may include machine-readable storage having stored thereon a computer program having a plurality of code sections executable by a machine for causing the machine to perform the foregoing. | 08-28-2008 |
| 20080205381 | METHOD FOR PROVIDING ON-LINE CHARGING AND DEVICE AND SYSTEM THEREOF - The present invention discloses a method for providing an online-charging to solve a problem that a related charging can not be processed correctly for a service involving simultaneously a CS domain and an IMS. The method includes: not invoking an online charging for the user in the CS domain when the user, who subscribes for the service involving simultaneously the CS domain and the IMS and the online charging service, originates or terminates a call in the CS domain; and performing the credit control in the CS domain and/or the IMS for the user in the IMS when the call of the user is processed through the IMS. The present invention also discloses a device and a system for an online credit control. | 08-28-2008 |
| 20080205382 | Intelligent routing of VoIP traffic - A method of routing communications traffic includes receiving, at a switch or server of an originating service provider, incoming communications traffic including a terminating number or address. The switch or server is programmed to send a query to a transaction server for a carrier routing. The transaction server determines whether the terminating number or address matches a number in an ENUM database in communication with the transaction server. If a match is found, the carrier associated with the matched number is added to a routing matrix generated by an offline routing matrix generation process to form a supplemental routing matrix. The transaction server then generates using the supplemental routing matrix a list of available route options for delivering the communications traffic to the terminating number or address. | 08-28-2008 |
| 20080205383 | METHOD AND APPARATUS FOR ROUTING DATA - A method and apparatus for handling internet access telephone calls made via cable company telephone services. A head end data terminal receives cable signals and converts them into individual signals. An intelligent switch detects signals destined for an internet service provider and routes those signals on a separate path to the internet service provider. A central switch routes the other signals along a telephone network. A computer program can control the steps of receiving cable signals, converting them into voice band signals, routing the signals that are not for the intended recipient to a central switch, multiplexing the signals for the intended recipient together, and sending the multiplexed signals to the intended recipient. | 08-28-2008 |
| 20080205384 | METHOD AND APPARATUS FOR IMPLEMENTING A HIGH-RELIABILITY LOAD BALANCED EASILY UPGRADEABLE PACKET TECHNOLOGY - A network is defined with several alternative softswitches/proxies, which may be used for communication. Each softswitch/proxy has a unique Internet Protocol (IP) address. The softswitches/proxies receive configuration data from a centralized user-profile server, which maintains user-profile information. A centralized call-detail record (CDR) server also is connected to each softswitch/proxy and maintains CDRs on each user on each softswitch/proxy. Based on the network configuration, an end-device configuration system generates a provisioning file. The provisioning file includes the IP addresses of each softswitch/proxy. The provisioning file is communicated to user devices. Each user device accesses the provisioning file and uses the IP address for communication. Should the communication fail for any reason, the user device may autonomously access the provisioning file and initiate another call using the next IP address in the provisioning file. This process may continue until a call is completed. | 08-28-2008 |
| 20080212566 | Method and apparatus for transmitting and receiving VOIP packet with UDP checksum in wireless communication system - A method and apparatus for transmitting a voice packet through a radio link in a mobile communication system providing a voice service through a packet network inter-working with the Internet are provided, in which a voice packet based on an Internet protocol (e.g. a VoIP packet) is received, the voice packet comprising headers including a User Datagram Protocol (UDP) checksum; the voice packet is verified by using the UDP checksum, to determine if the voice packet has an error; the headers are compressed to construct a header-compressed packet including the UDP checksum and a Cyclic Redundancy Check (CRC) code calculated for other header fields, except for the UDP checksum from among the headers, when the voice packet has no error, the UDP checksum from the header-compressed packet is deleted to construct a header-compressed packet from which the UDP checksum has been deleted, and the header-compressed packet is transmitted without the UDP checksum through a wireless channel. | 09-04-2008 |
| 20080212567 | Method And Apparatus For Non-Intrusive Single-Ended Voice Quality Assessment In Voip - An apparatus ( | 09-04-2008 |
| 20080212568 | WIRE AND WIRELESS INTERNET PHONE TERMINAL USING WIDEBAND VOICE CODEC - A wire/wireless Internet phone terminal using a wideband voice codec is provided. The wire/wireless Internet phone terminal using a wideband voice codec includes: a multimedia application processor for including a process core to perform a protocol according to a wire and wireless interface communication scheme and supporting wideband voice service; an Ethernet processing unit for connecting the multimedia application processor to the Ethernet to perform an Ethernet physical-layer function of and transforming a power input from the Ethernet to supply a driving power to the multimedia application processor; a PSTN (public switched telephone network) processing unit for connected to the multimedia application processor and a PSTN to emulate a telephone function; and a wireless processing unit for connecting the multimedia application processor to an AP (access point) in a wireless manner. Accordingly, it is possible to provide a wideband service without a limitation to wire/wireless implementation. | 09-04-2008 |
| 20080212569 | Method and Apparatus for Allocating Application Servers in an Ims - A method, application server, and Serving Call/Session Control Function (S-CSCF) for allocating a SIP Application Server to a subscriber in an IP Multimedia Subsystem. A Home Subscriber Server (HSS) identifies for the subscriber, a set of provisioned initial filter criteria, which contains at least one generic SIP Application Server identity. The HSS sends the filter criteria to an S-CSCF allocated to the subscriber. The S-CSCF resolves the generic SIP Application Server identity into a plurality of application server addresses, with one of the addresses being allocated to the subscriber for use in provisioning a service to the subscriber. The S-CSCF caches the address allocated to the subscriber for subsequent use. | 09-04-2008 |
| 20080212570 | System and method for selectively coupling various communication devices through common channel - A communication system with a common channel for selectively coupling a subscriber line port of a telephone system to one of a plurality of voice communication devices including a fixed network communication module, an internet-based phone module, and a wireless internet-based phone module is provided. When a phone unit of the telephone system dials a called end phone number, the telephone system is connected to the common channel, a ring current generated by the ring current generator is then supplied to the phone unit, the called end phone number is decoded by the tone decoder, and thereby the microprocessor in correspondence to the called end phone number couple the telephone system to one of the fixed network communication module, the internet-based phone module, or the wireless internet-based phone module to establish a voice communication therebetween according to a preset data table stored in the memory. | 09-04-2008 |
| 20080212571 | METHOD AND SYSTEM FOR MONITORING AND RECORDING VOICE FROM CIRCUIT-SWITCHED SWITCHES VIA A PACKET-SWITCHED NETWORK - Some embodiments of the present invention are directed to a method and system for monitoring and recording voice from circuit-switched switches via a_packet-switched network. A circuit-switched or VoIP recording system may record and/or live-monitor telephone calls by trunk and/or extension tapping over a packet-switched network. Alternatively, a circuit-switched or VoIP recording system may record and/or live-monitor telephone calls over a packet-switched network by activating the service observation feature of the circuit-switched switch either by feature code dialing or a computer telephony integration (CTI) link command. | 09-04-2008 |
| 20080212572 | Extended Handset Functionality and Mobility - A system includes an enterprise network having a call control system that manages telephony services for wireless handsets. At a remote site, a computing device establishes a secure, wireline communication session with the enterprise network. The computing device also establishes wireless, packet-based links with one or more handsets. The device acts as a relay to enable the handsets to receive telephony services managed by the enterprise network even though outside of the service area of the enterprise network. | 09-04-2008 |
| 20080219240 | Digital browser phone - A telephone system wherein all the functions of a digital telephone can be accessed and implemented on a personal computer alone, thereby eliminating the need for a telephone set. By means of the computer display and mouse, keyboard or other input/output command devices, a user accesses and implement all digital telephone functions without the physical telephone set, the personal computer also providing the audio function. | 09-11-2008 |
| 20080219241 | Subscriber access authorization - A method for registering a session initiation protocol (SIP) client to an internet protocol multimedia subsystem (IMS), in which a SIP client having a given IP address, public identity and private identity sends a registration request to a session border controller (SBC) for registering the public identity to the IMS, the SBC responsively causes an authorization request to be sent to another network entity in the IMS, the authorization request indicating the IP address of the SIP client and a private identity, the another network entity obtaining from an LDAP/AAA server a reference address based on the private identity and deciding whether to allow the authorization of the public identity to the IMS based on the correspondence between the reference address and the IP address of the SIP client. | 09-11-2008 |
| 20080219242 | Method for Charging for a Communication Link Routed Via a Packet-Switched Communication Network - Disclosed is a method for charging for a communication link established from a first communication terminal (A) to a target communication terminal (B) via a packet-switched communication by transmitting message packets, the target communication terminal (B) featuring forwarding to at least one additional communication device (C). The inventive method comprises the following steps: —the first communication terminal (A) sends a signaling message ( | 09-11-2008 |
| 20080219243 | SYSTEMS AND METHODS FOR MONITORING QUALITY OF CUSTOMER SERVICE IN CUSTOMER/AGENT CALLS OVER A VOIP NETWORK - A system and method for monitoring call quality for calling centers using packet based call technology. A distributed system manages packet flow between a caller and a call center agent and storage servers. The distributed system is used to monitor, record and analyze real time communications between the caller and the agent and to identify whether certain predetermined parameters are occurring in any particular call. In the event that such a predetermined parameter does exist, a message can be sent to a supervisory station or dialog guidance messages may be sent to the agent. | 09-11-2008 |
| 20080219244 | Speech codec selection for improved voice quality in a packet based network - A method of improving voice quality in a packet based network. The method includes receiving an incoming call from a first endpoint and matching capabilities between the first endpoint and the second endpoint. The method also includes completing the incoming call if the capabilities match and tracking the packet loss associated with the network. The method also includes negotiating the voice quality based on the tracking and the capabilities. Also described is a devices and system for a similar method. | 09-11-2008 |
| 20080219245 | Automated method and system for selectively updating communications parameters representing subscriber services in telecommunications networks - Methods and systems for updating subscriber service parameters in a communications network wherein a central provisioning unit automatically locates subscriber ports having subscriber service parameters that require updating and automatically issues commands for updating the subscriber service parameters of the ports requiring updating. | 09-11-2008 |
| 20080225830 | CIRCUIT WITH GENERATING PHONE-CALL RING VIA COMPUTER SYSTEM AND INTERNET PHONE SYSTEM USING THE CIRCUIT - An internet phone system is implemented in a computer system, including an internet-phone software unit, for storing a software used by the computer system to operate as an internet phone. An audio-file storage software unit is for storing a plurality of audio files. A USB (Universal Serial Bus) audio interface software unit is coupled with the internet-phone software unit and the audio-file storage unit. A USB audio apparatus is coupled to the audio interface unit, and comprising at least a microphone and a speaker. Wherein, the USB audio interface unit stores a software for selecting one of the audio files to serve as the phone-call ring, and for driving the speaker in the USB audio apparatus by the computer system for generating ringing sound for an incoming call. | 09-18-2008 |
| 20080225831 | Methods, Apparatuses, and Computer Program Products for Processing Session Related Protocol Signaling Measures - An apparatus for processing session related protocol signaling messages includes a simplification element. The simplification element may be configured to receive a message associated with a session related protocol, to determine whether the message is a per call based message and, in response to a determination that the message is the per call based message, to interpret the per call based message without the removed per session based information. The per call based message is free of per session based information that has been removed. | 09-18-2008 |
| 20080225832 | MULTI-PROTOCOL TELECOMMUNICATIONS ROUTING OPTIMIZATION - A telecommunications switching system employing multi-protocol routing optimization which utilizes predetermined and measured parameters in accordance with a set of user priorities in determining the selection of a telecommunications path to be utilized for transmitting a data file to a remote destination. The switching system has a first memory for storing the data file to be transferred, a second memory for storing predetermined parameters such as cost data associated with each of the telecommunications paths, a third memory for storing a set of user priorities regarding the transmission of data files, and means for measuring the value of variable parameters such as file transfer speed associated with each of the telecommunications paths. Processor means are operatively associated with the second and third memories and the variable parameter measuring means for determining which of the plurality of telecommunications paths should be utilized for transferring the data file in accordance with the set of user priorities, the predetermined telecommunications path parameters, and the measured variable parameters. The switching system further comprises input means for allowing a user to change the user priorities in the third memory prior to transmitting a file. | 09-18-2008 |
| 20080225833 | Apparatus For Guaranteeing the Availability of Subscribers in Communication Networks Over Network Boundaries - According to prior art, for example, when danger or environmental catastrophe are imminent, the population is warned by information guided via the communication network. This type of warning is however limited when the communication network consists of a plurality of sub-networks and the (warning) information must thus be transmitted over network boundaries to the subscriber The invention solves this problem in that a warning profile is associated with each route/transit switching centre trunk group. The association is carried out during the establishment of the route/transit switching centre trunk group, and is updated as required as changes of the network configuration. Each network transition implicitly thus has one such warning profile. | 09-18-2008 |
| 20080225834 | Method for Supporting the Service Features "Call Hold", "Conference Calling" and "Three-Party Service" in Fmc Networks - The prior art cannot implement the feature call hold in a FMC network (fixed mobile conversion, i.e. mixed mobile fixed networks). This is due to the fact that the feature call hold as well as the features conference calling and three-party service have different definitions in the two standards ITU-T Q1912.5 and 3GPP TS 24.228 with which incompatible procedures are defined. Compatibility problems arise since all involved units in the FMC networks with different units such as clients and network transition units necessitate an inter-working of all interlinked units. This problem is resolved by providing a mapping functionality that converts the protocol elements of both standards into one another. | 09-18-2008 |
| 20080225835 | COMMUNICATION SERVER - The SIP server includes a SIP communication controller for controlling transmission/reception of a SIP message complying with SIP, a SIP call controller for controlling calls by using SIP, a call data manager for managing communication state information which indicates communication state with the opposing server and bypass requirement information which indicates the need of skipping the opposing server determined for each session communication on the basis of recovery state of the opposing server, a bypass controller for determining whether to skip the opposing server which is set to be the next destination for each session communication in relaying the SIP message on the basis of the communication state information and the bypass requirement information of the opposing server, and a SIP header controller for editing a Route header of the SIP message in accordance with the result of the determination by the bypass controller. | 09-18-2008 |
| 20080225836 | Method of Transmitting Messages - To provide a method of transmitting messages, using a transmission protocol having time-slots, that allows the messages to be transmitted in a flexible way, it is proposed that each message has a message identifier assigned to it, that the messages are placed in order in a queue, and that the queue has a set of slot identifiers assigned to it. | 09-18-2008 |
| 20080232350 | Computer-Telephony Integration - The invention provides a modification to the operation of the intelligent network instruction set whereby on transmitting an Initiate Call Attempt message between the SCP | 09-25-2008 |
| 20080232351 | IP communication system and IP telephone apparatus - During an incoming call, when a client apparatus of an incoming side notifies a client apparatus of an outgoing side of hold of the incoming call, the client apparatus notifies the client apparatus of the hold of the incoming call with OK message which is a response to an INVITE message. | 09-25-2008 |
| 20080232352 | Method and Apparatus for Distributing Application Server Addresses in an Ims - A method of distributing application server addresses in an IP Multimedia Subsystem. Upon allocation of an application server to a subscriber, the allocated or another application server sends at least one application server address to a Home Subscriber Server (HSS). The HSS stores the address(es) in association with the subscriber identity and sends the address(es) to a Serving Call/Session State Control Function (S-CSCF) allocated to the subscriber. The S-CSCF caches the address(es) in association with the subscriber identity and uses the address(es) to send subscriber-related messages to the allocated application server. | 09-25-2008 |
| 20080232353 | Method of transmitting data in a communication system - A method of receiving at a terminal a first signal transmitted via a communication network, said method comprising the steps of; receiving at the terminal the first signal comprising a plurality of data elements; analysing characteristics of the first signal; receiving from a user of the terminal a second signal to be transmitted from the terminal; analysing characteristics of the second signal to detect audio activity in the second signal; and applying a delay between receiving at the terminal and outputting from the terminal at least one of said plurality of data elements; and adjusting the delay based on the analysed characteristics of the first signal and on the detection of audio activity in the second signal. | 09-25-2008 |
| 20080232354 | IP TELEPHONE SYSTEM, IP EXCHANGE, IP TERMINAL, IP EXCHANGE BACKUP METHOD, AND LOGIN METHOD FOR IP TERMINAL - The present invention provides a technique suitable for the case where three or more IP exchanges are multiplexed each other. Priorities are respectively given to a plurality of IP exchanges | 09-25-2008 |
| 20080232355 | SESSION INITIATION PROTOCOL TRUNK GATEWAY APPARATUS - According to one embodiment, a session initiation protocol trunk gateway apparatus includes a register which registers each connection ID of the plurality of terminals registered in a registration table into a registration server on the session initiation protocol network in accordance with a prescribed registration period, a connector which connects among the plurality of terminals and the session initiation protocol network, a processor which divides the registration period into a plurality of distribution intervals in response to the number of the connection IDs and executes registration processing of the next second connection IDs by spacing of the distribution interval from a registration start of a first connection ID, and a controller which executes registration processing of the corresponding-connection IDs into the registration server in preference to the processor based on prescribed conditions in processing by the processor. | 09-25-2008 |
| 20080240079 | Communicating Processing Capabilities Along a Communications Path - The present invention provides a technique for determining which nodes are to provide various functions on traffic along a particular communication path. Generally, a communication path may include multiple nodes between which and through which traffic is routed. These nodes may include the communication terminals at either end of the communication path, as well as various types of routing nodes along the communication path. Each node will send to other nodes in the communication path information identifying the local functions it is capable of providing to the traffic carried in the communication path, and if available, remote functions capable of being provided to the traffic by other nodes in the communication path. Each node will receive from other nodes in the communication path information bearing on the remote functions. Each node will access criteria to determine whether any local functions should be applied to the traffic. | 10-02-2008 |
| 20080240080 | SYSTEM AND METHOD FOR MEDIA-LEVEL REDUNDANCY IN VOICE-OVER INTERNET PROTOCOL SYSTEMS - A system and method for providing media-level redundancy in voice-over Internet Protocol (VoIP) systems are disclosed. A central controller receives VoIP calls including a media transmission and call setup data and a standby allocation module included in the central controller transmits the VoIP calls to an active card and a standby card. The active card processes the media transmission of the VoIP call using an array of signal processing modules. The VoIP call setup data is also transmitted to a standby card which stores the call setup data and data identifying the active card processing the VoIP transmission in a profile database. When an active card malfunctions, the central controller transmits an activation signal to the standby card which then loads the contents of the profile database into an array of signal processing modules on the standby card to process the VoIP calls previously processed by the malfunctioning active card. | 10-02-2008 |
| 20080240081 | Method, system and apparatus for providing rules-based restriction of incoming calls - In a method, system and apparatus for providing rules-based restriction of incoming calls, a network entity such as a call manager receives a call request from a caller to setup a call to the client. The call manager includes a database including a client profile for the client, the client profile including identification data for the client and one or more client-defined conditions for accepting calls. The call manager also includes a processor coupled to the database that is configured to: query the database to obtain the client profile for the client in response to the call request received from the caller; determine if the call request from the caller satisfies the one or more client-defined conditions; and reject the call request if the call request is determined not to satisfy the one or more client-defined conditions. | 10-02-2008 |
| 20080240082 | SYSTEM AND METHOD FOR MANAGING INTEROPERABILITY OF INTERNET TELEPHONY NETWORKS AND LEGACY TELEPHONY NETWORKS - A system and method for providing interoperability between Internet telephony networks and legacy telephony networks includes conveying an address of an Internet telephony endpoint in a legacy telephony protocol. A globally unique Uniform Resource Identifier, referred to as a Universal Global Title, may be assigned as the address of the Internet telephony endpoint. The URI-based address of the Internet telephony endpoint can be conveyed to a legacy telephony network as an Internet Address Parameter, implemented as an extension to the ANSI ISDN User Part legacy telephony protocol. As such, a Universal Teletraffic EXchange may be provided where Internet telephony networks and legacy telephony networks can exchange addressing and signaling information while interoperating at a peer-to-peer level. | 10-02-2008 |
| 20080240083 | SYSTEM AND METHOD FOR MANAGING INTEROPERABILITY OF INTERNET TELEPHONY NETWORKS AND LEGACY TELEPHONY NETWORKS - A system and method for providing interoperability between Internet telephony networks and legacy telephony networks includes conveying an address of an Internet telephony endpoint in a legacy telephony protocol. A globally unique Uniform Resource Identifier, referred to as a Universal Global Title, may be assigned as the address of the Internet telephony endpoint. The URI-based address of the Internet telephony endpoint can be conveyed to a legacy telephony network as an Internet Address Parameter, implemented as an extension to the ANSI ISDN User Part legacy telephony protocol. As such, a Universal Teletraffic EXchange may be provided where Internet telephony networks and legacy telephony networks can exchange addressing and signaling information while interoperating at a peer-to-peer level. | 10-02-2008 |
| 20080240084 | COMMUNICATION SYSTEM, SUBSCRIBER MANAGEMENT SERVER AND COMMUNICATION SYSTEM CONTROL METHOD - There is provided a communication system or the like including an IP phone network, a circuit switched network, and a shared terminal capable of connecting any one of the IP phone network and the circuit switched network, wherein, when the position of the shared terminal is registered or when a call is made from the shared terminal, the subscriber management server of the IP phone network and the subscriber management server of the circuit switched network give and receive the subscriber information about the shared terminal to share the subscriber information, and a call control server of the IP phone network and a call control server of the circuit switched network mediate the call from the shared terminal with the use of the subscriber information. | 10-02-2008 |
| 20080240085 | SIP COMMUNICATION SYSTEM, CALL CONTROL SERVER AND CALL CONTROL METHOD - There is provided a SIP communication system or the like which, by performing SIP communication between a terminal and a call control server via an access network configured on a different network infrastructure, performs position registration or calling control of the terminal, wherein, when position registration or calling control of the terminal is performed, a SIP message including an access type identifying the communication method used by the terminal on the access network or access information identifying position information about the position of the terminal in the access network as control information is transmitted and received between the terminal and the call control server. | 10-02-2008 |
| 20080240086 | Signaling status information of an application service - There is provided a method of signaling status information of an application service. The method is performed by a signaling gateway which interconnects an internet protocol network and a signaling system | 10-02-2008 |
| 20080247381 | Provisioning of Redundant Sip Proxy Resources - A resolution of the address of an SIP proxy in an SIP network, with redundant SIP proxy resources, is provided. To establish a connection in an SIP network, an SIP client typically transmits a request to a DNS server system to obtain an address to gain access to SIP proxy resources. The SIP proxy resources are provided in the form of SIP proxy servers which are part of a peer-to-peer group. Messages are exchanged within the peer-to-peer group via a peer-to-peer protocol in order to communicate responsibilities for SIP domains or user-agent addresses. Adjustable responsibilities are defined within group. The address of the SIP proxy server responsible for the request of the SIP client is made available to the DNS server system so the DNS server system can forward the address to the SIP client so as to allow the SIP client to access the relevant Sip proxy server. | 10-09-2008 |
| 20080247382 | System and method for providing improved VoIP services - The invention provides a system and method for providing voice over Internet protocol (“VoIP”) services with an improved quality of service (“QoS”). In one embodiment, a number of gateways or proxies are connected with dedicated links to form a dedicated network. Each gateway is also on one or multiple communication networks, such as individual communication networks provided by different network providers. Each user of the system uses an ATA to connect to the dedicated network, which then connects the user to another ATA of another user, thereby establishing a connection between these two users. The connection between each of the ATAs and the dedicated network is made through communication network or networks. As the quality of a public network is generally not managed by a VoIP service provider, it is desirable to minimize the effects of public network or networks. A method is provided for selecting a gateway that provides the best connection possible to the dedicated network. The selection may be based, for example, network latency. The selected gateway then selects the best receiver gateway for routing the connection request and to connect to the receiver's ATA. | 10-09-2008 |
| 20080247383 | Voice over internet protocol phone - A VoIP phone includes a signal port, a comparison and distribution module, a first communication protocol module, a second communication protocol module, and a communication interface for a user to dial and talk. The comparison and distribution module is connected with the signal port for sorting signal outputted from the signal port into a first signal and a second signal by respective signal formats of the signals. The first communication protocol module is connected with the comparison and distribution module for processing the first signal and output a processed first signal, and the second communication protocol module is connected with the comparison and distribution module for processing the second signal and output a second processed signal. | 10-09-2008 |
| 20080247384 | Ims Call Routing Using tel-UrIs - The present invention proposes a specific handling of tel URIs in an IMS terminating network so as to enable routing of calls using telephone numbers (and not SIP URIs with embedded telephone numbers) as identifiers of the target users of those calls. Specifically, the present invention introduces a conversion module which is located within the IMS terminating network and is capable of converting SIP URIs with embedded telephone numbers into equivalent tel URIs which are then used by a terminating I-CSCF and S-CSCF to query the SLF and/or HSS so that they can route the calls to the target users. | 10-09-2008 |
| 20080247385 | Provision of Ims Services Via Circuit-Switched Access - The present invention proposes a solution for providing IMS services to users having circuit-switched controlled terminal being not adapted to provide IMS services to the users. In particular, it is proposed, in order to allow IMS to take the full call and service control, to place a user agent being responsible for the user ported to the IMS in a new node type called Mobile Access Gateway Control Function (MAGCF). This new node combines the logical functionality of a cellular switching center and the logical functionality of IMS. Further it is proposed to enhance a user's register, like HLR, to provide also information about the availability of a MAGCF node and about whether a particular user is enabled to use the MAGCF functionality. | 10-09-2008 |
| 20080253356 | METHOD AND SYSTEM FOR A POWER REDUCTION SCHEME FOR ETHERNET PHYS - Aspects of a method and system for a power reduction scheme for Ethernet PHYs are provided. An Ethernet PHY in a link partner may disable transmission via a transmit DAC integrated during an inactive connection, 10Base-T autonegotiation operation, and/or active 10Base-T connection with no data packet transmission. The DAC may be a voltage mode or current mode DAC. The PHY or a MAC device may determine when to disable transmission via the DAC. In this regard, the PHY or the MAC device may generate appropriate signals for disabling the transmission. The DAC may be enabled for transmission by the PHY or the MAC device when a connection becomes active or when an active 10Base-T connection is ready to transmit data. Moreover, the PHY may enable transmission via the DAC when operating in a forced 10Base-T mode of operation and the connection to the link partner is active. | 10-16-2008 |
| 20080253357 | COMPUTER SYSTEM WITH INTERNET PHONE FUNCTIONALITY - A computer system with internet phone functionality is provided. The computer system keeps running application programs when it turns the processing rate and the voltage of the central processor unit down (e.g. enter to the sleep mode). When the internet phone or a traditional telephone has an incoming call, the computer would be waked up immediately to prevent missing any incoming call from the internet phone or the traditional telephone. | 10-16-2008 |
| 20080253358 | Terminal-to-terminal communication connection control method using IP transfer network - Both a connection server and a relay connection server are installed in an IP transfer network; a function similar to a line connection control of a subscriber exchanger is applied to a connection server; a function similar to a line connection control of a relay exchanger is applied to the relay connection server; and a terminal-to-terminal communication connection control method with using the IP transfer network is realized in such a manner that a telephone set and a terminal such as an IP terminal and a video terminal transmit/receive an initial address message, an address completion message, a call pass message, a response message, a release message and a release completion message, which can be made in a 1-to-1 correspondence relationship with line connection control messages of the common line signal system. Furthermore, while an address administration table is set to a network node apparatus of an IP transfer network, means for registering addresses of the terminals into this address administration table is employed, so that an IP packet communication by a multicast manner can be realized with improving information security performance. | 10-16-2008 |
| 20080253359 | Terminal-to-terminal communication connection control method using IP transfer network - Both a connection server and a relay connection server are installed in an IP transfer network; a function similar to a line connection control of a subscriber exchanger is applied to a connection server; a function similar to a line connection control of a relay exchanger is applied to the relay connection server; and a terminal-to-terminal communication connection control method with using the IP transfer network is realized in such a manner that a telephone set and a terminal such as an IP terminal and a video terminal transmit/receive an initial address message, an address completion message, a call pass message, a response message, a release message and a release completion message, which can be made in a 1-to-1 correspondence relationship with line connection control messages of the common line signal system. Furthermore, while an address administration table is set to a network node apparatus of an IP transfer network, means for registering addresses of the terminals into this address administration table is employed, so that an IP packet communication by a multicast manner can be realized with improving information security performance. | 10-16-2008 |
| 20080253360 | TERMINAL APPARATUS AND COMPUTER PROGRAM - A terminal apparatus includes a communication unit connected to a private branch exchange and other terminal apparatus; a memory unit for storing address information for the other terminal apparatus; and a controller which, when receiving an incoming call command containing address information for the terminal apparatus as destination information from the private branch exchange via the communication unit, informs a user of the terminal apparatus of an incoming call addressed to the terminal apparatus, generates a substituted incoming call command in which the address information contained in the incoming call command is substituted with the address information for the other terminal apparatus stored in the memory unit, and transmits the substituted incoming call command to the other terminal apparatus via the communication unit so that the other terminal apparatus informs a user of the other terminal apparatus of the incoming call addressed to the terminal apparatus. | 10-16-2008 |
| 20080253361 | Receiving party based web-to-phone communication - A method for communicating in a communication system, said communication system comprising a network terminal of a network user, a server of a telecommunication company, and a telephone of a receiver, the method comprises the steps of: (A) the receiver registering an encoded object from the telecommunication company, the encoded object is associated with a telephone number; (B) the receiver distributing the encoded object for the network user to identify, (C) the network user clicking the encoded object; (D) in response to the network user's clicking, the server connecting the network terminal to the telephone which uses the telephone number for a communication connection; and (E) the telecommunication company collecting service fee from the receiver. The method can also be applied to Internet-based communication companies where the encoded object can be its customer membership for identification. | 10-16-2008 |
| 20080253362 | METHOD FOR PROVIDING LOCAL AND TOLL SERVICES WITH LNP, AND TOLL-FREE SERVICES TO A CALLING PARTY WHICH ORIGINATES THE CALL FROM AN IP LOCATION CONNECTED TO A SIP-ENABLED IP NETWORK - A method for providing combined local, toll, toll-free services, and number portability, to a calling party originating calls from an IP-based communication devices which are coupled to an IP-based multi-media service provider. The method includes receiving a SIP INVITE message, which includes a multi-media service identifier, at the multi-provider. The SIP INVITE message represents a call request for a multi-media service. The SIP INVITE message is processed at the multi-media service provider for determining the call request can be satisfied. If the multi-media service provider includes resources for satisfying the call, the multi-media service identifier of the SIP INVITE message is set to a first predetermined state and the call is processed. If the multi-media service provider does not include resources for satisfying the call, the multi-media service identifier of the SIP INVITE message is set to a second predetermined state and the call is processed elsewhere. | 10-16-2008 |
| 20080253363 | Systems and Methods to Facilitate Real Time Communications and Commerce via Answers to Questions - Methods and systems to facilitate real time communications and commerce via answers presented to questions. One embodiment includes one or more web servers to receive from a second user an answer to a question of a first user, and to present the answer with a communication reference of a connection provider to a third user; a session border controller of the connection provider to interface with a packet switched network; and one or more telecommunication servers of the connection provider coupled with the session border controller to connect the third user to the second user for real time communications in response to a request made via the communication referenced. One embodiment includes: receiving from a second user an answer to a question of a first user; and presenting to a third user the answer with a reference for requesting fee-based content from or real time communications with the second user. | 10-16-2008 |
| 20080253364 | INFORMATION DELIVERY SYSTEM AND INFORMATION DELIVERY METHOD USING THE SAME - A mobile type service provider terminal registers a service providing area to a presence management server before a service starts. The presence management server creates a status management table of the mobile type service provider terminal. When the presence of the mobile type service provider terminal is detected in an area, the terminal registers information contents, which are delivered to subscribers, to an information delivery server. When the information delivery server stores the registered information contents to an information contents database and completes the creation of an information providing table, it notifies a service control server of the information. The service control server creates a message with reference to the information notified from the information delivery server and delivers it to the subscribers. | 10-16-2008 |
| 20080253365 | Wireless networking communication system and communicating method for VOIP - A wireless networking communication system is disclosed. The wireless networking communication system includes a wireless communicating gateway connected to a personal computer, which in turn connects with a network, and a handheld communicating device in communication with a first communicating transceiver. The wireless communicating gateway includes the first communicating transceiver, and a first processor for determining type of operating system of the computer and detecting connection of the computer to the network. The handheld communicating device includes a second communicating transceiver in communication with the wireless communicating gateway, and a second processor for converting voice signals into digital signals and detecting connection of the wireless communicating gateway to the network. | 10-16-2008 |
| 20080259906 | TARGETED TELEVISION ADVERTISEMENTS BASED ON ONLINE BEHAVIOR - In a method for delivering targeted television advertisements based on online behavior, IP addresses indicating online access devices and IP addresses indicating television set-top boxes are electronically associated for a multitude of users. Using user profile information derived from online activity from one of the online access IP addresses, a television advertisement is selected, such as by using behavioral targeting or demographic information, and automatically directed to the set-top box indicated by the set-top IP address associated with that online access IP address. Preferably neither the user profile information nor the electronic association of online access and set-top box IP addresses includes personally identifiable information. | 10-23-2008 |
| 20080259907 | INTERWORKING BETWEEN H.320/H.324 AND SIP - Disclosed are a method, apparatus and system for interworking between H.320 or H.324 and SIP. The method comprises receiving a SIP message indicative of capabilities supported by a first endpoint device and deferring responding to the SIP message. The method further comprises receiving from a second endpoint device an H.320 or H.324 message indicative of the capabilities supported by the second endpoint device in response thereto responding to the SIP message thereby to establish a media communication channel between the first and second endpoint devices based on their respective capabilities. | 10-23-2008 |
| 20080259908 | Location object proxy - The function of determination of location is separated from the function of gathering information based on determined location by use of a Location Object (LO) proxy between an initiating VoIP capable device and a positioning center. The LOProxy queries an appropriate location database using a location key, and injects a PIDF-LO into a routing SIP message otherwise without location. A SIP request without location is received from a VoIP capable device. The SIP request contains messages indicating the type of location generator or service needed. A location key (like a telephone number or SIP URI), in addition to the type of location generator or service needed, is included in a SIP request. A location object (LO) broker may be used between a routing SIP message and a positioning center to direct a routing SIP message to an appropriate one of a plurality of location object (LO) proxies. | 10-23-2008 |
| 20080259909 | Signaling of Early Media Capabilities in IMS Terminals - Methods and apparatus are disclosed for processing Session Initiation Protocol (SIP) INVITE messages that include an early media capability indicator. The early media capability indicator provides information regarding the early media handling capabilities of the device originating the message, and is used by various network nodes that process the INVITE message to determine if, and under what circumstances, early media sessions may be established with the originating communication device. In an exemplary method, a SIP INVITE message is received from a SIP User Agent Client for a first communication device, the SIP INVITE message comprising an early media capability indicator. The exemplary method further comprises forwarding the SIP INVITE message to one or more remote communication devices, and selectively allowing one or more media sessions between the first communication device and the one or more remote communication devices, based on the early media capability indicator. | 10-23-2008 |
| 20080259910 | Dynamic Media Content For Collaborators With VOIP Support For Client Communications - Methods, systems, and computer program products are provided for delivering dynamic media content to collaborators. Embodiments include providing collaborative event media content, wherein the collaborative event media content includes a grammar and a structured document; selecting a VOIP protocol for communications between a client and a dynamic context generation server; generating a dynamic client context for a client by the dynamic context generation server in dependence upon communications from the client through the selected VOIP protocol; detecting an event in dependence upon the dynamic client context; identifying one or more collaborators in dependence upon the dynamic client context and the event; selecting from the structured document a classified structural element in dependence upon an event type and a collaborator classification; and transmitting the selected structural element to the collaborator. | 10-23-2008 |
| 20080267166 | Method and Apparatus for Providing a Multimedia Service - A method of providing a combinational communication service to a user of a communication network where a physical channel is defined for the user in respect of a circuit switched connection primarily for carrying real time data, the method comprising interrupting the transmission of real time data on said physical channel in order to allow non-real time data to be transmitted on the physical channel or to allow signalling, for setting up a packet switched bearer for carrying non-real time data, to be transmitted on the physical channel. | 10-30-2008 |
| 20080267167 | SYSTEM AND METHOD FOR SET UP OF AN IP COMMUNICATION TO THE ORIGIN OF A CIRCUIT SWITCHED CALL - A mobile device for operation within a service provider's wide area wireless network system comprises a wireless communication system, a circuit switched application, and an IP application. The circuit switched application receives circuit switched call signaling from the wireless service provider network. The circuit switched call signaling comprises an MSISDN identifying the origin of a circuit switched call. The IP application determines whether the origin of the circuit switched call is capable of internet protocol communications by querying a contact database (or receiving an alert from the service provider network) to determine whether the MSISDN identifying the origin of the circuit switched call is associated with a unique uniform resource identifier. If the origin of the circuit switched call is capable of internet protocol communications, initiating an internet protocol communication thereto. | 10-30-2008 |
| 20080267168 | Slow Adaptation of Modulation and Coding for Packet Transmission - Systems and methods for performing MCS adaptation are provided. In some cases, the network performs MCS adaptation based on received ACKs. | 10-30-2008 |
| 20080267169 | Method and system for remote diagnosis of a device over a communication network - A device in a voice over packet (VOP) network includes a transceiver operable to transmit and receive session initiation protocol (SIP) communications over at least a portion of a VOP network. Also included is a processor cooperatively operable with the transceiver. The processor is configured to facilitate, responsive to receipt of an invite request, determining if the invite request is a diagnostic invite. If the invite request is a diagnostic invite, then the audible alert and display of caller id are suppressed. The processor is configured to generate a final response to establish two-way media flow and a dialog. After the dialog is established, diagnostics is launched on the device, and responsive to receipt of diagnostic information requests and after two-way media flow is established, the device responds with diagnostic information responses. | 10-30-2008 |
| 20080267170 | System and method for presenting media to multiple parties in a SIP environment - In one embodiment, a network node is operable, responsive to receiving a SIP-based request from a calling party, for presenting media in a SIP network environment by establishing an inbound media session leg with the calling party and one or more outbound media session legs with a corresponding number of target parties. The network node includes functionality for patching the inbound and outbound media session legs to establish an end-to-end communications path respectively between the calling party and one or more target parties. | 10-30-2008 |
| 20080267171 | METHODS AND APPARATUS FOR OBTAINING VARIABLE CALL PARAMETERS SUITABLE FOR USE IN ORIGINATING A SIP CALL VIA A CIRCUIT-SWITCHED NETWORK FROM A USER EQUIPMENT DEVICE - Methods and apparatus for use in processing Session Initiation Protocol (SIP) calls in a network environment which includes a circuit-switched (CS) network and an Internet Protocol (IP) multimedia subsystem (IMS) network. In one illustrative technique, a SIP Register message is sent from a mobile communication device to the IMS network for registration of the mobile device. A SIP 200 OK message is received by the mobile device from the IMS network in response to sending the SIP Register message. The SIP 200 OK message has one or more variable call parameters or a network address at which to obtain the variable call parameters. The variable call parameters may include an E.164 number which may be dynamically assigned to the mobile device by the IMS network, and/or a time or timer value which defines a time period for which the E.164 number remains assigned to the mobile device. Other information may be included such as preferred access network/technology information. The variable call parameters are stored in memory of the mobile device and utilized for processing each one of a plurality of SIP calls involving the mobile device. After registration, the mobile device may initialize a timer with the timer value, run the timer and, when processing a SIP call, cause a CS call setup message which includes the E.164 number to be sent to the IMS network for routing of the call if the timer has not yet expired. If the timer has expired, the mobile device may refrain from utilizing the deassigned E.164 number in the CS call setup message and alternatively obtain and utilize a new E.164 number or an altogether different technique for processing of the SIP call. Alternative techniques for obtaining parameters and formatting the data are also described. | 10-30-2008 |
| 20080267173 | Method for the Improved Use of an Interface System with Address Components - The invention relates to a method for the improved use of an interface system (GW) for connections of subscribers (A-TIn, B-TIn) of at least two separate communications networks (KN | 10-30-2008 |
| 20080273523 | Providing Service Information For Charging A Subscriber For A Service - Providing service information includes receiving session initiation protocol (SIP) packets from a SIP proxy. Service information is extracted from the SIP packets. The service information describes a service provided to an access terminal associated with a subscriber. The service information is sent to a charging/enforcement point operable to charge the subscriber for the service. | 11-06-2008 |
| 20080273524 | SPLIT AND SEQUENTIAL PAGING FOR VOICE CALL CONTINUITY - Systems and methodologies are described that facilitate paging for establishing a Voice Call Continuity (VCC)-supported voice call in a network containing access point(s) that can support packet switched (PS) voice communication, such as Voice over Internet Protocol (VoIP), and access point(s) that can support only circuit switched (CS) voice communication. Paging signals as described herein are selectively delivered, such that a desired terminal receives a single PS paging signal if located at a VoIP-capable access point and a single CS paging signal otherwise. A split paging technique is described herein, wherein PS paging signals are delivered to VoIP-capable access points and CS paging signals are delivered to non-VoIP-capable access points substantially simultaneously. Additionally, a sequential paging technique is described herein, wherein PS paging signals are delivered to VoIP-capable access points and, if no response is received from a desired terminal, CS-domain paging is conducted. | 11-06-2008 |
| 20080273525 | COMMUNICATION METHOD AND COMMUNICATION DEVICE AS WELL AS COMPUTER PROGRAM - Communication performed within a network including a plurality of communication stations is provided, in which when an access control is performed so that communication timing of a packet can not collide with that of another station by detecting a signal which is transmitted from another station, “a header area processed not to become easily an error such as a physical layer header portion of a packet” which is transmitted from a communication station is made to have at least information which is required for extracting information in a payload of the packet and a field for controlling an access reservation of transmission of a packet which is generated as a result of transmission of another packet so that processing using the field can be performed. | 11-06-2008 |
| 20080273526 | Method and system of supporting abbreviated dialing between affiliated phones wherein at least one phone is associated with a non-affiliated network - A method and system of supporting abbreviated dialing between phones include determining dialing by a first phone of an abbreviated number associated with a second phone. The first phone is in a circuit-switched network associated with a Centrex system and the second phone is in an Internet Protocol (IP) packet-switched network disassociated from the Centrex system. The phones in the circuit-switched network are associated with the Centrex system. The Centrex system supports abbreviated dialing between phones associated with the Centrex system. A routing number for a phone call from the first phone to the second phone is determined as a function of the abbreviated number. The routing number includes IP addressing sufficient to support routing of the phone call through the packet-switched network to the second phone such as if both phones were associated with the Centrex system. | 11-06-2008 |
| 20080279176 | BASE STATION SYSTEM AND MOBILE STATION SUPPORTING DTMF PROTOCOL - A protocol is proposed for a mobile station to transmit Dual Tone Multi-Frequency (DTMF) to another mobile station. When a user presses a key on a source mobile station, which is connected to a target mobile station, DTMF data are transmitted via a digital channel to a base station first. Instead of converting the received DTMF data to a corresponding analog voice to the target mobile station, the base station forwards the DTMF data to the target station via a digital channel. The target mobile station parses the DTMF data and activates certain operation in response to the key pressed by the user. | 11-13-2008 |
| 20080279177 | Conjoined Telephony Communication System - There is provided a communication method for the receiving a first call-initiation request, generating a second call-initiation request in response to the first call-initiation request and generating a third call-initiation request in response to the first call-initiation request. Moreover, there is provided a communication method comprising receiving call-session information, separating the call-session information into core audio information and into supplementary information, routing core audio information on a first path and routing supplemental information on a second path. Further, there is provided a communication method, comprising receiving core audio information on a first communication path, receiving audio-enhancement information on a second communication path, uncompressing audio-enhancement information, combining core audio information with audio-enhancement information to generate combined audio information and providing combined audio information to audio terminals. | 11-13-2008 |
| 20080279178 | PORT REDUCTION FOR VOICE OVER INTERNET PROTOCOL ROUTER - An apparatus and method for increasing available ports on a voice router is provided. A first gateway and a second gateway are assigned a single port number for a data stream, the direction of packet flow is identified to determine a destination gateway. The destination gateway is one of the first and second gateways, depending on the direction of the packet flow. The packets are then forwarded to the destination gateway. The voice router can further consolidate RTCP streams from a plurality of gateways into a single port on the voice router. | 11-13-2008 |
| 20080285543 | METHODS AND APPARATUS TO MANAGE INTERNET PROTCOL (IP) MULTIMEDIA SUBSYSTEM (IMS) NETWORK CAPACITY - Methods and apparatus to manage Internet Protocol (IP) Multimedia Subsystem (IMS) network capacity are disclosed. An example method comprises identifying a terminating voice over Internet protocol (VoIP) call server associated with a called device, and returning a call rejection indicator when the terminating VoIP call server is in a first condition, the call rejection indicator returned without a call initiation request message sent to the terminating VoIP call server. | 11-20-2008 |
| 20080285544 | METHOD AND APPARATUS FOR PROVIDING MOBILITY FOR A VOICE OVER INTERNET PROTOCOL SERVICE - A method and an apparatus for providing mobility for a Voice over Internet Protocol Service (VoIP) provided on packet networks are disclosed. For example, the method receives a register request from a user endpoint device and retrieves an Access Point-address parameter from a contact header in the register request. The method then determines a physical location of an access point device in accordance with the Access Point-address parameter and updates location information for the user endpoint device in accordance with the physical location of the access point device. | 11-20-2008 |
| 20080285545 | VOICE OVER IP CUSTOMER PREMISES EQUIPMENT - A system and method for increasing the cost effectiveness of a service provider in meeting the needs of the multiple dwelling unit (MDU) market. To reduce the cost of connectivity between a network unit and a customer premises in the MDU, customer premises equipment functionality is embedded in a voice over Internet protocol (VOIP) phone. | 11-20-2008 |
| 20080285546 | SYSTEM AND METHOD FOR ENABLING OPERATION OF AN ETHERNET DEVICE OVER AN EXTENDED DISTANCE - A system and method for enabling operation of an Ethernet device over an extended distance. In a multiple dwelling unit (MDU) a customer premises equipment (CPE) can be coupled to a network unit via a broad reach Ethernet link that is greater than 100 meters (e.g., 500 meters). In this example, a CPE having a conventional Ethernet port can be operated over the broad reach Ethernet link using a converter device. | 11-20-2008 |
| 20080285547 | Voice over internet protocol gateway and a method for controlling the same - A voice over Internet protocol (VoIP) gateway includes a foreign exchange office (FXO), a foreign exchange station (FXS), and a VoIP processor. A controller of the VoIP gateway sets the VoIP gateway to either a TANDEM (trunk and ENM (ear and mouth)) mode or a standalone mode. In the TANDEM mode, the VoIP gateway transmits an incoming call from the VoIP processor to the FXO and an outgoing call from the FXS to the VoIP processor. In the standalone mode, the VoIP gateway transmits the incoming call from the VoIP processor to the FXS and the outgoing call from the FXS to the VoIP. | 11-20-2008 |
| 20080285548 | SYSTEM AND METHOD FOR PROCESSING A PLURALITY OF REQUESTS FOR A PLURALITY OF MULTI-MEDIA SERVICES - A system and method for processing a plurality of requests for a plurality of multi-media services received at a Private Service Exchange (PSX) from a plurality of IP-communication devices. The system includes an IP Segmentation Directory (IP-SD) coupled to the PSX and to a plurality of IP Service Control Points (IP-SCPs), which are operative to process the plurality of requests for the plurality of multi-media services. The PSX is adapted to receive, process and redirect the plurality of requests for the plurality of multi-media services to the IP-SD. The IP-SD further receives, processes and selectively redirects the plurality of requests for the plurality of multi-media services to a predetermined IP-SCP of the plurality of IP-SCPs based on attributes of each of the plurality of requests for the plurality of multi-media services. | 11-20-2008 |
| 20080291894 | METHODS AND APPARATUS TO COMMUNICATE USING A MULTI-FIDELITY AUDIO GATEWAY - Methods and apparatus to communicate using a multi-fidelity audio gateway are described. One example method receives information associated with at least one communication service via a multi-fidelity audio gateway and selects at least one communication path for use by the gateway device based on the communication service. | 11-27-2008 |
| 20080291895 | Integrated access device, voice over internet protocol system and backup method thereof - An integrated access device, a voice over Internet protocol (VOIP) system and a backup method thereof. The VOIP system includes a first remote server, a second remote server and an integrated access device. The integrated access device includes a first Internet protocol interface, a second Internet protocol interface, a memory unit and a processing unit. A first network connection is built between the first Internet protocol interface and the first remote server, and a second network connection is built between the second Internet protocol interface and the second remote server. The memory unit is for storing a program, and the processing unit executes the program to judge whether a voice packet in the integrated access device can be transmitted through the first network connection or not. When the voice packet cannot be transmitted through the first network connection, the voice packet will be timely transmitted through the second network connection. | 11-27-2008 |
| 20080291896 | Detection of communication states - A method of determining an overall presence state for a user of a communication system in which the user is connected to the communication system using a plurality of devices. The method includes each of the plurality of devices storing in a device memory a presence state for that device; detecting a change in the presence state in at least one of the plurality of devices; each of the plurality of devices transmitting a message via the communication system to the remainder of the plurality of devices, the message comprising the presence state; receiving the messages at the remainder of the plurality of devices; and executing a decision-making code sequence in a processor at each of the remainder of the plurality of devices to determine whether to synchronise the presence state of that device with the presence state from one of the messages based on the origin of an event causing the change in presence state at the at least one of the plurality of devices. | 11-27-2008 |
| 20080291897 | Access System for the Provisioning of Different Communications Sevices, and Method for Using Same - An access communication system is provided which comprises at least one aggregation device comprising at least one automated switching matrix for connecting a plurality of communication devices with a plurality of subscribers. When a new subscriber is to be connected through the at least one aggregation device, the at least one automated switching matrix is operative to enable the provisioning of a required service to the new subscriber either by using one of these communication devices or by communicating with a communication device installed at a different location and operative to enable the provisioning of the service required by the new subscriber. By a preferred embodiment the at least one aggregation device and automated switching matrix and the plurality of communication devices are managed by a single managing entity. | 11-27-2008 |
| 20080291898 | Provision of Telecommunication Services - An apparatus ( | 11-27-2008 |
| 20080291899 | METHOD AND SYSTEM FOR SENDING, ROUTING, AND RECEIVING INFORMATION USING CONCISE MESSAGES - A system and method are provided for communication between a communication device and a content provider associated with an internet domain name and a server. The system includes a network with a user interface, an internet connection, and an interface to the content provider's internet domain. A communication device user enters a concise message request which includes a channel, a designator and, optionally, a request instruction. The combination of the channel and the designator specify a location on the internet at which routing instructions reside for responding to the concise message request and generating a concise message response for output to the communication device. Concise message documents can be generated for effecting financial transactions such as purchases and payments via SMS. CMRL can also be used to route person-to-person messaging through a content provider's internet domain at which the users may be registered. | 11-27-2008 |
| 20080291900 | Delivering Unified User Experience By Automatically Teaming Up Information Appliances With General Purpose PC Through Internet - An embodiment of the present invention is a method for server side integration of communication devices and the general purpose PC of the same user through a computer network wherein no physical connection is required between the PC and the communication device. The user registers with PnC (phone and computer) server for subscribing to one or more PnC services such as drop-to-call, conference-call-dropping service, webpage sharing, caller kaleidoscope etc., via user interface of communications device and/or PC. Various features for subscribing and unsubscribing to services are provided along with authenticating the user using the name and the phone number of the user while registering with the server. | 11-27-2008 |
| 20080291901 | Network architecture for call processing - A network system for call processing, including customer premises equipment originating a call across a network, wherein the call includes private and public information; a session border controller directing the call information from the customer premises equipment to a public switched telephone network gateway, wherein the session border controller parses out the private information from the call information transmitted to the gateway; and one or more servers coupled to the session border controller, routing only the non-private call information to the public switched telephone network gateway. | 11-27-2008 |
| 20080291902 | MANAGING A BUFFER FOR MEDIA PROCESSING - A method and apparatus to perform buffer management for media processing are described. | 11-27-2008 |
| 20080291903 | SUBSCRIBER LINE CIRCUIT FOR COMMUNICATION SYSTEMS AND COMMUNICATION SYSTEM - In one aspect a communication system is provided wherein subscriber line circuits are connected to the communication network via a packet-based network. The subscriber line circuit comprises protocol interfaces for communicating with different network elements of the communication system and for the bi-directional conversion of information, which is transmitted from and to subscriber terminals on the subscriber side, into the information that is transmitted from and to the communication system on the network side. | 11-27-2008 |
| 20080291904 | Telecommunications System for Minimizing the Effect of White Noise Data Packets for the Generation of Required White Noise on Transmission Channel Utilization - Minimizing the effects of the requisite AGWN packets on transmission channel utilization without diminishing any of the aesthetic quality of the AGWN white noise on the voice or audio communication. A system for minimizing the effect of required generated background noise on said transmission channel utilization comprising the combination of an implementation for forming a transmission stream of sequential digital audio data packets, associating with each audio packet a data code representation of the payload data packet enabling the generation of said background noise and an implementation at a receiving station, responsive to each of said data representations for forming the represented payload data packet enabling said generation of background noise together with means at said receiving station for interspersing said formed payload packets enabling background noise generation between said associated audio data packets and background noise generating means, at said receiving station, responsive to said enabling payload packets for generating the background noise between the audio data packets. | 11-27-2008 |
| 20080298343 | VOIP PHONE NUMBER DISCOVERY ON PSTNS USING TWO WAY FXO COMMUNICATION - A method for placing a telephone call using a Voice over Internet Protocol (VoIP), the method including using a foreign exchange office (FXO) of a first VoIP system, making a public switching telephone network (PSTN) connection with an FXO of a second VoIP system; exchanging at least an Internet telephone address between the FXOs; terminating the PSTN connection; and placing the telephone call from the first VoIP system to the second VolP system over the Internet using the Internet telephone address. | 12-04-2008 |
| 20080298344 | Method and Apparatus to Facilitate Assessing and Using Security State Information Regarding a Wireless Communications Device - Upon determining ( | 12-04-2008 |
| 20080298345 | Cross-connect for emulated circuit-base communications - In one embodiment, a software-based cross-connect routes packets in an asynchronous Ethernet packet network to emulate circuit-based switching in a conventional circuit-oriented synchronous TDM network. The cross-connect has Ethernet interfaces for receiving and storing incoming Ethernet packets transmitted over the packet network from various source nodes. A cross-connect processor manipulates the stored Ethernet packets based on a stored connection table that defines emulated circuit-based connections in order to generate and store outgoing Ethernet packets in appropriate interfaces that then transmit the outgoing Ethernet packets via the packet network towards their defined destination nodes. | 12-04-2008 |
| 20080298346 | Apparatus and System for Controlling Signal Filtering - According to embodiments of the present invention, there is provided an apparatus and system for controlling signal filtering. According to some non-limiting embodiments, a selective filtering apparatus is provided. The selective filtering apparatus comprises an input interface connectable to a source of a composite signal within a first frequency range and a filtering device, coupled to the input interface. The filtering device comprises a filter and an output interface, the filter being operable to filter the composite signal and output an output signal within a second frequency range, the second frequency range being a subset of the first frequency range; the output interface being connectable to at least a portion of an in-premises telephone wiring. The selective filtering apparatus further comprises a triggering module being operable to cause the output interface to selectively output one of the output signal and the composite signal responsive to detection of a triggering event. | 12-04-2008 |
| 20080298347 | METHOD AND SYSTEM FOR PLAYING PACKETIZED ANNOUCEMENTS TO TELEPHONE NETWORK END OFFICE SWITCHING SYSTEMS FROM A CENTRALIZED DIGITAL RECORDED ANNOUNCEMENT UNIT - A computer readable medium stores a computer program that provisions an announcement to be played to a requester. The computer readable medium stores and include code segments. A storing code segment stores multiple announcements. A receiving code segment receives a request to send the announcement to be played to the requester to an announcement device, in response to a determination that the announcement is not stored locally at the announcement device. A sending code segment sends the announcement to the announcement device in response to the request. | 12-04-2008 |
| 20080298348 | System and method for providing audio cues in operation of a VoIP service - An exemplary VoIP service provides call participants cues to indicate that an enhanced service is being employed. When calling, the standard dial tone may be replaced with a distinctive dial tone or sound that indicates to the call participant that enhanced service is active (e.g., a service active sound). In some embodiments, the person called by the VoIP user hears a viral sound that indicates that an enhanced telephone service is being used. Furthermore, communication audio cues may be provided during the communication to provide further information to the call participants. | 12-04-2008 |
| 20080298349 | SYSTEM FOR TRANSMITTING HIGH QUALITY SPEECH SIGNALS ON A VOICE OVER INTERNET PROTOCOL NETWORK - The VoIP quality speech process is activated when a subscriber accesses a speech quality sensitive resource or in response to an activation of the feature by the subscriber, or when it is determined that the originating subscriber terminal device requires the transmission of high quality speech signals. A transmit buffer, associated with the port circuit that serves the originating device, stores a predetermined number of packets as they are transmitted from the originating device. In the case of lost or damaged packets, the VoIP quality speech system activates the transmit buffer to retransmit the missing or damaged packet to the destination device. Intelligent buffer management is provided, where the destination device can regulate the size of the transmit buffer as well as the size of its jitter buffer. | 12-04-2008 |
| 20080298350 | METHOD AND APPARATUS FOR AUTOMATED CALENDAR SELECTIONS - A method and apparatus for reserving an appointment in a communications network is described. In one embodiment, a request is received from a caller to schedule an appointment with an enterprise customer, wherein the request is processed in accordance with a media server in the communications network. A scheduling calendar is then accessed. Afterwards, the appointment is reserved with one of a plurality available appointment time slots. | 12-04-2008 |
| 20080304470 | METHOD AND SYSTEM FOR PROVIDING INTELLIGENT CALL REJECTION AND CALL ROLLOVER IN A DATA NETWORK - A system and method may include receiving an invite message associated with a calling device over a data network, the invite message requesting establishment of a voice over data communication session, and presenting a plurality of call rejection options, each of the plurality of call rejection options being associated with separate call rejection messages. The system and method may further include determining which one of the plurality of call rejection options is selected, where the plurality of call rejection options permit a called party to intelligently reject a voice over data communication session. | 12-11-2008 |
| 20080304471 | METHODS AND APPARATUS TO PERFORM CALL SCREENING IN A VOICE OVER INTERNET PROTOCOL (VOIP) NETWORK - Methods and apparatus to perform call screening in a voice over Internet protocol (VoIP) network are disclosed. An example method comprises sending a call initiation request to a VoIP endpoint, receiving a call screening indication from a user of the VoIP endpoint in response to the call initiation request and prior to a no answer determination made by an initiating call server, and playing a message for the user while a calling party is leaving the message. | 12-11-2008 |
| 20080304472 | Communication embodiments and low latency path selection in a multi-topology network - In one embodiment, a source device (e.g., a VOIP phone) establishes a call connection with a remote device depending on which of multiple network paths provides an acceptable latency (e.g., a lower latency). For example, in response to receiving a request to establish a connection with a remote destination device over a network, the source device (e.g., a caller's phone) obtains multiple service code values. The source device encodes each of multiple data packets to include a unique service code value for transmission of the messages over different network topologies to a remote destination. Thus, when transmitted, each of the multiple messages follows a different logical network topology of a network as specified by a respective service code value. Based on feedback from a remote device that receives the multiple messages, the source learns a preferred logical network topology of the network for establishing the call connection. | 12-11-2008 |
| 20080304473 | Enhanced terminal adapter - An enhanced terminal adapter (ETA) allows multiple communication devices connected to either similar or dissimilar networks, and typically accessible through different access numbers/addresses, to be used as extensions of each other when any of the devices are accessed via their access number/address. In the case of a PSTN call, communications devices connected to the ETA are considered POTS extensions. For a VoIP call, the communications devices connected to the ETA could be considered enhanced extensions in a VoIP network. In this way, the communications devices are each extensions of each other, depending on which device is accessed. | 12-11-2008 |
| 20080304474 | Techniques to Synchronize Packet Rate In Voice Over Packet Networks - Method and apparatus to synchronize packet rate for audio information are described. | 12-11-2008 |
| 20080310397 | Method and Device for Session Control in Hybrid Telecommunications Network - Combinational networks may provide simultaneous connectivity over networks of different type between terminals. Communication sessions on different network types such as Circuit switched and Packet switched, belonging to the same user equipment can be correlated. In case a communication session on a circuit switched network is halted by a supplementary service e.g. at an event such as acceptance of Call Hold, a communication session on a correlated packet switched network should be halted as well. A user equipment that detects the event sends a halt message to the circuit switched network and a message to the packet switched network or a session state manager node. The session state manager node either forwards the halt-message to the packet switched network, or sends a halt-message to the packet switched network when the packet switched network does not notify that a halt has occurred. | 12-18-2008 |
| 20080310398 | CALL PRIORITY BASED ON AUDIO STREAM ANALYSIS - A method and system of prioritizing calls based on audio stream analysis includes receiving a plurality of calls, wherein each call comprises an audio stream. The audio stream associated with one of the calls is analyzed for pre-determined audio characteristics. The call is processed based on the audio characteristics of the call. A system for prioritizing calls includes a multipoint control unit for receiving calls. An audio stream capture system captures an audio stream from the calls. The audio stream is analyzed by the capture system according to one or more selected criteria and an urgency priority ranking is determined for each call. The calls are ranked in a queue database according to urgency priority. A controller manages the audio stream capture system, the audio analyzer and queue database computer system. | 12-18-2008 |
| 20080310399 | Methods and systems for connecting phones to internet users - Methods and systems for connecting a phone to a client via a network by using an electronic notification system, which may include a third-party internet electronic messenger service, such as, Yahoo messenger™, Google Talk™, MSN messenger™ software, or email systems. The system includes a server that is connected to the network and receives a call from the phone and that sends a notification of the incoming call to the client via the electronic notification system like internet messenger service of choice of the client/receiver. The user of the client, receiver, launches an internet phone on the client in response to the notification by clicking on the called-context link. Then, the server connects the phone to the client via the internet phone whereby providing a communication between the receiver and a user of the phone, caller. | 12-18-2008 |
| 20080310400 | System and Method for Link Adaptation Overhead Reduction - Systems and methods of providing link adaptation information feedback are provided. A mobile device that receives packets generates link adaptation information based on incorrectly received packets. This can involve sending link adaptation information in association with NACKs (negative acknowledgements) generated by the mobile device. The network receives this link adaptation information and performs link adaptation accordingly. | 12-18-2008 |
| 20080310401 | Systems and Methods to Provide Communication References Based on Recommendations to Connect People for Real Time Communications - Methods and apparatuses to selectively present communication references based on recommendations from related entities to connect people for real time communications. One embodiment includes: receiving from a user a selection of a first listing, including a reference to be used to request a connection for real time communications between the user and a first entity; responsive to the selection of the first listing, determining one or more entities related to the first listing; selecting a second listing based at least in part on data representing one or more recommendations from the one or more entities; and presenting to the user the second listing, the including a reference for the user to request a connection with a second entity for real time communications. In one embodiment, the first and second entities provide services over connections established via the references included in the first and second listings for real time communications. | 12-18-2008 |
| 20080310402 | COMMUNICATION SYSTEMS AND QSIG COMMUNICATIONS METHODS - This invention relates to communication systems and QSIG communication methods. According to a first aspect, a communication system includes a control component; and a data network configured to communicate packets of information intermediate an originating location and a terminating location, the originating location being configured to receive a QSIG communication including a content portion and a signaling portion, wherein the data network is configured to communicate the signaling portion to the control component and the control component is configured to establish a connection within the data network intermediate the originating location and the terminating location responsive to the signaling portion, and wherein the data network is further configured to communicate the content portion of the communication within a plurality of packets intermediate the originating location and the terminating location using the connection. | 12-18-2008 |
| 20080310403 | Method for Switching Connections Between an IP-Only Phone and a Soft Phone to a Server - A method for switching connections between an IP-only phone and a soft phone to an IP gateway server is disclosed. An identical telephone number is allocated to the IP-only phone and the soft phone. When the soft phone has been moved such that a connection destination to a local-area network changes from a first connector to a second connector, a relay device is also changed from a first relay device to a second relay device. After recognizing its present position by a MAC address of the relay device to which it is currently connected, the soft phone issues a request to an IP gateway server to change an IP address to the soft phone when it is determined that the soft phone is located far away from the IP-only phone. When it is determined that the soft phone is located near the IP-only phone, the soft phone issues a request to the IP gateway server so that the IP address is changed to the IP-only phone. | 12-18-2008 |
| 20080316998 | Method, and Related Mobile Communications System, for Providing Combinational Network Services - In a mobile communication system including a circuit-switched (CS) mobile communications network, a packet-switched (PS) mobile communications network and an interworking function adapted to enable a signaling exchange between the CS and PS mobile communications network, a method of providing combinational CS+PS services to mobile users includes receiving, at a serving network entity in the PS mobile communications network, a user request issued from a first user on the PS mobile communications network, the user request relating to combinational services and having the serving network entity managing the received request, wherein the managing of the received request includes controlling an establishment of a session in the CS mobile communication network through the interworking function. | 12-25-2008 |
| 20080316999 | SYSTEM FOR DEPLOYING VOICE OVER INTERNET PROTOCOL SERVICES - A system for deploying Voice over Internet Protocol (VoIP) services is provided. A system that incorporates teachings of the present disclosure may include, for example, a Call Session Control Function (CSCF) having a controller element to receive a Session Initiation Protocol (SIP)message from an originating communication device requesting communications with a terminating communication device, and establish an Internet Protocol (IP) connection between the originating communication device and an advertisement media system to present at the originating communication device an advertisement message that replaces a ringback tone associated with the terminating communication device. Additional embodiments are disclosed. | 12-25-2008 |
| 20080317000 | METHODS AND APPARATUS TO PROVIDE A CALL-ASSOCIATED CONTENT SERVICE - Methods and apparatus to provide a call-associated content service to voice over Internet protocol (VoIP) devices are disclosed. An example method comprises receiving a message comprising a uniform resource identifier (URI) and a call dialog parameter at a content mediator, the call dialog parameter associated with a first communication session between a voice over Internet protocol (VoIP) endpoint and a destination, establishing a second communication session from the mediator to the destination based on the URI and the call dialog parameter, receiving content associated with the first communication session via the second communication session, and providing the content to the VoIP endpoint. | 12-25-2008 |
| 20080317001 | SYSTEM AND METHOD FOR DISTRIBUTED PROCESSING IN AN INTERNET PROTOCOL NETWORK - A system and method for distributed processing in an Internet Protocol network is provided. A system that incorporates teachings of the present disclosure may include, for example, an application server can have a controller element to receive a Session Initiation Protocol (SIP) INVITE message from a communication device, establish a Real Time Protocol (RTP) channel between the communication device and the application server responsive to the SIP INVITE message, and submit a SIP SUBSCRIBE message to an intermediate communication node (ICN) directing the ICN to engage one or more Digital Signal Processing (DSP) resources for processing signals in the RTP channel. Additional embodiments are disclosed. | 12-25-2008 |
| 20080317002 | Tamper-resistant communication layer for attack mitigation and reliable intrusion detection - A Tamper-Resistant Communication layer (TRC) adapted to mitigate ad hoc network attacks launched by malicious nodes is presented. One embodiment of the invention utilizes TRC, which is a lean communication layer placed between a network layer and the link layer of a network protocol stack. All aspects of the network protocol stack, with the exception of the routing protocol and data packet forwarding mechanism in the network layer, are unchanged. TRC takes charge of certain key functions of a routing protocol in order to minimize network attacks. Additionally, TRC implements highly accurate self-monitoring and reporting functionality that can be used by nodes in the network to detect compromised nodes. TRC of a node controls its ability to communicate with other nodes by providing non-repudiation of communications. The tamper-resistant nature of TRC provides high assurance that it cannot be bypassed or compromised. | 12-25-2008 |
| 20080317003 | Adaptive routing for packet-based calls using a circuit-based call routing infrastructure - A method in one example has: implementing an incoming voice call routing preference in a telecommunication system, routing bias settings for incoming packet calls and incoming circuit calls being set for any candidate trunk group lists as per desired call routing preferences; and selecting one of packet routing and a non-packet routing for call routing, the candidate trunk group lists for individual routing destinations being updated to indicate if packet voice technology is to be used for call delivery, wherein if packet voice technology is preferred, then a packet core access trunk group is added to a front of a respective trunk group list, and wherein if packet voice technology is to be used only if no circuit trunks are available, then packet core access trunk groups are added at an end of the list. | 12-25-2008 |
| 20080317004 | SIP ENDPOINT CONFIGURATION IN VoIP NETWORKS - VoIP networks and methods are disclosed for configuring SIP endpoints of VoIP networks. An application server of a VoIP network identifies an endpoint configuration for the SIP endpoints, and generates a configuration command based on the endpoint configuration. The application server formats a SIP message to include the configuration command, and transmits the SIP message to the SIP endpoints. Responsive to receiving the SIP message, the SIP endpoints process the SIP message to identify the configuration command, and set local configuration parameters based on the configuration command. | 12-25-2008 |
| 20080317005 | Computer Telephony System - A method and apparatus for securely registering an association between a computer terminal and a selected one of a plurality of communications terminals in a computer telephony system. An association is established according to a known technique between the identity of the selected communications terminal and the identity of the computer terminal to allow for control of the communications terminal by a user via the computer terminal. An abstract representation of the identity of the communications terminal is; generated and provided to a third party system accessible by the user. The user can then implement control of the selected communications terminal via the third party system whilst not prejudicing the security of the system. | 12-25-2008 |
| 20080317006 | METHOD FOR MANAGING A COMMUNICATION TERMINAL DEVICE, A COMMMUNICATION TERMINAL AND A COMMUNICATION SYSTEM - A method for managing communication terminal device includes sending a device management command which includes a control instruction for an activity state of a designated function, to a terminal; and performing, by the terminal, an operation on the activity state of the designated function according to the control instruction. The invention further provides a corresponding communication terminal and system. Using the present invention may avoid running the service client-end function in the communication terminal all the time, saving electric energy or terminal resources. | 12-25-2008 |
| 20080317007 | SYSTEM AND METHOD FOR SUPPORTING CONCURRENT COMMUNICATION OVER MULTIPLE ACCESS POINTS AND PHYSICAL MEDIA - A system and method for enabling communication concurrently over multiple access points and multiple physical media including but not limited to: cellular, network (e.g., Ethernet), broadband wireless, audio communication schemes. | 12-25-2008 |
| 20080317008 | Voice-over-IP Hybrid Digital Loop Carrier - Certain exemplary embodiments can comprise a method of use comprising: for a call between a local IP network and a remote non-IP network, converting between IP packets and PCM robbed bit signaling via a VoIP channelized router; providing the PCM robbed bit signaling to a TDM switch via the VoIP channelized router; and/or converting between IP packets and GR303 call reference values via the VoIP channelized router. | 12-25-2008 |
| 20090003312 | METHODS AND APPARATUS TO PROVIDE ENHANCED 911 (E911) SERVICES FOR NOMADIC USERS - Methods and apparatus provide enhanced 911 (E911) services for nomadic users are disclosed. An example method comprises receiving an Internet protocol (IP) address associated with a voice over internet protocol (VoIP) device and a media access control (MAC) address associated with the VoIP device, detecting when the VoIP device is outside an access network based on the MAC address and the IP address, prompting a user of the VoIP device to provide geographic location information for the VoIP device when the VoIP device is outside the access network, and updating enhanced 911 (E911) information for the VoIP device based on the geographic location information. | 01-01-2009 |
| 20090003313 | Activating a Tunnel upon Receiving a Control Packet - Packet switch operating methods and packet switches receive, at a packet switch, a control packet from another packet switch. The packet switch and the other packet switch are coupled together by two or more tunnels. The control packet indicates that a particular one of the tunnels is active on the other packet switch. In response, the packet switch operating methods and packet switches activate the particular tunnel indicated by the received control packet on the packet switch. | 01-01-2009 |
| 20090003314 | Systems and Methods For Verification of IP Device Location - This application discloses systems and methods for associating the geographic location of VoIP devices and monitoring and updating these locations such that emergency personnel can be directed to a caller's location based on the stored geographic-location information. | 01-01-2009 |
| 20090003315 | METHODS AND APPARATUS FOR DUAL-TONE MULTI-FREQUENCY SIGNAL CONVERSION WITHIN A MEDIA OVER INTERNET PROTOCOL NETWORK - In one embodiment, a method includes receiving a first internet protocol (IP) packet having information associated with a dual-tone multi-frequency (DTMF) signal. The information of the first IP packet is configured based on a protocol associated with a first layer of a media over internet protocol (MoIP) network. The first IP packet is associated with a destination endpoint. The method also includes producing a second IP packet having information associated with the DTMF signal. The information of the second IP packet is configured based on a protocol associated with a second layer of the MoIP network and a DTMF conversion policy associated with the destination endpoint. The second layer is different than the first layer. | 01-01-2009 |
| 20090003316 | METHOD AND SYSTEM FOR MANAGING ENTERPRISE-RELATED MOBILE CALLS - Methods, systems, and mobile devices for managing mobile calls to or from an enterprise-associated mobile device. The system and mobile device are configured to ensure all calls over a public land mobile network are routed through an enterprise communications system. The mobile device is prevented from directly calling remote parties through the public land mobile network and the public land mobile network forwards all calls addressed to the mobile device to the enterprise communications system. The enterprise communication system responds to a request to connect the mobile device and the remote party by establishing a first call with the mobile device, establishing a second call with the remote party, and bridging the two calls to connect the mobile device to the remote party. | 01-01-2009 |
| 20090003317 | METHOD AND MECHANISM FOR PORT REDIRECTS IN A NETWORK SWITCH - A method for selectively redirecting a data packet to a port on a switching device which is associated with a corresponding network service. In one embodiment, the data packet is redirected to an intrusion prevention service (IPS) for security analysis of the data packet. In another embodiment, the switching device performs a data link layer redirecting of the data packet based at least in part on whether the data packet is to be flooded from the switching device. | 01-01-2009 |
| 20090003318 | System and method for voice redundancy service - A system and method for providing voice redundancy service. A digital packet telephony service is monitored for continuity of the digital packet telephony service. Voice communication service is switched to a plain old telephone connection in response to determining the digital packet telephony service is unavailable. | 01-01-2009 |
| 20090003319 | Network interface apparatus - An intelligent network interface apparatus to provide always-on, always-connected processing for call signals is described. One embodiment of the apparatus includes logic to selectively handle incoming call signals even when a computer to which the apparatus is operably connected is unavailable (e.g., asleep). The apparatus may also include logic for selectively waking up a sleeping computer upon determining that incoming call signals indicate that a communication with the computer is desired. The incoming call signals may be associated with a voice over internet protocol (VoIP) communication. | 01-01-2009 |
| 20090003320 | System for seamless redundancy in IP communication network - In a seamless redundancy or failover system for an IP network, data intended for a master component is received at a seamless redundancy component, where the data is routed both to the master component and to a standby component. The standby component is configured to process the data in the same manner as the master component, e.g., the standby component may be a duplicate of the master component, or another component configured to perform the same data processing functions. For seamless redundancy/failover, the data output of the standby component is suppressed unless and until the master component enters a failure condition, at which time the data output of the standby component is enabled for transmission to a downstream network component. “Failure condition” refers to an operational state of the master component where the master component is unable to process received data in its intended and normal manner. | 01-01-2009 |
| 20090003321 | FACILITATING NON-SIP USERS CALLING SIP USERS - A technique for allowing a non-SIP user to call a SIP user includes dialing an established service number that indicates a desire to place a call to a SIP user. The SIP URI of the intended call recipient is included in a call setup protocol message associated with dialing the service number. A non-SIP network recognizes the call to the service number and the SIP URI from the UUI parameter of the call setup message. The call is then routed to a gateway for interfacing between the non-SIP network and the appropriate SIP network where the SIP URI is extracted from the message received by the gateway and used to generate an SIP INVITE message for establishing the call with the intended SIP user. | 01-01-2009 |
| 20090003322 | IP DEVICE EXCHANGE APPARATUS AND CALL CONNECTION CHANGING METHOD - An IP device exchange apparatus includes: a connector that is connected to a first IP phone, a second IP phone, and a third IP phone; a memory for storing a coding scheme obtained by call setting which is negotiated between the first and second IP phones; and a controller that, when receiving a call instruction for call connection to the second IP phone from the third IP phone during communication between the first and second IP phones, employs the coding scheme stored in the memory to perform, between the third IP phone and the IP device exchange apparatus, call setting between the second and third IP phones, while maintaining call connection between the first and second IP phones, thereby changing the call connection between the first and second IP phones to call connection between the second and third IP phones. | 01-01-2009 |
| 20090003323 | IP TELEPHONE SYSTEM AND IP TELEPHONE TERMINAL USED THEREIN - An IP telephone system includes a first IP telephone terminal and a second IP telephone terminal. IP telephone communications are established between the first and second IP telephone terminals via Internet when the second IP telephone terminal has acquired identification data identifying the first IP telephone terminal. The second IP telephone terminal further acquires terminal data identifying a function that the first IP telephone terminal can control via Internet. The first IP telephone terminal receives from the second IP telephone terminal data instructing to execute the function that the first IP telephone terminal can control. Then, the first IP telephone terminal controls execution of the function identified by the data received from the second IP telephone terminal. | 01-01-2009 |
| 20090010246 | Cordless telephone systems - A multimode home telephone system includes a computer, a base unit, a wireless handset and a wireless headset. The computer is programmed with a “soft phone” program to make and receive VOIP calls via the Internet and couple them to and from the base through a USB connection. The base is also coupled to a public switched telephone network (PSTN) and is operable to effect full duplex communication via both the PSTN and the Internet. The base, in turn, is wirelessly coupled to the handset via the DECT/UPCS protocol, and the handset is wirelessly coupled to the headset via the Bluetooth protocol, such that the user can selectively place and receive telephone calls via any one of the Internet, the PSTN or an optional Bluetooth enabled cell phone. Optionally, a DECT/UPCS enabled headset can communicate directly with the base in addition to or in lieu of the Bluetooth enabled headset. | 01-08-2009 |
| 20090010247 | Method and Arrangement for Enabling a Multimedia Communication Session - A method and arrangement for enabling multimedia during an ongoing circuit-switched call between a first mobile terminal and a second terminal, wherein the first terminal uses a first access having constraints by not admitting simultaneous packet-switched and circuit-switched communication. A change of connection is detected from the first access to a second access having no such constraints by admitting simultaneous packet-switched and circuit-switched communication. A capability query is then sent to the second terminal in response to said detection. When the requested capabilities are received from the second terminal, possible multimedia applications and/or services are indicated to the user according to the received capabilities. | 01-08-2009 |
| 20090010248 | Data Communication System and Data Communication Method - An IP terminal | 01-08-2009 |
| 20090010249 | METHOD OF DISTRIBUTING GEO-LOCALISATION INFORMATION - The invention concerns a method of distributing geo-localisation information associated with an endpoint device ( | 01-08-2009 |
| 20090010250 | REVERSE ENUM BASED ROUTING FOR COMMUNICATION NETWORKS - A network and method of routing a call between communication networks includes a first step of establishing a reverse ENUM DNS server containing a table of NAPTR records that associate E.164 telephone numbers with identifiers. A next step includes routing a call from an originating PSTN system to a first gateway. A next step includes sending an ENUM query containing an E.164 telephone number to an ENUM DNS server, which returns an identifier associated with the E.164 telephone number. A next step includes routing the call to a second gateway. A next step includes launching a reverse ENUM query containing the identifier to the reverse ENUM DNS server, which looks up an E.164 telephone number associated with the identifier, and returns it to the second gateway. A next step includes routing the call from the second gateway to the returned E.164 telephone number in the terminating PSTN system | 01-08-2009 |
| 20090010251 | Simplifying DSL Deployment via Analog/DSL Combination Solution - A method using a combination analog/DSL modem for deploying DSL services is disclosed. A combination analog/DSL modem is utilized at the subscriber premises. A telephone line is tested using the analog portion of the modem. In combination with information provided by the subscriber and records, suitability of the service line for DSL services may be accurately determined. DSL service is then ordered by the subscriber. Preferably, DSL services are deployed on top of the existing analog voice service line allowing service turn-on within a short period of time. The subscriber can have the ability to access a network using the DSL portion of the combination modem. If during modem testing, it is determined that the telephone line would not support DSL service, the subscriber would be informed that DSL service is currently not available for them. However, the subscriber could continue to use the analog portion of the combination modem. | 01-08-2009 |
| 20090016323 | System, Method, and Apparatus for Maintaining Call State Information for Real-Time Call Sessions - A method for facilitating communication sessions includes establishing a communication session between a first endpoint and a second endpoint, sending a hibernation message from the first endpoint, and receiving the hibernation message by the second endpoint. The method further includes storing, by the first and second endpoint, session state information associated with the communication session in response to receiving the hibernation message, and deactivating at least a portion of the communication session. After storing the session state information by the first and second endpoints, the method further includes retrieving the session state information by the first and second endpoints, and reestablishing the deactivated portion of the communication session. | 01-15-2009 |
| 20090016324 | Method and Gateway for Connecting IP Communication Entities via a Residential Gateway - The invention concerns a method for connection to IP communication entities (E | 01-15-2009 |
| 20090016325 | Multimode Customer Premises Gateway Providing Access to Internet Protocol Multimedia Subsystem (IMS) Services and Non-IMS Service - A multimode customer premises gateway supports multiple different types of telecommunications signaling protocols to allow different types of user devices to connect to the gateway and access both Internet Protocol multimedia subsystem (IMS), and non-IMS services. The gateway includes a connection manager configured to provide a non-IMS user device connected to the gateway with access to an IMS service using an IP multimedia services identity module (ISIM) for the gateway. A protocol converter translates information between an IMS protocol and a non-IMS protocol to enable the connection manager to provide the non-IMS device with access to the IMS service. The connection manager is also configured to provide an IMS user device with access to the IMS service and to provide a non-IMS user device with access to a non-IMS service. | 01-15-2009 |
| 20090016326 | MANAGED PRIVATE NETWORK SYSTEM - A managed private network (“MPN”) system for interconnecting enterprise entities to subscriber entities. The MPN system uses the ATM protocol and segregates data for an enterprise on to virtual connections dedicated to the enterprise. Each enterprise may have a single connection to the MPN system. The MPN system may forward data to various service providers through which subscriber entities may be connected to the MPN system. Thus, the enterprise entities need not have a separate physical connection to each service provider. Also, the MPN system can offer services (e.g., archival storage) to the enterprise entities. The MPN system ensures that data for one enterprise will not be intermingled with the data of another enterprise. | 01-15-2009 |
| 20090016327 | Technique for Communicating Information Over a Broadband Communications Network - A system and method for enabling communications to be transmitted between at least two associated counterpart devices at different locations, e.g., at work and home. One or more communications devices having a first identification code applicable to a first communications network are associated with one or more counterpart devices having a second identification code applicable to a second communications network. | 01-15-2009 |
| 20090016328 | METHOD FOR SCHEDULING VoIP TRAFFIC FLOWS - The present invention relates to a method for scheduling a data transmission to a user equipment in a communication system comprising at least one radio network controller (RNC) governing a number of base stations, wherein the communication system supports data transmission from a base station to an user equipment on a high speed packet access (HSPA) bearer or a dedicated channel (DHC) or on similar bearers in a CDMA2000 system. The method comprises the steps of: identifying at least one predetermined scheduling condition for data transmissions for the user equipment; determining at least one current scheduling conditions of the user equipment; comparing the predetermined scheduling conditions of the user equipment with the current conditions of the user equipment; selecting a bearer for the data transmissions during a session from a base station based on the comparison; and using the selected data bearer during the data transmission session or until a new data bearer has been selected. Furthermore, the invention relates to a user equipment, a radio network controller, a computer readable medium and mobile communication system for data transmissions such as VoIP service transmissions in wireless communications systems | 01-15-2009 |
| 20090016329 | Managing a System Between a Telecommunications System and a Server - A call is managed between a network and a telecommunications system ( | 01-15-2009 |
| 20090016330 | PROVISION OF PACKET-BASED SERVICES VIA CIRCUIT-SWITCHED ACCESS - The present invention proposes a solution for providing IMS services and in particular mid-call services to users having circuit-switched controlled terminals and being not adapted to provide IMS services to the users. In particular, it is proposed to introduce a new node type called Mobile Access Gateway Control Function (MAGCF). This new node combines the logical functionality of a cellular switching center and the logical functionality of packet-based logic. The invention discusses a concept of the MAGCF handling mid-calls, which comprises identification of the received mid-call request, generating in accordance to the identified mid-call a corresponding message, tracking the status of the performed mid-calls. | 01-15-2009 |
| 20090022140 | SYSTEMS, METHODS AND COMPUTER PRODUCTS FOR VOICEMAIL VIA INTERNET PROTOCOL TELEVISION - Systems, methods and computer products for voicemail via Internet Protocol Television. Exemplary embodiments include a method for providing voicemail to an Internet-Protocol-enabled device, the method including receiving a communication that a voicemail to a called party has been deposited in a voicemail infrastructure, mapping the called party number to an Internet Protocol-enabled device address of the called party, and sending the voicemail to the Internet Protocol-enabled device address corresponding to the called party number. | 01-22-2009 |
| 20090022141 | SYSTEMS, METHODS AND COMPUTER PRODUCTS FOR PLACING TELEPHONE CALLS VIA INTERNET PROTOCOL TELEVISION CALL LOGS - Systems, methods and computer products for placing phone calls via Internet Protocol Television call logs. Exemplary embodiments include a method for generating communication requests via an Internet-Protocol-enabled device, the method including receiving a request to initiate a communication request from an Internet-Protocol-enabled device having an Internet-Protocol-enabled device address, mapping the Internet Protocol-enabled device address of a calling party to a calling party number and sending a first communication request to a calling party communication device associated with the Internet Protocol-enabled device. | 01-22-2009 |
| 20090022142 | SYSTEMS, METHODS AND COMPUTER PRODUCTS FOR LOGGING OF OUTGOING CALLS TO AN INTERNET PROTOCOL TELEVISION CALL LOG - Systems, methods and computer products for the logging of outgoing calls to an Internet Protocol Television call log. Exemplary embodiments include a method for logging outgoing communication requests related to an Internet-Protocol-enabled device, the method including receiving a request to initiate a communication request from an Internet Protocol-enabled device having an Internet-Protocol-enabled device address, mapping the Internet Protocol-enabled device address of a calling party to a calling party number, retrieving caller identification information associated with the called party and recording the caller identification information in a log associated with the Internet Protocol-enabled device. | 01-22-2009 |
| 20090022143 | SYSTEMS, METHODS AND COMPUTER PRODUCTS FOR LOGGING OF INCOMING CALLS TO AN INTERNET PROTOCOL TELEVISION CALL LOG - Systems, methods and computer products for the logging of incoming calls to an Interact Protocol Television call log. Exemplary embodiments include a method for logging incoming communication requests related to an Internet-Protocol-enabled device, the method including receiving a communication request from a caller device over a voice network, the communication request including a caller party number and name of the caller device and a called party number and name of a called device associated with the communication request, mapping the called party number and name to an Internet Protocol-enabled device address of a called party, sending the caller party number to the Internet Protocol-enabled device address corresponding to the called party number and recording the caller identification information in a log associated with the Internet Protocol-enabled device. | 01-22-2009 |
| 20090022144 | IP Telephony Service Interoperability - The invention concerns a residential gateway device designed for a decentralized client equipment and comprising converting means for providing interoperability between to separate IP telephony services. | 01-22-2009 |
| 20090022145 | SYSTEMS, METHODS, APPARATUS AND COMPUTER PROGRAM PRODUCTS FOR NETWORKING TRADING TURRET SYSTEMS USING SIP - Systems, methods, apparatus and computer program products are provided for sharing a resource including a subscription engine configured to subscribe to a first turret system to share the resource, a state change engine configured to receive a state change notification corresponding to the resource, from the turret system, and a failover engine configured to invite the turret system to initiate a connection to the resource. | 01-22-2009 |
| 20090022146 | METHOD AND SYSTEM OF SCREENING AND CONTROL OF TELEPHONE CALLS WHILE USING A PACKET-SWITCHED DATA NETWORK - A Call Alerting and Control System is provided in a communication environment to allow an Internet user (“user”) approximately real-time monitoring of information about an incoming call from a calling party while maintaining a connection with the Internet. The monitored information can include the calling party's name and telephone number. The system could also allow the user to provide an answering machine-type message to the calling party and the user to listen to the calling party's response to the message while still connected to the Internet. The system can further allow the user to reroute, answer or otherwise treat the incoming call while, at the user's discretion, either maintaining or disconnecting a connection to the Internet. | 01-22-2009 |
| 20090022147 | TELEPHONY COMMUNICATION VIA VARIED REDUNDANT NETWORKS - A switched telephone network is arranged in a manner to enable packet voice communication between telephone terminals via multiple redundant packet switched networks. The packet switched networks may utilize different protocols, be operated by different entities, and have primary functions other than voice communication. One example of such a network may be internetworked networks, such as the Internet. One example of an alternate packet switched network may be a network whose primary function is control of a circuit switched telephone network. The common channel interoffice switching system (CCIS) of a public switched telephone network (PSTN) is a preferred example. | 01-22-2009 |
| 20090028130 | CALL IDENTIFICATION MECHANISM FOR MULTI-PROTOCOL TELEPHONES - In one embodiment, a system identifies an Internet Protocol (IP) device as a calling party for calls from either a Voice over Internet Protocol (VoIP) portion or a cellular portion of a multi-protocol phone. As a result, return calls to the multi-protocol phone are always sent through an IP device to allow call handling or Single Number Reach (SNR) functionality for the return calls. | 01-29-2009 |
| 20090028131 | TEST AUTOMATION FOR AN INTEGRATED TELEPHONY CALL MANAGEMENT SERVICE - A test application includes instructions for configuring a digital cross connect. A first modem and a second modem are each connected to at least one telecommunications network and a digital cross connect. An integrated telephony call management service (ITCMS) client includes computer-executable instructions stored on a computer-readable medium included in the test computer. | 01-29-2009 |
| 20090028132 | System and method for transferring interaction metadata messages over communication services - System and method for transmitting interaction metadata messages, for example, computer telephony integration (CTI) messages, from one or more network end points and/ox from a central network device to a recording system using a light-weight interaction metadata protocol, for example, a light-weight CTI protocol, over one or more communication services. | 01-29-2009 |
| 20090028133 | Method for providing hysteresis to fluctuating signaling link - A signaling node within a telecommunications network automatically detects that a signaling link is fluctuating in and out of service and provides hysteresis to the fluctuating signaling link. The signaling node includes a link controller that monitors the state(s) of the signaling link over a time period and a link blocking module that blocks the signaling link from carrying SS7 traffic when the signaling link fluctuates between a failed state and a stable state over the time period. | 01-29-2009 |
| 20090028134 | Management and Control of Call Center and Office Telephony Assets - Call center and office telephony assets, including telephones, headsets, on-line indicators (OLI), and handset lifters, are managed and controlled over a network by a remote computer system. Each asset has associated therewith one or more network addresses, in some cases the network addresses mapped from an electronic identifier stored within the particular asset or determined by a proxy. In one embodiment, an asset's network address is mapped from the asset's unique media access control (MAC) address. The computer system communicates with the assets over the network to manage and control the assets. | 01-29-2009 |
| 20090028135 | SYSTEM AND METHOD FOR UNIFIED COMMUNICATIONS THREAT MANAGEMENT (UCTM) FOR CONVERGED VOICE, VIDEO AND MULTI-MEDIA OVER IP FLOWS - A method and system for unified communications threat management (UCTM) for converged voice and video over IP is disclosed. A computer-implemented method for threat management receives an incoming packet. The incoming packet is broken into sub-packets and fed to a plurality of packet processing engines. Each packet processing engine inspects the sub-packets and annotate the sub-packets with meta-data. The annotated sub-packets are combined and processed by a plurality of application engine to generate a processed packet. The processed packet is classified and stored in a database. | 01-29-2009 |
| 20090028136 | Method And Apparatus For Controlling Preset Events - A method and apparatus for controlling preset events. The method includes: setting a preset event for a set object on a media gateway controller and a media gateway respectively; judging whether the preset event needs to be monitored continuously on the set object; if so, holding the preset event to be in the active state on the set object; otherwise, no longer holding the preset event to be in the active state on the set object. The apparatus includes a judging module, a controlling module and a canceling module. | 01-29-2009 |
| 20090028137 | METHOD AND APPARATUS FOR STORING AND ACTIVATING UNIVERSAL RESOURCE LOCATORS AND PHONE NUMBERS - A method and apparatus for enabling subscribers to store telephone numbers and/or URLs embedded in streaming video contents associated with a video session being shown on video display devices into address books hosted in the network. Subscribers can then access these phone numbers to place phone calls using information stored in the network address books by activating a voice session. Similarly, subscribers can also access URLs to browse websites using URLs stored in the network address books by activation a web session. | 01-29-2009 |
| 20090034509 | METHOD AND SYSTEM FOR REDUCING UPSTREAM NOISE IN A NETWORK USING AN ACTIVE MULTIPLEXER - An active multiplexer contains a switching mechanism that connects one of a plurality of upstream links from corresponding nodes in a communication network to an upstream output based on information contained in a MAP. The MAP contains scheduling information of the next user stations, which are coupled to the nodes, and which are scheduled to transmit in the upstream direction during a period following the current time. Station identifiers are associated with their corresponding node identifier during a ranging burst interval into a station/node table, which is used in conjunction with the MAP to control the active multiplexer. Based on the MAP, the active multiplexer connects the node, as determined from the station/node table, that serves the station that is scheduled to transmit upstream traffic and disconnects other nodes. | 02-05-2009 |
| 20090034510 | Method and apparatus for securely transmitting lawfully intercepted VOIP data - A method, apparatus, and computer usable program product for transmitting intercepted VOIP data are provided in the illustrative embodiments. A VOIP call is intercepted in response to a lawful request for intercept by a law enforcement agency. VOIP data associated with the intercepted VOIP call is encrypted. The encryption may use a virtual private network an encryption using a key of a specific length, bit stuffing, or other encryption methods. The encrypted VOIP data is transmitted to the law enforcement agency using a public data network either during the VOIP call or after the VOIP call. The intercept request may be made during the VOIP call, or before the VOIP call. Furthermore, the VOIP data of the VOIP call may be stored before transmitting to the law enforcement agency, and archived based on archiving rules. The request for the intercept may be queued for processing according to queuing rules. Notifications based on the request for intercept, VOIP call characteristics, or characteristics of the VOIP data may be sent to one or more law enforcement agencies, and may also be encrypted. | 02-05-2009 |
| 20090034511 | TECHNIQUE FOR INTERCONNECTING CIRCUIT-SWITCHED AND PACKET-SWITCHED DOMAINS - A technique for interconnecting circuit-switched (CS) and packet-switched (PS) domains enables network components ( | 02-05-2009 |
| 20090034512 | METHODS, SYSTEMS, AND COMPUTER READABLE MEDIA FOR MANAGING THE FLOW OF SIGNALING TRAFFIC ENTERING A SIGNALING SYSTEM 7 (SS7) BASED NETWORK - Methods, systems, and computer readable media for managing the flow of signaling traffic entering a signaling system 7 (SS7) based network having a plurality of gateways for connecting the SS7 network to a non-SS7 network are disclosed. According to one aspect, a method for managing the flow of signaling traffic entering the SS7 based network includes generating, at a signaling node within the SS7 network, a route management message including information for identifying one of the plurality of gateways as the preferred gateway for traffic into the SS7 network. The message is sent to a node in the non-SS7 network for directing traffic into the SS7 network via the identified gateway. | 02-05-2009 |
| 20090034513 | INTERNET BASED TELEPHONE LINE - A telephone service method that provides subscribers with the functionality of an extra telephone line during data/Internet sessions. Each subscriber has a unique telephone number Dns that can be dialed by anyone with access to the PSTN. When the Dns is dialed the call will be routed via the PSTN to the ILTD server. The ILTD server upon receiving the call attempt from the Dnc will analyze the dialed number (Dns) and determine if the subscriber's computer is able to receive the telephone call. If the subscriber's computer is actively engaged in an Internet Protocol session, with the ILTD client software running, the ILTD server will connect the call over the Internet to the ILTD client software. The ILTD client software will activate the subscriber's sound card and the microphone to play audio and receive input from the microphone to allow the subscriber and the calling party to have a full duplex telephone conversation (i.e. using voice-over-IP technology). | 02-05-2009 |
| 20090034514 | Integrated Mobile Computing and Telephony Device and Services - Disclosed is an integrated handheld computer and telephony system. Integration of the handheld computer and telephony system is at the physical and operational level. For example, the integrated handheld computer and telephony system physically integrates a handheld computer with a mobile (e.g., cellular) telephone. In addition, the handheld computer is distinct from telephony system in that they are logically separable. However, they are also operationally integrated, for example, the telephony system executes a telephone application on the processor of the handheld computer. Likewise, the handheld computer can execute applications, for example, a phone book, that can be used to launch the telephony application. | 02-05-2009 |
| 20090034515 | Call Setup Request Confirmation - At least one exemplary embodiment of the present invention includes a method comprising receiving a call setup request, and automatically providing an indication that the call setup request is being processed. At least one exemplary embodiment of the present invention includes a method comprising providing a call setup request to a network, and receiving an indication that the call setup request is being processed. It is emphasized that this abstract is provided to comply with the rules requiring an abstract that will allow a searcher or other reader to quickly ascertain the subject matter of the technical disclosure. This abstract is submitted with the understanding that it will not be used to interpret or limit the scope. | 02-05-2009 |
| 20090041004 | Inline power system and method for network communications - An adapter and method for coupling an inline powered communications device to a primary network and to a secondary network, the communications device configured for having an assigned device identification and configurable for using an assigned network address for use in routing data over at least one of the networks. The adapter and method comprise a first port for connecting to the communications device to facilitate the communication of the data and inline power between the adapter and the communications device, the inline power for use in operating the communications device. The adapter and method comprise a second port for connecting to the primary network to facilitate the communication of the data between the primary network and the communications device via the first port, the second port coupled to the first port, and a third port for connecting to the secondary network to facilitate the supply of the inline power between the secondary network and the communications device via the first port, the second port coupled to the first port. The adapter and method also comprise a power coupling module configured for providing a transmission path of the inline power between the first port and the third port when the inline power is unavailable from primary network via the second port. | 02-12-2009 |
| 20090041005 | Method for activating an internet telephony hardware device - Systems and methods for activating an Internet telephony hardware device that is pre-configured with connection information are described. One embodiment of the method of the invention for activating an Internet telephony hardware device ( | 02-12-2009 |
| 20090041006 | METHOD AND SYSTEM FOR PROVIDING INTERNET KEY EXCHANGE - In a method and system for providing Internet Key Exchange (IKE) during a Session Initiation Protocol (SIP) signaling session, the method includes: enabling a caller end node device to send a first SIP request message to a callee end node device, wherein the first SIP request message includes a payload unit of a first IKE quick mode initial message; enabling the callee end node device to respond to the first SIP request message with an SIP response message, wherein the SIP response message including includes a payload unit of an IKE quick mode response message; and enabling the caller end node device to send a second SIP request message to the callee end node device, wherein the second SIP request message includes a payload of a second IKE quick mode initial message. | 02-12-2009 |
| 20090041007 | METHOD AND SYSTEM FOR OBTAINING INTERNET RADIO RESOURCES BASED ON SESSION INITIATION PROTOCOL - A method for obtaining Internet radio resources based on Session Initiation Protocol (SIP) is adapted for use between an administrator server and at least one client terminal. The method includes the following steps: (a) enabling the client terminal and the administrator server to set up a tunnel based on the SIP; (b) enabling the client terminal to send a SIP message requesting radio station data to the administrator server through the tunnel; and (c) enabling the administrator server to provide the radio station data to the client terminal through the tunnel. Since SIP has very good flexibility and functionality, obtaining Internet radio resources based on the SIP can overcome inconveniences associated with searching by the user. | 02-12-2009 |
| 20090041008 | HPNA HUB - Analog HPNA hub including at least one group of coils, the coils inducing HPNA signals there between, a plurality of filters, each of the filters coupled with a respective one of the coils and further coupled, via respective telephone wiring, with at least a respective HPNA node, wherein each of the filters enables transmission of HPNA data signals there through, and wherein each of the filters prevents transmission of conventional telephony signals there through. | 02-12-2009 |
| 20090041009 | IP TELEPHONE TERMINAL, IP TELEPHONE SYSTEM AND RECORDING MEDIUM - An IP telephone terminal outputs a call request for communication with a prescribed terminal and, if it is determined that connection failed, records the input voice. Thereafter, the IP telephone terminal determines that the prescribed telephone terminal has reached a state connectable through IP network, outputs a call request for communication with the prescribed terminal and if connection is determined to be established, transmits the recorded voice to the prescribed terminal. | 02-12-2009 |
| 20090041010 | Communication Diversion with a Globally Routable User Agent Uniform Resource Identifier System and Method - A method for diverting a Session Initiation Protocol (SIP) message is provided. The method includes using at least one Globally Routable User Agent Uniform Resource Identifier (GRUU) to determine a recipient to which the SIP message is diverted, and concealing an identity present in the SIP message. | 02-12-2009 |
| 20090046703 | USING AN IP REGISTRATION TO AUTOMATE SIP REGISTRATION - In one embodiment, a network device receives an Internet protocol (IP) registration request, such as a mobile IP registration request, from an access terminal. The network device may be a home agent that is configured to register the access terminal for IP services at the network layer. In addition to registering the access terminal at the network layer, the network device may facilitate registration at another layer, such as the application layer. In one example, registration information for the access terminal for an application layer registration, such as information needed to register for a session initiation protocol (SIP) services, is determined. The network device then facilitates registration at the application layer automatically using the registration information. | 02-19-2009 |
| 20090046704 | Providing Effective Advertising Via Synchronized Telephone and Data Streams - Information, such as advertising, is presented to VoIP users ( | 02-19-2009 |
| 20090046705 | Method and System for Facilitating Establishment of an Ip-Link in a Telecommunications System - A mobile switching center (MSC) | 02-19-2009 |
| 20090046706 | Managed Wireless Mesh Telephone Network And Method For Communicating High Quality Of Service Voice And Data - A telecommunications system includes a managed wireless mesh network capable of transmitting Internet Protocol (IP) packets therethrough, a controller in communication with the mesh network and in communication with the Public Switched Telephone Network, and a communication device in communication with the mesh network. The communication device converts a sound communication into at least one VOIP packet, and transmits and receives VOIP and non-VOIP packets to and from the mesh network, respectively. The packet containing the converted sound communication is set to a higher priority than a non-VOIP packet that does not contain the converted sound communication. | 02-19-2009 |
| 20090046707 | Apparatus for enhanced information display in end user devices of a packet-based communication network - Method and apparatus for information conveyance in an end user device of a packet-based communication service where the end user device is connected to a PSTN-based communication device includes detecting a power up condition of the end user device connected to the PSTN-based communication device, detecting a packet-based network connection, retrieving an end user profile from the packet-based communication service attempting a communication registration operation and displaying one or more non-binary type messages at the end user device regarding the status of the communication service. The apparatus for enhanced information conveyance includes a main body having a local area packet network connection means, a wide area packet network connection means and a non-packet network connection means for connection of a PSTN-based communication device and a display panel body adapted to display information regarding the status of the communication service in a non-binary manner. | 02-19-2009 |
| 20090052434 | METHODS AND APPARATUS TO SELECT A VOICE OVER INTERNET PROTOCOL (VOIP) BORDER ELEMENT - Methods and apparatus to select a voice over Internet protocol (VoIP) border element are disclosed. An example method comprises sending a first session initiation protocol (SIP) protocol message from a first voice over Internet protocol (VoIP) device, the first SIP message comprising an Internet protocol (IP) address shared by at least two VoIP border elements, and receiving a second SIP message at the first VoIP device from a second VoIP device, the second SIP message comprising a unique address for the second VoIP device, the second VoIP device to be selected based on the shared IP address. | 02-26-2009 |
| 20090052435 | RELAY DEVICE, COMMUNICATION SYSTEM, AND CONTROL METHOD AND PROGRAM FOR THEM - Relay devices T are installed in opposition to each other across an FW to implement an FW traversal communication between communication addresses such as IP addresses. Each relay unit | 02-26-2009 |
| 20090052436 | IP Telephone System - An IP telephone system according to the present invention includes a plurality of communication devices T | 02-26-2009 |
| 20090052437 | System and Method for Dynamic Telephony Resource Allocation Between Premise and Hosted Facilities - A population of networked Application Gateway Centers or voice centers provides telephony resources. The telephony application for a call number is typically created by a user in XML (Extended Markup Language) with predefined telephony XML tags and deployed on a website. A voice center provides facility for retrieving the associated XML application from its website and processing the call accordingly. The individual voice centers are either operated at a hosted facility or at a customer's premise. Provisioning Management Servers help to allocate telephony resources among the voice centers. This is accomplished by suitably updating a voice center directory. In this way, the original capacity at a premise, predetermined by the hardware installed, can be adjusted up or down. If the premise is under capacity, it can be supplemented by that from a hosted facility. If the premise has surplus capacity, it can be reallocated for use by others outside the premise. | 02-26-2009 |
| 20090052438 | METHOD, SYSTEM AND DEVICE FOR PROCESSING SUPPLEMENTARY SERVICES - Method, system and apparatus for processing supplementary services, and MGCF enhanced method and apparatus as well. The method for processing supplementary services is used for supplementary services in packet network when CSI terminal and IMS terminal are interworking. The method includes: after receiving session message relating to supplementary services interworking control function entity extracts detailed content from the session message; according to the detailed content, interworking control function entity executes corresponding supplementary services. The interworking control function entity includes: supplementary service information receiving unit, which receives session message relating to supplementary services and extracts detailed content from the session message; supplementary service operating unit, which executes corresponding supplementary services according to the detailed content. | 02-26-2009 |
| 20090052439 | PACKET TELEPHONY APPLIANCE - A packet telephony appliance includes a Euphony network processor that integrates networking and DSP functions to provide a low cost and efficient solution in building a networked appliance. In particular, a Euphony ATM Telephone (EAT) is built around the Euphony network processor. The EAT uses a real-time operating system to provide predictable processing and networking support. The EAT implements IObufs, which provides a unified buffering scheme that allows zero-copy data movement. Furthermore, the EAT uses an Event Exchange (EVX), which provides a flexible mechanism for event distribution, allowing software modules to be composed together in an extensible manner. EVX and IObufs are used together to provide highly efficient intra-appliance communication. The EAT provides a platform that can evolve gracefully to support new protocols, advanced telephony services and enhanced user interfaces. | 02-26-2009 |
| 20090052440 | SYSTEM AND METHOD FOR FACILITATING COMMUNICATION BETWEEN A CMTS AND AN APPLICATION SERVER IN A CABLE NETWORK - A system and method for facilitating communication between a CMTS and a VoIP application server in a cable network. VoIP-enabled customer premises equipment (CPE) generates packets that are sent through a cable modem (CM) to a cable modem termination system (CMTS). A packet is parsed by CMTS and the destination IP address and port number compared to the destination IP address-port tuples received by the CMTS from a datastore. A packet that is directed to an IP address-port tuple on the target list (a “service request packet”) is modified to incorporate CMTS-identifying information and subscriber-identifying information in the packet header. When the VoIP application server communicates with the CMTS to reserve the network resources, the VoIP application server provides the CMTS with the CM MAC and CM IP addresses to facilitate resource allocation, subscriber identification and billing. This Abstract is not to be considered limiting, since other embodiments may deviate from the features described in this Abstract. | 02-26-2009 |
| 20090059894 | METHODS AND APPARATUS TO SELECT A PEERED VOICE OVER INTERNET PROTOCOL (VOIP) BORDER ELEMENT - Methods and apparatus to select a peered voice over Internet protocol (VoIP) border element are disclosed. An example method comprises receiving a session initiation protocol (SIP) message that includes an identifier representative of a location of a voice over Internet protocol (VoIP) access border element, querying a telephone number mapping (ENUM) database to identify two or more peered VoIP border elements, and selecting a one of the two or more peered VoIP border elements based on the identifier. | 03-05-2009 |
| 20090059895 | METHODS AND APPARATUS TO DYNAMICALLY SELECT A PEERED VOICE OVER INTERNET PROTOCOL (VOIP) BORDER ELEMENT - Methods and apparatus to select a dynamically peered voice over Internet protocol (VoIP) border element are disclosed. An example method comprises collecting data representative of a dynamic performance of a voice over Internet protocol network, prioritizing a selection of a peered border element based on the collected data, and modifying a telephone number mapping (ENUM) database based on the prioritized selection. | 03-05-2009 |
| 20090059896 | REMOTE CONNECTION TO A TELEPHONE LINE VIA INTERNET - A device receives a request from an Internet Protocol (IP)-based device to create a virtual extension of a plain old telephone service (POTS)-based telephone line, authenticates the IP-based device for association with the POTS-based telephone line, and creates the virtual extension of the POTS-based telephone line to the IP-based device when the IP-based device is authenticated. | 03-05-2009 |
| 20090059897 | IDENTITY-BASED INTERACTIVE RESPONSE MESSAGE - A system that can deliver a tailored message based upon characteristics surrounding an incoming communication. In one aspect, the system is a targeted voice-mail system that has the capability to provide a unique voice-mail depending upon the communication characteristics which include the identity of caller or the initiator of the call, whether a specific identity or within a group, the identity for which the call is targeted, and the intent of the caller. Additionally, other contextual factors can be considered in generating, locating and/or rendering a tailored response message. | 03-05-2009 |
| 20090059898 | Method and Apparatus for Signaling the Subscriber Type of IP and Non-IP Subscribers Using the Hostpart of the SIP URI - To identify a subscriber type of IP and non-IP subscribers with SIP without adding additional signaling elements to existing SIP headers, a method for identification of subscriber type with SIP makes use of the originating subscriber URIs are in form of userpart@hostpart. The userpart uniquely identifies the originating subscriber. A first switch identifies itself to a second switch through the hostpart. The first switch signals via SIP to the second switch the originating subscriber-type by using the hostpart to define a logical grouping identifying the originating subscriber-type. | 03-05-2009 |
| 20090059899 | OPTIMIZED PACKET PROCESSING ARCHITECTURE FOR BATTERY POWERED MOBILE COMMUNICATION DEVICE - A transceiver includes a peripheral device, a first processor configured to control an operation of the peripheral device, at least one second processor configured to transport data between the transceiver and at least one wireless network, and a third processor connected between the first processor and the at least one second processor. The third processor is configured to control the at least one second processor for executing a network operation independently of the first processor. | 03-05-2009 |
| 20090059900 | External System Access to Telephone Line through VOIP Telephony Device - A telephony device is configured to provide VoIP service at a customer premises and is also configured to provide an external system connected to the telephony device with the ability to seize a telephone line at the customer premises when needed. The telephony device includes an embedded MTA (EMTA), a telephone circuit, and a switch connector configured to connect the external system with the telephony device. When the external system is connected to the telephony device via the switch connector, the switch connector routes telephone signals between the EMTA and the telephone circuit though the external system, and the external system, such as an alarm system, may seize the line when needed. When the external system is not connected, the switch connector connects the EMTA and the telephone circuit. | 03-05-2009 |
| 20090059901 | VOIP network phone forwarding device and application thereof - A VOIP network phone forwarding device is used to forward a call of an extension to a dual-mode mobile terminal. The VOIP network phone forwarding device comprises a wireless network module, a network interface, a phone connection interface, and a processor. The wireless network module connects to a wireless network for signal transmission with the dual-mode mobile terminal via the wireless network. The network interface is coupled to a modem device for connecting to the Internet. The phone connection interface is coupled to a PBX, which is further coupled to several extensions. The processor is coupled to all the above components. When a dialing source dials a phone call to an extension, the processor forwards this call to the dual-mode mobile terminal via the wireless network or the Internet so that a user can answer this call. A mobile extension communication architecture can thus be realized to let users be able to answer any phone call anytime, anywhere. | 03-05-2009 |
| 20090059902 | IP TELEPHONE TERMINAL AND COMPUTER READABLE STORAGE MEDIUM - An IP telephone terminal and a method for controlling IP telephone terminal capable of performing IP telephone communication upon receiving information from a conversation application through a user interface provided with a microphone and a loud speaker when a conversation is to be performed with the microphone and the laud speaker. The IP telephone terminal is connected to an internet. The terminal has a mouse and a keyboard operable as an operation input device, and a microphone and a loud speaker operable as an audio input device. A first selecting unit selects the operation input device of an interface device, and a second selecting unit selects the audio input device of the interface device. A third selecting unit selects one of the interface devices. A first control part controls the first selecting unit so that the first selecting unit selects the operation input device of the interface device selected by the third selecting unit. A second control part controls the second selecting unit so that the second selecting unit selects the audio input device of the interface device selected by the third selecting unit. | 03-05-2009 |
| 20090059903 | HIERARCHICAL DATA COLLECTION NETWORK SUPPORTING PACKETIZED VOICE COMMUNICATIONS AMONG WIRELESS TERMINALS AND TELEPHONES - A packet-based, hierarchical communication system, arranged in a spanning tree configuration, is described in which wired and wireless communication networks exhibiting substantially different characteristics are employed in an overall scheme to link portable or mobile computing devices. The network accommodates real time voice transmission both through dedicated, scheduled bandwidth and through a packet-based routing within the confines and constraints of a data network. Conversion and call processing circuitry is also disclosed which enables access devices and personal computers to adapt voice information between analog voice stream and digital voice packet formats as proves necessary. Routing pathways include wireless spanning tree networks, wide area networks, telephone switching networks, internet, etc., in a manner virtually transparent to the user. A voice session and associate call setup simulates that of conventional telephone switching network, providing well-understood functionality common to any mobile, remote or stationary terminal, phone, computer, etc. | 03-05-2009 |
| 20090059904 | PROVIDING A NETWORK NODE WITH SERVICE REFERENCE INFORMATION - Service reference information is added to an IP telephony signaling protocol message and the IP telephony signaling protocol message is then sent to the network node in order to provide a network node using the IP telephony signaling protocol, e.g., SIP, with service reference information needed for billing purposes. | 03-05-2009 |
| 20090059905 | SYSTEM AND METHOD OF PROVIDING A HIGH-QUALITY VOICE NETWORK ARCHITECTURE - Embodiments of the invention include a system and method for providing high quality voice/sound communications over a local loop of a telephone network. The method aspect of the invention comprises receiving a voice signal, digitizing the voice signal into a high quality voice signal, utilizing sampling rates greater than 8000 samples per second and/or sample sizes greater than 8 bits per sample, negotiating voice processing characteristics between a customer premises equipment and a network element such as a softswitch, receiving speech from a user at a customer premises equipment according to the negotiation, converting the received speech into high bandwidth signal and transmitting the high bandwidth signal to a telephone local loop, transmitting the high bandwidth signal from the local loop to wideband node that packetizes the high bandwidth signal for transmission to a packet network and receiving the packetized signal from the packet network at a switch that switches between an on-network or off-network status. A voice over IP platform may also be used to route packetized signals from the packet network to either the telephone network or another packet network. | 03-05-2009 |
| 20090059906 | ROUTING OF TELECOMMUNICATIONS - A gateway ( | 03-05-2009 |
| 20090067409 | SYSTEM FOR COMMUNICATING BETWEEN INTERNET PROTOCOL MULTIMEDIA SUBSYSTEM NETWORKS - A system that incorporates teachings of the present disclosure may include, for example, a Telephone Number Mapping system operating in a first IP Multimedia Subsystem (IMS) communication system having a controller adapted to receive first contact information of a communication device registered through a terminating Serving Call Session Control Function (S-CSCF) operating in a second IMS communication system and second contact information of at least one among an Interrogating CSCF (I-CSCF) of the second IMS communication system and the terminating S-CSCF. Additional embodiments are disclosed. | 03-12-2009 |
| 20090067410 | DETECTION OF SPIT ON VOIP CALLS - A method for packet telephony includes receiving over a packet communication network ( | 03-12-2009 |
| 20090067411 | Call Forwarding in an IP Multimedia Subsystem (IMS) - A method and Serving Call/State Control Function (S-CSCF) for handling a Session Initiation Protocol (SIP) communication within an IP Multimedia Subsystem (IMS), wherein the communication is subject to a call-forwarding operation handled by a SIP Application Server (AS). An INVITE is received at the S-CSCF, which serves a user equipment (UE) identified by an R-URI. The S-CSCF adds a URI for the S-CSCF to the INVITE route header together with an Original Dialog Identifier (ODI) mapped to the R-URI. The S-CSCF forwards the INVITE to the AS, which changes the R-URI to a URI of a UE to which the call is to be forwarded. The AS adds a forwarding indicator to the INVITE and returns it to the S-CSCF. The S-CSCF identifies the forwarding indicator and determines the original R-URI based on the ODI received in the returned INVITE. The S-CSCF determines call restrictions and Initial Filter Criteria (IFCs) based on the original R-URI. | 03-12-2009 |
| 20090067412 | METHOD FOR SUPPORTING MULTIPLE DEVICES FROM A BROADBAND CONNECTION - A method adds a MAC address per line for a multiline EMTA. After the EMTA initializes, the method creates “Virtual MTA” instances corresponding to each analog line/MAC address. The method facilitates MTA emulation of each of the Virtual MTA instances. For each virtual EMTA line, the emulation method includes acquiring an IP address via DHCP and acquiring a configuration file via TFTP for each virtual MTA instance. | 03-12-2009 |
| 20090067413 | PACKET NETWORK BASED EMERGENCY BACKUP TELEPHONE SYSTEM - In an emergency backup telephone system, members of an enterprise use their personal computers to log into an emergency communications web page. Upon logging in, software that enables the personal computer to act as a webphone is automatically downloaded. This software allows a person to initiate a call from personal computer to a conventional PSTN number destination using a PSTN gateway, or to another party's computer at a specified URL using VoIP telephony. Upon logging in, an authoritative index of employees reachable via the backup system updated to include information such as a phone number and/or IP address where the member can be reached in order to allow calls originating from the PSTN to be routed to the member's computer. The index is made available to other members of the enterprise via the enterprise's intranet, and, in some embodiments, to the public via a web page on the internet and/or email. | 03-12-2009 |
| 20090073957 | Apparatus and methods for data distribution devices having selectable power supplies - A network apparatus includes an independent power supply providing a first power signal, and a data distribution device which is operably coupled to the independent power supply and a remote data distribution device, where the remote distribution data device exchanges data and provides a second power signal to the data distribution device, and further where the data distribution device selects the first power signal or the second power signal for operational power. A method includes scanning a plurality of sensors, each coupled to a plurality of power inputs, to ascertain if a power signal is present, determining whether a power signal associated with an independent power supply is present at a power input, sourcing power from the independent power supply if the power signal is associated with an independent power supply, and sourcing power from an alternative supply if the power signal is not associated with an independent power supply. | 03-19-2009 |
| 20090073958 | METHOD AND SYSTEM FOR TRANSMISSION OF CHANNEL QUALITY INDICATORS (CQIs) BY MOBILE DEVICES IN A WIRELESS COMMUNICATIONS NETWORK - A method and system for optimizing channel quality indicator (CQI) transmissions by mobile devices in a cellular network allows transmission of CQIs at a slower rate and with fewer bits during voice-over-internet-protocol (VoIP) sessions than during non-real-time (NRT) data transmissions. A VoIP transmission typically includes “talkspurt” periods, during which VoIP packets are transmitted, and silence periods, which start with a silence indication (SID) packet and continue with periodic SID packets until a VoIP packet is received. When the base station is transmitting NRT data, the mobile device transmits CQIs to the base station at a first rate, with each CQI having a first fixed number of bits. When the base station is transmitting VoIP to the mobile device, then during a talkspurt period, the mobile device may transmit CQIs to the base station at a second rate slower than the first rate, and each CQI may have a second fixed number of bits less than the first fixed number of bits. However, during a silence period, the mobile device does not transmit CQIs to the base station, and uplink channel resources allocated for the CQIs can be reallocated to other mobile devices. | 03-19-2009 |
| 20090073959 | METHOD AND SYSTEM FOR VOICE-OVER-INTERNET-PROTOCOL (VoIP) TRANSMISSION IN A WIRELESS COMMUNICATIONS NETWORK - The invention is a method and system for reliably detecting the start and/or end of silence periods during voice-over-internet-protocol (VoIP) sessions in a wireless communications network. A VoIP session typically includes “talkspurt” periods, during which VoIP packets are transmitted, and silence periods, during which silence indication (SID) packets are transmitted. Both the base station (eNodeB or eNb) and the mobile device (user equipment or UE) may inspect the packets to identify them as VoIP or SID packets. Alternatively, only the eNB inspects the packets. The eNB then flags the first SID packet after a VoIP packet as the start of a silence period, and flags the first VoIP packet after a SID packet as the end of a silence period. The eNB then modifies the header of the medium access control (MAC) protocol data unit (PDU) prior to transmission to the UE. The UE then detects the modified MAC headers to identify the start and/or end of silence periods. | 03-19-2009 |
| 20090073960 | BRIDGING PHONE NETWORKS USING VOIP TO PRESERVE IN-NETWORK CALLING ADVANTAGES - A call may be accomplished from a first mobile network to a second mobile network by bridging the first and second mobile networks using VoIP. A first communication is initiated from a first mobile device to a first Voice over Internet Protocol (VoIP) server circuitry emulating a mobile phone. The first VoIP server circuitry receives an indication of a destination number of a mobile phone on the second mobile network. Communication takes place between the first VoIP server circuitry and the second VoIP server circuitry according to an internet protocol to conference the first mobile device on a first telephone call, to the first VoIP server, to the second mobile device, on a second telephone call to the second VoIP server. The first and second telephone call are each intra-network, and the VoIP communication accomplishes a bridge between the first mobile network and the second mobile network. The second telephone call may be via a plain old telephone service (POTS) rather than via a second mobile network. | 03-19-2009 |
| 20090073961 | METHOD, COMPUTER PROGRAM PRODUCT, AND APPARATUS FOR PROVIDING AUTOMATIC GAIN CONTROL VIA SIGNAL SAMPLING AND CATEGORIZATION - An apparatus for detecting and adjusting volumes levels may include a processor capable of receiving data from a carrier(s). The processor is also capable of receiving trigger control signals from a trigger control and arranging the data into frames that are stored in a buffer. The processor is also capable of determining whether the trigger control signals include data indicating whether a determination regarding adjustment of a volume level of the data is required. The apparatus also includes a packet analyzer capable of calculating an average volume level associated with the frames when a determination reveals that adjustment of the volume level is required and is capable of generating categories, corresponding to intensity levels and categorizing the frames according to the intensity levels based on the average. The packet analyzer is also capable of determining whether to adjust the volume level based on a category assigned to the frames. | 03-19-2009 |
| 20090073962 | Modular messaging log application on an IP phone - Presented is a method for selectively retrieving voice messages. The method includes accessing a list of received calls, and accessing a subset of the list of received calls. Each of the calls in the subset has an associated stored message. The method further includes selectively accessing information associated with a particular call in the subset of received calls, and retrieving the message associated with the particular call. Also presented is an internet protocol (IP) phone that includes a screen configured to display a list of received calls, a soft key configured to display a subset of the list of received calls when pressed. Each of the calls in the subset has an associated stored message. The phone further includes soft keys configured to selectively display information associated with a particular call in the subset of received calls when pressed, and retrieve the message associated with the particular call when pressed. | 03-19-2009 |
| 20090073963 | Method and network unit for setting up a connection in a second network - Method and network unit for setting up a connection in a second network ( | 03-19-2009 |
| 20090073964 | Providing services in case of call diversion in a communication system - The present invention relates to an S-CSCF receiving a terminating request associated with a called user and executing services for the called user. The S-CSCF determines an indication in Session Case indicating originating services handling in call forwarding situation, and based on this executes a subset of services for the user. | 03-19-2009 |
| 20090073965 | Methods, smart cards, and systems for providing portable computer, voip, and application services - A smart card is used with a network based system to providing portable telecommunication and computing services. In an exemplary embodiment the smart card holds user application programs and/or user data such as a calling list, account information, a list of local or remote application programs, and user interface configuration settings. The smart card transfers the user data to one of a plurality of geographically dispersed card readers which are each connected to a local computerized device such as a computer or a telephony device. When the smart card is plugged into a first card reader, the user's customized settings and/or user interface is configured at a first local computerized device. When the smart card is plugged into a second smart card reader, the user's customized settings and/or user interface is configured at a second local computerized device. Hence the user can use different computerized devices and still have the same configuration and user interface as though the various computerized devices had each been individually customized for the user. | 03-19-2009 |
| 20090073966 | Distribution of Identifiers in Serverless Networks - A method for assigning identifiers in a distributed system involves establishing a circle as a locus of all identifiers, with the value of any point on the circle being the portion of one complete revolution in a first direction around the circle to the point, measured from a first zero point, and selecting values to be assigned as identifiers as needed by rounds of assignment, wherein the beginning and end of any round of assignment has identifiers assigned with point values that divide the circle into equal-length sectors. The method is useful in and applied to serverless telephony systems. | 03-19-2009 |
| 20090080409 | METHOD, COMPUTER PROGRAM PRODUCT AND APPARATUS FOR PROVIDING NON-INTRUSIVE VOICE OVER INTERNET PROTOCOL (VoIP) MONITORING AND RECORDING - An apparatus for non-intrusively monitoring and recording data (e.g., speech data) associated with a call(s) as well as addition and/or removal of a user(s) to/from a communication may include a processor capable of receiving speech data generated by a user of a device that subscribes to a network(s). The processor is further capable of receiving trigger control signals and determining whether the trigger control signals contain data indicating whether recording and monitoring of the data is required as well as addition and/or removal of a user to a communication is required. The processor is further capable of generating one or more copies of the speech data when the determination reveals that the recording and monitoring of the speech data is required and is further capable of generating sound corresponding to the speech data when the determination reveals that the recording and monitoring of the speech data is not required. | 03-26-2009 |
| 20090080410 | Speech Processing Peripheral Device and IP Telephone System - There are provided with an IP telephone system having both convenience of the softphone and durability of the hardphone, and a speech processing peripheral device ( | 03-26-2009 |
| 20090080411 | System and method for providing carrier-independent VoIP communication - Systems and methods for seamlessly providing carrier-independent VoIP calls initiated using an existing carrier-issued telephone number are provided. In exemplary embodiments, the existing carrier-issued telephone number to be called is received. Subsequently, a status regarding if the existing carrier-issued telephone number is a registered telephone number stored in a carrier-independent database is determined. If the existing carrier-issued telephone number comprises a registered telephone number in the carrier-independent database, a call is established via peer-to-peer connection using an address associated with the registered telephone number. However, if the existing carrier-issued telephone number is not a registered telephone number in the carrier-independent database, the call is placed via a standard route. | 03-26-2009 |
| 20090080412 | COMMUNICATION APPARATUS AND COMMUNICATION CONTROL METHOD - A communication apparatus for dividing voice data into a plurality of packets and transmitting the packets to a destination communication apparatus includes a detection unit configured to detect one of predetermined events which are triggers of change processing of a packet division length used for the dividing, a determination unit configured to determine, when the detection unit detects the one of the predetermined events, a possible range of the packet division length after the change processing based on predetermined external information which influence the packet division length, a negotiation unit configured to negotiate the packet division length after the change processing with the destination communication apparatus based on the range determined by the determination unit, and, a control unit configured to control the packet transmission based on the packet division length negotiated by the negotiation unit. | 03-26-2009 |
| 20090080413 | IP Telephone System - An Internet Protocol (IP) telephone has a constant impedance filter that is capable of being continuously attached to the physical layer of a computer chip in the IP telephone. The constant impedance filter is located outside the physical layer and is connected to a relay on the physical layer. The relay is configured using native FET devices, which are normally conductive without a supply voltage. Therefore, the relay is capable of operating during the discovery mode of IP telephone operation, where no power is applied to the substrate. Rectifier circuits rectify an incoming signal during discovery mode, and apply the rectified signal to the gate of the relay to improve conductivity of the relay. This allows for faster detection of the IP telephone during discovery mode. During normal operation mode, voltage is applied to the physical layer, and the relay is opened by grounding the native devices. Also, during the normal operation mode, any signal coming from the constant impedance filter is terminated in a switchable termination resistor that is also disposed on the physical layer. | 03-26-2009 |
| 20090080414 | METHOD OF PROPOGATING MULTIPLE IP TELEPHONY ROUTES, AND A CORRESPONDING LOCATION SERVER AND COMPUTER PROGRAM - A method is provided for propagating routes between a first location server of a first IP telephony domain and a second location server of a second IP telephony domain. The method includes the following stages: the first location server receives a first propagation message from at least one neighboring location server and containing at least two routes enabling a destination to be reached, referred to as propagation routes; and the first location server advertises the at least two routes to at least one second location server of a second telephony domain neighboring the first. | 03-26-2009 |
| 20090080415 | LATE FRAME RECOVERY METHOD - Method of processing a transmitted encoded media data stream is received. If a data element arrives prior to, or at, a predetermined playout deadline, the data element is decoded, the media represented by the decoded data element is played, and the data element is provided to a decoder state machine to update a decoder state. If a data element arrives after the predetermined playout deadline, the data element is provided to the decoder state machine to update the decoder state. In one embodiment, if the specified data element fails to arrive by the playout deadline, a subsequently received data element is saved in memory. Then, if the specified data element arrives after the predetermined playout deadline, the specified data element and the saved, subsequently received, data element are provided to the decoder state machine to update the decoder state. | 03-26-2009 |
| 20090086715 | Method and system for implementing dynamic signaling routing - A method for implementing dynamic signaling routing includes: A. sending a register request from a Terminal Element (TE) to Service Elements (SEs) via a Network Element (NE); B. upon receiving the register request, determining one of the SEs which will provide signaling service for the TE in accordance with association information recorded in the NE. Further, a system for implementing dynamic signaling routing, comprising: Service Elements (SEs), for providing signaling service; Terminal Elements (TEs), for sending register requests to the SEs; and Network Elements (NEs), between the TEs and the SEs, for determining one of the SEs which has provided signaling service for the TE in accordance with association information recorded in the NEs upon receiving the register request. | 04-02-2009 |
| 20090086716 | Real Time Measurement Of Network Delay - Delay is measured associated with the transfer of voice signals involving a telephone connected to a PSTN carrier (e.g., non VoIP based) where the call is terminated by an operator agent using a workstation connected to a VoIP based network. A test tone is provided to the telephone causing a tone to be generated at a headset of the workstation. An oscilloscope measures the delay using an input of a first signal associated with the generation of the test tone at the telephone, and a second signal associated with the generation of the resulting tone at the headset. The tone at the headset can be looped back into the headset microphone, causing a return signal to be generated and measured. Once the overall delay is known, and the delay of certain elements are estimated, the delay associated with other network elements, including the workstation, can be determined. | 04-02-2009 |
| 20090086717 | METHODS AND APPARATUS FOR BANDWIDTH MANAGEMENT WITHIN A MEDIA OVER INTERNET PROTOCOL NETWORK BASED ON A SESSION DESCRIPTION - In one embodiment, a method includes receiving a request to establish at least a portion of a media session between a session exchange device and a network entity based on at least a portion of a session description. The session exchange device and the network entity being associated with a media over internet protocol (MoIP) network. The method includes receiving an indicator at the session exchange device that the portion of the session description is not associated with a predefined data-transfer-rate value. A request for a user-defined data-transfer-rate value is sent in response to the indicator. | 04-02-2009 |
| 20090086718 | Method and apparatus for facilitating telecommunication network selection - A method, apparatus, and computer usable program product for facilitating a selection of a telecommunication network are provided. A request for a type of network associated with a called identifier is received from a calling communication device. A repository of information about caller identifiers is searched. Information corresponding to the called identifier is selected. The selected information includes the type of network associated with the called identifier. The selected information is returned to the calling communication device. The information in the repository is updated by adding information about new caller identifiers, updating information about the several caller identifiers existing in the repository, or both. | 04-02-2009 |
| 20090086719 | Dynamic initiation of I1-ps signaling in IMS centralized services - A device is described which comprises a sender and a receiver configured to perform a circuit switched session, wherein the sender and the receiver are capable to also send and receive session control messages according to a session control protocol, and a controller configured to receive a specific session control message, wherein the session control message includes information for establishing a specific packet switched connection capable of supporting a specific packet services to a network node, wherein the information is encoded in a format suitable for the session control protocol, wherein the controller is further configured to obtain the information from the specific session control message, and to establish the specific packet switching connection based on the obtained information. | 04-02-2009 |
| 20090086720 | IDENTITY ASSOCIATION WITHIN A COMMUNICATION SYSTEM - A system is disclosed that combines social network linkage information with a caller's phone number or identity. This linkage enables communication systems, such as Voice-over-Internet Protocol (VoIP) phone systems, to provide rich caller-ID information. The system can also combine other contextual and mobility-based information to provide rich information to a user with respect to incoming (or outgoing) communications. | 04-02-2009 |
| 20090086721 | SYSTEM AND METHOD TO DETERMINE A LOCATION ASSOCIATED WITH AN INTERNET PHONE - An Internet phone may be physically located based on its credential. The credential is related to a MAC address of the Internet phone. The MAC address is related to a port identifier of a network switch in communication with the Internet phone. The port identifier is related to a physical location of the Internet phone. | 04-02-2009 |
| 20090086722 | COMMUNICATION APPARATUS AND TERMINAL REGISTRATION METHOD FOR USE IN COMMUNICATION SYSTEM - According to one embodiment, a communication apparatus includes an agent server module configured to connect to the telephone terminal connected to the public network, while bypassing the public network and the private network, and configured to receive a registration request including the terminal ID of the telephone terminal from the telephone terminal, and a controller which performs registration processing based on the registration request received by the agent server module via the bypass, and generates a session for terminal control with respect to the telephone terminal of a requester via the NAT router. | 04-02-2009 |
| 20090086723 | METHOD FOR SETTING UP A COMMUNICATION CONNECTION AND PRIVATE BRANCH EXCHANGE FOR CARRYING OUT THE METHOD - A communication connection between a calling communications terminal and a further communications terminal, the connection setup being initiated through the exchange of an invite message and a number of acknowledgment messages between the calling communications terminal, the further communications terminal and a higher-ranking communication-management module, the connection modalities relevant for the calling communications terminal being agreed in a first connection initiation sequence between the higher-ranking communication-management module and an application-management module allocated to the calling communications terminal, and a second connection initiation sequence being provided for agreeing the connection modalities relevant for the further communications terminal between the higher-ranking communication-management module and an application-management module allocated to the further communications terminal. | 04-02-2009 |
| 20090086724 | METHOD OF SELECTING A TELEPHONY ROUTE WITHIN AN IP TELEPHONY DOMAIN, AND CORRESPONDING APPARATUS AND COMPUTER PROGRAM - A method is provided for selecting a telephony route for at least one digital stream serving a telephony destination. The method is performed within a first location server belonging to a first IP telephony domain deployed on at least one autonomous system. The autonomous system exchanges IP routing information with its neighbors designating at least one IP destination for updating an IP routing table. The method includes: the first location server searching for the IP routing information, the IP routing information including an identifier of a second IP telephony domain having associated therewith the at least one IP telephony destination, referred to as the destination identifier; and selecting the IP telephony route to reach the at least one telephony destination, applying a predetermined criterion for selecting the second telephony domain as a function of the destination identifier. | 04-02-2009 |
| 20090086725 | METHOD AND SYSTEM FOR TRANSMITTING MESSAGE SERVICE DATA - This invention discloses a method for transmitting message service data, which includes: adding, at a message sender, the message service data into a Session Initiated Protocol message or a Message Session Relay Protocol message, and transmitting the message service data to a message receiver through the Session Initiated Protocol message or a Message Session Relay Protocol message. The invention also discloses a system for transmitting message service data, which includes a message service data processing module, including a message adding module and a transmitting module. Different types of message service data can be transmitted according to the invention. | 04-02-2009 |
| 20090092126 | METHOD AND SYSTEM FOR RETRIEVING LOG MESSAGES FROM CUSTOMER PREMISE EQUIPMENT - An approach is provided for retrieving a system log. Packets that are destined for a predetermined network address and network port are detected and captured. The packets represent a log file corresponding to a customer premise equipment (CPE) for troubleshooting. A data file is generated to contain the log file, wherein the packets are discarded, by at a firewall, before reaching the predetermined network address and network port. | 04-09-2009 |
| 20090092127 | METHOD FOR ESTABLISHING A COMMUNICATION CONNECTION AND COMMUNICATION DEVICES - A method for establishing a communication connection includes transmitting an identification for identifying a first communication device to the first communication device or a second communication device via a packet-switched first communication connection between the first communication device and the second communication devices and using the identification to establish a circuit-switched second communication connection between the first communication device and the second communication device. | 04-09-2009 |
| 20090092128 | IP TELEPHONE SYSTEM AND COMPUTER READABLE STORAGE MEDIUM - The invention provides an IP telephone terminal. An IP telephone terminal includes an identification data receiving unit, a communicating unit, an IP telephone function controlling unit, a determining unit, a terminal data acquiring unit, and a terminal data acquiring unit. The identification data receiving unit receives, over an Internet, identification data identifying another IP telephone terminal. The communicating unit establishes IP telephone communications with the other IP telephone terminal identified by the identification data via the Internet. The IP telephone function controlling unit controls execution of an IP telephone function used to implement a telephone call with the another IP telephone terminal via the communicating unit. The determining unit determines the identification data received by the identification data receiving unit. The terminal data acquiring unit acquires terminal data associated with the identification data transmitted from the other IP telephone terminal over the Internet, in case the determining unit determines that the terminal data identifying functions that the other IP telephone terminal can control is associated with the identification data received by the identification data receiving unit. The process data transmission controlling unit controls transmission of a process data to the other IP telephone terminal via the IP telephone communications, where the process data is data used in the function identified by the terminal data acquired by the terminal data acquiring unit. | 04-09-2009 |
| 20090092129 | Data Driven Configuration of Call Management Applications - A call manager uses a call management application in conjunction with a live dial database to control routing of calls for managed devices. To generate the live dial database, the call management application accesses configured route patterns and enters these patterns into the live dial database. Upon identifying an expansion indicator in a configured route pattern, the call management application accesses dial plan data that includes multiple route pattern definitions that each define a pattern using one or more sub-strings and, for each sub-string, an associated tag. The call management application then enters patterns defined by the route pattern definitions into the live dial database based on various other criteria established for the configured route pattern having the expansion indicator. | 04-09-2009 |
| 20090092130 | NETWORK SWITCHING SYSTEM WITH ASYNCHRONOUS AND ISOCHRONOUS INTERFACE - To provide a switching system with telephone switching function mainly on the basis of hardware processing by using isochronous channel which is a real time communication channel. The switching system comprises a gateway node connected with ISDN (Integrated Services Digital Network) and PSTN (Public Switched Telephone Network), and one or more extension nodes, and a serial bus such as IEEE 1394 bus. The gateway node transforms data rate of outside line into data rate of extension node, and the other way around, and secure a seamless communication channel. Concretely, the gateway node secures an isochronous channel, according to a request from the extension nodes or the outside line, and executes switching such as transfer or reservation. A resource manager holds a table for managing the gateway node and extension node. | 04-09-2009 |
| 20090097471 | METHOD AND APPARATUS FOR CALL PROCESSING FOR SIP AND ISUP INTERWORKING - A system that incorporates teachings of the present disclosure may include, for example, a server having a controller to adjust a call processing logic for Session Initiated Protocol to Integrated Services Digital Network User Part (ISUP) calls based at least in part on interworking profiles assigned to ISUP trunk groups supporting the calls. Additional embodiments are disclosed. | 04-16-2009 |
| 20090097472 | Method and apparatus for optimizing telephony communications - There is provided a method and apparatus for determining and optimizing a transmission route for a phone call. The phone call may be either local or long distance. For a local phone call the voice data is transmitted via known local methods. For a long distance phone call, middleware determines an optimal internet terminal service provider for carrying or transmitting the voice data. The middleware determines optimal internet terminal service provider by performing a comparative analysis of several internet terminal service providers according to cost and transmission quality. | 04-16-2009 |
| 20090097473 | Functional extended system of network communication device and its method - A functional extended system of network communication device and its method are disclosed according to the present invention. The functional extended system of network communication device and its method are applicable to a network communication function extended equipment that is connecting to a network communication device that has universal communication transmit port, mainly it connects to the universal communication transmit port of the network communication device via an extended communication transmit port, and it also has its data processing unit and the network communication device produce a handshake based on a preset communication protocol, and then it has the data processing unit execute data process that includes at least encoding/decoding data transmitted from/to the network communication device based on the preset communication protocol, thus the data processing unit is able to drive corresponding functional extended module to execute extending function, and/or it can further connect with other functional extended system of network communication device, therefore, the network communication device is capable of extending continually. | 04-16-2009 |
| 20090097474 | System and method for providing location information to a public safety answering point during an emergency 911 call from a softphone - A system and method for providing location information to a public safety answering point from a softphone may include receiving, at a network access point, an emergency 911 call from the softphone. The emergency 911 call may be communicated to a public safety answering point. In response to a call connection message being received, an address location of the network access point to which the softphone is in communication in placing the emergency 911 call to the public safety answering point may be communicated in a type II caller ID data packet. The softphone may generate the type II caller ID data packet with the address location in a data field, such as a data field typically used for name information of a caller. | 04-16-2009 |
| 20090097475 | CODEC automatic setting system of IAD and control method thereof in DSL network - A codec automatic setting system of an IAD in a DSL network includes a codec table, a codec negotiator, a codec detector, and a codec selector. The codec table stores at least one codec list. The codec negotiator incorporates an applicable codec list from the codec table into a call request message, transmits the same to a reception IAD, and receives a response message when an arbitrary terminal requests a call. The codec detector detects a codec list from the response message. The codec selector compares the codec list of the reception IAD with the codec list stored in the codec table, and selects a settable codec. An automatic codec switching function in an IAD system of a network supporting a DSL automatically switches a codec into an optimum codec according to a DSL connection band and forming and release of a VoIP call, thereby securing an optimum communication quality for an Internet phone. | 04-16-2009 |
| 20090097476 | METHOD AND APPARATUS FOR SUPPORTING VOICE COMMUNICATIONS - An apparatus comprises two processors ( | 04-16-2009 |
| 20090097477 | METHOD AND SYSTEM FOR REALIZING MEDIA STREAM INTERACTION AND MEDIA GATEWAY CONTROLLER AND MEDIA GATEWAY - A method and system for realizing media stream interaction are provided. The method includes the following steps: the MGC obtains the public network address corresponding to the media gateway MG in the private network, and the public network address is used as the remote address of the opposite side of the MG; then, the MGC sends the public network address to the opposite side; and the opposite side realizes media stream interaction with the MG in the private network by the public network address. Using this method, the media stream passing through across different IP domains in the media gateway can be realized. Also, a media gateway controller and a media gateway are provided. | 04-16-2009 |
| 20090103517 | Acoustic signal packet communication method, transmission method, reception method, and device and program thereof - When acoustic signal packets are communicated over an IP communication network, data corresponding to an acoustic signal (acoustic signal corresponding data) has been included and transmitted in a packet different from a packet containing the acoustic signal. However, conventionally, a packet in which the acoustic signal corresponding data is to be included must be determined beforehand and cannot dynamically be changed. | 04-23-2009 |
| 20090103518 | CALL ORIGINATION BY AN APPLICATION SERVER IN AN INTERNET PROTOGOL MULTIMEDIA CORE NETWORK SUBSYSTEM - A system and method for call origination by an application server in an internet protocol multimedia core network subsystem includes a first step of providing a public user identity for a user. A next step includes storing a service parameter in a service profile of the user, the service parameter indicating whether to allow/disallow the application server to initiate call requests on behalf of the public user identity. If the service parameter allows the application server to initiate call requests, the system unblocks calls originated by the application server on behalf of the user. If the service parameter disallows the application server to initiate call requests, the system blocks calls originated by the application server on behalf of the user. | 04-23-2009 |
| 20090103519 | Method and Computer Product for Switching Subsequent Messages With Higher Priority Than Invite Messages in a Softswitch - A method for switching invite messages and subsequent messages in a softswitch directing the invite messages to a first list, and the subsequent messages to a second list. The method processes the subsequent messages of the second list with a higher priority than the messages of the first list. | 04-23-2009 |
| 20090103520 | TRANSPARENT SIGNAL RELAY SYSTEM FOR PACKET TRANSMISSION SERVICES - A system for sending a data packet from a first communication network ( | 04-23-2009 |
| 20090103521 | TELECOMMUNICATION AND MULTIMEDIA MANAGEMENT METHOD AND APPARATUS - An improved communication method for sending media between a sending node and receiving node during a conversation. When network bandwidth is insufficient to transmit a full bit rate representation of time-sensitive media, then a reduced bit rate representation of the media is transmitted for the purpose of increasing the ability of the recipient to review the media upon receipt and continue the conversation in the real-time mode when the bandwidth on the network is insufficient to support the transmission of the full bit rate representation. Media that is ascertained as not time-sensitive on the other hand is transmitted when bandwidth in excess of what is needed for time-sensitive media becomes available. When the media ascertained as not time-sensitive is transmitted, the rate of transmission is adjusted at the sending node based on network conditions, the adjusted rate of transmission being set for network efficiency and reliable delivery of the media ascertained as not time-sensitive relative to the timeliness of the delivery of the media ascertained as time-sensitive. | 04-23-2009 |
| 20090103522 | TELECOMMUNICATION AND MULTIMEDIA MANAGEMENT METHOD AND APPARATUS - An apparatus for transferring a complete copy of media designated as time sensitive over a network. The apparatus includes a sending node including a network ascertaining element configured to ascertain if conditions on the network are adequate to transmit a full bit rate representation of media designated as time-sensitive at a first bit rate and a first packetization interval between the sending node and a receiving node over the network, where the full bit rate representation being derived from when the time-sensitive media was originally encoded. If the network conditions are ascertained as being inadequate, then a transmitter at the sending node generates and transmits a reduced bit rate representation of the media designated as time-sensitive. The transmitting node is also configured to receive receipt reports from the receiving node that identify the received reduced bit rate representation of the media. In response to the receipt reports, the transmitter at the sending node retransmits the corresponding full bit rate representation of the media. Eventually a complete full bit representation of the media is obtained at the receiving node after the retransmitted media is received. | 04-23-2009 |
| 20090103523 | TELECOMMUNICATION AND MULTIMEDIA MANAGEMENT METHOD AND APPARATUS - A method for transferring a complete copy of media designated as time-sensitive over a network. The method involves transmitting media designated as time-sensitive from sending node to a receiving node and receiving the media designated as time-sensitive at the receiving node. At the receiving node, any missing media designated as time sensitive is noted. One or more receipt reports are generated at the receiving node and are sent back to the sending node, the receipt reports including a low priority request for retransmission of the identified missing media. In response, the sending node retransmits the low priority request for retransmission, the retransmission occurring when bandwidth on the network in excess of what is needed to transmit time-sensitive media becomes available. Eventually a complete copy of the media including the missing media is obtained at the receiving node after the retransmission. | 04-23-2009 |
| 20090103524 | SYSTEM AND METHOD TO PRECISELY LEARN AND ABSTRACT THE POSITIVE FLOW BEHAVIOR OF A UNIFIED COMMUNICATION (UC) APPLICATION AND ENDPOINTS - A system and method to precisely learn and enforce security rules for Unified Communication (UC) applications and endpoints is disclosed. According to one embodiment, a behavioral learning system learns and abstracts positive flow behaviors of UC applications and endpoints. The properties of previously received messages from the endpoints and learned behaviors of the plurality of endpoints are stored in a database. A message from a endpoint is received by a message scanner and correlated with the AOR records in the database. The message is classified into one of a whitelist, a blacklist, and a graylist based on the results of analysis by the analysis engine. The whitelist contains the AOR records that are legitimate, the blacklist contains the AOR records that are a potential attack, and the graylist contains the AOR records that belong to neither the whitelist nor the blacklist. Based on the analysis and inspection of the message in light of the learned behaviors, a decision is made to allow, deny, quarantine or redirect the message. | 04-23-2009 |
| 20090103525 | RELEASE LINK TRUNKING FOR IP TELEPHONY - Methods and systems are provided that use resources more efficiently for calls originating and terminating in a first address space that use services in an IP address space. A call is established from an originator in a first address space to an IP device within an IP address space. The IP device sends a message to a switch in the first address space indicating a new destination in the first address space. The established call is released and a second call is established from the originator in the first address space to the new destination in the first address space. In another implementation, a first leg of a call is established to an IP device from a first address space. The IP device establishes a second leg of a call to a destination in the first address space. The calls are bridged and resources released. | 04-23-2009 |
| 20090103526 | METHOD AND APPARATUS FOR PROVIDING DISASTER RECOVERY USING NETWORK PEERING ARRANGEMENTS - The present invention enables network providers to create peering arrangements with other providers that allow them to fail over to other networks in the event of a site failure. This invention would lower the cost to provide site diversity within a provider's network by allowing cost sharing between the provider's network and other networks. For example, when an Application Server (AS) in a network fails, the network provider can send a call to a partner's network and uses an AS in the partner's network to process the call request. | 04-23-2009 |
| 20090109957 | Content Delivery During Call Setup - According to one aspect of the present invention, there is provided a system for enabling a caller terminal to establish a call with a callee terminal over a communications network such that usable communications may be entered into between users of each terminal, wherein prior to the call being established a call setup phase is entered in which signaling messages are sent over the network for the purpose of establishing the call, wherein the callee terminal is adapted for receiving content during the call setup phase as indicated in the signaling messages. | 04-30-2009 |
| 20090109958 | IDENTIFYING PHONE CALLS FOR INTERNET TELEPHONY FEATURE HANDLING BY ROUTING THE PHONE CALLS TO A SOFTSWITCH VIA A DEDICATED TRUNK - A communication system includes a plurality of time division multiplexed (TDM) public switched telephone network (PSTN) trunks, a Signal Control Point (SCP), a Softswitch, and an Internet telephony call controller. The SCP a plurality of phone numbers for time division multiplexed TDM PSTN lines with an Internet telephony feature group that has Internet telephony feature handling. The SCP routes phone calls that are directed to phone numbers in the Internet telephony feature group through at least one dedicated trunk of the PSTN to the Softswitch. The Softswitch routes phone calls that it receives on the dedicated trunk to the Internet telephony call controller for Internet telephony feature handling. | 04-30-2009 |
| 20090109959 | System and method for providing requested quality of service in a hybrid network - Telephone calls, data and other multimedia information is routed through a hybrid network which includes transfer of information across the internet. A media order entry captures complete user profile information for a user. This profile information is utilized by the system throughout the media experience for routing, billing, monitoring, reporting and other media control functions. Users can manage more aspects of a network than previously possible, and control network activities from a central site. The hybrid network also contains logic for responding to requests for quality of service and reserving the resources to provide the requested services. | 04-30-2009 |
| 20090109960 | METHOD AND APPARATUS FOR A VIRTUAL CIRCUIT DATA AREA WITHIN A PACKET DATA FRAME - A method and apparatus for a virtual circuit data area within a packet data frame is disclosed. The method may include operating ( | 04-30-2009 |
| 20090109961 | MULTIPLE SIMULTANEOUS CALL MANAGEMENT USING VOICE OVER INTERNET PROTOCOL - Illustrative embodiments provide a computer implemented method, apparatus, and computer program product for more effectively managing multiple call situations using voice over internet protocol. In one illustrative embodiment, the computer implemented method comprising, responsive to receiving a request to monitor a call from among multiple simultaneous calls using voice over internet protocol, creating a set of trigger criteria for the call and monitoring the call for the set of trigger criteria. Responsive to one of the set of trigger criteria having been met, identifying a triggered criteria and selectively invoking a rule with respect to the triggered criteria to produce a result, and notifying a requester of the result. | 04-30-2009 |
| 20090109962 | Method and apparatus for dynamically allocating and routing telephony endpoints - A client application sends a message to an endpoint server. The message contains parameters including allocation constraints. The endpoint server allocates an endpoint, such as a PSTN number, according to those constraints. The endpoint server maintains the state of the endpoint in a database. Various authentication and credentialing information is included in the database. The endpoint server configures a route through a proxy PBX for the endpoint, and returns the endpoint to the client. When desired, the client releases the endpoint by sending another message to the endpoint server to de-allocate the endpoint. The endpoint server removes the route to the endpoint, and marks the endpoint as available in the database. | 04-30-2009 |
| 20090109963 | Apparatus, method, and computer program product for registering user address information - The storage unit stores therein authentication IDs that are used for authentication of users and address information in association with one another. The authentication processing unit receives from a PC an authentication message that includes an authentication ID and is used for the authentication of the user of a communication terminal, and performs authentication on the user based on the received authentication message. The SIP address acquiring unit acquires from the storage unit address information that corresponds to the authentication ID included in the authentication message when the user is authenticated. The SIP address registering unit sends the SIP location server a registration request for registering the acquired address information as the address information of the user of the IP telephone terminal associated with the PC that transmits the authentication message. | 04-30-2009 |
| 20090109964 | APPARATUS AND METHOD FOR PLAYOUT SCHEDULING IN VOICE OVER INTERNET PROTOCOL (VoIP) SYSTEM - A method and an apparatus for playout scheduling in a Voice over Internet Protocol (VoIP) system are provided. The method includes acquiring Pulse Code Modulation (PCM) samples by decoding a received packet; setting a first scale ratio according to a length of PCM samples stored in a playout buffer based on a preset scale ratio table; setting a second scale ratio by predicting a packet delay; setting a final scale ratio using the first scale ratio and the second scale ratio; and adjusting the length of the acquired PCM samples at the final scale ratio. | 04-30-2009 |
| 20090116474 | TERMINAL, METHOD, AND COMPUTER PROGRAM PRODUCT FOR REGISTERING USER ADDRESS INFORMATION - The transferring unit transfers a message received from a PC or a network to a designated destination address. The judging unit judges whether an authentication message including the identification information of the user and a grant message indicating that the user is authenticated are transferred. The identification information acquiring unit acquires the identification information from the transferred authentication message. The SIP message processing unit creates a registration message that includes the address information of the user having the acquired identification information and transmits the created registration message to the SIP server when the grant message is transferred. | 05-07-2009 |
| 20090116475 | SYSTEM AND METHOD FOR INTER-PROCESSOR COMMUNICATION - A means for reliable inter-processor communication in a multi-processor system is described. In accordance with one aspect, a specially-configured serial bus is used as a general-purpose data link between a first processor and a second processor. The serial bus may be an Inter-IC Sound (I | 05-07-2009 |
| 20090116476 | METHOD FOR FORWARDING AND STORING SESSION PACKETS ACCORDING TO PRESET AND/OR DYNAMIC RULES - A system and method for recording and/or monitoring data by forwarding it, with or without analyzing or otherwise filtering the data itself are provided. According to embodiments of the invention, the system and method are operative over IP networks. According to an embodiment of the invention, there is provided a system and method for forwarding data according to at least one characteristic of the data, such as the session's metadata for example, without analyzing or otherwise filtering the data itself. According to another embodiment of the invention, before the data is forwarded to the recording device, pre-processing algorithms are performed according to a system preset or according to one or more rules. | 05-07-2009 |
| 20090122785 | VoIP ADAPTER, IP NETWORK DEVICE AND METHOD FOR PERFORMING ADVANCED VoIP FUNCTIONS - A VoIP adapter for POTS a phone comprises: a POTS phone connector, an IP network interface, two sets of signaling senders, signaling receivers, media senders and media receivers for the POTS phone and the IP network respectively, and a controller for controlling the operations of above components. The VoIP adapter enables the user to carry out VoIP communications using a normal POTS phone and further enables use of advanced VoIP functions via the normal POTS phone, such as Call Hold, Call Transfer, Ad Hoc Conference, etc. | 05-14-2009 |
| 20090122786 | SIGNALING METHOD IN IP TELEPHONE SYSTEM , IP TELEPHONE SYSTEM, AND IP TELEPHONE DEVICE - There is provided a signaling method for an IP telephony system ( | 05-14-2009 |
| 20090122787 | ALERT FOR ADDING CLIENT DEVICES TO A NETWORK - In one embodiment, a method of configuring a network connectivity device comprises receiving a network association request from a prospective client device requesting access to a network and sending an alert signal to a control device to cause the control device to emit an audible signal indicative of a network association request from a prospective client device. A prompt message is sent to the control device to cause the control device to provide instructions on providing feedback to the network connectivity device. The network connectivity device receives feedback from the control device and permits or denies access to the network responsive to the feedback from the control device. | 05-14-2009 |
| 20090122788 | Wireless Communication System - A disclosed wireless communication system for realizing a packet switching communication and a circuit switching communication between a mobile station and a fixed station is provided. In this system, said fixed station is configured to, when the packet switching communication is established between said mobile station and said fixed station and the line quality of that packet switching communication deteriorates to a level equal to a predetermined level or less, attempt to establish the circuit switching communication with said mobile station; and said mobile station is configured to, when the packet switching communication is established between said mobile station and said fixed station and said mobile station receives a call from the fixed station on the circuit switching communication, establish the circuit switching communication with the fixed station by responding to that call and then disconnect the packet switching communication with the fixed station. | 05-14-2009 |
| 20090122789 | APPARATUS AND METHOD FOR FILE TRANSFER USING IMS SERVICE IN A MOBILE COMMUNICATION TERMINAL - An apparatus and method for file transfer using IMS service in a mobile communication terminal is provided. In the method, Circuit-Switched (CS) call is connected with a corresponding terminal through a CS network. Capability information of the corresponding terminal is acquired according to performing a Session Initiation Protocol (SIP) capability negotiation process with an IP Multimedia Subsystem (IMS) proxy server through a Packet Switched (PS) network. Files are transferred to the corresponding terminal through a File Transfer Protocol (FTP) service according to using the acquired capability information of the corresponding terminal. | 05-14-2009 |
| 20090122790 | VOICE COMMUNICATION METHOD AND SYSTEM IN UBIQUITOUS ROBOTIC COMPANION ENVIRONMENT - A voice communication method and system that enables establishing a voice communication in a URC environment is provided. The voice communication method and system of the present invention allow establishing a voice communication channel between terminals (private IP address to private IP address, public IP address to public IP address, and private IP address to public IP address) in a URC environment. Particularly, the voice communication system of the present invention is implemented with a call server acting as a STUN server for supporting voice communication between two terminals of which one is assigned private IP address and controlling the communication session in the URC environment. | 05-14-2009 |
| 20090122791 | METHOD AND APPARATUS FOR SELECTIVE RECOVERY FROM BRANCH ISOLATION IN VERY LARGE VOIP NETWORKS - A digital telecommunications system, a method of reconnecting branches to a softswitch in a communications network and a program product for reconnecting branches to a softswitch in a communications network. A softswitch manages communications between devices at network endpoints, e.g., session initiation protocol (SIP) devices. When a branch is disconnected from the softswitch, the softswitch manages reconnects, prioritizing reconnects when multiple branches request reconnecting. | 05-14-2009 |
| 20090122792 | GUIDANCE CONFIRMATION APPARATUS AND METHOD - A guidance confirmation apparatus includes a pseudo terminal which belongs to an IMS network including a CSCF server which performs call/connection processing of an SIP terminal, an HSS server which permits use of a registered SIP terminal in the IMS network, and an MRF system which includes sound sources for generating various kinds of guidance. Upon receiving a test connection command to instruct a guidance connection test from the SIP terminal, the pseudo terminal is registered in the HSS server as a virtual terminal usable in the IMS network and connects the SIP terminal to a sound source corresponding to a guidance number based on the test connection command. A guidance confirmation method is also disclosed. | 05-14-2009 |
| 20090122793 | Method And System For Establishing Emergency Call - A method for establishing an emergency call includes: if an emergency call request message sent by a User Equipment (UE) contains an Internet Protocol Multimedia Subsystem Public User Identity (IMPU) in a TEL URI format, a Proxy-Call Session Control Function entity (P-CSCF) generates an IMPU in a Session Initiation Protocol (SIP) URI format according to the IMPU in the TEL URI format, sends both IMPUs to a Public Safety Answering Point (PSAP), and receives an emergency callback initiated by the PSAP. The PSAP initiates the emergency callback according to one of the two IMPUs. A system for establishing an emergency call includes a UE, a P-CSCF and a PSAP. The PSAP can always acquire the IMPU in the TEL URI format and the IMPU in the SIP URI format of the UE, and initiate an emergency callback to the UE according to the IMPU in the SIP URI format. | 05-14-2009 |
| 20090122794 | PACKET NETWORK AND METHOD IMPLEMENTING THE SAME - The invention provides a packet network with enhanced service filter criteria, including: a service filter criteria library, configured to store and generate a service filter criterion for a user; a service control point, configured to provide a service to the user; and a service trigger point, configured to obtain the service filter criterion from the service filter criteria library, and determine, according to the service filter criterion, whether a currently processed SIP communication needs to be triggered to the service control point or processed locally, wherein the service filter criterion includes at least one of the following: a message body other than an SIP message Session Description, a session state, a message event other than an SIP initial request message, time, user presence information, a user state, a service invocation message, related information for another criterion, a virtual application server address, filter criterion validity, processing of service invocation result, a service identification, and a ban criterion for service control point invocation. The invention further provides a service triggering method for implementing the service triggering in a packet network. | 05-14-2009 |
| 20090129369 | APPARATUS AND METHOD FOR SUPPORTING MULTIPLE TRAFFIC CATEGORIES AT A SINGLE NETWORKED DEVICE - An apparatus comprising a first and a second functional entity operable for supporting traffic in, respectively, first and second traffic categories across a communications network. The second traffic category is associated with specific routing requirements. A network interface releases a request for a first address and a request for a second address. The request for a second address comprises data that is instrumental in causing the second address to be assigned by an address-assigning entity from a particular set of at least one address. The network is pre-configured to route traffic destined for a given address in the particular set of at least one address in accordance with the specific routing requirements. Receipt of the first address from the address-assigning entity enables the first functional entity to act as a receptor of traffic in the first traffic category, while receipt of the second address enables the second functional entity to act as a receptor of traffic in the second traffic category. | 05-21-2009 |
| 20090129370 | Voice-Over-IP Capable Sideshow Device - A Voice-over-IP capable SideShow device is disclosed. Specifically, according to one embodiment of the present invention, a SideShow device capable of supporting Voice-over-Internet-Protocol (VoIP) includes a modifiable content endpoint and a virtual UART. The content endpoint enables the SideShow device to support a set of configurable functions and associate customized events with the set of configurable functions for the SideShow device to display customized graphical user interface. The virtual UART facilitates the accessing of hardware resources in the SideShow device. | 05-21-2009 |
| 20090129371 | Method and system to enable mobile roaming over ip networks and local number portability - A method and system for creating a virtual roaming solution for a MSISDN using a softphone over an IP network. The system involves (i) implementation of a novel virtual mobile network (VMN) comprising virtual visitor location register (vVLR), virtual home location register (vHLR) and virtual multiple switching centre (vMSC) on an IP server responsible for managing IP call traffic administration, and (ii) implementation of a novel mobile to internet gateway (MIG) comprising an VoIP gateway for diverting call traffic from the mobile network to the IP network, and an IP server with vMSC functionality to translate routing information from the VMN to GSM network so as to appear to the GSM network as a traditional mobile operator. The system dynamically registers the subscriber to the IP network, and provides valid routing information to the MSC (Mobile Switching Centre) or public telephone switch to route the call over to the NGN (next generation network) operator in the IP space. | 05-21-2009 |
| 20090129372 | IMS AND SMS INTERWORKING - Providing for inter-working between SMS network architectures and IMS network architectures in a mobile environment is described herein. By way of example, a next generation (NG) short message service center (SMSC) is provided that can receive SMS messages in mobile application protocol (MAP) and convert such messages to IMS protocol. In addition, the NG SMSC can also receive IMS data and convert the IMS data to an SMS MAP message. The NG SMSC can reference an IMS or an SMS location registry to determine a location of the target device, and convert from IMS to SMS MAP, and vice versa, as suitable. Accordingly, the NG SMSC can provide an efficient interface between legacy SMS and NG IMS network components while preserving legacy protocols associated with such networks. | 05-21-2009 |
| 20090129373 | EXTRACTION OF SUBSCRIBER TERMINAL INFORMATION SIGNAL - A first terminal sends a monitoring execution information for monitoring an information signal sent from a subscriber terminal, together with subscriber identification information, to a subscriber information storage apparatus. The subscriber information storage apparatus sends the subscriber identification information to a call control apparatus which is searched for based on the subscriber identification information. The call control apparatus sends the subscriber identification information to a network band managing apparatus which is searched for based on the subscriber identification information. The network band managing apparatus sends the subscriber identification information and a physical port number and a TCP/UDP port number which are searched for based on the subscriber identification information to a transmission apparatus which is searched for based on the subscriber identification information. The transmission apparatus extracts an information signal which uses the physical port represented by the physical port number and the TCP/UDP port represented by the TCP/UDP port number, and sends the extracted information signal together with the subscriber identification information to a second terminal. | 05-21-2009 |
| 20090129374 | SYSTEM AND METHOD FOR USING EXCEPTION ROUTING TABLES IN AN INTERNET BASED TELEPHONE CALL ROUTING SYSTEM - In a Voice Over Internet Protocol (VoIP) system for completing telephone calls over the Internet, the system uses a general routing table and client exception routing tables to instruct originating gateways about how to complete calls. When a call request for a particular client is received, the system first looks to that client's exception routing table to see if routing information for the call is available. If so, the system will use the routing information in the client's exception routing table to complete the call. The routing information in the client's exception routing table could include information about preferred destination gateways and/or preferred Internet Service Providers. If the client's exception routing table does not contain information that could be used to route the call, then the system simply uses the routing information in the general routing table. In some situations, the system could utilize multiple general routing tables. Likewise, a single client could have multiple exception routing tables. | 05-21-2009 |
| 20090135806 | ENABLING AD-HOC DATA COMMUNICATION OVER ESTABLISHED MOBILE VOICE COMMUNICATIONS - In one embodiment, a first PC may receive a trigger to establish a data communication session with a second PC over an established voice call between first and second phones over a WAN. In response, the first PC may discover the first phone as an authorized personal area network (PAN) device, and may establish a first PAN communication session between the first PC and the first phone. A request may then be transmitted to the second phone over the established voice call to establish the data communication session between the first and second PCs, and in response, the second phone may discover the second PC as an authorized PAN device from the second phone. A second PAN communication session may thus be established between the second phone and the second PC, and data may be exchanged between the PCs using the PAN communication sessions and the established voice call. | 05-28-2009 |
| 20090135807 | PERSISTENT SCHEDULING OF HARQ RETRANSMISSIONS - Briefly, in accordance with one or more embodiments, HARQ retransmissions may be persistently scheduled so as to efficiently allocate network without requiring the HARQ retransmissions to be scheduled for every frame or nearly every frame. Furthermore, grouping of users may occur using a bitmap for the HARQ retransmissions using the same bitmap as used for scheduling of the original packet transmission or using a separate bitmap for the HARQ retransmissions. In the event one or more scheduled HARQ retransmissions are not needed, the base station is capable of reallocating the previously scheduled resources. | 05-28-2009 |
| 20090135808 | LINE TERMINATION ARRANGEMENT WITH COMBINED BROADBAND AND NARROWBAND SERVICES - A combined line termination arrangement ( | 05-28-2009 |
| 20090135809 | METHOD AND APPARATUS FOR ESTABLISHING A VOICE BEARER IN A TELECOMMUNICATIONS SYSTEM - A method for establishing a voice bearer in a telecommunications system in which packetized voice traffic is carried over a user plane, includes negotiating at least one of header compression and voice multiplexing options for a call and, in the negotiation process, using information about the options that is not sent over the user plane during transmission of voice traffic over the user plane. The information may be sent by signaling out of band and not via the user plane, thus not impacting on transmission of voices data. In another method, option information from a previous call or calls between the same IP addresses is used to set up the options for a new call without any additional signaling. | 05-28-2009 |
| 20090135810 | Device to terminate a modem relay channel directly to an IP network - A modem data aggregating gateway that supports modem relay functionality for permitting reliable switching of modem traffic between a VoIP network and a data packet switch Internet Protocol (IP) network, s.a. the Internet. The modem relay aggregator may receive modem data encapsulated as Voice over IP (VoIP) data packets in accordance with a Simple Packet Relay Transport (SPRT) mechanism. The packet data may be error corrected and/or decompressed before being repackaged for forwarding to the ultimate destination. In the event that the destination is itself an IP device, the modem relay aggregator may forward the packets directly over the IP network. As a result, if the destination of a modem call is an IP device (such as a Web site or other Internet-enabled device) the technique eliminates two points from a processing path in which digital signal processing (DSPs) would otherwise have to perform modem protocol processing. Otherwise, minimal modem reformatting can be performed at the aggregation point. | 05-28-2009 |
| 20090135811 | HYBRID PACKET-SWITCHED AND CIRCUIT-SWITCHED TELEPHONY SYSTEM - A hybrid telephony system with packet switching as well as circuit switching optimizes utilization of transport networks, and is accessible from any conventional telephone set. A call originating from a circuit-switched network is passed through a gateway computer to a backbone packet-switched network, and then through a second gateway computer to a second circuit-switched network where it terminates. The voice of both the originating party and the terminating party is converted to data packets by the near-end gateway computer and then converted back to voice by the far-end gateway computer. In an alternative scenario, the originating party uses a computer on the packet-switched network, which replaces the originating circuit-switched network and the originating computer. Powered by CPUs, DSPs, ASICs disks, telephony interfaces, and packet network interfaces, the gateway computers may have media conversion modules, speech processing modules and routing resolution modules, and are capable of translating telephony call signaling as well as voice between circuit-switched and packet-switched networks. Optionally, the gateway computers may also have analog trunking modules, MF and DTMF digit modules and special services modules, in order to support analog circuit-switched networks and secure telephone calls. | 05-28-2009 |
| 20090135812 | CALL TRANSFER METHOD AND COMMUNICATION SYSTEM - The present disclosure discloses a call transfer method and a communication system, and the call transfer method includes: transferring, by a source CTI platform, a call of a user terminal processed by a source traffic resource device to a target traffic resource device upon a call hang and transfer request of the source traffic resource device, and setting the source traffic resource device in a suspended state according to a call processing response message provided from a target CTI platform; and receiving a call release notification message sent from the target CTI platform after the target traffic resource device processes the call, setting the source traffic resource device in an active state according to the call release notification message, and transferring the call of the user terminal back to the source traffic resource device. Embodiments of the disclosure may reduce a coupling degree between the platform and the service side in the call transfer service and hence improve reliability of the system. | 05-28-2009 |
| 20090135813 | TRANSMITTING MESSAGES IN TELECOMMUNICATIONS SYSTEM COMPRISING A PACKET RADIO NETWORK - A method of transmitting messages in a telecommunication system includes a first network offering circuit-switched services, a second network offering packet-switched services, and at least one mobile station supporting the first and the second network. When the need arises to transmit at least one message, a check is made to see if the mobile station is attached to the second network. The message is transmitted to the second network if the mobile station is attached to the second network. The message is transmitted to the first network in case of a failure to transmit the message via the second network. | 05-28-2009 |
| 20090141703 | SYSTEMS AND METHODS FOR CARRIER ETHERNET USING REFERENTIAL TABLES FOR FORWARDING DECISIONS - The present invention utilizes specific referential tables for forwarding decisions while maintaining current mechanisms of Ethernet addressing and QoS marking. The referential tables are utilized for forwarding decisions based on any and/or multiple fields within the packets simultaneously, such as, for example, incoming port number, incoming MAC, incoming VLAN, outgoing MAC, outgoing VLAN, P-bits, DSCP, MPLS label, TCP/UDP port numbers, IP, SIP, HTTP, and the like. A user can define the forwarding criteria based on any combination/permutation fields in the packet. Advantageously, the present invention removes the need to introduce explicit tunnel labels in the Ethernet frame in order to maintain the desired QoS within the network removing explicit labeling requirements. | 06-04-2009 |
| 20090141704 | Hybrid Protocol Voice Over the Internet Calling - A click to talk system for use in a data network is disclosed. In response to a user selection on a browser, a click to talk server bridges an IP capable voice device to the browser by translating between data network protocols. Additionally, a media server may be manually or automatically contacted to provide a media stream simultaneously with a voice connection between a client computer running the browser and the IP capable voice device. | 06-04-2009 |
| 20090141705 | Device and method for address-mapping - To perform address mapping, a configuration client determines port numbers required for a network service and a network address conversion unit converts external network addresses into internal network addresses and vice versa. A configuration server requests required port numbers from the network address conversion unit which directly provides the network service with an external network address with the required port number. A device located in an internal address domain can thus be allocated a unique external network address. | 06-04-2009 |
| 20090141706 | SYSTEM AND METHOD FOR THE AUTOMATIC PROVISIONING OF AN OPENLINE CIRCUIT - A system and method for the automatic provisioning of an openline circuit, specifically for use with a network management system, the system comprising a web GUI (Graphical User Interface) and middleware, wherein the middleware may be synchronized with and instructs a prior art network management system is disclosed. The GUI can be accessed by personnel in an IT department, or by dealers over the Internet via their dealer board using “secure sockets” to be able to make the changes to their openline circuits, including the provisioning of a new openline circuit, the various different inputs being processed by the middleware. The middleware manipulates many SQL (Structured Query Language) databases and is adaptable to work with any network management platform and further is able to provide billing information. | 06-04-2009 |
| 20090141707 | SYSTEMS AND METHODS FOR PROVIDING EMERGENCY SERVICE TRUST IN PACKET DATA NETWORKS - A method and apparatus for providing in a packet data telecommunication network serving one or more end terminals and/or Mobile Stations (MSs), a method for establishing, managing, modifying, and terminating an End-to-End (E2E) Emergency Service (ES) Chain-of-Trust (CoT) from an Access Serving Network (ASN) and Connectivity Service Network (CSN) to a PSAP, PSAP proxy, or PSAP (i.e. PSTN) gateway that results in the creation of a trust relationship amongst the components in the established ES CoT necessary to allow or validate the granting of any unauthenticated or unprovisioned ES network access and ES operation establishment, modification, and termination requests from amongst the components in an ES CoT to assist a particular terminal/MS or ES network component attempting to establish an ES session between the ES user agent of the terminal/MS and a serving PSAP. | 06-04-2009 |
| 20090141708 | VOIP ANALOG TELEPHONE SYSTEM - A multi-port VoIP telecommunications system that allows the user to gain access to telephone connectivity through the Internet by connecting directly to the Internet or by connecting to the Internet through the existing Internet connection of a computer or cell phone device. | 06-04-2009 |
| 20090141709 | METHOD FOR INITIATING INTERNET TELEPHONE SERVICE FROM A WEB PAGE - A direct telephone dialing scheme for initiating internet telephone service from a web page is provided. The scheme allows a caller, using an internet telephone service, to place telephone call to a telephone number appearing on any web page directly from that web page. In one embodiment, a caller navigates to a desired web page on the internet and the caller dials a telephone number on that web page directly to initiate a two-way audio communication with the destination telephone number using an internet telephone service. The direct telephone dialing scheme of the present invention improves the accessibility and ease of use of internet telephone services. Furthermore, the direct telephone dialing scheme can be used with video, data, and fax communications which are supported by the VoIP data communication standard. | 06-04-2009 |
| 20090147771 | Mobile Communication Device Providing Integrated Access to Telephony and Internet - Provided are devices, methods, communication managers and user interface solutions that enable access to multiple services from a mobile communications device. A mobile communication device that provides telephony services via a PSTN also includes multiple communication channels that exploit packet data transfer via an IP network, for example enabling VoIP, instant messaging and other internet-based communication services to be initiated from a mobile telephone. | 06-11-2009 |
| 20090147772 | SYSTEMS AND METHODS FOR PROVIDING PRESENCE INFORMATION IN COMMUNICATION - A method for facilitating communication between at least a first user who uses a first device and a second user who uses a second device. The method may include associating possible device states with possible presence states. The possible device states pertain to the first device, and the possible presence states pertain to the first user. The method may also include determining a device state of the first device. The method may also include setting a communication presence state of the first user to be a first presence state if the device state is a first device state and setting the communication presence state of the first user to be a second presence state if the device state is a second device state. The method may also include providing information concerning the communication presence state of the first user to at least the second device. | 06-11-2009 |
| 20090147773 | Intelligent end user devices for clearinghouse services in an internet telephony system - Clearinghouse services architectures that support the use of end user devices, such as personal computers, Internet Protocol (IP) phones, cable multimedia terminal adapters, and residential gateways, in an Internet telephony system. The innovative architectures include a proxy-based system model, a direct communication model, and a hybrid proxy/direct communication model. A user can operate an “intelligent” end user device. i.e., a device running a client program with knowledge of the architecture particulars, to access a clearinghouse service on an IP network. This enables the user to communicate a telephony call over the IP network and via the combination of a terminating gateway identified by the clearinghouse service and the Public Switched Telephone Network. | 06-11-2009 |
| 20090147774 | Multimedia interactive telephony services - In a multimedia interactive telephony system, a voice service server generates dynamic content intended for consumption by a communication device. The dynamic content is sent to a gateway where it is transformed from to an intermediate content format appropriate for rendering at the communication device. The user may interact with the transformed dynamic content rendered on the communication device, causing the arguments to be sent to the voice server, thus allowing user interactivity with the voice service. The voice services server may also generate dynamic content for simultaneous consumption by multiple communication devices, each of which may independently render an intermediate content format appropriate to it. The voice services server may also generate the dynamic content for the communication device while the communication device is not currently engaged in an active call. | 06-11-2009 |
| 20090147775 | Ancillary data support in session initiation protocol (SIP) messaging - A SIP ancillary data server provides host to auxiliary data for an emergency SIP session (call) uniquely referred to in a transported SIP header. In a manner similar to how location is represented in an emergency call, a SIP header is extended. The extended SIP Header contains one of two possible types of content elements: either (a) a content pointer element to a SIP Message body part (a “cid:”, or content identifier); or (b) an (a.k.a, “info_URI” in this document). | 06-11-2009 |
| 20090154448 | TERMINAL EQUIPMENT OF COMMUNICATION SYSTEM AND METHOD THEREOF - Disclosed is a transmitting and receiving apparatus and method in a communication system. The transmitting and receiving apparatus and method can provide a data service for exchanging user data including characters, images, computer files, messages, etc. as well as voice over a voice physical channel for providing a voice service in a wireless communication system including IS-95A/B, CDMA 1x, GSM and W-CDMA and in a communication system including a voice service for providing a VoIP service through a wired/wireless packet network. That is, the transmitting and receiving apparatus and method can provide a data service which transfers user data information while a voice service is provided or plays a game etc. during a call. | 06-18-2009 |
| 20090154449 | TELEPHONE SYSTEM, AND MAIN UNIT AND TERMINAL REGISTRATION METHOD THEREFOR - According to one embodiment of the invention, there is provided a telephone system comprises a plurality of telephone terminals and a main unit. The main unit comprises an authentication processing unit performs login authentication MAC address authentication, and a mode specification unit receives specification of a plural terminal registration mode. The MAC address authentication refuses logins from telephone terminals differing in MAC address from a telephone terminal that has been allowed to log in firstly even if the logins are made by the same extension numbers. The plural terminal registration mode exclusively allows the simultaneous login by the same extension numbers from a plurality of telephone terminals having different MAC addresses. The authentication processing unit gives priority over the plural terminal registration mode higher than the MAC address authentication and makes the MAC address authentication void in the plural terminal registration mode. | 06-18-2009 |
| 20090154450 | SERVICE DELIVERY METHOD, SERVICE EXECUTION METHOD, PC AND SWITCH - The present invention provides a Computer Supported Telecommunications Applications (CSTA) protocol-based service delivery method, which includes: obtaining a switching function service that carries voice parameters; and sending the switching function service that carries voice parameters down to the switch side. The present invention also provides a CSTA-based service execution method, a Personal Computer (PC), a switch, and a voice service system. In the present invention, when the PC sends a switching function service to the switch, voice parameters can be carried in the switching function service. In this way, when the switch obtains the switching function service with voice parameters, the switch can play voice according to the voice parameters. Therefore, the present invention enables play of voice without modifying the existing service process, thus, simplifying the implementation. | 06-18-2009 |
| 20090161656 | METHOD AND SYSTEM FOR FRAME SIZE ADAPTATION IN REAL-TIME TRANSPORT PROTOCOL - A system and method for adapting circuit-switched payload transport between a mobile station, MS, ( | 06-25-2009 |
| 20090161657 | METHOD AND APPARATUS FOR ASSURING VOICE OVER INTERNET PROTOCOL SERVICE - In one embodiment, the present invention is a method and apparatus for assuring Voice over Internet Protocol service. In one embodiment, a system for assuring Voice over Internet Protocol service includes a performance management platform for collecting performance management data from a plurality of sources in a Voice over Internet Protocol network, for detecting at least one abnormal event in accordance with the collected performance management data, and for reporting a volume of traffic in the Voice over Internet Protocol network and a trouble ticketing system for generating a ticket identifying a root cause of the abnormal event(s). | 06-25-2009 |
| 20090161658 | Method for selecting VOIP call path to monitor - The disclosed invention enables a user to select a Voice over IP (VOIP) call path to monitor. In particular, a user interface presents data regarding nodes within a VOIP network. The user may select between different possible configurations to monitor, including fully meshed whereby every site including a test probe router is connected to every other site; hub-and-spoke in which a subset of the sites are designated by the user as hubs connected to every other site or spokes connected only to hubs; or a custom configuration in which the user selects which individual call paths to monitor. Embodiments of the present application provide a tool that accepts the user's selections and implements the commands needed to define the desired VOIP network nodes to be monitored, preferably by configuring IP SLA or other tools to provide synthetic data in the selected VOIP call path, and then measuring the performance of the network elements in the selected call path when transferring the synthetic data. | 06-25-2009 |
| 20090161659 | ON-CHIP APPARATUS AND METHOD OF NETWORK CONTROLLING - An apparatus and method of controlling an on-chip network is provided. An apparatus for controlling a network includes an arbiter which generates a switch control signal based on first route information received from a first router, and a switch which receives from the first router a first data packet associated with the first route information, controls at least one output port according to the switch control signal during a first time interval, and outputs the first data packet via the at least one controlled output port during a second time interval. | 06-25-2009 |
| 20090161660 | SYSTEM, METHOD, AND RECORDING MEDIUM FOR SCHEDULING PACKETS TO BE TRANSMITTED - A system for scheduling packets to be transmitted is provided. The system includes a soft delay bound calculator module and a frame determination module. The soft delay bound calculator module calculates a soft delay bound for a non-real-time packet based on a packet size of the non-real-time packet and a minimum reserved traffic rate of a channel. The frame determination module determines whether a real-time packet must be transmitted at a current frame according to a delay bound, a transmission time, and a possible retransmission time thereof, and whether a non-real-time packet must be transmitted at a current frame according to a soft delay bound, a transmission time, and a possible retransmission time thereof. Thus, it is possible to improve the performance of the system while keeping the QoS thereof in a mixed service environment. | 06-25-2009 |
| 20090161661 | METHOD, SYSTEM AND SOFTWARE FOR ESTABLISHING A COMMUNICATION CHANNEL OVER A COMMUNICATIONS NETWORK - The establishment of a VoIP connection between first and second telecommunication devices ( | 06-25-2009 |
| 20090161662 | POWER MANAGEMENT SYSTEMS AND METHODS FOR ELECTRONIC DEVICES - Power management systems and methods for use in an electronic device are provided. The system comprises a baseband processing unit, a wireless communication module, and an application processing unit. The baseband processing unit connects to a base station via a communication network, thereby enabling the electronic device equipped with a communication capability. The wireless communication module receives a data packet via an Internet, and determines whether the data packet conforms to a packet pattern. If so, the wireless communication module transmits a wake-up signal to the application processing unit. In response to the wake-up signal, the application processing unit enters a normal state from a sleep state, and performs an application operation in the normal state according to the data packet. | 06-25-2009 |
| 20090161663 | METHOD AND SYSTEM FOR SERVERLESS VOIP SERVICE IN PERSONAL COMMUNICATION NETWORK - Method and system for supporting serverless VoIP service are provided. Network information of a first device and a second device is exchanged through a telecommunication network. A VoIP connection between the first and second devices can be established through an internet based network according to the exchanged network information. The network information may comprise an IP address and a port number, and can be delivered by short message service. | 06-25-2009 |
| 20090161664 | SYSTEM AND METHOD FOR INSTANT VoIP MESSAGING - There is provided an instant voice messaging system (and method) for delivering instant messages over a packet-switched network, the system comprising: a client connected to the network, the client selecting one or more recipients, generating an instant voice message therefor, and transmitting the selected recipients and the instant voice message therefor over the network; and a server connected to the network, the server receiving the selected recipients and the instant voice message therefor, and delivering the instant voice message to the selected recipients over the network, the selected recipients being enabled to audibly play the instant voice message. | 06-25-2009 |
| 20090161665 | SYSTEM AND METHOD FOR INSTANT VoIP MESSAGING - There is provided an instant voice messaging system (and method) for delivering instant messages over a packet-switched network, the system comprising: a client connected to the network, the client selecting one or more recipients, generating an instant voice message therefor, and transmitting the selected recipients and the instant voice message therefor over the network; and a server connected to the network, the server receiving the selected recipients and the instant voice message therefor, and delivering the instant voice message to the selected recipients over the network, the selected recipients being enabled to audibly play the instant voice message. | 06-25-2009 |
| 20090168754 | Systems and methods for WiMAX and 3GPP interworking by using GGSN - Embodiments include systems and methods for interoperability between WiMax and 3GPP systems reusing a GGSN. Embodiments comprise implementing GTP functions within the ASN of a WiMAX system to enable communication of data between the ASN Gateway (ASN-GW) of the ASN and the GGSN of the 3GPP system. Embodiments also implement a WiMAX AAA server in the 3GPP system to enable communication of control information between ASN-GW and the 3GPP system. | 07-02-2009 |
| 20090168755 | Enforcement of privacy in a VoIP system - Systems and methods for providing privacy in a VoIP system are provided. In exemplary embodiments, an incoming call is received. A caller ID associated with the incoming call is determined. A category based on the caller ID is associated with the incoming call. Based on the category, a call treatment database is accessed to determine at least one call treatment associated with the category. The at least one call treatment is then applied to the incoming call. | 07-02-2009 |
| 20090168756 | System, Method and Apparatus for Clientless Two Factor Authentication in VoIP Networks - The present invention provides a system, method and apparatus for authenticating an Internet Protocol (IP) phone and a user of the IP phone by determining whether the IP phone is an authorized device, and whenever the IP phone is authorized and a trigger condition occurs, determining whether the user of the IP phone is authorized. The user authorization process initiates a call to the IP phone, sends a request for a passcode to the IP phone, sends a message to disable the IP phone whenever the passcode is invalid, and terminates the call. The user authentication process uses an in-band channel and the IP phone does not run a two factor authentication client application during the authentication process. | 07-02-2009 |
| 20090168757 | TRANSPARENTLY ROUTING A TELEPHONE CALL BETWEEN MOBILE AND VOIP SERVICES - Systems and methods are provided for routing a telephone call intended for a communications device between a mobile network and a VOIP service, where the mobile network and VOIP service may be connected through the PSTN. The VOIP service may receive telephone calls and may direct the telephone calls to the communications device through the Internet when a stable Internet connection is present, and may route telephone calls to the mobile network through the PSTN otherwise. When a call is routed to the mobile network, the mobile network may make the call the communications device to establish a telephone connection through a cellular link. While a telephone call is in progress, the VOIP service and communications device may be configured to seamlessly switch the telephone call to a different service depending on the status of the communications device's Internet connection. | 07-02-2009 |
| 20090168758 | METHODS FOR FACILITATING COMMUNICATION BETWEEN INTERNET PROTOCOL MULTIMEDIA SUBSYSTEM (IMS) DEVICES AND NON-IMS DEVICES AND BETWEEN IMS DEVICES ON DIFFERENT IMS NETWORKS AND RELATED ELECTRONIC DEVICES AND COMPUTER PROGRAM PRODUCTS - A bridge device is used to setup communication sessions between Internet Protocol (IP) Multimedia Subsystem (IMS) devices and non-IMS devices or between non-IMS devices over an IMS network. Once a communication session is established, the IMS Bridge device may translate messages received from the respective endpoint devices between IMS and non-IMS formats. | 07-02-2009 |
| 20090168759 | METHOD AND APPARATUS FOR NEAR REAL-TIME SYNCHRONIZATION OF VOICE COMMUNICATIONS - A method and system for synchronizing in real-time the voice media of a conversation conducted over a network between a first communication device and a second communication. The method includes at each of the first and second communication devices progressively storing in first and second storage elements and transmitting the voice media created using the first and second communication devices to the other communication device respectively. Both the first and second communication devices store in the first and second storage elements the progressively received media from the other device respectively. A mechanism to continually review, ascertain and request the media stored in the first storage element, but not the second storage element, and vice-versa is provided to ensure that the two storage elements contain the same voice media. As a result, the first and second storage elements each maintain real-time synchronized copies of the voice media of the conversation respectively. | 07-02-2009 |
| 20090168760 | METHOD AND SYSTEM FOR REAL-TIME SYNCHRONIZATION ACROSS A DISTRIBUTED SERVICES COMMUNICATION NETWORK - A method for progressively synchronizing stored copies of indexed media transmitted between nodes on a network. The method includes progressively transmitting available indexed media from a sending node to a receiving node with a packet size and packetization interval sufficient to enable the near real-time rendering of the indexed media, wherein the near real-time rendering of the indexed media provides a recipient with an experience of reviewing the transmitted media live. At the receiving node, the transmitted indexed media is progressively receive and any indexed media that is not already locally stored at the receiving node is noted. The receiving node further continually generates and transmits to the sending node requests as needed for the noted indexed media. In response, the sending node transmits the noted indexed media to the receiving node. Both the sending node and the receiving node store the indexed media. As a result, both the sending node and the receiving node each have synchronized copies of the indexed media. | 07-02-2009 |
| 20090168761 | SIGNALING GATEWAY, NETWORK SYSTEM AND DATA TRANSMISSION METHOD - A signaling gateway can handle the signaling communication of SS7 among the STP in PSTN and a plurality of nodes using different types of SIGTRAN protocols in the IP network. The signaling gateway includes a routing function unit which discriminates SIGTRAN protocol for each of the nodes using routing information contained in a received SS7 message from the STP, and a plurality of protocol units, each being provided for corresponding type of SIGTRAN protocol to be used for each of the nodes. The routing function unit outputs a data transfer request to the protocol unit which corresponds to the discriminated SIGTRAN protocol. The protocol unit constructs a corresponding protocol format of SIGTRAN protocol using data contained in the data transfer request and sets an originating IP address to the same value regardless the SIGTRAN protocol and a port number corresponding to the type of SIGTRAN protocol, and requests signal transmission. | 07-02-2009 |
| 20090168762 | Method and System for Setting Up a Voice Connection - Method for setting up a voice connection between a first terminal set (T | 07-02-2009 |
| 20090168763 | APPARATUS AND METHOD FOR MANAGING CHANNEL CAPACITY AND DECT BASE STATION FOR THE SAME - Wireless channel capacity of a digital enhanced cordless telecommunication (DECT) base station may be expanded by selecting an unused timeslot from a plurality of timeslots of a downlink channel. A dummy bearer may be created in a corresponding timeslot and it may be determined whether a traffic bearer is created in an unused timeslot. If so, the dummy bearer may be removed, and dummy bearer information may be periodically transmitted to a handset through a traffic bearer on every frame. Dummy bearer information may be inserted into a header of the traffic bearer, and the previously created dummy bearer may be removed after the insertion is completed. Usage of the timeslot occupied by the dummy bearer may be changed to voice communication. When the traffic bearer is released, the dummy bearer may be re-created, and dummy bearer information may be periodically transmitted to the handset through the re-created dummy bearer on every frame. | 07-02-2009 |
| 20090168764 | CALL CONTROL ELEMENT CONSTRUCTING A SESSION INITIATION PROTOCOL (SIP) MESSAGE INCLUDING PROVISIONS FOR INCORPORATING ADDRESS RELATED INFORMATION OF PUBLIC SWITCHED TELEPHONE NETWORK (PSTN) BASED DEVICES - A Session Initiation Protocol (SIP) message adapted for use by a multi-media services provider system to form a multi-media communication path between at least a calling communication device adapted to operate using a first protocol (e.g. SIP) and at least a destination communication device adapted to operate using a second protocol, such as Integrated Services Digital Network User Part (ISUP). The SIP message includes a header region having a number of header fields, a first body region having Session Description Protocol (SDP) information related to the calling communication device and a second body region having ISUP related addressing information associated with the destination communication device. | 07-02-2009 |
| 20090168765 | ENDPOINT SELECTION FOR A CALL COMPLETION RESPONSE - Techniques for selecting a call completion response from a group of call completion responses based on weights associated with the call completion responses, are provided. A server processes a call invitation for a callee by forwarding the call invitation to each of the callee's endpoints. Each of the callee's endpoints associates a weight to its call completion response it generates to accept or reject the call invitation. The server waits to receive the call completion responses from each of the callee's endpoints or for a predetermined period of time (i.e., a timeout), and uses the weights associated with the received call completion responses to decide which of the received call completion responses to use to complete the call invitation. | 07-02-2009 |
| 20090175262 | VOIP With Internet Access - Systems and methods allow an analog phone line to concurrently carry both non-packetized data from a telephone handset and packetized data from a computer to a common network access number. An access device is provided to connect the handset and the computer to a common phone line, which provides both VoIP connectivity and Internet access. Since the access number is located within the user's local calling area, the inventive system avoids long distance charges that would otherwise be applied to Internet connectivity and long distance phone calls. | 07-09-2009 |
| 20090175263 | APPARATUS, AND ASSOCIATED METHOD, FOR INFORMING DEDICATED MODE CONNECTED MOBILE STATION OF PACKET SERVICE CAPABILITIES IN A COVERAGE AREA - An apparatus, and an associated methodology, for identifying to a circuit-switch-connected mobile station with an indication of packet-service capabilities available to a mobile station. A message generator at the network generates a message that includes an indication of the network-entity capabilities with respect to packet communications. A field of the message identifies the packet-service capabilities. A message is sent by the network and detected by a detector of the mobile station. A report is formed indicative of the value contained in the delivered message, and a user display displays an indication of the detected information. | 07-09-2009 |
| 20090175264 | User interface - A method of initiating a communication event via a communication system at a communication device comprising storing a plurality of memory items, wherein each memory item is associated with a user of the communication system; selecting a first set of memory items from said plurality of memory items in accordance with a predetermined selection method; displaying the first set of memory items as a first set of icons, wherein each icon represents at least one memory item and receiving a selection signal associated with one of said icons from the user of the communication device to initiate the communication event with the user of the communication system associated with the memory item represented by the selected icon. | 07-09-2009 |
| 20090175265 | Message Routing in the IP Multimedia Subsystem - A method of routing a SIP message within an IP Multimedia Subsystem, where the message originates at an IMS/SIP client attached to a visited network and contains as its destination address a TEL URI including a telephone number. The method comprises, at the IMS/SIP client, specifying that the telephone number is a local number of the visited network or of a home network of the client, including at the IMS/SIP client a phone-context within the TEL URI identifying the home network or the visited network according to the specification, and delivering the message from the client to the home network and receiving the message within the home network, and routing the message according to the phone-context contained within the TEL URI. | 07-09-2009 |
| 20090175266 | METHOD OF TAKING ACCOUNT OF QUALITY OF SERVICE BETWEEN DISTINCT IP TELEPHONY DOMAINS, AND A CORRESPONDING LOCATION SERVER AND COMPUTER PROGRAM - A method for propagating at least one route for at least one digital stream between a first location server of a first IP telephony domain and a second location server of a second IP telephony domain. The first location server belongs to a first autonomous system and the second location server belongs to a second autonomous system. The method includes sending digital stream routing update messages to the second location server. The update messages contain information for managing quality of service, and, prior to being propagated towards the second server, the information is updated by the first server. The information includes at least one of the following: information about a quality of service component associated with at least one autonomous system, referred to as a system component; and information about a quality of service component associated with at least one IP telephony domain, referred to as a domain component. | 07-09-2009 |
| 20090175267 | METHOD OF PROPAGATING IP CONNECTIVITY INFORMATION BETWEEN DISTINCT IP TELEPHONY DOMAINS, AND A CORRESPONDING LOCATION SERVER AND COMPUTER PROGRAM - A method is provided for propagating at least one route for at least one digital stream between a first location server of a first IP telephony domain and a second location server of a second IP telephony domain, the first location server belonging to an autonomous system, and the route for transferring the at least one digital stream. The method includes a stage of propagating at least one identification relating to the autonomous system of the first location server towards the second server. | 07-09-2009 |
| 20090175268 | METHOD, DEVICE AND SYSTEM FOR COMMUNICATION - A communication system adapted to be connected to a calling device through the Internet includes a proxy device and a plurality of communication devices. The proxy device receives messages sent from the calling device through the Internet. Each of the communication devices has specific media processing capability, and receives the messages sent by the calling device through the proxy device and the Internet. The proxy device and the communication devices store the media processing capabilities of the communication devices, and upon receipt of a message requesting connection from the calling device, select one of the communication devices with the media processing capability matching that required by the connection according to the media processing capabilities of the communication devices stored therein. The selected communication device sets up a connection with the calling device through the proxy device, or selects another communication device to set up the connection. | 07-09-2009 |
| 20090175269 | VOICE RELAYING APPARATUS AND VOICE RELAYING METHOD - A voice relaying apparatus includes a receiving section for receiving a cell from an asynchronous transfer mode (ATM) network, a plurality of cell assembling/disassembling units for assembling and disassembling the cells, and a transmitting section for transmitting the cells assembled by each of the plurality of cell assembling/disassembling units. Each of the plurality of cell assembling/disassembling units is composed of a cell disassembling section for disassembling the cell received by the receiving section, a detecting section for detecting whether or not the voice relaying apparatus is carrying out a relay switch operation, and a cell assembling section for assembling the cell disassembled by the cell disassembling section and then sending to the transmitting section, if the fact that the voice relaying apparatus is carrying out the relay switch operation is detected by the detecting section. | 07-09-2009 |
| 20090175270 | TELEPHONE RECORDING AND STORING ARBITRARY KEYSTROKES SEQUENCE WITH REPLAY WITH A SINGLE STROKE - A telephone is described that allows any arbitrary combination of key strokes, including numerical keys, extension keys, as well as function keys such as TRANSFER, CONFERENCE, etc., to be programmed such that the entire sequence of key strokes can be recalled with the touch of a single button. The phone can be programmed directly by operation of the telephone user interface on the phone (i.e., the keys, phone display, and speaker prompting the user) and a program button dedicated to the feature of programming a separate programmable button to map to the specified key sequence. The feature can be implemented in advanced telephones capable of voice over Internet Protocol networks, and supporting the Session Initiation Protocol. In these more advanced phones, the programming can be done by a system administrator or by the user of the phone via a computer with internet access. | 07-09-2009 |
| 20090180467 | System and Method for Connecting Remote Callers with PBX Extensions Using Internet Telephony - A telecommunications system configured to provide access to a company's directory via a simple-to-use client software program; to integrate directory access with Internet telephony call establishment (click to dial); to automate the dialing of DTMF tones to connect to a specific extension; and to provide searching of the company's directory. | 07-16-2009 |
| 20090180468 | Universal plug and play method and apparatus to provide remote access service - Disclosed are a universal plug and play (UPnP) method and a UPnP apparatus providing remote access service. The method includes receiving external inputs of an identifier of a remote access server (RAS) to generate a credential and a session initiation protocol (SIP) identifier of the RAS, generating a payload of a SIP packet including a credential identifier (ID) generated based on the identifier of the RAS, remote access transport agent (RATA) capability information, and a transport address (TA) set corresponding to candidate IP addresses to access a remote access client (RAC), and transmitting the SIP packet to the RAS. | 07-16-2009 |
| 20090180469 | IP COMMUNICATION APPARATUS - An IP communication apparatus employed in a telephone voice/moving picture recording system is comprised of: an IP packet transmitting/receiving I/F for transmitting/receiving an IP packet; an IP address acquiring unit of acquiring an IP address corresponding to a transmission source of the IP packet; a signal judging unit for performing a signal judging operation by employing data contained in an IP packet; a recording unit for recording the data in relation to the IP address based upon a judgement result made by the signal judging unit; and a recording control unit for controlling the recording unit. | 07-16-2009 |
| 20090180470 | EFFICIENT INTERWORKING BETWEEN CIRCUIT-SWITCHED AND PACKET-SWITCHED MULTIMEDIA SERVICES - Techniques for signaling a packet size limitation of a circuit-switched terminal to a packet-switched terminal during a multimedia session such as a multimedia telephony session. In one aspect, an interworking node obtains information from the circuit-switched terminal during call setup, and signals to a packet-switched terminal that another end of the telephony session is a circuit-switched terminal. In a further aspect, the interworking node signals to the packet-switched terminal a maximum packet size limitation negotiated with the circuit-switched terminal. Further techniques for the packet-switched terminal to accommodate the maximum negotiated packet size to minimize data reformatting by the interworking node are described. | 07-16-2009 |
| 20090185552 | Audio/Visual Information Dissemination System - Systems and methods for providing information to the public by way of publicly accessible devices. A network of video displays are deployed at publicly accessible locations such as inside public transportation vehicles or at public transportation stations. The video feed to these video displays are provided by a video distribution hub which receives the video feed from a network hub. Different audio feeds are accessible to end users or by telephone. End users can call a telephone interface which receives and routes audio feeds from an audio distribution hub. End users can access audio feeds which may be synchronized with a video feed to provide a complete audio visual experience to the end user. For more useful content, the video displayed at any location may be adjusted to be relevant to the area where the video display is deployed. Audio content synchronized to one of these disparate video feeds can be accessed by the end user by dialing different options through the telephone interface. Audio feeds not tied to a specific video feed, such as radio stations or themed audio feeds, may also be accessed by the end user through the telephone interface. | 07-23-2009 |
| 20090185553 | TELEPHONY SYSTEM - In IP telephony systems, it has become impossible to detect the location of installation of a telephone terminal from the telephone number, since an IP telephone terminal can be installed in an arbitrary location. Also, even if one observes the calling party number presentation at the time of an incoming call, it has become impossible to grasp from where the calling party is placing the outgoing call. It is possible that, within an IP telephony system, a terminal location detection means is configured and the installation location of a telephone terminal is detected simultaneously with the registration of the telephone terminal. In addition, the problem can be solved by configuring, in an IP telephony server, a device of reporting location information about the correspondent to the telephone terminal and by configuring, in the telephone terminal, a device of displaying the received positional information. | 07-23-2009 |
| 20090185554 | Method of managing a call addressed to a terminal associated to an access device - A method of managing a call addressed to a first terminal operating in a telephone system, which includes a mobile network, a packet-switched network and an access device allowing connection of dual mode terminals to the packet-switched network, wherein the method includes: a) providing configuration information by associating information related to a set of terminals to an identifier of the access device, the set of terminals including a dual mode terminal, which is adapted to operate in the mobile network and in the packet-switched network; b) providing status information related to the at least one dual mode terminal; c) upon reception of a request for the call, checking whether the first terminal belongs to the set of terminals; d) in the affirmative, routing the call to at least one selected terminal of the set of terminals, the selection being performed based on the configuration information and the status information; and d) in the negative, routing the call to the first terminal. | 07-23-2009 |
| 20090185555 | METHOD AND DEVICE FOR PROVIDING MULTIMEDIA DATA WHEN ESTABLISHING A TELEPHONE CALL - The invention concerns a method and a device for providing multimedia data when setting up a telephone call. The terminal being connected via Internet to a platform ( | 07-23-2009 |
| 20090185556 | Method and apparatus for controlling telephone calls using a computer assistant - Systems and methods for monitoring, making, managing and controlling telephone communications with a computer call assistant with an integrated voice/data communications system are disclosed. A call assistant computer application preferably runs on a personal computer (“PC”) coupled to the integrated system over a packet bus. The call assistant exchanges control and/or status packets with the integrated system preferably over a packet bus. The call assistant enables the user to make, receive and control telephone calls, monitor the status of the user's extension, voice mail, etc., and preferably operates with integrated systems capable of transmitting and receiving voice and data in multiple modes. In preferred embodiments, the computer call assistant operates with systems that are capable of multiple native mode voice and data transmissions and receptions with a communications system having a multi-bus structure, including, for example, a time division multiplexed (“TDM”) bus, a packet bus, and a control bus, and multi-protocol framing engines, preferably including subsystem functions such as PBX, voice mail, file server, web server, communications server, telephony server, LAN hub and data router. | 07-23-2009 |
| 20090185557 | Method and Device for Selecting Service Domain - A method and device for selecting a service domain in a system for setting a session for at least one or more services between at least two or more terminals, in which when receiving an INVITE message for setting a voice service related session from an originating terminal, it is decided whether to send the INVITE message to a server or to directly send the INVITE message to a terminating terminal according to user pre-registered information for a domain selection, and then the terminating terminal having received the INVITE message sends to a network a response message including domain selecting information for directly selecting a domain with respect to the voice service related session according to a user's selection. | 07-23-2009 |
| 20090185558 | IP converged system and call processing method thereof - A call processing method in an Internet Protocol (IP) converged system includes: requesting an incoming call to be routed through an IP network; checking a data traffic-processing state of a traffic manager in response to the request; and rerouting the call through the IP network or rerouting the call through a Public Switched Telephone Network (PSTN) according to the checked data traffic-processing state. | 07-23-2009 |
| 20090190573 | SYSTEM AND METHOD OF PROVIDING IMS SERVICES TO USERS ON TERMINATING NON IMS DEVICES - Disclosed is a network-based device in an IP Multimedia Subsystem (IMS) that provides IMS services to terminating non-IP devices. The method embodiment includes receiving a REGISTER message that initiates registration of a Public User Identity (PUID) at a terminating non-IP device at a network-based device in an IMS, wherein the terminating non-IP device is specified as being at an E.164 routing address, establishing, in a network device that accepts communications regarding where to send sessions destined for a specific PUID and provide information regarding where to send sessions, the E.164 routing address as a final destination of sessions to the PUID based on the registration and an IP address of a network device that includes routing functionality based on telephone numbers as an immediate destination for the session and using this information from the registration to route to the terminating non-IP device after providing IMS services by using the relationship between the PUID, the E.164 routing address and the IP address of the network device that includes routing functionality based on telephone numbers that completes the routing of messages to the terminating non-IP device that was established during registration. | 07-30-2009 |
| 20090190574 | TELEPHONE FEATURE SELECTION BASED ON FEATURES RECEIVED FROM A SERVICE PROVIDER - Implementations described herein may provide for VoIP phone or server devices, where the phones include visual menus through which a user of the phone can modify options or features relating to the user's account. In one implementation, the phone may receive a data structure defining features relating to communication services for the phone device. The phone may parse the data structure to obtain a menu corresponding to the features and present the menu to a user of the phone device. | 07-30-2009 |
| 20090190575 | PACKET CAPTURING APPARATUS, PACKET CAPTURING METHOD AND PACKET CAPTURING PROGRAM - A packet capturing apparatus and method are provided. The packet capturing apparatus includes an acquisition part acquiring a voice packet having voice information, after receiving each packet transferred in a network, from received packets. The packet capturing apparatus also includes a measuring part measuring an elapsed time after receiving the acquired voice packet and an acquisition part starting to receive a voice packet transferred in a direction opposite to a transfer direction of the acquired voice packet when the elapsed time reaches a predetermined standby time. | 07-30-2009 |
| 20090190576 | DATA PROCESSING METHOD AND SYSTEM - A method of managing an IP call between a calling party and a called party, the method comprising receiving, at a gateway, a request to set up the call from the calling party; determining, from the request, a requirement to route the call through an interceptor; forwarding the request from the gateway to the interceptor; setting up an IP call between the interceptor and the called party; setting up an IP call between the interceptor and the calling party; and operating the interceptor as a back-to-back user agent (B2BUA) between the calling party and the called party. | 07-30-2009 |
| 20090190577 | Providing Session Initiation Protocol Request Contents Method and System - An embodiment provides a user equipment that includes a processor configured to receive a Session Initiation Protocol (SIP) NOTIFY message transmitted by a network component as a result of a registration event. The SIP NOTIFY message contains at least a portion of information included in a first SIP message sent between a first user equipment and the network component. Another embodiment provides method and apparatus for a network node to determine whether filter criteria include one or more indicators that specify the need for information, and including in a second SIP message the information specified by the one or more indicators. | 07-30-2009 |
| 20090190578 | Routing Methods and Systems Using ENUM Servers - A method of processing a Voice over Internet Protocol (VoIP) call is disclosed. The method includes receiving a Uniform Resource Identifier (URI) associated with a destination telephone number from a telephone number mapping (ENUM) server associated with a third service provider. The method also includes receiving an Internet Protocol (IP) address of a Session Initiation Protocol (SIP) server associated with a second service provider in response to a query by a first service provider to a Domain Name Service (DNS) server. The query is based on the URI. Additionally, the method includes contacting the SIP server using the IP address of the SIP server to set up a bearer path of the VoIP call. | 07-30-2009 |
| 20090196281 | ENERGY STAR compliant Voice over Internet Protocol (VoIP) telecommunications network including ENERGY STAR compliant VoIP devices - A Voice over Internet Protocol (VoIP) communications system, a method of managing a communications network in such a system and a program product therefore. The system/network includes an ENERGY STAR (E-star) aware softswitch and E-star compliant communications devices at system endpoints. The E-star aware softswitch allows E-star compliant communications devices to enter and remain in power saving mode. The E-star aware softswitch spools messages and forwards only selected messages (e.g., calls) to the devices in power saving mode. When the E-star compliant communications devices exit power saving mode, the E-star aware softswitch forwards spooled messages. | 08-06-2009 |
| 20090196282 | METHODS AND APPARATUS FOR PROVIDING QUALITY-OF-SERVICE GUARANTEES IN COMPUTER NETWORKS - An arbitration mechanism provides quality of service guarantees for time-sensitive signals sharing a local area computer network with non-time-sensitive traffic. Device adapters are placed at all access points to an Ethernet network. The device adapters limit admission rates and control the timing of all packets entering the network. By doing so, collisions are eliminated for timesensitive traffic, thereby guaranteeing timely delivery. A common time reference is established for the device adapters. The time reference includes a frame with a plurality of phases. Each of the phases is assigned to a device adapter. Each device adapter is allowed to transmit packets of data onto the network only during the phase assigned thereto. The length of the phases may be modified in accordance with the number of packets to be transmitted by a particular device adapter. A master device adapter may be appointed to synchronize each of the device adapters. | 08-06-2009 |
| 20090196283 | MULTI-CHANNEL GENERATING SYSTEM ON WIRED NETWORK - The present invention relates to a multi-channel provision system for a wired network. The multi-channel provision system of the present invention includes a tap-off unit ( | 08-06-2009 |
| 20090196284 | Method for Providing an Emergency Call Service for VoIP Subscribers - A requirement for an emergency call is to reach the nearest emergency call centre and to transmit information to said centre concerning the location of the caller. However, a pre-defined assignment of the subscriber number of the VoIP subscriber is not necessarily given, as a VoIP subscriber can use the telephone service from any Internet connection. To solve this problem, the emergency call service is implemented by the local VoIP service provider responsible in the roaming network of the VoIP subscriber. | 08-06-2009 |
| 20090196285 | METHOD AND APPARATUS FOR PROVISIONING DUAL MODE WIRELESS CLIENT DEVICES IN A TELECOMMUNICATIONS SYSTEM - Method and apparatus for provisioning a wireless multi-modal client device in a telecommunications system includes determining a provisioning environment within and a provisioning condition under which the client device is operating, determining a state of a configuration file of the client device and obtaining an updated configuration file based on the provisioning environment and provisioning conditions. The provisioning environment is determined by detecting one or more wireless networks accessible by the client device such as an IP-based network and a PSTN-based network. Connection to the IP-based network is made via WiFi and to the PSTN-based network via GSM/GPRS. The wireless networks defining the provisioning environment have characteristic timing intervals for configuration file updating. There is also an incremental timer that determines the elapsed time for each characteristic timing interval and switches between characteristic timing intervals depending upon the provisioning environment that the client device is operating within. | 08-06-2009 |
| 20090196286 | Domain Transfer Method, Server and Controller - The present embodiments disclose a domain transfer method, a server and a controller. The domain transfer method includes: receiving a call request from a terminal, where the request carries a session transfer identifier allocated by a server in advance for identifying the session and domain transfer of the session; and transferring the session to another domain according to the session transfer identifier. With the present invention, domain transfer is based on a dynamically allocated Session Transfer Identifier (STID) so as to guarantee the correctness and effectiveness of domain transfer and promote the diversification of network services. Network resources are saved and the efficiency of domain transfer is higher. | 08-06-2009 |
| 20090201910 | APPARATUS AND METHOD TO HANDLE DYNAMIC PAYLOADS IN A HETEROGENEOUS NETWORK - Various embodiments provide an apparatus and method for handling dynamic payloads in a heterogeneous network. An example embodiment includes a first node interface to receive a first request for data communication from a first node, the first request being coded in a first protocol and including information identifying a first payload type. The example embodiment includes a second node interface to receive a second request for data communication from a second node, the second request being coded in a second protocol and including information identifying a second payload type. The first node interface of the example embodiment configures a message coded in the first protocol to include the information identifying the second payload type and to send the message to the first node. | 08-13-2009 |
| 20090201911 | Highly Scalable Internet Protocol-Based Communications System - A highly scalable Internet Protocol (IP) based communications system which provides voice and other communication services to end-users. The instant system incorporates a unique architecture which simplifies scaling of services to hundreds of thousands and even millions of subscribers. The instant system architecture includes a means for directly connecting a plurality of peered service providers thereby obviating the need to move the communications across the PSTN. By bypassing the PSTN, the instant system can leverage the advantages of IP-based networks in meeting subscriber communications needs such as, without limitation, quality of service, service up-time, and advanced feature sets. Bypassing the PSTN also allows the peered partners to negotiate communications rates between themselves, without incurring PSTN carrier fees. | 08-13-2009 |
| 20090201912 | METHOD AND SYSTEM FOR UPDATING THE TELECOMMUNICATION NETWORK SERVICE ACCESS CONDITIONS OF A TELECOMMUNICATION DEVICE - A system is provided for updating the conditions under which a telecommunication device accesses services provided by a telecommunication network. The system includes a network access point through which the device accesses the network, and a database, wherein the system authenticates the device via the access point on the basis of authentication data transferred by the device as well as the database storing the profile associated with the authentication data. The access point controls the conditions under which the device accesses the network services once the device has been authenticated and on the basis of the device profile. The system generates a second authentication command for the device via the access point following an alteration of the profiled associated with the authentication data. | 08-13-2009 |
| 20090201913 | Learning the Expiry Time of an Address Binding Within an Address Translation Device for an Sip Signaling Server - A signaling server (SS) comprising means for transmitting SIP signaling messages with a client (T) through a NAT address translation device temporarily binding a public address to the client's private address, including means for receiving registration messages from the client and for sending the client a validity duration, at the end of which it must transmit a new registration message. The invention resides in the fact that if the client is located behind an address translation device, it determines an approximate expiry time for the temporary binding by successively sending test messages after an increasing wait time until the termination of the binding is detected. This approximate time is then used by being transmitted as the SIP validity period. | 08-13-2009 |
| 20090201914 | METHOD AND SYSTEM FOR AUTHORIZATION CHECKING WHEN SETTING UP A CONNECTION - Present-day on-line charging mechanisms make it possible, in a non-blocking application, for the setting up of a connection to be allowed initially even though it is not yet clear, or cannot be clarified at this time whether a prepaid subscriber (SIP client) still has sufficient money in his account. This therefore leads to a call being signalled to a subscriber (Subs) to be called (it rings), and the call must then be cleared if there is not enough money in the account. In order to avoid this “ghost ringing”, but nevertheless to ensure that connections are set up quickly, a method is proposed which includes an address element, for example the last digit, being withheld, until authorization is obtained from the On-line Charging System (OCS). Alternatively, a connection is set up with a prior condition of “do not ring”, in accordance with IETF RFC 3212. The invention can also be used for other non-monetary authorizations, for example for “lawful interception”. | 08-13-2009 |
| 20090201915 | INTERNET NETWORK COMMUNICATIONS SYSTEM AND A METHOD OF PUTTING A COMMUNICATIONS UNIT INTO COMMUNICATION WITH AN INTERNET NETWORK - A method of putting a communications unit into communication with an Internet network in which the communications unit calls the services platform, the services platform identifies the calling line, and an operating system is automatically installed in the communications unit enabling a connection to be provided with an Internet network and enabling navigation on the Internet network to be performed, and automatically enabling the operation of the operating system to be tracked and proceeding with updating and repair operations. | 08-13-2009 |
| 20090201916 | METHOD, SYSTEM AND APPARATUS FOR VERIFYING VALIDITY OF LOCATION INFORMATION IN A PACKET-SWITCHED NETWORK - According to embodiments of the present invention, there are provided a method, system and apparatus for determining validity of location information in a packet-switched network. A method for determining if location information associated with an endpoint in a packet-switched network is valid, the location information having been stored in a memory, comprises obtaining an access device identifier associated with an access device responsible for handling a communication session between the endpoint and the packet-switched network. The access device identifier is then compared with a last known access device identifier associated with the endpoint to enable determining if the location information is valid. | 08-13-2009 |
| 20090201917 | PRAGMATIC APPROACHES TO IMS - Embodiments of the invention provide systems and methods for providing services such as provided by Internet Protocol (IP) Multimedia Subsystem (IMS) with an IP network that is not the IMS. According to one embodiment, a system for providing communication services can comprise a communication network, one or more subsystems communicatively coupled with the network and adapted to provide one or more telco functions, and one or more applications communicatively coupled with the network and adapted to utilize the telco functions. | 08-13-2009 |
| 20090201918 | METHOD FOR INITIATING INTERNET TELEPHONE SERVICE FROM A WEB PAGE - A direct telephone dialing scheme for initiating internet telephone service from a web page is provided. The scheme allows a caller, using an internet telephone service, to place telephone call to a telephone number appearing on any web page directly from that web page. In one embodiment, a caller navigates to a desired web page on the internet and the caller dials a telephone number on that web page directly to initiate a two-way audio communication with the destination telephone number using an internet telephone service. The direct telephone dialing scheme of the present invention improves the accessibility and ease of use of internet telephone services. Furthermore, the direct telephone dialing scheme can be used with video, data, and fax communications which are supported by the VoIP data communication standard. | 08-13-2009 |
| 20090201919 | System for providing hosted telephone services to a subscriber via the internet - An Internet controlled telephony system employing a host services processor connected to a subscriber via the Internet and further connected to the public switched telephone system (PSTN). The subscriber employs a web interface to populate a database with preference data which is used by the host services processor to handle incoming calls and establish outgoing telephone connections in accordance with the preference data provided by the subscriber. Incoming calls to a telephone number assigned to the subscriber may be automatically forwarded to any telephone number specified by the preference data. The subscriber may also use the web interface to specify whether call waiting is to be activated, to screen or reroute calls from designated numbers, for recording voice mail messages in designated voice mailboxes, for selectively playing back voice mail messages via the web interface or for forwarding voice mail as an email attachment, for handling incoming fax transmissions using character recognition and email attachment functions, and for automatically paging the subscriber when incoming voice mail, fax or email messages are received, all in accordance with the preference data supplied by the subscriber using the web interface. Outgoing connections and conference calls may be initiated using the web interface, and the subscriber may block the operation of caller identification functions. Call progress information may be visually displayed to the subscriber during calls by transmitting web pages from the host services computer to the subscriber's web browser. | 08-13-2009 |
| 20090201920 | Enhancing voice QoS over unmanaged bandwidth limited packet network - An improved telephony adapter compresses voice data, creates IP packets, and prioritizes the voice IP packets over the data IP packets. Preferably, the compression and packetization interval is such that the bandwidth occupied by the voice IP packets is approximately half of the minimum average available bandwidth in the upstream direction, thereby maintaining acceptable latency and voice quality of the speech. Further enhancement is achieved by causing the ISP to also give priority to voice packets that are destined to the telephony adapter, over the data packets that are destined to the telephony adapter. | 08-13-2009 |
| 20090201921 | WIDE AREA COMMUNICATION NETWORKING - A communications network is disclosed and includes a broadband communication line having a first derived voice channel and a second derived voice channel, wherein the first and second derived voice channels are established as a function of an available bandwidth associated with the broadband communication line. The communication network further includes a residential gateway in communication with the broadband communication line. The residential gateway includes a switch, a network interface device in communication with the switch, and wherein the switch is configured to select at least one of the first or second derived voice channels for voice communication over the broadband communication line as a function of the available bandwidth. | 08-13-2009 |
| 20090201922 | METHOD FOR CHANGING SESSION MEDIA, METHOD FOR ESTABLISHING A CALL, AND EQUIPMENT THEREOF - A method for changing ICS session media includes: receiving a media type change request including a new media type sent from a terminal equipment or a MSC, releasing a CS call leg based on an original media type between an ICCF and the terminal equipment, establishing a CS call leg based on the new media type between the ICCF and the terminal equipment, and updating a media type of a second call leg between the ICCF and a second party into the new media type; or, receiving a media type change request including a new media type sent from a second party, updating a media type of a second call leg between an ICCF and the second party into the new media type, releasing a CS call leg based on an original media type between the ICCF and a terminal equipment, and establishing a CS call leg based on the new media type between the ICCF and the terminal equipment. | 08-13-2009 |
| 20090207833 | EFFICIENT KEY SQUENCER - A method includes for determining a plurality of fields of a packet associated with a routing of the packet, wherein each field of the plurality of fields includes one or more bits. Arranging the bits of the plurality of fields into a plurality of ordered partitions of a search sequence, the search sequence being associated with a plurality of searches, wherein the searches are based on the bits included in one or more of the ordered partitions. Providing, to a routing table including routing information associated with the routing of the packet, one or more of the ordered partitions of the search sequence, wherein the routing table is structured based on the search sequence. Receiving, based on the plurality of searches, the routing information associated with the routing of the packet from the routing table. Routing the packet based on the routing information. | 08-20-2009 |
| 20090207834 | TRANSMITTING A PACKET FROM A DISTRIBUTED TRUNK SWITCH - A method of transmitting an upstream communication packet from a distributed trunk (DT) switch is described. The method comprises receiving a packet from a device connected to a DT port of the DT switch; and transmitting the received packet via a non-DT port of the DT switch if the DT switch is the owner of the device and transmitting the received packet via a DT interconnect (DTI) port of the DT switch if the DT switch is not the owner of the device. | 08-20-2009 |
| 20090207835 | Enterprise Collection Bus - Systems and methods are presented to collect raw data from a plurality of servers and nodes on a network. A Distributed Enterprise Collection Bus (DECB) architecture is employed at various points on a network. The DECB comprises a collector unit that is protocol agnostic, an orchestration unit, a rule database, a filtering unit, and a distribution unit. Packets of raw data such as Call Detail Records (CDRs) generated by switching centers are received, and distributed to relevant destinations. Relevant destinations include data warehouses, mediation, analytics, etc. The goal is to alleviate collection and filtration duties of the source and destination. | 08-20-2009 |
| 20090207836 | TRANSMISSION METHOD AND APPARATUS - A voice packet transmission method and apparatus for transmitting a voice packet with a header, wherein a voice packet with a compressed header is transmitted, monitoring is performed to detect whether a necessity to send a voice packet with an uncompressed header is generated during the transmission, the voice packet data with an uncompressed header is divided into a plurality of portions when the necessity is generated, and each divided data is transmitted via different antennas by spatial multiplexing. | 08-20-2009 |
| 20090213834 | Technique for coordinating CS and PS registrations in a multi-operator core network - A technique for coordinating the registration of a terminal (UE) in circuit-switched (CS) and packet-switched (PS) domains of a multi-operator core network (MOCN) with multiple core networks (CN) is described. According to a method approach, a notification message indicating the necessity of coordinated CS and PS registrations for a terminal (UE) is received from a first core network (CN). In a next step, and based on a global permanent identity (IMSI) associated with the terminal (UE), a second core network (CN) responsible for CS and PS registrations is determined. A registration message for coordinated CS and PS registrations is then sent to the second core network (CN) that has been determined based on the global permanent identity (IMSI). | 08-27-2009 |
| 20090213835 | METHOD AND APPARATUS FOR MEASURING ONE WAY TRANSMISSION DELAY - A method and apparatus enabling the measurement of one way delay in each of the two directions of transmission from a single location are disclosed. The method measures a first roundtrip delay at a first location between a first endpoint and a second endpoint over a first communication network, and measures a second roundtrip delay between a third endpoint and a fourth endpoint over a second communication network with symmetric delay characteristics. The method performs synchronous recordings of a test signal that is sent simultaneously from the second endpoint to the first endpoint and from the fourth endpoint to the third endpoint, to measure an arrival time (t | 08-27-2009 |
| 20090213836 | WEB PAGE TELEPHONE SYSTEM - A web page telephone system uses a web page information processor to connect to telecommunication exchanges of a VoIP network and is built in with the registration data and link files for web page owners. Each of the link files corresponds to a contact label on a web page belong to the web page owner. The contact label establishes a link with the web page information processor. When any end user uses a computer browser to link to and display the web page, the link file is activated by clicking the contact label. The web page information processor completes automatic identification and finds the corresponding telecommunication exchange. Once the end user's computer sends out a dial request, the telecommunication exchange is notified to establish a VoIP network connection between the communication device of the web page owner and the end user computer. | 08-27-2009 |
| 20090213837 | SYSTEMS AND METHODS TO SELECT PEERED BORDER ELEMENTS FOR AN IP MULTIMEDIA SESSION BASED ON QUALITY-OF-SERVICE - Systems and methods to select peered border elements for a communication session based on Quality-of-Service are disclosed. In particular, an example method for peered border element assignment is disclosed, comprising determining a composite Quality-of-Service result based on a plurality of Quality-of-Service parameters associated with a communication session, querying a telephone number mapping server for a status of each of a plurality of peered border elements, and assigning the communication session to be handled by one of the plurality of peered border elements based on the composite Quality-of-Service result and the status of each of the plurality of peered border elements. | 08-27-2009 |
| 20090213838 | MESSAGE HANDLING IN AN IP MULTIMEDIA SUBSYSTEM - A Session Initiation Protocol Application Server of an IP Multimedia Subsystem having processing means for handling a message received from a Serving Call/State Control Function, the means being arranged to handle the message based upon a header of the message containing the URI of the served user, this header having been introduced by the Serving Call/State Control Function and being other than the P-Asserted Identity and the R-URI. | 08-27-2009 |
| 20090213839 | System and Method for Distributed Call Monitoring/Recording Using the Session Initiation Protocol (SIP) - The system and method described herein allows for full monitoring and recording of SIP calls by using standard SIP messages. During the call set up between a first SIP device and a second SIP device, information is derived from a first SIP INVITE message from a first SIP device. Information is then derived from a response message from the second SIP device. | 08-27-2009 |
| 20090213840 | Integrated information communication system - To provide an integrated information communication system without using dedicated lines or the Internet, ensuring communication speed, communication quality, communication trouble countermeasures in a unified manner, wherein security and reliability in communication is ensured. The system is comprised of an access control apparatus for connecting a plurality of computer communication networks or information communication equipment to each, and a relay device for networking the aforementioned access control apparatus, the system having functions for performing routing by transferring information by a unified address system, and is configured such that the aforementioned plurality of computer communication networks or information communication equipment can perform communications in an interactive manner. | 08-27-2009 |
| 20090213841 | TERMINAL AND METHOD FOR STORING AND RETRIEVING MESSAGES IN A CONVERGED IP MESSAGING SERVICE - A terminal, server and method for storing and selectively retrieving SIP-based messages, are discussed. According to an embodiment, the present invention provides a method for controlling a SIP-based message by a control server, which includes receiving a SIP-based message; determining a manner in which the SIP-based message is to be processed based on user preference information; transmitting the SIP-based message and indication information to a storage server based on the determination result, the indication information indicating if the SIP-based message is to be sent back with link information, the link information including a reference to the SIP-based message; receiving a part of the SIP-based message and the link information from the storage server; and transmitting the part of the SIP-based message and the link information to a terminal, whereby the SIP-based message can be selectively retrieved. | 08-27-2009 |
| 20090213842 | COMMUNICATION LINK ESTABLISHING METHOD - When a communication device receives a request of connection, the communication device selects a destination address from multiple destination addresses according to data provided by a server. After a destination address is selected from the multiple destination addresses, the communication device connects a source address and the selected destination address. | 08-27-2009 |
| 20090213843 | METHOD AND APPARATUS FOR ENABLING VOICE COMMUNICATION - An embodiment of the invention is directed to a method, wherein it is received, at an origin device, input from a first telephonic device via an origin telephonic landline. An initiation request is then output based on the input to a destination device, wherein the destination device is configured to output a call request to a second telephonic device via a destination telephonic landline. Another embodiment of the invention is directed to a method, including receiving, at a server, a request from an origin device, wherein the origin device is configured to receive input from a first telephonic device via an origin telephonic landline. Information is then output, based on the input, regarding a destination device to the origin device, wherein the destination device is configured to output a call request to a second telephonic device via a destination telephonic landline. | 08-27-2009 |
| 20090213844 | TELEPHONY - In one embodiment of an improvement to telephony, a solution to the problem of communicating to “the many” is made by enabling telecommunications service providers to: accept digital dialog as well as conventional dialog, enable augmented phone service to be added to conventional phone services, handle non-calls in addition to calls, and turn content into content-of-interest. | 08-27-2009 |
| 20090213845 | VOICE AND DATA EXCHANGE OVER A PACKET BASED NETWORK WITH TIMING RECOVERY - A signal processing system which discriminates between voice signals and data signals modulated by a voiceband carrier. The signal processing system includes a voice exchange, a data exchange and a call discriminator. The voice exchange is capable of exchanging voice signals between a switched circuit network and a packet based network. The signal processing system also includes a data exchange capable of exchanging data signals modulated by a voiceband carrier on the switched circuit network with unmodulated data signal packets on the packet based network. The data exchange is performed by demodulating data signals from the switched circuit network for transmission on the packet based network, and modulating data signal packets from the packet based network for transmission on the switched circuit network. The call discriminator is used to selectively enable the voice exchange and data exchange. | 08-27-2009 |
| 20090213846 | SYSTEM AND METHOD FOR VOICE OVER INTERNET PROTOCOL (VoIP) AND FACSIMILE OVER INTERNET PROTOCOL (FoIP) CALLING OVER THE INTERNET - A system and method for sending Long distance telephone calls over the Internet utilizes cost and quality of service data to optimize system performance and to minimize the cost of completing the calls. The system utilizes a network of gateways connected to the Internet. The gateways receive calls from various service providers and convert the analog calls into data packets which are then placed onto the Internet. Similarly, the gateways take data packets off the Internet, convert the data packets back into analog format, and provide the analog telephone calls to the same or another service provider. Then system periodically checks the quality of communications between each of the gateways, and uses this information, in combination with cost information, to determine how to route the calls over the Internet. Special addressing protocols can be used by a system embodying the invention to reduce or eliminate unnecessary signaling between gateways as call setup procedures are carried out. The system can also use information about calls that has been recorded in more than one location to determine how much to charge for completing a call. | 08-27-2009 |
| 20090219920 | VOICE-OVER-IP-(VOIO-) TELEPHONY COMPUTER SYSTEM - A Voice-over-IP-(VoIP-) telephony computer system includes a client computer ( | 09-03-2009 |
| 20090219921 | METHOD FOR LOCALIZATION AND LOCATION-RELATED CONNECTION OF A MOBILE VOICE-OVER-IP SUBSCRIBER TO AN EMERGENCY CALL STATION - The invention relates to a method for localization and location-related connection of a mobile voice-over-IP subscriber to an emergency call station even when the subscriber is temporarily registered in the voice-over-IP network with an address of a location other than his home address ( | 09-03-2009 |
| 20090219922 | Exchange system and server device - An exchange apparatus serves a plurality of telephone terminals and a server device is connected to the exchange apparatus via an exchange network and that, in a case in which the plurality of telephone terminals are grouped, stores in association and administers, for each group, a different single number and identification information of the at least one telephone terminal belonging to the group. When a call is originated with a predetermined single number from an originating side telephone terminal, the exchange apparatus performs a query to the server device for a connection destination corresponding to the predetermined single number, and implements a call connection between a telephone terminal of a connection destination, which is a result from the server device in response to the query, and the originating side telephone terminal. | 09-03-2009 |
| 20090219923 | Method and Apparatus for Accessing Communication Data Relevant to a Target Entity Identified by a Number String - Service resource items for use in call setup in a telephone system are held on servers that are connected to a computer network which is logically distinct from the telephone system infrastructure; this computer network may, for example, make use of the Internet. Each service item is locatable on the network at a corresponding URI and is associated with a particular telephone number. A mapping is provided between telephone numbers and the URIs of associated service resource items. When it is desired to access a service resource item associated with a particular telephone number, this mapping is used to retrieve the corresponding URI which is then used to access the desired service resource item. | 09-03-2009 |
| 20090219924 | VOICE CALL COMMUNICATION SWITCHING SYSTEM - A voice call communication switching system in which a first network and a second network are connected to each other and which includes a user equipment that establishes communication using the networks and an application server that controls communication exchanged. The application server includes a message control unit that receives a message to be sent from the user equipment when a situation is changed from one where the user equipment communicates with another user equipment using resources of the first network to another where the user equipment communicates with the another user equipment using resources of the second network and a session end requesting unit that sends a resource release instruction to disconnect the communication between the first network and the user equipment to the user equipment. The user equipment includes a resource releasing unit that sends a message indicating release of the resources to a node. | 09-03-2009 |
| 20090219925 | INTERNET PROTOCOL TELEPHONY VOICE/VIDEO MESSAGE DEPOSIT AND RETRIEVAL - A method for signaling an Integrated Messaging System (IMS) on an Internet Protocol (IP) based network to deposit a message, including the steps of sending a Session Initiation Protocol (SIP) SIP INVITE request to the IMS indicating a message deposit action; receiving a corresponding SIP menage from the IMS agreeing to participate in the message deposit action; and sending an SIP acknowledge message to the IMS confirming receipt of the corresponding SIP message; and depositing the message in a destination mailbox. A method of signaling an IMS on an IP based network to retrieve a deposited message, the method including the steps of sending a SIP INVITE request to the IMS indicating a message retrieval action; receiving a corresponding SIP message from the IMS agreeing to participate in the message retrieval action; sending an SIP acknowledge message to the IMS confirming receipt of the corresponding SIP message; and retrieving the deposited message from a mailbox corresponding to known account information. | 09-03-2009 |
| 20090219926 | CALL CONTROL METHOD AND IMS CS CONTROL APPARATUS - A call control method and an IP multimedia subsystem (IMS) circuit-switched (CS) control apparatus are disclosed. The call control method includes these steps: a terminal device and a second party set up a call through a CS call leg set up between the terminal device and an IMS CS control function (ICCF) and a second call leg set up between the ICCF and the second party; and the ICCF receives a media type change request, and rejects the change of media type for the call between the terminal device and the second party if more than one session is available on the terminal device. Embodiments of the present invention avoid call failure upon session transfer due to the change of media type in the prior art, thus improving the reliability and stability of session transfer. | 09-03-2009 |
| 20090232127 | UPD-Based Soft Phone State Monitoring for CTI Applications - A supervisor computer directly communicates, via User Datagram Protocol (UDP) packets, with a call control application software in a soft phone. The UDP packets provide real-time information, from a desktop of the soft phone, describing call activity and usage status of the soft phone. The supervisor computer is able to remotely control usage of the soft phone according to information provided by the UDP packets. | 09-17-2009 |
| 20090232128 | Method for Lawful Interception During Call Forwarding in a Packet-Oriented Telecommunication Network - The invention relates to a method for lawful interception in the case of call forwarding (AW_TlnB) in a packet-oriented telecommunications network (TK | 09-17-2009 |
| 20090232129 | METHOD AND APPARATUS FOR VIDEO SERVICES - A method for providing a multimedia service to a multimedia terminal includes establishing an audio link between the multimedia terminal and a server over an audio channel, and detecting one or more media capabilities of the multimedia terminal. The method also includes providing an application logic for the multimedia service, establishing a visual link between the multimedia terminal and the server over a video channel, providing an audio stream for the multimedia service over the audio link, and providing a visual stream for the multimedia service over the video link. The method further includes combining the video link and the audio link, and adjusting a transmission time of one or more packets in the visual stream to synchronize the visual stream with the audio stream. | 09-17-2009 |
| 20090232130 | GATEWAY ROUTER AND PRIORITY CONTROL FOR EMERGENCY CALL IN IP TELEPHONY SYSTEM - A gateway router in an IP telephony system includes a determination section which determines whether or not an arriving session control signal is of an emergency call, and a controller which dynamically assigns to emergency calls and other general calls a predetermined number of sessions that can be simultaneously connected and thereby performs control such that connection processing is carried out preferentially for a session control signal of an emergency call. | 09-17-2009 |
| 20090232131 | METHOD AND APPARATUS FOR PROVIDING EMERGENCY CALLS TO A DISABLED ENDPOINT DEVICE - The present invention enables the remote activation of a device by a packet-switched service, e.g., VoIP network service for the purposes of receiving calls identified as urgent from a pre-identified calling party when the device is disabled. The present invention enables registered users to select the calling parties they wish to receive emergency calls from. | 09-17-2009 |
| 20090232132 | COMMON MOBILITY MANAGEMENT PROTOCOL FOR MULTIMEDIA APPLICATIONS, SYSTEMS AND SERVICES - A framework of a common mobility management protocol for Q.5/16 includes a high level protocol for performing the functions of address resolution, routing, location update and authentication. The common mobility management protocol can be used by existing and future multimedia applications (MA's) to support mobility management for messaging among mobility management authentication function (AuF), home location function (HLF) and visitor location function (VLF) databases/servers, and the corresponding multimedia application functional entities (MAFEs) of the multimedia applications (MA's). The common mobility management protocol may replace, act in concert with or in sequence with existent interworking protocols for the various multimedia applications. Reference point architectures, functional characteristics, features, and capabilities of the protocol are described including call flows and message syntax. The disclosure presents the scope of Q.5/16 and how H.MMS.1 (H.323 Mobility), H.MMS.2 (Global Mobility), and H.MMS.3 (Presence/Instant Messaging Mobility) can be a part of the same common mobility management protocol. | 09-17-2009 |
| 20090238168 | COMMUNICATION NODE AND METHOD FOR HANDLING SIP COMMUNICATION - The present invention relates to a communication node and method for connecting and maintaining a call between a caller device and a callee device. The communication node comprises a session controller module and a plurality of internal SIP UA (Session Initiation Protocol User Agent) servers. The session controller module is adapted to remain in communication with the caller device and the callee device through a whole duration of the call. The plurality of internal SIP UA servers is adapted to communicate with the session controller module by open protocol. | 09-24-2009 |
| 20090238169 | CALL INTERCEPT FOR VOICE OVER INTERNET PROTOCOL (VoIP) - A device receives, from a calling party, a call to a Voice over Internet Protocol (VoIP) subscriber, and generates a request for the calling party to record information. The device also receives information from the calling party based on the request, and provides the information from the calling party and call handling options to the VoIP subscriber. The device further receives a response from the VoIP subscriber to the call handling options, and handles the call based on the VoIP subscriber response. | 09-24-2009 |
| 20090238170 | METHOD AND SYSTEM FOR PROVIDING VOICE OVER IP (VOIP) TO WIRELESS MOBILE COMMUNICATION DEVICES - A wireless voice over Internet Protocol (VOIP) system comprises a VOIP-enabled wireless communication device (WCD) and a VOIP gateway. The WCD includes a short-range wireless interface and a client application configured to place and receive VOIP calls through the short-range wireless interface. The VOIP gateway includes a short-range wireless interface for communicating with the WCD, a network interface for communicating with the Internet, and a VOIP service client configured to communicate with a VOIP service over the Internet by way of the network interface. The VOIP gateway also includes a proxy server configured to act as an interface between the WCD client application and the VOIP service client and to route the VOIP calls through the gateway's short-range wireless interface. | 09-24-2009 |
| 20090238171 | TELECOMMUNICATIONS SYSTEM AND METHOD FOR CONNECTING SEVERAL CSTA CLIENTS TO A PBX - A system connects a plurality of CSTA Clients to a communications system that supports only one CSTA Client at a time, such as a PBX. The system includes a server or other processor programmed to provide a CSTA dialog with each of the plurality of CSTA Clients, and a single CSTA dialog with the PBX. | 09-24-2009 |
| 20090238172 | IP PHONE TERMINAL, SERVER, AUTHENTICATING APPARATUS, COMMUNICATION SYSTEM, COMMUNICATION METHOD, AND RECORDING MEDIUM - A transfer unit transfers a message between a network and an external terminal. An input unit inputs a user ID for identifying a user. A generating unit generates a registration message requesting a registration of address information of the user. A transmitting unit transmits the registration message to a server. A receiving unit receives a response message including registration information and connection information from the server. When the connection information indicates a permission of a connection of the external terminal to the network, a control unit controls the transfer unit to transfer the message between the network and the external terminal. | 09-24-2009 |
| 20090238173 | VoIP Integrating System and Method Thereof - An internet phone integrating system includes a PC, a VoIP phone, a softphone, an HID signal-transmitting unit, and a media transmitting unit. The VoIP phone provides an HID inputting signal. The softphone provides an HID outputting signal and a media controlling signal and decodes an audio coding streaming to a media data flow. The HID signal-transmitting unit receives the HID outputting signal from the softphone and sends the HID outputting signal to the VoIP phone, and receives the HID inputting signal from the VoIP phone and sends the HID inputting signal to the softphone. The media transmitting unit receives the media controlling signal and media data flow from the softphone and sends the media controlling signal and media data flow to the VoIP phone, and receives the media data flow from the VoIP phone and sends the media data flow to the softphone. | 09-24-2009 |
| 20090238174 | Service Handling in a Service Providing Network - A method is described for handling services in a service providing network. The network comprises a serving network node connected to one or more application servers. The method comprises the steps of a first terminal comprising one or more services, preferably VoIP services, sending a registration message to the serving network node associated with the user terminal; providing the serving network node in response to the registration message, with service routing information associated with the first terminal, the service routing information arranged to prevent registration of the first terminal to services residing on the application servers and corresponding with one or more services in the first terminal. | 09-24-2009 |
| 20090238175 | SYSTEM AND METHOD FOR VOICE OVER INTERNET PROTOCOL (VoIP) AND FACSIMILE OVER INTERNET PROTOCOL (FoIP) CALLING OVER THE INTERNET - A system and method for sending long distance telephone calls over the Internet utilizes cost and quality of service data to optimize system performance and to minimize the cost of completing the calls. The system utilizes a network of gateways connected to the Internet. The gateways receive calls from various service providers and convert the analog calls into data packets which are then placed onto the Internet. Similarly, the gateways take data packets off the Internet, convert the data packets back into analog format, and provide the analog telephone calls to the same or another service provider. Then system periodically checks the quality of communications between each of the gateways, and uses this information, in combination with cost information, to determine how to route the calls over the Internet. Special addressing protocols can be used by a system embodying the invention to reduce or eliminate unnecessary signaling between gateways as call setup procedures are carried out. The system can also use information about calls that has been recorded in more than one location to determine how much to charge for completing a call. | 09-24-2009 |
| 20090238176 | METHOD, TELEPHONE SYSTEM AND TELEPHONE TERMINAL FOR CALL SESSION - A method for call session, an IP telephone system and an IP telephone terminal are disclosed, the method including: receiving a call session operation signal from a first IP telephone terminal; detecting based on the call session operation signal whether a second IP telephone terminal using the same telephone number as the first IP telephone terminal is communicating with a third IP telephone terminal; and controlling the first IP telephone terminal and the second IP telephone terminal to exchange information with the third IP telephone terminal when detecting that the second IP telephone terminal is communicating with the third IP telephone terminal. A definition is added for implementation of the IP telephone terminal combination function in an IP telephone system so that IP telephone terminals can use the same telephone number to join or quit a call session for the combined terminals, and accordingly the user satisfaction may be enhanced. | 09-24-2009 |
| 20090238177 | INTERNET PROTOCOL TELEPHONY VOICE/VIDEO MESSAGE DEPOSIT AND RETRIEVAL - A method for signaling an Integrated Messaging System (IMS) on an Internet Protocol (IP) based network to deposit a message, including the steps of sending a Session Initiation Protocol (SIP) SIP INVITE request to the IMS indicating a message deposit action; receiving a corresponding SIP message from the IMS agreeing to participate in the message deposit action; and sending an SIP acknowledge message to the IMS confirming receipt of the corresponding SIP message; and depositing the message in a destination mailbox. A method of signaling an IMS on an IP based network to retrieve a deposited message, the method including the steps of sending a SIP INVITE request to the IMS indicating a message retrieval action; receiving a corresponding SIP message from the IMS agreeing to participate in the message retrieval action; sending an SIP acknowledge message to the IMS confirming receipt of the corresponding SIP message; and retrieving the deposited message from a mailbox corresponding to known account information. | 09-24-2009 |
| 20090245232 | GROUP PAGING SYNCHRONIZATION FOR VOIP SYSTEM - This invention overcomes the problem of delay associated with establishing connections with individual phones by providing a method for sending a virtual real time voice message processed through a VOIP system to a group of phones concurrently. The method includes assembling a portion of the voice message. The voice message includes a voice portion and an address portion. The voice portion of the voice message is buffered in a digital buffer. The address portion is used to determine the address of each phone in the group. After the address of each phone in the group is determined, an attempt is made to establish a connection with each phone. The method further includes waiting for a period of time. The period of time is determined based at least in part on a time duration required to establish a connection with a phone. After waiting the period of time, the voice portion of the voice message is sent to at least one phone in the group of phones. | 10-01-2009 |
| 20090245233 | UNIFIED SESSION SIGNALING SYSTEM FOR USE IN MULTIMEDIA COMMUNICATIONS - A design for a unified session signaling system for use in multimedia communications is disclosed. In one embodiment, a method includes interfacing, via an application interface, with an associated application and a session, tracking, via a call state/session manager, a call state and session properties across multiple calls associated with the session, managing, via a server interoperation module, registration and proxying services associated with the session, managing, via a basic SIP services module using a third party SIP stack, a basic set of SIP services associated with the application and the session, and determining and advertising, via a media negotiator module, media capabilities of devices associated with the session. The method may also include managing, via an additional SIP services module using the third party SIP stack, a set of additional services associated with the session. | 10-01-2009 |
| 20090245234 | DYNAMIC REROUTING OF VOIP SESSIONS - Network connections that are used in routing VoIP session to their end destination are monitored to determine when a call fails to reach its destination. When a network failure is detected, the call is automatically routed to another device that is not dependent on the failed network connection. The call may be automatically rerouted to one or more secondary numbers, such as a mobile telephone number and/or a PSTN number. | 10-01-2009 |
| 20090245235 | Relay apparatus and memory product - If a call session has been established between terminal apparatuses on the sending side and receiving side when a voice packet is received from the terminal apparatus on the receiving side, a firewall apparatus sends the received voice packet to the terminal apparatus on the sending side. On the other hand, if a call session has not been established when a voice packet is received, the firewall apparatus starts buffering received voice packets. When a call session is established, the firewall apparatus sends the buffered voice packets to the terminal apparatus on the sending side. | 10-01-2009 |
| 20090245236 | Method, System, and Computer Program Product for Managing Routing Servers and Services - A method, system, and computer program product for routing network traffic (calls in a Voice over Internet Protocol (VoIP)), which expands the capabilities of existing systems by providing faster and more efficient direction of network traffic, is disclosed. A routing management system includes a routing manager which maintains a list of local routes, establishes and manages connections to the routing server(s), exports routes to the routing server(s), imports disseminated routes from the routing server(s), obtains static global and dynamic routes from the routing server(s), caches those routes for future use, finds all matching routes for a particular number dialed by the user, and prioritizing those routes based on timing, access and ordering information. An additional embodiment contains at least one routing server which provides look-up services for gateway server(s), allows export of local routes from gateway server(s), and distributes translation data; and at least one gateway server which handles calls received on either the Internet protocol (IP) or traditional telephony networks. The gateway server bridges calls between the different kinds of networks, interacts with users, interfaces with the routing system. | 10-01-2009 |
| 20090245237 | Architectures for clearing and settlement services between internet telephony clearinghouses - A system for routing voice telephone calls over IP networks as opposed to traditional switched circuit networks. The voice communications during the telephone call are packaged as digital data and access the Internet through gateways. The system supports the linking of a source gateway in a first clearinghouse to a destination gateway in a second clearinghouse. The system further supports the selection of a destination gateway based on factors such as cost, speed of routing, and transmission quality of the voice data. The components of the system are arranged so as to minimize the number of signals sent between clearinghouses in identifying the optimal destination gateway. | 10-01-2009 |
| 20090245238 | Telephone System, Associated Exchange, and Transmission Control Method - According to one embodiment, a telephone system which realizes voice communication by using a packet network comprises an exchange which accommodates a telephone terminal as its extension and a call processing server which processes calls on the packet network. The exchange comprises a first trunk connected to the packet network, a second trunk connected to a public network having a different protocol from that of the packet network, a monitoring module which monitors the call processing server, and when a failure occurs in the call processing server, deactivates the first trunk, and a call control module which transfers a transmission request which is made from the telephone terminal to the packet network to the second trunk when the transmission request is made and performs a detour transmission to the public network in a status where the first trunk is inactive. | 10-01-2009 |
| 20090245239 | PERFORMING OPERATIONS ON IP TELEPHONY DEVICE FROM A REMOTE CLIENT - A method and system for remotely accessing an intelligent IP telephony device is provided. Information about at least one IP telephony device associated with a user is stored in a database. The database is accessible to a user through a secured environment. From a remote location, the user may logon to the database and select one or more actions to be performed on any of the IP telephony devices to which they have access. | 10-01-2009 |
| 20090245240 | METHOD, NETWORK AND APPARATUS FOR ROUTING SESSIONS - The present disclosure provides a method, network and apparatus for routing sessions so as to route sessions correctly according to the IDs of the domain-related users when the domain-related users are not registered on the IMS network. The method includes: obtaining a wildcard route ID; determining the corresponding domain user ID according to the wildcard route ID; and routing the session through the route set corresponding to the domain user ID. | 10-01-2009 |
| 20090252149 | METHOD FOR PROCESSING BEARER CONTROL - A method for bearer control includes: a first control entity routes a call which is routed from a TDM bearer based network domain or originated by an intra-office POTS subscriber, into an IP bearer based network domain; a second control entity receives the call which is routed back from said IP bearer based network domain and is destined to the TDM bearer based network domain or the intra-office POTS subscriber; when it's determined that a media stream that enters into and exits from said IP bearer based network domain is exchanged on the same media gateway, the first and the second control entities perform a media negotiation during the SIP session to select a same media codec type as the one used on TDM circuits or subscriber lines at two sides, and control a corresponding media gateway to transmit the media stream according to said selected media codec type. | 10-08-2009 |
| 20090252150 | System and Method for Secure Transaction Routing on Demand - A secure transaction routing system, which includes a transaction terminal for generating a routing type based on a message received from a transaction instrument, is provided. Many host servers that link to the transaction terminal through one or more communication gateways are also included. The communication gateways route a call session to at least one of the many host servers based on the routing type. | 10-08-2009 |
| 20090252151 | Method and Network Elements for Content Duplication in Packet Networks - There is provided method for duplicating communications content in a telecommunications network, wherein the content is transported in a layered communications protocol comprising at least one protocol layer. The method comprises receiving first data identifying the content to be duplicated, receiving second data identifying a lowest protocol layers to be duplicated, and duplicating the content as identified by said first data including all protocol information of the lowest protocol layer as identified by said second data, further including all higher layer protocol information. An advantage thereof is that, by means of the second data, the protocol depth of the duplication may be influenced. For example, if the content is transported by the protocols RTP (real-time protocol), UDP (user datagram protocol), and IP (internet protocol), then by means of the second data the content alone, or the content plus the entire RTP protocol information (of which the content is the payload), or the entire IP traffice associated with the content to be duplicated could be selected for duplication. A preferred application of the duplication method is lawful interception (LI), wherein the duplicated content and protocol information along with labels and/or parameters, if applicable, is forwarded to a monitoring facility or monitoring center. | 10-08-2009 |
| 20090252152 | METHOD OF VoIP NUMBER PORTABILITY USING WIRELESS DEVICE - A method of processing a number portability call, the method including: transmitting a call request message to a donor network server based on dialed number information of a called terminal; receiving a response message according to number portability of the called terminal from the donor network server, in correspondence to the call request message; detecting routing number information of the called terminal based on the dialed number information, according to reception of the response message; and performing call setup to a recipient network server associated with the called terminal based on at least one of the dialed number information and the routing number information is provided. | 10-08-2009 |
| 20090252153 | METHOD FOR PROVIDING EARLY-MEDIA SERVICE BASED ON SESSION INITIATION PROTOCOL - The present invention relates to a method of providing an early media service based on a session initiation protocol (SIP), wherein early media of a multimedia form can be provided under SIP-based B2BUA mode operation. According to the present invention, in a case where early media are provided to an originating terminal when a call connection with a terminating terminal is established at the request of the originating terminal, the early media is provided in the form of multimedia data, such as text, image, moving image, flash animation and the like, as well as audio data, and thus users desires are fulfilled and users, satisfactions are maximized. In addition, with individual operation management of the terminating terminal and the originating terminal according to B2BUA mode operation based on the session initiation protocol and an early session initiation with the originating terminal, an early media service can be normally provided to the originating terminal even when the terminating terminal is in an abnormal operation state. | 10-08-2009 |
| 20090252154 | SYSTEM FOR INTEGRATING AND TRANSMITTING NETWORK PHONE SIGNALS AND METHOD APPLIED THEREIN - A system for integrating and transmitting network phone signals and a method applied therein. The system and method provide a medium server device applicable between network phone client ends and network phone message exchange system, the medium server device enabling the network phone client ends and the network phone message exchange system to process message exchanges between TCP port 80 and UDP port 5060 and/or UDP port 1024˜65535 transmission protocols. Accordingly, a network phone client end is capable of directly communicating with other local phone client ends and/or network phone client ends in the UDP port 80 transmission protocol. | 10-08-2009 |
| 20090252155 | REDUNDANT GATEWAY SYSTEM - First and second gateway devices perform TDM conversion on data from multiple packets supplied from the packet networks to generate TDM signals. A TDM exchange unit switches to the first gateway device from the second gateway device to supply the TDM network with only the TDM signal generated by the first gateway device. When the TDM exchange unit switches to the first gateway device from the second gateway device, a jitter buffer controller of the second gateway device notifies the first gateway device of the packet read order determined by the jitter buffer controller of the second gateway device, and the first gateway device determines a packet read order as the packet read order determined by the jitter buffer controller of the second gateway device. | 10-08-2009 |
| 20090252156 | Voice over internet protocol switch devices - A VOIP switch device has a first terminal ( | 10-08-2009 |
| 20090252157 | Method of setting up a call in an internet protocol multimedia subsystem network - The present invention is directed to a method of setting up a call from an originating user in an internet protocol (IP) multimedia subsystem (IMS) network. The originating user provides a signalling message containing an originating identifier of the user to a first node of the network. According to the method, a first node of the network receives the signalling message. The first node performs a verification on whether the originating identifier is associated with a wildcard identifier, wherein the wildcard identifier identifies a plurality of identifiers which are entitled to using a group service. If the originating identifier is associated with a wildcard identifier, the first node forwards the wildcard identifier and the originating identifier to a further node for setting up the call. | 10-08-2009 |
| 20090252158 | APPLICATION SERVER ALLOWING THE DISTRIBUTION OF A CALL INTENDED FOR A TERMINAL CONNECTED TO A GATEWAY TO ALL TERMINALS CONNECTED TO THIS GATEWAY - According to the invention, an application server (AS) for a telecommunication network (IPN) supporting the SIP protocol, includes:
| 10-08-2009 |
| 20090252159 | SYSTEM AND METHOD FOR PROCESSING TELEPHONY SESSIONS - In one embodiment, the method of processing telephony sessions includes: communicating with an application server using an application layer protocol; processing telephony instructions with a call router; and creating call router resources accessible through a call router Application Programming Interface (API). In another embodiment, the system for processing telephony sessions includes: a call router, a URI for an application server, a telephony instruction executed by the call router, and a call router API resource. | 10-08-2009 |
| 20090257428 | VoIP-BASED INVOCATION OF PSTN-BASED AIN/IN SERVICES - A method may include receiving an Advanced Intelligent Network/Intelligent Network (AIN/IN) service request from a Voice over Internet Protocol (VoIP) subscriber, generating an IP-based message for invoking the AIN/IN service based on the AIN/IN service request, routing the IP-based message to an AIN/IN service control device via an IP signaling gateway, receiving an AIN/IN response from the AIN/IN service control device based on the IP-based message, and connecting the VoIP subscriber to the AIN/IN service based on the AIN/IN response. | 10-15-2009 |
| 20090257429 | SYSTEM, METHOD, AND COMPUTER-READABLE MEDIUM FOR PROCESSING CALL ORIGINATIONS BY A FEMTOCELL SYSTEM - A system, method, and computer readable medium for processing a call setup in a network system are provided. A femtocell system receives a call origination from a user equipment located within a service area of the femtocell system and performs a service connection with the user equipment. The femtocell system creates a connection for an Internet Protocol Multimedia Subsystem core network, transmits an INVITE message to a called telephone device via the Internet Protocol Multimedia Subsystem, and completes the call setup between the user equipment and the called telephone device. | 10-15-2009 |
| 20090257430 | Method and System for Preventing Data Loss in a Real-Time Computer System - A method and system are provided for preventing data loss in a VoIP system. In particular, during a VoIP call, it is determined whether incoming ringing on a POTS line causes an unacceptable level of signal loss or errors. If so, for subsequent VoIP calls, the CO handling calls to the POTS line is instructed to either answer each call with a busy signal or automatically forward calls to the POTS line to the VoIP line or other selected telephone. Calling returns to normal upon ending of the VoIP call. In this manner, incoming ringing on the POTS line does not result in call dropping or lengthy retraining processes. | 10-15-2009 |
| 20090262723 | Systems and methods for accessing IP transmissions - Various systems and methods for intercepting transmissions are disclosed. In one embodiment, a system is disclosed that includes an internet protocol media gateway. The internet protocol media gateway is communicably coupled to a soft switch, an acquisition facility, and a communicator. The internet protocol media gateway is associated with a processor and a computer readable medium, and the computer readable medium includes instructions executable by the processor to receive a transmission identified with the communicator, and to direct the transmission to the acquisition facility. Various other systems and methods are also disclosed. | 10-22-2009 |
| 20090262724 | PROXY SERVER, COMMUNICATION SYSTEM, COMMUNICATION METHOD AND PROGRAM - Provided in a proxy server which enables a plurality of SIP servers to copy registration information after presenting, to an SIP server requesting SIP Digest authentication, properness of one who makes an access and of a REGISTER request to be transmitted. With a REGISTER request generation function provided in addition to a function that a common SIP proxy server holds, a proxy server disposed between a user agent and an SIP server generates a REGISTER request to a spare SIP server and transmits the same to the spare SIP server, thereby realizing registration information copying, and holds a user identifier and a password of the proxy server to execute Digest authentication with the spare SIP server. Moreover, after confirming a REGISTER request processing completion response (200 OK) from a working SIP server from which registration information is copied, the proxy server transmits a copied REGISTER request to the spare SIP server. | 10-22-2009 |
| 20090262725 | DISTRIBUTED TRANSCODING ON IP PHONES WITH IDLE DSP CHANNELS - Idle DSP channels of an IP phone can be used to respond to a request to transcode a codec of an incoming call in a distributed IP phone system but only if sufficient idle channels remain available to the phone to handle basic call functions and a possible non-basic call feature (such as conferencing) of the phone. | 10-22-2009 |
| 20090262726 | Method, system and apparatus for accessing communication features - A method and apparatus for accessing communication features in a communication session between at least two communication devices is provided. A first communication path is established between the at least two communication devices via a first communication protocol, the first communication protocol associated with a first set of communication features. A second communication path is established between the at least two communication devices via a second communication protocol, the second communication protocol associated with a second set of communication features thereby giving at least one of the at least two communication devices access to the second set of communication features. | 10-22-2009 |
| 20090262727 | Communication system - A method of initiating a call from a device executing a client program via an access network is provided. The method comprises providing a network node with information associated with the device, receiving from the network node an indication of whether at least one access number for accessing the access network is available, wherein the availability of the access number is based on the information associated with the device, and selectively enabling an input means to receive a selection signal from a user of said device to initiate the call using the access number, wherein the input means is only enabled if it is indicated that the access number is available. | 10-22-2009 |
| 20090262728 | METHOD FOR ROUTING OF CONNECTIONS IN A PACKET-SWITCHED COMMUNICATION NETWORK - The invention relates to a method for routing connections in a packet-switched communication network. Said method is characterized in that the communication network comprises a plurality of virtual local area networks, to each of which at least one network transition between the communication network and another network is allocated, while said connection is assigned to a virtual local area network based on an operator selection code when a connection is established. | 10-22-2009 |
| 20090262729 | SYSTEM FOR EFFECTING A TELEPHONE CALL OVER A COMPUTER NETWORK WITHOUT ALPHANUMERIC KEYPAD OPERATION - A system for effecting a telephone call between telephonic devices is operative to use a computer network, without manual use of the alphanumeric keypads. A third party call control (3PCC) application program interface (API) provides the capability for users to use a web browser or other Internet capable software to place a call, rather than using the telephone keypad. A third party call control application program interface includes a uniform resource locator operable over the Internet to cause a call between a first telephonic device and a second telephonic device to be completed. The uniform resource locator includes identification of the first telephonic device and identification of the second telephonic device. | 10-22-2009 |
| 20090262730 | INTELLIGENT NETWORK AND METHOD FOR PROVIDING VOICE TELEPHONY OVER ATM AND PRIVATE ADDRESS TRANSLATION - An illustrative intelligent network and method for providing voice telephony over Asynchronous Transfer Mode (“ATM”) and private address translation are provided that can provide significant advantages. The method includes generating an input ATM setup message at the calling party CPE that includes a VToA designator and a called party phone number, extracting information from the input ATM setup message such as the VToA designator and the called party phone number, analyzing the information, designating an ATM address of a called party CPE to be stored in the first parameter of an output ATM setup message, determining if private address translation is needed, designating the ATM address of the called party CPE to be stored in a first instance of the second parameter of the output ATM setup message, designating an ATM address of an egress ATM edge switch to be stored in the first parameter of the output ATM setup message, and generating an output ATM setup message. The method also includes extracting information from the output ATM setup message such as the ATM address of the called party CPE, designating the ATM address of the called party CPE that was stored in the first instance of the second parameter of the output ATM setup message to be stored in the first parameter of a destination ATM setup message, and generating a destination ATM setup message that includes the ATM address of the called party CPE stored in the first parameter and the called party phone number value stored in the second parameter. An illustrative intelligent network for providing VToA and private address translation is also provided. | 10-22-2009 |
| 20090268712 | Method for Establishing a Multimedia Session With a Remote User of a Communications Network - For establishing a multimedia session with a remote user's terminal, a terminal starts a signaling intended to establish the multimedia session addressed to the remote user's terminal. Predetermined acknowledgement messages indicate to the terminal and to the remote terminal that the multimedia session is established. The terminal and/or the remote terminal run at least a module of a multimedia application before reception of the predetermined acknowledgement messages. | 10-29-2009 |
| 20090268713 | METHOD AND APPARATUS FOR TESTING IN A COMMUNICATION NETWORK - Method and apparatus for testing in a communication network is described. One example of the invention relates to a method of testing in a voice over internet protocol (VOIP) network. At least one test script is obtained from the VOIP network at an enhanced terminal adapter. The enhanced terminal adapter is configured to couple at least one communication device to the VOIP network. The at least one test script is executed within a scripting framework of the enhanced terminal adapter to interact with at least one component coupled to the VOIP network. Results of the execution of the at least one test script are transmitted from the enhanced terminal adapter to the VOIP network. | 10-29-2009 |
| 20090268714 | APPARATUS AND METHOD FOR PROCESSING VOICE OVER INTERNET PROTOCOL PACKETS - A method for processing Voice over Internet Protocol (VoIP) packets is provided. The method includes: determining if the arrived VoIP packet arrives out of order according to a sequence number of the arrived VoIP packet and a sequence number of a preceding VoIP packet of the arrived VoIP packet; determining whether the buffer has a packet having a same sequence number as the arrived VoIP packet if the arrived VoIP packet arrives out of order; calculating the difference between the sequence number of the arrived VoIP packet and that of the preceding VoIP packet if the buffer has no such packet having the same sequence number as the arrived VoIP packet; and counting a number of pseudo packets needed to be inserted into the buffer according to the calculated difference and generating and inserting the number of pseudo packets into the buffer. | 10-29-2009 |
| 20090268715 | System and Method for Providing Service Correlation in a Service Access Gateway Environment - A network service access gateway is described that provides service correlation for incoming and outgoing invocations. The service requests can be received to the gateway from telecommunication mobile devices as well as from external service provider applications. A first service request can be received to the gateway and processed. The service correlation identifier (SCID) of the request can be persisted within the gateway prior to forwarding the request to the recipient. When a second and related service invocation is later received to the gateway, the two invocations can be associated based on the SCID. Based on the association, various custom functionality can be performed, such as invoking the charging system to treat the multiple services as a single unified transaction. | 10-29-2009 |
| 20090268716 | Communication method and apparatus - A method of sorting communication events at a user terminal connected to a communication network and executing a communication client arranged to be operable by a user is provided. The method comprises storing an event list comprising a list of identifiers, each identifier having information relating to at least one previously received communication event associated therewith, wherein the identifier identifies the initiator of the associated at least one previously received communication event and each identifier is listed only once in the list of identifiers. The event list is displayed in a user interface of the communication client. The method further comprises receiving an incoming communication event at the user terminal from an initiating user over the communication network and determining whether the initiating user is present in the list of identifiers stored in the event list. In the case that the initiating user is present in the list of identifiers, the event list is amended by adding information relating to the incoming communication event to the information relating to the at least one previously received communication event associated with the identifier of the initiating user. In the case that the initiating user is not present in the list of identifiers, a new entry is created at the top of the event list comprising an identifier for the initiating user and having information relating to the incoming communication event associated therewith. The display of the event list is updated in the user interface of the communication client. | 10-29-2009 |
| 20090268717 | NETWORK DEVICE AND METHOD FOR ESTABLISHING QUALITY OF SERVICE - A network device for establishing quality of service (QoS) between two terminal devices includes a transceiver module and a state-machine setting module. The transceiver module is configured for receiving establishing requests, request responses, acknowledge messages, and QoS requests from any one of the two terminal devices. The state-machine setting module is configured for setting a state of the network device according to a current state of the network device and messages received by the transceiver module, and the state of the network device includes an idle state, an inviting state, a trying state, an acknowledge state, and a QoS state. | 10-29-2009 |
| 20090268718 | COMMUNICATION METHOD AND SYSTEM OF INTERNET - An Internet communication system including a first access point, a second access point, a first caller and a first callee is provided. The first access point and the second access point are respectively located in a first LAN and a second LAN. The first caller, having a probing-based mechanism, accesses the Internet via the first access point and has voice packets with a first transmission priority. The first callee accesses the Internet via the second access point. The first caller transmits a simulation packet to the first callee for probing a transmission quality of an end-to-end transmission path of the first caller and the first callee to determine whether to invite the first callee to communicate via the Internet. | 10-29-2009 |
| 20090268719 | Telephone System and Terminal Device Therein - According to one embodiment, a telephone system comprises a plurality of terminal devices and a main unit. The terminal device realizes telephone communication via a packet-switched network. The main unit accommodates the terminal devices via the packet-switched network. Each of the terminal device comprises an update module, a storing unit, a read module and an access module. The update module updates firmware functioning inside the device in accordance with an instruction from the main unit. The storing unit stores access information for accessing an ante-unit, to which the terminal device is currently connected, before update of the firmware. The read module reads the access information from the storing unit after update of the firmware if an post-unit to which the terminal device is to be connected under the updated firmware differs from the ante-unit. The access module which accesses the ante-unit by using the read access information. | 10-29-2009 |
| 20090268720 | Service Controlling in a Service Provisioning System - A method and a system is described for controlling a service in a service provisioning network. The method including the steps of: a serving network node associated with a user terminal receiving a registration message, the user terminal having one or more of services, preferably VoIP services; and, the serving network node retrieving in response to the registration message service routing information associated with the first user terminal, the service routing information being arranged to route service messages associated with the first user terminal via a stateless application server, the stateless application server being adapted to perform control actions on said service messages. | 10-29-2009 |
| 20090268721 | TELEPHONE SYSTEM, ITS SERVER UNIT, AND DATABASE SYNCHRONIZATION METHOD - According to one embodiment, a telephone system comprises networks connected mutually, terminals and servers. The terminals belong to any one of the networks. The servers control each of networks and accommodate terminals. The server comprises database, manager and controller. The terminals and the server of their assignment destinations are associated with one another in the database. The manager updates to synchronize among the databases and the servers at movement destinations as the terminals under its own control move to control by other servers. The controller specifies a server for controlling a terminal at incoming call destination of outgoing call, destined to another network, from the databases transmit a call message to the specified server when the outgoing call toward another network is generated from the terminal under its own control to another network. The databases are sequentially synchronized among the servers as calls are generated among the terminals. | 10-29-2009 |
| 20090268722 | User Equipment and System Architecture for Voice over Long Term Evolution via Generic Access - Some embodiments provide a communication system that includes (1) an evolved packet system (EPS) that includes an evolved packet core (EPC) and several evolved Universal Mobile Telecommunication System (UMTS) Terrestrial Radio Access Network (E-UTRANs) for communicatively coupling a user equipment (UE) to the EPC, where the EPC is not capable of providing circuit switched (CS) services for the UE and (2) a Voice over long term evolution (LTE) via Generic Access (VOLGA) network controller (VANC) communicatively coupling the UE through the EPS to a legacy circuit switched (CS) core network, where the legacy CS core network is capable of providing CS services to the UE. | 10-29-2009 |
| 20090268723 | Methods and Apparatuses for Transporting Signalling Connectivity Status Information Relating to the Signalling Connection Between a Terminal and P-CSCF in IMS - A system, method, and Proxy Call/Session Control Function (P-CSCF) for transporting signaling connectivity status information relating to a signaling connection between a terminal and the P-CSCF in an IP Multimedia Subsystem (IMS) network. In one embodiment, when the P-CSCF detects that the connectivity status has changed, the P-CSCF sends a SIP request such as a REGISTER request to a Serving CSCF (S-CSCF) indicating the new status. Alternatively, the registration event package of the terminal may be extended to include the connectivity status, and the P-CSCF then sends the status in a PUBLISH request. In an alternative embodiment, the P-CSCF maintains a new SIP event package. The S-CSCF subscribes to the SIP event package and the P-CSCF notifies the S-CSCF upon a change of connectivity status. | 10-29-2009 |
| 20090268724 | SYSTEMS, PROCESSES AND INTEGRATED CIRCUITS FOR RATE AND/OR DIVERSITY ADAPTATION FOR PACKET COMMUNICATIONS - Packets of real-time information are sent with a source rate greater than zero kilobits per second, and a time or path or combined time/path diversity rate initially being zero kilobits per second. This results in a quality of service QoS, optionally measured at the sender or the receiver. When the QoS is on an unacceptable side of a threshold of acceptability, the sender sends diversity packets at an increased rate. Increasing the diversity rate while either reducing or maintaining the overall transmission rate is new. CELP-based multiple-description data partitioning sends the base or important information plus a subset of fixed excitation in one packet and sends the base or important information plus the complementary subset of fixed excitation in another packet. Reconstruction produces acceptable quality when only one of the two packets is received and better quality when both packets are received. Reconstruction provides for single and multiple lost packets. | 10-29-2009 |
| 20090274141 | IP TELEPHONE SYSTEM AND IP TELEPHONE METHOD - There are provided an IP telephone system and method for establishing a connection to the IP network | 11-05-2009 |
| 20090274142 | Device and Method for the Recognition of Call Numbers for Voice-Over-Ip Telephony - Call numbers are recognized in order to establish a connection from a lie-switched network to a packet-switched network. In one aspect, a device comprises a unit for detecting a selected string of digits as a selected call number, a unit for storing a plurality of authorized call numbers, a comparator unit for comparing the selected all number to the plurality of stored call numbers, and a unit for converting the selected call number into an associated IP address as soon as the comparator unit detest that the selected call number matches one of the stored all numbers. | 11-05-2009 |
| 20090274143 | State Machine Profiling for Voice Over IP Calls - An apparatus and method for detecting potentially-improper call behavior (e.g., SPIT, etc.) are disclosed. The illustrative embodiment of the present invention is based on finite-state machines (FSMs) that represent the legal states and state transitions of a communications protocol at a node during a Voice over Internet Protocol (VoIP) call. In accordance with the illustrative embodiment, a library of FSM execution profiles associated with improper call behavior is maintained. When there is a match between the behavior of a finite-state machine during a call and an execution profile in the library, an alert is generated. | 11-05-2009 |
| 20090274144 | Multi-Node and Multi-Call State Machine Profiling for Detecting SPIT - An apparatus and method for detecting potentially-improper call behavior (e.g., SPIT, etc.) are disclosed. The illustrative embodiment of the present invention is based on finite-state machines (FSMs) that represent the legal states and state transitions of communications protocols at nodes during Voice over Internet Protocol (VoIP) calls. In accordance with the illustrative embodiment, a library of FSM execution profiles associated with improper call behavior and a set of rules (or rule base) associated with improper FSM behavior over one or more calls are maintained. When the behavior of one or more finite-state machines during one or more calls matches either an execution profile in the library or a rule in the rule base, an alert is generated. | 11-05-2009 |
| 20090274145 | Methods, Systems, and Products for Emergency Communications - Methods, systems, and products are disclosed for processing emergency communications. A database of addresses is queried to determine if a communications address is an emergency communications address. When the communications address is the emergency communications address, then a location coordinate is retrieved and mapped to a location of an emergency services provider. | 11-05-2009 |
| 20090274146 | METHOD, SYSTEM AND DEVICE FOR IMPLEMENTING NETWORK ADDRESS TRANSLATION TRAVERSAL - A method for implementing NAT traversal is disclosed herein. The first MGW and the ICE mechanism supporting device obtain the local candidate information and the candidate information of the peer end; the first MGW and the ICE mechanism supporting device perform connectivity check according to the candidate information; and the first MGW and the ICE mechanism supporting device transmit media streams according to the result of the connectivity check. A system and a device for implementing NAT traversal are also disclosed. The method, the system and the device under the present disclosure improve stability of transmitting media streams in a network inclusive of an MGC and an MGW (for example, an NGN). | 11-05-2009 |
| 20090274147 | ELECTRONIC LOOP PROVISIONING - The present invention is directed to a local network access architecture and method of providing local services that advantageously replaces portions of the physical hardwired local loop with a path that is software-defined. In one embodiment the system comprises a remote terminal comprising a packet processor that converts an analog signal carried on a customer loop into digital packets and a packet node connected to the remote terminal configured to selectively forward the digital packets based on an identifier in the digital packets to equipment of one of a plurality of local exchange carriers, wherein said plurality of local exchange carriers are different companies and each one of said plurality of local exchange carriers provides at least one different service subscribed to by a subscriber. | 11-05-2009 |
| 20090279532 | TCP/IP BASED VOICE COMMUNICATION SYSTEM - In various embodiments described herein a TCP/IP based voice communication system is described. The TCP/IP based voice communication system may be useful in a correctional facility or other environments such as college campus, hospitals or other institutions. In addition to providing voice communication from a source to a destination, the voice communication system can perform additional functions for example, validating destination numbers, maintaining user records and storing call details. | 11-12-2009 |
| 20090279533 | EXTENSIBLE AND SECURE TRANSMISSION OF MULTIPLE CONVERSATION CONTEXTS - The entry and transmission of notes to recipients along the conversation chain. Notes can be created based on an incoming caller. The notes can be transmitted to the conversation recipient for viewing before, during, and after the recipient accepts the conversation. This is facilitated by a communications client that operates to allow entry of the notes, and forwarding of the call recipient via a SIP framework. Moreover, notes previously taken and/or information provided manually and/or automatically by the communications system can be provided to an agent (e.g., ACD, receptionist) receiving the conversation, at any point in the conversation chain for quick identification not only of the conversation source but of previous information already collected. | 11-12-2009 |
| 20090279534 | Method and System for Placing a VOIP Call - The present document describes a method and system for placing a VoIP call from a user using a user voice interface device in a given geographical area to a contact using a contact voice interface device in a distant geographical area. The method comprises: assigning an individual local access phone number per geographical area thereby resulting in a list of individual access phone numbers; the user placing a call from the user voice interface device to the individual local access phone number assigned to the given geographical area thereby initiating a first leg of the call from the user voice interface device to the bridge server through the PSTN; switching the call from the PSTN to a given URL which points to a bridge server accessible through the Internet; the user providing the identity of the contact to which the call must be completed, the identity of the contact corresponding to the contact voice interface to which a second leg of the call will be established; the bridge server establishing the second leg of the call from the bridge server to the contact voice interface device; and the bridge server bridging the first and second legs of the call thereby establishing the VoIP call from the user to the contact. | 11-12-2009 |
| 20090279535 | Providing Dynamic Services During a VOIP Call - The present document describes a method and system for providing services during a call established between a user making the call and a contact. The call being established using a voice interface device having a key. The method comprises: providing an electronic assistant in a background mode; using the key to produce a summoning signal; upon detection of the summoning signal, summoning the electronic assistant to a foreground mode; issuing a command to the electronic assistant for the provision of a service; and upon detection of the command, providing the service. | 11-12-2009 |
| 20090279536 | IP forwarding across a link state protocol controlled ethernet network - Nodes on an Ethernet network run a link state protocol on the control plane and install shortest path forwarding state into their FIBs to allow packets to follow shortest paths through the network without requiring MAC header replacement at each hop through the network. When a node learns an IP address, it will insert the IP address into its link state advertisement to advertise reachability of the IP address to the other nodes on the network. Each node will add this IP address to its link state database. If a packet arrives at an ingress node, the ingress node will read the IP address, determine which node on the link state protocol controlled Ethernet network is aware of the IP address, and construct a MAC header to forward the packet to the correct node. The DA/VID of the MAC header is the nodal MAC of the node that advertised the IP address. Unicast and multicast IP forwarding may be implemented. | 11-12-2009 |
| 20090279537 | METHOD AND SYSTEM FOR NETWORK ADDRESS TRANSLATION (NAT) TRAVERSAL OF REAL TIME PROTOCOL (RTP) MEDIA - A solution for the Network Address Translation (NAT) traversal problem for Real Time Protocol (RTP) is provided, which uses an RTP Proxy (e.g., a Session Border Controller (SBC)), instead of being logically located between the NAT and the Feature Server (FS), but instead, for devices which use a protocol unsupported by the SBC, having these devices first signal the Feature Server, which determines whether and how an RTP proxy should be invoked. An RTP proxy should be invoked by the FS if Both endpoints (e.g., devices) are behind different NATs (or one of the endpoints is behind a NAT and the other is not) and neither of the endpoints are already signaled through an RTP proxy. For example, the SBC is interposed (at least logically) between the Feature Server and other shared components. | 11-12-2009 |
| 20090279538 | DYNAMIC COMMUNICATION LINE ASSIGNMENT - A system that enables a calling party to communicate with a called party over a communications network comprises: (a) a web page storage device that is operable to send, over the internet, (i) web pages to a calling party device, the web pages including a data entry screen into which a user enters a required telephone number or VoIP user name with which communication is sought and (ii) a call-in number; (b) a conversion device that is operable to receive over the internet, from the calling party device, the telephone number or VoIP user name and can cause the altering of call forwarding settings at a switch, such that a call from the calling party device to a call-in number will be automatically forwarded to a device associated with the telephone number or VoIP user name; (c) a dynamic line assignment module that can dynamically assign the call-in number. | 11-12-2009 |
| 20090279539 | POST ANSWER CALL REDIRECTION VIA VOICE OVER IP - A method is provided for forming a multi-media communication path between at least first, second and third communication devices coupled to a multi-media provider system during post answer call redirecting and/or teleconferencing. The method includes receiving and processing a first call request at a circuit-based portion of the multi-media provider system for forming a first communication link between the first and second communication devices. Thereafter, predetermined attributes of the first communication link may be sent to an IP-based portion of the multi-media provider system for configuring the IP-based portion of the multi-media provider system to process a subsequent request to execute post answer call redirecting and/or teleconferencing. Upon detecting the request to execute the post answer call redirecting and/or teleconferencing in the first communication link, the IP-based portion of the multi-media provider systems responds by forming the multi-media communication path between at least first, second and third communication devices. | 11-12-2009 |
| 20090285198 | APPARATUS AND METHODS FOR PROVIDING MEDIA PACKET FLOW BETWEEN TWO USERS OPERATING BEHIND A GATEWAY DEVICE - A method for supporting communication between a source internet protocol phone and a destination internet protocol phone is provided. The source internet protocol phone communicates via a Network Address Translator (“NAT”) gateway. The method includes receiving a packet from the source phone at the NAT. The packet is for communication with the destination phone. The method further includes querying whether the destination phone is located in the subnetwork serviced by the NAT gateway. If the destination phone is not located in the subnetwork serviced by the NAT gateway, then the method includes sending the packet upstream to the destination phone via the Internet. If the destination phone is located in the subnetwork serviced by the NAT gateway, then the method includes directing the packet to the destination phone. | 11-19-2009 |
| 20090285199 | METHOD AND APPARATUS FOR SUPPORTING ENTERPRISE ADDRESSING IN NETWORKS - A method and apparatus for supporting enterprise addressing in networks are disclosed. For example, the method creates a Domain Name System (DNS) service record and loading said DNS service record in a public DNS server for a customer, wherein the DNS service record supports a mapping of a domain name of the customer to a sub-domain name of a service provider. The method receives a call destined to a customer endpoint device for the customer; and forwards the call to the customer in accordance with the DNS service record. | 11-19-2009 |
| 20090285200 | DEVICE AND METHOD FOR ENABLING SIP DECT TERMINAL MOBILITY - The present invention concerns a networking device comprising a first interface to a first network and a second interface to a second network. The device comprises connecting means for associating to a terminal located on the first network and storing a unique and permanent identifier of the terminal, means for registering with an address comprising the terminal identifier to a server located on the second network for using a service, and means for enabling the terminal to use the service on the address. | 11-19-2009 |
| 20090285201 | OPTIMZATION OF INTERNET TRAFFIC BASED ON APPLICATION PRIORITIZATION - A method of classifying, scheduling, prioritizing, and optimizing data to provide a final data packet ready for transmission by the modem to the head end. Additionally, a feedback loop is provided to improve scheduling, prioritizing and optimizing data by providing real-time bandwidth availability related information and maximum packet size to be sent over the physical layer. | 11-19-2009 |
| 20090285202 | METHOD FOR COMPLETING INTERNET TELEPHONY CALLS - A call between a calling party and a called party, one or both of whom may be subscribers to Internet Telephony (IT) services, commences upon the receipt of a call dialed by the calling party to the Plain Old Telephony Service (POTS) number associated with the calling party. A first hub receives the call and routes it to the called party if that party is not an IT services subscriber that is currently on line. If the called party is an IT services subscriber that is on-line, the call is received at an Internet Services Provider serving the called party. The ISP converts the call to an IT format if the call is not already in that format and thereafter delivers the call to the called party. | 11-19-2009 |
| 20090285203 | FORCED HOLD CALL HANDLING IN A VOP ENVIRONMENT - The present invention provides a technique for providing a forced hold service such as is used for an emergency services call, which is supported at least in part over a packet network. The forced hold service acts to effectively hold a connection for the call with a called party, even when the caller takes an action that would normally end a call, such as going on hook, pressing end, or the like. When the caller takes an action that would normally end the call, the forced hold service allows the caller to automatically reconnect to the emergency services provider over the held connection upon going offhook, pressing send, or the like. Alternatively, the emergency services provider can effectively re-engage the call wherein the caller is reconnected over the held connection upon going offhook, pressing send, or the like. | 11-19-2009 |
| 20090285204 | RECURSIVE QUERY FOR COMMUNICATIONS NETWORK DATA - An approach for providing telephony services over a data network is disclosed. A communications system includes a location server that receives a request from a calling station to establish a call with a station associated with a called party. The location server generates a message specifying a set of addresses relating to the called party and context information. A proxy server communicates with the location server and is configured to receive the message and to attempt to establish the call based on the set of addresses. The proxy server iteratively queries the location server to obtain another set of addresses if no prior address results in establishment of the call. | 11-19-2009 |
| 20090285205 | UNIFIED MESSAGE SYSTEM - The present invention provides a method and devices for unified messaging. One method provides for receiving a message having a first identifier associated with a user, translating the first identifier associated with the user to a second identifier comprising a zip code and a street address, the second identifier being associated with a network address, and sending the message to the user at the network address. A line interface device of the present invention is associated with an address that comprises a zip code. | 11-19-2009 |
| 20090290573 | Method for Establishing a Video Telephone Connection and/or a Multimedia Telephone Connection in a Data Network - A method establishes a video telephone connection in a data network that includes a telephone network and an IP network based on the internet protocol. The expression video telephony connection is to be taken generally in this context and encompasses multimedia telephony in addition to pure video telephony. | 11-26-2009 |
| 20090290574 | Method for Handling Unanswered Calls - Reliable and interactive communication between parties is allowed even in those cases in which an incoming call cannot be answered due to inconvenience, inopportunity and/or impoliteness, e.g., during an important meeting, a conference or a ceremony. A packet-switched connection between the called party and the calling party is established in response to an action intended for terminating an incoming call performed at the called party's terminal. | 11-26-2009 |
| 20090290575 | METHOD OF CORRESPONDENCE BETWEEN GROUP COMMUNICATION IDENTIFIERS AND MULTICAST ADDRESSES - An identifier (IDp) designates a group of terminals accessible via the network (RP) by packets having a multicast address as destination address. An equipment (EIm) between the network and a terminal (Tn) belonging to a group calculates the multicast address of the group using a function depending on the identifier (IDp) of the group each time that a packet including a message transmitted by the terminal is to be transmitted to the network. The equipment also calculates a group identifier (IDp) using a function depending on a multicast address extracted each time that a packet transmitted from the network is received by the equipment in order to transmit a message content extracted from the packet to be received by the terminal if the calculated group identifier is identical to the identifier of the group to which the terminal belongs. No table of mappings between the identifiers of the groups and the multicast addresses is stored in the equipment. | 11-26-2009 |
| 20090290576 | CALL CONTROL METHOD, CIRCUIT-SWITCHED DOMAIN ADAPTER AND TERMINAL DEVICE - A call control method which includes: establishing a circuit-switched call leg with a terminal device; establishing a packet-switched call leg with a second party; and establishing a call connection between the terminal device and the second party through binding the circuit-switched call leg and the packet-switched call leg. A circuit-switched domain adapter and a terminal device are also provided to realize a call control to the circuit-switched terminal device by a packet-switched control platform. | 11-26-2009 |
| 20090290577 | METHODS, APPARATUS AND COMPUTER PROGRAM PRODUCTS FOR ASSOCIATING LOCAL TELEPHONE NUMBERS WITH EMERGENCY PHONE CALLS IN A PACKET SWITCHED TELEPHONE SYSTEM - A packet switched telephone system includes a packet switched routing apparatus. The packet switched routing apparatus selectively associates a local telephone number with a phone call based on a called telephone number, and routes the phone call based on the called telephone number. The local telephone number may be substituted for a calling telephone number when the called telephone number corresponds to a predefined number, such as an emergency number. When the called telephone number corresponds to an emergency number, the phone call may be routed with the substituted local telephone number to a Public Safety Access Point (PSAP) that services the local area of the subscriber. | 11-26-2009 |
| 20090290578 | Screening Inbound Calls in a Packet-Based Communications Network - A method and system is provided for performing inbound call screening in a packet-based network, such as an H.323 Voice over IP (VoIP) network. The inbound gateways on the network are registered with inbound gatekeepers, and standard messages are used between an inbound gateway, an inbound gatekeeper and an inbound screening database to decide: whether an inbound call to a particular called number (DID) is to be allowed into the network; whether the called number should be translated into a different called number; and whether a routing index should be included in the called number to indicate the destination of the call. | 11-26-2009 |
| 20090290579 | Method and Apparatus for Controlling the Quality of Service of Voice and Data Services Over Variable Bandwidth Access Networks - A terminal adapter for guaranteeing the quality of service of both voice and data packets is disclosed. When a data packet is received in a first data input queue of a terminal adapter, a determination is made whether a voice packet is present in a voice input queue. Another determination is made as to whether the sum of the size of the data packet and the size of all packets in a terminal adapter output queue would exceed a first size threshold established for the output queue. If voice packets are present in the voice input queue, or if the aforementioned sum exceeds the size threshold, the data packet is not forwarded to the output queue. If no voice packets are present in the voice input queue and if the aforementioned sum is below the first size threshold, then the data packet is forwarded to the output queue. | 11-26-2009 |
| 20090296686 | METHODS, COMMUNICATIONS DEVICES, AND COMPUTER PROGRAM PRODUCTS FOR SELECTING AN ADVERTISEMENT TO INITIATE DEVICE-TO-DEVICE COMMUNICATIONS - Methods, communications devices, and computer program products for selecting an advertisement to initiate communications between communication devices using an Internet protocol enabled television infrastructure are provided. Input of a call back number is received. Advertisement data of an enterprise is accessed via an Internet protocol enabled device. A selection is received to initiate a communication to the enterprise. A selection of the call back number is received. The call back number is contacted, in response to an indication that the enterprise has been contacted for initiation of the communication. | 12-03-2009 |
| 20090296687 | BYPASSING ROUTING RULES DURING A CONVERSATION - Communication requests added to a conversation are routed directly to a user without following the pre-configured routing rules for the user during a breakthrough period. The breakthrough period may last for the duration of the conversation or for some other period of time. A conversation may be initiated using any supported type of communication. For example, if a user initially sets up an IM conversation with a remote user, then when a voice call is made to the user from the remote user, the voice call is routed directly to the user without applying the routing rules that are configured for the user. Once the breakthrough period has elapsed, the routing rules become active again and are applied to communications received from the remote user that are directed to the user. | 12-03-2009 |
| 20090296688 | Coding and Behavior when Receiving an IMS Emergency Session Indicator from Authorized Source - A method is provided for a user equipment (UE) to respond to an emergency-related message sent to the UE. The method comprises the UE receiving a first message containing an indicator indicating that an emergency-related request has been made, the UE recognizing the indicator as an indication that the emergency-related request is related to an emergency, and the UE sending a second message containing emergency-related information about itself. | 12-03-2009 |
| 20090296689 | Privacy-Related Requests for an IMS Emergency Session - A network component is provided that includes a processor configured, upon the network component receiving an IMS (Internet Protocol Multimedia Subsystem) emergency call from a user equipment (UE), to detect in the emergency call an indicator requesting the network component to restrict presentation of private information related to the UE. The processor is further configured, when the indicator is present, to transmit the emergency call without at least some of the private information to a Public Safety Answering Point (PSAP). | 12-03-2009 |
| 20090296690 | Method And Management Of Public Identities In An Information Transmission Network, Server For Managing Public Identity Records, Equipment For Managing A Group Public Identity And Corresponding Computer Programs - This method of managing public identities in an information transmission network ( | 12-03-2009 |
| 20090296691 | METHOD FOR MAKING TELEPHONE APPARATUS OPERATIVE WITH MULTIPLE NETWORKS - An apparatus having a telephonic communication capability with multiple networks enables users to make telephone calls in a simplified manner. According to an exemplary embodiment, the apparatus includes a memory for storing a first telephone number including an area code, and a processor for receiving a signal to dial the stored first telephone number and for determining whether the first network or the second network is selected. If the first network is selected, the processor causes the stored first telephone number to be dialed. If the second network is selected, the processor enables a user to select between the stored first telephone number and a second telephone number derived from the stored first telephone number by deleting at least the area code. The processor also causes a selected one of the stored first telephone number and the second telephone number to be dialed. | 12-03-2009 |
| 20090296692 | End-to-end Internet connections establishment - Methods and apparatus, including computer program products, for signaling in a network. A method of signaling in a network includes determining in a first end station a destination telephone network address of a second end station and determining in the first end station an intermediate Internet address corresponding to the destination telephone network address. In response to determining the intermediate Internet address, the method retrieves an Internet address of the second end station from an address list at the intermediate Internet address and establishes an end-to-end Internet connection between the first end station and the second end station with the Internet address of the second end station. | 12-03-2009 |
| 20090296693 | Session Initiation Protocol Telephone System, Data Transmission Method, Server Unit, and Telephone Terminal - According to an aspect of the present invention, there is provided a Session Initiation Protocol (SIP) telephone system comprises a server unit, telephone terminals and a module. The server is connected to Internet Protocol (IP) network. The telephone terminals transmit and receive SIP messages to and from the server unit via the IP network. The module applies SIP messages regarding event notification to data transmission to form interactive communication paths among the server unit and each of the telephone terminals. | 12-03-2009 |
| 20090296694 | METHODS, SYSTEMS, AND COMPUTER READABLE MEDIA FOR PROVIDING NEXT GENERATION NETWORK (NGN)-BASED END USER SERVICES TO LEGACY SUBSCRIBERS IN A COMMUNICATIONS NETWORK - The subject matter described herein includes methods, systems, and computer readable media for providing NGN-based end user services to legacy subscribers in a communications network. According to one aspect, the subject matter described herein includes a method for providing NGN-based end user services to legacy subscribers in a communications network that includes, at a service creation system (SCS) node having at least one processor, using the at least one processor for receiving a SS7 call setup message associated with a call involving a legacy subscriber access device and holding the SS7 call setup message. The method also includes, while holding the SS7 call setup message, generating a SIP call setup message related to the SS7 call setup message, and initiating the providing of at least one NGN-based end user service for the call using the SIP call setup message. The method further includes determining whether to modify the SS7 call setup message based on the at least one NGN-based end user service, and, in response to determining to modify the SS7 call setup message, modifying the SS7 call setup message, and routing the SS7 call setup message towards a destination. | 12-03-2009 |
| 20090296695 | HYBRID TYPE TELEPHONY SYSTEM - A hybrid type telephony system capable of establishing a connection between conventional type telephone sets contained in an exchange unit and LAN type telephone sets contained in an IP network, the system comprising: a gateway circuit connected between the exchange unit and the IP network and performing voice data format conversion, and a central control unit connected to the LAN of the IP network for establishing a communication path to the exchange unit via a control bus, controlling switching of IP packets of the IP network, managing IP address information of the LAN type telephone sets and the gateway circuit via the LAN, and controlling connection between the LAN type telephone sets and connection between the LAN type telephone sets and the gateway circuit. | 12-03-2009 |
| 20090296696 | VOICE OVER INTERNET PROTOCOL MULTI-ROUTING WITH PACKET INTERLEAVING - A method and system for processing data packets is described within. The method executed by the system includes the steps of receiving a first data packet, determining if the first data packet is a first expected data packet, determining if the first data packet is a next expected date packet, storing the first data patent if the first data packet is the next expected data packet and waiting a period of time for a second data packet. | 12-03-2009 |
| 20090303983 | Method, Server Device and Converting Device for Setting Up a Payload-Data Connection - There is described a transmission of user data from a source communications device provided with a first encoder for encoding users data to a target communications device provided with a first decoder for decoding said user data via a communication network which is provided with several converting devices comprising additional encoders and additional decoders for carrying out a verification of the converting devices. Via the verification it is determined, whether the first encoder is compatible with the decoder of a given converting device and, whether the first decoder is compatible with the encoder of said converting device. One of the converting devices for which the compatibility is ascertained by the verification is selected for transmitting user data. During transmission of the user data, said user data encoded with the aid of the first encoder is decoded with the aid of the compatible decoder of the converting device and the user data decodable with the aid of the first decoder is encoded with the aid of the compatible encoder of the selected converting device. | 12-10-2009 |
| 20090303984 | System and method for private conversation in a public space of a virtual world - A system and method for allowing a first user and a second user to converse privately in a public place in a metaverse application. The metaverse system includes a metaverse server and a privacy engine. The metaverse server executes a metaverse application. The metaverse application includes a metaverse virtual world that enables a first user to interact with a second user in a public place of the metaverse virtual world. The privacy engine is coupled to the metaverse server. The privacy engine recognizes a private conversation trigger and creates a virtual private space in the public place of the metaverse virtual world in response to the private conversation trigger. The virtual private space facilitates a private audio conversation between the first user and the second user within the public place of the metaverse virtual world. | 12-10-2009 |
| 20090303985 | COMMUNICATION CONTROL METHOD AND COMMUNICATION CONTROL APPARATUS - In a communication system which performs a communication of the user communication information between a user terminal and a user terminal, the communication system including a plurality of network domains each having different types of destination information (for example, host address) of the user communication information transmitted from the user terminal to the user terminal, a communication control method that switches a U-PLANE includes: allocating, to the communication path of the user communication information, path identification information identifying the communication path of the user communication information; notifying, to the user terminal, the path identification information; and transmitting, from the user terminal, the user communication information, by using the path identification information as the destination information. | 12-10-2009 |
| 20090303986 | IP TELEPHONE SYSTEM, NETWORK DEVICE, COMMUNICATION METHOD IN DISASTER SITUATIONS USED THEREFOR AND IP TELEPHONE TERMINAL - An IP (Internet Protocol) telephone system according to the present invention is an IP telephone system including an IP telephone terminal communicating with an opposite party using SIP (Session Initiation Protocol), and a network device transferring a packet from the IP telephone terminal, wherein the IP telephone terminal includes a CPU (Central Processing Unit) transmitting an SIP packet indicating an e-mail address related to a telephone number of the opposite party when the terminal resides in a non-disaster area and calls the opposite party in a disaster area with a disaster mode being set, and the network device includes an SIP packet processing part terminating an SIP packet whose destination is an e-mail address, an e-mail creation part converting a voice packet of RTP (Real Time Protocol) from the IP telephone terminal into text data and creating an e-mail, and a packet transmitting part transmitting the e-mail to the opposite party. | 12-10-2009 |
| 20090303987 | MEDIA RESOURCE ADAPTATION METHOD, MEDIA GATEWAY CONTROLLER AND SERVER - The embodiment of the invention provides a media resource adaptation method, a media gateway controller and a server. In an embodiment of the invention, the services, the CTIS and the MGC are deployed in the center, each dispersed area is only equipped with the access equipment or the resource equipment, the deployment structure is simple, and the maintenance workload is less. The CTIS decides the resource adaptation, and the resource can be extended to the service sides. Therefore, the resource usage is flexible. The user terminal call can obtain the local media services, which reduces the occupancy of a VoIP long-distance link, thus lowering the operation cost. Furthermore, the global share of the media resources is realized, which is favorable for load balance and reduces the cost of redundancy devices. | 12-10-2009 |
| 20090303988 | Communication System for Home Automation Using Advanced ADSL - Communication system for home automation using advanced asymmetric digital subscriber line (ADSL) is provided. Communication system for home automation using advanced ADSL includes: a home automation communication server means for providing home automation service; a home automation service channel means for interchanging data through wire or wireless line of baseband signal with object apparatus of home automation, the home automation service channel means included in ADSL terminal; and a home automation service multiplexing means for connecting the home automation communication server means and home automation service channel means through baseband signal, the home automation service multiplexing means included in ADSL connection apparatus. | 12-10-2009 |
| 20090310595 | PROVIDING SESSION INITIATION PROTOCOL (SIP) CALL CONTROL FUNCTIONS TO PUBLIC SWITCHED TELEPHONE NETWORK (PSTN)-BASED CALL CONTROLLER - A device receives information associated with an outbound call from a calling party via a Public Switched Telephone Network (PSTN), and generates a request for the outbound call from the Public Switched Telephone Network (PSTN) using a remote procedure call (RPC) interface. The device also enables communication, via the remote procedure call (RPC) interface, of the outbound call with a Session Initiation Protocol (SIP)-based device associated with a called party. | 12-17-2009 |
| 20090310596 | APPARATUS, METHOD AND SYSTEM FOR MANAGING BYPASS ENCAPSULATION OF INTERNET CONTENT WITHIN A BYPASS ARCHITECTURE - An apparatus, method and system for delivering Internet content within a system that includes a bypass architecture, such as a bypass architecture that transmits content from the Internet or an Internet content source to a downstream modulator, such as an Edge Quadrature Amplitude Modulation (EQAM) modulator, in a manner that bypasses the system's Cable Modem Termination System (CMTS). Content from the Internet or an Internet source is transmitted to a last-hop router, which is configured to identify content for bypass encapsulation. The last-hop router also can be configured to perform at least a portion of the necessary bypass encapsulation for proper bypass flows of the identified content. Alternatively, the EQAM is configured to perform the bypass encapsulation, and the last-hop router transmits the identified content to the EQAM, which performs at least a portion of the necessary bypass encapsulation on the identified content. | 12-17-2009 |
| 20090310597 | METHOD FOR HANDLING CS CALLS IN VOICE CALL CONTINUITY, VCC APPLICATION SERVER AND TRANSLATION ENTITY - A method for handling CS (Circuit Switching) calls in a VCC (Voice Call Continuity), a VCC application server and an apparatus thereof, wherein first, the VCC application identifies whether a called party number of CS domain calls routed thereto is in an international format, secondly, the VCC application may request to convert the format of the called party number into the international format (referred to as ‘routable number’) if the called party number is not in the international format, thirdly, a translation entity may convert the called party number into the routable number by adding an international prefix suitable for a current location of the originating terminal based on both the called party number and location information on the originating terminal, and then the calls may continue in an IMS domain or a CS domain by the routable number. | 12-17-2009 |
| 20090310598 | TELEPHONE COMMUNICATION - A telephone connection is established between a first terminal (A) and a second terminal (B). The first terminal (A) is presumed to be associated with a subscription in a first home telephone network (α) in which the terminal (A) is identified by means of a first network identity (CLI | 12-17-2009 |
| 20090310599 | Apparatus and method for providing mirroring service in VoIP system including IP-PBX - An apparatus and method for automatically mirroring Real Time Protocol (RTP) packets in a Voice over Internet Protocol (VoIP) system including an Internet Protocol-Private Branch Exchange (IP-PBX). It is possible automatically detects call startup and/or termination from an RTP packet or an RTP Control Protocol (RTCP) packet provided through a mirroring port of the IP-PBX, and based on the detection of call startup and/or termination, automatically mirrors the RTP/RTCP packet, which is transmitted/received due to call establishment. | 12-17-2009 |
| 20090310600 | Personal Control of Address Assignments & Greeting Options for Multiple BRG Ports - A method and apparatus for providing multiple telephone lines using a single directory number. A method and apparatus for associating multiple directory numbers with multiple telephone lines. A broadband residential gateway (BRG) is a user interface to a broadband communication system providing packetized telephone service and other media services. The BRG has multiple ports, and each port is connected to one or more telephones. The multiple ports may be mapped to a single directory number, or the multiple ports may be mapped to multiple directory numbers. The BRG can provide greeting and message features. A greeting may instruct a caller to select a port which is associated with a party the caller is attempting to reach. Also, a message, played after the greeting, may further instruct the caller. | 12-17-2009 |
| 20090310601 | COMMUNICATION CONTROL DEVICE, COMMUNICATION TERMINAL DEVICE, COMMUNICATION SYSTEM, AND COMMUNICATION CONTROL METHOD - A communication control device includes: a network communication unit connected to plural communication terminal devices via a network that enables communication in a booked band for communication with the plural communication terminal devices; and a control unit that, when a connection request is made from the communication terminal device but the band is not assignable, changes the band with the communication terminal device during communication to secure the band for communication with the communication terminal device that has made the connection request. | 12-17-2009 |
| 20090310602 | MAPPING OF IP PHONES FOR E911 - A system including a first network configured to receive IP device data from an IP device and to provide one or more IP addresses of the IP device based on the IP device data; a second network comprising: a second network location database configured to store physical location information, and a second network location server configured to receive the one or more IP addresses of the IP device from the first network, and query the second network location database to determine physical location information of the IP device based on the one or more IP addresses. | 12-17-2009 |
| 20090310603 | Memory Optimization Packet Loss Concealment in a Voice Over Packet Network - A method to reduce memory requirements for a packet loss concealment algorithm in the event of packet loss in a receiver of pulse code modulated voice signals. A voice playout unit in the receiver shares its nominal delay buffer with a history buffer of a packet loss concealment algorithm up to a maximum limit described in a standard. This reduces or eliminates need to allocate memory for the history buffer. A history buffer can also be extended to retain an original portion of voice signal packets received prior to a packet loss as well as generated voice signals as they are generated. A scratch buffer is used as a working buffer and replaces the function of a pitch buffer. | 12-17-2009 |
| 20090310604 | Method for Service Processor Discrimination and Precedence in a Voice-Over-Internet-Protocol (VoIP) Network - A method and apparatus for identifying and prioritizing applications and application servers in a Voice over IP network is disclosed. In a first embodiment, elements of signaling information are extracted from a call and are mapped to parameters associated with the call. These mapped parameters are then used by a service broker in a VoIP network to identify one or more application servers adapted to process the values of the respective parameter. The service broker may illustratively identify the application servers by a pointer to permit flexible reassignment of processing of a given parameter. The matched pointer/parameter combinations are then mapped to a precedence index. Then, according to this precedence index, the aforementioned pointers are mapped to specific addresses of application servers and the elements of signaling information are forwarded to those addresses for processing of applications. | 12-17-2009 |
| 20090316683 | METHOD FOR SETTING UP AN EMERGENCY CALL IN A COMPUTER LOCAL AREA NETWORK, TERMINAL AND SERVER FOR IMPLEMENTING THE METHOD - If a given terminal (IPP | 12-24-2009 |
| 20090316684 | Method for a Network Component to Route a Communication Session - A network node is provided. The network node includes a component configured to use a value in a Session Initiation Protocol message. The value indicates a supported transport addressing scheme and is used to determine whether to route a related communication session through a transport addressing scheme translation component. | 12-24-2009 |
| 20090316685 | Communication system - A method of communicating user participation status information for a communication event in a communication system is provided. The method comprises: transmitting a group communication event connection request from one user of the communication system to a plurality of second users of the communication system; detecting if at least one of said second users has established a communication event connection in response to receiving the request; generating a notification message indicating the participation status of said at least one second user, wherein said participation status indicates if said at least one second user has established a communication event connection in response to receiving the request; and transmitting the notification message to at least one other of said second users. | 12-24-2009 |
| 20090316686 | Communication system - A method is provided of authorising a user of a communication system to be added to a group communication event. The method comprises: selecting a group of users of the communication system; initiating from a host node the group communication event with the group of users; responsive to receiving a group communication acceptance from at least a first user in the group, establishing the group communication event with the first user in the group; receiving at the host node a communication set up request from another user of the communication system; analysing the communication set up request to determine if said communication set up request is associated with said group communication event initiated by the host node; and adding said other user to the group communication event if it is determined that the communication set up request is associated with said group communication event. | 12-24-2009 |
| 20090316687 | PEER TO PEER INBOUND CONTACT CENTER - A system and method for implementing a contact center on a device node connected to a data network. The system includes a peer-to-peer inbound contact center system that executes in each device node to enable peer-to-peer connections between users making interaction requests at a requesting device and a destination interaction endpoint. Device nodes may be VoIP telephones, computers having soft-phones, computers having a CTI-enabled PBX interface to implement CTI-enabled telephones as interaction endpoints. | 12-24-2009 |
| 20090316688 | METHOD FOR CONTROLLING ADVANCED MULTIMEDIA FEATURES AND SUPPLEMTARY SERVICES IN SIP-BASED PHONES AND A SYSTEM EMPLOYING THEREOF - A method for controlling the advanced multimedia features and supplementary services, such as integration with Internet TV powered by online advertising including interactive video, banner, text ads, online tracking tool which tracks users behaviour, along with various features of communication and infotainment like Phone calls including PC to PC, PC to land line, PC to mobile and vice versa, Video Phone, Chat & TV (Internet Television), on-line shopping/store with facilities like searching of web, follow me facility, world clock and adding of favorites etc that are implemented within Internet Protocol (IP) based telephony technology using Session Initiation Protocol (SIP) for its communications. Moreover the present application provides an interactive solution by offering free talk time to the users based on the duration of watching of the internet television, using and interacting with the disclosed soft phone and also incorporating some viewer friendly solutions for entertainment, communication across the globe. | 12-24-2009 |
| 20090316689 | JITTER BUFFER AND JITTER BUFFER CONTROLLING METHOD - A jitter buffer controlling method includes a data writing step, a data buffering step and a data reading step. The data writing step and the data reading step are executed synchronously and repeatedly. The data writing step includes detecting whether a data packet that comprises a series of voice data frames is normally received, and calculating a storage address for each of the voice data frames. The data buffering step includes buffering the voice data frames, and storing each of the voice data frames in a corresponding storage address calculated in the data writing step. The data reading step includes transmitting the voice data frames to a voice digital signal processor (VDSP) for playing. | 12-24-2009 |
| 20090316690 | Method for Delivering Device and Server Capabilities - A method is provided for delivering the capabilities of user agents. The method includes a user agent sending a session initiation protocol (SIP) message containing a Contact Header containing a Push Resource Identifier feature tag containing at least one push resource. | 12-24-2009 |
| 20090316691 | METHOD AND APPARATUS FOR ENABLING PEER-TO-PEER COMMUNICATION BETWEEN ENDPOINTS ON A PER CALL BASIS - A method and apparatus for enabling a user to signal to the network that a call to be initiated or a call that is in progress needs to occur in a peer-to-peer relationship with the terminating endpoint. The network will then remove itself from the call signaling and media path and direct the signaling and media communication to occur directly between the two endpoints. | 12-24-2009 |
| 20090316692 | UNIFIED RECEPTION AND PROCESSING OF MULTI-PROTOCOL COMMUNICATION SERVICES - A method and an apparatus and server for the receipt of a message addressed to a single identifier for forwarding to a customer is described in which the message uses one of a plurality of message formats, The method comprises receiving the message at a receiving one of a plurality of receivers in one or more of a plurality of telecommunications networks in accordance with the one of the plurality of message formats, wherein the message uses one of a plurality of message formats, the one of the plurality of message formats being independent of the single identifier, passing the message from the receiving one of the plurality of receivers to a central platform and forwarding the message from the central platform to the customer, wherein the single identifier is chosen from a plurality of identifiers provided to the central platform by the one or more telecommunications networks, the single identifier being assigned to the customer. The apparatus comprises a plurality of receivers for receiving the message, a central platform connected to the plurality of receivers, and a connection to the customer for forwarding the message from the central platform to the customer. The central platform comprises a central server and a central database server comprises a database for managing the identifiers. | 12-24-2009 |
| 20090316693 | Convergence of Ancillary Call Services Across Multiple Communication Domains - A method for communication in an environment including a circuit-switched network and a packet-switched network, both of which include a respective connectivity layer including one or more switching elements and a respective service layer including one or more service platforms. A request is accepted to set up a call for a communication terminal associated with one or more of the networks. The call is established responsively to the request via one or more of the switching elements. At least one service platform in the service layer of the circuit-switched network is invoked to provide a first ancillary call service to the call, and at least one second service platform in the service layer of the packet-switched network is invoked to provide a second ancillary call service to the call. | 12-24-2009 |
| 20090323670 | Systems and Methods to Facilitate Searches of Communication References - Methods and apparatuses to facilitate searches of communication references for real time communication connections. One embodiment includes: one or more web servers to assign a communication reference to an advisor for distribution by the advisor in one or more documents, to associate at least one keyword with the communication reference, to receive from the advisor a bid price on the keyword associated with the communication reference, and to present the communication reference selected based at least in part on the bid price in response to a search related to the keyword; a session border controller to interface with a packet switched network; and one or more telecommunication servers to determine contact information of the advisor based on the communication reference used by a customer to request a communication connection to the advisor, and to connect the customer to the advisor for real time communications using the determined contact information. | 12-31-2009 |
| 20090323671 | METHOD FOR DETERMINING RLC DATA PDU SIZE IN WIRELESS COMMUNICATIONS SYSTEM ACCORDING TO CONTROL DATA - A method of determining a size of Data PDUs of an RLC AM entity includes: (a) utilizing a MAC layer to set a transmission payload size; (b) determining whether the transmission payload size is larger than or equal to at least a Control PDU size; (c) when the transmission payload size is larger than or equal to the Control PDU size, submitting the Control PDU to the MAC layer; (d) adjusting the transmission payload size by subtracting the size of the submitted Control PDU; (e) repeating steps (b), (c) and (d) for all Control PDUs; and (f) utilizing a final adjusted transmission payload size in step (d) to determine a size of a Data PDU. | 12-31-2009 |
| 20090323672 | Techniques to enable emergency services in an unauthenticated state on wireless networks - An embodiment of the present invention provides a method of enabling emergency services in an unauthenticated state on wireless networks, comprising attempting Extensible Authentication Protocol (EAP) authentication with a public user account by a client whose identity indicates the need to place an emergency call, authenticating the client by a Subscription Service Provider Network's (SSPN's) authentication, authorization and accounting (AAA) server and providing keying material to an authenticator and supplicant, thereby securing wireless link, providing by the SSPN's AAA server a virtual local area network identification (VLAN ID) back to an access point (AP), performing by the AP or a distribution system (DS) infrastructure a per-user policing for the VLAN ID ensuring upper-limit on resource usage commensurate with an emergency call, and routing the emergency call to a Public Safety Answering Point (PSAP) by the SSPN's call manager. | 12-31-2009 |
| 20090323673 | Portable Soft Phone - A communication device ( | 12-31-2009 |
| 20090323674 | METHOD FOR ESTABLISHING A TELEPHONE CONNECTION - The invention relates to a method for establishing a telephone connection between a caller and a party to be called via a service provider. According to said method, the caller establishes an Internet connection to the service provider with the aid of a data processing unit and transmits information identifying both the caller and the party to be called to the service provider via the Internet connection. In response, the service provider makes available a transmission path for the telephone connection, by analysing at least the information identifying either the caller or the party to be called in accordance with a predefined criterion and by transmitting selected information, depending on the analysis result, to at least one of the parties via the Internet connection and/or the transmission path. | 12-31-2009 |
| 20090323675 | METHOD FOR IMPLEMENTING DISTRIBUTED VOICE FUNCTIONS INTO SOFTWARE APPLICATIONS - A system includes application software that issues voice function requests to one or more web services server. A web services server receives the requests from the application software. In response to the voice function request, the web services server selects at least one to perform one or more actions to provide the voice function request and issues implementation specific messages to the selected device or devices to perform the actions. | 12-31-2009 |
| 20090323676 | NETWORK ADDRESS TRANSLATION DEVICE AND PACKET PROCESSING METHOD THEREOF - A network address translation device for processing session initiation protocol (SIP) packet is provided. The network address translation device receives a first SIP packet and a second SIP packet. The first SIP packet at least includes a former part of a message and the second SIP packet includes a latter part of the message. The network address translation device further obtains the former part of the message from the first SIP packet, reassembles the second SIP packet by combining the latter part with the obtained former part of the message from the first SIP packet, and translates and sends out the first SIP packet and the reassembled second SIP packet. | 12-31-2009 |
| 20090323677 | SEPARATION OF VALIDATION SERVICES IN VOIP ADDRESS DISCOVERY SYSTEM - In one embodiment, an apparatus may receive at least one call attribute of a public switched telephone network (PSTN) call initiated to a destination telephone number. The apparatus may verify a destination Voice-over-Internet-Protocol (VoIP) call agent for the destination telephone number based on demonstrated knowledge of the PSTN call. The apparatus may transmit an indication the destination VoIP call agent is verified for the destination telephone number. | 12-31-2009 |
| 20090323678 | SYSTEM AND METHOD FOR ALLOCATING SESSION INITIATION PROTOCOL (SIP) IDENTIFICATIONS (IDs) TO USER AGENTS - A communications system includes a Session Initiation Protocol (SIP) user agent. A server communicates with the SIP user agent and allocates an SIP ID for the user agent for subsequent communications using SIP. A database can be associated with the server and contain data relating to free SIP ID's that can be allocated to the SIP user agent and allocated SIP ID's. | 12-31-2009 |
| 20090323679 | SYSTEMS, PROCESSES AND INTEGRATED CIRCUITS FOR RATE AND/OR DIVERSITY ADAPTATION FOR PACKET COMMUNICATIONS - Packets of real-time information are sent with a source rate greater than zero kilobits per second, and a time or path or combined time/path diversity rate initially being zero kilobits per second. This results in a quality of service QoS, optionally measured at the sender or the receiver. When the QoS is on an unacceptable side of a threshold of acceptability, the sender sends diversity packets at an increased rate. Increasing the diversity rate while either reducing or maintaining the overall transmission rate is new. CELP-based multiple-description data partitioning sends the base or important information plus a subset of fixed excitation in one packet and sends the base or important information plus the complementary subset of fixed excitation in another packet. Reconstruction produces acceptable quality when only one of the two packets is received and better quality when both packets are received. Reconstruction provides for single and multiple lost packets. | 12-31-2009 |
| 20090323680 | Hierarchical data collection network supporting packetized voice communications among wireless terminals and telephones - A packet-based, hierarchical communication system, arranged in a spanning tree configuration, is described in which wired and wireless communication networks exhibiting substantially different characteristics are employed in an overall scheme to link portable or mobile computing devices. The network accommodates real time voice transmission both through dedicated, scheduled bandwidth and through a packet-based routing within the confines and constraints of a data network. Conversion and call processing circuitry is also disclosed which enables access devices and personal computers to adapt voice information between analog voice stream and digital voice packet formats as proves necessary. Routing pathways include wireless spanning tree networks, wide area networks, telephone switching networks, internet, etc., in a manner virtually transparent to the user. A voice session and associate call setup simulates that of conventional telephone switching network, providing well-understood functionality common to any mobile, remote or stationary terminal, phone, computer, etc. | 12-31-2009 |
| 20100002680 | VOIP LINE SEIZURE SYSTEM AND METHOD - A system for yielding control of a network to a device configured to operate on a PSTN. The system includes a network configured to couple one or more devices to a PSTN, and a PSTN telephone, a PSTN security system, and an ATA and modem coupled to the network. The ATA and modem are configured to provide a VoIP interface between the network and the Internet and to provide a dial tone to the network. An access detector is coupled to the network to detect when the security system attempts to use the network. | 01-07-2010 |
| 20100002681 | Techniques for enhanced persistent scheduling with efficient link adaptation capability - An embodiment of the present invention provides a method, comprising, enhancing persistent scheduling with efficient link adaptation capability by grouping Voice over internet Protocol (VoIP) users and using an intelligent bitmap mechanism to compactly represent persistent allocations for the users within the group. | 01-07-2010 |
| 20100002682 | INTERWORKING METHOD AND INTERWORKING CONTROL UNIT, METHOD AND SYSTEM FOR IMPLEMENTING SIMULATION SERVICES - The present invention discloses a method for implementing simulation services, including the following: an interworking control unit obtains the CS domain network user identifier after detecting that the call signaling message from the CS domain network carries no CS domain network user identifier, puts the obtained user identifier into the SIP message, which is subsequently sent to the IMS network; the IMS network processes the PSTN/ISDN simulation services according to the user identifier. The present invention also discloses a system for implementing simulation service, The present invention makes it unnecessary to extend the existing SIP protocol and to perform signaling interaction between the IMS network and the interworking control unit for obtaining the user identifier, thus simplifying the implementation of the PSTN/ISDN simulation services. | 01-07-2010 |
| 20100002683 | CLOCK SKEW COMPENSATION - A method and arrangement in a receiving communication device for compensating for the difference between the clock-frequency controlled sample rate of the receiving device and the sample rate of a sending communication device. The sending device transmits packets comprising M audio samples to be stored in a buffer in the receiving device accommodating at least 2·M samples before play-out. An estimation of the clock skew is continuously updated from a calculated accumulated difference between an expected and an actual point of time of reception of the M audio samples. Before play-out, an adjusted number N of audio samples to be read from the buffer before play-out is calculated using the estimated clock skew. Thereafter, the N audio samples are resampled by interpolation to M audio samples to play-out. | 01-07-2010 |
| 20100002684 | CALL PROCESSING METHOD AND APPARATUS IN VOIP SYSTEM - A Voice over Internet Protocol (VoIP) system includes at least one Internet Protocol (IP) terminal setting up a call through a switching system according to a signaling protocol, and at least one computer terminal capable of remotely accessing the IP terminal. The IP terminal sets up the call by exchanging a signaling message with a counterpart IP terminal when the computer terminal has remotely accessed the IP terminal, by transmitting a packet received through the call to the computer terminal, and by transmitting a packet received from the computer terminal to the counterpart IP terminal. A subscriber can receive and originate a call by accessing the subscriber's IP terminal irrespective of the subscriber's location. | 01-07-2010 |
| 20100002685 | METHOD AND SYSTEM FOR PROVIDING COMMUNICATION - A communication system, the system including: (i) a first network interface for communicating with a remote system over a network, wherein the first network interface is configured to: (a) receive a conversation initiation request generated in response to an interaction with a conversation trigger that is included in a web page that is displayed at a remote system; wherein the conversation initiation request includes context metadata that pertains to content of the web-page; and (b) provide to the remote system a communication widget that is configured in response to the context metadata; and (ii) a management unit, configured to initiate a communication session between the communication widget and a recipient, for transmitting conversation signals between the remote system and the recipient. | 01-07-2010 |
| 20100002686 | RESTRICTION OF COMMUNICATION IN VOIP ADDRESS DISCOVERY SYSTEM - In one embodiment, a system is provided to restrict VoIP communication. The system may validate a Voice over Internet Protocol (VoIP) call initiation message based on demonstrated knowledge of a Public Switched Telephone Network (PSTN) call. | 01-07-2010 |
| 20100002687 | INTEGRATION OF VOIP ADDRESS DISCOVERY WITH PBXs - A system for verifying VoIP call routing information. The system may include an apparatus integrated with a private branch exchange (PBX). The apparatus may store at least one call attribute of a public switched telephone network (PSTN) call initiated to a destination telephone number. The apparatus may verify a destination Voice-over-Internet-Protocol (VoIP) call agent for the destination telephone number based on demonstrated knowledge of the PSTN call. The apparatus may route a new call either over a VoIP network to the destination VoIP call agent or over a circuit switched network based on whether the destination VoIP call agent is verified for the destination telephone number. | 01-07-2010 |
| 20100002688 | QoS CONTROL SYSTEM AND METHOD OF VoIP MEDIA PACKET RECEIVED FROM BROADBAND PORT IN ROUTER/GATEWAY-INTEGRATED VoIP SYSTEM - A Quality-of-Service (QoS) control system and method of a Voice over Internet Protocol (VoIP) packet received from a broadband port in a router/gateway integrated VoIP system, which can process an incoming VoIP call by detecting in real-time an available bandwidth of the VoIP packet through interaction with a QoS module, determining whether to allow the VoIP call based on the result of the detection, and responding to the VoIP call based on the result of the determination. The QoS can be ensured according to the size of a VoIP media packet received through a broadband port. | 01-07-2010 |
| 20100002689 | VOICE OVER IP ADAPTER - A headset and adapter that converts between Voice Over Internet Protocol (VOIP) network of a contact center and an audio signal received and generated by an agent of the contact center wearing the headset is disclosed. A VoIP terminal receives and transmits VoIP instructions and application program interface instructions over a contact center network. An audio terminal receives an audio signal from a microphone and transmits an audio signal to a speaker. A processor of the adapter converts a received VoIP signal to the transmitted audio signal and converts the received audio signal to a transmitted VoIP instruction. | 01-07-2010 |
| 20100002690 | NETWORK TELEPHONY APPLIANCE AND SYSTEM FOR INTER/INTRANET TELEPHONY - A network appliance ( | 01-07-2010 |
| 20100002691 | METHOD AND APPARATUS FOR PROVIDING ASYNCHRONOUS AUDIO MESSAGING - The present invention provides audio messaging in a communications network, e.g., a VoIP network. More specifically, the present invention establishes a non-duplex communication link between a first subscriber and a second subscriber. Audio messages are transmitted between the first subscriber and the second subscriber via the non-duplex communication link. | 01-07-2010 |
| 20100008352 | Methods and Apparatus for Registering or Deregistering a User to or From an IP Multimedia Subsystem - A method of registering or deregistering a user to or from an IP Multimedia Subsystem (IMS) network. A SIP Application Server (SIP-AS) performs a registration or deregistration with the IMS network on a user's behalf. The registration or deregistration is performed over one of the following interfaces: a Service Control Interface (ISC) with a Serving Call State Control Function (S-CSCF), a Gm interface with a Proxy CSCF (P-CSCF), or an Ma interface with an Interrogating CSCF (I-CSCF). | 01-14-2010 |
| 20100008353 | METHOD AND SYSTEM FOR ENDING-CALL ANCHORING OF CIRCUIT SWITCHED DOMAIN - The present invention discloses a method and a system for call termination anchoring of CS (Circuit Switching) domain, the method including the following steps: when a call coming from CS domain network reaches the mobile switch center in the home circuit switched domain of a voice call continuity subscriber, the mobile switch center sends a LOCREQ message to a home location register to query a location; Step | 01-14-2010 |
| 20100008354 | METHOD FOR BIDIRECTIONAL DATA TRANSMISSION VIA A PACKET-ORIENTED NETWORK DEVICE - A telecommunication system for bidirectional data transmission of a data set between a data transmission device and a data reception device via at least one packet-oriented network device, which includes encapsulation of the data set to enable a connection-oriented data transmission of the data set; connection-oriented transmission of the encapsulated data set by means of at least one mobile telephone from the data transmission device to a base station of a mobile telephone network; evaluation of the data encapsulation protocol in the base station for an unpacking of the data set to enable a packet-oriented data transmission of the data set; and packet-oriented transmission of the data set from the base station to the data reception device. | 01-14-2010 |
| 20100008355 | Method And System For Computer-Based Private Branch Exchange - A computer-based distributed private branch exchange (PBX). Preferred embodiments route calls and perform other functions of a PBX as well as performing services not commonly available on a PBX, such as Internet telephony. In one embodiment, the invention control and operations is distributed among several computers or Personal Computers (PCs) on a computer network. | 01-14-2010 |
| 20100008356 | METHODS AND SYSTEMS FOR PRESENCE-BASED TELEPHONY COMMUNICATIONS - A system and method can enable a user of a communications network, such as a Public Switched Telephone Network (PSTN), wireless and/or voice over IP network to participate in Presence Availability Management (PAM) and Instant Messaging (IM) activities of a PAM/IM network. In response to phone network triggers, a phone network Service Control Point (SCP) can generate requests to a web server. The web server can translate the requests to presence information that can be forwarded to presence user agents for participants of the PAM/IM network. The presence user agents can present the user's presence information to participants having the user on their “buddy list”. In turn, the presence user agent for the user can forward the presence information for participants on the user's “buddy list” to a media server that can communicate the information to the user through Automatic Speech Recognition, Text to Speech and/or Dual Tone MultiFrequency technology | 01-14-2010 |
| 20100014506 | SYSTEM AND METHOD FOR SELECTIVELY PROVISIONING TELECOMMUNICATIONS SERVICES BETWEEN AN ACCESS POINT AND A TELECOMMUNICATIONS NETWORK BASED ON LANDLINE TELEPHONE DETECTION - A method and system for reducing network load by selectively provisioning connections between an access point and a carrier network is disclosed. The access point supports voice and data connections over an IP network. The access point includes a network connection and a telephone connector capable of connecting to a standard landline telephone. The access point also includes at least one detection component that detects whether a landline telephone is plugged in to the telephone connector. The access point is configured to provision a voice connection through the IP network when it detects that a landline telephone is plugged in. | 01-21-2010 |
| 20100014507 | SYSTEM AND METHOD FOR SELECTIVELY PROVISIONING TELECOMMUNICATIONS SERVICES BETWEEN AN ACCESS POINT AND A TELECOMMUNICATIONS NETWORK USING A SUBSCRIBER IDENTIFIER - A method and system for reducing network load by selectively provisioning services between an access point and a carrier network is disclosed. The access point supports telecommunications services over an IP network. The access point includes a network connection and a telephone connector capable of connecting to a standard landline telephone. The access point also includes at least one detection component that detects whether a landline telephone is plugged in to the telephone connector. The access point is configured to provision a telecommunications services through the IP network when it detects that an identification module is present. | 01-21-2010 |
| 20100014508 | Method And System For Emergency Call - An emergency call includes sending, by a Call Session Control Function entity (CSCF) to User Equipment (UE), a session reject message containing indication information indicating re-initiating an emergence call according to a local policy upon detecting that a session request sent by the UE is an emergency session request; and re-initiating, by the UE, an emergency call of CS domain or Internet Protocol Multimedia Subsystem (IMS) domain according to the indication information. A system for an emergency call includes UE and a CSCF. The CSCF includes an emergency call determination unit and an emergency call domain selection unit. Upon detecting that a call is an emergency call, the CSCF instructs the UE to re-initiate an emergency call of CS or IMS domain according to whether a network supports the emergency call of CS or IMS domain. | 01-21-2010 |
| 20100014509 | Digital telecommunications system, program product for, and method of managing such a system - A digital telecommunications system, a method of managing a communications network in such a system and a program product for managing audio transmission in a digital communications system. A softswitch manages communications between devices at network endpoints, e.g., session initiation protocol (SIP) devices, and detects when communications include a non-human, e.g., an audio system, at an endpoint. The softswitch selects conversational communications for calls between voice devices and messaging communications parameters with lower overhead for communications with an audio system, e.g., messaging systems such as voice mail. | 01-21-2010 |
| 20100014510 | PACKET BASED COMMUNICATIONS - This invention is applicable to packet based, rate limited radio links, such as satellite or terrestrial wireless digital communications systems. These communications networks concurrently carry time-critical traffic, such as voice or multimedia, and non time-critical traffic, such as generic data traffic, between two or more communication end points. The communication end points may be connected through a number of heterogenous networks and the end to end throughput characteristics may vary over time. A first aspect of the invention concerns a method for generating packets. In other aspects the invention concerns a computer system for use in packet based communications, a computer protocol for packet based communications and a communications packet. The invention involves determining a “time slice” packet size from the link speed and the interval of time extending between the times at which packets are selected for output from a buffer to the transmission interface. It also involves creating a network packet from frames of time-critical data generated during the interval, where the packet is synchronised to both existing timing requirements of the time-critical frames and the link speed. Then, adding non time-critical data to the network packet if its size had not exceeded the determined “time slice” packet size. | 01-21-2010 |
| 20100014511 | CALL CENTERS FOR PROVIDING CUSTOMER SERVICES IN A TELECOMMUNICATIONS NETWORK - A software-based distributed architecture allows rapid provisioning and flexible management of fault-tolerant call centers for interaction between companies' agents and outside customers via multi-media messages, using both real time and non-real time messages. The real time messages include web-based chat, forms and applications sharing, PSTN calls, and incoming and outgoing Voice over IP calls. The non-real time messages include web call-back requests, voice messages, fax messages, and email messages. The architecture provides for sharing of non-dedicated resources among multiple companies, mirrored hot backup, dynamic resource provisioning and allocation, dynamic load balancing, and implementation of service controls on individual companies in accordance with subscription service limits. | 01-21-2010 |
| 20100014512 | IP TELEPHONE SYSTEM AND CALLING METHOD - An IP telephone number query system includes a terminal, a Web server, and an ENUM server. The terminal displays a call recipient profile hypertext markup language (html) that is assigned a HTML document file name. The Web server includes a phonebook searcher that has a plurality of call recipient profile htmls, and returns a selected call recipient profile html in response to a request from the terminal. The ENUM server has a database, a query issuer and a reversed query issuer. The database stores a plurality of NAPTR resource records in association with an ENUM domain name, each NAPTR resource record containing a URI that at least includes a telephone number and a HTML document file name. The query issuer searches the database in response to a query by an ENUM domain name and returns a NAPTR resource record corresponding to the ENUM domain name. The reversed query issuer searches the database in response to a query by a URI of a HTML document file name and returns a URI of a telephone number corresponding to the ENUM domain name having the URI of the HTML document file name. | 01-21-2010 |
| 20100020788 | Method for Establishing Multimedia Connections Across the Borders of Packet-Switching Communications Networks - The invention relates to a method for establishing multimedia connections across the borders of packet-switching communications networks according to an Internet protocol and the ITU-Standard H.323, consisting in inserting (connect) a rearwardly pointing authorisation cycle into a standard connection set-up, thereby making it possible to overcome in a simple manner the FIREWALLS restrictions for multimedia connections, in particular voice connections, over IP. | 01-28-2010 |
| 20100020789 | Provision of Telecommunication Services - An apparatus ( | 01-28-2010 |
| 20100020790 | SERVICE ADAPTATION IN AN IP MULTIMEDIA SUBSYSTEM NETWORK - A method of operating a Call Session Control Function node within an IP Multimedia Subsystem network. The method comprises establishing a first session corresponding to a first IP Multimedia Subsystem communication service using a first Application Server, receiving a request for a further session corresponding to a further IP Multimedia Subsystem communication service, and forwarding said request to a further Application Server. Said further Application Server is additionally notified that said first communication service is ongoing and of the nature of said first session. | 01-28-2010 |
| 20100020791 | Method and System for a Gigabit Ethernet IP Telephone Chip with No DSP Core, Which Uses a RISC Core With Instruction Extensions to Support Voice Processing - Methods and systems for processing data are disclosed and may comprise receiving packetized data comprising voice data and network data via an Ethernet switch integrated within a single gigabit Ethernet IP phone chip. The received packetized data may be processed via a single main processor core integrated within the single gigabit Ethernet IP phone chip. The single main processor core may comprise circuitry that is controlled by an instruction set for handling processing of the voice data for a plurality of voice channels without the use of a separate DSP. It may be determined whether data to be processed by the single main processor core is voice data or network data. If the data to be processed by the single main processor core is voice data, at least one modified instruction may be selected from the modified instruction set for processing the voice data. | 01-28-2010 |
| 20100020792 | Media Proxy Able to Detect Blocking - A media proxy receive a first message from a near end of a path of a communications session, and before receiving a corresponding message from a far end, the media proxy is arranged to detect a blocking situation where another device in the path is awaiting the first message before forwarding the corresponding message. Detecting such a blocking situation enables it to be overcome, and enables the communication session to proceed. The media proxy can send a probe message to discover if there is another media proxy along the path causing the blocking. This is useful where the only information about the far end is the media path which is in the call set up, e.g. IP address and port. Sending the probe message can be under the control of a call server. | 01-28-2010 |
| 20100020793 | METHOD AND APPARATUS FOR USING A SINGLE LOCAL PHONE NUMBER FOR ROUTING OUT OF AREA PHONE NUMBERS - A method and apparatus for providing a single shadow number to be associated with one or more out of area phone numbers that have registered service addresses in the same local area. For instance, if multiple subscribers with service addresses within the same local calling area choose to use out of area phone numbers, these multiple out of area phone numbers will all be associated with a single shadow phone number that is local within the local calling area. In one embodiment, when a subscriber using an out of area phone number places an E911 call, the out of area phone number as well as the associated shadow number will be sent to the E911 PSAP. | 01-28-2010 |
| 20100027528 | Notification of Impending Media Gateway Resource Exhaustion - A method is disclosed that enables a media gateway controller to optimize the selection of a media gateway from which to acquire call-related resources, in a multi-gateway environment. In accordance with the illustrative embodiment, the controller sets a high utilization threshold and a low utilization threshold for each media gateway it controls, for the purpose of receiving a notification when a threshold is crossed. As resources are utilized, removed from service, or become available for use, the media gateway recalculates the resource utilization of one or more predetermined resources and notifies the controller if a threshold for a particular resource has been crossed. The controller, in turn, uses the current threshold states as part of the selection of media gateway to serve one or more subsequent calls. The disclosed method can increase the probability of selecting a media gateway with sufficient resources for a successful call completion on the first attempt. | 02-04-2010 |
| 20100027529 | METHODS AND APPARATUS TO CONTROL SYNCHRONIZATION IN VOICE OVER INTERNET PROTOCOL NETWORKS AFTER CATASTROPHES - Example methods and apparatus to control synchronization in voice over Internet protocol (VoIP) networks after catastrophes are disclosed. An example border element comprises a network interface to receive a VoIP network registration request message from a VoIP endpoint, a catastrophe detector to determine whether a catastrophe has been detected, a backoff time module to compute a backoff time using a priority assigned to the VoIP endpoint and an expected number of VoIP network registration request messages, a recovery module to determine whether the VoIP endpoint is currently registered with the VoIP network, and to send a response message having a header representing the backoff time to the VoIP endpoint when the catastrophe has been detected and the VoIP endpoint is not currently registered with a VoIP network, and a signaling processor to process the VoIP network registration request message when the VoIP endpoint is currently registered with the VoIP network. | 02-04-2010 |
| 20100027530 | ADAPTIVE NETWORK PHONE DEVICE AND CONTROL METHOD THEREOF - The present invention relates an adaptive network phone device and control method thereof, in which the adaptive network phone device includes: a storage unit adapted to store a virtual operating system, in which a VoIP software is carried; a control unit adapted to start automatically the virtual operating system and the VoIP software stored in the storage unit; and a voice conversion unit adapted to convert voice signals of senders into data packets and send the data packets to receivers by the Internet, and to convert data packets sent by the receivers into voice signals. The adaptive network phone device and control method thereof works with the network phones by the virtual operating system and the VoIP software carried in the virtual operating system on a computer without installing the VoIP software and the device driver. | 02-04-2010 |
| 20100027531 | COMMUNICATION CONTROL APPARATUS, SYSTEM, METHOD AND PROGRAM - A communication control apparatus includes: a communication control unit connected with a relay apparatus relaying a communication between first and second terminals; a request receiver receiving a group hold request from the first terminal for setting a communication status into a group hold state in which the communication is terminated by the relay apparatus and can be responded by third terminal in a group including the first terminal; a hold direction unit making the relay apparatus change the status into the group hold state, if the group hold request is received by the request receiver; a status information provider providing status information to the third terminal; and a communication starting unit making the relay apparatus start a communication between the second and the third terminal, if the communication starting unit receives a response to the group held communication from the third terminal. | 02-04-2010 |
| 20100027532 | METHODS, SYSTEMS, AND COMPUTER READABLE MEDIA FOR PROVIDING SEDATION SERVICE IN A TELECOMMUNICATIONS NETWORK - Methods, systems, and computer readable media for providing sedation service in a telecommunications network are disclosed. According to one aspect, a method for providing sedation service in a telecommunications network is provided. The method includes steps that are performed at a session initiation protocol (SIP) sedation node. The method includes receiving a first message sent from a SIP user agent and intended for a SIP server. The method further includes determining whether the SIP server is unavailable. The method further includes responsive to a determination that the SIP server is unavailable to respond to the first message, sending, to the SIP client, a SIP sedation message for reducing the number or frequency of messages sent by the SIP user agent to the SIP server. | 02-04-2010 |
| 20100034193 | METHOD AND APPARATUS FOR PROVIDING A NETWORK ASSISTING SWITCH FUNCTION - A method and apparatus for providing a network assisting switch function are disclosed. For example, the method receives a query for feature processing for a call from a switch deployed in a switched network, and determines if the feature processing for the call requires one or more switching services. The method determines if the switch is able to provide the one or more switching services, if the one or more switching services are determined to be required, and initiates a temporary connection to a network assisting switch function in a packet network, if the switch is unable to provide the one or more switching services. | 02-11-2010 |
| 20100034194 | Eliminating unreachable subscribers in voice-over-ip networks - A method detects an unreachable endpoint in a voice over IP network operating according to standard protocol. The endpoints include a first endpoint as a call originator and a second endpoint as a VoIP destination. The endpoints are connectable via a soft-switch. After each call, a check is performed to determine whether the second endpoint responded to the call. If the second endpoint did not respond to the call, a non-call related message found in the standard protocol is sent from the soft-switch to the second endpoint. If the second endpoint does not respond to the non-call related message, the second endpoint is deactivated so that further calls are not | 02-11-2010 |
| 20100034195 | Incremental addition and scale-back of resources adapting to network resource availability - An exemplary method includes receiving at a first network a notice of an intended communication to a called party network, wherein the intended communication requires a resource for supporting a streaming data protocol in each network between a calling party network and the called party network; forwarding the notice of an intended communication to a second network and toward the called party network; in parallel with said forwarding, initiating for the intended communication a determination of resource availability for the first network; performing for the intended communication the determination of resource availability for the first network, wherein the determination is for a first resource for the first network; and verifying resource sufficiency for the intended communication. Verification of resource sufficiency is based on resource, (e.g., bandwidth) availability being greater than a threshold for plural network segment of the calling party to calling network required for the intended call. | 02-11-2010 |
| 20100034196 | RPH mapping and defaulting behavior - An exemplary management method includes receiving at a first network a notice of an intended communication to a called party network, the notice including a first priority indicator, the intended communication requiring a resource for supporting a streaming data protocol in each network between a calling party network and the called party network; forwarding the notice to a second network and toward the called party network, the notice including the first priority indicator; in parallel with said forwarding, mapping the first priority indicator to a second priority indicator and initiating for the intended communication a determination of resource availability for the first network based on the second priority indicator; determining the determination of resource availability for the first network based on the second priority indicator, wherein the determination is for a first resource for the first network; and verifying resource availability for the intended communication. | 02-11-2010 |
| 20100034197 | End-to-end capacity and priority management through multiple packet network segments - Apparatus and method for management of a communications network are provided. An exemplary method includes receiving at a first network a notice of an intended communication to a called party network, wherein the intended communication requires at least one resource for supporting a streaming data protocol between a calling party network and the called party network; verifying resource availability of at least one resource in the first network; and in parallel with said verifying, forwarding the notice of an intended communication to a second network and toward the called party network prior to receiving an indication of resource availability of the at least one resource in the first network required for the intended communication. | 02-11-2010 |
| 20100034198 | METHOD AND GATEWAY FOR ROUTING INTERNATIONAL MOBILE TELEPHONE CALLS - A gateway for routing an international mobile telephone call comprises a storage device and a cost-saving routing module. The storage device is configured to store a mapping table and a call record table. The mapping table records a mobile phone number of a roaming subscriber and a fixed network number, and the call record table records a caller's phone number and the mobile phone number of the roaming subscriber. The cost-saving routing module is configured to establish a connection in accordance with the mapping table and call record table. | 02-11-2010 |
| 20100034199 | METHOD FOR REQUESTING DOMAIN TRANSFER AND TERMINAL AND SERVER THEREOF - A method, terminal and server for controlling a domain transfer operation, are discussed. According to an embodiment, the method includes receiving, by a terminal, a domain transfer request from a network server, the domain transfer request including domain transfer related information; evaluating, by the terminal, the domain transfer related information when deciding whether or not to initiate a domain transfer; determining, by the terminal, whether to initiate the domain transfer of an ongoing call based on the evaluation result; and initiating, by the terminal, the domain transfer of the ongoing call when the evaluated domain transfer related information indicates that the domain transfer of the ongoing call needs to be initiated, wherein the domain transfer is for voice call continuity that is capable of transferring voice calls between a circuit switched (CS) domain and an (IMS) domain. | 02-11-2010 |
| 20100034200 | System and Method for Assisting in Controlling Real-Time Transport Protocol Flow Through Multiple Networks - Methods and systems for routing call signaling messages are disclosed. One such method is performed in a session router. The method includes: maintaining a telephony route information base (TRIB) stored in the session router as a result of participation of the session router in telephony routing over internet protocol (TRIP). The TRIB allows multiple routes to the same destination. The method further comprises: using the TRIB to route the received call signaling messages to another session router. One such system includes memory and a processor. The processor is configured by instructions retrieved from the memory to: build and maintain, as a result of participation of the router in telephony routing over internet protocol (TRIP), a telephony route information base (TRIB) that allows multiple routes to the same destination; and use the TRIB to route a received call signaling message to another router. | 02-11-2010 |
| 20100040046 | VOIP DATA PROCESSING METHOD - A method of processing data in a communication apparatus in a local network is provided. The method comprises receiving, at the communication apparatus, a first Internet Protocol (IP) data packet, comparing at least one bit of leading bytes with a predetermined value, determining the first IP data packet belongs to a control signal data packet and processing the first IP data packet according to the control signal data packet when the bit of leading bytes is less than or equal to the predetermined value, and determining the first IP data packet belongs to a multimedia data packet and processing the first IP data packet according to the multimedia data packet when the bit of leading bytes exceeds the predetermined value. | 02-18-2010 |
| 20100040047 | Loss of Signalling Bearer Transport - Being aware of a loss of signalling bearer transport through an IP Connectivity Access Network is an important issue. Therefore, the present invention relies on amending the Policing and Charging Control model with means to provide the IMS infrastructure with subscriptions to and notifications about signalling session events detected on the signalling IP flow transported through the bearer layer. To this end, a P-CSCF, or AF included therein, is amended to allow the establishment of a signalling session for subscription to notification of bearer level events for a signalling IP flow. Apart from that, new processing rules are required at the AF and PCRF for handling the signalling session, the notification of events and the termination of the signalling session. | 02-18-2010 |
| 20100040048 | Address Resolution in a Communication System - Apparatus for resolving a local TEL Uniform Resource Identifier to a service Uniform Resource Identifier for routing a message over an IP-based communication system. The apparatus comprises first processing means ( | 02-18-2010 |
| 20100040049 | METHODS, SYSTEMS, AND COMPUTER PROGRAM PRODUCTS FOR COMMUNICATING CALLING NAME (CNAM) SERVICES FOR SESSION INITIATION PROTOCOL (SIP) ORIGINATED CALLS TERMINATING IN A CIRCUIT SWITCHED NETWORK - Methods, systems, and computer program products for communicating CNAM services for SIP originated calls terminating in a circuit switched network is described. In one embodiment, the method includes, at a SIP-SS7 gateway, receiving a SIP call setup message that includes a SIP calling subscriber identifier information, associating a temporary telephone number with the SIP calling subscriber identifier information, generating an SS7 call setup message associated with the SIP call setup message, wherein the SS7 call setup message includes the temporary telephone number, and communicating the temporary telephone number and SIP calling subscriber identifier information to a calling name interworking function (CIF) module. The method also includes, at the CIF module, storing the temporary telephone number and the associated SIP calling subscriber identifier information in a local cache, receiving a CNAM query message containing the temporary telephone number from a terminating switching office, and transmitting a CNAM response message to the terminating switching office including the SIP calling subscriber identifier information. | 02-18-2010 |
| 20100040050 | COMMUNICATION SESSION QUALITY INDICATOR - An approach for providing a quality indicator in support of a communication session between a near end station and a far end station over a data network. The quality of the communication session in a direction of the near end station sending to the far end station is determined. A message containing statistics associated with the communication session is transmitted according to a prescribed protocol to the near end station to notify the near end station of the quality of the communication session. The prescribed protocol supports real-time data exchange. The present invention has particular applicability to SIP (Session Initiation Protocol) IP (Internet Protocol) telephony services. | 02-18-2010 |
| 20100040051 | Systems and Methods for Serial Packet Synchronization in a Voice Processing System - A serial packet sync encoder is used to encode a serial packet sync datastream. In an embodiment, the serial packet sync datastream is made up of the packet sync vector and a unique preamble bit sequence that is preselected. In another embodiment, the serial packet sync datastream is made up of a non-unique bit sequence. A serial packet sync transmitter is used to transmit the serial packet sync datastream. A serial packet sync receiver is provided for receiving the serial packet sync datastream. In an embodiment, the serial packet sync transmitter and the serial packet sync receiver are shift registers. In this way, the serial packet sync datastream can be transmitted and received using only a single pin. The serial packet sync datastream is useful for providing an indication that an event, such as a grant arrival, has occurred. A preamble comparator is provided to compare the received serial packet sync datastream and the preselected preamble to determine if the two match. In cases where a match is made, the packet sync vector is written into a holding register for access from other applications and or system components such as a digital signal processor. | 02-18-2010 |
| 20100046499 | APPARATUS FOR A TRADITIONAL TERMINAL TO ACCESS AN IMS SYSTEM AND THE METHOD THEREOF - An apparatus and method for realizing the access of a legacy terminal to an IMS system. The apparatus includes a session control module, a downlink signaling interface function module, a downlink bearer interface function module, an uplink signaling interface function module, an uplink bearer interface function module and a media interworking module. The session control module registers the terminal that has entered service status to the I-CSCF on IMS side. During the session, the uplink signaling interface function module provides SIP signaling interaction with the CSCF function entity of IMS core network; the downlink signaling interface function module provides signaling interaction with the legacy terminal; the media interworking module provides the connection and media adaptation between the uplink bearer interface function module and the downlink bearer interface function module. The invention enables the services of the legacy networks such as PSTN/ISDN and the like to be integrated with those of IMS networks, thus reducing the cost of network construction and operation. | 02-25-2010 |
| 20100046500 | APPARATUS, METHOD AND SYSTEM FOR PROVIDING NEW COMMUNICATION SERVICES OVER EXISTING WIRING - Various embodiments of the invention provides apparatus for providing a next-generation communication system over existing wiring. In one form the apparatus includes an input to receive broadband signals carrying next-generation communication data, a processor to extract the next-generation communication data from the broadband signals and a converter to convert the next-generation communication data into analogue telephone signals. The apparatus is arranged to output the analogue telephone signals at the input of the apparatus. Also described is a related method of providing a next-generation communication system over existing wiring. | 02-25-2010 |
| 20100046501 | METHODS AND APPARATUSES FOR REGISTERING A TERMINAL IN THE IMS OVER A CIRCUIT-SWITCHED ACCESS DOMAIN - The invention provides a solution for registering a terminal having a packet-switched and circuit-switched functionality in a packet-switched service domain, such as the IMS over a circuit-switched access domain. In particular it is proposed to send a packet-switched registration message packed in a circuit-switched transport bearer (USSD) to a circuit node (HLR, MSC, dispatcher) which selects an adapter node (IA) being responsible for performing a registration in the packet-switched service domain on behalf of the user using the information provided with the packet-switched registration message and by deriving and adding additional information. | 02-25-2010 |
| 20100046502 | METHOD AND MEANS FOR ROUTE SELECTION OF A SESSION - In the present invention, it is provided a method and means for selecting a route for a session requested by a calling user equipment in a session manager, and correspondingly, it is provided a method and means for selecting a route for a session requested by a calling user equipment in a media gateway control function, it is characterized as selecting an MGW having relative lighter load to bear the session on a basis of load related information of MGWs. By applying the methods and means of the present invention, load of every MGW is balanced; performance degradation, caused by heavy load, of a certain MGW is avoided; MGW having stopped working is bypassed; a success ratio of session setup is increased; session performance is improved; and benefit is brought to multi-network integration, such as inter-working between a packet switching network and a circuit switching network in an IMS network. | 02-25-2010 |
| 20100046503 | CONCENTRATOR FOR SPEECH TELEPHONES AND METHOD OF COMMUNICATION OVER LAN USING SAME - Speech telephones are incorporated in a LAN, and, for example, when a voice telephone | 02-25-2010 |
| 20100046504 | AUDIO COMMUNICATIONS SYSTEM USING NETWORKING PROTOCOLS - Methods for providing improvement in Voice-over-IP communication systems, and hardware for implementing the methods, are disclosed. A first aspect provides a method of improving on the efficiency of RTP used to transport VoIP voice calls by reducing the overhead of second and subsequent calls on a link to almost zero using trunking. A second aspect uses bandwidth awareness to compress RTP payload data captured from the network. This involves capturing G.711 encoded RTP data directly from the network ( as opposed to at source ) and transcoding that data in such a way as to take account of the available bandwidth on an outbound link. A third aspect uses dynamic and transparent packet fragmentation and reassembly based on RTP interval to reduce VoIP latency and jitter. A fourth aspect uses dynamic re-writing of SIP messages to provides automatic fail-over and load balancing of SIP servers. This involves capturing SIP call set-up messages and re-writing and duplicating them to direct them to multiple servers. The response is monitored to determine which server responds most quickly and allowing only that reply back to the source device. A fifth aspect provides dynamic sizing of trunk payload packets. Given that the above scheme has been set up on a link, it is trivial for the receiving trunk device to determine if the received packets are too big or small, and to signal the transmitter to adjust its payload size accordingly. | 02-25-2010 |
| 20100046505 | Internet Telephony Device and Method of Monitoring User Status - An Internet telephony device is provided, which comprises a voice processing unit capable of processing voice signals into electrical signals and vice versa, and a microprocessor comprising an embedded client application for communicating said electrical signals to an Internet telephony server over a network. The Internet telephony device further comprises at least one motion detector adapted to generate motion signals based on the detection of motion in the vicinity of the Internet telephony device. The motion detector is coupled to the microprocessor to determine user status of the Internet telephony device based on the motion signals. | 02-25-2010 |
| 20100046506 | SYSTEM AND METHOD FOR LOCATION IDENTIFICATION - A telecommunications outlet providing location identification in a local area network, the telecommunications outlet constituted of: a network side connection adapted to be connected to a networking device via horizontal cabling; a data terminal side connection adapted to be connected to a data terminal equipment; a control circuitry; a memory adapted for storage of multi-bit data; a transmitter in communication with the memory; and a first switch responsive to the control circuitry, the first switch arranged in a first mode to connect data from the network side connection to the data terminal side connection and in a second mode to connect data from the transmitter to the network side connection and disconnect data from the network side connection to the data terminal side connection. | 02-25-2010 |
| 20100046507 | USING PSTN REACHABILITY IN ANONYMOUS VERIFICATION OF VOIP CALL ROUTING INFORMATION - In one embodiment, an apparatus may verify an identity of a destination Voice-over-Internet-Protocol (VoIP) call agent for a destination telephone number based on demonstrated knowledge of at least one public switched telephone network (PSTN) call initiated to the destination telephone number. The apparatus may also receive the identity of the destination VoIP call agent based on the demonstrated knowledge of the at least one PSTN call initiated to the destination telephone number. | 02-25-2010 |
| 20100046508 | TIME-SLOT INTERCHANGE CIRCUIT - A circuit and method are presented for signal processing and routing of digital voice telephony signals, using a specialized high-density integrated circuit voice processor. The voice processor performs several essential functions required for telephony processing, including echo cancellation, protocol conversion, and dynamic range compression/expansion. These functions are traditionally performed by multiple circuits or modules. By combining these capabilities in a single device, power and circuit board area requirements are reduced. The embodiment of the circuit and method disclosed herein include novel implementations of a time-slot interchange circuit and a telephony signaling circuit. Both of these circuits are designed to minimize demands on the signal processing engines incorporated within the voice processor, and account for very little of the on-chip circuitry. | 02-25-2010 |
| 20100046509 | CALL CONNECTION METHOD, EQUIPMENT, AND SYSTEM IN IP MULTIMEDIA SUBSYSTEM - A call connection method in an IP multimedia subsystem (IMS) is provided. The method includes the following steps. An entrance network element (NE) of a called network receives a session request carrying called user identification (ID) information from a calling network. When determining that the called user ID information is incomplete, the entrance NE of the called network sends a response message indicating that the called user ID information is incomplete to the calling network. The calling network updates the called user ID information according to the response message, and sends the updated called user ID information to the entrance NE of the called network. An interface NE, a called network system, a call connection system, and a method of informing a call connection failure are also provided. | 02-25-2010 |
| 20100054238 | TELECOMMUNICATION NETWORK, NETWORK NODE DEVICE, AND ROUTING METHOD - There is provided a network node device capable of selecting a route without causing an increase in the size and complexity of the information management system. Input lines ( | 03-04-2010 |
| 20100054239 | DATA NETWORK AND METHOD THEREFORE - A data network comprises proxy-call session control functions (P-CSCFs) serving user equipments. Each P-CSCF can request resource reservation from an associated policy manager. A serving-call session control function receives a first call session setup message and determines a set of terminating user equipments associated with a terminating user identity of the setup message. It then transmits a call session initialization message to each identified terminating user equipment via an associated P-CSCF. This message includes a session identity indication and a forking indication which indicates if the first call session is a forked call session. The P-CSCFs and/or the policy managers then restrict the resource reservation for two or more user equipments having the same session identity and forking indications indicative of a forked call session setup to the resource requirement for only one of the user equipments. This may reduce resource usage for forked call sessions. | 03-04-2010 |
| 20100054240 | Single-Rotator Circulating Switch - Switch elements, each receiving data from external sources and transmitting data to external sinks, are interconnected through a single rotator to form a switching node. The single rotator has a number of inlets equal to the number of switch elements and a number of outlets equal to the number of switch elements. A first set of channels connects the switch elements to inlets of the rotator and a second set of channels connects the outlets of the rotator to the switch elements. The connectivity pattern of the second set of channels is a transposition of the connectivity pattern of the first set of channels in order to preserve sequential data order of switched data. A controller communicatively coupled to the switch elements exchanges timing data with external nodes of a time-coherent network and schedules data transfer among the switch elements. | 03-04-2010 |
| 20100061363 | SYSTEM AND METHOD FOR MEDIA GATEWAY NEGOTIATION - A system and method of negotiating Media Gateways (MGs) between a plurality of call control nodes (CCNs). The system includes a first CCN which builds an original list of identifiers associated with at least one MG capable of being used in a call by the first CCN. The system also includes a second CCN for receiving the original list of identifiers from the first CCN. The second CCN removes from the original list any identifiers associated with any MG in the original list of identifiers which is not capable of being used in the call by the second CCN. The second CCN then forms a modified list of identifiers associated with at least one MG capable of being used in a call by the first CCN and the second CCN. The second CCN also selects a specified MG from the modified list and sends a first backward message from the second CCN to the first CCN identifying the specified MG. The first CCN may then validate that the specified MG is on the original list of identifiers and selects the specified MG for the call. | 03-11-2010 |
| 20100061364 | Home Gateway Device for Providing Multiple Services to Customer Devices - A telecommunication node such as a home gateway and a method of routing data packets received from customer premises devices connected to the node. The node includes an operator-configurable service profile table for storing service profiles and a user-configurable customer devices table for storing the source addresses of the customer premises devices and associations between each source address and at least one of the service profiles. The operator controls service provisioning while the user can freely allocate the customer premises devices to different service profiles and can access a plurality of services from the same device. | 03-11-2010 |
| 20100061365 | METHOD AND APPARATUS FOR PROVIDING EXTENSION MANAGEMENT IN VOICE OVER INTERNET PROTOCOL CUSTOMER PREMISES - A method and apparatus for allowing all the extensions connected to an enhanced Terminal Adaptor (TA) associated with a single phone number to place and receive phone calls independently are disclosed. For example, in the case of a call waiting scenario, if an extension is already engaged in an ongoing phone call, then the enhanced TA provides call waiting handling to the engaged extension similar to traditional call waiting when a subsequent incoming call is received. However, the enhanced TA also rings the remaining extensions that are not currently engaged in phone calls when the subsequent incoming call is received. | 03-11-2010 |
| 20100067519 | METHOD AND APPARATUS FOR PRIORITIZING VOICE OVER INTERNET PROTOCOL SIGNALING MESSAGES - A method and apparatus for enabling prioritization of signaling messages in a communication network are disclosed. For example, the method receives at least one signaling message, and classifies each of the at least one signaling message. The method schedules each of the at least one signaling message for processing, and discards selectively one or more signaling messages that have been scheduled under an overload condition. | 03-18-2010 |
| 20100067520 | INFORMATION COMMUNICATION TERMINAL - A information communication terminal is provided which includes: a voice communication device that transmits and receives voice signals to and from an other telephone equipment via a public switched telephone network; a data communication device that transmits and receives call data signals as digitized voice signals to and from an other terminal via an IP network; a message communication device that transmits and receives data signals of an instant message which contains character information to and from an other terminal via an IP network; and a control device that makes the message communication device transmit the data signals of the instant message to a destination of the call data signals so as to enable communication by voice as well as character information, when one of the voice signals and the call data signals is received while the other is being transmitted and received. | 03-18-2010 |
| 20100067521 | Internet protocol telephone system - An internet protocol telephone includes a substrate having an input and an output that are capable of being connected to the internet protocol (IP) network. A relay is disposed on the substrate and is connected between the input and the output of the substrate. The relay includes first and second native FETs that have a threshold voltage of approximately zero volts. Therefore, the relay is nominally turned-on, even when little or no voltage (or power) is applied to the IP telephone substrate, as during the discovery mode of IP telephone operation. During discovery mode, The IP phone is configured to be responsive to extended link pulses and block data packets that are associated with legacy devices. Data packets have a higher signal duration and are more continuous than extended link pulses. The IP phone includes a switchable ground that is connected to the gates of the native devices, and is controlled by a rectifier and filter circuit that are connected to the substrate input. If the IP phone receives legacy data packets during discovery mode, then the high signal duration and continuous nature of the data packets are sufficient to cause the rectifier to generate a rectified signal having sufficient amplitude to activate the switchable ground, so as to ground the gates of the native devices and therefore turn-off the native devices. Therefore, the data packets are rejected and are not passed back to the switch. Extended link pulses have a frequency that is too low to generate a rectified signal that is sufficient to activate the switchable ground, and therefore the native devices remain turned-on. Accordingly, the extended link pulses are passed back to the switch. | 03-18-2010 |
| 20100067522 | METHOD FOR DETERMINING PACKET TYPE FOR SVC VIDEO BITSTREAM, AND RTP PACKETIZING APPARATUS AND METHOD USING THE SAME - Provided are a method for determining the packet type for a Scalable Video Coded (SVC) video bitstream, and a Real-time Transport Protocol (RTP) packetizing apparatus and method using the same. The method for determining a packet type for a Scalable Video Coded (SVC) video bitstream, which includes the steps of: a) deriving temporal and spatial hierarchy information between Network Abstraction Layer (NAL) units from field information defined in the NAL unit headers of scalable layers; b) detecting the type of encoding information by applying combined scalability encoding to the hierarchical structure of the Scalable Video Coding (SVC); and c) determining a Real-time Transport Protocol (RTP) packet type for the corresponding SVC video bitstream by using the derived temporal and spatial hierarchy information between the NAL units and the detected type of encoding information. | 03-18-2010 |
| 20100067523 | INTERCONNECT NETWORK FOR OPERATION WITHIN A COMMUNICATION NODE - An interconnect network for operation within communication node, wherein the interconnect network may have features including the ability to transfer a variety of communication protocols, scalable bandwidth and reduced down-time. According to one embodiment of the invention, the communication node includes a plurality of I/O channels for coupling information into and out of the node, and the interconnect network includes at least one local interconnect module having local transfer elements for transferring information between the plurality of I/O channels; and scaling elements for expanding the interconnect network to include additional local interconnect modules, such that information can be transferred between the local interconnect modules included in the interconnect network. | 03-18-2010 |
| 20100074247 | METHOD, SYSTEM AND APPARATUS FOR INTELLIGENTLY HANDLING A REQUEST FOR A COMMUNICATION SESSION - According to embodiments of the present invention, there is provided a method, system and apparatus for handling a request for a communication session. The method comprises receiving, at a processing time, a request for a communication session, the request comprising a destination network identifier, the destination network identifier having been registered in association with a plurality of communication clients; the request having been originated by an originating party associated with an originating identifier. The method further comprises identifying, based on at least one of the originating network identifier and the processing time, a subset of the plurality of communication clients. The method further comprises delivering the request to the subset of the plurality of communication clients. | 03-25-2010 |
| 20100074248 | VOICE OVER THE INTERNET METHOD AND SYSTEM - A voice over internet method and system is disclosed to enable radio devices to initiate or terminate a session initiation protocol for transmission of audio data over the internet. A gateway ( | 03-25-2010 |
| 20100074249 | SERVICE CONTINUITY MANAGEMENT IN A NETWORK - A service is provided to a user of a terminal in a network, comprising a service platform and network equipment including session border controllers. Each of the controllers is capable of communicating with the service platform. At least one given session border controller is capable of routing messages received from the terminal, intended for the service platform. This controller decides that the service platform is not accessible on the basis of an accessibility check ( | 03-25-2010 |
| 20100074250 | METHOD AND BROADBAND ACCESS SYSTEM FOR REMOTE-CONTROLLING A VOICE INTERFACE OF AN ACCESS NODE - A method for controlling an access node interface connected to a VoIP server via an IP-based network, wherein subscriber lines connect a plurality of subscriber terminals to the access node, includes storing subscriber-specific data in a memory device associated with the VoIP server, where the data contains information to configure an access node voice interface. The access node determines whether at least one of the plurality of subscriber terminals is connected to the interface. If at least one of the subscribers is connected to the interface, then interface-associated identification data is transmitted from the access node to the VoIP server using an IP-based protocol. In response to the interface identification data received, subscriber-specific data filed for the connected interface is transmitted from the VoIP server to the access node using the IP-based protocol. The access node is configured, using the subscriber-specific data, so that the interface is operated as a voice interface. | 03-25-2010 |
| 20100074251 | AUTOMATIC TERMINATION PATH CONFIGURATION - There is provided herein a system and method for automatic configuration of data routings for use with electronic data such as phone calls, faxes, etc. In an exemplary embodiment, when more than one carrier might potentially terminate the transmission, the carriers are ordered based on some screening criterion (e.g., transmission price). Data transmissions are then assigned to the carriers based on the sorting order, with the second place and lower carriers (e.g., the higher priced carriers) not being selected unless the first carrier cannot complete the transaction. The switch instructions necessary to implement this scheme may be generated automatically. | 03-25-2010 |
| 20100074252 | INTERNET TELEPHONY SYSTEM WITH AUTOMATED CALL ANSWERING - A system and method for automatically answering a call from a calling party to a called party that originates via the Internet, includes and involves a data storage system and processor that is coupled to the data storage system. The processor is operative to initiate an automated call answering service in response to an Internet telephony call from the calling party which is intended to be received by the called party, to receive a message from the calling party via the Internet in accordance with the automated call answering service, and to store the message in the data storage system for processing by the processor in accordance with the automated call answering service. | 03-25-2010 |
| 20100080211 | METHODS AND APPARATUS FOR COMMUNICATING INTERNET PROTOCOL BASED CONTROL SIGNALING THROUGH A COMMUNICATIONS SYSTEM - An embodiment of a method for communicating call control signaling information in a communications system that includes a user equipment (UE) and a base includes the UE formatting the call control signaling information, transmitting the call control signaling information over a first logical channel that is mapped to a first transport channel, and transmitting user traffic over a second logical channel that is mapped to a second transport channel. In an embodiment, the base receives the call control signaling information from the UE over the first logical channel, receives the user traffic from the UE over the second logical channel, and transmits the call control signaling information to a core network. In an embodiment, the communication system is an IP network in which information is exchanged between the UE and the base using a W-CDMA transmission protocol. The base may form a portion of a satellite-based radio network. | 04-01-2010 |
| 20100080212 | IMPAIRMENT REDUCTION FOR TANDEM VOIP CALLS - A method and apparatus are provided for allowing IP endpoints to communicate over a PSTN with improved signal quality. Watermarks are used in the handshaking between the end-points when a communication session is being established, the watermarks indicating that the endpoints are capable of VoIP. If the two end-points establish that they are each VoIP-capable then packet data is inserted into a TDM channel using a framing technique managed by the gateways, with the bearer data being native to the VoIP devices, avoiding the lossy conversion of packet-voice data to 64 kb/s PCM and back to packet data again, realizing that the other end-point will be able to decode the data. If an IP-enabled endpoint determines that the other endpoint is not IP-enabled, then the data is inserted into the TDM channel by the gateway after conversion to 64 kb/s PCM so that the resulting TDM stream remains compatible with the PSTN and non-IP endpoints. | 04-01-2010 |
| 20100080213 | SYSTEMS AND METHODS FOR UTILIZING A SPARE SWITCH IN A DISTRIBUTED VOIP SYSTEM - A distributed VoIP system includes a network and a first switch at a first site coupled to the network. The first switch is configured to provide telephony services to a first communication device. The system also includes a second switch at a second site coupled to the network. The second switch is configured to provide telephony services to a second communication device. The system also includes a spare switch coupled to the network. The spare switch is configured to provide telephony services to the first communication device if the first communication device is unable to register with the first switch, and the spare switch is configured to provide telephony services to the second communication device if the second communication device is unable to register with the second switch. | 04-01-2010 |
| 20100080214 | INTEGRATION OF A PRIVATE CELLULAR SYSTEM INTO A UNIFIED COMMUNICATIONS SOLUTION - In one embodiment, a communication system includes a private cellular base station subsystem to communicate, using a cellular radio frequency air radio interface, with home cellular wireless devices and visiting cellular wireless devices located within a coverage area associated with the private cellular base station subsystem. Each of the home cellular wireless devices having associated therewith (i) a public cellular number from a home public land mobile network, and (ii) a private cellular number from a private network associated with the communication system. The communication system further includes a private cellular switching subsystem to provide cellular switching functionality within the private network for the home cellular wireless devices in connection with sessions that are associated with the respective private cellular numbers of the respective home cellular wireless devices. The communication system further includes unified communications (UC) functionality to interface the private cellular switching subsystem to a unified communications server in order to provide unified communications services using the home cellular wireless devices. | 04-01-2010 |
| 20100080215 | METHOD AND SYSTEM FOR MEASURING MARKET SHARE FOR VOICE OVER INTERNET PROTOCOL CARRIERS - Methods and apparatus to measure market share for voice over Internet protocol (VoIP) carriers is disclosed. An example method includes querying a plurality of VoIP carrier servers to determine the VoIP carrier server that owns the telephone subscriber number and, in response to the querying, receiving a plurality of messages operable to determine whether the telephone subscriber number is found within any one of the plurality of VoIP carrier servers. The example method also includes, when the received plurality of messages is at least one of inconclusive or when the telephone subscriber number is not found within any one of the plurality of VoIP carrier servers, placing a first partial call to the telephone subscriber number from a first VoIP number within a first VoIP carrier network. Further, the example method includes, in response to placing the first partial call, receiving a first signal from the first VoIP carrier network, and based on the first received signal, determining whether the telephone subscriber number belongs to the first VoIP carrier network. | 04-01-2010 |
| 20100080216 | Real-time communication blocking for Dot Not Call" registered information - A real-time call blocking system based on Session Internet Protocol (SIP), e.g., Voice Over Internet Protocol (VoIP) over both wireline and/or wireless systems using relevant Internet Protocol (IP) based systems. This also includes communications originating on traditional legacy or other non-SIP protocols that are converted to SIP somewhere during the call processing (e.g., using a media gateway to terminate a non-SIP device). A Session Internet Protocol (SIP)-based real time communication blocker comprises a do not call database, and a communication blocking proxy to intercept a communication from a commercial source. An intended recipient's identity is compared to entries in the do not call database. The intercepted communication (e.g., phone call, email, short message, etc. is blocked from being routed to an intended recipient if the intended recipient is listed in the do not call database. | 04-01-2010 |
| 20100080217 | Sip Telephone System and Method for Controlling Line Key Display - According to one embodiment, a SIP telephone system includes a plurality of terminals each configured to include a plurality line keys configured to blink or light and distinguish a plurality of lines, and an SIP server apparatus configured to house the plurality of terminals and be connected to a communication network, and establish communication among the terminals via a selected line in a case where an arbitrary line key is selected from among the plurality of line keys by the terminals, wherein the SIP server apparatus includes a transmitter which adds each item of identification information of the plurality of line keys to be selection candidates to a control message, to transmit the identification information to the terminals, and each of the terminals includes a controller which blinks or lights the corresponding-line key among the plurality of line keys based on the identification information. | 04-01-2010 |
| 20100085957 | Methods and Apparatus to Form Secure Cross-Virtual Private Network Communications Sessions - Example methods and apparatus to form secure cross-VPN (virtual private network) communication sessions in multiprotocol label switching (MPLS)-based networks are disclosed. An example method comprises receiving a communication session initiation request from a first device associated with a first MPLS-based VPN, determining whether the communication session initiation request is directed to a second device associated with a second MPLS-based VPN, sending a cross-VPN link setup request to a route reflector to enable a cross-VPN communication path over which the first and second devices are permitted to communicate when the communication session initiation request is directed to the second device associated with the second VPN, and facilitating a communication session between the first and second devices via the communication path enabled by the route reflector. | 04-08-2010 |
| 20100085958 | Method and System For Service Preparation of a Residential Network Access Device - The invention relates to a method and system of service preparation of a residential network access device from one or more remote provisioning devices to prepare said residential network access device to receive a network service over a communications network. The method comprises the steps of receiving a line identifier indicating a physical line used by said residential network access device to connect to said communication network; transmitting an IP address from said one or more provisioning devices to said residential network access device for which said line identifier has been received, said IP address being a source address for said residential network access device, and transmitting software code portions to said IP address of said residential network access device, said software code portions being required for receiving said network service. | 04-08-2010 |
| 20100085959 | SYSTEM AND METHOD FOR ACHIEVING INTEROPERABILITY BETWEEN ENDPOINTS OPERATING UNDER DIFFERENT PROTOCOLS - A teleconferencing system for achieving interoperability between a multiple endpoints including a first endpoint following SIP protocol, a second endpoint following H.323 protocol and a third endpoint following a proprietary protocol. The teleconferencing system incorporates a signaling gateway and a call control server. In the teleconferencing system, the signaling gateway and a call control server are configured to perform policy-based management of calls between the first endpoint, the second endpoint and the third endpoint. | 04-08-2010 |
| 20100085960 | ATM Telecommunications Systems and Method for Routing Narrow Band Traffic - A telecommunications system comprises an asynchronous transfer mode (ATM) network having uncommitted bandwidth, and a plurality of adaptive grooming routers (AGR) coupled to the network. The AGRs comprise a group adapted to function as a virtual transit exchange whose fabric and control are distributed over the group. The visual comprising the AGRs incorporates independent connection control and call routing functions and has means for determining the current system status whereby to set up narrow band connections across the ATM network based on that status determination. | 04-08-2010 |
| 20100085961 | METHOD, DEVICE, AND SYSTEM FOR SYNCHRONIZING TERMINAL STATE IN GENERIC ACCESS NETWORK - The present invention relates to wireless communication, and discloses a method, a device, and a system for synchronizing a terminal state in a GAN to ensure that the GAN can know the relevant state context information of the terminal correctly. In the present invention, a network device of the GAN receives CS domain and/or PS domain state information reported by a terminal; and the network device processes the terminal registration information according to the received CS domain and/or PS domain state information. After the connection is reestablished between the terminal and the GAN, the terminal reports the CS domain and/or PS domain state information in the GAN mode to the GANC. The terminal may report the CS domain and/or PS domain state information in the GERAN/UTRAN mode to the GANC in the registration process. The terminal may also initiate a registration update process after the CS domain and/or PS domain state in the GERAN/UTRAN mode changes. | 04-08-2010 |
| 20100091761 | System and Method for Placing a Call Using a Local Access Number Shared by Multiple Users - The present document describes a method and system for placing a call through an Internet Protocol (IP) network, from a contact voice interface device for use by a contact user located in a first geographical area, to a subscribed voice interface device for use by a subscribed user located in a second geographical area, each geographical area defined by an area in which a local call can be made. The method comprises: assigning a local access phone number to the first geographical area; the contact user initiating a first leg of the call, from the contact voice interface device to a first IP switch, by dialing the local access phone number using the contact voice interface device; the contact user providing an identity of the subscribed voice interface device to which the call is to be completed; transmitting the identity from the first IP switch to a second IP switch via the IP network, the second IP switch associated with the identity of the subscribed voice interface device provided; the second IP switch establishing a second leg of the call at a local calling rate to the subscribed voice interface; and bridging the first leg of the call to the second leg of the call through the IP network, thereby completing the call from the contact voice interface device to the subscribed voice interface device through the IP network. | 04-15-2010 |
| 20100091762 | SYSTEM, METHOD, AND APPARATUS FOR USER-INITIATED PROVISIONING OF A COMMUNICATION DEVICE - An embodiment of a method and apparatus for provisioning of a communication device includes receiving a registration request from a first communication device. The registration request includes an address associated with the first communication device. The method further includes registering the first communication device in response to receiving the registration request, placing a call request to the first communication device, and establishing a call session with the first communication device. The method further includes prompting a user of the first communication device for a user identifier, and receiving a user identifier from the user of the first communication device. The method still further includes sending one or more configuration parameters associated with the user identifier to the first communication device. The one or more configuration parameters are operable to configure the first communication device. | 04-15-2010 |
| 20100091763 | Handling information - A communication method, for use with a telecommunications system, enables special communications to be routed to respective devices in a group of the devices by addressing the devices with a respective special identifier. The telecommunications system includes a network core and a plurality of subscriber devices registered with the network core. The core enables communications to be routed to the devices by a respective ID allocated to each device. The method further includes maintaining a store of the special identifier of each device in the group and a corresponding value derived from the ID. The method enables the special identifier of a device in the group to be obtained from the store by providing the store with the corresponding value derived from that device's ID. | 04-15-2010 |
| 20100091764 | Communication System for VOIP Using an Internet Protocol Converter - A proprietary internet converter (PIC) is disclosed, which allows a calling party end-user device with internet access such as a mobile telephone, to initiate voice communication with a called party VoIP (Voice Over Internet protocol) end-user device. The ID (Internet Device with a built-in PIC) converts the protocols used by the calling party end-user device so that the switch that routes calls to the called party VoIP end-user device understands instructions sent from the calling party end-user device. The switch has a call forwarding function. The calling party gives the calling party user name (e.g. ISP user name/contact or VoIP user name/contact) to the PIC over the internet. The PIC then sets call forwarding function on the switch, for that particular calling party, so that an incoming call from the calling party is automatically forwarded to the ISP user or VoIP user defined by the calling party. | 04-15-2010 |
| 20100091765 | APPARATUS AND METHOD FOR ENABLING OPTIMIZED GATEWAY SELECTION FOR INTER-WORKING BETWEEN CIRCUIT-SWITCHED AND INTERNET TELEPHONY - An optimized gateway selection process of the present invention is based on a universal mobility manager (UMM), a component for inter-technology location management. The UMM is capable of holding location information for diverse cellular networks, as well as for Internet telephony systems. For cellular networks, UMM acts as a traditional HLR; for an Internet telephony network, it acts as the entities that are responsible for user/terminal registration (registrar in SIP, gatekeeper in H.323) and address resolution (proxy server in SIP, gatekeeper in H.323). An optimal gateway selection is possible based on location related information provided by the UMM which had not previously been available. Utilizing the newly available information enables a gateway to be selected which may, for example, enable the circuit switched portion of a call to now be minimized. | 04-15-2010 |
| 20100091766 | ABBREVIATED DIALING USING A VOIP PLATFORM - A feature server provides an abbreviated dialing feature via an internet protocol based network to facilitate abbreviated dialing between a first phone system that serves a first location and a second phone system that serves a second location. A routing table at the first phone system stores a number range of a voice over internet protocol customer local area network located at a third location. The routing table includes instructions to route abbreviated-dialed calls to a first integrated access device. The feature server receives a query from the second phone system when an abbreviated-dialed call originating from the second phone system is not recognized by the second phone system. The abbreviated-dialed call is communicated to the internet protocol based network by a second integrated access device. | 04-15-2010 |
| 20100091767 | METHODS, SYSTEMS, AND DEVICES FOR PROVIDING VOICE-CALL SERVICES RESPONSIVE TO A DIALED SEQUENCE - A connection is established in a communications network responsive to receiving a Dual Tone Multi-Frequency (DTMF) signal at a port having an assigned sequence associated therewith. A dialed sequence corresponding to the received DTMF signal is identified. If the dialed sequence is associated with a request for a specified service, first and second fields of a packet-switched signaling protocol message are populated with the assigned sequence associated with the port. The populated packet-switched signaling protocol message is transmitted over a packet-switched network to request the specified service, and a connection is established to provide the specified service through the port. Related systems and devices are also discussed. | 04-15-2010 |
| 20100091768 | Coordination of User Information across Session Initiation Protocol-based Proxy Servers - An improvement in the design and operation of telecommunications networks is disclosed, in which when a calling party's telecommunication terminal does not know the address of the called party's terminal, the calling party's telecommunication terminal contacts its home Session Initiation Protocol (SIP) proxy server (or “home proxy”). Upon determining that it does not already have the called party's address, the home proxy employs one or more techniques in order to obtain that party's address, as well as to retain that address. The first technique of the illustrative embodiment features the usage of a registration event package, which includes SIP-based subscribe and notify mechanisms. The second technique of the illustrative embodiment features the usage of a data distribution service, which operates in a data distribution layer in contrast to utilizing, for example, a SIP mechanism. | 04-15-2010 |
| 20100091769 | Method And System For Improving Real-Time Data Communications - A system and method for improving real-time data communications by accounting for sampling rate mismatches between a transmitter and a receiver. Based on an analysis of the average number of packets received at a receiver over a period of time, a buffer monitor cooperating with the receiver can trigger an adjustment to the playback sampling rate to account for mismatches in the sampling rates of the transmitter and receiver. The buffer monitor may adjust the playback sampling rate more dramatically if the average is dangerously high or low, adjust the playback sampling rate less dramatically if the average is near satisfactory conditions, and not adjust the playback sampling rate if the average falls is satisfactory. | 04-15-2010 |
| 20100098054 | METHOD AND APPARATUS FOR PROVIDING INTERNET PROTOCOL SERVICES TO A USER OF A PRIVATE BRANCH EXCHANGE - A method and apparatus for providing one or more Internet Protocol (IP) services to users of a private branch exchange (PBX) in a network are disclosed. For example, the method receives user phone information from the private branch exchange (PBX) via a data feed, and stores the user phone information in a storage device located within the network. | 04-22-2010 |
| 20100098055 | Communication system and method - A method of controlling a connection between a user terminal and an access node connected to a communication network is provided. The user terminal establishes a data connection with the access node, periodically generates a message at predetermined intervals and transmits the periodic message to at least one network node via the access node over the communication network. Responses to the periodic messages are received from the at least one network node. The responses are analysed to determine whether to terminate the connection to the access node, and in the case that the connection to the access node should be terminated, a disconnect message is transmitted to the access node from the user terminal. | 04-22-2010 |
| 20100098056 | IMS Surrogate Registration - A method and arrangement in a telecommunication system for facilitating communication between a first terminal A configured to use a first session model and a second terminal B configured to use a second session model for media transportation. A first feature tag representing a contact between the first terminal A and the second terminal B is registered in a control domain in the system. When setup of a first media session (Voice) between the first terminal A and the second terminal B is initiated, the registered first feature tag is detected, and the first media session is routed via a circuit-switched domain. | 04-22-2010 |
| 20100098057 | TELEPHONE CALL PROCESSING - Embodiments of the invention provide methods and apparatus for providing one-telephone dialing number telephony services where only a single telephone dialing number is required for each subscriber, despite each subscriber having multiple telephony devices on which they wish to be contacted. Calls to a one-telephone dialing number telephony service subscriber may be detected at a telephone switch using one or more triggers configured in association with a device-shared telephone dialing number allocated to the subscriber. Upon receipt of a call connection request to a subscriber, control of the call is assumed, for example by redirecting the call to a service platform capable of generating multiple call connection requests. Multiple outgoing call connection requests are transmitted to multiple telephony devices, including a mobile telephone, associated with the device-shared telephone dialing number allocated to a subscriber. | 04-22-2010 |
| 20100098058 | BROADBAND COMMUNICATIONS DEVICE - The Residential Communications Gateway (RCG) is a broadband communications device that combines all voice, data and video communications to and from a typical residence or small business for transmission over a single, or a plurality of Plain Old Telephone Service (POTS) lines separately or in conjunction with, a wireless broadband backbone. The RCG does this by employing packetized data with Voice over Internet Protocol (VoIP) technologies combined with RF communications technologies. A key consideration to the design of the RCG is that no additional or special transmission equipment must be installed at the Central Office or anywhere else in the network to enable new calling features provided by the RCG as is the case with DSL and Cable systems. By eliminating the requirement for costly infrastructure enhancements, ubiquitous high speed communications and services can be deployed to every POTS subscriber. | 04-22-2010 |
| 20100098059 | SPLITTER WALL PLATES FOR DIGITAL SUBSCRIBER LINE (DSL) COMMUNICATION SYSTEMS AND METHODS TO USE THE SAME - Splitter wall plates for digital subscriber line (DSL) communication systems and methods to use the same are disclosed. An example apparatus comprises a splitter to separate a digital subscriber line (DSL) signal from a plain old telephone signal (POTS) signal, and a switch to selectively couple a VoIP signal received via a first jack or the POTS signal to a second jack. | 04-22-2010 |
| 20100098060 | METHOD AND APPARATUS FOR CONNECTING PACKET TELEPHONY CALLS BETWEEN SECURE AND NON-SECURE NETWORKS - Described herein is a method and apparatus for connecting packet telephony calls between secure networks and non-secure networks. A first telephony stream having information content for delivery to a first address may be received wherein the first telephony stream is formatted according to a first communication protocol used by a first network. The first telephony stream may be terminated at a secure boundary between the first network and a second network. A second address associated with the first address may be identified. A second telephony stream having the information content and formatted according to the second communication protocol may be delivered to the second address | 04-22-2010 |
| 20100098061 | METHODS AND APPARATUS FOR MAINTAINING CONNECTIVITY WITH AN INTERNET PROTOCOL PHONE OPERATING BEHIND A FIREWALL - Methods and apparatus for maintaining connectivity with an Internet protocol (IP) phone operating behind a firewall are disclosed. An example method disclosed herein comprises registering the IP phone in response to receiving a first registration request from the IP phone, the first registration request including first registration information, the first registration information including a first public IP address associated with the firewall, storing the first registration information, reregistering the IP phone in response to receiving a second registration request from the IP phone, the second registration request including second registration information, the second registration information including a second public IP address associated with the firewall, the second public IP address different from the first public IP address, and reverting to the stored first registration information to process calls associated with the IP phone. | 04-22-2010 |
| 20100098062 | METHOD AND APPARATUS FOR PROVIDING E911 SERVICES VIA NETWORK ANNOUNCEMENTS - A method and apparatus for providing emergency services, e.g., E911 services, for nomadic users by utilizing network announcements to remind customers to update location information used to provide services on packet networks, such as Voice over Internet Protocol (VoIP) and Service over Internet Protocol (SoIP) networks, are disclosed. For example, the method enables the VoIP or SoIP service provider to detect a change in the IP address associated with either the broadband modem or the router through which a terminal adaptor is used to access services when a customer is logging on from a new location. In turn, the method sends a reminder network announcement message to the terminal adaptor, e.g., to be played when the terminal adaptor goes off-hook. | 04-22-2010 |
| 20100098063 | METHOD AND APPARATUS FOR SUPPORTING MULTIPLE ACTIVE SESSIONS ON A PER USER BASIS - A method and apparatus for establishing multiple application sessions, such as video, audio, voice, and data sessions, and displaying them on a video display device such as a television are disclosed. These sessions can be independent of each other or the user can request the network to join these sessions so that a single session is created. For example, a user can request the network to create a video session and a music session and combine them into one session, so the audio portion of the video session is replaced by the user specified music contents and so on. | 04-22-2010 |
| 20100098064 | METHOD AND APPARATUS FOR DYNAMICALLY PROVIDING COMFORT NOISE - A method and apparatus for dynamically enabling the activation and deactivation of comfort noise over a VoIP media path or channel are disclosed. The present method detects all sound levels in the media path and only activates the comfort noise in the absence of sound and when the background noise level or the telephone line noise level is low rather than only in the absence of speech. | 04-22-2010 |
| 20100098065 | METHOD AND APPARATUS FOR PROVIDING EMERGENCY RING TONES FOR URGENT CALLS - A method and apparatus for enabling calling parties to request the VoIP network to provide a special ring tone to be signaled as the occurrence of an urgent call to called parties are disclosed. Alternatively, a high frequency intercept tone or call waiting tone is also provided when the called parties are already engaged in conversation when an urgent call is incoming. | 04-22-2010 |
| 20100098066 | METHOD AND APPARATUS FOR PROVIDING SHARED SERVICES - The present invention enables an overlay capability to be invoked on network systems and elements that are designed to support multiple customer bases. Depending on the registered identification of the user, screens and other user interfaces that provide access to functions can be overlaid on the network component and segmented along customer classifications. | 04-22-2010 |
| 20100098067 | METHOD AND APPARATUS FOR ROUTING CALLS TO AN ALTERNATIVE ENDPOINT DURING NETWORK DISRUPTIONS - A method and apparatus for enabling calls destined for a terminating point on a packet network, e.g., a VoIP network, that is experiencing a service disruption to be forwarded by the network to another endpoint is disclosed. The method enables subscribers to register an alternative number, such as a cell phone number, a relative's phone number, or a work number, that the network can use to forward calls in the event of a service disruption. In one embodiment, the provider can even use an alternative transport network, such as the PSTN, to forward these calls until the VoIP network service is restored. | 04-22-2010 |
| 20100098068 | METHOD AND APPARATUS FOR SENDING ALERTS TO INTERNET PROTOCOL PHONES - The present invention enables an alert message and the display of calling party identity on all on-hook phones associated with an extension sharing the same phone number, when one phone is off-hook and in use. In one exemplary embodiment, this capability enables all other members of a household to receive information regarding an incoming call even when one phone is in use by another member. | 04-22-2010 |
| 20100098069 | SYSTEM AND METHOD FOR PROVIDING A PLURALITY OF MULTI-MEDIA SERVICES USING A NUMBER OF MEDIA SERVERS TO FORM A PRELIMINARY INTERACTIVE COMMUNICATION RELATIONSHIP WITH A CALLING COMMUNICATION DEVICE - A system and method for processing a plurality of requests for multi-media services received at a call control element (CCE) defined on the system from a calling communication device. The system includes a Network Routing Element, a Service Broker (SB), at least a primary media severs (MS) and at least a secondary MS, a plurality of application servers (ASs) and a plurality of border elements, all of which are coupled to the CCE. The SB is adapted to receive a plurality of requests including parameters for requesting multi-media services, via the CCE, and to selectively redirect the requests to one or more ASs for providing feature processing for the requests. The ASs can instruct either the primary MS or secondary MS, via the CCE, to form a preliminary interactive communication path with the calling communication device for collecting caller-entered data, which can be validated prior to providing the feature processing. | 04-22-2010 |
| 20100103925 | System, method, and apparatus to correlate a TCAP web service request to an application server session - A method includes encoding a session identifier into a uniform resource identifier (URI) associated with a TCAP Begin message request originating at an application server, where the session identifier identifies a communication session. The method also includes transmitting the TCAP Begin message request from the application server to a transaction capabilities application part (TCAP) interface and receiving a TCAP Continue message request from the TCAP interface with the TCAP Continue message request including the encoded URI. The method includes correlating the TCAP Continue message request to the communication session that originated the Begin request identified by the session identifier in the received URI and routing the TCAP Continue message request to the communication session. | 04-29-2010 |
| 20100103926 | COMMUNICATION APPARATUS AND SERVER, AND METHODS AND COMPUTER PROGRAMS THEREFORE - A communication apparatus enabled to communicate over at least one communication bearer is disclosed. The communication apparatus comprises a receiver arranged to receive an page message from a public land mobile network node, the page message being present when another party requests communication with the communication apparatus; and a connection controller arranged to establish a connection to the Internet over at least one of the communication bearers for providing an IP connection to the another party, and to send a notification over the established connection to the Internet to a page server managing the paging by the public land mobile network for enabling closing of the paging. Further, a page server connected to the Internet is disclosed. The page server comprises a connection request receiver arranged to receive a request from a first party requesting communication with a second party; an interface for communicating with a public land mobile network, wherein the interface is arranged to provide a page request to the public land mobile network, upon the received request from the first party, on provision of an page message; and a notification receiver arranged to receive a notification, over an established connection between the second party and the Internet, that the page message is received, wherein the interface is further arranged to provide a page release request, upon the reception of the notification, to the public land mobile network for closing of the paging. Methods and computer programs are also disclosed. | 04-29-2010 |
| 20100103927 | METHOD AND APPARATUS FOR INTERWORKING SIP COMMUNICATION WAITING WITH CIRCUIT SWITCHING AND PACKET SWITCHING NODES - A method and apparatus for interworking a session by a Mobile Switching Center (MSC) server enhanced for IP Multimedia Subsystem (IMS) Centralized Services (ICS), the method comprising receiving at the MSC an invite message and the MSC sending a setup message comprising an information element indicating a call waiting tone on when the invite message comprises a call waiting indication. | 04-29-2010 |
| 20100103928 | TELEPHONE OUTLET WITH PACKET TELEPHONY ADAPTER, AND A NETWORK USING SAME - An outlet for a Local Area Network (LAN), containing an integrated adapter that converts VoIP to and from analog telephony, and a standard telephone jack (e.g. RJ-11 in North America) for connecting an ordinary analog (POTS) telephone set. Such an outlet allows using analog telephone sets in a VoIP environment, eliminating the need for an IP telephone set or external adapter. The outlet may also include a hub that allows connecting both an analog telephone set via an adapter, as well as retaining the data network connection, which may be accessed by a network jack. The invention may also be applied to a telephone line-based data networking system. In such an environment, the data networking circuitry as well as the VoIP/POTS adapters are integrated into a telephone outlet, providing for regular analog service, VoIP telephony service using an analog telephone set, and data networking as well. In such a configuration, the outlet requires two standard telephone jacks and a data-networking jack. Outlets according to the invention can be used to retrofit existing LAN and in-building telephone wiring, as well as original equipment in new installation. | 04-29-2010 |
| 20100111071 | COMMUNICATION DEVICE FOR PROVIDING VALUE-ADDED INFORMATION BASED UPON CONTENT AND/OR CONTEXT INFORMATION - A communication device, for use in a communication network, provides value-added information to a user of the communication device. The communication device includes a transceiver, operable to transmit and receive communications over the communication network, and a processor. The processor is operable to facilitate detecting context information representative of an environment in which the communication device is operated, detecting content information of a multi-directional communication stream by identifying significant words in the communication stream, encoding the detected context and content information as meta-information, transmitting the meta-information as a request for value-added information, receiving value-added information in response, and providing the value-added information to the user of the communication device. A method for providing value-added information to a user of a communication device and a communication system for providing value-added information are also disclosed. | 05-06-2010 |
| 20100111072 | Internet Phone Service System and Internet Phone Service Method by Using Softphone Created by Users - Provided are an Internet phone service system and method by using softphone created by users. The Internet phone service system includes: a service system creating a softphone created by users based on information input through a webpage and connecting calls made over the Internet by using the softphone created by users; a creator requesting the creation of the softphone created by users by inputting the information to the service system and, after the softphone created by users is created, downloading the softphone created by users in the form of an execution file or inserting a universal resource locator (URL) code into a webpage as a link to the softphone created by users; a user using the softphone created by users in the form of the execution file in a personal computer (PC) environment or clicking on the URL code, which is inserted into the webpage, to make phone calls over the Internet. | 05-06-2010 |
| 20100111073 | Universal plug and play method and apparatus to provide remote access service - Provided are a universal plug and play (UPnP) method and an apparatus thereof to provide remote access service, where the method includes receiving external inputs of an identifier of a remote access server (RAS) to generate a credential and a session initiation protocol (SIP) identifier of the RAS, generating a payload of a SIP packet written in extensible markup language (XML), which includes a credential ID generated based on the identifier of the RAS and remote access transport agent (RATA) capability information, and transmitting the SIP packet to the RAS identified by the SIP identifier, where the payload of the SIP packet includes multipurpose internet mail extensions (MIME)-type information to be identified as information used to provide remote access service. | 05-06-2010 |
| 20100111074 | Transcoders and mixers for Voice-over-IP conferencing - Transcoders and mixers having reduced algorithmic delay and processing complexity. An improved mixer for signals having encoded speech parameters wherein the parameters obtained through decoding are used by a parameter estimator to improve the encoding by providing a parameter estimate for the mixed signal. In the case of pitch parameters, the mixer uses the principle of strong-pitch-domination. The mixing of wideband signals is simplified by performing mixing of individual lower and upper sub-bands. A transcoder and a mixer that converts a wideband signal into a narrowband signal relies upon high frequency suppression. A transcoder and a mixer that converts a narrowband signal into a wideband signal relies upon filter combination. | 05-06-2010 |
| 20100111075 | Main Apparatus and Bandwidth Allocating Method - According to one embodiment, a main apparatus includes a memory configured to store a priority information table showing correspondence relationships among the terminals or lines and priority of the use bandwidth on the communication network, a monitor module configured to monitor a use bandwidth on the communication network, and a controller configured to refer to priority corresponding to terminals or lines to be subjects of session establishment from the priority information table in session establishment, and allocate use bandwidth after the session establishment based on a reference result of the table and a monitor result from the monitor module. | 05-06-2010 |
| 20100111076 | METHOD AND APPARATUS FOR ENABLING CUSTOMER PREMISE PUBLIC BRANCH EXCHANGE SERVICE FEATURE PROCESSING - A method and apparatus for enabling customer premise Public Branch eXchange (PBX) service feature processing to be performed in a service provider network using an intermediary device are disclosed. For example, the method receives a signaling message by an intermediary device managed by a service provider of a communication network, where the signaling message requires processing by a customer premise Public Branch eXchange (PBX), wherein the signaling message is in accordance with a network signaling format. The method interworks the signaling message into a signaling message in accordance with a PBX signaling format, and sends the interworked signaling message to the customer premise PBX to retrieve service logic and data associated with the signaling message. | 05-06-2010 |
| 20100111077 | METHOD AND APPARATUS FOR ENABLING CUSTOMER PREMISE PUBLIC BRANCH EXCHANGE SERVICE FEATURE PROCESSING - A method and apparatus for enabling customer premise Public Branch eXchange (PBX) service feature processing to be performed in a service provider network are disclosed. For example, the method receives a signaling message associated with a user, and accesses a customer premise Internet Protocol (IP) Public Branch eXchange (PBX) to retrieve customer premise IP PBX based service logic and data associated with the user by a Serving Call Session Control Function (S-CSCF) network element. The method then completes a service feature associated with the service logic and data in the communication network. | 05-06-2010 |
| 20100111078 | METHOD AND APPARATUS FOR ENABLING CUSTOMER PREMISE PUBLIC BRANCH EXCHANGE SERVICE FEATURE PROCESSING - A method and apparatus for enabling customer premise Public Branch eXchange (PBX) service feature processing to be performed in a service provider network are disclosed. For example, the method receives a signaling message associated with a user by an application server deployed in a communication network, and accesses a customer premise Internet Protocol (IP) Public Branch eXchange (PBX) to retrieve customer premise IP PBX based service logic and data associated with the user by the application server. The method forwards an updated signaling message by the application server to a Serving Call Session Control Function (S-CSCF) network element for completing a service feature associated with the service logic and data in the communication network. | 05-06-2010 |
| 20100111079 | METHOD AND APPARATUS FOR NETWORK BASED FIXED MOBILE CONVERGENCE - A method and apparatus for providing a network based Fixed Mobile Convergence (FMC) service are disclosed. For example, the method receives a NB-FMC call request originating from a Session Initiation Protocol (SIP) NB-FMC endpoint device or a non-SIP NB-FMC endpoint device, and processes the NB-FMC call request using a single hosted NB-FMC Application Server (AS). | 05-06-2010 |
| 20100118859 | ROUTINE COMMUNICATION SESSIONS FOR RECORDING - Systems and methods for recording a communication session between a customer and an agent are provided. In this regard, a representative method comprises: routing a media stream associated with the communication session based on information corresponding to routing criteria, wherein the routing criteria include call control protocols or policies; receiving the media stream associated with the communication session from the customer center communication system; and recording the received media stream. | 05-13-2010 |
| 20100118860 | METHOD AND ARRANGEMENT FOR ALLOWING ENTERPRISE AND PERSONAL DOMAINS IN THE IMS - The present invention relates to a method and an arrangement for allowing private domains in the IMS, which makes it possible to use a SIP URI like ID@private-domain.TLD. This is achieved by providing an administration support and an interface to the IMS interconnect DNS and the DNS system of the operator network. The identity associated with private domain is established as a Private domain name based IMPU. The private domain name based IMPU and the Operator domain name based IMPU is associated by using the implicit registered identity set provided by the IMS. | 05-13-2010 |
| 20100118861 | Inter-Working Between a Packet-Switched Domain and a Circuit-Switched Domain - The present invention proposes a solution for providing packet-switched services to a user, even in case the user is accessibly only via a circuit-switched access. For this purpose it is proposed to force the user to report changes in the reachability status and to keep the user's registration alive in a packet-switched domain as long as the user is reachable. A packet-switched adapter located between the circuit-switched domain and a packet-switched domain aligns the registration status of the user in the packet-switched domain to the reachability status and performs a registration procedure depending on the outcome of the alignments. The registration procedure might be in detail either a registration or a re-registration or a de-registration procedure. | 05-13-2010 |
| 20100118862 | METHOD OF AND A SYSTEM FOR ESTABLISHING A CALL OVER AN IP MULTI MEDIA COMMUNICATIONS SYSTEM AND A CIRCUIT SWITCHED COMMUNICATIONS SYSTEM - A system and method for simultaneously supporting IMS signaling and Circuit Switched signaling during a call between a calling user terminal in an IP Multi media System (IMS) and a called user terminal in a Circuit Switched CS network. A first IP address is determined by the calling user terminal, the calling user terminal initiates the call using IMS signaling towards the called user terminal, the IMS signaling comprising the first IP address. The called user terminal then initiates a CS connection towards an IP Access Converter using CS signaling and comprising the first IP address. The IP Access Converter allocates a second IP address (IP | 05-13-2010 |
| 20100118863 | Method for Realizing a Re-Answer Call - A method for realizing a re-answer call, comprises: S | 05-13-2010 |
| 20100118864 | Hierarchical Data Collection Network Supporting Packetized Voice Communications Among Wireless Terminals And Telephones - A packet-based, hierarchical communication system, arranged in a spanning tree configuration, is described in which wired and wireless communication networks exhibiting substantially different characteristics are employed in an overall scheme to link portable or mobile computing devices. The network accommodates real time voice transmission both through dedicated, scheduled bandwidth and through a packet-based routing within the confines and constraints of a data network. Conversion and call processing circuitry is also disclosed which enables access devices and personal computers to adapt voice information between analog voice stream and digital voice packet formats as proves necessary. Routing pathways include wireless spanning tree networks, wide area networks, telephone switching networks, internet, etc., in a manner virtually transparent to the user. A voice session and associate call setup simulates that of conventional telephone switching network, providing well-understood functionality common to any mobile, remote or stationary terminal, phone, computer, etc. | 05-13-2010 |
| 20100118865 | APPARATUS AND METHOD FOR PROVIDING RECORDING SERVICE IN IP MULTIMEDIA SUBSYSTEM - An apparatus and method are provided for proving the recording service in an Internet Protocol (IP) Multimedia Subsystem (IMS). The apparatus includes a communication unit for receiving a recording request from a calling portable terminal or a called portable terminal, and a recording service manager unit coupled to the communication unit for setting a path of bearer traffic for recording a conversation between the calling portable terminal and the called portable terminal. | 05-13-2010 |
| 20100118866 | INTERNET PROTOCOL TRANSPORT OF PSTN-TO-PSTN TELEPHONY SERVICES - A system for transporting public switched network (PSTN) terminated signaling across an Internet protocol (IP) network includes a gateway between the PSTN and the IP network. The gateway receives a telephony signaling message from the PSTN and determines if the telephony signaling message maps to an IP signaling message. If the telephony signaling message does not map to an IP signaling message, the gateway packages the telephony signaling message in a special IP signaling message for transport over the IP network. If the gateway receives a special IP signaling special message, the gateway unpackages the telephony signaling message from the special message for transport over the PSTN. If the gateway receives DTMF signals from the PSTN, the gateway translates the DTMF signals to digits and packages the digits in a special IP signaling message for transport over the IP network. The gateway also packages the DTMF signals in an IP media transport protocol message for transport over the IP network. | 05-13-2010 |
| 20100124216 | METHOD AND APPARATUS FOR PROVIDING CALL ROUTING IN A NETWORK - A method and an apparatus for providing call routing in a network are disclosed. For example, the method receives a signaling message for a call, and determines if the signaling message contains information for determining if routing of the call requires an ENUM (tElephone Numbering Mapping) query. The method then processes the call by bypassing the ENUM query if the signaling message contains the information. | 05-20-2010 |
| 20100124217 | Apparatus and method for connection control with media negotiation successful on different media formats - In an apparatus for connection control between two terminals, a communication unit transmits or receives a connection control signal to or from the terminals. A storage stores media format information usable on the two terminals, which are to be supplied with the media format information on media formats converted by a media format converter. A media format information supplementing unit references the media format information storage, based on the connection control signal received from the terminal, and verifies a possible presence of common media format information usable by the terminals to be interconnected. If there is no common media format information, predetermined media format information is supplemented to the connection control signal, and a resulting connection control signal is delivered to the communication device. Thus, media negotiation may be made even when media formats usable on the two terminals differ from each other. | 05-20-2010 |
| 20100124218 | METHOD AND SYSTEM FOR CONNECTING A VOICE CALL USING A DOMAIN NAME DATABASE - A method for connecting a telephone call includes receiving, at a server, from a communication terminal, a first message including at least one word corresponding to a name of an individual or an organization, wherein the first message is transmitted using an Internet-compatible protocol; searching for the at least one word in a server database; at the server, comparing the at least one word with domain names stored in the database and, if domain names are found such that at least a part of the domain name matches the word, transmitting to the communication terminal a list of domain names, each domain name including an identifier of an Internet resource; receiving, at the server, a second message containing the domain name selected by the user from the list; identifying a phone number associated with the selected domain name; at the server, transmitting the phone number to the communication terminal; and connecting the communication terminal to the phone number via the communications network. | 05-20-2010 |
| 20100128715 | Protocol Conversion System in Media Communication between a Packet-Switching Network and Circuit-Switiching Network - In media communication by way of a packet-switching network and a circuit-switching network, a protocol conversion device for converting protocols between the packet-switching network and the circuit-switching network includes a call connection unit and a protocol converter. The call connection unit carries out call connection processes of media communication between terminals of the packet-switching network side and terminals of the circuit-switching network side. The protocol converter analyzes packets of speech received from the packet-switching network to specify the encoding bit rate of speech data in the speech packets. The protocol converter then specifies the multiplex table used in multiplexing frames on the circuit-switching network from the encoding bit rate. The protocol converter further generates frames by using the multiplex table that was specified to multiplex data in the payload of packets received from the packet-switching network and transmits to the circuit-switching network. | 05-27-2010 |
| 20100128716 | Method and apparatus for providing network based services to private branch exchange endpoints - Many of the current IMS standards and enriched services were originally designed for the individual subscribers that are serviced by the wireless network. However, the IMS standards do not fully address the problem of providing the IMS enriched services and features to users connected to a PBX. The present invention discloses a method for providing IMS enriched services and features to users connected to a PBX or an IP-PBX. Access to network services can be secured through a web-friendly interface via the IMS, enabling third-party developers, service providers and even subscribers to self-manage their service experience while the network operator retains control over network resources. | 05-27-2010 |
| 20100128717 | METHOD AND APPARATUS FOR OPERATING A COMPUTER-TELEPHONY SYSTEM - One embodiment of the invention provides a method of operating a computer-telephony system. The method comprises providing computer-telephony support for a plurality of customers. Each customer maintains customer relationship management, CRM data. The CRM data is uploaded from the plurality of customers into a computer-telephony database. The uploading includes transforming the CRM data from an original format maintained by the respective customer into a standardised format for the computer-telephony database. Telephone calls can then be handled using the transformed CRM data in the computer-telephony database. | 05-27-2010 |
| 20100128718 | Supporting Method for REFER Message Expansion Parameter - The present invention discloses a method for supporting extended parameter(s) in a REFER message, applied in a system comprising a parameter proxy server that can receive and forward the REFER message; upon receiving the REFER message including the extended parameter(s), the parameter proxy server performing the following processing: storing said extended parameter(s); sending the REFER message to the indicated party of the message; upon receiving a third party message sent by said indicated party and indicated by the method parameter in the REFER message, adding the extended parameter(s) stored into the third party message; sending the third party message to said third party. With the present invention, those IMS intelligent terminals that do not support the extended parameter(s) in the REFER message can more fully utilize the REFER message to use the abundant services provided by NGN. | 05-27-2010 |
| 20100128719 | Server Apparatus and Terminal Apparatus - According to one embodiment, a server apparatus includes a memory configured to store a management table, wherein the management table showing a correspondence relationship among the terminal ID, a remaining amount of the battery, and an additional service concerning call origination and call termination, an acquisition module configured to acquire remaining information of the battery from the first terminal, and a controller configured to refer to the management table based on the remaining information of the battery, when an execution request for the additional service is issued, and execute an additional service corresponding to the remaining amount of the battery based on a reference result of the management table. | 05-27-2010 |
| 20100128720 | ENTERPRISE CONTACT SERVER WITH ENHANCED ROUTING FEATURES - A server may include logic configured to receive a call-back request from a customer, where the call-back request includes an identifier associated with the customer. The server may further include logic configured to identify, based on the identifier, a call center from a group of call centers having a group of agents qualified to handle the call-back request; and logic configured to forward the call-back request to the identified call center, where the call-back request causes the identified call center to select one of the group of agents to handle the call-back request. | 05-27-2010 |
| 20100128721 | APPARATUS AND METHOD FOR PROVIDING OTHER SERVICE IN IP MULTIMEDIA SUBSYSTEM (IMS) - An apparatus and a method for providing other Voice over Internet Protocol (VoIP) services (e.g., Skype, Google talk, and the like) using a terminal which supports an IP Multimedia Subsystem (IMS) network are provided. The apparatus includes an interworking apparatus for converting information received from a VoIP service network to information supportable by an IMS terminal to interwork the IMS terminal and other VoIP services not supported by the IMS terminal, and converting information received from the IMS terminal to information supportable by a VoIP service network. | 05-27-2010 |
| 20100128722 | QUEUING MECHANISMS FOR LTE ACCESS AND SAE NETWORKS ENABLING END-TO-END IMS BASED PRIORITY SERVICE - A system and method for queuing emergency telecommunication service requests prevent dropped connections by sending messages to nodes requesting the emergency telecommunication service that the request has been queued. This allows for an orderly queuing process and allows congestion related issues to be overcome without preempting existing network traffic. | 05-27-2010 |
| 20100128723 | COMPUTER, INTERNET AND TELECOMMUNICATIONS BASED NETWORK - A method and apparatus for a computer and telecommunication network which can receive, send and manage information from or to a subscriber of the network, based on the subscriber's configuration. The network is made up of at least one cluster containing voice servers which allow for telephony, speech recognition, text-to-speech and conferencing functions, and is accessible by the subscriber through standard telephone connections or through internet connections. The network also utilizes a database and file server allowing the subscriber to maintain and manage certain contact lists and administrative information. A web server is also connected to the cluster thereby allowing access to all functions through internet connections. | 05-27-2010 |
| 20100135277 | VOICE PORT UTILIZATION MONITOR - A method for monitoring utilization of a voice over internet protocol platform in a mass calling application includes receiving calls via voice ports established by servers. Utilization information for each of the servers is aggregated in accordance with applications associated with the calls. The aggregated utilization information is separately displayed for each of the applications, each of the applications corresponding to a distinct subset of the calls. | 06-03-2010 |
| 20100135278 | SYSTEM AND METHOD TO INITIATE A PRESENCE DRIVEN PEER TO PEER COMMUNICATIONS SESSION ON NON-IMS AND IMS NETWORKS - An architecture and method is provided for call routing using both IMS and non-IMS frameworks. The method includes receiving presence information of a third party from a non-IP Multimedia Subsystem (IMS) network device. The method further includes routing the third party to at least one callee designated device based on configurable preferences provided by the callee and correlated to presence information using an IMS compliant component. The method additionally includes providing a charging record for the routing on an IMS complaint charging platform. | 06-03-2010 |
| 20100135279 | Method and Arrangement for Remotely Controlling Multimedia Communication Across Local Networks - A method and arrangement for remotely controlling the communication of media between devices in different local networks ( | 06-03-2010 |
| 20100135280 | TELECOMMUNICATIONS SYSTEM AND TELECOMMUNICATIONS MANAGEMENT APPARATUS - The migration of telephone services by a telecommunications carrier from a PSTN to an IP network entails that problem that when it is not possible for some telephone subscribers to migrate to the IP network due to the types of their telephone lines or services they subscribe to, other subscribers also cannot migrate to the IP network until the former subscribers migrate to the IP network. | 06-03-2010 |
| 20100135281 | METHOD AND APPARATUS FOR SENDING UPDATES TO A CALL CONTROL ELEMENT FROM AN APPLICATION SERVER - A method and apparatus for enabling Application Servers to automatically update the databases used by Call Control Elements as changes occur between customer data, such as customer specific logic, and the Application Servers, such as the IP addresses of the Application Servers are disclosed. Whenever there is a change in the location of customer specific data needed by the CCEs, e.g., switching from one AS to a new AS, the new AS will automatically update the relevant database in the CCEs to indicate such an update has occurred. After the automatic update is performed, the CCEs will be able to communicate with the correct AS to retrieve and process the customer specific service logic. | 06-03-2010 |
| 20100135282 | Implementation Method, System and Device of IMS Interception - The embodiment of the present invention discloses an implementation method of IMS interception, a system and a device thereof. The method includes allocating media anchor points to communication parties; and copying communication content to complete interception of the communication content when the communication parties communicate through the media anchor points. In the present invention, the interception of the communication content can be implemented on the media anchor points instead of the existing media access equipment, thus implementing centralized interception of a media on an IMS network, and ensuring that users accessing the IMS network in various modes can implement the centralized interception of the media on an IMS core without requirement for access side equipment or an extended private interface between the IMS core and the access side equipment. | 06-03-2010 |
| 20100135283 | Voice-Over-IP Enabled Chat - A network-based system and method for providing anonymous voice communications using the telephone network and data communications links under the direction of a Call Broker and associated network elements. A user (the call initiator) present in a text chat room session establishes a data connection to Call Broker and, after qualifying for access (e.g., using credit card information) and providing a callback number, receives voice session information and participant access codes for each desired participant in a voice call. The initiator causes session information and participant codes to be passed to one or more selected chat participants in the current text chat room. When a selected participant uses the received session information, and enters the received participant code and a callback number, the Call Broker in cooperation with a Network Adjunct Processor (NAP) completes voice links to the initiator and the selected participant(s). The need for each party to have a second subscriber line is advantageously avoided by having the Call Broker arrange to have one or more voice links completed through a VoIP link, and further reduces the need for second lines for participants by forwarding a Call Broker—placed call to a busy participant line to the participant's Internet Service Provider (ISP), which then sends a message to the participant announcing one or more options for receiving the incoming call, including receiving the incoming call through a VoIP link. | 06-03-2010 |
| 20100135284 | Method and system for routing calls from a standard telephone device to a voice over Internet Protocol network - The invention enables accessing and using a Voice over Internet Protocol network, and can use a standard telephone to automatically access a VoIP network. A first aspect of the invention uses an auto dialer to transmit digits, such as a network access number, an account number and a PIN, which remain unchanged from call to call made through a given network service provider. A second aspect of the invention provides a speed dial feature for placing VoIP telephone calls. Speed dial numbers are recorded in a VoIP service provider's database on a server, which is accessible through the Internet from a personal computer (PC) or a conventional telephone. A third aspect of the invention enables callers to complete calls from conventional telephones to personal computers connected to the Internet. The VoIP network detects a flag such as leading “0,” determines that the call recipient station is a personal computer, looks up the IP address of the PC and routes the call to the PC. | 06-03-2010 |
| 20100135285 | Multi-Networking Communication System and Method - The architecture of the present invention includes a multi-media multi-network communication server connected to a variety of access and delivery platforms via a variety of communication networks. The access platforms are used by senders, recipients or agents to access their digital mailboxes on a multi-network communication server and to send and receive calls and messages. The messages can be in electronic format such as text, audio, graphic images, video, and audio-video. The multi-network communication may send a notification message to the recipient, indicating that a message has been received. Messages can be accessed remotely or wirelessly and can be viewed, heard, or both, depending on the capability of the delivery platform being used by the recipient user. | 06-03-2010 |
| 20100142512 | METHOD AND ARRANGEMENT FOR AUTOMATICALLY UPDATING A WHITE LIST - The invention relates to a method and a system and devices for session control in a communications network, whereby subscriber-specific data (D) of a subscriber (B) called by a calling subscriber (A) for the purpose of call completion (C) are stored in a list (WL), associated with the subscriber (A) to be called, for administering subscriber-specific data (D) of trustworthy subscribers. The subscriber-specific data (D) concerning the called subscriber (B) are automatically stored in the list (WL). | 06-10-2010 |
| 20100142513 | Method for Measuring Processing Delays of Voice-Over IP Devices - A system and method for recording analog signals exchanged between a telephone device and a VoIP device, capturing packets exchanged between the VoIP device and an IP network, determining analog time values corresponding to analog characteristics of the analog signals, determining digital time values corresponding to digital characteristics of the packets, determining a common reference time for the analog time values and digital time values and determining a processing delay based on the analog time values and the digital time values. | 06-10-2010 |
| 20100142514 | METHOD AND APPARATUS FOR CORRELATION OF DATA SOURCES IN A VOICE OVER INTERNET PROTOCOL NETWORK - In one embodiment, a method for managing a Voice over IP (VoIP) network includes collecting a first set of data from a first source of network performance management data, each data item in the first set of data corresponding to a call made using the VoIP network; collecting a second set of data from a second source of network performance management data, each data item in the second set of data corresponding to a call made using the VoIP network and being of a different type than the first set of data; correlating the first set of data and the second set of data such that a data item from the first set of data is matched with a data item from the second set of data that corresponds to a common call made over the VoIP network; and outputting a performance report based on the correlating. | 06-10-2010 |
| 20100142515 | BLENDING TELEPHONY SERVICES IN AN INTERNET PROTOCOL MULTIMEDIA SUBSYSTEM - An Internet protocol Multimedia Subsystem (IMS) gateway application server includes an originating application server module adapted to invoke call control services in response to requests initiated by a voice over Internet Protocol (IP) (VoIP) client associated with a communication device such as an IP telephone. Disclosed gateway application servers include a proxy server module adapted to notify the communication client of session control messages intended for the communication device. | 06-10-2010 |
| 20100142516 | SYSTEM AND METHOD FOR PROCESSING MEDIA REQUESTS DURING A TELEPHONY SESSIONS - In a preferred embodiment, the method of caching media used in a telephony application includes: receiving a media request; sending the media request to a media layer using HTTP; the a media layer performing the steps of checking in a cache for the media resource; processing the media request within a media processing server; and storing the processed media in the cache as a telephony compatible resource specified by a persistent address. The system of the preferred embodiment includes a call router and a media layer composed of a cache and media processing server. | 06-10-2010 |
| 20100142517 | Method and System for Supporting SIP Session Policy Using Existing Authorization Architecture and Protocols - A method for sending a session policy request to a network component is provided. The method comprises a user agent sending the session policy request to the network component using a lower layer protocol. The lower layer protocol is at least one of Extensible Authentication Protocol (EAP), Point to Point Protocol (PPP), and General Packet Radio Service (GPRS) Activate Packet Data Protocol (PDP) context. | 06-10-2010 |
| 20100142518 | Hierarchical Data Collection Network Supporting Packetized Voice Communications Among Wireless Terminals and Telephones - A packet-based, hierarchical communication system, arranged in a spanning tree configuration, is described in which wired and wireless communication networks exhibiting substantially different characteristics are employed in an overall scheme to link portable or mobile computing devices. The network accommodates real time voice transmission both through dedicated, scheduled bandwidth and through a packet-based routing within the confines and constraints of a data network. Conversion and call processing circuitry is also disclosed which enables access devices and personal computers to adapt voice information between analog voice stream and digital voice packet formats as proves necessary. Routing pathways include wireless spanning tree networks, wide area networks, telephone switching networks, internet, etc., in a manner virtually transparent to the user. A voice session and associate call setup simulates that of conventional telephone switching network, providing well-understood functionality common to any mobile, remote or stationary terminal, phone, computer, etc. | 06-10-2010 |
| 20100142519 | METHOD AND SYSTEM FOR AN ETHERNET IP TELEPHONE CHIP - Methods and systems for an Ethernet IP phone chip are provided. In this regard, data may be received via a first port of an Ethernet switch in the Ethernet IP phone chip, and the port(s) via which the data is forwarded may be determined based on characteristics of the data. The Ethernet switch may receive data from a network via a first port, and communicate the received data to one or more on-chip interfaces via a second port. The on-chip interfaces may process the received data and may communicate video contained in the data to an off-chip video processing device. The Ethernet IP phone chip may receive video data from an off-chip video processing device via one or more on-chip interfaces, packetize the video data into one or more Ethernet packets; and communicate the packet(s) onto a network link via the Ethernet switch. | 06-10-2010 |
| 20100150133 | METHOD AND APPARATUS FOR PROVIDING IMS SERVICES TO CIRCUIT-SWITCHED CONTROLLED TERMINALS - The present invention proposes a solution for providing IMS services to users having circuit-switched controlled terminals being not adapted to provide IMS services to the users. In particular, it is proposed, in order to allow IMS to take the full call and service control, to place a user agent being responsible for the user ported to the IMS in a new node type called Mobile Access Gateway Control Function (MAGCF). This new node combines the logical functionality of a cellular switching center and the logical functionality of IMS. The invention discusses a concept of a static MAGCF being deployed in a network and being assigned for handling a user. | 06-17-2010 |
| 20100150134 | METHOD AND APPARATUS FOR PROVIDING REPEAT CALLING - A method and apparatus for providing a repeat calling service feature in a communication network are disclosed. For example, the method captures call session information between a calling endpoint and a called endpoint of a failed call due to an unavailability of required network resource. The method then receives a repeat calling service request from the calling endpoint, and processes the repeat calling service request to reestablish a call between the calling endpoint and the called endpoint. | 06-17-2010 |
| 20100150135 | DEVICE BASED EMERGENCY SERVICES FOR CROSS CLUSTER EXTENSION MOBILITY - A system is disclosed. The system has a call data receiver arranged to receive call data comprising number data indicative of a telephone number associated with a call connection request. The system also has a translator arranged to translate the received number data to obtain translated number data indicative of another telephone number. There is also a number data associator for associating the call connection request with the translated number data. | 06-17-2010 |
| 20100150136 | VOICE-OVER-INTERNET PROTOCOL DEVICE LOAD PROFILING - A device may obtain, from a remote device on a network, information regarding loads and Session Initiation Protocol (SIP) devices on which the loads are installed. In addition, the device may access a database storing load compatibility information, identify problematic loads based on the obtained information and the load compatibility information, determine fixes for one or more of the problematic loads, and apply the fixes to the one or more of the problematic loads over the network. | 06-17-2010 |
| 20100150137 | IMS and Method of Multiple S-CSCF Operation in Support of Single PUID - A method for providing multimedia services to subscriber user equipment (UE) within an IP multimedia subsystem network (IMS) may include configuring the IMS to enable a single UE to fork register and cooperate with multiple serving-call session control functions (S-CSCFs). After obtaining IP connectivity, the single UE signals to register with the IMS and the IMS determines whether the UE is configured to fork register with multiple S-CSCFs. If the UE is configured, the IMS determines which S-CSCFs are eligible for the UE registration and fork registers the UE to multiple S-CSCFs of the eligible S-CSCFs. Consequently, incoming and outgoing calls to/from the UE are routed by the IMS to any of the multiple registered S-CSCFs. | 06-17-2010 |
| 20100150138 | INTERCEPTING VOICE OVER IP COMMUNICATIONS AND OTHER DATA COMMUNICATIONS - Methods and apparatus for intercepting communications in an Internet Protocol (IP) network involve maintaining dialing profiles for respective subscribers to the IP network, each dialing profile including a username associated with the corresponding subscriber, and associating intercept information with the dialing profile of a subscriber whose communications are to be monitored. Intercept information will include determination information for determining whether to intercept a communication involving the subscriber, and destination information identifying a device to which intercepted communications involving the subscriber are to be sent. When the determination information meets intercept criteria communications are established with a media relay through which communications involving the subscriber will be conducted or are being conducted to cause the media relay to send a copy of the communications involving the subscriber to a mediation device specified by the destination information. | 06-17-2010 |
| 20100150139 | Telephony Web Event System and Method - An embodiment of the system for publishing events of a telephony application to a client includes a call router that generates events from the telephony application and an event router that manages the publication of events generated by the call router and that manages the subscription to events by clients. The system can be used with a telephony application that interfaces with a telephony device and an application server | 06-17-2010 |
| 20100150140 | IP MULTIMEDIA SUBSYSTEM (IMS) AND METHOD FOR ROUTING AN HTTP MESSAGE VIA AN IMS - The invention relates to an IP Multimedia Subsystem, IMS, for providing a service via a network to at least one subscriber, the system comprising: at least a first proxy function and a first server function for handling messages with a first protocol, a subscriber database connected via a first interface to the server function, at least a second proxy function and a second server function for handling messages with a second protocol, the second server functionally is connected via a second interface to the database. | 06-17-2010 |
| 20100150141 | METHOD AND APPARATUS FOR DETERMINING MEDIA CODEC IN SIP-BASED VOIP NETWORK - A method and apparatus for determining a media codec in a Session Initiation Protocol (SIP)-based Voice over Internet Protocol (VoIP) network are provided. The method includes determining an SIP entity to be assigned with a priority for determining the media codec, generating an SIP message including information on the SIP entity having the priority for determining the media codec, and transmitting the SIP message. | 06-17-2010 |
| 20100150142 | Telephone Service Via Packet-Switched Networking - A system and method for providing telephone type services over the internetwork commonly known as the Internet. Public switched telephone networks utilizing program controlled switching systems are arranged in an architecture with the Internet to provide a methodology for facilitating telephone use of the Internet by customers on an impromptu basis. Provision is made to permit a caller to set-up and carry out a telephone call over the Internet from telephone station to telephone station without access to computer equipment, without the necessity of maintaining a subscription to any Internet service, and without the requiring Internet literacy or knowledge. Calls may be made on an inter or intra LATA, region or state, nationwide or worldwide basis. Billing may be implemented on a per call, timed, time and distance or other basis. Usage may be made of common channel interoffice signaling to set up the call and establish the necessary Internet connections and addressing. Calls may be made from telephone station to telephone station, from telephone station to computer or computer to telephone station. | 06-17-2010 |
| 20100157976 | NETWORK DEVICE - A first telephone, which is connected to a network device, can be prevented from becoming communicable with a second telephone, which is connected to the network device over a public network, when the second telephone and a third telephone, which is an Internet telephone, are communicable with each other and a device operating voltage for operating the network device is no longer supplied. | 06-24-2010 |
| 20100157977 | METHODS, SYSTEMS, AND COMPUTER PROGRAM PRODUCTS FOR PROVIDING INTRA-CARRIER IP-BASED CONNECTIONS USING A COMMON TELEPHONE NUMBER MAPPING ARCHITECTURE - Internet protocol (IP) based calls from a first terminal in an IP based communications system are routed to a second terminal in another communications system. In response to a call setup request at a common communications core that is common to both the IP based communications system and the other communications system, a query is transmitted to a private telephone number mapping database that contains routing information for terminals in both the IP based communications system and the other communications system requesting routing information for the second terminal. Routing information for the call setup request is received from the private telephone number mapping database for routing the call. | 06-24-2010 |
| 20100157978 | APPARATUS AND METHOD FOR MANAGING A PRESENTATION OF MEDIA CONTENT - A system that incorporates teachings of the present disclosure may include, for example, a communication device having a controller to detect a media device operating externally to the communication device actively engaged in presenting media content, detect an incoming communication session initiated by another communication device, present a notice identifying the media device and the media content being presented by the media device, detect a directive to modify an operation of the media device to mitigate interrupting a communication session with the other communication device, and instruct the media device to modify its operation according to the directive. Other embodiments are disclosed. | 06-24-2010 |
| 20100157979 | System and Methods for Improving Interaction Routing Performance - An interaction router includes a computerized server executing a routing engine stored on a machine-readable medium, an interface at the server receiving information from an interaction switching element, the information regarding an interaction received at the switching element to be routed, an interface at the server to a wide area network (WAN), a function of the routing engine judging if one or more business-logic determinations are to be made to select a routing destination for the interaction, and a function for controlling the switch to route the interaction. If if one or more business-logic determinations are to be made, the routing engine requests the business-logic determination from a remote server over the WAN, and upon receiving the determination from the remote server, uses the determination in controlling the switching element to route the interaction. | 06-24-2010 |
| 20100157980 | SIP PRESENCE BASED NOTIFICATIONS - An exemplary embodiment builds on the idea of presence and SIP messaging in conjunction with a profile comprising, for example, a lookup table and a rules module, to assist with one or more of reminders, actions, endpoint maintenance, availability, endpoint capabilities and session management. One exemplary embodiment provides a message-based notification system. SIP allows a user to associate themselves with a number of different User Agents (UAs). This means that a user may have a presence on more than one UA at any given time, e.g., soft phone, PDA, and workstation. One exemplary embodiment utilizes this fact in connection with monitoring and determining if a message is a trigger-type message to provide more timely and relevant notifications to users. | 06-24-2010 |
| 20100157981 | DIFFERENTIATED PRIORITY LEVEL COMMUNICATION - Provided are methods, apparatuses and systems for providing prioritized data distribution at a customer premise. A network access component may determine a particular hardware identifier associated with data received from a communication entity. The hardware identifier may uniquely identifying a piece of hardware originating data. The network access component may also determine a particular priority level associated with the data based on the particular hardware identifier. The network access component may also prioritize at least a portion of the data on a basis of the particular priority level. | 06-24-2010 |
| 20100157982 | TRANSMITTER AND METHOD FOR TRANSMITTING AND RECEIVING OF VOICE DATA FOR VOIP SERVICE - Provided are a transmitting apparatus and voice data transmitting and receiving methods for providing VoIP services. When a call is started and an analog signal including a voice signal is input, the transmitting apparatus divides the analog signal into a plurality of voice data packets for transmission. Here, the plurality of voice data packets are generated by sampling with different phases in the same frequency. In addition, the transmitting apparatus inserts a time indication bit that is changed every transmission period into each of the voice data packets and transmits the voice data packets, and distinguishes voice data corresponding to a current transmission period based on the time indication bit. | 06-24-2010 |
| 20100157983 | System and Method for Providing Alternate Routing in a Network - Methods and systems are described for performing alternate routing of communications in a network. The system initiates a communication from an origination endpoint in a packet-switched network, such as a VOIP network, to a destination endpoint, and determining, according to selection criteria, whether to route the communication to the destination endpoint using at least a second circuit-switched network, such as the PSTN. | 06-24-2010 |
| 20100157984 | WIDEBAND VOIP TERMINAL - A wideband Voice over Internet Protocol (VoIP) terminal is provided. The wideband VoIP terminal includes a synchronous serial interface which processes audio data input thereto or output therefrom in series in synchronization with a clock; and an audio accelerator which encodes or decodes the audio data, wherein the synchronous serial interface includes a buffer buffering the audio data and a buffer controller controlling the buffer and the audio accelerator includes a memory storing the audio data processed by the synchronous serial interface under the control of the buffer controller, a memory controller controlling the memory and an encoder/decoder encoding/decoding the audio data. The wideband VoIP terminal can facilitate the input and output of data. | 06-24-2010 |
| 20100157985 | SYSTEM AND METHOD FOR INDICATING CIRCUIT SWITCHED ACCESS AT IMS REGISTRATION - In IP Multimedia Subsystem (IMS) IMS Control Channel Protocol (ICCP) is used between a user equipment (UE) and IMS Control Channel Function (ICCF) and Session Initiated Protocol (SIP) interface (between to ICCF, Call Session Control Function and Application Server) to support the indication of Circuit Switched (CS) access utilizing P-Access-Network-Information header. The indication can be used by an S-CSCF or AS for different purposes such as routing decision, charging, and presence info. | 06-24-2010 |
| 20100157986 | SYSTEMS, METHODS, AND COMPUTER READABLE MEDIA FOR LOCATION-SENSITIVE CALLED-PARTY NUMBER TRANSLATION IN A TELECOMMUNICATIONS NETWORK - Systems, methods, and computer readable media for location-sensitive identifier translation in a telecommunications network are disclosed. According to one aspect, the subject matter described herein includes a method for providing location-sensitive called-party identifier translation in a telecommunications network. The method includes, at a signaling node that includes at least one processor: receiving a first signaling message that includes a called party identifier; determining proximity information associated with the calling party; performing a location-sensitive called party identifier translation based on the proximity information associated with the calling party; and sending the first signaling message or a second signaling message, the sent message including the translated called party identifier. | 06-24-2010 |
| 20100157987 | TELEPHONE SWITCHING SYSTEMS - The invention relates to the generation of configuration data for use in the migration of telephone switching systems. Configuration data for use in the migration of subscribers from a first telephone switching system over to a second telephone switching system in a telecommunications network is generated by monitoring signaling information on telephone channels associated with subscribers for telephone calls conducted via the first telephone switching system. The monitored signaling information is then analyzed in relation to call data produced by the first telephone switching system for the calls to identify relationships between the monitored signaling information and call data for calls conducted by subscribers. Configuration data based on the identified relationships is then stored and used to configure the second telephone switching system with mappings between the associated telephone channels and the telephone dialing numbers for subscribers. | 06-24-2010 |
| 20100157988 | IP TELEPHONE DEVICE, IP TELEPHONE SYSTEM, AND SETTING CONFIRMATION METHOD - An IP telephone system comprises a main device that manages outgoing and incoming calls of an IP telephone device, an external storage device storing network configuration information and telephone device configuration information, and an IP telephone device comprising a first interface section that uses in connection to the external storage device and a second interface section that uses in connection to a network. When automatically carrying out internal setting by connecting the external storage device to the first interface section, the IP telephone device obtains the network configuration information and the telephone device configuration information from the external storage device and, based on the obtained network configuration information and telephone device configuration information, carries out network setting and telephone device setting. The IP telephone device accesses the main device through the second interface section based on the setting and performs confirmation of the set contents. | 06-24-2010 |
| 20100157989 | APPLICATION STORE AND INTELLIGENCE SYSTEM FOR NETWORKED TELEPHONY AND DIGITAL MEDIA SERVICES DEVICES - Telephony and digital media services may be provided to a plurality of locations, such as to a plurality of homes and offices, though the deployment of telephony and digital media services devices to the locations, wherein each device is configured to function as a voice, data and media information center. A system in accordance with one embodiment of the present invention includes an application store and an application intelligence subsystem implemented on one or more computers. Each of the application store and the application intelligence subsystem is communicatively connected via a network to a plurality of such telephony and digital media services devices. The application store is operable to provide applications via the network for installation and execution on each of the plurality of devices. The application intelligence subsystem is operable to obtain and report information about applications installed and executed on each of the plurality of devices. | 06-24-2010 |
| 20100157990 | SYSTEMS FOR PROVIDING TELEPHONY AND DIGITAL MEDIA SERVICES - A system, method and apparatus for providing telephony and digital media services to a location, such as a home or office, is described herein. In one embodiment, the system includes a telephony and digital media services device that is configured to function as an all-in-one voice, data and media information center. The device provides telephony functionality both directly and through associated handsets. The device pairs a user-friendly touch-screen interface with a high-performance hardware/software architecture capable of delivering advanced media applications and graphics combined with landline quality telephony service all in one integrated system. | 06-24-2010 |
| 20100157991 | APPARATUS AND METHOD FOR RECORDING CELLULAR CALL IN AN INTERNET TELEPHONE SYSTEM - Call recording in an Internet telephone system is provided. A dual-mode terminal includes a call server interworker for, when a cellular call commences, determining whether it is possible to access a call server which controls Voice over Internet Protocol (VoIP) calls; a recording interface processor for, when it is possible to access the call server, setting a connection to a recording server; a recorder for generating recording data packets comprising a cellular phone conversation; and a data communicator for transmitting the recording data packets to the recording server. | 06-24-2010 |
| 20100157992 | DATA SIN/DATA SOURCE, DATA TRANSMISSION DEVICE AND DATA TERMINAL DEVICE FOR A CIRCUIT-SWITCHED AND PACKET-SWITCHED NETWORK - The present invention is directed toward, a data sink/data source data transmission device and data terminal device for a circuit-switched and packet-switched network, the ability to eliminate the logical separation between applications, which are based on the circuit-switched network (e.g., PSTN, ISDN), and applications, which are based on the packet-switched network, (e.g., Internet). To this end, a data transmission device for transmitting and receiving data into/from the circuit-switched network includes controllable switchover parts. This data transmission device is or can be assigned to a universally useable unit for automatically processing data and for transmitting and receiving data to/from the packet-switched network and is assigned or can be assigned to the at least one data terminal device for transmitting and receiving data into/from the circuit-switched network. The switch-over parts can be controlled in such a manner that the data terminal device which, in a first operating mode is connected to the circuit-switched device, can be switched from the first operating mode into a second operating mode, during which the data terminal device is connected to the packet-switched network via the data transmission device and the data processing device, and from the second operating mode into the first operating mode. | 06-24-2010 |
| 20100157993 | ACCESS GATEWAY AND METHOD OF OPERATION BY THE SAME - An access gateway containing IP telephone service functions for subscribers under an integrated access device (IAD), forming a PSTN network side speech path or IP network side speech path selectively for each subscriber, and, further automatically switching, when trouble occurs at the IP network side, the IP network side speech path to the PSTN network side speech path. | 06-24-2010 |
| 20100165976 | HANDLING EARLY MEDIA IN VOIP COMMUNICATION WITH MULTIPLE ENDPOINTS - Technologies for handling early media in VoIP communications with multiple endpoints are provided. A calling device sends an initial VoIP call request to multiple destination devices, or endpoints. The calling device then receives a provisional response from one or more of the destination devices that includes media streaming parameters regarding the destination device. The calling device creates a media context associated with the destination device that contains the media streaming parameters and stores the media context. The calling device uses the media context to establish a media connection with the destination. One of the destination devices returning a provisional response is selected to exchange early media over the media connection established with the destination device. | 07-01-2010 |
| 20100165977 | System for Scheduling Routing Rules in a Contact Center Based on Forcasted and Actual Interaction Load and Staffing Requirements - A system for scheduling resources and rules for routing includes a server connected to a network, a scheduling application executable from the server, and at least one programmable software agent for scheduling routing rules. The scheduling application receives statistics about forecast arrival rates for incoming interactions and current resource availability data and schedules resources and routing rules according to the forecast requirements the software agent propagating the portion of scheduling relative to the routing rules. | 07-01-2010 |
| 20100165978 | METHOD AND APPARATUS FOR PROVIDING AN AUTOMATED SHOPPING SERVICE IN A TELECOMMUNICATION SYSTEM - Method and apparatus for providing an automated shopping service in a telecommunication system are described. In some examples, a call is received via the telecommunication system initiated by a caller. An electronic prompt is played to the caller. An electronic response is received from the caller in response to the electronic prompt. At least one item requested by the caller is automatically detected in the electronic response. A search is performed of at least one shopping source to obtain information associated with the at least one item. The information is sent to towards the caller. | 07-01-2010 |
| 20100165979 | METHOD AND APPARATUS FOR GENERALIZED THIRD-PARTY CALL CONTROL IN SESSION INITIATION PROTOCOL NETWORKS - In one embodiment, the present invention is a method and apparatus for generalized third party call control in session initiation protocol networks. In one embodiment, a method for controlling a media negotiation with one or more endpoints in a network includes determining, for each endpoint, a current state of a corresponding port on a third-party controller and transitioning the corresponding port to a new state in accordance with a finite state machine that tracks the state of the media negotiation. | 07-01-2010 |
| 20100165980 | Usage Of Physical Layer Information In Combination With Signaling And Media Parameters - A plurality of subscriber connections for a plurality of subscribers is established, where the establishment of each subscriber connection includes receiving a connection request message from a subscriber that includes physical layer information identifying a physical access connection on which the connection request message was received. A physical layer identifier is assigned for the subscriber connection that uniquely identifies the subscriber connection and is based on the physical layer information. A first signaling message is received on a first one of the established subscriber connections and includes a subscriber identifier of a subscriber. The subscriber identifier is associated with the physical layer identifier of the first subscriber connection. Subsequently, messages are received that include the subscriber identifier of the subscriber. The ones of those messages that were received on the first subscriber connection are processed. | 07-01-2010 |
| 20100165981 | METHOD AND APPARATUS FOR CONTROLLING THE ACCESS OF A USER TO A SERVICE PROVIDED IN A DATA NETWORK - Process for controlling the access of a user to a service provided in a data network, to protect user data stored in a data base of the service from unauthorized access, the method comprising: (a) inputting, in a VoIP session, a voice sample of the user at a user data terminal which is at least temporarily connected to the data network, (b) processing, in a first processing step, the user's voice sample using a dedicated client implemented at the user data terminal, to obtain a pre-processed voice sample or a current voice profile of the user, (c) further processing, in a second processing step, the pre-processed voice sample or the current voice profile, including a comparison step of the current voice profile with an initial voice profile stored in a data base, and (d) outputting an access control signal for granting or rejecting access to the service, taking the result of the comparison step into account. | 07-01-2010 |
| 20100165982 | Method and Apparatus for Creating and Distributing COST Telephony-Switching Functionality within an IP Network - A system for providing and managing IP telephone calls establishes separate and distinct call legs between IP-capable appliances and routers and between routers, and creates calls, changes calls, and manages telephony functions by joining and disjoining calls legs. In some instances one or more call legs disjoined from an active call are maintained as established to be joined later to other call legs to create other active calls. By managing IP calls as separate and distinct legs functions of intelligent, connection-oriented telephony networks may be simulated in IP telephony systems. The management is provided by software running on processors coupled to routers in the IP network. | 07-01-2010 |
| 20100172342 | Session Initiation Protocol Message Payload Compression - A method, user terminal, and Session Initiation Protocol (SIP) Application Server for transporting SIP messages across an IP Multimedia network between the user terminal and the SIP Application Server. The sending side compresses message payloads within the application layer and the receiving side decompresses them at the application layer. The compressed message payloads are passed between the application layer and a SIP User Agent via an appropriate Application Programming Interface (API). | 07-08-2010 |
| 20100172343 | Dynamic Network Classification - A round trip time (“RTT”) is measured between a Voice over Internet Protocol (“VoIP”) endpoint and a mediation server across a network. A determination is made whether the measured RTT is consistent with one of a plurality of network classification values. Each of the plurality of network classification values may correspond to a network policy. In response to determining that the measured RTT is consistent with one of the plurality of network classification values, the corresponding network policy is applied to configure bandwidth management on the VoIP endpoint. | 07-08-2010 |
| 20100172344 | WEB SERVICE ASSISTED REAL-TIME SESSION PEERING BETWEEN ENTERPRISE VOIP NETWORKS VIA INTERNET - A system and method enables Voice over IP (VoIP) session peering via Internet. A converged enterprise web server can receive a request from a caller to initiate a session, associate a Service Level Agreement (SLA) and address information of the caller with the request, and then provide the request to a receiver using a first communication protocol. After accepting from the receiver a response to the request if the caller is an allowed partner of the receiver based on the SLA, wherein the response is associated with address information of the receiver, the converged enterprise web server can establish the session between the caller and the receiver using a second communication protocol. | 07-08-2010 |
| 20100172345 | EMERGENCY ASSISTANCE CALLING FOR VOICE OVER IP COMMUNICATIONS SYSTEMS - In accordance with one aspect of the invention there is provided a process for handling emergency calls from a caller in a voice over IP system. The process involves receiving a routing request message including a caller identifier and a callee identifier. The process also involves setting an emergency call flag active in response to the callee identifier matching an emergency call identifier pre-associated with the caller. The process further involves producing an emergency response center identifier in response to the emergency call identifier. The process also involves determining whether the caller identifier is associated with a pre-associated direct inward dialing (DID) identifier. The process further involves producing a direct inward dialing (DID) identifier for the caller by associating a temporary DID identifier with the caller identifier when the emergency call flag is active and it is determined that the caller has no pre-associated DID. The process also involves producing a routing message including the emergency response center identifier and the temporary DID identifier for receipt by a routing controller operable to cause a route to be established between the caller and the emergency response center. | 07-08-2010 |
| 20100172346 | METHOD AND APPARATUS FOR TRANSMITTING GROUPCAST TO SUPPORT VOICE PAGING SERVICE IN VOICE OVER INTERNET PROTOCOL SYSTEM - A method and apparatus for transmitting a groupcast to support a voice paging service in a VoIP system are provided. In the method, a voice paging message is received from a voice paging transmitting terminal and one or more voice paging messages are reproduced from the received voice paging message. A group table is used to change the destination address and port of each of the reproduced voice paging messages into the IP address and port of each of one or more voice paging receiving terminals. Each of the reproduced voice paging messages is unicast on the basis of the changed IP address and port. | 07-08-2010 |
| 20100172347 | VOICE COMMUNICATION BETWEEN USER EQUIPMENT AND NETWORK - User equipment (UE), for communicating wirelessly with communication networks, is disclosed, the UE being adapted so as to be capable of voice communication with communication networks via a plurality of mechanisms, the plurality of mechanisms comprising at least one packet-switched (PS) mechanism and at least one circuit-switched (CS) mechanism. The UE is further adapted, when at a location at which the UE is able to communicate wirelessly with a particular communication network, to communicate with the network to determine which voice communication mechanisms the network is adapted to use. The UE is further adapted to select a voice communication mechanism according to a result of the determination and according to said plurality of mechanisms the UE is adapted to use, and to provide voice communication with the network via the selected mechanism. A corresponding method is disclosed. | 07-08-2010 |
| 20100177764 | Technique for Interconnecting Circuit-Switched and Packet-Switched Domains - A technique for providing circuit-switched cal services for a call stretching between a packet-switched domain and circuit-switched domain is provided. A possible server implementation of this technique includes a first interface adapted to receive packet-switched protocol messages requesting circuit-switched call services, a service component providing the requested call services, and a second interface adapted to pass call control towards the circuit-switched domain after the call services have been provided. | 07-15-2010 |
| 20100177765 | WAKING UP A VOIP TERMINAL DEVICE FROM A POWER-SAVING STATE - A VoIP terminal device is configured to enter a power-saving state upon the occurrence of a specified condition. The VoIP terminal device is further configured to wake up from the power-saving state when a communication associated with a specified communication operation is received by the VoIP terminal device. In particular, the operating power of the VoIP terminal device is increased to an extent sufficient to perform the specified communication operation. | 07-15-2010 |
| 20100177766 | Self-forming VoIP Network - A self-forming VoIP connection capability is described that may be superimposed over wired networks, wireless networks, or combinations thereof. As described herein, a local network cluster forms while isolated from a conventional SIP server, or alternately may exist as a cluster of network nodes and clients that later becomes isolated from a conventional SIP server by a break in the network. Either way, each network node thus enabled with distributed SIP registry functionality according to this invention independently constructs a local SIP registry and SIP server capability within that node. Subsequently, while isolated from a conventional SIP server, VoIP conversations among client devices connected to nodes within an isolated cluster will continue, and nodes and clients may join or leave an isolated cluster with conversations able to be initiated or continued while a node has network connectivity to the cluster. | 07-15-2010 |
| 20100177767 | IMS NETWORK SYSTEM AND DATA RESTORING METHOD - On detecting that subscriber data has been lost, an S-CSCF notifies a P-CSCF of a loss of the subscriber data and rebuilds the lost subscriber data by incorporating with the P-CSCF. The S-CSCF and the P-CSCF may be included in a single server or different servers. | 07-15-2010 |
| 20100177768 | METHOD TO ADAPT THE ROUTING OF A CUSTOMER'S COMMUNICATIONS WITHIN AN IMS TYPE NETWORK - The invention concerns a method for the adaptation of the routing of the communications of a Customer (C) within an IMS type network, the said method providing for the ability of the Customer (C) to transmit additional routing rules (2), and wherein the said additional routing rules are concatenated with the routing rules (1) defined by the IMS network operator, such concatenated routing rules (3) being subsequently available for use by the IMS network to adapt the routing of the customer's communications as a function of his/her needs. The invention also pertains to an IMS type network. | 07-15-2010 |
| 20100177769 | Method and Arrangement For Handling Profiles in a Multimedia Service Network - A method and apparatus for sharing an application profile for plural public IMS identities across different IMS subscriptions. A home application profile for a first public IMS identity (IMPUx) of a first IMS subscription, is stored in its entirety at a first HSS storage. A profile reference is stored as an abbreviated foreign application profile for a second public IMS identity (IMPUy) of a second IMS subscription at a second HSS storage. The profile reference points to the home application profile in the first HSS storage. An authorizing identifier for the second public IMS identity that authorizes access to the home application profile, is also stored at the first HSS storage. | 07-15-2010 |
| 20100177770 | Peer-To-Peer Telephone System - There is provided a peer-to-peer telephone system comprising a plurality of end-users and a communication structure through which one or more end-users are couplable for communication purposes. The system is distinguished in that the communication structure is substantially de-centralized with regard to communication route switching therein for connecting the one or more end-users. One or more end-users are operable to establish their own communication routes through the structure based on exchange of one or more authorisation certificates, namely User Identity Certificates (UIC), to acquire access to the structure. The structure comprises an administration arrangement for issuing said one or more certificates to said one or more end-users. | 07-15-2010 |
| 20100177771 | System and Method for Originating a Call Via a Circuit-Switched Network from a User Equipment Device - Methods and apparatus for originating a Session Initiation Protocol (SIP) call from a user equipment (UE) device in a network environment including a circuit-switched (CS) network and an IP multimedia subsystem (IMS) network to a called party are disclosed. In one illustrative example, when the SIP call is originated by the UE device in the CS network domain, a SIP Invite message which includes a SIP Uniform Resource Indicator (URI) or Tel URI of the called party is sent from the UE device to the IMS network (e.g. to an application server (AS) node). At the AS node, a pool of E.164 numbers are maintained as IP multimedia routing numbers (IMRNs) which are utilized for mapping to or otherwise associating with called party URIs. Thus, the AS node dynamically allocates a select E.164 number with respect to the called party's URI received from the UE device, and returns it to the UE device in a SIP Response message, e.g., a SIP 380 (Alternative Service) message. Subsequently, the dynamically-allocated E.164 number is sent from the UE device in a call setup message for identification of the URI at the AS node, via the mapping, so that the SIP call may be properly routed towards the called party. | 07-15-2010 |
| 20100182994 | IP TELEPHONY ON A HOME NETWORK DEVICE - In one embodiment, a method for providing voice communications in a packet switched network protocol through a home network is provided, the method comprising: receiving, at a first home network device, an incoming call in the packet switched network protocol; notifying a second home network device of the incoming call; receiving an indication from the second home network device that the second home network device accepts the call; and forwarding the incoming call to the second home network device. | 07-22-2010 |
| 20100182995 | NAT traversal method in Session Initial Protocol - The present invention provides an NAT (Network Address Translator) traversal method in Session Initiation Protocol (SIP) for solving the problems of SIP in Internet phone (VoIP) under current Internet environment. In other words, the present invention solves the SIP problems caused by NAT (Network Address Translator) that P2P (Peer to Peer) transmission cannot traverse the NAT firewall directly. The major content of the present invention is that the computer conducts multiple detections before issueing an Invite message in order to detect the rule of the NAT server to assign port number | 07-22-2010 |
| 20100182996 | Feature Interaction Detection During Calls With Multiple-Leg Signaling Paths - Methods are disclosed for detecting feature interactions during a call that has a signaling path comprising two or more legs. In accordance with the illustrative embodiment, feature state information is maintained for each of the legs of the call and is propagated along the signaling path. The illustrative embodiment is capable of detecting interactions between features in different legs of a call, as well as interactions between features in the same leg of a call. Moreover, the illustrative embodiment is capable of accommodating a variety of feature resolution techniques. In one illustrative embodiment specific to Voice over Internet Protocol (VoIP) telephony, a Back-to-Back User Agent (B2BUA) stores and propagates the feature state information, and performs address mapping for two specially-defined headers in addition to the usual Session Initiation Protocol (SIP) headers. | 07-22-2010 |
| 20100182997 | METHOD, APPARATUS AND SYSTEM FOR TRANSMITTING USER EQUIPMENT INFORMATION IN A MULTIMEDIA SUBSYSTEM - The present disclosure discloses a method, apparatus and system for transmitting UE information in a multimedia subsystem. The method includes: a call session control function entity obtains capability information of UE, and transmits the capability information of the UE to an AS; the AS obtaining the capability information of the UE sent from the call session control function entity. The solution of the present disclosure ensures that the AS in the IMS can obtain the capability information of the UE. Therefore, services based on the capability information of the UE can be implemented on the AS successfully. | 07-22-2010 |
| 20100182998 | Access Domain Selection In A Communications Network - A method and apparatus for managing access domain selection for a user device accessing an IP Multimedia Subsystem (IMS) network. A Call Session Control Function (CSCF) in the IMS network stores an access domain indicator associated with a user s contact address. The access domain indicator is associated with the user s contact address when the user registers with the IMS network. The CSCF sends the access domain indicator to an Access Domain Selection (ADS) function in the IMS network, the access domain indicator to be used by the ADS function in selecting an access domain. This allows the ADS function to select the correct access domain to use when sending messages to the user device. | 07-22-2010 |
| 20100182999 | SYSTEM AND METHOD FOR PROVIDING A LOCAL NUMBER FOR AN OVERSEAS CALLER TO CALL OR SEND A MESSAGE TO A CALLEE - A system and method for providing a local number for an overseas caller. The system comprises a service provider unit; an interface unit for communication between a callee and the service provider unit, wherein the service provider unit receives a communication from the callee via the interface unit instructing to allocate a DDI number to a caller and creates an association between the DDI number and both the caller and the callee; and a communication device for communication between the service provider unit and the caller for providing the DDI number to the caller. | 07-22-2010 |
| 20100183000 | VIDEO DELIVERING SYSTEM, VIDEO DELIVERING DEVICE, AND SYNCHRONIZATION CORRECTING DEVICE - The video receiving device delivers reproducible video streams by synchronizing video images. The video delivering device determines the delivery time for each RTP packet based on the time information for plural video streams corresponding to plural contents, adds the determined delivery time (timestamp) to each RTP packet, and delivers RTP packets by using the counter common among plural contents. The video relaying device corrects the transfer timing for RTP packets based on the counter common among plural contents and the delivery time (timestamp) and sends them to the video receiving device. The video receiving device plays back the video images from the received RTP packets. | 07-22-2010 |
| 20100183001 | INTERCEPT SYSTEM, ROUTE CHANGING DEVICE AND RECORDING MEDIUM - An intercept system includes: a call controller that controls a call between a plurality of communication devices connected through a packet network; a route setting device that sets a route along which communication on a call between the communication devices is relayed; a duplicating device that duplicates a packet to be intercepted; an acquiring unit that acquires communication device identification information for identifying positions on the packet network of the communication devices from the call controller; a setting unit that sets the route setting device in such a way that a packet received from one communication device is transmitted to the duplicating device as to communication on a call between the communication devices identified by acquired communication device identification information; and a returning unit that returns a received packet to the route setting device after duplicating the received packet by the duplicating device for use in interception. | 07-22-2010 |
| 20100183002 | POLICY CONTROL AND BILLING SUPPORT FOR CALL TRANSFER IN A SESSION INITIATION PROTOCOL (SIP) NETWORK - A session initiation protocol (SIP) server adds billing and authentication information to conventional SIP messages used in establishing call transfers. This additional information is later verified by a SIP server and used to enable advanced billing and fraud protection features for call transfers in a SIP telecommunications network. | 07-22-2010 |
| 20100189094 | System and method for transition of association between communication devices - A system and method of providing an assignable registration between a device and a user for IP telephony, wireless telephony and other forms of collaborative systems is provided wherein loss of association is detected and a policy language is used to capture and execute user preferences in the event of such loss. A method and system are also provided for utilizing coupling between a thin client and a telephone to provide for device association. | 07-29-2010 |
| 20100189095 | Method and apparatus for voice traffic management in a data network - Method and apparatus for voice traffic management in a data network includes establishing a default maximum bandwidth setting at a LAN egress port when voice-type traffic is not present in a LAN portion of the data network, detecting voice-type traffic, reducing the bandwidth setting at the LAN egress port to effect a change in a rate of non voice type traffic and monitoring non voice type traffic and voice quality statistics to determine if the rate of non voice type traffic entering the data network has changed. Once the desired change has occurred, performing a linear increase of the bandwidth setting at the LAN egress port to a first value while monitoring voice quality statistics, determining if voice quality has degraded during increase of the bandwidth setting and repeating the last two steps if voice quality has not degraded. | 07-29-2010 |
| 20100189096 | SINGLE SUBSCRIPTION MANAGEMENT FOR MULTIPLE DEVICES - System(s) and method(s) are provided that facilitate managing routing voice and data traffic, associated with a subscription, when there are multiple devices. A client component can manage which communication device of multiple communication devices of a subscriber is active on the network at a given time for the subscriber based in part on location of a mobile device associated with the subscriber, a subscriber profile, and predefined routing criteria, which can facilitate optimal device selection. The mobile device can communicate via a macro network when outside of an area served by consumer premise equipment of the subscriber; and when the mobile device is in the area served by the consumer premise device, voice and data traffic directed to the mobile device can be automatically routed to one of multiple communication devices connected to the consumer premise equipment. The subscriber profile can specify routing preferences of the subscriber. | 07-29-2010 |
| 20100189097 | SEAMLESS SWITCH OVER FROM CENTRALIZED TO DECENTRALIZED MEDIA STREAMING - A media gateway is provided that enables seamless switchover between a centralized media stream passing between first and second endpoints and through the media gateway and a decentralized media stream passing between the first and second endpoints, but bypassing the media gateway. The gateway provides synchronization information to the first and second endpoints to enable synchronization of the centralized and decentralized media streams. After synchronization is completed, the centralized media stream is disconnected in favor of the decentralized media stream. | 07-29-2010 |
| 20100189098 | TELEPHONE OUTLET WITH PACKET TELEPHONY ADAPTOR, AND A NETWORK USING SAME - An outlet for a Local Area Network (LAN), containing an integrated adapter that converts VoIP to and from analog telephony, and a standard telephone jack (e.g. RJ-11 in North America) for connecting an ordinary analog (POTS) telephone set. Such an outlet allows using analog telephone sets in a VoIP environment, eliminating the need for an IP telephone set or external adapter. The outlet may also include a hub that allows connecting both an analog telephone set via an adapter, as well as retaining the data network connection, which may be accessed by a network jack. The invention may also be applied to a telephone line-based data networking system. In such an environment, the data networking circuitry as well as the VoIP/POTS adapters are integrated into a telephone outlet, providing for regular analog service, VoIP telephony service using an analog telephone set, and data networking as well. In such a configuration, the outlet requires two standard telephone jacks and a data-networking jack. Outlets according to the invention can be used to retrofit existing LAN and in-building telephone wiring, as well as original equipment in new installation. | 07-29-2010 |
| 20100189099 | METHOD AND SYSTEM FOR PROVIDING INTERDOMAIN TRAVERSAL IN SUPPORT OF PACKETIZED VOICE TRANSMISSIONS - An approach provides interdomain traversal to support packetized voice transmissions. A request for establishing a voice call is received from a source endpoint behind a first network address translator of a first domain, wherein the request specifies a directory number of a destination endpoint within a second domain. A network address is determined for communicating with the destination endpoint based on the directory number. Additionally, existence of a second network address translator within the second domain is determined. If the network address can be determined, a media path is established between the source endpoint and the destination endpoint based on the network address to support the voice call. | 07-29-2010 |
| 20100189100 | Communication Between Call Controllers By Amending Call Processing Messages - Call Control entities in a network communicate between themselves by amending call processing messages to include encrypted network information. As such, a call may be established whose path through the network is dependent on the paths of other calls. Information of a scope larger than a Call Controller normally possesses can, as a result of this communication, be made available to Call Controllers for constraining call establishment. This information could relate to other calls and connections associated with those other calls. The information may also relate to gateways in and to adjacent networks and the Call Controllers in the adjacent networks that are related to the current Call Controller. | 07-29-2010 |
| 20100195641 | Seamless multi-mode voice - A multi-mode mobile phone device is equipped to store both PSTN and VoIP phone numbers in a unified, multi-formatted manner. Automatic registration of VoIP new user accounts is conducted using an existing cellular phone number, an existing MAC address, or an existing VoIP identifier, without active participation from the user. Registrations of an existing VoIP account's IP addresses are also conducted without the knowledge of the user of a VoIP device. Unified electronic phonebooks and graphical user interfaces present all phone (PSTN and VoIP) numbers with the same format, with an option to display the mode (PSTN or VoIP) associated with each number. Four-way switching between entire inbound and outbound circuit and VoIP calls is accomplished by intercepting CALL and ANSWER commands issued by the user of a mobile dual-mode phone device. Seamless end-to-send call setup is enabled by using a social network of phone devices using a DHT-based search algorithm on a distributed database. | 08-05-2010 |
| 20100195642 | System and Method for Routing Calls Associated with Private Dialing Plans - A method for routing a call associated with a private dialing plan includes receiving a call directed to a destination endpoint associated with a private dialing plan (PDP), receiving an internal egress path identifier associated with the destination endpoint, and routing the call to an egress path identified by the egress path identifier. A system for routing a call including a destination number associated with a PDP including a routing engine operable to route the call to a PDP call resolution server, and a first switch operable to receive an egress path identifier and a PDP telephone number from the PDP call resolution server, the egress path identifier identifying an egress path for routing the call to a destination endpoint associated with the destination number, and the PDP telephone number identifying a selected PDP destination endpoint and a second switch operable to receive the call based on the egress path identifier and route the call to the selected PDP destination endpoint using the PDP telephone number. | 08-05-2010 |
| 20100195643 | Domain Specific PLMN Selection - A mobile communication device includes a domain selection feature that allows a user to select a domain preference such as a circuit switched (CS) voice domain preference, a packet switched (PS) data domain preference, or a (CS+PS) domain preference. The mobile device receives Public Land Mobile Network (PLMN) ID and domain availability information from one or more PLMNs. A PLMN priority list is generated on the basis of the received PLMN information and the user domain preference selection. PLMNs having the user selected service available are assigned a higher priority than those that don't currently have the service, whereby an original PLMN list may be updated. Thereby, the mobile device is more likely to obtain the desired service without resorting to a time consuming manual selection process. | 08-05-2010 |
| 20100195644 | METHOD FOR SWITCHING THE SESSION CONTROL PATH OF IP MULTIMEDIA CORE NETWORK SUBSYSTEM CENTRALIZED SERVICE - A method for switching the session control path of IMS centralized services is provided. When the condition for switching the session control path is satisfied during the ICS session based on the first session control path, the following steps are performed: one party of the ICS UE and ICCF of the ICS session transmits the request of switching the session control path to the other party; the receiving party identifies the ICS session corresponding to the request, and transmits an acknowledgement response to the transmitting party; and the ICCF and ICS UE set the session control path corresponding to the identified ICS session as the second session control path and transfer the subsequent session control information associated with the ICS session via the second session control path; wherein the first or second session control path is one of the PS session control path and the CS session control path. | 08-05-2010 |
| 20100202437 | TELECOMMUNICATIONS SYSTEM AND METHOD FOR CONNECTING A CSTA CLIENT TO SEVERAL PBXS - The presently disclosed Demultiplexer Application associated with a server or other processor (S) (collectively “Demultiplexer”) enables a computer telephony Client Application (CA), for example a Computer-Supported Telecommunications Application (CSTA) Client Application, to connect to several Communication Devices (PBX | 08-12-2010 |
| 20100202438 | AUTO-CONFIGURED VOICE OVER INTERNET PROTOCOL - In one embodiment, an apparatus may receive a call over a Public Switched Telephone Network (PSTN) from a Voice over Internet Protocol (VoIP) adapter. The VoIP adapter may be one or more devices that may create and accept VoIP connections over a network, such as the Internet, and that may transmit a call over the PSTN. The apparatus may store a call detail of the received call in a registry service, where the call detail is associated with a node identifier of the apparatus in the registry service. The apparatus may further determine a dial sequence at which the apparatus may be reached over the PSTN based on corresponding call details also stored in the registry service. | 08-12-2010 |
| 20100202439 | PREVENTION OF VOICE OVER IP SPAM - In one embodiment, a system is provided to prevent VoIP spam. The system may store call data that is associated with a call to a phone number made over a Public Switched Telephone Network. Subsequently, the system may accept an Internet Protocol telephony connection in response to verification of a demonstrated knowledge of the call. The demonstrated knowledge of the call may be verified based on the call data. | 08-12-2010 |
| 20100202440 | METHOD AND APPARATUS FOR ESTABLISHING DATA LINK BASED ON AUDIO CONNECTION - In a communications system, after parties form a voice telephone connection, the parties respective communications devices automatically create or leverage machine readable features or content of the telephone connection to identify the parties to each other or to a rendezvous server, and thereafter the communications devices and/or the rendezvous server automatically establishes a data link between the parties. | 08-12-2010 |
| 20100202441 | METHOD AND APPARATUS FOR THE USER-SPECIFIC CONFIGURATION OF A COMMUNICATIONS PORT - A method and an apparatus for the user-specific configuration of a communications port includes provisioning a default profile that references a predetermined user, assigning the default profile to a user-specific configuration profile that is assigned to the predetermined user, and configuring the communications port using the user-specific configuration profile. | 08-12-2010 |
| 20100202442 | TELEPHONY AND DATA NETWORK SERVICES AT A TELEPHONE - A packetised data network includes IP telephones (ITs) and a network intelligence (NI). All of the keys of each IT are “soft” keys (i.e., they have no fixed function). The NI associates a configuration data structure with the IT which correlates the keys with functions, and, based on this, may control the display of the IT to indicate the current function of certain of the soft keys. Some of the functions are requests for data services at the telephone (e.g., video or programmed audio over the Internet). When a user requests such a service with a key press, the NI sets up the service between the data source and the telephone. This may require associating a new configuration data structure with the keys of the IT. The IT user may activate multiple data services through the NI. | 08-12-2010 |
| 20100202443 | Voice communications system - Voice communications apparatus is connected to a general subscriber telephone set or a broadband telephone set, to communicate over the public switched telephone network. The apparatus includes a filter converting signals such as to satisfy signal conditions prescribed for the telephone network. The apparatus also includes a terminal class determiner for determining the class of the telephone set connected, and a circuit for changing at least the sampling frequency at which the analog signals from the telephone set are sampled. This establishes high quality in broadband voice communications. The terminal class determiner may automatically determine the class of the telephone set at any timing of a call sequence by detecting a band component of the signals from the telephone set, a predetermined frequency of a signal intermittently transmitted from the telephone set or the characteristics of the telephone set. | 08-12-2010 |
| 20100202444 | METHOD FOR PROCESSING THE BUSYNESS OF A FLEXIBLE ALERT GROUP WITH SINGLE-USER TYPE - A method for processing the busyness of flexible alert group with single-user type, the method comprises: a caller dials a guiding number of Flexible Alert (FA), and the calling is connected to an application server, the application server acquires member numbers of the FA group based on the guiding number, and establishes the callings to each member in the FA group; when one member in the FA group returns a busyness message, if the FA is of the single-user type, the application server determines the FA group is busy; and the application server releases all the callings to the other members in the FA group, and returns FA group being busy to the caller. | 08-12-2010 |
| 20100202445 | SERVER DEVICE AND INFORMATION REGISTRATION METHOD - The present invention provides an information registration system, a server device, a server processing program, and an information registration method which are capable of efficiently registering a telephone number without hesitation of the user and recognizing an identical person using the telephone number thus registered. | 08-12-2010 |
| 20100202446 | METHODS, SYSTEMS, AND COMPUTER READABLE MEDIA FOR CENTRALIZED ROUTING AND CALL INSTANCE CODE MANAGEMENT FOR BEARER INDEPENDENT CALL CONTROL (BICC) SIGNALING MESSAGES - The subject matter described herein includes methods, systems and computer readable media for centralized routing and call instance code management for bearer independent call control (BICC) signaling messages. One aspect of the subject matter described herein includes a system for routing BICC signaling messages and managing call instance code assignments. The system includes a BICC signaling router. The BICC signaling router includes a routing module for centralized routing of BICC signaling messages between a plurality of BICC signaling nodes. The BICC signaling router further includes a call instance code management module for centralized assignment of call instance codes for BICC signaling sessions routed through the BICC signaling router. | 08-12-2010 |
| 20100202447 | Call Transfer Method, System and Device - A call transfer method includes releasing a call signaling connection between a call transfer server and the called UE after knowing that a called user equipment performs call transfer. A service request is sent for redirecting to a third party UE to a telephony application server. | 08-12-2010 |
| 20100208723 | SYSTEMS AND METHODS FOR NETWORK FACSIMILE TRANSMISSIONS - Disclosed herein are systems and methods for sending and receiving facsimile transmissions in a voice-over-IP system. In certain embodiments, a facsimile machine may include a network interface and a call set up protocol client configured to interface with a call set up protocol server. The call set up protocol client may communicate with the call set up protocol server using the network interface to establish a communication channel with the public switched telephone network. The call set up protocol client may operate according to the session initiation protocol. The facsimile machine may be configured to send and receive facsimile transmissions according to the T.30 protocol. In alternative embodiments, the facsimile machine may be configured to send and receive facsimile transmissions according to the T.38 protocol. | 08-19-2010 |
| 20100208724 | Power Savings For Network Telephones - In an example embodiment, an IP phone is connected to a network switch that has an established communications channel to a call control server. The network switch acts as a proxy for the phone exchanging registration information that in effect keeps the phone registered while the phone itself can go to sleep as defined by periods of the day, periods where the phone is unused, or presence, which can then wake-up quickly with assistance from the switch. If the switch is not able to act as proxy, the phone can switch from sleep mode to “wake” up mode at predetermined intervals, such as every 30 seconds, in order to respond to keep alive packets from the call control server. | 08-19-2010 |
| 20100208725 | Methods, apparatuses, system, related computer programs and data structures for subscription information delivery - A method and corresponding apparatus are configured to transmit, during terminal attachment to an evolved packet system and in at least a portion of a diameter command, centralized service-related subscription information. The method and apparatus are also configured to transmit, in an initial message of a call continuity procedure, received centralized service-related subscription information. The method and apparatus are configured to operate according to the received centralized service-related subscription information in the initial message of the call continuity procedure. | 08-19-2010 |
| 20100208726 | Systems and Methods for the Reliable Transmission of Facsimiles over Packet Networks - Described herein is a facsimile to voice over IP adapter for the real-time reliable transmission of audio messages using HTTP, the voice over IP adapter having audio adapter interfaces, the audio adapter interfaces capable of receiving a audio encoded facsimile data stream; ethernet adapter interfaces, the ethernet adapter interface capable of transmitting an HTTP encoded facsimile image; a fax processor, the real-time fax processor capable of receiving a one or more audio streams from the audio adapter interface and packetizing the one or more audio streams into an HTTP encode facsimile image; where the facsimile is capable of being transmitted from a source facsimile machine through an voice over IP adapter, and further transmitted to a destination facsimile machine. | 08-19-2010 |
| 20100215033 | PREFERENTIAL ROUTING OF SECURED CALLS - Installed in an IGAR gateway is intelligence for determining the capabilities of an endpoint. Many older generation secure phones are not IP capable and are thus not directly capable of operating in a VOIP environment. The intelligence allows backwards compatibility of IGAR to legacy phones by recognizing that the endpoint is not IP capable and forcing the secure connection to be routed over PSTN. IGAR could also be included between independent instances of a communications manager (CM). Currently IGAR is supported on only a single CM controlling PSTN gateways, and not between independent CMs. This embodiment recognizes that incoming PSTN call based on a DN and once answered, in-band digits are passed from the originating PBX to the destination PBX in order to route the call within the answering PBX. | 08-26-2010 |
| 20100215034 | ADAPTIVE R99 AND HS PS (HIGH SPEED PACKET-SWITCHED) LINK DIVERSITY FOR COVERAGE AND CAPACITY ENHANCEMENT OF CIRCUIT-SWITCHED CALLS - A system and methodology that facilitates adaptive link diversity for enhanced coverage and capacity during user data communication in a UMTS (Universal Mobile Telecommunications System) is provided. Specifically, current radio conditions associated with the user data are monitored and analyzed. Moreover, a switching and/or concurrent transport mechanism is implemented for communication between a NodeB and UE (User Equipment), when the current radio conditions change beyond a predefined level. In particular, a CS (Circuit Switched) over HSPA (High Speed Packet Access) connection is reconfigured to an R99 (Release 99) CS connection, or a concurrent R99 CS connection is provided along with the CS over HSPA connection, when detected that radio conditions have degraded beyond a predefined threshold. In one aspect, the selection between switching to a new transport mechanism and, adding a concurrent transport mechanism is based on an analysis and/or operator defined conditions. | 08-26-2010 |
| 20100215035 | EMBEDDED COMMUNICATION APPARATUS, METHOD AND SYSTEM FOR USING THE SAME - A method for network connectivity of an embedded communication apparatus comprises the steps of: registering the domain name and the dynamic IP address of an embedded communication apparatus on a gateway, wherein the dynamic IP address comprises the ID code of the embedded communication apparatus and the domain name of the gateway; connecting an Internet user intending to connect with the embedded communication apparatus according to the domain name thereof to the gateway; dispatching the connection request from the Internet user to the embedded communication apparatus via the gateway; and connecting the embedded communication apparatus to the Internet user. | 08-26-2010 |
| 20100215036 | METHOD FOR TRANSFERRING SESSION IN CONVERGED INTERNET PROTOCOL MESSAGING SYSTEM - A method is provided for transferring a session between multiple devices by a target device, in which the target device selects a particular session of a source device and sends a request for session transfer for the selected session to a call server, the target device acquires from the call server data that has been transmitted from the remote party's device of the particular session and temporarily stored in the call server after the session transfer request, and the target device sends a message indicating completed acquisition of the temporarily stored data to the call server, and receives the particular session transferred in response thereto. | 08-26-2010 |
| 20100215037 | MULTIMEDIA SESSION CALL CONTROL METHOD AND APPLICATION SERVER - A multimedia session call control method and an Application Server (AS) are provided. The multimedia session call control method includes these steps: a multi-UE party performs a multimedia session with a peer under control of an AS; a master UE of the multi-UE party establishes a session with a third party under control of the AS; and the AS binds a call leg between a slave UE of the multi-UE party and the AS to the session established with the third party. | 08-26-2010 |
| 20100215038 | METERING IN PACKET-BASED TELEPHONY NETWORKS - One embodiment of the present invention facilitates efficient metering in a packet network environment by providing a single metering message, which contains sufficient information to provide the complete call tariff model for a particular call. The media gateway receiving the message can analyze the information provided in the message to determine how to provide metering pulses for all phases of the call, as well as any one-time charges, such as setup and add-on charges. Another embodiment of the invention provides a way for handling fractional pulse counts in an efficient manner. Yet another embodiment facilitates the handling of situations where charge intervals do not divide evenly into the phase duration of the phase associated with the call. In still another embodiment, the amount of information necessary to deliver the parameters of the call tariff model is minimized to reduce the overhead necessary for facilitating the metering process. | 08-26-2010 |
| 20100215039 | Intelligent Interactive Call Handling - An intelligent interactive call handling system is provided that typically includes a central office, a call-handling device, and an internet call routing system. The central office typically triggers a query responsive to receiving a call request. The call-handling device is coupled to the central office, receives the query, and triggers an internet call routing query. The internet call routing system, which is coupled to the call-handling device, typically receives the internet call routing query, determines presence of the called party with respect to at least one registered communication device, sends a prompt to the called party at said at least one registered communication device responsive to the presence determination, receives a reply from said at least one registered communication device, and routes the call responsive to the reply. Methods and other systems are also provided. | 08-26-2010 |
| 20100220714 | METHOD AND SYSTEM FOR MANAGING INTERNAL AND EXTERNAL CALLS FOR A GROUP OF COMMUNICATION CLIENTS SHARING A COMMON CUSTOMER IDENTIFIER - A method and network element for implementing a virtual PBX feature for a customer associated with a plurality of endpoints. The method comprises receiving information regarding a call. Based on the information regarding the call, it is determined if the call is an external inbound call or an internal call that identifies a particular one of the endpoints. Responsive to determining that the call is an external inbound call, the call is caused to be routed to the plurality of endpoints associated with the customer, while responsive to determining that the call is an internal call that identifies a particular one of the endpoints, the call is caused to be routed to the particular one of the endpoints. This allows members of a small business or household to share a common external subscriber line, while also allowing the members to reach one another with ease. | 09-02-2010 |
| 20100220715 | TECHNIQUE FOR PROVIDING TRANSLATION BETWEEN THE PACKET ENVIRONMENT AND THE PSTN ENVIRONMENT - Voice over Internet Protocol (VoIP) calls received in a Hybrid Fiber Coax (HFC) network ( | 09-02-2010 |
| 20100220716 | Methods for Enhancing SDP Preconditions Signalling for IP Multimedia Sessions - This application describes how Session Description Protocol (SDP) preconditions signaling can be enhanced to support lead role negotiation, precondition capability exchange, premature precondition attempts and concatenated preconditions processing. The application describes the use of send and receive tags in an SDP message for a given media line. In a given message, a success or failure tag may be associated with a send or receive tag in addition to an optional or mandatory condition indicator tag. A lead role indicator may also be associated with a send or receive tag to indicate a desired preference with regard to the sender or receiver taking the lead role. These additions lead to a greater chance of successful session set-up completion, reduce the number of signaling exchanges in general, and enable precondition attempts to be started earlier and to be executed in parallel. | 09-02-2010 |
| 20100220717 | Method and apparatus for controlling rate of voice service in a mobile communication system supporting voice service via packet network - A method for controlling a rate of a voice service in a mobile communication system supporting the voice service via a packet network. The method includes the steps of receiving a control message at a terminal from a radio network controller (RNC); if the control message indicates control of a downlink rate, determining a downlink rate according to the control message; setting a Change Mode Request (CMR) field of an uplink Voice over Internet Protocol (VoIP) packet according to the downlink rate, and transmitting the uplink VoIP packet from the terminal to the RNC; if the received control message indicates control of an uplink rate, determining an uplink rate according to the control message; and generating an uplink VoIP packet including uplink voice data generated according to the determined uplink rate and frame type (FT) information indicating the determined uplink rate, and transmitting the uplink VoIP packet from the terminal to the RNC. | 09-02-2010 |
| 20100220718 | Method for detecting calls and corresponding units - A method for detecting calls is disclosed. A call request initiated by a calling terminal device is received by an access unit of a called terminal device of a data packet transmission network. An invite message is sent from the access unit to the called terminal device. A 200-OK message from the called terminal device is received by the access device. In the event of a detection request, a detection message is received by the access unit in order to initiate a detection of the calling terminal device. Additionally, a telephone terminal device includes a controller, which responds with a 200-OK to an invite message, the terminal device sends a detection message to the access unit in order to initiate a detection of a calling terminal device. An access unit which automatically stores an identifier of a calling terminal device is also disclosed. | 09-02-2010 |
| 20100220719 | Call processing method, system and equipment of same number service - A call processing method, a call processing system and call processing equipment of a same number service are disclosed. The method includes: receiving a call which is initiated by a calling client and carries an initial called number, and sending a message of the called number with a same number service characteristic to first switching equipment in an IP network when the initial called number is a number of the same number service; and receiving a call request initiated by the first switching equipment, starting same number service processing of the initial called number according to the message carried in the call request, and calling a same number terminal corresponding to the initial called number. The embodiment of the invention helps realize the same number service between a SIP intelligent terminal in the IP network and ordinary terminals in other communication networks. | 09-02-2010 |
| 20100226361 | Multi-Vendor IMS Architecture - In an internet protocol multimedia subsystem architecture including multiple home subscriber servers, an apparatus (home subscriber server proxy) is interposed between a call session controlling function and/or an application server and the multiple home subscriber servers for adapting signaling messages exchanged between the call session controlling function (and/or the application server) and the home subscriber servers, so as to overcome interoperability issues. Such apparatus can be exploited for correctly routing the signaling messages toward the home subscriber server, thus rendering unnecessary the presence of a service locator function in a multi home subscriber server internet protocol multimedia subscription architecture. | 09-09-2010 |
| 20100226362 | Intelligent Call Mapping and Routing for Low Cost Global Calling on Mobile Devices Including SmartPhones - A method for providing international telephone call service to a calling party using a PSTN enabled communication device includes dialing the destination telephone number and establishing a connection between a software application installed on the communication device and an application server, authenticating the calling party using the user ID and the caller ID. When the calling party is authenticated, the method includes assigning a local DID number having the same or a nearby area code as the caller ID, notifying the communication device of the assigned local DID number, storing the destination telephone number and the assigned local DID number in a database, initiating a telephone connection over the PSTN to control signaling servers by dialing the assigned local DID number, retrieving the destination telephone number associated with the local DID number from the database, and establishing a voice-based connection between the caller and the callee. | 09-09-2010 |
| 20100226363 | ALTERNATE ROUTING OF VOICE COMMUNICATION IN A PACKET-BASED NETWORK - A method for performing alternate and therefore least cost routing in distributed H.323 Voice over IP (VoIP) networks is provided. With this method, the VoIP network consists of a hierarchy of gatekeeper (GK) functions to provide alternate routing, network element redundancy, and scalability. The alternate routing function is performed by a directory gatekeeper with route selection advancing from a first route to a second route by either of two conditions: (1) there are no resources available to terminate the call in the first zone; and (2) a lack of response to the directory GK request for such resources. | 09-09-2010 |
| 20100226364 | System and Method for Device Registration Replication in a Communication Network - A system for device registration replication in a packet-based network includes a first call manager and a second call manager that are coupled to the packet-based network. The first and second call managers each control one or more devices and store composite registration information associated with the devices. The first call manager communicates status information to the second call manager in response to a change in the control status of a device controlled by the first call manager. The second call manager updates the composite registration information stored by the second call manager in response to receiving status information from the first call manager. | 09-09-2010 |
| 20100226365 | VOICE OVER INTERNET PROTOCOL MARKER INSERTION - A watermark is inserted or overwritten into a packetized voice stream in a VoIP environment to characterize the voice data stream for various functions, such as providing certain in-band audible information or markers for detection. A visual type of marker can be inserted to measure delay for various applications, such as the round trip delay associated with providing directory assistance services, including measuring the delay from providing a prompt to a caller to the their response. The visual marker facilitates use of processes to detect measuring points for measuring delays. Audible markers can be used to provide various types of audible signals, including informational tones to agents, as well as announcements to callers. | 09-09-2010 |
| 20100232417 | MOVING SERVICE CONTROL WITHIN A MOBILE TELEPHONY SERVICE PROVIDER NETWORK FROM A CHANNEL ACCESS DOMAIN TO AN IP DOMAIN - A telecommunication call attempt involving a mobile phone number can be identified at a mobile telephony service provider network. Call control for the call can be moved from a circuit switched, channel access domain to an IP domain by moving control form a mobile switching controller (MSC) component to a media gateway controller (MGC) component. Signaling and routing can them be handled by an IMS core. A SIP registry (e.g., HSS or HLR) can associated enterprise applications with mobile phone numbers. Based upon SIP registry entries, the IMS core can communicate with an enterprise, and receive results from SIP applications executing in the enterprise. The results can change signaling and/or routing behavior of the call attempt. | 09-16-2010 |
| 20100232418 | Method and Apparatus for One Number Mapping Directory Presence Service - A method includes associating an e-mail address with a plurality of telephone numbers; associating one of the telephone numbers with a one number service ( | 09-16-2010 |
| 20100232419 | PROVIDING FIBRE CHANNEL SERVICES AND FORWARDING FIBRE CHANNEL OVER ETHERNET FRAMES - In one embodiment, an apparatus may include a first interface configured to be communicatively coupled, via a network, to a second interface and a fibre channel services module. The first interface may be configured to receive a fibre channel service from the fibre channel services module, establish communication with the second interface, and communicate a fibre-channel-over-Ethernet (FCoE) frame to the second interface, via a forwarder that forwards the FCoE frame without employing a fibre channel switching element. Other embodiments are described and claimed. | 09-16-2010 |
| 20100232420 | COMMUNICATION APPARATUS AND RELATED METHOD - A communication apparatus includes a network interface, a voice processor, a data bus and a microprocessor. The voice processor includes a judging module and a voice codec module coupled with the judging module. The judging module is coupled with the network interface. The microprocessor is coupled with the voice-processor through the data bus. A packet-based local area connection is formed between the voice processor and the microprocessor. The network interface is used to receive a network resource message and a VoIP message. When the network resource message is received by the network interface and sorted by the judging module, the voice processor transfers the network resource message to the microprocessor through the local area connection. When the VoIP message is received by the network interface and sorted by the judging module, the judging module transmits the VoIP message to the voice codec module for further processing. | 09-16-2010 |
| 20100232421 | AUDIO/VIDEO COMMUNICATIONS SYSTEM - An audio/video communication system is provided which includes: a web server providing a user system with a phone icon or button indicating a call receiver and transmitting a phone identifier LN for identifying the receiver allocated to the phone button when a user clicks the icon or button; and a gateway module performing a call setup in response to a data connection request for the audio/video communication from the user system, specifying the user identifier DN for identifying the user system from another user system, transmitting the phone identifier LN to the IP-based telephone exchanger, and relaying a communication between a phone connected to the IP-based telephone exchanger and the user system to progress the audio/video communication. | 09-16-2010 |
| 20100232422 | GROUPING OF USER IDENTITIES IN AN IP MULTIMEDIA SUBSYSTEM - The present invention is aimed to provide a more flexible data structure where any IMPU, even those of the SIP URI type, may be shared by more than one IRS in order to simplify the registration of an IMPU for users of a Fixed-Mobile Convergent network. To this end, there is provided a flexible data structure wherein a number n of IMPUs of a user may be distributed in a number m of Implicit Registration Sets, wherein a given IMPU may be shared by more than one IRS, each IRS is associated with an access condition, and the explicit registration of said given IMPU under a given access condition triggers the implicit registration of those IMPUs in the IRS associated with said access condition, whilst the registration status of IMPUs in any other IRS remain unchanged. | 09-16-2010 |
| 20100232423 | IP PHONE SYSTEM AND IP PHONE TERMINAL REGISTRATION METHOD - A registration method of a voice terminal in a call connecting apparatus, wherein; when a registration request of said voice terminal including a first distinguishing information is received, said call connecting apparatus refers to a storage unit which stores distinguishing information and telephone number of said voice terminal; said call connecting apparatus judges whether there is a distinguishing information that is consist with said first distinguishing information; and when there is no consisted distinguishing information, said call connecting apparatus registers said first distinguishing information in said storage unit with correspondence to a telephone number which is not taken correspondence to any said distinguishing information. | 09-16-2010 |
| 20100232424 | METHOD OF AND SYSTEM FOR PROVIDING QUALITY OF SERVICE IN IP TELEPHONY - A method and system for providing quality of service in an IP telephony session between a calling party and a called party establishes a high quality of service ATM virtual circuit for the session between first and second devices, each of the devices having ATM capability and IP capability. The first and second devices provide bidirectional translation between IP media and ATM media. The system transports IP media for the session between the calling party and the first device, and between said called party and a second device. The virtual circuit transports ATM media for the session between the first and second devices. An intelligent control layer provides IP and ATM signaling to set up the session. | 09-16-2010 |
| 20100238918 | Method and Device for Transmitting Data Using DSL Technology - The invention relates to a method and a device for transmitting data, wherein when transmitting data with DSL technology transmission rates are compared. | 09-23-2010 |
| 20100238919 | SYSTEM AND METHOD FOR TELECOMMUNICATION WITH A WEB-BASED NETWORK, SUCH AS A SOCIAL NETWORK - A system and method is described for establishing a communications session between a telecommunications device and one or more registered users on web-based networks, such as social networks. Further details and features are described herein. | 09-23-2010 |
| 20100246565 | System and method for displaying a called party calendar on a voice over IP phone display - A system and method for displaying a contact's availability information on a display of a voice over internet protocol (IP) phone is disclosed. The method includes sending a request for a selected telephone contact's availability information from the IP phone to a web service calendar module operable on a web server connected to the IP phone. The telephone contact's availability information is extracted from an application server connected to the web server. The availability information is formatted for display in a graphical user interface on the IP phone. The availability information for the telephone contact is the displayed on the IP phone to enable a user to determine when the selected telephone contact is available to receive a telephone call. | 09-30-2010 |
| 20100246566 | Serverless gateway infrastructure for voice or video - A system and method to provide voice or video over IP without a centralized control infrastructure is enabled by an overlay network of software devices. Such a device is comprised of VoIP server, PBX, PBX database, and a control module. The control module is used to store and retrieve items stored in the distributed databases hosted on the overlay networks. Two main functions provide by the serverless infrastructure are: VoIP call setup and tear-down, and accounting for a service provider. | 09-30-2010 |
| 20100246567 | SYSTEM AND METHOD FOR MANAGING CREATED LOCATION CONTEXTS IN A LOCATION SERVER - A system and method for creating a location uniform resource identifier (“URI”) for determining the location of a target device. A location request may be received for a target device. Location context information may be collected for the target device including starting information, validating information and policy information. This collected location context information may be encrypted in a location information server and converted to a form compatible with URI syntax. A location URI may then be constructed as a function of the converted information. | 09-30-2010 |
| 20100246568 | TELEPHONY SYSTEM WITH INTELLIGENT ENDPOINTS OR INTELLIGENT SWITCHES TO REDUCE DEPENDENCY OF ENDPOINTS ON APPLICATION SERVER - A system and a method are disclosed for reducing interaction between a server and an endpoint while executing features on an endpoint. The endpoint, and not the application server, includes part or all of the implementation of UT logic and feature logic. The endpoint therefore does not have to rely on server's instructions for executing a feature. The endpoint also includes an endpoint determination module for determining the parts of the UT logic and feature logic implemented on the endpoint and the parts implemented on a switch or a server. | 09-30-2010 |
| 20100246569 | TEMPORARY CONNECTION NUMBER MANAGEMENT SYSTEM, TERMINAL, TEMPORARY CONNECTION NUMBER MANAGEMENT METHOD, AND TEMPORARY CONNECTION NUMBER MANAGEMENT PROGRAM - The present invention makes it possible to connect to and communicate with a connect destination by using a temporary number. In the present invention, a temporary number user terminal ( | 09-30-2010 |
| 20100246570 | COMMUNICATIONS SESSION PREPARATION METHOD AND APPARATUS - Systems and methods for providing contextual information to one or more parties to a communications session are provided. More particularly, context information relevant to a party to a communications session is delivered to another party to the communications session as part of a communications message. In addition to information providing an identification of a party for whom context information is provided, embodiments of the present invention may make use of supplemental information in selecting context information for delivery. | 09-30-2010 |
| 20100246571 | SYSTEM AND METHOD FOR MANAGING MULTIPLE CONCURRENT COMMUNICATION SESSIONS USING A GRAPHICAL CALL CONNECTION METAPHOR - Disclosed herein are systems, methods, and non-transitory computer-readable storage media for managing a plurality of concurrent communication sessions via a graphical user interface (GUI). A system configured to practice the method presents a set of connected graphical elements representing a structure of the respective communication session via the GUI for each of a plurality of concurrent communication sessions. Each communication session has at least two participants and the appearance of the set of connected graphical elements is based on a communication mode. The system receives user input associated with one set of connected graphical elements and having an action associated with the respective communication session, and performs the action based on the received user input. The communication mode is one of voice over IP (VoIP), phone, videoconference, instant messaging, text messaging, and email. The action can combine two communication sessions or split one communication session into multiple communication sessions. | 09-30-2010 |
| 20100246572 | METHOD AND APPARATUS FOR PROVIDING USER ACCESS VIA MULTIPLE PARTNER CARRIERS FOR INTERNATIONAL CALLS - A method and apparatus for providing subscribers of a VoIP service provider to take advantage of wholesale arrangements made by the VoIP service provider with one or more international partner carrier network providers to one or more international countries are disclosed. Specifically, the present method enables a VoIP service provider to display a web page to their subscribers, for each destination country, with one or more international partner network providers and their corresponding calling rates and/or call completion success rates to each particular destination country. | 09-30-2010 |
| 20100246573 | TELECOMMUNICATION SYSTEM WITH PACKET-SWITCHED-MULTIMEDIA-SESSION-TO-CIRCUIT-SWITCHED-CALL TRANSFERRAL - Telecommunication systems with packet-switched multimedia terminals and nodes for packet-switched multimedia sessions and with servers for exchanging multimedia signaling information for the packet-switched multimedia sessions and with terminating units are provided with gateways for in response to transferral messages originating from the packet-switched multimedia terminals and arriving at the servers transferring packet-switched multimedia sessions between packet-switched multimedia terminals and nodes to circuit-switched calls between gateways and circuit-switched terminals via switches, to continue possibly interrupted sessions via replacing calls. The servers send invitation messages to gateways, which send setup messages to switches for setting up circuit-switched calls via partly alternative communication paths. The servers send information messages to terminating units for bringing the terminating units from session level to call level. Preferably, a packet-switched multimedia terminal and a circuit-switched terminal are one and the same terminal. | 09-30-2010 |
| 20100246574 | System and Method for Processing Packet Domain Signal - Embodiments of the present invention disclose a system and a method for processing a packet domain service signal, which enable a terminal that does not support an access control protocol of an Internet Protocol Multimedia Subsystem (IMS) to access the IMS and acquire the services in the IMS. An AGCF is added for shielding the differences of the users on the basis of the IMS defined in the | 09-30-2010 |
| 20100246575 | Virtual PBX based on Feature Server Modules - A virtual private branch exchange is formed by a plurality of interconnected feature server modules, each having an integral feature server that is configured and operates independently of the other feature server modules. Within a virtual private branch exchange, the feature server modules may be logically arranged in a hierarchy having at least a main feature server module and one or more subordinate feature server modules. A particular feature server module may operate in multiple virtual private branch exchanges, and may have a distinct set of rules for handling calls originating in different virtual private branch exchanges. | 09-30-2010 |
| 20100254370 | METHOD AND APPARATUS FOR MANAGING COMMUNICATION SESSIONS - A system that incorporates teachings of the present disclosure may include, for example, a server operably couplable to an Internet Protocol Multimedia Subsystem (IMS) network and an Interactive Television (ITV) system, where the server includes a controller to receive a session transfer request from a first communication device operably connected to the ITV system and presenting media content where the session transfer request includes identification information associated with a second communication device operably connected to the ITV system, and transmit an INVITE message to the second communication device and transmit a media adjustment message to a Media Resource Function Processor (MRFP) of the IMS network, where the media content is adjusted and transmitted to the second communication device based on receipt of the media adjustment message, where the adjusted media content is generated by the MRFP based on the identification information associated with the second communication device. Other embodiments are disclosed. | 10-07-2010 |
| 20100254371 | VOIP DEVICE AND METHOD OF PREVENTING NOISE GENERATION THEREBY - A Voice over Internet protocol (VoIP) device includes a time division multiplexing (TDM) bus, a plurality of digital signal processors (DSPs), and a plurality of subscriber line interface circuits (SLICs). The SLICs are respectively connected to a corresponding plurality of telephones. The VoIP device distributes the TDM bus to a plurality of calling timeslots and a special timeslot, and allocates at least one of the calling timeslots to each of the SLICs, selecting one of the DSPs as a special DSP. The VoIP device further directs the special DSP to generate an alternating voltage signal to the special timeslot, and directs the SLICs to receive the alternating voltage signal from the special DSP by the special timeslot to prevent the VoIP device from being locked at a high voltage and from generating noise. | 10-07-2010 |
| 20100254372 | SYSTEM AND METHOD FOR ENHANCING IMS CENTRALIZED SERVICES - A system and method of providing an enhanced service in a telecommunications network. The system includes a telecommunications network utilizing a circuit switched and packet switched access capability. The system also includes a sending User Equipment (UE) originating a call to a receiving UE. The sending UE sends an indicator informing the network to wait for further call information before proceeding with a call request towards the receiving UE. The system also includes a control node for routing calls within the network. The control node combines the call information received from the sending UE with call routing information of the call for connecting the call with the receiving UE to form a call request to the receiving UE. Upon receiving the call request message by the receiving UE, the call is connected and a media path is established between the receiving UE and the sending UE. | 10-07-2010 |
| 20100254373 | System and Method for Coordinating between Multiple Telephony Channels - A system comprising: an IP telephony interface communicatively coupled to an IP telephony service; a secondary telephony interface communicatively coupled to a secondary telephony service; and a telephone connection module to select between the IP telephony service and the secondary telephone service based on one or more specified telephony connection conditions. | 10-07-2010 |
| 20100254374 | PATCH PANEL FOR USE IN DELIVERING VOICE AND DATA TO END USERS - A patch panel that comprises a housing exhibiting a front face, in addition to first, second and third connectors. Each second connector corresponds to one of the first connectors, while each third connector also corresponds to one of the first connectors. Each first, second and third connector provides access, via the front face of the housing, to a respective set of terminals disposed at a set of positions relative to the respective connector. Each terminal in a first subset of the terminals to which one of the first connectors provides access is connected to a corresponding terminal to which the corresponding second connector provides access. Each terminal in a second, complementary subset of the terminals to which that same one of the first connectors provides access is connected to a corresponding terminal to which the corresponding third connector provides access. | 10-07-2010 |
| 20100254375 | INSTANT INTERNET BROWSER BASED VoIP SYSTEM - The present invention is an instant Internet browser based VoIP system with a VoIP client in the form of temporary VoIP applets that can start in a Web browser and can establish an instant peer-to-peer connection with another web-based or hardware embedded/installed VoIP client using session initiation protocol (SIP) and real-time transport protocol (RTP) audio streaming. The applet is a small file that is easily loaded onto a user's browser and uses application program interfaces (APIs) that require no additional libraries. The applet is written in JAVA, although other programming languages may also be used to write the applet. | 10-07-2010 |
| 20100260170 | SYSTEM AND METHOD FOR DYNAMIC CALL ROUTING - A telecommunications system is disclosed which may include a VOIP (Voice Over Internet Protocol) network having a plurality of network elements; at least one carrier data center; communication links enabling communication between the network elements and the at least one carrier data center, wherein the carrier data center is operable to receive a query describing a communication session active at a given one of the network elements, over one of the communication links, and to generate a routing table in response to the query. | 10-14-2010 |
| 20100260171 | METHOD AND APPARATUS FOR PROCESSING NUMBER PORTABILITY IN INTERNET PHONE - The present invention relates to a method and apparatus for processing a number portability call and a request for number portability in a VoIP, wherein a VoIP network access to a L-NPDB of each communication carrier to process the number portability call. Especially, to process a number portability call and request for number portability between various types of communication network such as VoIP networks, wired phone network, and mobile network, the apparatus includes an mobile number portability management system comprised of computer systems such as an NPDB, DB system where a VoIP carrier accesses, a router, an NPMS, and the like, a computer system of a VoIP network which can access and search the NPDB to process a phone call in VoIP network or can performs a relay-access of the phone call, and a switch board. | 10-14-2010 |
| 20100265938 | Enhanced system operation by virtualization - A system and method for enhanced operation of internet protocol (IP) telephony is disclosed. The system comprises a call server. The call server includes a shared resources module, a virtual machine environment in communication with the shared resources module the virtual machine having a plurality of virtual machines. A plurality of telecommunication function applications operate on separate instances. Each instance is operable to be connected to a customer through a network connection in communication with at least one of the plurality of virtual machines. | 10-21-2010 |
| 20100265939 | SYSTEM AND METHOD FOR PROCESSING A PLURALITY OF REQUESTS FOR A PLURALITY OF MULTI-MEDIA SERVICES - A system and method for processing a plurality of requests for a plurality of multi-media services received at a Private Service Exchange (PSX) defined on the system from a plurality of IP-communication devices. The system further includes a media server (MS) coupled to the PSX and to at least one IP Service Control Point (IP-SCP), which is operative to process the plurality of requests for the plurality of multi-media services. The IP-SCP further selectively directs the requests to the media server, which operates to form a preliminary multi-media communication path with a calling communication device. The MS further operates to play a plurality of announcements to the calling communication device over the preliminary multi-media communication path, as well as to collect caller-entered data from the calling communication device over the preliminary multi-media communication path. | 10-21-2010 |
| 20100272096 | CALL HANDLING FOR IMS REGISTERED USER - The present invention proposes a solution for providing IMS services to users having circuit-switched controlled terminals. In particular, it is proposed, in order to allow IMS to take the full call and service control, to combine circuit-switched and packet-based multimedia functionality in a new node type called Mobile Access Gateway Control Function (MAGCF). In particular the present invention provides a method for ensuring that the MAGCF node acts as a roaming anchor point in order to enforce the handling of originating and terminating calls in the IMS. | 10-28-2010 |
| 20100272097 | INTERNET PHONE TERMINAL USING WIDEBAND VOICE CODEC AND COMMUNICATION METHOD FOR INTERNET PHONE - Provided are an Internet phone terminal that applies a wideband voice codec, and an Internet phone communication method. A wideband voice signal received from the Internet through a wired line or wirelessly is decoded using the wideband voice codec, and a wideband voice signal received through a microphone supporting a wideband is encoded using the wideband voice codec, so that the Internet phone terminal can provide high quality voice communication. | 10-28-2010 |
| 20100272098 | METHOD AND SYSTEM FOR VOIP PBX CONFIGURATION - A VOIP PBX configuration system and method enable remotely configuring a VOIP PBX server connected to the Internet. The system includes a central server connected to the Internet and having a database of VOIP PBX configuration templates. The central server is accessible via a website user interface having configuration fields to be populated by a user. The central server includes computer readable program code components configured to modify the configuration templates to generate configuration instructions based on user entries in the configuration fields, and transmit the configuration instructions to the VOIP PBX server to configure the VOIP PBX server based on the user entries. | 10-28-2010 |
| 20100272099 | Internet Telephony with Interactive Information - A subscriber ( | 10-28-2010 |
| 20100272100 | System and Method for Managing Call Routing in a Network Environment Including IMS - In one embodiment, a scheme is disclosed for managing call routing in a network environment including a circuit-switched (CS) network and an IP multimedia subsystem (IMS) network. When a call is originated by a user equipment (UE) device in the CS network, call information associated with the call is provided to a call continuity control function (CCCF) network node disposed in the IMS network. At the CCCF node, a pool of E.164 numbers are maintained as IP multimedia routing numbers (IMRNs) which are mapped to or otherwise associated with called party numbers. The CCCF node dynamically allocates a select IMRN with respect to a called party number received from the UE device and returns it to the UE device. The dynamically allocated IMRN is then utilized for routing the call towards the called party. | 10-28-2010 |
| 20100272101 | SYSTEM AND METHOD OF COMMUNICATION IN AN IP MULTIMEDIA SUBSYSTEM NETWORK - A system and method of communication in an IMS network is disclosed. An apparatus that incorporates teachings of the present disclosure may include, for example, a call processing server having a controller element that receives from a terminal device a calling ID for establishing communications with a called party, submits to a telephone number mapping (ENUM) server a query corresponding to the calling ID, receives from the ENUM server a plurality of communication identifiers retrieved from a Naming Authority Pointer record according to a grade of service (GoS) of the called party, and selects according to the GoS of the called party a communication identifier from the plurality of communication identifiers to establish communications with the called party. Additional embodiments are disclosed. | 10-28-2010 |
| 20100278171 | Methods and Apparatus for Enhancing the Scalability of IMS in VoIP Service Deployment - Methods for Enhancing the Scalability of IMS in VoIP Service Deployment lower the number of messages transmitted between functions of an IMS network. The number of messages transmitted between functions of an IMS network are lowered by storing and utilizing predetermined configuration information pertaining to the calling and called parties including the media and codecs the parties support. The predetermined configuration information, which may be based on a prior peering business agreement, supports the implementation of a one round procedure for establishing an IMS communication session. | 11-04-2010 |
| 20100278172 | IMAGE COMMUNICATION APPARATUS - An image communication apparatus comprises: a call connection control unit that establishes a session with a communication partner using an SIP message; and an image communication control unit that controls an image communication, wherein (i) when the call connection control unit receives, as a calling party, from a called party, an INVITE SIP message in which a T.38 communication and a first priority transport are specified in a session description protocol, and when a second priority transport is set in the image communication apparatus of the calling party, the call connection control unit opens the second priority transport, and (ii) when no priority transport is set in the image communication apparatus of the calling party, the call connection control unit opens the first priority transport specified by the called party, and performs a T.38 communication using the opened transport. | 11-04-2010 |
| 20100278173 | METHOD AND SYSTEM FOR ROUTING CALLS OVER A PACKET SWITCHED COMPUTER NETWORK - The present invention describes how a trusted network routing authority, such as a VoIP inter-exchange carrier or clearinghouse can provide routing and secure access control across multiple network domains with a single routing and admission request. This technology can improve network efficiency and quality of service when an Internet Protocol (IP) communication transaction, such as a Voice over IP (VoIP), must be routed across multiple devices or administrative domains. This technology defines the technique of performing multiple route look-ups at the source of the call path to determine all possible routes across intermediate domains to the final destination. The VoIP inter-exchange carrier or clearinghouse then provides routing and access permission tokens for the entire call path to the call source. | 11-04-2010 |
| 20100278174 | Method and Arrangement for Network Roaming of Corporate Extension Identities - The present invention is a method, a node and a user terminal for roaming in an IP based main network. A node, in a sub-network of said main network receives a registration request from a user terminal belonging to said sub-network, wherein the registration request comprises an unique extension identity. The node is configured to determine whether said unique extension identity belongs to the sub-network of the requesting user terminal. In case said extension identity belongs to another sub-network, the node will request a routing server of the sub-network for an IP-address of a home sub-network for said unique extension identity. The routing server is configured to respond the IP address of the requested home sub-network. The node will transmit the IP address to the requesting user terminal for enabling the transmission of a registration request using the IP address to the home sub-network from said user terminal. | 11-04-2010 |
| 20100284395 | Method and apparatus for frame selection - To address the need for new techniques that are able to reduce delays in frame selection, a method such as that depicted in diagram | 11-11-2010 |
| 20100284396 | Communication system and method - A method, program and system for use in a communication system comprising at least a packet-based network. The method comprises: receiving names of users of the communication system retrieved from a first storage unit; and interacting with a document-browser application executed on a first user terminal, the document browser being configured to retrieve an electronic document from a second storage unit and display it on a screen, the document comprising at least a portion of text; wherein said interaction comprises analysing the text of the retrieved document in order to match one or more of said names with one or more respective corresponding text strings in said document, and displaying presence information in conjunction with the document to indicate an availability status of the user corresponding to the matched name. | 11-11-2010 |
| 20100284397 | HANDLING OUT-OF-SEQUENCE PACKETS IN A CIRCUIT EMULATION SERVICE - Various exemplary embodiments relate to a method and related network node having a playout buffer including one or more of the following: receiving a first packet, a second packet, a first set of at least one subsequent packet, wherein each packet includes a sequence number (SN); determining that the second packet is not in sequence with the first packet by determining that the SN of the second packet is not equal to the SN of the first packet plus an expected increment value; determining whether the second packet represents a jump in SNs by determining whether the SN of a first subsequent packet is equal to the SN of the second packet plus the expected increment value; and when the second packet represents a jump in SNs, gradually normalizing the playout buffer upon receipt of each subsequent packet. | 11-11-2010 |
| 20100284398 | SYSTEM AND METHOD FOR PROVIDING PHONE RELATED SERVICES TO DEVICES USING UPnP ON A HOME NETWORK - A system and method for exchanging call data between multiple devices using Universal Plug and Play (UPnP) on a home network. The system includes a telephony terminal, a first electronic device, a second electronic device, and a control point for selecting the telephony terminal and the first and second electronic devices for exchanging the call data, for setting a call reception connection between the telephony terminal and the first and second electronic devices, and forming a plurality of sessions for exchanging the call data between the selected telephony terminal and the first and second electronic devices. | 11-11-2010 |
| 20100284399 | MEDIA PATH OPTIMIZATION FOR MULTIMEDIA OVER INTERNET PROTOCOL - Methods for optimizing the media path between multimedia endpoints in a network are described. One embodiment allows avoiding having to relay the media traffic through a central device, such as a border controller's media controller element, and lets endpoints communicate directly under various conditions. | 11-11-2010 |
| 20100290452 | Method and Device for Establishing a Subject-Related Communication link - A method and a device establish a subject-related communication link between two users in a communication network. At least the called user is equipped with a telecommunication terminal and a computer workstation suitable for electronic messages. Electronic messages are provided with subject-related identifiers when the same are transmitted from the calling user to the called user. The calling user can indicate a subject-related identifier to a communication link between the calling user and the called user in the communication network when the communication link is established. The subject-related identifier is then transmitted along with the call information and is forwarded from the communication network to the computer workstation of the called user. The subject-related identifier is evaluated in the computer workstation, and messages including the subject-related identifier are then indicated to the called user. | 11-18-2010 |
| 20100290453 | Method and Communication Terminal for Providing VOIP - The invention relates to a method for providing Voice over IP (VoIP) in a communication system with a number of terminals operating with VoIP, between which a transmission of voice data according to VoIP or a signalling is achieved, wherein the signalling is achieved using the Computer Supported Telecommunication Application (CSTA) interface standard. Telephone services can be controlled by a computer using the CSTA protocol. As for conventional application, H323 protocol and SIP are used for IP telephony processing of audio/video streams of a conversation. The invention is based on replacing H323 protocol and SIP by a CSTA protocol only when the latter is correspondingly extended. | 11-18-2010 |
| 20100290454 | Play-Out Delay Estimation - A receiving terminal estimates a required jitter buffer depth for each received audio frame, by locating ( | 11-18-2010 |
| 20100290455 | METHOD AND APPARATUS FOR COMMUNICATION REQUEST TERMINATION ROUTING - A method and apparatus for call termination routing. The method comprises determining one or more characteristics of an incoming call, mapping the one or more characteristics to a termination policy, and routing the incoming call to a communication device. The incoming call is routed to the communication device in accordance with the mapped termination policy. The determining, mapping, and routing steps are performed by a controller computing device as known in the art. The apparatus comprises means for determining one or more characteristics of an incoming call, means for mapping the one or more characteristics to a termination policy, and means for routing the incoming call to a communication device. The incoming call is routed to the communication device in accordance with the mapped termination policy. | 11-18-2010 |
| 20100290456 | APPARATUS, METHOD AND COMPUTER-READABLE STORAGE MEDIUM FOR REGISTERING USER IDENTITIES - An apparatus is provided that includes a processor configured to maintain a first implicit registration set for a first apparatus, where the first implicit registration set includes a first identity unique to the first apparatus and a shared identity. The processor is also configured to maintain a second implicit registration set for a second apparatus, where the second implicit registration set includes a second identity unique to the second apparatus and the shared identity. In this regard, the first and second implicit registration sets may be maintained to enable registration of the first and second apparatuses with a network such that each of the first and second apparatuses are configured to receive communication requests to the respective first and second identities, and such that both of the first and second apparatuses are configured to receive communication requests to the shared identity. | 11-18-2010 |
| 20100296507 | Analog Voice Bridge - According to one embodiment, a communication system includes an analog voice bridge coupling one secure network domain to another. The analog voice bridge includes two codecs that are each coupled to a secure network domain and to each other through an analog voice line. One codec decapsulates an analog voice signal from a digital voice stream received from a terminal, and transmits the analog voice signal to the other codec through the analog voice line. The other codec encapsulate the analog voice signal in another digital voice stream and transmit the encapsulated digital voice stream to another terminal coupled through the other secure network domain. The analog voice line conveys the analog voice signal from the first codec to the second codec while restricting communication of the digital packet stream between the two secure network domains. | 11-25-2010 |
| 20100296508 | SYSTEM AND METHOD FOR BYPASSING DATA FROM EGRESS FACILITIES - An open architecture platform bypasses data from the facilities of a telecommunications carrier, e.g. an incumbent local exchange carrier, by distinguishing between voice and data traffic, and handling voice and data traffic separately. An SS7 gateway receives and transmits SS7 signaling messages with the platform. When signaling for a call arrives, the SS7 gateway informs a control server on the platform. The control server manages the platform resources, including the SS7 gateway, tandem network access servers (NASs) and modem NASs. A tandem NAS receives the call over bearer channels. The control server determines whether the incoming call is voice traffic or data traffic, by the dialed number, and instructs the tandem NAS how to handle the call. Voiced traffic is transmitted to a switch for transmission from the platform. Data traffic is terminated at a modem NAS, where it is converted into a form suitable for a data network, such as a private data network or an Internet services provider (ISP). The converted data is sent by routers to the data network. The data network need not convert the data, as the function has already been provided by the platform. In lieu of a conversion, the modems can create a tunnel (a virtual private network) between a remote server and the data network. | 11-25-2010 |
| 20100296509 | Method and Apparatus for Coordinating Internet Multi-Media Content with Telephone and Audio Communications - Internet content is coordinated with audio communications, such that two or more parties can view the same media content on the Internet while simultaneously communicating over a traditional telephony network or via voice over network. A user computer displays shared content that corresponds to a second computer's display, such that both parties view the same content on their browsers. Either of the parties is allowed to update the visual content of their browsers. Updates in the visual content are transmitted to the other parties so that all parties view the same, shared content. The shared content can include web pages, forms, applications, images, conferences, and files among other information. | 11-25-2010 |
| 20100296510 | METHOD AND SYSTEM FOR DELIVERY OF A CALLING PARTY'S LOCATION - A method and system for providing a service that delivers location information associated with a caller. The service operates in both wireline and wireless networks, providing called parties with the location information of calling parties who use either stationary terminal devices or mobile devices. The service can operate as a stand alone service or can be a part of a calling name delivery service (or caller-ID service), delivering location information in addition to the conventional name, number, date, and time. The components of the present invention include a service control point, an address database in communication with the service control point, and a network that tracks the locations of mobile network users. The system further includes a mapping converter if the location data provided by the network is not meaningful to a subscriber. | 11-25-2010 |
| 20100303057 | COMPUTER ASSISTED VOIP COMMUNICATION METHOD AND SYSTEM - A computer assisted VoIP communication method and system. A remote server comprises a database in which is stored generic data of a plurality of displayed mobile phones. A personal computer in communication with the remote server comprises a local memory device in which is storable user-specific data of a known mobile phone, and an input device for selecting a desired mobile phone displayed in the remote website and for downloading the generic data associated with the selected phone to the memory device. Following generation of a virtual phone having a shape, key arrangement and functionality similar to those of the known mobile phone on the computer screen, a type and recipient of a session to be established are defined by virtually selecting a desired number and sequence of keys of the virtual phone or by entering commands. A session is established by a VoIP application residing on the computer. | 12-02-2010 |
| 20100303058 | Providing session-based services to event-based networks using partial information - A method for communication includes, during a call conducted among two or more subscribers in a circuit-switched network, which operates in accordance with a first communication protocol that manages calls among the subscribers by exchanging discrete events among elements of the circuit-switched network, receiving from the circuit-switched network an incomplete subset of the events related to the call. Based on the incomplete subset of the events, at least one emulated communication session is generated in a packet-switched network that operates in accordance with a second communication protocol. Using the emulated session, a service platform in the packet-switched network is caused to provide a communication service to the call conducted in the circuit-switched network. | 12-02-2010 |
| 20100303059 | Providing session-based service orchestration to event-based networks - A method for communication includes, during a call conducted among two or more subscribers in a circuit-switched network, which operates in accordance with a first communication protocol that manages calls among the subscribers by exchanging discrete events among elements of the circuit-switched network, receiving from the circuit-switched network a sequence of the events related to the call. Based on the sequence of the events, at least one emulated communication session is generated in a packet-switched network that operates in accordance with a second communication protocol. Multiple call services are provided to the call conducted in the circuit-switched network from the packet-switched network by cascading multiple service sessions, each providing a respective one of the call services, in the packet-switched network responsively to the emulated communication session. | 12-02-2010 |
| 20100303060 | SECOND CALL MODE CALL SET-UP BETWEEN TWO USERS - A method is provided of setting up a call between first and second users having respective first and second application identifiers, shared between the users, the first user being connected to a communications application and being adapted to communicate in a first call mode with the second user, adapted to receive a connection request, to obtain an identifier of a second call mode of the first user and an identifier of the second mode of the second user and to set up a second call mode call between the two users as a function of the identifiers obtained. In addition, a method is provided of requesting the setting up of a second mode call between the first user and the second user, the first user being connected to a communications application and being adapted to communicate in a first call mode with the second user, adapted to send a connection request. A terminal and a server respectively are provided for implementing the call set-up request method and the call set-up method. | 12-02-2010 |
| 20100303061 | NETWORK COMMUNICATION SYSTEM FOR SUPPORTING NON-SPECIFIC NETWORK PROTOCOLS AND NETWORK COMMUNICATION METHOD THEREOF - The present invention is related to a technology of unified communication systems. In detail, the present invention is applied to a management of the communications of an enterprise end, that is, the combinations of the on-line presence of each user end and gateways in the enterprise end are brought into practice through variable embodiments. Hence, the number of each user end and the representative number of the enterprise end can be shown to represent the presence of on-line of the user end. More particularly, the present invention is focused on a network communication system for supporting non-specific network protocols and a network communication method thereof. | 12-02-2010 |
| 20100303062 | IP telephone and method for controlling supplementary services - A request destination management table that stores therein a service ID uniquely identifying a service being interrelated with a request destination to which execution of a service is requested and a report destination management table that stores therein a call ID uniquely identifying a call being interrelated with a report destination of a call state transition report indicative of state transition of the call are provided. An IP telephone receives a request for a supplementary service, obtains a request destination corresponding to a service ID of the supplementary service from the request destination management table, and transfers the request for the supplementary service to the request destination. The IP telephone receives a call state transition report, obtains a report destination corresponding to a call ID of the call state transition report from the report destination management table, and transfers the call state transition report to the report destination. | 12-02-2010 |
| 20100303063 | SYSTEM AND METHOD FOR IMPLEMENTING AND ACCESSING CALL FORWARDING SERVICES - A communications redirection service is implemented when current communications service data for an account is forwarded through a packet switching network. An instruction is received through the packet switching network for the communications redirection service to redirect communications addressed to the first receiving communications address to a second receiving communications address when the communications are from a specified sending communications address. The instruction for the communications redirection service is forwarded to a communications service manager through a data network. Communications addressed to the first receiving communications address are redirected to the second receiving communications address in accordance with the instruction for the communications redirection service when the communications are from the specified sending communications address. | 12-02-2010 |
| 20100303064 | HANDLING EMERGENCY CALLS USING EAP - A user (terminal) is allowed to make an emergency voice-over-Internet Protocol (VoIP) phone call through an access network, such as a wireless local area network (WLAN) using Extensible Authentication Protocol (EAP). The emergency call can be made with or without authentication credentials and is identified by the user's terminal transmitting a Network Access Identifier (NAI) having a user part and/or realm part that indicates the emergency nature of the call, such as e911@e911.com. In response to such an NAI, the caller can be immediately granted limited authentication for the purpose of connecting to an emergency call center. Alternatively, the user (terminal) can be authenticated through networks supporting emergency calls, such as the user's home network, if the terminal indicates to the access network authentication server a preference or requirement for using such networks. The call can be routed to the emergency call center either directly or via one or more intermediary networks, such as networks that support emergency VoIP phone calls. | 12-02-2010 |
| 20100303065 | Method and Apparatus for Accessing Service Resource Items That are For Use in a Telecommunications System - Service resource items for use in call setup in a telephone system are held on servers that are connected to a computer network which is logically distinct from the telephone system infrastructure; this computer network may, for example, make use of the Internet. Each service item is locatable on the network at a corresponding URI and is associated with a particular telephone number. A mapping is provided between telephone numbers and the URIs of associated service resource items. When it is desired to access a service resource item associated with a particular telephone number, this mapping is used to retrieve the corresponding URI which is then used to access the desired service resource item. | 12-02-2010 |
| 20100309904 | Power management in an internet protocol (IP) telephone - Power management is provided in an Internet protocol (IP) telephone and system to provide energy savings during times that the IP telephone is not in use or use is not expected. A low-power operating mode disables at least a portion of the IP telephone. The low-power operating mode may be initiated by a command received by the IP telephone from the IP telephone controller according to a schedule, which may be modified locally by the user to individualize the user's schedule. The low-power operating mode may alternatively be activated manually by a user pressing a special key, sequence or combination. The low-power operating mode is canceled upon an indication that a user either is or should be present at the IP telephone. | 12-09-2010 |
| 20100309905 | NETWORK PHONE - A network phone includes a base unit, a handset unit and a coiled wire cord connecting the handset unit and the base unit. The base unit is connected to at least one network line for receiving a first digital communication signal or transmitting a second digital communication signal via the network line, and further combining the power of the handset unit with the first digital communication signal so as to form an integrated communication signal. The handset unit decomposes the integrated communication signal back into power and the first digital communication signal. The coiled wire cord is adapted to transmit the integrated communication signal from the base unit to the handset unit, or to transmit the second digital communication signal from the handset unit to the base unit, thereby providing a net-phone device with signal conversion functionality with a coiled wire cord. | 12-09-2010 |
| 20100309906 | METHODS AND APPARATUS FOR MULTISTAGE ROUTING OF PACKETS USING CALL TEMPLATES - A method for multistage routing of packets using call templates is disclosed. An ingress call is filtered based on a plurality of ingress-call parameter values. A parameter value for the ingress call is modified based on a plurality of ingress-call-peer parameter values. A filtered ingress-call parameter value and at least one filtered ingress-call-peer parameter value from a plurality of ingress-call-peer parameter values are converted to an egress-call parameter value and an egress-call-peer parameter value, respectively. An egress call is filtered based on a plurality of egress-call parameter values. A parameter value for the egress call is modified based on a plurality of egress-call-peer parameter values. | 12-09-2010 |
| 20100316045 | Prioritising Messages in a Communications Network - A method and communications network node for allocating a priority level to an Internet Protocol (IP) packet containing all or part of a Session Initiation Protocol (SIP) message. One or more characteristics of the SIP message are determined, and the characteristics are mapped to a Differentiated Services Code Point (DSCP) value. The DSCP value is then applied to the IP packet header. | 12-16-2010 |
| 20100316046 | Method for performing gate coordination on a per-call basis - Network resources for a call between a calling party and a called party are allocated. The network resources for the call are reserved based on a reservation request. The network resources are reserved before any one network resource from the reserved network resources is committed. The reserved network resources for the call are committed when a called party indicates acceptance for the call. | 12-16-2010 |
| 20100316047 | COMMUNICATION TERMINAL DEVICE, DEVICE FOR DETERMINING POSSIBILITY OF DISCRIMINATING RELATION OF PSEUDONYMOUS-NAME COMMUNICATION IDENTIFIER, COMMUNICATION SYSTEM, COMMUNICATION METHOD AND STORAGE MEDIUM - To determine a relation discrimination possibility of a pseudonymous-name communication identifier so that, in each communication layer, no mismatch occurs between a pseudonymous-name communication identifier whose relation can be discriminated and a pseudonymous-name communication identifier whose relation cannot be discriminated. The Relation discrimination possibility determination means | 12-16-2010 |
| 20100322231 | VOIP DEVICE AND METHOD FOR ADJUSTING INTERRUPT TIME THEREOF - A voice over Internet protocol (VoIP) device for providing VoIP service for a telephone includes a time detecting module and a time adjusting module. The time detecting module is operable to receive a dual tone multiple frequency (DTMF) signal, detect interrupt time of the DTMF signal, and determine whether the interrupt time is less than a predefined time interval. The time adjusting module is operable to adjust the interrupt time to the predefined time interval upon the condition that the interrupt time is less than the predefined time interval. | 12-23-2010 |
| 20100322232 | MODEM AND CALLING PACKET PROCESSING METHOD THEREOF - A modem to process calling packets includes receiving a calling request packet from a software phone of a communication terminal, and determining if the calling request packet includes a special tag. If the IP phone is idle, the modem records a source IP address of the calling request packet, and modifies the source IP of the calling request packet to be an IP address of the IP phone, then the modem transmits the modified calling request packet to a server, and receives a calling reply packet from the server, then modifies a destination IP address of the calling reply packet to be the IP address of the communication terminal. The modem transmits the modified calling reply packet to the software phone to establish the call. | 12-23-2010 |
| 20100322233 | Switchboard For Multiple Data Rate Communication System - A switchboard device and methods of operation of same are disclosed. Embodiments of the invention may provide a flexible means of interconnecting wideband and narrowband communications interfaces, where wideband communications interfaces may transfer low-band data and high-band data, and narrowband communication interfaces may transfer low-band data. Low-band data may be combined and sent to a narrowband communications interface or a wideband communications interface. High-band data may be combined and sent to a wideband communications interface. The low-band data may represent audio signals below a predetermined frequency, while the high-band data may represent audio signals above the predetermined frequency. The predetermined frequency may be, for example, approximately 4 kHz. The spectral mask of the low-band data may meet the spectral mask of G.712. Methods of operating embodiments of the present invention are included. An additional aspect of the present invention may include machine-readable storage having stored thereon a computer program having a plurality of code sections executable by a machine for causing the machine to perform the foregoing. | 12-23-2010 |
| 20100322234 | IP TELECOMMUNICATION SYSTEM, METHOD FOR CONTROLLING COMMUNICATION IN IP NETWORK, CLIENT TERMINAL AND CLIENT SERVER - A terminal including: a remote control section for transmitting and receiving data with respect to a main device which performs call control processing with a target device via a telephony server in place of the terminal; and a call communication section for performing audio communication with the target device, wherein the remote control section transmits a command including a calling request for the target device and an IP address of the terminal to the main device, and receives an IP address of the target device from the main device, and the call control section performs audio communication with the target device using the IP address of the terminal and the IP address of the target device. | 12-23-2010 |
| 20100322235 | METHOD AND SYSTEM FOR AUTHENTICATED FAST CHANNEL CHANGE OF MEDIA PROVIDED OVER A DSL CONNECTION - A method and system for fast channel changes of media that is provided by carriers over an xDSL connection to a home. Each customer's subscriber information is stored at the DSLAM that supports the xDSL connection to the home. Also, each DSLAM supports multicast protocols so that only one instance of a channel is provided on the core network regardless of how many customers have requested access to the channel. | 12-23-2010 |
| 20100329238 | SYSTEM AND METHOD FOR EXPOSING THIRD PARTY CALL FUNCTIONS OF THE INTELLIGENT NETWORK APPLICATION PART (INAP) AS A WEB SERVICE INTERFACE - Systems and methods are described for exposing the third party call control functionality of a telecom signaling network as a web services interface. An intelligent network application part (INAP) plug-in is used to provide the translation logic of simple web service interface calls received from a client application, into the lower-level signaling protocol invocations needed to provide the third party call functionality at the network level. The INAP plug-in is deployed in a service access gateway positioned between the telecommunications signaling-based network and a multitude of service provider applications that seek to access various functions in the network. By implementing the INAP plug-in, applications are provided with access to third party call control (3PCC) within the network, without the necessity of invoking low-level signaling needed to establish calls, terminate or cancel calls, as well as obtain various call information. | 12-30-2010 |
| 20100329239 | SIP SERVLET APPLICATIONS CO-HOSTING - Methods, devices, and systems are provided for allowing a single machine, such as a server, to co-host multi-SIP Archive (SAR) applications offering SIP servlet based products. The concept of a Root Application Router is introduced that is adapted to coordinate other Sub-Application Routers rather than individual SARs. These other Sub-Application Routers are fully fledged Application Routers in their own right, but are unaware of the controlling Root Application Router. | 12-30-2010 |
| 20100329240 | Method for dialing between internet extensions - A method for dialing between Internet extensions is disclosed. When dialing between Internet extensions, just dial the switchboard phone number of SIP proxy server plus “-” and then dial the extension phone number of the opposite Internet extension directly. It is not necessary to use a voice guidance for asking dialing of the extension phone number of the opposite Internet extension. | 12-30-2010 |
| 20100329241 | APPARATUS AND METHOD FOR PREVENTING SPAMS IN VOIP SYSTEM - A system for preventing a spam-call for a VoIP system includes a communication network, a plurality of terminals connected via the network, and a server. The server includes a server black list DB, a connection control module, a membership information management module, and a server-side management module. Each of the terminals includes a terminal-side management module. With the system, VoIP spam can be prevented in a cost-effective way and users' convenience can be increased. | 12-30-2010 |
| 20100329242 | SERVER APPARATUS AND SPEECH CONNECTION METHOD - According to one embodiment, a server apparatus includes a memory, a determination module and a controller. The memory stores a management table associating terminal IDs specifying the terminals with media processing abilities owned by the terminals. The determination module refers the management table and determines whether information showing a media processing ability corresponding to the first terminal and information showing a media processing ability corresponding to the second terminal coincide with each other based on the reference result. The controller executes first processing for making speech connection between the first terminal and the second terminal by a peer-to-peer when the media processing abilities coincide with each other, and executes second processing for leading in a speech path between the first terminal and the second terminal to convert into the same media processing ability when the media processing abilities non-coincide with each other. | 12-30-2010 |
| 20100329243 | System And Method For Voice Service In An Evolved Packet System - A system and method for accessing voice services using a user equipment (UE) in a communication system is provided. The UE is configured to receive a first message which may include an audio session indication. The method includes the step of sending a second message in response to the first message, with the second message being based on one or more voice service indicators comprising at least one value. The second message may be a response indicating not to select an alternative domain. The second message may also be a not acceptable response. | 12-30-2010 |
| 20100329244 | System And Method For Voice Service In An Evolved Packet System - A system and method for enabling voice services using a network component in a communication system including a user equipment (UE) is provided. The UE is configured to receive a first message which may be a SIP request. The network component is configured to create a second message, or SIP request, based upon the first message. The network component further configured to subsequently receive a SIP response and select a subsequent action upon receiving the SIP response. | 12-30-2010 |
| 20110002325 | MULTIMEDIA TERMINAL DEVICE HAVING INTEGRATED TELEPHONE SYSTEM AND USER INTERFACE METHOD - Customer premise equipment provides a communication gateway with a network of a service provider and includes a multimedia terminal device for installation on the customer's premises typically at an out-of-the-way location. The multimedia terminal device includes a modem having an embedded media terminal adaptor and an integrated telephone base station, for instance, to provide both Internet connectivity and Voice-over-Internet-Protocol telephone service to the customer premises. A portable cordless telephone handset communicates via wireless communication signals with the telephone base station thereby providing telephone service to the premises. The handset is also capable of transmitting commands to the telephone base station for purposes of providing a user interface for the components of the multimedia terminal device. For example, as a result of a sent command, status or other information can be forwarded to the handset, the modem can be instructed to reboot, a test can be initiated on the multimedia terminal device, or a set up operation can be accomplished. The display screen of the handset can be used to provide the customer with the requested information or results. | 01-06-2011 |
| 20110002326 | Method for dialing from internet extension to conventional extension - A method for dialing from Internet extension to conventional extension is disclosed. A VoIP gateway or an IP auto attendants is used for dialing from Internet extension to conventional extension. The phone number of the Private Branch Exchange and the voice guidance are not needed. The calling number of SIP message is interpreted directly and converted into DTMF (Dual-tone multi-frequency) messages for dialing into a conventional extension. | 01-06-2011 |
| 20110002327 | VOICE SERVICE IN EVOLVED PACKET SYSTEM - Methods and apparatus to manage voice service in evolved packet systems are disclosed. An example method in a user equipment (UE) with a first indicator related to voice services in an Evolved Packet System (EPS) comprises receiving a Non Access Stratum (NAS) protocol response message with a second indicator and responsive to at least one of the first indicator or the second indicator, sending a notification that voice services are not currently able to be provided. | 01-06-2011 |
| 20110002328 | METHOD, SYSTEM, AND DEVICE FOR SETTING UP A CALL USING A GLOBAL REGISTRY - A method, system, and device for establishing a call using a single identifier, which includes receiving contact information relating to the single identifier from an uploading device, the contact information identifying at least one protocol, storing the contact information received from the uploading device, retrieving the stored contact information and transmitting a message including the contact information, in response to a request from a call server for the contact information relating to the single identifier, receiving a request from a first communication device to establish a call to a second communication device associated with the single identifier, requesting the contact information relating to the single identifier, receiving the contact information relating to the single identifier, and establishing a call between the first communication device and the second communication device associated with the single identifier using the at least one protocol. | 01-06-2011 |
| 20110002329 | METHOD, EQUIPMENT AND MOBILE COMMUNICATION SYSTEM FOR REALIZING EXPLICIT CALL TRANSFER - A method, equipment, and a mobile communication system for realizing explicit call transfer are provided. The method for realizing explicit call transfer includes the following steps. A service centralization & continuity application server (SCC AS) receives a call request sent by a second user equipment (UE), and sends the call request to a third UE, in which an instruction for replacing a call between a first UE and the third UE is carried in the call request. A message returned by the third UE according to the call request is received, and the third UE is controlled to establish a connection with the second UE and to break a connection with the first UE. The third UE is an IP multimedia subsystem centralized service user equipment (ICS UE). | 01-06-2011 |
| 20110002330 | SYSTEMS AND METHODS OF DECIDING HOW TO ROUTE CALLS OVER A VOICE OVER INTERNET PROTOCOL TELEPHONE CALL ROUTING SYSTEM - A system and method of monitoring Voice over the Internet Protocol (VoIP) and facsimile over Internet Protocol (FoIP) calling over the Internet includes compiling information about each call after the call is terminated. By compiling information about each of the calls immediately after they are terminated, the system can quickly generate billing reports. The system can also quickly react to developing problems. | 01-06-2011 |
| 20110007732 | Unified Communication System - A unified communication system is disclosed that allows a variety of end point types to participate in a communication event using a common, unified communication system. In some implementations, a calling party interacts with a client application residing on an endpoint to make a communication request to another endpoint. A communication event manager residing in the unified communication system selects a script from a repository of scripts based on the communication event and the capabilities of the endpoints. A communication event execution engine receives a user profile associated with at least one of the endpoints. The user profile can be configured by the user to describe the user's preferences for how the communication should be processed by the unified communication system. | 01-13-2011 |
| 20110007733 | Hierarchical Data Collection Network Supporting Packetized Voice Communications Among Wireless Terminals And Telephones - A packet-based, hierarchical communication system, arranged in a spanning tree configuration, is described in which wired and wireless communication networks exhibiting substantially different characteristics are employed in an overall scheme to link portable or mobile computing devices. The network accommodates real time voice transmission both through dedicated, scheduled bandwidth and through a packet-based routing within the confines and constraints of a data network. Conversion and call processing circuitry is also disclosed which enables access devices and personal computers to adapt voice information between analog voice stream and digital voice packet formats as proves necessary. Routing pathways include wireless spanning tree networks, wide area networks, telephone switching networks, internet, etc., in a manner virtually transparent to the user. A voice session and associate call setup simulates that of conventional telephone switching network, providing well-understood functionality common to any mobile, remote or stationary terminal, phone, computer, etc. | 01-13-2011 |
| 20110007734 | ARBITER CIRCUIT AND METHOD OF CARRYING OUT ARBITRATION - A method of carrying out arbitration in a packet exchanger including an input buffer temporarily storing a packet having arrived at an input port, and a packet switch which switches a packet between a specific input port and a specific output port, includes the steps of (a) concurrently carrying out a first plurality of sequences in each of the sequences basic processes for at least one of the input buffer and the output port are carried out in a predetermined order, and (b) making an allowance in each of the sequences for packets to be output through output through output ports at different times from one another. | 01-13-2011 |
| 20110007735 | CALL SETUP FROM A CIRCUIT SWITCHED NETWORK TO A TERMINAL RESIDING WITHIN A PACKET SWITCHED NETWORK - A user with a terminal residing in a Circuit Switched (CS) telecommunication network calls a party having a terminal residing at a Packet Switched (PS) telecommunication network, the CS and PS networks connected to each other by gateway entity. The party to be called at the PS network is addressed by means of a Session Initiation Protocol Universal Resource Identifier (SIP-URI). The call setup is performed in a two step process. In a first step, the terminal sends a the SIP-URI in a message together with the address of this terminal to a network entity which stores said message. In a second step, the terminal calls the network entity, wherein the network entity selects the stored SIP-URI and resolves the SIP-URI into an address of the terminal at the PS network and instructs the gateway entity to connect the calling terminal to the terminal. | 01-13-2011 |
| 20110007736 | INTERNET PROTOCOL TRUNK GROUPS - A system includes a core routing engine operable to receive a call setup request and identify one or more IP trunk groups through which the call setup request can be routed, select one of the one or more identified IP trunk groups and route the call setup request to an internal IP address associated with the selected IP trunk group. The system may further include an IP edge node associated with the internal IP address, the IP edge node in the backbone network and operable to receive the call setup request and route the call setup request to one of a plurality of IP addresses associated with a plurality of carrier edge nodes in the carrier network. | 01-13-2011 |
| 20110013618 | Method Of Processing Sequential Information In Packets Streamed Over A Network - A method of processing sequential information in near real-time data packets streamed over a network includes providing a process running according to a process clock. The process buffers and decodes the streamed data packets. The speed of the process clock is dynamically controlled in accordance with a receipt time value of a data packet. The speed of the process clock is run faster or slower than a system clock. | 01-20-2011 |
| 20110013619 | Universal Service Transport Transitional Encoding - An apparatus comprising a switch fabric coupled to a plurality of interfaces and configured to switch a plurality of universal service transport (UST) multiplexing (USTM) data streams between the interfaces, wherein the USTM data streams comprise packet-switched traffic, circuit-switched traffic, and transitional signaling that indicates a change of state between the packet-switched traffic and the circuit-switched traffic, wherein the transitional signaling does not indicate the state in every octet of the USTM data streams. Also disclosed is a network component comprising at least one processor coupled to a memory and configured to receive a data that corresponds to a flow, identify the flow using a flow map, determine whether there is a change in a state of the flow, send transitional signaling on a USTM data stream that indicates the state of the flow if the state of flow has changed, and send the data on the USTM data stream. | 01-20-2011 |
| 20110013620 | System for Accessing End-to-End Broadband Network Via Network Access Server Platform - A system is described for providing personalized network access and services in a distributed end-to-end broadband transport network having a telecommunication device used by a user having a unique personal identifier, a premises-based broadband access agent (BAA), the BAA connected to and in communication with the telecommunication device, a switch specific to an underlying transport medium, the switch connected to and in communication with the distributed end-to-end broadband transport network, a network access server platform (NASP), the NASP connected to and in communication with the BAA and the switch, the NASP provides personalized network access and services on demand and a call connection agent (CCA) to complete a call placed by the user to a terminating user. | 01-20-2011 |
| 20110013621 | Upstream Data Rate Estimation - In one embodiment, a device includes: a transceiver operable to transmit packets to and receive packets from a modem; and a logic engine configured to transmit first packets at a rate through an upstream path for a modem to an Internet node such that no throttling is triggered in the modem, the logic engine being further configured to transmit second packets through the upstream path for the modem to the Internet node at a rate sufficient to trigger throttling in the modem if the modem implements throttling, the logic engine being further configured to compare an average transmission time for first packets to an average transmission time for the second packets to determine whether the modem implements throttling. | 01-20-2011 |
| 20110013622 | VOICE COMMUNICATION SYSTEM AND VOICE COMMUNICATION METHOD - A voice communication system, which is connected to a LAN to which communication terminals are connected and to a public network to which telephones are connected, is provided with a communication server between the LAN and public network having different protocols from each other. The communication server enables a voice communication between a telephone on the public network and a communication terminal connected to the LAN by performing processing similar to that for a voice communication between two communication terminals connected to the LAN. The communication server determines whether an address of the other party inputted by a user is a communication terminal address or a telephone number, and transmits a voice communication request to a communication terminal of the other party when the address is a communication terminal address. When the address is a telephone number, the user acquires the communication terminal address of the communication server, and transmits a voice communication request to the communication server. Thereafter, the voice communication processing is performed through the communication server. | 01-20-2011 |
| 20110019660 | Plug and Play Provisioning of Voice Over IP Network Devices - Techniques are provided for sending from a client in a network device a request message configured to request configuration parameters to allow the network device to operate as a source or destination node for packet switched network telephony activity. In response to receiving the request message, sending the configuration parameters from a server configured to retrieve the configuration parameters from a call provisioning server. The configuration parameters are received at the client and passed to a call agent in the network device in order to configure the network device to operate as a source or destination node for packet switched network telephony activity. | 01-27-2011 |
| 20110019661 | METHOD AND APPARATUS RESOLVING ENUM DATA COLLISIONS - A system that incorporates teachings of the present disclosure may include, for example, a telephone Number Mapping (ENUM) system having a subsystem to monitor one or more operations of the ENUM system, determine if ENUM data packets that are received are one of provisioning packets or query packets, send the query packets to a Virtual Internet Protocol (VIP) address of an ENUM domain name system (DNS) server when the ENUM data packets are query packets, send the provisioning packets to a VIP address of an ENUM Lightweight Directory Access Protocol (LDAP) server when the ENUM data packets are provisioning packets, and cause the subsystem to wait and send traffic to one LDAP server at a time after determining if the ENUM data packets are one of provisioning packets or query packets. Other embodiments are disclosed. | 01-27-2011 |
| 20110019662 | METHOD FOR DOWNLOADING AND USING A COMMUNICATION APPLICATION THROUGH A WEB BROWSER - A method of enabling communication over a network by maintaining a server on a network and receiving a request at the server from a user of a communication device. In response to the request, a communication application is downloading over the network to the communication device. The communication application enabling the user to participate in a conversation on the communication device in either (i) a real-time mode or (ii) a time-shifted mode and (iii) to seamlessly transition the conversation between the two modes (i) and (ii). | 01-27-2011 |
| 20110019663 | METHOD, SYSTEM AND GATEWAY FOR SUPPLYING INTELLIGENT SERVICE - An intelligent service system and method are provided. The method includes: receiving a calling information transmitted by a switching device through an Internet-protocol-based call control protocol and which carries the identification information of the service control function requested by the call as well as the identification information of the call; parsing the received calling information, and after the identification information of the service control function and the identification information of this calling are obtained, initiating an assist request to the service control device; receiving the resource operation instruction, and supplying service resource service accordingly; said instruction is sent out by the service control device after said device has received the assist request and according to the service requirement. The media stream leading to the media gateway is established through the media stream transport protocol based on the Internet protocol. The method simplifies the management and maintenance of the intelligent service network. | 01-27-2011 |
| 20110019664 | Emergency alert for voice over internet protocol (VoIP) - A voice over Internet Protocol (VoIP) positioning center (VPC) is implemented in configuration with support from a text-to-voice module, emergency routing database, and VoIP switching points (VSPs) to allow a public safety access point (PSAP) or other emergency center to effectively communicate the nature of an emergency alert notification and the area of notification to the VoIP positioning center (VPC). The inventive VPC in turn determines which phones (including wireless and/or VoIP phones) are currently in the area for notification, and reliably and quickly issues the required warning to all affected wireless and VoIP phones. | 01-27-2011 |
| 20110019665 | Method of Terminating a Call and Voice-Over-IP Terminal - A method of terminating a call is used by a voice over IP terminal ( | 01-27-2011 |
| 20110026515 | COMMUNICATION NETWORK WITH LINE-AND PACKET-SWITCHING CONTROL - The invention relates to a common communication network with line- and packet-switching control, with telecommunication services such as call-forwarding being carried out by mean of a link between a control device and a communication network. The invention is characterized in that at least partially synchronized control ( | 02-03-2011 |
| 20110026516 | SYSTEM AND METHOD FOR REGISTERING AN IP TELEPHONE - A system and method for establishing connection of an IP telephone to a network may include, in response to receiving a registration request from an IP telephone, generating a command to cause network access devices to ping the IP telephone. The command may be communicated to the network access devices. Ping information may be received in response to the network access devices pinging the IP telephone. A network access device may be selected that has the highest quality network access path to the IP telephone. In response to selecting the network access device that has the highest quality network access path, a network address of the selected network access device may be communicated to a network device to enable the IP telephone to communicate with the selected network access device. Credentials may be communicated to the IP telephone to register with the selected network access device. | 02-03-2011 |
| 20110026517 | Session Initiation Protocol (SIP) - An adaptation proxy, a computer system, a computer-implemented method, and a computer program product for enabling presence and remote call control services between client devices served by different SIP servers. In one aspect, the adaptation proxy integrable into a computer system for enabling presence and remote call control services between client devices served by different SIP servers may comprise an SIP adaptor operable to transform and to transport SIP messages between the client devices served by the different SIP servers; a CSTA gateway operable to convert a CSTA event supported by a second SIP server of the SIP servers into a format supported by a first SIP server of the SIP servers, wherein the CSTA event independently operates over the SIP messages to communicate remote control commands; and a presence integrator operable to notify a change to a call state of a first client device from the client devices served by the first SIP server to the second SIP server after having performed a mapping between the changed call state and a corresponding presence state of a second client device from the client devices served by the second SIP server so as to integrate presence information of the first client device and the second client device. | 02-03-2011 |
| 20110026518 | METHOD, DEVICE, AND SYSTEM FOR TRANSFERRING SERVICE CONTROL SIGNALLING PATH - A method, device, and system for transferring a Service Control Signalling Path are provided. The method for transferring a Service Control Signalling Path includes: establishing a connection with an opposite end by a User Equipment (UE), where the UE uses a Circuit Switched (CS) bearer in a CS network and a Service Control Signalling Path in a first Packet Switched (PS) network; sending a transfer request via a second PS network, to instruct a network side to transfer the Service Control Signalling Path according to the transfer request. Thus, the UE can replace a current Gm reference point with a Gm reference point of a new and available PS network when the PS network where the current Gm reference point is located is unavailable, so as to ensure smooth data transmission. | 02-03-2011 |
| 20110032927 | METHODS, SYSTEMS, AND COMPUTER READABLE MEDIA FOR INTELLIGENT OPTIMIZATION OF DIGITAL SIGNAL PROCESSOR (DSP) RESOURCE UTILIZATION IN A MEDIA GATEWAY - The subject matter described herein includes methods, systems, and computer readable media for intelligent optimization of digital signal processor (DSP) resource utilization in a media gateway. In one method, it is determined in a media gateway whether predetermined conditions exist for DSP-less IP-IP switching for a call. In response to determining that the predetermined conditions exist, DSP-less IP-IP switching is implemented for the call in the media gateway. After implementing the DSP-less IP-IP switching for the call, it is determined whether a predetermined event occurs that requires insertion of DSP resources during the call. In response to determining that the predetermined event occurs, the DSP resources are inserted into the call during the call. | 02-10-2011 |
| 20110032928 | SYSTEMS AND METHODS FOR INITIATING ANNOUNCEMENTS IN A SIP TELECOMMUNICATIONS NETWORK - Network servers in a session initiation protocol (SIP) telecommunication network implement playback of announcements to end-users by embedding programming scripts defining how the announcements are to be played in a SIP message. In particular, the scripts may define the sequence in which a series of announcements are to be played, duration information relating to a playback length of the announcements, and repetition information defining how many times an announcement is to be repeated. By including a script in a single message, announcement instructions may be efficiently communicated in the network. | 02-10-2011 |
| 20110032929 | AUDIO/VIDEO COMMUNICATION SYSTEM - An audio/video communication system is provided which includes: a web server providing a user system with a phone icon or button indicating a call receiver and transmitting a phone identifier LN for identifying the receiver allocated to the phone button when a user clicks the icon or button; and a gateway module performing a call setup in response to a data connection request for the audio/video communication from the user system, specifying the user identifier DN for identifying the user system from another user system, transmitting the phone identifier LN to the IP-based telephone exchanger, and relaying a communication between a phone connected to the IP-based telephone exchanger and the user system to progress the audio/video communication. | 02-10-2011 |
| 20110032930 | Network Entity Selection - There are disclosed measures of network entity selection, for example including furnishing an identity of a network entity being pre-selected by a first network apparatus, and providing verification information for said pre-selected network entity identity, enabling to verify whether the pre-selected network entity identity is applicable for network entity selection at a second network apparatus. | 02-10-2011 |
| 20110038362 | Controlling multi-party communications - A first user terminal, host terminal, method and program. The first terminal comprises: a transceiver for communicating with a plurality of other user terminals over a communication network; and communications processing apparatus, coupled to the transceiver, and arranged to participate in a call with a selected number of the other user terminals via the transceiver and communication network, the call including transmission of a voice signal from the first user terminal. The communications processing apparatus is operable in a mode whereby it temporarily discontinues transmission of the voice signal in response to detecting less than a predetermined level of activity on said voice signal, and the communications processing apparatus is further configured to selectively enable that mode in dependence on the selected number of other user terminals in the call. | 02-17-2011 |
| 20110038363 | METHOD AND ARRANGEMENT FOR PROVIDING VOIP COMMUNICATION - The invention relates to a method for providing communication in a VoIP communication network having a multiplicity of network nodes, in which a) at least one first subscriber terminal in the VoIP communication network stores a first item of information containing at least one VoIP connection property desired by the user of the first subscriber terminal in at least one first network node of the VoIP communication network, wherein b) when a second subscriber terminal wishes to connect to the first subscriber terminal, b1) the second subscriber terminal requests the first item of information from the first network node, b2) the second subscriber terminal forms at least one data element that describes the connection on the basis of at least the first item of information, b3) the second subscriber terminal transmits the data element that describes the connection to a functional element which is assigned to the network and switches through direct connections between communication partners, and wherein c) the functional element evaluates the data element in such a manner that it establishes the connection between the first subscriber terminal and the second subscriber terminal on the basis of at least the first item of information. The invention also relates to an arrangement for carrying out the method. | 02-17-2011 |
| 20110038364 | SYSTEM AND METHOD FOR SWITCHING BETWEEN PHONE SERVICES - The present invention concerns a gateway device and a method at the gateway, the gateway device comprising an interface to a residential phone wiring comprising more than one plugging means for connecting at least one analogue phone, a broadband interface to a network comprising a central office, the central office being adapted to provide a first voice service type to the at least one analog phone, an FXS module for providing a voice over IP service over the broadband interface to the at least one analogue phone when the first voice service type is disabled, unbundling detection means for detecting the presence of the first voice service type, connecting the FXS module to the residential phone wiring when the first voice service type is disabled, and disconnecting the FXS module from the residential phone wiring when the first voice service type is enabled, and a management agent for informing a gateway management server when changing from the first voice service type to the voice over IP service and vice versa, so that the same phone number can be used when using the first voice service type or the voice over IP service. | 02-17-2011 |
| 20110038365 | Systems and methods for voice and data communications including a network drop and insert interface for an external data routing resource - Systems and methods by which voice/data communications may occur are disclosed. In particular, systems and methods are provided with a computing system having a multi-bus structure, including, for example, a TDM bus and a packet bus. An integrated communication system is coupled to a digital telecommunications link, the communication system providing voice and data communications to a plurality of users. At least a first packet bus is coupled to one or more packet-based devices and adapted for transferring packetized data to and from the system. One or more time division multiplex (TDM) buses are coupled to one or more telephony devices. Data routing resources are provided internal to the integrated system. A network interface module couples data to and from a data router external to the integrated system. The data router external to the integrated system is coupled to the first packet bus. Data is routed via the external data router through the network interface module and coupled to data channels of the digital telecommunications link, while voice data is selectively coupled to voice channels of the digital telecommunications link. | 02-17-2011 |
| 20110038366 | SWITCHING DATA STREAMS BETWEEN CORE NETWORKS - The present disclosure is directed to switching data streams between core networks. In some implementations, a method can include identifying a plurality of different RTP streams from a SIP device with at least one stream associated with a supplementary service. A plurality of single media streams for a plurality of different mobile devices in a cellular core network is identified. Dynamically switching connections between each RTP stream in the plurality of different RTP streams and a corresponding single media stream in the plurality of single media streams based, at least in part, on SIP signaling from the SIP device. | 02-17-2011 |
| 20110038367 | AUTOMATED COMMUNICATIONS RESPONSE SYSTEM - In one embodiment, a system provides for end-user control over the automatic recognition of communication situations by detection of unique telecommunication event characteristics and the consequential responses to those situations by invocation of related programmatic responses. The system allows an end user to specify various patterns of telecommunication event characteristics that describe various situational aspects of incoming communications, such as the timing and originator of voice calls, the content of, timing of, and author of chat messages, etc., as well as appropriate sets of programmatic response actions to be performed in response to those communications, such as initiating conference calls, sending chat messages, routing calls to other users, etc. The system monitors incoming communications, matches characteristic patterns to recognize the situations, and then invokes the matching response actions, thereby automating many functions of the communication system that previously would have had to be performed manually. | 02-17-2011 |
| 20110038368 | TELEPHONE COMMUNICATION SYSTEM AND METHOD OVER LOCAL AREA NETWORK WIRING - A device for enabling a local area network wiring structure to simultaneously carry digital data and analog telephone signals on the same transmission medium. It is particularly applicable to a network in star topology, in which remote data units (e.g. personal computers) are each connected to a hub through a cable comprising at least two pairs of conductors, providing a data communication path in each direction. Modules at each end of the cable provide a phantom path for telephony (voice band), signals between a telephone near the data set and a PBX, through both conductor pairs in a phantom circuit arrangement. All such communication paths function simultaneously and without mutual interference. The modules comprise simple and inexpensive passive circuit components. | 02-17-2011 |
| 20110044317 | REAL-TIME VOICE LOGGING OF TELEPHONE CALLS - An office telephone system contains packet switched network and network telephone sets coupled to said packet switched network for transmitting and receiving speech data in addressed packets. A packet switched network interface taps the packet switched network and processes packets received from the packet switched network by identifying first and second packets that contain network voice call data for respective sides of a network telephone calls. The packet switched network interface mixing speech data from the first and second packets into streams while the call proceeds. Each stream comprising a mix of speech data from both sides of a respective one of the network telephone calls. An application program interface defines provides access to the streams to a programmable set of applications. In addition a line interface circuit taps call dedicate telephone lines outside the network and generates further speech data streams from signals from the call dedicated telephone lines. The application program interface defines provides interchangeable types of calls to access streams generated from both sources. | 02-24-2011 |
| 20110044318 | Interoperability of Legacy Alarm System - A base station and system configured to support interfacing digital and analog devices with a legacy alarm system in a manner that allows messages from the legacy alarm system to pre-empt other messaging, if needed, when the messaging takes place through the same gateway as that which is used by the legacy alarm system. | 02-24-2011 |
| 20110044319 | EARLY MEDIA AND FORKING IN 3PCC - A control server initiates a call to a first device. After creating a connection to the device, the control server reverses the direction of the message flow between the device and the control server such that the device becomes the initiator of the call (the caller) and the control server becomes the device that is called (the callee). A connection is also established between the first device, the control server and a second device that is an endpoint for the call. Early media and forking is available to the first device after reversing the direction of the message flow between the first device and the control server and the callee has been contacted. Additionally, information flows between the first device and the second device through the control server as if the first device and the second device were directly connected. | 02-24-2011 |
| 20110044320 | MECHANISM FOR FAST EVALUATION OF POLICIES IN WORK ASSIGNMENT - A work item routing mechanism is provided that is capable of employing a state map which compresses routing decisions and results of comparisons into a single bit. Thus, comparisons and determinations made in connection with work item routing are made prior to the routing mechanism receiving a work item. Once a work item is received, the routing mechanism only has to refer to the bit map to see if it is allowed to route the work item to a particular processing resource and if that resource is the best among all candidate processing resources. All of the work item routing decisions can, therefore, be made very quickly thereby reducing processing delay and wait time. | 02-24-2011 |
| 20110044321 | MIDCALL FALLBACK FOR VOICE OVER INTERNET PROTOCOL (VOIP) CALLS - A method for performing midcall fallback is provided. The method includes assigning a direct inward dialing (DID) number to a first client. The DID number may be selected from a list of direct inward dialing numbers. The method may further include establishing a VoIP phone call between the first client and a second client and sending a DID number representing the first client and receiving a dial sequence identifying a call agent serving the second client. The dial sequence may define a phone number to be dialed to reach the call agent. The method may also include determining that mid-call fallback should be performed, and performing midcall fall back, midcall fallback including establishing a public switched telephone network (PSTN) phone call between the first client and the second client. | 02-24-2011 |
| 20110044322 | Method To Share Phone Line - First method for sharing telephone resources includes a VoIP device connecting to a first device over an IP network, receiving a request from the first device to call a second device with a telephone number, connecting to the second device through a telephone system, and transferring voice signals between the first and the second devices. Second method for sharing telephone resources includes a first VoIP device joining with a group of VoIP devices connected to an IP network to share their telephone resources, receiving from a caller a telephone number to call a device, connecting to a second VoIP device from the group over the IP network, transmitting the telephone number to the second VoIP device so the second VoIP device connects to the device through a telephone system, and transmitting to and receiving from the second VoIP device voice signals between the caller and a recipient at the device. | 02-24-2011 |
| 20110044323 | METHOD AND APPARATUS FOR CONCEALING LOST FRAME - A method for concealing lost frame includes: using history signals before the lost frame that corresponds to a lost MDCT coefficient to generate a first synthesized signal when it is detected that the MDCT coefficient is lost; performing fast IMDCT for the first synthesized signal to obtain an IMDCT coefficient corresponding to a lost MDCT coefficient; and using the IMDCT coefficient corresponding to the lost MDCT coefficient and an IMDCT coefficient adjacent to the IMDCT coefficient corresponding to the lost MDCT coefficient to perform TDAC and obtain signals corresponding to the lost frame. An apparatus for concealing lost frame is also disclosed herein. The method and the apparatus for concealing lost frames in the embodiments of the present invention make full use of the received partial signals to recover high-quality voice signals and improve the QoS. | 02-24-2011 |
| 20110044324 | Method and Apparatus for Voice Communication Based on Instant Messaging System - Embodiments of the present invention provide a method and apparatus for voice communication based on an IM system. The method includes: a) establishing a tone-modified voice communication channel between second IM client and first IM client; b) processing inputted original voice information through tone modification to obtain tone-modified voice; sending the tone-modified voice to the first IM client via the tone-modified voice communication channel. According to embodiments of the present invention, the voice information collected in the IM system is first processed through tone modification, thereby tone-modified voice communication based on the IM system is implemented. | 02-24-2011 |
| 20110044325 | System and Method for Effectuating a SIP Call in a Network Environment Including IMS - In one embodiment, a scheme is disclosed for effectuating a call in a network environment including a circuit-switched (CS) network and an IP multimedia subsystem (IMS) network. Call information associated with a call is sent from a user equipment (UE) device to an application server (AS) node disposed in the IMS network. The call information includes at least one of a call reference number and a called party's URI. When a message is received at the UE device from the AS node, which message includes the call reference number and an IP multimedia routing number (IMRN), the returned call reference number is verified that it remains valid based on a local timer mechanism associated with the UE device. The IMRN is then sent to the application server in order to facilitate a session with respect to the called party. | 02-24-2011 |
| 20110044326 | IDENTIFY A SECURE END-TO-END VOICE CALL - We describe a system embodiment comprising generating a Secure Real-Time Transport Protocol (SRTP) encapsulated packet and including a secure media indicator into the SRTP encapsulated packet. The method further comprises inserting the SRTP encapsulated packet into an SRTP voice stream associated with an active call between a source and a destination endpoint and indicating an end-to-end secure call between the source and destination endpoints responsive to the secure media indicator. | 02-24-2011 |
| 20110044327 | Support for Continuity of Single Radio Voice Call Communications in a Transition to a Circuit Switched Communications Network - The present invention establishes a new protocol that supports the continuity of a single radio voice call onto a circuit switched communications system through the use of a special addressing identifier. This special identifier is called the single radio voice call identifier, and it designates the use of a single radio voice call continuity procedure for the transition to the circuit switched communication system. The applications server receives the single radio identifier and performs the transfer of the single radio voice session without the need for other address or identifier information, and also uses the single radio identifier or a new message type to initiate the correlation of parameters related to service control session establishment in later steps. | 02-24-2011 |
| 20110051712 | INTERNET PROTOCOL MULTIMEDIA SYSTEM (IMS) MOBILE SESSION INITIATION PROTOCOL (SIP) AGENT - A first phone obtains an identifier of a second phone from a phone list, and sends a request for the second phone's Session Initiation Protocol (SIP) type to a remote server. The first phone receives the second phone's SIP type from the remote server, and sends a message to one or more nodes in a network, based on the received second phone's SIP type, for a SIP session between the first phone and the second phone. | 03-03-2011 |
| 20110051713 | FACSIMILE PRIORITIZATION WITHIN INTERNET PROTOCOL CALL NETWORKS - A method and apparatus maintain a facsimile number priority hierarchy within a computer storage medium and process a first facsimile call being transmitted through a computerized call processor. The first facsimile call is made between a first telephone number associated with the computerized call processor and a second telephone number not associated with the computerized call processor. While processing the first facsimile call, the computerized call processor receives an indication of an attempt to connect a second facsimile call between the first telephone number and a third telephone number. The third telephone number is not associated with the computerized call processor. The method and apparatus determine a priority between the second telephone number and the third telephone number based on the facsimile number priority hierarchy. If the second telephone number has a higher priority than the third telephone number, the computerized call processor does not connect the second facsimile call. However, if the third telephone number has a higher priority than the second telephone number, the computerized call processor terminates the first facsimile call and connects the second facsimile call by connecting the third telephone number to the first telephone number. | 03-03-2011 |
| 20110051714 | Apparatuses, Methods and Systems For Tiered Routing Engine - The APPARATUSES, METHODS AND SYSTEMS FOR TIERED ROUTING ENGINE (“TRE”) provides an automatic routing, selecting, processing for calls placed in an international network according to a selected International Tier Level for premium or guaranteed delivery. In one embodiment, a platform initiates international tiered routing information to a gateway based on a pre-set platforms' knowledge of the terminating gateway topology, Automatic Number Identification, and assigned services that requires such transmission. In one embodiment, a user may select a tier to route an international call. In another embodiment, the contextual fields of a communication mechanism define tags and tier levels indicating determining, routing and handling information to be sent to a validated gateway, or routing devices. In one embodiment, contextual tags includes customized domain name and global descriptors of compatible network components, delivery control, trunk group service ID, trunk-related tier level, and other trunk-related service attributes for tiered routing. | 03-03-2011 |
| 20110051715 | METHOD AND SYSTEM FOR PLATFORM-INDEPENDENT VOIP DIAL PLAN DESIGN, VALIDATION, AND DEPLOYMENT - A system and method for designing a dial plan for Voice over Internet Protocol (VoIP) systems includes generating an abstract dial plan design which is platform independent, the dial plan including rules for routing communications in a VoIP network structure. The dial plan is validated through simulations prior to deployment to evaluate the dial plan performance under simulated conditions. The dial plan design is translated to provide compatibility with a deployed network using platform specific configuration adaptors. | 03-03-2011 |
| 20110051716 | TV ACTING AS POTS PHONE SWITCH - A TV receives IP calls and POTS calls. When a POTS call is received the TV passively passes the call to a non-IP phone, and when an IP call is received the TV processes the IP packets as appropriate for the non-IP phone and passes the call to the phone. The non-IP phone can also signal using a special code a desired to place an IP call, with the signals from the phone being rendered into IP packets by the TV. In this way, a non-IP phone may be used to place and receive both POTS calls and IP-based calls. | 03-03-2011 |
| 20110051717 | SYSTEM AND METHOD FOR PROVIDING REDUNDANCY IN A DISTRIBUTED TELECOMMUNICATIONS ARCHITECTURE - A telecommunications platform that provides redundant interfaces to a telecommunications system for multiple IP based telecommunication devices. The telecommunications platform includes a gateway cluster with two or more signaling gateways. Each signaling gateway is assigned a point code for being accessed by devices in the telecommunications system. The gateway cluster is assigned a virtual point code. Any of the IP based telecommunications devices can be accessed by the telecommunications system by routing to the virtual point code through one of the signaling gateways in the gateway cluster. Thus, if one of the signaling gateways is not available, the IP based telecommunications devices can still be accessed through one of the other signaling gateways in the gateway cluster. | 03-03-2011 |
| 20110051718 | METHODS AND APPARATUS FOR DELIVERING AUDIO CONTENT TO A CALLER PLACED ON HOLD - Several methods and systems for providing audio content to callers placed on hold are described. In some on-hold phonecasting methods, a two-way telecommunications link is established between a caller and a call terminus. The caller or the call terminus is temporarily isolated from the link. The audio content is provided via the link while the caller or call terminus is isolated to indicate that the link is still in place. At least a portion of the audio content is specified by a really simple syndication feed. The audio content may include one or more podcasts publicly available via the Internet. The audio content may be generated according to configuration information and by concatenating an audio advertisement or public service essage with the portion of the audio content. The method may also include periodically checking the RSS feed for updates to the audio content, and downloading updated audio content. | 03-03-2011 |
| 20110051719 | PROVIDING A CALL SERVICE IN A COMMUNICATION NETWORK - Methods and systems for providing company call service in wireless and wired integrated network are provided. For example, a call between an employee's wireless device and a client's device can be connected while indicating the employee's wired telephone number as a caller's telephone number. When an employee is receiving a call, one example is to call an employee's wired device first, and if there is no response, employee's wireless device may be called subsequently. In another example, employee's wired and wireless device may be called simultaneously. | 03-03-2011 |
| 20110051720 | SIP TELEPHONE SET, AND FILE TRANSFER SYSTEM, FILE TRANSFER METHOD AND FILE TRANSFER PROGRAM THEREOF - Provided is service for transmitting and receiving a file between a calling device and a call receiving device without depending on a capacity of a proxy server. | 03-03-2011 |
| 20110058544 | METHODS, SYSTEMS, AND COMPUTER READABLE MEDIA FOR VERIFYING THE AVAILABILITY OF AN INTERNET PROTOCOL (IP) MEDIA ROUTER DURING A CALL SETUP - Methods, systems, and computer readable media for verifying the availability of an IP media router during a call setup are described. In one embodiment, the method comprises receiving, from a first endpoint device, a call setup signaling message requesting to establish a call session with a second endpoint device. The method also includes selecting a first media router to establish a first call leg of the call session, performing a route query and MAC address resolution to determine if the first media router is available, and if the first media router is determined to be available, creating a first redirect stream to communicate media packets received from the second endpoint device to the first endpoint device via the first call leg. | 03-10-2011 |
| 20110064073 | METHODS, APPARATUS AND ARTICLES OF MANUFACTURE TO PROVIDE UNIFORM RESOURCE IDENTIFIER PORTABILITY - Example methods, apparatus and articles of manufacture to provide uniform resource identifier (URI) portability for communication networks are disclosed. A disclosed example method includes receiving a first communication session initiation message identifying a called party, identifying a URI associated with the called party, querying a global URI database based on the URI to identify a domain name associated with a service provider network based on the URI, and sending a second communication session initiation message including the URI to the service provider network via the domain name. | 03-17-2011 |
| 20110064074 | Presence information - A method, program and user node for use in a communication system implemented over a network comprising a plurality of user nodes, each being associated with a respective presence status indicating an availability of the user node for communication within the communication system. The method comprises, at each of a first one or more of the user nodes: maintaining a contact list specifying a selection of contacts from the plurality of user nodes; associating a presence update priority level with each of the contacts, the presence update priority level relating to an estimated likelihood of communication between the first user node and the respective contact; and transmitting a presence message to each of a plurality of the contacts in dependence on the respective presence update priority level, each of the presence messages comprising at least one of: a request for the presence status of the contact, and a notification of the presence status of the first user node. | 03-17-2011 |
| 20110064075 | METHODS AND SYSTEMS FOR COMMUNICATING SIGNALING SYSTEM 7 (SS7) USER PART MESSAGES AMONG SS7 SIGNALING POINTS (SPs) AND INTERNET PROTOCOL (IP) NODES USING SIGNAL TRANSFER POINTS (STPs) - Methods and systems for transmitting user part messages between signaling system seven (SS7) signaling points over an internet protocol (IP) network include receiving, at a signal transfer point, a first SS7 user part message. The first SS7 user part message can be received from a first SS7 signaling point, such as a service switching point (SSP). The first SS7 signaling point is encapsulated in a first IP packet. The first IP packet is transmitted to a second SS7 signaling point over an IP network. | 03-17-2011 |
| 20110069699 | Method for Telephony Client Synchronization in Telephone Virtualization - A method is provided for the use of a signaling protocol stack by telephony applications which run on different system software images. When a telecommunications session is conducted by a first telephony application, the first telephony application typically controls the state of the telecommunications session through a signaling protocol stack executing on the same system software image as the first telephony application. When control over the telecommunications session is passed from the first telephony application to a second telephony application, the second telephony applications begins controlling the state of the telecommunications session through the same signaling protocol stack by using remote procedure calls. | 03-24-2011 |
| 20110069700 | SYSTEM FOR AND METHOD OF INFORMATION ENCODING - A system for and method of information encoding is presented. The system and method include encoding information within other information of a protocol, and then decoding the information and performing actions based on the decoded information. | 03-24-2011 |
| 20110069701 | GATEWAY AND METHOD FOR PROCESSING PACKETS UTILIZED THEREBY - A gateway includes a plurality of line cards and a management board. One of the plurality of line cards connected to one user terminal transmits an Internet control message (ICM) packet with off-hook information of the user terminal. The management board receives the ICM packet with the off-hook information from the line card, and transmits a call request packet to the media gateway controller according to an Internet protocol (IP) address of the management board. The management board further receives a call response packet including a dial tone from the media gateway controller, and transmits an ICM packet with the dial tone to the line card connected to the user terminal. The line card connected to the user terminal further transmits the dial tone to the user terminal. Thus the signaling connection between the gateway and the media gateway controller is established. | 03-24-2011 |
| 20110069702 | BRANDED VOIP SERVICE PORTAL - A method is provided for a Voice over IP (VoIP) Operator to enable another entity, the call branding company, to offer a phone service of its own brand using that VoIP service company's physical service infrastructure which includes, for example, the VoIP client application, the VoIP network elements, and provisioning, billing and ordering systems, the method including the steps of receiving an incoming VoIP call over a network from a caller's VoIP client, matching one or more parameters of the incoming call against one or more call branding activation triggers; and, in the event of a match, applying call branding to the VoIP call as specified by a call branding configuration profile associated with the matched call branding activation trigger. Additional methods of applying call branding provided include the call branding company setting up a call branding configuration profile with the VoIP service provider, the call branding configuration profile including one or more call branding activation triggers, advertisements, VoIP soft-phone client skins, VoIP service options, sponsorship details and call redirection/forwarding rules. | 03-24-2011 |
| 20110075653 | SYSTEMS, METHODS, AND COMPUTER PROGRAM PRODUCTS FOR PROVIDING A MANUAL RING-DOWN COMMUNICATION LINE USING SESSION INITIATION PROTOCOL - Systems, methods, and computer program products are provided for manual ring-down communication using Session Initiation Protocol (SIP). A first SIP user agent transmits a message to a second SIP user agent over an Internet Protocol (IP) network to establish a SIP session. The first SIP user agent determines that a signal key associated with a first communication device has been selected and transmits, to the second SIP user agent over the IP network, a start event message to cause a second communication device to activate an alert. The first SIP user agent determines that the signal key has been released and transmits over the IP network an end event message to deactivate the alert. The first SIP user agent transmits, to the second SIP user agent over the IP network, one or more subsequent INVITE messages at a predetermined repetition rate to refresh the SIP session. | 03-31-2011 |
| 20110075654 | Method and System for Implementing Redundancy at Signaling Gateway Using Dynamic SIGTRAN Architecture - Described are a method, a computer program product and apparatus for implementing signaling gateway redundancy. A first SIGTRAN protocol application server process maintenance message is received, at a first signaling gateway, from a first application server process. Connection control information associated with one or more connections to the first signaling gateway is updated based on the first SIGTRAN protocol application server process maintenance message. A second SIGTRAN protocol application server process maintenance message is transmitted, from the first signaling gateway, to a second signaling gateway. The second SIGTRAN protocol application server process maintenance message is based on the first SIGTRAN protocol application server process maintenance message. The second signaling gateway is mated with the first signaling gateway. | 03-31-2011 |
| 20110075655 | METHOD TO OPTIMIZE CALL ESTABLISHMENT IN MOBILE SATELLITE COMMUNICATION SYSTEMS - Call placement to or from satellite UEs is optimized by reducing IMS message exchanges, the originating party has control over QoS parameters; a HPA subscription service is made available, and calls to a terminating satellite UE that is shielded from satellite coverage are completed by selectively employing HPA pages. For a call request without preconditions, an IMS node associated with an originating UE uses the NRSCPA on Offer instead of using the standard terminating node initiated NRSCPA on Answer. An IMS node associated with a terminating UE checks for HPA subscription by the user. If subscribed, the terminating INVITE request is for a “Conversational” or “Interactive” service, and the terminating UE is in PMM_IDLE state, the satellite RAN pages the terminating UE using HPA. | 03-31-2011 |
| 20110075656 | CIRCUIT ARRANGEMENT, NETWORK-ON-CHIP AND METHOD FOR TRANSMITTING INFORMATION - A circuit arrangement, network-on-chip, and a method for transmitting information are disclosed. In one embodiment, an electrical circuit is provided comprising a plurality of circuit blocks comprising a first circuit block, a second circuit block, and a third circuit block, and a connection structure coupled to the plurality of circuit blocks, wherein the first circuit block is configured to send a request comprising information corresponding to the request and an address onto the connection structure, wherein the second circuit block is configured to initiate a transmission onto the connection structure in response to receiving the request, and wherein the third circuit block is configured to receive the transmission and wherein the address is assigned to the third circuit block. | 03-31-2011 |
| 20110075657 | SYSTEM AND METHOD OF PROVIDING MULTIMEDIA COMMUNICATION SERVICES - In a particular embodiment, a method of providing multimedia communication services includes receiving, at an intelligent service switch (ISS) of an integrated wireline-wireless (IWW) network, a service request from a wireline communication device. The method includes receiving contextual information associated with the service request, the contextual information including a time of the service request, a range of times of the service request, a day of the service request, a range of days of the service request, a date of the service request, a range of dates of the service request, a location of the wireline communication device, a type of the wireline communication device, or a combination thereof. At least one multimedia communication service is provided to the wireline communication device. The at least one multimedia communication service includes audio entertainment content, news content, weather content, traffic content, securities market content, sports content, financial content, local business location content, local event schedule content, network address book content, calendar content, appointment content, map content, call log content, or a combination thereof. | 03-31-2011 |
| 20110075658 | HANDLING OF TERMINATING CALLS FOR A SHARED PUBLIC USER IDENTITY IN AN IP MULTIMEDIA SUBSYSTEM - A single IMPI is determined, allowing the progress of a terminating call that addresses a given IMPU shared by more than one IMPI of an IMS subscription. A number of policies are applied per IMPI basis on how to progress the terminating call. A HSS is provided where the policies are configured and, a method is disclosed including IMPU with a number of policies. Additionally, the method may also include a step of configuring at the HSS the more than one IMPI with a priority indication usable to set the order in which the more than one IMPI are checked to determine at least one for which the policies allow to progress the terminating call. | 03-31-2011 |
| 20110080904 | Subscriber Line Interface Circuitry with POTS Detection - A method of controlling a subscriber line interface circuit (SLIC) includes performing a plain old telephone services (POTS) detect at a customer premises using a customer premises SLIC. Injection of POTS services by the customer premises SLIC is disabled, if POTS is detected. | 04-07-2011 |
| 20110080905 | METHOD AND INTERNET PROTOCOL SHORT MESSAGE GATEWAY (IP-SM-GW) FOR PROVIDING AN INTERWORKING SERVICE BETWEEN CONVERGED IP MESSAGING (CPM) AND SHORT MESSAGE SERVICE (SMS) - A method and an IP-SM-GW for providing an interworking service between CPM and SMS are provided. The method comprises the steps of receiving a chat session invitation, in a IP-SM-GW, the chat session invitation originating from a CPM UE and being sent toward an SMS enabled UE. Assigning an identifier with the chat session in the IP-SM-GW and sending an invitation acknowledgement from the IP-SM-GW to the CPM UE. Receiving a message containing data, within the chat session, in the IP-SM-GW, from the CPM UE and being sent to the SMS enabled UE, formatting the message into an SMS message, wherein the identifier assigned to the chat session is inserted as a sender of the SMS message to ensure that an SMS response is sent back to the IP-SM-GW and forwarding the formatted SMS message to the SMS enabled UE. | 04-07-2011 |
| 20110080906 | METHOD FOR COLLECT CALL SERVICE BASED ON VOIP TECHNOLOGY AND SYSTEM THEREOF - One embodiment of the present invention provides a collect call method and system thereof, more particularly, in order to charge the called party with a uniform toll for collect call, which is determined by only the type and location of called party terminal. In one embodiment, the collect call method, system and a counsel service providing method use a free VoIP network for part of the voice call link and a charge PSTN network for the rest of the voice call link. In one embodiment, if the first link corresponding to the collect call request is established, the collect call switch calls the called party terminal to establish the second link, and billing on the second link is initiated. | 04-07-2011 |
| 20110085541 | LOCAL ROUTING MANAGEMENT IN A TELECOMMUNICATIONS NETWORK - An embodiment of a method includes determining a customer service plan identifier from information associated with a received call, determining a route plan associated with the identified customer service plan, and routing the call on a trunk group identified in the determined route plan. The method may further include determining a jurisdiction of the call based on a dialed number identified in the call, determining a local routing number (LRN) associated with the call, and using a portion of the LRN to determine the trunk group. An embodiment of a system includes a switch operable to select a route for routing a call received on an ingress trunk associated with a customer that subscribes to a service plan, wherein the switch is further operable to select the route based on the service plan subscribed to by the customer associated with the ingress trunk. | 04-14-2011 |
| 20110090898 | Methods and Apparatus for Enabling Media Functionality in a Content-Based Network - Methods and apparatus for providing unified access to interactive media applications and services in a network. In one embodiment, the network comprises a content-based network such as a cable television or satellite network, and the applications are disposed at the network headend. A servlet is provided to facilitate communication between the applications and client devices. The servlet acts as a proxy for applications utilizing a different content format than the client devices. The applications obtain data from e.g., an internet host server via a gateway device. The client application(s) may comprise Enhanced TV Binary Interchange Format (EBIF) pages, and are configured so as to permit use via a common interface (e.g., the user's set top box and television display). These client applications enable a user to, for example, search the internet for data relating to displayed content, post and navigate micro-blogs, instant messaging or SMS, making telephone calls (e.g., VoIP), address/contact management, or provide the user with additional information about a product or service. An application providing internet content to the client device is also provided. | 04-21-2011 |
| 20110090899 | Multimedia Routing System for Securing Third Party Participation in Call Consultation or Call Transfer of a Call in Progress - A multimedia router has code executable on the router from storage on a machine readable medium coupled to the router, the code providing routing functions, and a routing point identified in the router code for establishing at least one non-voice communications session between two or more communications appliances enabled for non-voice communications. During a voice call established between a calling party and one of the two or more communications appliances, the routing point is invoked from the called communications appliance by issuance of a non-voice routing request to establish at least one non-voice communications session between the called communications appliance and another of the two or more communications appliances. | 04-21-2011 |
| 20110090900 | Controlling registration floods in VoIP networks via DNS - A mechanism controls global synchronization, or registration floods, that may result when a large number of endpoints in a Voice over Internet Protocol (VoIP) network such as an Internet Protocol Multimedia Subsystem (IMS) come online simultaneously after a catastrophic failure. The mechanism allows the Domain Name System (DNS) infrastructure to efficiently control the overload condition by registering user end points with backup border elements, and by staggering and by randomizing the time-to-live (TTL) parameter in registrations with backup border elements. | 04-21-2011 |
| 20110090901 | DETERMINATION OF PERSONA INFORMATION AVAILABILITY AND DELIVERY ON PEER-TO-PEER NETWORKS - A method of operating a communication system to establish communication sessions between an origination network and a peer-to-peer network comprises receiving session signaling to establish a session between an origination device in the origination network and a destination node in the peer-to-peer network, wherein the session signaling includes a participant identifier associated with the origination device. The method further comprises processing the participant identifier to determine if persona information that identifies an originating participant and an entity associated with the originating participant is available for display by a destination device registered as the destination node on the peer-to-peer network and, if the persona information is available, transferring the persona information for delivery to and display by the destination device to a destination participant. The method further comprises establishing the session over the origination network and the peer-to-peer network and exchanging user communications for the session between the origination device and the destination device. | 04-21-2011 |
| 20110090902 | System and method for providing quality of service considering priorities of terminals in a communication system - Quality of Service (QoS) is provided based on a priority of a terminal in a communication system. A communication server receives Session Description Protocol (SDP) information and priority information of each of first and second terminals, and transmits, to a Policy Decision Function block (PDF), the SDP information of each of first and second terminals and priority information corresponding to a highest priority in the priority information of the first and second terminals. The PDF performs authentication based on QoS profile information of the first and second terminals acquired from a service profile server upon request for SDP information of each of the terminals, generates a QoS decision value based on the authentication results, and reserves resources that the first terminal will use to perform a communication service with the second terminal, using the QoS decision value. The PDF upgrades the QoS decision value based on the highest-priority information. | 04-21-2011 |
| 20110090903 | PROVIDING LOCATION INFORMATION IN AN IP MULTIMEDIA SUBSYSTEM NETWORK - A method and apparatus for providing location information to a CSCF in an IMS network. An S-CSCF registers a first contact associated with an IMPU, and receives location information associated with the first contact. A second contact associated with the same IMPU, and also with a mobile access, is then registered at the S-CSCF. The S-CSCF receives location information associated with the second contact. | 04-21-2011 |
| 20110090904 | METHOD AND NETWORK ELEMENT FOR IMPLEMENTING A CUSTOMIZED VIDEO SERVICE IN IMS NETWORKS - The present invention proposes a method for implementing a customized video service in IMS networks and a network element for controlling sessions between terminals, wherein a first terminal is calling a second terminal under the control of a first network element, and the second terminal has subscribed a customized video service provided by a second network element. The method comprises: the first network element transmits to the second network element a message of requesting for playing video to the first terminal which includes media information about the first terminal, after having known that the second terminal has the customized video service; the second network element transmits to the first network element an acknowledgement with information on the video to be played, if it determines that the first terminal supports the format of video to be played based on the media information about the first terminal; the first network element transmits to the second network element a message with information on the customized video service, and transmits to the first terminal a reply with the media information of video to be played after having received from the second network element a response with the media information of video to be played; and a media path is established between the first terminal and the second network element, thereby the video customized by the second terminal being played to the first terminal, and the call request sent by the first terminal while calling the second terminal is forwarded to the second terminal by the first network element. | 04-21-2011 |
| 20110096769 | Method and System for Providing an Emergency Location Service - Provided are a method and a system for providing an emergency location service using an IMS core. In the method, when the IMS core receives an emergency call initiating request message from a user equipment, the IMS core transmits a location service request message requesting for retrieving a location of the user equipment to a location retrieval subsystem in response to the emergency call initiating request message. Then, when the IMS core receives current location information of the user equipment, which is acquired through an access to the user equipment by the location retrieval subsystem having received the location service request message, from the location retrieval subsystem, the IMS core selects an emergency center on the basis of the current location information and transmits the emergency call initiating request message including the current location information to the selected emergency center. Then, an emergency call is established between the user equipment and the emergency center. | 04-28-2011 |
| 20110096770 | METHOD AND APPARATUS FOR PROVIDING CHANNEL SHARING AMONG WHITE SPACE NETWORKS - A method and an apparatus for providing channel sharing are disclosed. For example, the method receives a request for a white space channel assignment, and identifies one or more white space channels in accordance with the request. The method sends a response to the request comprising a white space channel assignment, wherein the white space channel assignment assigns one of the identified one or more white space channels. | 04-28-2011 |
| 20110096771 | VOICE OVER INTERNET PROTOCOL (VOIP) SYSTEMS, METHODS, NETWORK ELEMENTS AND APPLICATIONS - In accordance with at least one embodiment of the invention, methodologies and mechanisms are provided that enable methods, systems and software for supporting or implementing functionality to intercept a phone call and/or data transmission in a cellular network and direct it to at least one receivers' VoIP account if the account is active and provides VoIP connectivity. | 04-28-2011 |
| 20110096772 | CALLING PARTY NAME PROVISIONING - A system may receive a telephone call request for a Voice over Internet Protocol (VoIP) user. The telephone call request omits a name of a calling party. The system may further determine if the VoIP user has a calling party name feature enabled and obtaining, when the VoIP user has a calling party name feature enabled, the name of the calling party from a Public Switched Telephone Network (PSTN) based repository of calling party names. | 04-28-2011 |
| 20110096773 | DIRECTORY NUMBER MOBILITY UTILIZING DYNAMIC NETWORK DISTRIBUTED DIAL-PEER UPDATES - Methods, logic, apparatus, and systems are provided to support cross cluster directory number (DN) extension mobility (EM) using dynamic network distributed dial-peer updates in a communication networks, which includes a plurality of clusters or systems and each of the plurality of clusters including a call control agent (CCA). Identification data corresponding to an identity of an associated user is received into a first cluster of a multiple cluster telecommunication network. A directory number and associated first telecommunication device corresponding to the user are registered with a first call control agent of the first cluster in accordance with received identification data. Registration data corresponding to the registered directory number is communicated to at least a second cluster of the telecommunications network. An incoming connection request associated with the registered directory number is routed directly to the first CCA without redirection to any other CCAs within the multiple cluster telecommunication network. | 04-28-2011 |
| 20110103368 | METHODS FOR ENABLING E-COMMERCE VOICE COMMUNICATION - A method for operating a server includes receiving a page request for a web page from a client computer via the Internet, the web page including an icon, retrieving the web page from a storage of the server, sending the web page to the client computer via the Internet, receiving a request from the client computer to initiate a telephone call via the Internet in response to a selection of the icon on the web page, initiating a real-time communications channel between the client computer and the server via the Internet in response to the request, determining a telephone number in response to the request, using a voice modem, coupled to the server and to a telephone line, to dial the telephone number, receiving packets of voice data from the client computer from the Internet, reassembling the packets of voice data into a stream of digital voice data, converting the stream of digital voice data to a stream of analog voice data, outputting the stream of analog voice data to the voice modem, and outputting the stream of the analog voice data from the voice modem to the telephone line. | 05-05-2011 |
| 20110103369 | Method and Device for Managing Personal Communications of at Least One User - A device for managing calls sent by a local terminal (T | 05-05-2011 |
| 20110103370 | CALL MONITORING AND HUNG CALL PREVENTION - A hung call system includes a memory storing samples of voice data from packets for a VoIP call. A voice activity detector detects whether the stored voice data includes a voice from one or more parties to the call. A processing circuit determines whether the voice activity detector detects the voice, and the processing circuit facilitates release of the call if the voice activity detector does not detect the voice for a predetermined period of time. | 05-05-2011 |
| 20110103371 | METHOD AND SYSTEM FOR PROVIDING SIGNALING GATEWAY MANAGEMENT - An approach is provided for signaling gateway management. Data from a plurality of signaling gateways corresponding to a plurality of trunks of a telecommunications network is automatically retrieved, each signaling gateway being configured to convert circuit-switched signaling to packet-switched signaling. The data is stored. An operating state for each of the plurality of trunks is determined based on the data to perform trending analysis for the operation of one or more of the plurality of trunks. | 05-05-2011 |
| 20110103372 | SYSTEM AND METHOD FOR SESSION INITIATION PROTOCOL HEADER MODIFICATION - A method for modifying the contents of session initiation protocol (SIP) messages is presented. The method includes receiving a SIP message. The SIP message may include a set of message header fields. The method includes receiving an application policy. The application policy may specify how to modify the SIP message based on a characteristic of the SIP message. Alternatively, the application policy may be retrieved from a database such as one provided by a home subscriber server (HSS) or an application server. The method includes using the application policy to modify the SIP message resulting in a modified message, and sending the modified message. | 05-05-2011 |
| 20110103373 | SYSTEM AND METHOD FOR SESSION INITIATION PROTOCOL HEADER MODIFICATION - A user agent (UA) for communicating with a communications network implementing an internet protocol (IP) multimedia subsystem (IMS) is presented. The UA is configured to send and receive session initiation protocol (SIP) messages. The UA includes a processor configured to send a message to the network. The message identifies an application policy. The application policy defines at least one of a SIP message header field to include, a SIP message header field to remove, a SIP message header field to allow, and a SIP message header field to modify. The processor is configured to receive a SIP message from the network. The SIP message includes a set of SIP message header fields. The set of SIP message header fields are modified in accordance with the application policy. | 05-05-2011 |
| 20110103374 | METHODS AND APPARATUS FOR PACKETIZED CONTENT DELIVERY OVER A CONTENT DELIVERY NETWORK - Methods and apparatus for delivery of packetized content (e.g., video, audio, data, etc.) over a content delivery network. In one embodiment, the content is packetized using an Internet Protocol (IP), and delivered by a service provider over both managed and unmanaged networks to subscribers of the provider, so as to provide delivery at any time, at any location, and via any designated user device. The delivered content may originate from the service provider, third-party content sources (e.g., networks or studios), the subscriber(s) themselves, or other sources including the Internet. Use of a common control and service functions within the network afford the ability to integrate or blend services together, thereby affording the service provider and subscriber new service and economic opportunities. Content delivery sessions may also be migrated from one device to another. A network-based user interface infrastructure, and gateway-based client-side architecture, are also disclosed. | 05-05-2011 |
| 20110103375 | IP TELECOMMUNICATION SYSTEM, METHOD FOR CONTROLLING COMMUNICATION IN IP NETWORK, CLIENT TERMINAL AND CLIENT SERVER - A communication system including: a main device comprising a call control section for setting-up a call connection and a remote control section. The remote control section of the main device receives a command including calling request for the target device and IP address of the terminal from the remote control section of the terminal. The call control section of the main device transmits a call control message including the IP address of the terminal to the target device via a telephony server and receives IP address of the target device in response to the call control message. The remote control section of the main device transmits the IP address of the target device to the terminal. The call control section of the terminal performs audio communication with the target device using the IP address of the terminal and the IP address of the target device. | 05-05-2011 |
| 20110103376 | TELEPHONE SYSTEM AND EXCHANGE APPARATUS FOR USE IN THE SAME - According to one embodiment, a telephone system includes an exchange apparatus and a media server. The exchange apparatus include a detector and a controller. The detector detects an amount of resources remaining in the media server of the node including the exchange apparatus. The controller transmits a tone signal to a telephone terminal of a calling node from the resources available in the media server of the node including the exchange apparatus, which is a called node, in response to a call connection request made from the calling node to the exchange apparatus if the amount of resources remaining in the media server of the called node exceeds a prescribed threshold value. | 05-05-2011 |
| 20110103377 | Implementing a High Quality VOIP Device - A method is provided for Voice over Internet Protocol (VoIP) devices to communicate over an Internet Protocol (IP) network. The method includes synchronizing the VoIP devices using one or more dual-tone multi-frequency (DTMF) codes over a telephone network, retransmissions of voice packets in bursts, retransmissions of voice packets following a time lag, adjusting the number of retransmissions based on quality of service, retransmission of a missing voice packet identified in a list received from a peer device, discarding low energy voice frames in a jitter buffer to prevent overflow, stopping playout at a low energy voice frame when the jitter buffer is below a minimum buffer size, and selective transmission and retransmission of voice packets based on their energy levels. | 05-05-2011 |
| 20110103378 | INTELLIGENT SOFTPHONE INTERFACE - An interface unit ( | 05-05-2011 |
| 20110110361 | SYSTEM FOR AND METHOD OF VALIDATING A VOIP TELEPHONE NUMBER ORDER - A system for and method of validating a VoIP telephone number order is presented. The described systems and methods may allow for corrupt telephone numbers to be discovered and placed in a corrupt telephone number pool. To this end, data may be mined from class 5 switches and VoIP routers and then compared to the telephone numbers in the order. If a telephone number is found to be corrupt, it may be removed from the order and stored in the corrupt telephone number repository for later review. The ordering process for any other requested telephone numbers may then continue on without interruption to the processing of the entire order. | 05-12-2011 |
| 20110110362 | QUANTUM AND PROMISCUOUS USER AGENTS - A call processing system includes a call processing server. The call processing server processes calls for an internal network that employs SIP features and functions. The call processing server can receive calls from or send calls to one or more external communication endpoints that are not part of the internal network. However, the call processing server can associate a floating user agent with the communication from the external communication endpoint and lock the floating user agent to a gateway. After locking onto a gateway and initiating the call, the floating user agent can then publish call event status and receive SIP primitives similar to other SIP-enabled devices. | 05-12-2011 |
| 20110110363 | SIP PARSER/GENESYS-SIP PARSER-TO PARSE SIP TELEPHONY EVENTS AND DECRYPT THE USERDATA IN IP TELEPHONY - A tangible computer-readable medium encoded with an executable computer program for retrieving information from an internet protocol network is provided. The internet protocol network includes a plurality of tangible session initiation protocol entities that exchange session initiation protocol events via the internet protocol network, wherein each of the plurality of tangible session initiation protocol entities store exchanged session initiation protocol events. The tangible computer-readable medium includes an accessing code segment that, when executed, accesses the exchanged session initiation protocol events that are stored in one of the tangible session initiation protocol entities. A parsing code segment, when executed, parses the exchanged session initiation protocol events that are stored in the one of the tangible session initiation protocol entities based on a parsing parameter. Thereafter, a reporting code segment, when executed, displays results of the parsing code segment on a display. | 05-12-2011 |
| 20110110364 | SECURE CUSTOMER SERVICE PROXY PORTAL - A portal system for secure, aggregated and centralized management and access of disparate customer service and social networking environments is disclosed. A user interface provides multiple, parameter-based automated service scripts, each configured to utilize customer information. The scripts link to vendor-specific, scenario-specific, and social networking-specific interfaces that have common user interface elements. Shared and dedicated reverse automation gateways are configured to emulate the step-by-step self-service aspects of web sites and interactive voice response systems. The portal system eliminates or reduces inbound toll-free telephone charges for vendor contact centers and additionally links the same to social networking systems. | 05-12-2011 |
| 20110110365 | PAGE-MODE MESSAGING - A method, apparatus, and computer program product are provided for page-mode messaging. The method includes determining whether a page-mode message exceeds a predetermined size limit. The method further includes sending, using a terminal, the page-mode message using a session-mode messaging mechanism when it is determined that the page-mode message exceeds the predetermined size limit, with an indication indicating that a session-mode is for the page-mode message. Further, the method includes applying, using the terminal, a session description protocol to initiate a session in the session-mode messaging mechanism, and adding, using the terminal, the indication to a header of a session initiation message. | 05-12-2011 |
| 20110110366 | UNIVERSAL COMMUNICATIONS IDENTIFIER - An approach is provided for supporting a plurality of communication modes through universal identification. A core identifier is generated for uniquely identifying a user among a plurality of users within the communication system. One or more specific identifiers are derived based upon the core identifier, wherein the specific identifiers serve as addressing information to the respective communication modes. The specific identifiers and the core identifier are designated as a suite of identifiers allocated to the user. | 05-12-2011 |
| 20110110367 | Unauthorized Call Activity Detection And Prevention Systems And Methods For A Voice Over Internet Protocol Environment - Embodiments connect a call in which at least one party is a VoIP call party and monitoring resulting VoIP signals for unauthorized call activity, such as three-way call activity. The monitoring may include monitoring the call for suspend and/or resume events to detect the unauthorized call activity, the suspend and resume events may be generated by a telephone system and passed into a VoIP system associated with the VoIP call party. The monitoring may be carried out by an agent disposed between a VoIP gateway and the VoIP call party or by the VoIP gateway itself. | 05-12-2011 |
| 20110116492 | MEDIA FORKING - In an example embodiment, a Voice over IP (VoIP) system that provides for media forking at the caller's (ingress) gateway. The gateway receives data with a first address on a recording server for sending forked caller stream media and a second address on the recording server for sending forked called party stream media. The gateway sends forked media from the caller stream to the first address and forked media from the called party media to the second address. This provides a recording from the caller's perspective. By using this technique, the recording can include for example call transfer data and interactive voice response (IVR) data. | 05-19-2011 |
| 20110116493 | METHOD AND APPARATUS FOR PROVIDING MOBILE AND SOCIAL SERVICES VIA VIRTUAL INDIVIDUAL SERVERS - A method, computer readable medium and apparatus for providing a virtual individual server service within a communications network are disclosed. For example, the method receives a request from a subscriber of the communications network to subscribe to the virtual individual server service, provides a virtual individual server to the subscriber in response to the request and executes at least one application via the virtual individual server using at least one piece of personal information associated with the subscriber. | 05-19-2011 |
| 20110116494 | OPTIMIZATION OF CONSOLIDATING ENTITIES - A system and method for homogeneously merging locations in a telecommunications network including: calculating at least one first characteristic of at least one carrier at a first location and a second location; determining at least one penalty for excluding the at least one carrier based in part on the at least one first characteristic; comparing the at least one penalty determined to at least one preselected value; and, if the at least one penalty is less than the at least one preselected value, merging the at least one first location and the at least one second location, thereby forming at least one first merged location. | 05-19-2011 |
| 20110116495 | METHOD AND APPARATUS FOR INTER-DEVICE SESSION TRANSFER BETWEEN INTERNET PROTOCOL (IP) MULTIMEDIA SUBSYSTEM (IMS) AND H.323 BASED CLIENTS - Methods and apparatus for inter-device session transfer between devices operating according to different protocols are disclosed. The session may be a multimedia session supported by. Session Initiation Protocol (SIP) and the H.323 standard. The devices may include a SIP Client or an H.323 Client. The SIP Client may be an IMS-based SIP Client. An InterWorking Function (IWF) may support a session transfer from a SIP Client to an H.323 Client or from an H.323 Client to a SIP Client. A Media Gateway Control Function (MGCF) may support a session transfer from an Internet Protocol (IP) Multimedia Subsystem (IMS)-based SIP Client to an H.323 Client or from an H.323 Client to an IMS-based SIP Client. | 05-19-2011 |
| 20110116496 | METHOD AND APPARATUS FOR GIVING MONOPOLOY OF CALL IN CALL TRANSMISSION/RECEPTION SYSTEM USING UPNP - A method of giving a monopoly of a call in a call transmission/reception system using UPnP (Universal Plug and Play) includes a telephony server setting a user's authority to manage a session when the telephony server generates the session; the telephony server performing a user authentication when the telephony server receives a call for an action for managing the session from a control point; and the control point performing the action for managing the session if a user of the control point has an authority to manage the session as the result of authentication. | 05-19-2011 |
| 20110116497 | METHOD AND APPARATUS FOR DETECTING ONE OR MORE PREDETERMINED TONES TRANSMITTED OVER A COMMUNICATION NETWORK - An apparatus for detecting one or more predetermined tones transmitted over a communication network, each predetermined tone having a predetermined frequency, comprises a data memory for storing data including the predetermined frequency of each of the one or more predetermined tones, an input for receiving a signal transmitted over the communication network, and a frequency divider for dividing the received signal into at least two frequency sub bands so as to provide at least two components of the received signal in different frequency sub bands. The different frequency sub bands are selected based on the predetermined frequencies of the one or more predetermined tones. A frequency discriminator is arranged to determine a frequency of each tone in the at least two components and a decision logic block is arranged to provide an indication that a predetermined tone has been detected when the determined frequency of a tone in a component corresponds to the predetermined frequency of one of the one or more predetermined tones. | 05-19-2011 |
| 20110116498 | SYSTEM AND METHOD FOR ENABLING DTMF DETECTION IN A VOIP NETWORK - A method, mobile terminal, and system for selectively establishing an outgoing caller ID on a mobile terminal served by a wireless network, for identifying a line called on a mobile terminal, and for directing a call from a mobile terminal to a network subscriber based on accessed information of the subscriber in the subscriber's network. | 05-19-2011 |
| 20110122861 | Method to interact with packet-network based services and applications via intelligent network signaling - A cross domain server is configured to receive calls to at least one predetermined phone number. The cross domain server is a member of a packet-switched network. The cross domain server receives a call setup message for a call from a subscriber outside of the packet-switched network. The cross domain server performs an action in the packet-switched network on behalf of the subscriber and based on the call. The call is disconnected. | 05-26-2011 |
| 20110122862 | TERMINAL, METHOD FOR OPERATING THE TERMINAL, AND METHOD FOR INTERWORKING IN WIRELESS COMMUNICATION SYSTEM INCLUDING 3GPP LTE NETWORK AND 3GPP LEGACY NETWORK - A terminal, a method for operation of the terminal, and a method of interworking in a wireless communication system including an advanced network and a legacy network are provided. The method for operating the terminal includes monitoring a paging channel of the legacy network for a data paging message and a Circuit Switched (CS) paging message, when the terminal is in an idle state, receiving one of the CS paging message and data paging message, establishing a connection with the legacy network corresponding to the received one of the CS paging message and data paging message, wherein a CS voice connection is established with the legacy network if the CS paging message is received and a Packet Switched (PS) data connection is established with the legacy network if the data paging message is received, and performing a handover to the advanced network from the legacy network, if the PS data connection is established with the legacy network. | 05-26-2011 |
| 20110122863 | ENHANCED CALL PRESERVATION TECHNIQUES FOR SIP-BASED COMMUNICATION NETWORKS - Methods, devices, and systems are provided for preserving connections, especially in a SIP environment. More specifically, the connection preservation techniques presented in this document enhance the RFC 4028-based session refresh approach in order to provide media connection preservation for calls that experience end-to-end signaling loss or refresh failures. Specifically, participants on a call can continue to exchange media despite the loss of control at the SIP signaling plane. | 05-26-2011 |
| 20110122864 | METHOD AND SYSTEM FOR TRANSFERRING IN-PROGRESS COMMUNICATION BETWEEN COMMUNICATION DEVICES - An approach is provided for transferring in-progress communication between communication devices. A transfer code is received from a target terminal of a user. An in-progress call is detected between a first terminal and a second terminal, where the first terminal is associated with the user, and the second terminal is associated with another user. Transfer of the in-progress call from the first terminal to the target terminal is initiated in response to the received transfer code. | 05-26-2011 |
| 20110122865 | METHOD AND APPARATUS FOR PROVIDING ACCESS AND EGRESS UNIFORM RESOURCE IDENTIFIERS FOR ROUTING - A method and apparatus for providing routing of calls in a packet network, e.g., a Voice over Internet Protocol (IP) network, using one or more criteria extracted from signaling information to determine the routing for the calls are disclosed. In one embodiment, the routing criteria extracted from signaling messages comprises at least one of: an access Uniform Resource Identifier, a destination phone number, a destination URI host, a calling party number, a calling party URI host, an incoming IP address, or a requested codec. An access URI and the egress URI are used to enhance routing decisions in a VoIP network. For instance, the egress URI can be used to specify egress route selections from the egress point of a VoIP network. The access URI can be used to influence the routing decisions within the VoIP network as well as the routing decisions with regard to egress routes from the egress point of the VoIP network. | 05-26-2011 |
| 20110122866 | METHOD AND APPARATUS FOR TROUBLESHOOTING SUBSCRIBER ISSUES ON A TELECOMMUNICATIONS NETWORK - Methods, systems and computer readable media defining computer instructions for isolating subscriber and service issues on a service provider network methods by retrieving status information from active network elements of a service provider network, associating the status information with a subscriber session, call setup, or service, then displaying that information along with the KPI or SLAs associated with those active elements. | 05-26-2011 |
| 20110122867 | METHOD AND NODE FOR ROUTING A CALL WHICH HAS SERVICES PROVIDED BY A FIRST AND SECOND NETWORKS - A method for routing a call having services provided by a first network and a second network comprises: receiving the call in the first network; correlating the first network with the second network for the call; and sending the call to the second network. The correlation between the first network and the second network allows the call for returning back to the first network from the second network, after the services provided by the second network have been applied. A communication node for carrying out the method comprises: an input module for receiving the call in the first network; a generator of a correlation between the first and second networks for the call; and an output module for sending the call to the second network. | 05-26-2011 |
| 20110122868 | COMMUNICATION METHOD AND GATEWAY DEVICE BASED ON SIP PHONE - The present invention relates to the communication field and discloses a communication method and gateway device based on Session Initiation Protocol (SIP) phones. To enable a SIP phone to communicate over the Circuit Switched (CS) domain of a High Speed Packet Access (HSPA) network, the technical solution of the present invention includes: when receiving SIP signaling from a SIP phone, sending an HSPA call command associated with the SIP signaling to an HSPA network device; when receiving an HSPA call command from the HSPA network device, sending SIP signaling associated with the HSPA call command to the SIP phone; when receiving a Real-Time Transport (RTP) packet from the SIP phone, sending Pulse Code Modulation (PCM) data associated with the RTP packet to the HSPA network device; and when receiving PCM data from the HSPA network device, sending an RTP packet associated with the PCM data to the SIP phone. The present invention is applicable to SIP communications. | 05-26-2011 |
| 20110122869 | Method of Transmitting Data in a Communication System - A method of transmitting a first signal from a first terminal to a second terminal via a communication network including: receiving at the first terminal a second signal from the second terminal; outputting the second signal from an output device associated with the first terminal and determining information relating to a characteristic of the second signal. A processing resource of the second terminal used to transmit the second signal is estimated, wherein the estimation is based on the information relating to the characteristic of the second signal. A characteristic of the first signal is adjusted in dependence on the estimated processing resource of the second terminal used to transmit the second signal and the first signal is transmitted to the second terminal. | 05-26-2011 |
| 20110128953 | Method and System of Voice Carry Over for Instant Messaging Relay Services - A method of assisting communication for a user is provided. The method includes receiving an IM message including a request for a voice carry over from the user, and transmitting to the user an invitation to join a first voice connection. The method further includes initiating the first voice connection with the user, and initiating a second voice connection with a recipient. Additionally, the method includes communicating to the recipient a first voice communication from the user over the first and second voice connections, and communicating to the user a response IM message including a transcribed version of a second voice communication from the recipient. An apparatus for assisting communication for a user is provided. A computer-readable medium having stored thereon computer-executable instructions is provided. The computer-executable instructions cause a processor to perform a method when executed. | 06-02-2011 |
| 20110128954 | Automated Service Migration - A method and a system for automated service migration from a first network to a second network is described, wherein the first network is connected to a Residential Network Access Device (RNAD) via a first type of signaling, with electromagnetic properties representative of the first network. The second network comprises an User Agent (UA) Registration and Management Module (RMM) communicatively connected to a routing database and the RNAD comprises a User Agent (UA) and is configured for switching between a first signaling processing modus and a second signaling processing modus. The RNAD is switched to the second modus through transmitting to the RNAD a second type of signaling with electromagnetic properties, representative of the second network. The UA is activated, configured, and registered with the RMM. The routing database is updated so that the RNAD may be reached via the second network. | 06-02-2011 |
| 20110128955 | EMERGENCY SERVICES FOR PACKET NETWORKS - The present invention provides a technique for facilitating emergency services via packet networks. Emergency service providers will implement emergency proxies to ensure that proper call setup requests for emergency services are forwarded to the appropriate entities, even if those entities are in overload conditions. The emergency proxies may authenticate and filter call setup requests to ensure that only proper call setup requests are forwarded to help prevent such overload conditions. The emergency proxies may operate solely in a packet network, as well as at the interface between a packet network and a circuit-switched network to assist in call setup requests originating from either the packet network or the circuit-switched network. | 06-02-2011 |
| 20110134907 | METHOD AND APPARATUS FOR EFFICIENTLY ROUTING PACKETS ACROSS DISPARATE NETWORKS - A method, a system and a computer readable medium for routing packets across disparate networks are disclosed. For example, the method receives, via a media gateway controller (MGC), an external request from an external requestor for a reservation of a public switched telephone network (PSTN) trunk on a media gateway (MGW) for a communication session between a voice over Internet protocol (VoIP) network and a PSTN network, sends, via the MGC, a H.248 request to the MGW to make the reservation, establishes via the MGW, a communication path and sending a message to the MGC, retrieves, via the MGC, an assigned Internet protocol (IP) address and IP port on the MGW from the message from the MGW, sends, via the MGC, an allocation request to a media terminating session border controller (SBC) and allocates, via the media terminating SBC, a public IP address and a public IP port from an available pool of IP addresses and IP ports at the media terminating SBC. | 06-09-2011 |
| 20110134908 | SINGLE SLOT DTM FOR SPEECH/DATA TRANSMISSION - The present document relates to radio transmission. In particular, the present document relates to the single-slot dual transfer mode (DTM) available e.g. in GSM/GPRS/GERAN networks. A transmitter is described. The transmitter is configured to send circuit switched data over a traffic channel to a corresponding receiver, wherein the traffic channel is segmented into a plurality of frames. The transmitter if further configured to determine a vacant frame of the plurality of frames, wherein no circuit switched data is sent in the vacant frame due to discontinuous transmission; and to send packet switched data over the traffic channel using the vacant frame. | 06-09-2011 |
| 20110134909 | DATA COMMUNICATION WITH COMPENSATION FOR PACKET LOSS - Described is a technology by which a relay is coupled (e.g., by a wire) to a network and (e.g., by a wireless link) to an endpoint. Incoming data packets directed towards the endpoint are processed by the relay according to an error correction scheme, such as one that replicates packets. The reprocessed packets, which in general are more robust against packet loss, are then sent to the endpoint. For outgoing data packets received from the endpoint, the relay reprocesses the outgoing packets based upon the error correction scheme, such as to remove redundant packets before transmitting them to the network over the wire. Also described are various error correction schemes, and various types of computing devices that may be used as relays. The relay may be built into the network infrastructure, and/or a directory service may be employed to automatically find a suitable relay node for an endpoint device. | 06-09-2011 |
| 20110134910 | REAL-TIME VOIP COMMUNICATIONS USING N-WAY SELECTIVE LANGUAGE PROCESSING - A computer-implemented method and system of enabling concurrent real-time multi-language communication between multiple participants using a selective broadcast protocol, the method including receiving at a first server a real-time communication from a first participant, the real-time communication being addressed to a second participant constructed in a first spoken language. A preferred spoken language of receipt of real-time communication is identified by the second participant. A determination is made whether the preferred spoken language of receipt is different than that of the first spoken language of the real-time communication. The real-time communication from the first spoken language is translated and delivered to the preferred spoken language of receipt of the second participant to create a translated real-time communication whenever the preferred spoken language is different than the first spoken language and forwarded without translation when the preferred spoken language of the second participant is the same as the preferred spoken language of the first participant. | 06-09-2011 |
| 20110134911 | Selective filtering for digital transmission when analogue speech has to be recreated - A method, terminal and program for making a call in a packet switched network between a calling device and a called device. The method comprises receiving at a processor of the calling device samples of a speech signal and an identity of the called device, executing code on the processor to perform the steps of: determining based on the identity of the called device whether a filter should be applied to the samples, when it is determined that a filter should be applied, filtering the samples, and encoding the filtered samples for transmission on the packet switched network. | 06-09-2011 |
| 20110134912 | SYSTEM AND METHOD FOR PLATFORM RESILIENT VOIP PROCESSING - A system and method for platform resilient VoIP (Voice over Internet Protocol) processing in a partitioned environment. The system comprises a plurality of soft partitions. At least one soft partition is a sequestered partition. The sequestered partition includes one or more core processors having a controlled, real-time operating system and at least one network interface card (NIC) coupled to the one or more core processors. The NIC is dedicated to the sequestered partition, and the one or more core processors are used as an offload engine solely dedicated to Voice over Internet Protocol (VoIP) processing. | 06-09-2011 |
| 20110134913 | Auxiliary SIP Services - The invention includes a method of providing media data of a SIP-based Auxiliary Service from an Auxiliary Application Server, AS, to a recipient peer of an established communication exchange between peers. The method includes issuing an invocation to the Auxiliary AS, the invocation including an indication of the recipient peer of the Auxiliary Service. The Auxiliary Service media data is prepared and sent to the recipient together with a correlation ID identifying the established communication exchange, and an Application Classmark identifying the auxiliary service. | 06-09-2011 |
| 20110142031 | METHOD AND APPARATUS FOR DYNAMICALLY ASSIGNING BORDER ELEMENTS IN A VOICE OVER INTERNET PROTOCOL NETWORK - In one embodiment, the present disclosure is a method and apparatus for dynamically assigning border elements in a Voice over Internet Protocol network. In one embodiment, a method for registering an endpoint device to a core Internet Protocol network includes selecting a border element in the network, where the border element is selected based on monitored data relating to at least one of: a condition of the network and a condition of at least one component of the network and sending a message to the endpoint device instructing the endpoint device to register with the network via the border element. | 06-16-2011 |
| 20110142032 | REUSABLE AND EXTENSIBLE FRAMEWORK FOR MULTIMEDIA APPLICATION DEVELOPMENT - Systems and methods of developing and/or implementing multimedia applications. The system provides an extensible framework including an application layer, a framework utility layer, and a core engine layer. The framework utility layer includes an application programming interface, a video codec sub-framework (XCF), a video packetization sub-framework (XPF), and a video/text overlay sub-framework (XOF). The core engine layer includes one or more core codec engines and one or more core rendering engines. The XCF, XPF, and XOF sub-frameworks are effectively decoupled from the multimedia applications executing on the application layer, and the core codec and rendering engines of the core engine layer, allowing the XCF, XPF, and XOF sub-frameworks and core codec/rendering engines to be independently extensible. The system also fosters enhanced reuse of existing multimedia applications across a plurality of multimedia systems. | 06-16-2011 |
| 20110142033 | ELIMINATING FALSE AUDIO ASSOCIATED WITH VoIP COMMUNICATIONS - Embodiments are directed to eliminating false audio using an egress gateway in a communications network. At least one false audio packet is received by an egress gateway. The false audio packet includes false audio. A DTMF packet is received by the egress gateway. The DTMF packet is received subsequent to the at least one false audio packet. The false audio in the false audio packet is replaced with a substitute signal by the egress gateway. | 06-16-2011 |
| 20110142034 | CONTROL OF BIT-RATE AND PACKET DUPLICATION IN A REAL-TIME MEDIA STREAM - A method for controlling a real-time media stream between a sender and a receiver. The method includes streaming, from the sender, media packets over a network at a bit-rate, determining at the sender a loss-rate for the streamed media packets not received at the receiver. The sender optionally generates duplicate packets for a selected number of media packets and streams the duplicate packets over the network when the loss-rate is above a first loss-rate threshold, or varies the bit-rate of streaming the media packets over the network when the loss-rate is above a second loss-rate threshold. | 06-16-2011 |
| 20110142035 | Securing Uniform Resource Identifier Information for Multimedia Calls - A session request from a first subscriber is received at a first network component of a packet-based network. The session request comprises a request to establish a communications session between the first subscriber and a second subscriber. In the event the session request originated in a trusted network, the first network component permits access to unique resource identifier (URI) information associated with the second subscriber for use in establishing the communications session via the packet-based network. In the event the session request did not originate in a trusted network and in response to determining a security configuration associated with the second subscriber allows the first subscriber to access the URI information under the circumstances, the first network component permits access to the URI information for use in establishing the communications session via the packet-based network. In response to determining the security configuration prohibits access to the URI information by the first subscriber under the circumstances, the first network component forwards the session request to a second network component so as to establish the communications session via a public switched telephone network. | 06-16-2011 |
| 20110142036 | METHOD AND APPARATUS FOR SWITCHING PACKET/TIME DIVISION MULTIPLEXING (TDM) INCLUDING TDM CIRCUIT AND CARRIER ETHERNET PACKET SIGNAL - Provided is a packet/TDM switch that may classify a type of a received signal based on slot recognition information received from an Ethernet mapping unit or a TDM mapping unit, and may process the received signal using a dedicated switch corresponding to each of the Ethernet mapping unit and the TDM mapping unit according to the type of the received signal. | 06-16-2011 |
| 20110142037 | METHOD, SYSTEM AND APPARATUS FOR CONTROLLING PLAY OF CUSTOMIZED RING BACK TONE SERVICE - A method, a system, and an apparatus for controlling play of a Customized Ring Back Tone (CRBT) service are disclosed. The method may be: receiving a play control instruction sent by a CRBT receiving terminal; and sending a CRBT to the CRBT receiving terminal according to the play control instruction. The system may include: a CRBT receiving terminal, configured to send a play control instruction to a CRBT platform, and obtain a CRBT sent by the CRBT platform according to the play control instruction; and a CRBT platform, configured to send the CRBT to the CRBT receiving terminal according to the play control instruction. A User Equipment (UE) and a CRBT platform are also disclosed. Through the information interaction between the CRBT receiving terminal and the CRBT playing terminal, the CRBT receiving terminal can control the play of the CRBT, and the user experience of the CRBT receiving terminal is improved. | 06-16-2011 |
| 20110142038 | DIGITAL TELEPHONE DATA AND CONTROL SIGNAL TRANSMISSION SYSTEM - Techniques are disclosed for using Ethernet Layer | 06-16-2011 |
| 20110149948 | ON-NET DIRECT ACCESS TO VOICEMAIL - A device in a provider network receives a Session Initiation Protocol (SIP) request message from an originating device, where the SIP request message includes a general number for a voicemail service and where the voicemail service includes multiple voicemail systems. The device determines whether the originating device is associated with a voice-over-Internet-protocol (VoIP) account on the provider network and, when the originating device is associated with a VoIP account, selects a direct access number assigned to a voicemail system, from the multiple voicemail systems in the network, that services the VoIP account. The device also associates the direct access number and the SIP request message, and forwards, based on the direct access number, the SIP request message to an application server. | 06-23-2011 |
| 20110149949 | METHOD AND APPARATUS FOR CLEARING HANG CALLS - A method, computer readable medium and apparatus for clearing hang calls in a communication network are disclosed. For example, the method detects a failure of an adjacent call stateful network element in a signaling path, identifies one or more calls that are affected by the failure of the adjacent call stateful network element, tests a media path of the one or more calls for activity and clears the one or more calls that are affected by the failure of the adjacent call stateful network element if the media path of the one or more calls is inactive. | 06-23-2011 |
| 20110149950 | Systems, Methods, Devices and Arrangements for Cost-Effective Routing - A variety of methods, systems, devices and arrangements are implemented for assessing and/or controlling call routing for VoIP/VioIP calls. According to one such method, endpoint devices are used to monitor and/or assess the call-quality. The assessment is sent to a centralized server arrangement and call-routing is controlled therefrom. Endpoint devices employ a decentralized testing mechanism to further monitor and assess call quality including the use of test connections. Aspects of call quality are analyzed and attributed to endpoint devices and/or local connections or networks to distinguish intermediate routing issues from local/endpoint issues. | 06-23-2011 |
| 20110149951 | METHOD AND APPARATUS FOR MANAGING COMMUNICATION FAULTS - A system that incorporates teachings of the present disclosure may utilize, for example, a method involving receiving from a first communication device a service request while providing back-up services to an out-of-service network element, detecting a deficiency in call state information to process the service request, transmitting to the first communication device an error message that prevents termination of an active Internet Protocol (IP) communication path between the first communication device and a second communication device, and receiving from the first communication device a request for an alternate IP communication path for communicating between the first and second communication devices which resolves the deficiency in call state information. Additional embodiments are disclosed. | 06-23-2011 |
| 20110149952 | MULTIMEDIA TERMINAL ADAPTER AND REMOTE CONNECTION METHOD - A multimedia terminal adapter saves IP addresses of the multimedia terminal adapter and second communication devices as an IP address list, and relationships between the IP addresses of the second communication devices and user ports as a relationship list. The multimedia terminal adapter sends the IP address list to the first communication device, and receives a selected IP address of a selected second communication from the first communication device. The multimedia terminal adapter further searches one user port corresponding to the selected IP address, and opens the searched user port to establish a remote connection between the first communication device and the selected second communication device. | 06-23-2011 |
| 20110149953 | Tracking results of a v2 query in voice over internet (VoIP) emergency call systems - A simplified method of encoding information needed to set the NENA 08-001 v2 Result code based on two essential factors that are stored in a data store at runtime. An ESRResponse header is built with two fields created as simple enumerated types: LocationSrc and RoutedOnAlgo. For each emergency call there is one entry in this data store. The first field of the ESRResponse header comprises one of five possible unique values, as does the second field. | 06-23-2011 |
| 20110149954 | Wireless emergency services protocols translator between ANSI-41 and VoIP emergency services protocols - A protocol converter or translator between ANSI-41 ORREQs and VoIP V2 messaging. The protocol converter may alternatively (or also) provide conversion between GMS MAP and VoIP V2 messaging. Interaction of VSPs with a Mobile Positioning Center (MPC) or a Gateway Mobile Location Center (GMLC) is permitted, as is interaction of wireless carriers with a Voice Positioning Center (VPC). In this way existing GMLCs or MPCs may be used to service VoIP 9-1-1 calls. Moreover, operators of Voice Positioning Centers (VPCs) who implement wireless offerings can re-use their existing VPCs to service wireless 9-1-1 calls. | 06-23-2011 |
| 20110149955 | SYSTEMS AND METHODS FOR PREVENTING FRAUD IN AN INTERNET PROTOCOL TELEPHONY SYSTEM - Systems and methods for preventing fraud in an IP based telephony system include noting when an IP based telephony device sent to a new customer is not installed and registered with the system. If a new customer never attempts to register a device which was sent to the new customer, the system will assume that the new customer submitted false or erroneous address information. A new customer is prevented from taking any actions that would result in new charges until the new customer has registered an IP device sent to the new customer. Likewise, the system will act to prevent a phone verification service from reaching a new customer at his newly assigned telephone number until after the new customer has registered an IP based telephony device sent to the new customer. | 06-23-2011 |
| 20110149956 | METHOD AND SYSTEM FOR PROVIDING SECURE MEDIA GATEWAYS TO SUPPORT INTERDOMAIN TRAVERSAL - An approach provides interdomain traversal to support packetized voice transmissions. A signaling message is received for establishing a voice call from a first endpoint associated with a first domain to a second endpoint associated with a second domain. The first endpoint queries a STUN (Simple Traversal of UDP (User Datagram Protocol)) server to determine information relating to a firewall and network address translator that the first endpoint is behind, and to log into a TURN (Traversal Using Relay NAT (Network Address Translation)) server configured to establish a media path between the first endpoint and the second endpoint. A first proxy server serving the first endpoint communicates with an ENUM (Electronic Number) server to convert a directory number corresponding to the second endpoint to a network address. The first proxy server communicates with a second proxy server serving the second endpoint to establish the voice call. The STUN server, the TURN server and the ENUM server are maintained by service provider. The first endpoint is authenticated to permit exchange of a media stream over the media path. The media stream is relayed, if the first endpoint is successfully authenticated. | 06-23-2011 |
| 20110158222 | CELLULAR TELEPHONE SYSTEMS WITH SUPPORT FOR CONVERTING VOICE CALLS TO DATA SESSIONS - Wireless electronic devices such as cellular telephones may communicate with computing equipment such as servers over a network. Voice telephone calls may be routed over voice links in a voice network and data may be conveyed over data links in a data network. The voice network may be formed using the public switched telephone network. The data network may be formed using the Internet. Cellular base stations may form wireless links with the wireless devices. A server may store information on the current internet protocol address of a wireless device user. The user may place a voice telephone call to an organization. In response to receiving the voice telephone call, a server may automatically transmit information such as web pages or other data that includes interactive on-screen options to the wireless device using the current internet protocol address of the device. | 06-30-2011 |
| 20110158223 | METHOD, SYSTEM NETWORK AND COMPUTER-READABLE MEDIA FOR CONTROLLING OUTGOING TELEPHONY CALLS TO CAUSE INITIATION OF CALL FEATURES - The present invention discloses numerous implementations for IP-based call processing systems that can selectively control an outgoing call initiated by a source device to a destination device. The call processing system causes a Service Switching Point (SSP) associated with the source device to initiate a media connection between the IP-based call processing system and the source device. The call processing system further causes initiation of a call feature for the outgoing call using the media connection with the source device. The call feature may include a call restriction feature, a call feature for conveying an audio element to the source device, a call record feature and a call feature for conveying information to the source device. The call processing system further causes establishment of a media connection between the source and the destination devices. | 06-30-2011 |
| 20110158224 | COMMUNICATION SYSTEM AND TELEPHONE EXCHANGE APPARATUS - According to one embodiment, a communication system includes a Network Address Translator (NAT) rooter and a telephone exchange apparatus. The NAT router comprises a transfer module configured to transfer a communication packet brought from the global network to the telephone exchange apparatus. The telephone exchange apparatus comprises a memory configured to store a map table in which a terminal ID specifying the terminal, and an address and a port number specifying the network are correlated with each other, and a controller configured to refer to the map table, and notify the terminal connected to the global network of an address and a port number of the telephone exchange apparatus's own apparatus as an address and a port number of the communication partner, and bring the communication path between the terminals into the apparatus. | 06-30-2011 |
| 20110158225 | TELEPHONE APPARATUS AND COMPUTER READABLE MEDIUM - A telephone apparatus that can be connected to both an IP network and a public switched telephone network, the telephone apparatus includes: a microphone unit, a speaker unit, an operation unit that is operated by a user and a call control unit. The call control unit includes a call processing unit and a relay unit. The relay unit includes a call request transmission section; a conversion section; an analog voice data transmission section; and a digital voice data transmission section. Another telephone apparatus that can be connected to the IP network can perform a voice data communication with a public line telephone apparatus connected to the public switched telephone network, via the above-described telephone apparatus. | 06-30-2011 |
| 20110158226 | DIGITAL TELECOMMUNICATIONS SYSTEM, PROGRAM PRODUCT FOR, AND METHOD OF MANAGING SUCH A SYSTEM - A digital telecommunications system | 06-30-2011 |
| 20110158227 | GATEWAY HAVING DISTRIBUTED PROCESSING FUNCTION, AND COMMUNICATION TERMINAL - Conventionally, in a system where devices for handling multiple media data such as audio and video are present, there is a problem that the number of audio channels that can be processed at the gateway is limited. In light of this problem, this invention offers a gateway having distributed processing function for a telephone or a data processing system featuring the capability to request address conversion to another terminal within the system to replace the address of stream-type packet data such as audio and video meant for itself, and if the aforementioned terminal to which the request was sent responds that it can handle the requested processing, to notify the address of the terminal processing the address conversion to the terminal transmitting the stream-type packet data. | 06-30-2011 |
| 20110158228 | Methods, smart cards, and systems for providing portable computer, VOIP, andapplication services - A smart card is used with a network based system to providing portable telecommunication and computing services. In an exemplary embodiment the smart card holds a user authentication code and user telephony account information. The smart card transfers the user authentication code and the account information to one of a plurality of geographically dispersed card readers which are each connected to a local telephony device. When the smart card is plugged into a first card reader, telephone calls directed to the smart card user's follow-me telephone number are received at a first local telephony device. When the smart card is plugged into a second smart card reader, telephone calls directed to the follow-me telephone number are received at a second telephony local device. Hence the user is enabled to receive and place calls using any of the geographically dispersed telephony devices as though they were his/her own personal landline or cellular telephone supplied by his/her telephony services provider. | 06-30-2011 |
| 20110164608 | IP multimedia subsystem access method and apparatus - A method of and apparatus for facilitating access to IP Multimedia Subsystem, IMS, services by non-IMS enabled terminals. A non-IMS enabled terminal registers with a Home IMS gateway. In response to that registration, an IMS registration is performed on behalf of the terminal between the Home IMS gateway and the IMS using information obtained from an ISIM application present at the Home IMS gateway. | 07-07-2011 |
| 20110164609 | 1X MESSAGE PROCESSING - An apparatus for notifying of a circuit switched event over a packetized data network. The apparatus includes a packetized data modem and an internetworking interface. The packetized data modem is configured to transmit and receive packetized data over a packetized data radio link. The packetized data modem has a tunneling link access control processor that is configured to encapsulate/decapsulate data for a subset of sub-layers corresponding to a link access control layer of a circuit switched network model. The internetwork interface is operatively coupled to the packetized data modem via the packetized data network, and is configured to notify the packetized data modem of the circuit switched event. The internetworking interface has a link access control/tunneling link access control processor that is configured to encapsulate/decapsulate the data when performing notification of the circuit switched event. | 07-07-2011 |
| 20110164610 | METHODS TO ROUTE, TO ADDRESS AND TO RECEIVE A COMMUNICATION IN A CONTACT CENTER, CALLER ENDPOINT, COMMUNICATION SERVER, DOCUMENT SERVER FOR THESE METHODS - Click-to-dial function whereby the URL sent to the contact center is appended with additional information used within the contact center (ACD) for routing. Function is known under the terms such as: extended URL, URL Encoding, Percent-encoding and the query string (part of a URL that contains data to be passed to web applications such as CGI programs). The method to route a communication from a caller to a specific endpoint in a contact center comprises the routing ( | 07-07-2011 |
| 20110164611 | AUTOMATED ATTENDANT MULTIMEDIA SESSION - An automated attendant system is made multimedia capable by adding a combined user agent to the automated attendant. A search is done to verify that the caller to the automated attendant has combined user agent capabilities. If so, the caller receives multimedia content from the automated attendant's combined user agent so that the content may be presented on the caller's computer to assist the caller in navigating through the automated attendant's menus and options. Upon selection of a desired connection from the menus and options, the automated attendant's combined user agent helps the caller be connected by voice to the selected connection. | 07-07-2011 |
| 20110164612 | METHOD AND APPARATUS FOR BLOCKING A PAY-PER-USE FEATURE IN A COMMUNICATIONS NETWORK - A method and apparatus for blocking at least one pay-per-use feature in a communications network is described. In one embodiment, a request to initiate at least one pay-per-use feature from at least one endpoint device associated with a subscriber is received. A determination of whether a blocking function has been activated for the at least one pay-per-use feature is then made. Afterwards, the request to initiate the at least one pay-per-use feature is blocked if the blocking function is activated. | 07-07-2011 |
| 20110164613 | Media negotiation method for IP multimedia link - A media negotiation method for an IP multimedia link is used in the process of establishing an IP multimedia link between a first entity and a second entity via an application server (AS) of an IP multimedia subsystem (IMS). AS sends the second entity an invite message, which includes media resource information of the first entity; When AS receives a message with media resource information from the second entity before an answer message is received or after it receives a response message with media resource information from the second entity, AS sends an IMS re-invite message without media source information to the first entity; the AS, after receiving the IMS signaling message with media resource information from the first entity, sends the first entity the media resource information returned by the second entity. The present invention is applicable to an IMS centralized service and may effectively reduce the number of steps and the time required after response for media resource re-negotiation. | 07-07-2011 |
| 20110170537 | One Way and Round Trip Delays Using Telephony In-Band Tones - There is disclosed a method for measuring the delay between a source device and a destination device on a network. The source device may generate a timestamp and encode the generated timestamp as an in-band tone sequence. The source device may transmit the in-band tone sequence to the destination device. There is also disclosed a network device comprising a timer to generate a timestamp, a tone generator capable of encoding the timestamp as an in-band tone sequence and a tone transmitter to transmit the in-band tone sequence. | 07-14-2011 |
| 20110170538 | Method and System for Communications Roaming - A method and system for communications roaming discloses that end subscribers apply for numbers provided by roaming networks, connect their numbers in local networks to roaming network ones and then store them in the database in Global Roaming Interface Agent (GRIA). When a called number is dialed, the calling communications network first determines if the calling number or the called number belongs to a number in local network. If no, the calling network sends Global Roaming Interface Agent a request to search and check if the number in roaming network is stored under the calling number or the called number; if yes, the corresponding calling number or called number in roaming network will be used as the number for communication. Taking the technical solution from this invention, a flexible and low-cost global roaming communications system can be achieved by binding subscribers' multiple numbers to establish a uniform end-subscriber communications interface. | 07-14-2011 |
| 20110176536 | HEIRARCHICAL PROTOCOL CLASSIFICATION - A hierarchical protocol classification and signaling method specifies the interworking protocols used to send circuit-switched signaling messages to and from a mobile terminal in a packet-switched network. A set of possible interworking protocols are divided into two more classes that correspond to different types of interworking protocols. Within each class, different versions of the interworking protocol are specified by a revision value. The versions of the interworking protocols within a given class are may be denominated such that the versions with a higher revision value are backward compatible with versions having a lower value. When a circuit services domain message is sent from a sending device to a receiving device, an interworking option specifying the class/revision of the interworking protocol is transmitted along with circuit services domain messages. The interworking option may be inserted into the header of a tunneling packet containing the circuit services domain message. | 07-21-2011 |
| 20110176537 | METHOD AND SYSTEM FOR PRESERVING TELEPHONY SESSION STATE - A method and system for preserving session state in telephony communication including initializing a communication session of telephony communication between a telephony device and an application server; routing the telephony communication through a call router; storing session state for the communication session of the telephony device and the application server; and transmitting the stored session state in communication between the application server and the call router. | 07-21-2011 |
| 20110176538 | METHOD OF CALL TRACE ON MEDIA GATEWAY OF NEXT GENERATION NETWORK - The present invention discloses a method of call trace on a next generation network (NGN) media gateway (MG). According to the present invention, a softswitch device adds an extended trace indication to an H.248 message related to the call to be traced; the MG determines whether the received H.248 message needs to be traced according to the extended trace indication; and if the H.248 message needs to be traced, trace the call to which the transaction in the H.248 message belongs. By the method, the MG can trace the entire calling process of a call. | 07-21-2011 |
| 20110176539 | METHOD AND DEVICE FOR PROCESSING MULTIMEDIA MESSAGING SERVICE NOTIFICATION MESSAGE AND MULTIMEDIA MESSAGING SERVICE RECEIVING SYSTEM - A method and a device for processing a multimedia messaging service notification message and a multimedia messaging service receiving system are provided. The method for processing a multimedia messaging service notification message includes: receiving the multimedia messaging service notification message, and adding the multimedia messaging service notification message into a preset processing queue; setting a processing identifier which is used for indicating whether there is a circuit switch domain/a packet switch domain service being processed currently; judging whether there is a circuit switch domain/a packet switch domain service being processed currently according to the processing identifier, wherein if YES, it maintains the multimedia messaging service notification message in the processing queue for the purpose of being processed, and if NO, it reads a prior multimedia messaging service notification message from the processing queue for processing. | 07-21-2011 |
| 20110176540 | Network Telephony System - The present invention includes a network telephone having a microphone coupled to provide voice data to a network, a speaker coupled to facilitate listening to voice data from the network, a dialing device coupled to facilitate routing of voice data upon the network, a first port configured to facilitate communication with a first network device, a second port configured to facilitate communication with a second network device and a prioritization circuit coupled to apply prioritization to voice data provided by the microphone. | 07-21-2011 |
| 20110176541 | SYSTEM, METHOD AND APPARATUS FOR SUPPORTING E911 EMERGENCY SERVICES IN A DATA COMMUNICATIONS NETWORK - A system, method and apparatus for supporting enhanced 911 (E911) emergency services, in a data communications network that includes Voice over Internet Protocol (VoIP) telephones. A network system includes a host network communicatively coupled to an E911 database management system, a network access device, and a VoIP telephone communicatively coupled to an input port of the network access device. The network access device is adapted to assign a physical location identifier to an input port, to authenticate the VoIP telephone, wherein the authentication includes receiving a unique device identifier from the VoIP telephone, and to transmit the location identifier and the unique device identifier to the E911 database management system. The E911 database management system is permitted to store the physical location identifier in association with the unique device identifier. | 07-21-2011 |
| 20110182281 | FACILITATING VERIFICATION OF CALL LEG SETUP IN THIRD PARTY CALL CONTROL SYSTEMS - In a third party call control system, a controller sends a command to a PBX for causing the PBX to initiate setup of a call leg between the PBX and a telephone device. The PBX responsively places a telephone call to the telephone device and sends an indicator to the controller that the call is in a ringing state. Responsive to the indicator, the controller subscribes with the PBX for event notification of DTMF tones from the telephone device for verifying the setup of the call leg. Configuration of the PBX for providing the desired event notification to the controller may thus be completed before any DTMF tones arrive at the PBX. This may be true even if an audio channel of the telephone call is established before the PBX receives any indication that the call was answered. Verification of call leg setup by the controller may thus be facilitated. | 07-28-2011 |
| 20110182282 | MODEM AND METHOD SUPPORTING VARIOUS PACKET CABLE PROTOCOLS - A modem includes a communicating module, a multimedia terminal adapter (MTA) module, a parsing module, and a selecting module. The communicating module sends a configuration file request packet to a TFTP server to get configuration files including a file ID. The parsing module parses the configuration files to get the file ID. The selecting module configures the MTA module corresponding to the file ID. The communicating module further communicates with a VoIP network according to a protocol corresponding to the file ID. | 07-28-2011 |
| 20110182283 | Web-based, hosted, self-service outbound contact center utilizing speaker-independent interactive voice response and including enhanced IP telephony - Disclosed is an on demand, web-based, outbound contact center utilizing Voice over IP (VoIP) and speaker-independent voice recognition which automatically captures contact responses to question events in a pre-recorded, interactive voice call, the call launched by a user via a broadcast comprising a call sequence created by the user via a call center user interface comprising event add and logic add wizards, the call sequence comprising event prompts based on a user-generated script comprising message events and question events, the event prompts in the group consisting of voice recordings and text-to-speech inputs. | 07-28-2011 |
| 20110182284 | Proxy Server, Computer Program Product and Methods for Providing a Plurality of Internet Telephony Services - A proxy server including a system manager and a database is provided. The system manager includes an internal registrar module, an external registrar module, a session manager module and a signal routing module. The internal registrar module provides an internal register service for a plurality of nodes in a first service network. The external registrar module registers at an internet service provider providing network services in a second service network. The session manager module manages session processes in the first service network and the second service network and manages the network services shared between the registered nodes. The signal routing module routes control signals of the session processes between the first service network and the second service network. The database stores information related to the registered nodes. | 07-28-2011 |
| 20110182285 | Sessions In A Communication System - A method in a communications system for handling responses to messages includes a step of sending a message from a first party to a second party. A response to the message is sent, with the response including at least one parameter in breach of a policy for a communication between the first party and the second party. A network controller detects that the response includes at least one parameter breaching the policy. The at least one parameter is modified to be consistent with the policy. | 07-28-2011 |
| 20110182286 | SEPARATION DEVICE AND METHOD FOR TRANSMITTING VOICE SIGNAL - A separation device and a method for transmitting voice signal are disclosed. The separation device includes a line interface adapted to provide an interface for connecting to a telecommunication access system which provides public switched telephone network (PSTN) and data services; a data interface adapted to provide an interface connected with customer premises equipment (CPE) and connected with the line interface; a first communication interface adapted to provide an interface connected with a PSTN communication terminal; a second communication interface adapted to provide an interface connected with a foreign exchange station (FXS) interface of the CPE; and a connection switching unit adapted to connect the first communication interface with the line interface or connect the first communication interface with the second communication interface, and to perform switching between the two connections to implement PSTN communication or VoIP communication respectively. The solution in the embodiment may help perform switching with a PSTN communication terminal connection line according to different call types so that the PSTN communication terminal may implement the PSTN communication and the VoIP communication respectively. | 07-28-2011 |
| 20110182287 | Methods, Systems, and Computer Program Products for Enabling Non-IMS Queries of a Common Telephone Number Mapping System - Methods of routing a non-IP multimedia subsystem (IMS) message from a first user terminal that has telecommunications service provided by a first carrier to a second user terminal that has telecommunications service provided by a second carrier are provided. Pursuant to these methods, a first telephone number mapping (ENUM) database is queried to identify an address of a second ENUM database that is operated by the second carrier. The identified address is used to query the second ENUM database. Routing information for the non-IMS message is received from the second ENUM database in response to the query to the second ENUM database. The non-IMS message may then be routed to the second user terminal based on the routing information received from the second ENUM database. Related systems and computer program products are also provided. | 07-28-2011 |
| 20110182288 | END-POINT AWARE RESOURCE RESERVATION PROTOCOL PROXY - A method performed by a first network device may include receiving a request for a resource from an end-point device and acknowledging the request for the resource to the end-point device. The method may also include receiving a resource coordination message from a second network device and transmitting a return resource coordination message to the second network device. | 07-28-2011 |
| 20110188491 | SYSTEM FOR RAPIDLY ESTABLISHING HUMAN/MACHINE COMMUNICATION LINKS USING PRE-DISTRIBUTED STATIC NETWORK-ADDRESS MAPS IN SIP NETWORKS - A method and system are provided that enhance human/machine communication so as to more closely approximate natural human/human communication by more effectively establishing communications links for human-interactive media. Specifically, the speed and quality of the connection are improved by the method and system, resulting in a more natural user experience. The method includes receiving a Session Initiation Protocol (SIP) call request from an Endpoint (EP). The method determines an external address based on an internal address of the EP uniquely mapped to the external address and pre-distributed to SIP internal routers. The method includes modifying the SIP call request to replace the EP internal address with the EP external address. The method includes forwarding the EP external address to the EP. The method includes establishing a communication link for human-interactive media between the EP and the call provider, the EP using the EP external address, and the communication link not including the SIP Internal Router. | 08-04-2011 |
| 20110188492 | RESPONDING TO CALL CONTROL EVENTS USING SOCIAL NETWORK APPLICATIONS - Embodiments of methods of handling call control events are provided. An example method includes receiving, at an interpreter, information indicating a call control event associated with a call from a calling party to a called party. The calling party and/or the called party are subscribed to the social network. The example method also includes providing, from the interpreter, messaging information generated by an application server based on the call control event and information retrieved from the social network for the calling party and/or the called party. | 08-04-2011 |
| 20110188493 | TECHNIQUE FOR COMMUNICATION COMMANDS AND PARAMETERS IN AN INFORMATION ASSISTANCE SYSEM TO PROVIDE SERVICES - A method for use in a system for providing an information assistance service includes receiving an information assistance call including an identifier from a caller and storing one or more parameters in association with the identifier of the call in a first device in the system. A desired information assistance service is elicited from the caller, associated with one or more of the parameters required to provide the requested service. A message is sent to a second device in the system including a directive in the form of a uniform resource locator (URL) addressed to a third device in the system for providing the desired information assistance service, the directive including the identifier. The second device disseminates the directive addressed to the third device, and in response to the directive, the third device uses the URL of said directive retrieving from the first device the one or more parameters based on the identifier in the directive. The desired information assistance service is provided by the second device based on the one or more parameters. | 08-04-2011 |
| 20110188494 | DYNAMIC INTELLIGENT DATA ROUTING APPARATUS AND METHOD - A novel method and system for an LCR engine, herein referred to as a Dynamic Intelligent Routing Engine (DIRE) is disclosed that optimizes in real time the routing of data on a communication network. The method and system includes novel hardware architecture and software where routing queries from telecommunication switching equipment is sent to the DIRE. The DIRE responds to the queries by providing an optimized list of termination vendors. The DIRE provides real time or near real time solutions by addressing issues pertaining to mixed and fixed costs routes, control margins, weighted routing parameters, quality routing and other selected information that may affect routing costs. | 08-04-2011 |
| 20110188495 | METHOD AND APPARATUS FOR ENABLING DUAL TONE MULTI-FREQUENCY SIGNAL PROCESSING IN THE CORE VOICE OVER INTERNET PROTOCOL NETWORK - The invention provides a method and apparatus for enabling DTMF signal processing in the core VoIP network. More specifically, the present invention enables a VoIP network to recognize and respond to special DTMF signals entered by a user and initiate the appropriate service logic response to satisfy the user's service request. | 08-04-2011 |
| 20110188496 | METHOD, TELEPHONE, TELECOMMUNICATION SYSTEM AND DEVICE FOR CONTROLLING POWER CONSUMPTION OF A TELEPHONE - The invention relates to a method for controlling the power consumption of a telephone ( | 08-04-2011 |
| 20110194553 | NON-VALIDATED EMERGENCY CALLS FOR ALL-IP 3GPP IMS NETWORKS - An emergency call in an all Internet Protocol (IP) network having GPRS access is able to be completed without a valid SIM. A valid Subscriber Identity Module is substituted for the missing or invalid SIM only when an emergency call is attempted. The emergency call is either sent via an IMSI from an embedded SIM provided by the UE making the emergency call, or the emergency call is modified with an IMSI substituted by an Emergency SIM Pool Function prior to being sent to the HLR for validation. The SIM is valid for the UE's emergency call so the emergency call is completed because the UE is considered validated by the network. | 08-11-2011 |
| 20110194554 | SYSTEMS AND METHODS FOR IMPLEMENTING CALL PICK UP USING GRUU AN IMS NETWORK - A method and system for providing call pickup service in an IP Multimedia Subsystem (IMS) communication network is described. When call pickup is to be invoked, an IMS network node, e.g., a core network node or an application server, receives a Globally Routable User Agent URI (GRUU) associated with one of a plurality of devices which is to be used to pick up a call that is in the process of being placed to another device in a call pickup group. The IMS network node determines whether the one of the plurality of devices is authorized to pick up the call based, at least in part, on the received GRUU. Then, the IMS network node transmits a message to establish the call with that one of the plurality of devices. | 08-11-2011 |
| 20110200033 | COMMON ROUTING - A method and corresponding apparatus are provided to route a call from a customer to a destination by: i) intercepting a call setup message sent from a customer switch intended to signal a switch to perform a call routing function or request a call routing function be performed, the call routing function determines a route for the switch to use to carry the call to the destination, the route so determined is a switch-determined route, ii) intercepting a call release message sent from the switch intended to signal the customer switch of network congestion, iii) in response to either the call setup message or the call release message being intercepted, querying a routing engine with the destination of the call for a specific route over which to carry the call to the destination, the specific route is queried from a set of routes that is different from an other set of routes from which the call routing function determines the switch-determined route, iv) modifying the call setup message to include the specific route, the call setup message so modified is a modified call setup message, v) responding to the call release message with a re-route call message that includes the specific route, and vi) directing the switch with either the modified call setup message or re-route call message to use the specific route to carry the call from the customer to the destination. | 08-18-2011 |
| 20110200034 | Systems and methods for voice and data communications including a scalable TDM switch/multiplexer - Integrated communications systems having a scalable or upgradable TDM switch fabric (i.e., e.g., TDM-controlling switch/MUX) are disclosed. At a first point in time a system is first sold, installed and utilized with a first TDM capacity, using a first TDM switch/MUX controlling a first set of TDM streams operating at a first frequency. A first set of line and other cards (e.g., DSP resources) are provided to provide or receive the first set of TDM streams. At a second point in time the system is upgraded by installation of a second TDM switch/MUX; the second TDM switch MUX controls the first set of TDM streams operating at the first frequency and also controls a second set of TDM streams operating at second frequency, which is a frequency different and preferably higher as compared to the first frequency. With at least some of the first cards coupled to the TDM bus, the second TDM switch/MUX couples TDM streams to and from the first cards using the first streams at the first frequency, while concurrently coupling TDM streams to and from the second cards using the second streams at the second frequency. The first switch/MUX preferably operates concurrently with the second switch/MUX to couple streams to and from the TDM bus (e.g., from an HDLC or multi-protocol framing engine, etc.), while the first switch/MUX does not operate to control the TDM bus, as this function is carried out by the second switch/MUX. | 08-18-2011 |
| 20110200035 | COOPERATIVE EXTERNAL CONTROL OF DEVICE USER INTERFACE USING SIP PROTOCOL - Implementations described herein provide the ability for a network telephony device or a computer application device to co-operatively control various user interface elements of another network telephony device that has user interface elements such as a screen, physical and/or touch-screen buttons, and illuminated indicators. Upon a VoIP communication session being set up between two devices, one device can co-operatively modify user interface elements presented on the other device. In response to user input actions on a telephony network device with user interface elements, response messages are sent back to the other device within this communication session via VoIP DTMF responses. To maximize the end user interaction experience, the controlling device can specify to the recipient device what DTMF key responses to send when any non-dialpad keys are pressed. | 08-18-2011 |
| 20110200036 | Private Branch Exchange, VoIP Gateway Unit and Private Branch Exchange System - This private branch exchange includes a park portion capable of parking a call put through to a telephone terminal unit via a terminal connection portion and a control portion controlling the park portion, when a nuisance call from the public telephone network is put through to the telephone terminal unit via the terminal connection portion, to park the nuisance call on the basis of a prescribed operation of the user against the nuisance call. | 08-18-2011 |
| 20110200037 | AUTOMATED VOICE OVER IP DEVICE VLAN-ASSOCIATION SETUP - A system and method are disclosed for automatically registering various system attributes with a VoIP device such as an VoIP phone. The system attributes are provided by a network, preferably an adjacent switching device made aware of the system attributes through one of a number of learning mechanisms. The system attributes may include one or more of the following: the VLAN identification used for VoIP communications in the subnet in which the VoIP phone is connected; the switching device identification, switching device slot, and switching device port number to which the VoIP phone is connected. The switch, slot, and port are used in some embodiments by an IP PBX system to construct a relational database that associates the geographic location of the connection with the IP phone for purposes of reporting the physical location of the VoIP user to emergency response personnel. The system and method for automatically registering various system attributes enables the relational database to be updated prompt and accurate. | 08-18-2011 |
| 20110206036 | SYSTEM AND METHOD FOR METHOD FOR PROVIDING AN INDICATION OF CERTAINTY OF LOCATION OF ORIGIN OF AN INTERNET PROTOCOL EMERGENCY CALL - A method for providing an indication of certainty of location of origin of an internet protocol emergency call including: (a) routing the emergency call from an originating internet protocol calling instrument via an internet protocol telephone service provider unit to a call taker; the provider unit having a provider telephone number; the emergency call being accompanied by an internet address for the originating calling instrument; (b) in no particular order: (1) looking up the internet address in a data base to ascertain a first data element; and (2) looking up the provider telephone number in a data base to ascertain a second data element; (c) comparing the first and second data elements; (d) if the first and second data elements match, continuing handling the emergency call; and (e) if the first and second data elements do not match, presenting an alert to the call taker. | 08-25-2011 |
| 20110206037 | Proxy Media Service for Digital Telephony - A Session Initiation Protocol (SIP) service system includes a SIP-enabled soft switch at a telephony service provider, executing code from a coupled machine-readable medium, routing SIP transactions to remote destinations, a media server coupled to the SIP-enabled soft switch storing media including ring tones and music-on-hold for use in progressing transactions, and an interface to a wide-area-network (WAN) for transmitting transactions and media. | 08-25-2011 |
| 20110206038 | Digital Telecommunications Call Management and Monitoring System - The present invention discloses a centralized, digital, computer-based telephone call management system for authenticating users of a telephone system in an institutional facility. The system includes the capacity to allow an institution to control, record, monitor, and bill and report usage and access to a telephone network. The telephone call management system further includes both accounting and management software for use in controlling, monitoring, billing, recording, and reporting usage and access. Also, it can operate over both a Public Switch Telephone Network (PSTN) and a Voice over Internet Protocol (VOIP) infrastructure. | 08-25-2011 |
| 20110206039 | Systems And Methods For IP And VoIP Device Location Determination - A method and system for precise position determination of general Internet Protocol (IP) network-connected devices. A method enables use of remote intelligence located at strategic network points to distribute relevant assistance data to IP devices with embedded receivers. Assistance is tailored to provide physical timing, frequency and real time signal status data using general broadband communication protocols. Relevant assistance data enables several complementary forms of signal processing gain critical to acquire and measure weakened or distorted in-building Global Navigation Satellite Services (GNSS) signals and to ultimately extract corresponding pseudo-range time components. A method to assemble sets of GNSS measurements that are observed over long periods of time while using standard satellite navigation methods, and once compiled, convert using standard methods each pseudo-range into usable path distances used to calculate a precise geographic position to a known degree of accuracy. | 08-25-2011 |
| 20110206040 | SYSTEMS, METHODS, AND COMPUTER PROGRAM PRODUCTS FOR PROVIDING A MANUAL RING-DOWN COMMUNICATION LINE USING SESSION INITIATION PROTOCOL - Systems, methods, and computer program products are provided for manual ring-down communication using Session Initiation Protocol (SIP). A first SIP user agent transmits a message to a second SIP user agent over an Internet Protocol (IP) network to establish a SIP session. The first SIP user agent determines that a signal key associated with a first communication device has been selected and transmits, to the second SIP user agent over the IP network, a start event message to cause a second communication device to activate an alert. The first SIP user agent determines that the signal key has been released and transmits over the IP network an end event message to deactivate the alert. The first SIP user agent transmits, to the second SIP user agent over the IP network, one or more subsequent INVITE messages at a predetermined repetition rate to refresh the SIP session. | 08-25-2011 |
| 20110206041 | Method For Transmitting Data In a Telecommunications Network And Switch For Implementing Said Method - A method for transferring data from a first switch to a second switch selectively by line-switching or by packet-switching as well as to a switch for carrying out the method. Data packets are thereby first transferred packet-switched through a packet-switching network to the second switch. With the presence of a corresponding control signal a line-switching connection is established from the first switch to the second switch and the data are then transferred through this connection. Where applicable a renewed changeover to a packet-switching transfer is carried out. A flexible packet-switching or line-switching data transfer linked with dynamic costs between the junctions of a telecommunications network is enabled. | 08-25-2011 |
| 20110211572 | CALLER ID CALLBACK AUTHENTICATIONI FOR VOICE OVER INTERNET PROTOCOL ("VOIP") DEPLOYMENTS - Systems and methods are disclosed for authenticating caller identification in VoIP communication. A VoIP device receives an incoming call from an originating calling device; wherein the incoming call includes (1) a caller identification and (2) a unique identifier associated with the originating calling device. The VoIP device verifies that the caller identification in the received incoming call matches an entry in a trusted directory, wherein the trusted directory includes one or more entries of previously verified caller identifications. Upon verifying that the caller identification in the received incoming call matches a caller identification entry in the trusted directory, the VoIP device sends an inquiry to a unique locator associated with the matching caller identification in the trusted directory. | 09-01-2011 |
| 20110211573 | INTEGRATED CIRCUITS, SYSTEMS, APPARATUS, PACKETS AND PROCESSES UTILIZING PATH DIVERSITY FOR MEDIA OVER PACKET APPLICATIONS - In one form of the invention, a process of sending real-time information from a sender computer ( | 09-01-2011 |
| 20110211574 | METHOD, SYSTEM AND APPARATUS FOR SESSION ASSOCIATION - A session association method, system, and apparatus are disclosed. The method includes: receiving an Internet Protocol Connectivity Access Network (IP-CAN) session setup message and a gateway control session message; and associating an IP-CAN session with a gateway control session according to a temporary identity (ID) in the IP-CAN session setup message and the temporary ID in the gateway control session message. Therefore, the gateway control session is associated with the IP-CAN session by using a temporary ID; and the gateway control session is associated with the IP-CAN session when no user ID exists in the emergency service, which ensures the normal progress of the emergency service. | 09-01-2011 |
| 20110211575 | METHOD AND APPARATUS FOR CONTROLLING TELECOMMUNICATION SERVICES - Method and apparatus in a user terminal ( | 09-01-2011 |
| 20110216759 | METHOD FOR PUBLISHING, QUERYING AND SUBSCRIBING TO INFORMATION BY A SIP TERMINAL IN A VoIP NETWORK SYSTEM, SIP TERMINAL, SIP APPLICATION SERVER, SIP INFORMATION CENTER AND VoIP NETWORK SYSTEM - The invention provides method for publishing, querying, subscribing to information by a SIP terminal in a VoIP network system, a SIP terminal, a SIP application server, a SIP information center and the VoIP network system. Wherein the VoIP network system is deployed with SIP information center for storing and providing at least the information. The method for publishing information by a SIP terminal in a VoIP network system comprises: creating a publishing request with the information to be published embedded in at the SIP terminal; sending the publishing request from the SIP terminal to the SIP information center via the SIP application server; recording the information in the SIP information center's database; and notifying the new information update to the subscribed SIP terminals. | 09-08-2011 |
| 20110216760 | SYSTEM AND METHOD FOR WEIGHTED MULTI-ROUTE SELECTION IN IP TELEPHONY - Systems and methods can be used for selecting eligible egress routes for IP telephony termination when multiple eligible carriers exist. The method uses any of a variety of factors to determine eligible egress routes, including for example, cost, eligibility and carrier relationship as determining factors. The disclosed method includes returning multiple routes for route advancement to a next most preferred carrier in the event of preferred carrier failure. | 09-08-2011 |
| 20110216761 | System And Method Of Communicating A Priority Indication In A Call Control/Bearer Control Telecommunication System - The present invention relates generally to telecommunication services, and in particular, to communicating priority indications between telecommunication nodes in a telecommunication system having a separated call control and bearer control architecture. The present invention provides a number of solutions which map or assign the call level priority to the bearer level. | 09-08-2011 |
| 20110216762 | METHODS, SYSTEMS, AND COMPUTER READABLE MEDIA FOR PROVIDING E.164 NUMBER MAPPING (ENUM) TRANSLATION AT A BEARER INDEPENDENT CALL CONTROL (BICC) AND/OR SESSION INTIATION PROTOCOL (SIP) ROUTER - The subject matter described herein includes methods, systems and computer readable media for providing E.164 number mapping (ENUM) translation at a bearer independent call control (BICC) and/or session initiation protocol (SIP) router. One aspect of the subject matter described herein includes a system for providing ENUM translation. The system includes an ENUM database. The system also includes a signaling router for receiving a bearer independent call control (BICC) signaling message that includes a first call party identifier, for obtaining, from the ENUM database, a first SIP address associated with the first call party identifier, for generating a first SIP signaling message that includes the first SIP address, and for routing the first SIP signaling message to a destination SIP node. | 09-08-2011 |
| 20110216763 | Tel URI Handling Methods and Apparatus - A method of installing a default dialing or numbering plan identity into a user terminal comprising an IMS client. The method comprises, at or following registration of a user of the user terminal to an IMS network, receiving at the terminal from the network a dialing or numbering plan identity and saving the identity as a default identity at the terminal, wherein the default dialing or numbering plan identity is subsequently used by the IMS client as a phone context. | 09-08-2011 |
| 20110216764 | PACKET COMMUNICATION SYSTEM - The packet communication system enables communication between a communication unit connected in a conventional telephone network and a communication unit in a packet communication network. A device for exclusively selecting the connection network is provided so that, at the time of transmission of a signal from one communication unit to another, the connection path is selected according to the kind of the network to which the other communication unit belongs. At the time of signal reception, the communication unit is connected to only one of the conventional telephone network and the packet communication network. The packet communication system includes a packet processor for converting an information signal, such as speech, inputted from an input unit (for example, a transmitter microphone of a handset) into the form of a packet | 09-08-2011 |
| 20110216765 | MEDIA EXCHANGE NETWORK SUPPORTING MULTIPLE BROADBAND NETWORK AND SERVICE PROVIDER INFRASTRUCTURES - A method for communicating information includes establishing a logical communication path that is independent of a physical communication path that couples at least two end points via at least a first broadband network. At least a first portion of the logical communication path and at least a second portion of the logical communication path utilize different communication protocols. Both of the physical and logical communication paths are established through the same plurality of network nodes. Information that would be normally transferred over the physical communication path may be transferred between the at least two endpoints, via the established logical communication path over the corresponding redundant network connection. The established logical communication path may be provisioned for handling communication functions. The communication functions may include operations administration maintenance and provisioning (OAM&P), roaming, user authentication, media transfer, caching, storage management and/or addressing management. | 09-08-2011 |
| 20110216766 | SESSION INITIATION PROTOCOL (SIP) MESSAGE INCORPORATING A MULTI-PURPOSE INTERNET MAIL EXTENSION (MIME) MEDIA TYPE FOR DESCRIBING THE CONTENT AND FORMAT OF INFORMATION INCLUDED IN THE SIP MESSAGE - A system and method for processing a plurality of requests for multi-media services received at a call control element (CCE) defined on the system from a plurality of IP-communication devices. The system includes at least one Network Routing Element (NRE), a Service Broker (SB), a media sever, a plurality of application servers (ASs) and a plurality of border elements, all of which are coupled to the CCE. The CCE is adapted to receive requests for multi-media services and to generate subsequent requests for the multi-media services, which are communicated to the SB for processing. The subsequent requests can each include a Session Initiation Protocol (SIP) message including a message identifier portion having at least a first predetermined information field and a second predetermined information field. The message identifier portion of the SIP message declares the content and format of the SIP message to a recipient device defined on the system. | 09-08-2011 |
| 20110222529 | METHOD AND SYSTEM FOR STORING SESSION INFORMATION IN UNIVERSAL PLUG AND PLAY TELEPHONY SERVICE - A method, system and apparatus are provided for storing session information in a home network of an UPnP telephony service. The method is performed at a Telephony Server (TS). The method receives a request from a Telephony Control Point (TCP) to store session information while the session is in progress. The session information includes a session status and session related media. The method then divides the session information into meta information and session control information. Thereafter, the method stores the session information in a memory of the TS. | 09-15-2011 |
| 20110222530 | System and method for transmitting a telephone call over the Internet - A method and system for transmitting a call in a client/server architecture. A client device initiates a telephone call and converts first analog voice signals associated with the telephone call to digital signals. The digital signals are then transmitted over the Internet to a first gateway server. The first gateway server processes the digital signals using a codec algorithm and transmits the processed digital signals over the Internet to a second gateway server. The second gateway server converts the processed digital signals to second analog voice signals and transmits the second analog voice signals over a public switched telephone network. | 09-15-2011 |
| 20110222531 | voIP ACCESSORY - An accessory for electronic equipment includes an interface for exchanging data between the accessory and the electronic equipment, and a voice over internet protocol (VoIP) circuit. The VoIP circuit is operatively configured to implement at least a portion of VoIP in the electronic equipment or the accessory. | 09-15-2011 |
| 20110222532 | Routing A Call Setup Request To A Destination Serving Node In An IMS Network - A method of routing a call setup request to a destination serving node serving a destination subscriber in an IMS network. The method comprises the steps of a switching node receiving the call setup request having a destination number of the destination subscriber, the switching node querying a number conversion database node for destination routing information using the destination number. The method further comprises the steps of the number conversion database node querying subscriber information for destination serving node information related to the destination number, the number conversion database node receiving destination serving node information from the subscriber information, the number conversion database node replying to the switching node with destination routing information comprising the destination serving node information and the switching node routing the call set up request to the destination serving node using the destination serving node information. | 09-15-2011 |
| 20110228760 | Method and System for Find Me/ Follow Me in IMS Through Editing of IMS Registrations at S-CSCF - A method for operation of a Serving Call Session Control Function (S-CSCF) server is provided. The method includes storing a plurality of records, each record corresponding to a respective one of a plurality of Internet Protocol Multimedia Services (IMS) terminals associated with a user, the plurality of records indicating an order of the IMS terminals for attempting to establishing a communication link with the user. The method also includes receiving an input from an application server indicating a request to change at least one of the records to indicate a different order, and changing the record. A system for operation of a Serving Call Session Control Function (S-CSCF) server, and a computer-readable medium having stored thereon computer-executable instructions, the computer-executable instructions causing a processor to perform a method for operation of a Serving Call Session Control Function (S-CSCF) server when executed, are provided. | 09-22-2011 |
| 20110228761 | COMMUNICATION SYSTEM AND CONTROL SERVER - When an IP terminal on the Internet side transmits a name resolution request, which requests to resolve the FQDN of a public server connected to a router, to a SIP/DNS server in which SIP and DNS cooperate to manage the status of the data line of the router and when the data communication line status of the router is a disconnect status, a data communication line connection request instruction is transmitted to the router. The router connects the data communication line and notifies the SIP/DNS server about the result. The SIP/DNS server transmits the IP address of the router to the IP terminal as a response. | 09-22-2011 |
| 20110228762 | Telephone System, Telephone Exchange Apparatus, and Connection Control Method Used in Telephone Exchange Apparatus - According to one embodiment, a telephone exchange apparatus includes a connector, a memory, a determination module and a controller. The connector performs a part of the function of the media server and connect a distribution server that distributes an input media packet to the telephone terminals. The memory stores a connection management table indicating a correspondence relation between a terminal ID, a server ID and codec information, when a call connection related to the unicast packet distribution is established. The determination module refers to the connection management table when performing the unicast packet distribution, and determines whether there is a call connection using the same codec information based on reference result of the connection management table. The controller dynamically connects a plurality of telephone terminals using the same codec to a media server via the distribution server. | 09-22-2011 |
| 20110228763 | METHOD AND APPARATUS FOR ACCESSING SERVICES OF A DEVICE - A telephony device is provided including a display, a processor, and a network interface for communicating via a communications network. The telephony device runs a browser application, a server application, and a service application including services using the processor. The processor controls the server application to recognize a first command request that is provided to the server application using a first transfer protocol via the communications network, and provides instructions to the service application in response to the first command request. The processor controls the service application to control a function of the telephony device by executing a first set of commands, generate a first content, and provide the first content to the server application. The processor controls the server application to generate a first response to the first command request, the first response including the first content. | 09-22-2011 |
| 20110228764 | INTEGRATION OF AUDIO INPUT TO A SOFTWARE APPLICATION - The present invention concerns a method and a system for integrating voice inputs to a three-dimensional virtual application software, such as a game, on at least one computer during the execution of said virtual application software on said at least one computer, said system comprising: voice receiving means for receiving a voice input signal real-time sound streaming means, such as an application programming interface (API), receiving at least one external voice input from a user, wherein said voice audio input is encoded to an intermediate output voice sound stream; three-dimensional software application means comprising means adapted for subjecting said intermediate output voice sound stream data to predetermined software application logic defined in the application software, including identifying the game state of the user in the application software; output audio data stream generating means for manipulating output voice stream data and any activity related application software generated sounds, wherein the voice stream data with any activity related application selected sound are sampled and manipulated to the intermediate output voice data stream in accordance with the game state by selecting one or more predefined environmental sound effect; and processing means for routing the output audio data stream to a sound processor card on at least one recipient computer. Hereby, there is provided a method and a system which integrates voice inputs to a three-dimensional virtual application software, such as a game, wherein the game logic processes all sound inputs, i.e. both the predetermined game state selected sounds and the voice inputs so that the voice input is user-specifically played. Hereby, a user of the game will get an audio experience which is fully integrated with the game state. | 09-22-2011 |
| 20110228765 | TELEPHONY TERMINAL - Methods and apparatus implementing a telephony terminal for connecting a telephone to a data network. In one implementation, a telephony system includes: a phone connection for connecting to a telephone; a network connection for connecting to a network; and a controller connected to said phone connection and to said network connection; wherein said controller provides a phone service for processing information for said phone connection, said controller provides a network service for processing information for said network connection, and said controller provides a network voice service for converting information to and from a network voice format. | 09-22-2011 |
| 20110228766 | LOOP CONDITION PREVENTION AT INTERNETWORK INTERFACE BOUNDARY - The present invention provides a solution to maximize the chance of completion for an ISUP to SIP direction call by enabling a bigger factor for converting ISUP hop counter to SIP Max-Forwards value than the reverse direction thus enabling more hops in the SIP network. Enabling a bigger factor for ISUP to SIP direction can cause loops without special considerations. This invention provides an algorithm that prevents a “loop condition” that can arise at the interface boundary of two telephone networks, known by their standard names ISUP and SIP networks. The present invention solves the “loop condition” problem by adjusting the Hop Counter and Max-Forward parameter values in a predetermined manner such that the adjusted parameter values break the cycle of providing the same parameter values between networks at the network boundary for an uncompleted connection, or break an endless “loop condition”. | 09-22-2011 |
| 20110235630 | Techniques for prioritizing traffic - Techniques, at a subscriber station, for assigning packets to queues to prioritize real-time content over non-real time content. Packets with the same connection identifier are assigned to different priority queues. Block sequence numbers are assigned to packets after storage of packets to queues based on priority. | 09-29-2011 |
| 20110235631 | METHOD AND APPARATUS FOR AUTOMATIC VERIFICATION OF TELEPHONE NUMBER MAPPING - The present disclosure provides mechanisms for verification of mapping from one type of network address to another type of network address based on delivery of a one-time key over one type of the network and confirmation of its receipt over another type of network. A particular example of such mapping is mapping from a telephone number used in the PSTN or the like to a VoIP address such as a SIP URI. The mapping verification mechanisms can be provided without dependence on the records of past calls, manual calling, or the line information database in the PSTN system. Accordingly, a highly secure and efficient mapping verification mechanism is realized. | 09-29-2011 |
| 20110235632 | Method And Apparatus For Performing High-Quality Speech Communication Across Voice Over Internet Protocol (VoIP) Communications Networks - A communications terminal device and a method performed by a communications terminal device wherein packet data received from a Wireless Personal Area Network (WPAN) headset (such as, for example, a Bluetooth headset), which comprises an encoded audio signal, is directly convened by the terminal device to Internet Protocol (IP) packets which are transmitted across a Voice over Internet Protocol (VoIP) communications network, wherein speech encoding is not performed by the terminal device. Similarly, a communications terminal device and a method performed by a communications terminal device wherein IP packet data comprising an encoded audio signal is received from a VoIP communications network by the terminal device, and is directly converted by the terminal device to WPAN packets (such as, for example, Bluetooth protocol packets) which are transmitted to a WPAN headset (such as, for example, a Bluetooth headset), wherein speech decoding is not performed by the terminal device. | 09-29-2011 |
| 20110235633 | SYSTEMS AND METHODS FOR PROVIDING 9-1-1 SERVICES TO NOMADIC INTERNET TELEPHONY CALLERS - A system for facilitating 9-1-1 service delivery to internet telephony customers is provided. The system includes a server device for receiving a 9-1-1 call from a user device via a data network, where the 9-1-1 call are based on “9-1-1” digits dialed at the user device. The server device is configured to forward the received 9-1-1 call to an operator services interface operatively connected to the server device. | 09-29-2011 |
| 20110243123 | Noise Reduction During Voice Over IP Sessions - An approach is provided that, upon receiving a keyboard event, reduces a volume of an audio input channel from a first volume level to a lower volume level. After the volume of the audio input channel is reduced, the approach waits until a system event occurs, with the system event based at least in part on the occurrence of a nondeterministic event. The volume of the audio input channel is then increased from the lower volume level to a higher volume level when the system event occurs | 10-06-2011 |
| 20110243124 | METHOD AND APPARATUS FOR MANAGING A NETWORK - A method and an apparatus for managing a network are disclosed. For example, the method collects a plurality of call detail records (CDRs), and organizes one or more parameters of the CDRs in accordance with a plurality of cause codes. The method displays the one or more parameters of the CDRs in a hierarchical representation comprising a plurality of screen displays. | 10-06-2011 |
| 20110243125 | COMMUNICATION USING A USER TERMINAL - Provided is a method of communicating using a user terminal that comprises: a first interface for exchanging call data with a first interface of a mobile communication device, wherein the mobile communication device comprises a second interface for interfacing with a node of a mobile telecommunications network, and wherein the first interface of the mobile communication device is unsuitable for interfacing with a node of a mobile telecommunications network; a second interface for exchanging call data with a second user terminal over a packet-based communication network; and a processor for executing a communications client, which processor is coupled to the first interface of the user terminal and to the second interface of the user terminal and is configured to participate in a call with the second user terminal via the second interface of the user terminal and the packet-based communication network; wherein the method comprises: sending call data via one of the first interface of the user terminal and the second interface of the user terminal during the call, on the basis of call data received via the other of the first interface of the user terminal and the second interface of the user terminal. | 10-06-2011 |
| 20110243126 | Call Handling for IMS registered user - The present invention proposes a solution for providing IMS services to users having circuit-switched controlled terminals. In particular, it is proposed, in order to allow IMS to take the full call and service control, to combine circuit-switched and packet-based multimedia functionality in a new node type called Mobile Access Gateway Control Function (MAGCF). In particular the present invention provides a method for ensuring that the MAGCF node acts as a roaming anchor point in order to enforce the handling of originating and terminating calls in the IMS. | 10-06-2011 |
| 20110243127 | VOICE AND DATA EXCHANGE OVER A PACKET BASED NETWORK - A signal processing system which discriminates between voice signals and data signals modulated by a voiceband carrier. The signal processing system includes a voice exchange, a data exchange and a call discriminator. The voice exchange is capable of exchanging voice signals between a switched circuit network and a packet based network. The signal processing system also includes a data exchange capable of exchanging data signals modulated by a voiceband carrier on the switched circuit network with unmodulated data signal packets on the packet based network. The data exchange is performed by demodulating data signals from the switched circuit network for transmission on the packet based network, and modulating data signal packets from the packet based network for transmission on the switched circuit network. The call discriminator is used to selectively enable the voice exchange and data exchange. | 10-06-2011 |
| 20110243128 | COMBOPHONE WITH QoS ON CABLE ACCESS - A method of providing QoS to a session from a client to a first network includes providing data packets from the client to be conveyed in a session from the client to a first network, inserting each of the data packets into an encapsulating packet, and transmitting the encapsulating packets through the second network to the first network, forming a tunnel through the second network. The method includes receiving the encapsulating packets at a terminating device in the first network. The terminating device removes the encapsulating headers to recover the data packets. The method includes determining an association between the packet headers of the data packets and the encapsulating headers, identifying data packets requiring QoS, and using the association to identify encapsulating packets corresponding data packets requiring Quality of Service. The method includes applying QoS to the encapsulating packets, corresponding to the session of data packets requiring the QoS, being conveyed through the tunnel. | 10-06-2011 |
| 20110243129 | Multi-Mode Endpoint in a Communication Network System and Methods Thereof - A method, apparatus, and communication network system that allows an endpoint to be simultaneously registered with more than one communications server is described. In one embodiment, the communication network system includes a network, a plurality of communications servers that are coupled to the network, and a plurality of endpoints coupled to the network. Each endpoint is capable of being simultaneously registered with more than one communications server. A communication method for an endpoint involves registering a first logical line of the endpoint with a first communications server, and registering a second logical line of the endpoint with a second communications server. Consequently, flexibility is obtained by allowing an endpoint to choose the registering communications server for each logical line of the endpoint. | 10-06-2011 |
| 20110249666 | LOCATION BASED ROUTING - A method may include receiving a session initiation protocol (SIP) Invite message associated with a call and determining that the call involves location based processing. The method may also include identifying location information associated with the call based on header information included in the SIP Invite message and identifying a location ID based on the location information. The method may further include modifying the SIP Invite message to include the location ID, identifying a call type associated with the call and identifying a mobile switching center to which the SIP Invite message is to be forwarded based on the call type and the location information. | 10-13-2011 |
| 20110249667 | APPARATUS AND METHOD FOR TRANSMITTING MEDIA USING EITHER NETWORK EFFICIENT PROTOCOL OR A LOSS TOLERANT TRANSMISSION PROTOCOL - A method and apparatus for transmitting voice media over a network where the voice media may be consumed either in a real-time mode or a time-shifted mode. The method comprising transmitting the voice media over the network using a network efficient protocol when either (i) the media is not being consumed in the real-time mode or (ii) the condition on the network is good enough to support the real-time transmission and consumption of the voice media in the real-time mode. Alternatively, the voice media is transmitted using a loss tolerant transmission protocol when the media is being consumed in the real-time mode and the condition on the network is sufficiently poor to prevent the real-time consumption of the voice media in real-time using the network efficient protocol. The apparatus, which may be a communication device or a server, implements the above-described method. | 10-13-2011 |
| 20110249668 | Opportunistic Multitasking - Services for a personal electronic device are provided through which a form of background processing or multitasking is supported. The disclosed services permit user applications to take advantage of background processing without significant negative consequences to a user's experience of the foreground process or the personal electronic device's power resources. To effect the disclosed multitasking, one or more of a number of operational restrictions may be enforced. By way of example, inactive network applications (e.g., VOIP applications) may be placed in a suspended state until a message is received targeting the application (e.g., an incoming phone call or a heartbeat needed message). The user application may be placed into the background state to respond to the message and then returned to the non-active state (e.g., if the message was a heartbeat needed) message or to the foreground state if appropriate (e.g., the user elects to answer the incoming call). | 10-13-2011 |
| 20110249669 | METHOD FOR SERVICE INTER-WORKING AND SESSION CHANNEL ESTABLISHMENT, INTER-WORKING SELECTION FUNCTION MODULE AND DEVICE - A method for service inter-working and session channel establishment, an Inter-working Selection Function (ISF) module and device are provided. The method for service inter-working includes: receiving content which is corresponding to at least two media types and is sent by a calling party; and sending, according to the media types of the content, the content to Inter-Working Function (IWF) modules each corresponding to one media type, so that the IWF modules send the received content to a called party. In one aspect, a Converged IP Message (CPM) service may have sessions with a plurality of non-CPM services, thereby improving user experience; in another aspect, when the calling party intends to change the media types between the calling party and the called party with which the calling party has processed the service inter-working, it can be achieved just by directly changing the media types without disconnecting the session with the called party in advance. | 10-13-2011 |
| 20110249670 | EXECUTING A COMMUNICATION CONNECTION - According to one embodiment, a server apparatus for executing communication connection between a first terminal being connected to a first communication network, and a second terminal being connected to a second communication network, includes a memory, an acquisition module and a controller. The memory stores an electric quantity table and a media determining table. The acquisition module acquires remaining information of the battery from the first terminal. The controller refers to the electric quantity table and the media determining table to select a communication media based on the acquired remaining amount information of the battery and consumed electricity by the first terminal. When a request for the first terminal is received, controller assigns the selected communication media to the first terminal, when the battery remaining amount of the first terminal varies during communication, the controller re-determines to change the communication media. | 10-13-2011 |
| 20110249671 | System and Method for Computer Originated Audio File Transmission - A system and method for computer originated audio file transmission includes a server having a communications module operable to communicate with a terminal unit. The server may also include a storage module operable to store at least one file. A processor may be provided to separate the file into a plurality of packets. In accordance with one embodiment of the present invention, the communications module is operable to send an initial burst of packets to the terminal unit, wherein the initial burst of packets includes at least two of the plurality of packets. In accordance with another embodiment of the present invention, the communications module is further operable to send additional packets of the plurality of packets at a predetermined rate, until each of the plurality of packets has been sent to the terminal unit. | 10-13-2011 |
| 20110255529 | Conversion System and Method in Multioperator Environment - The invention relates to a method of performing signalling and media conversion in a multioperator environment. The invention comprises receiving a signalling from a first operator; detecting a second operator of the signalling; checking information of the second operator from a database; carrying out at least one conversion to the signalling from the first operator; and transmitting the signalling to the second operator. | 10-20-2011 |
| 20110255530 | SYSTEM AND METHOD FOR PROVIDING ENTERPRISE VOICE CALL CONTINUITY - An improved system and method are disclosed for providing voice call continuity in an enterprise network. For example, an enterprise public branch exchange (PBX) may be configured with a pilot number that is used to provide VCC services when called by a client. Digit collection via DMTF signaling or other means may be used to collect destination information from the client. The enterprise network may use the collected digits to establish a communication session with another device that corresponds to the destination information. | 10-20-2011 |
| 20110255531 | Setting Up A Call From A Non-IMS To An IMS Network Whereby The Gateway Interfaces The HSS - The application relates to a method for setting up a call from a non-IMS telecommunication network, comprising a Network Gateway Node (NGN), to a destination node in an IMS network. The method comprises the NGN interfaces a combined database node comprising a Home Location Register (HLR) and a Home Subscriber Server (HSS). The method further comprises routing the call to the destination node in the IMS network, of which address is determined by information received from the combined database node. The method further comprises sending, sending, in response to receiving an initial call setup request message, an information request message to the combined database node for obtaining routing information for the setup of the call, the information request message comprising an indicator indicating at least one type of response that the NGN is able to process. | 10-20-2011 |
| 20110255532 | Packet-Switched Telephony Call Server - A system and method for providing packet-switched telephony service. The system provides call control, signaling, and/or delivery of voice, video, and other media in substantially real time. One embodiment of the system includes a call client application on a user device, and a call server located at a packet-switched telephony service provider. The call server is preferably operable to communicate with the call client in a non-native protocol and with the gateway in a native protocol. | 10-20-2011 |
| 20110261807 | MONITORING INMATE CALLS USING SPEECH RECOGNITION SOFTWARE TO DETECT UNAUTHORIZED CALL CONNECTING - A system and method for managing and controlling telephone activity in a correctional facility comprises providing a first communicative connection between a caller and a recipient, delivering the conversation between the caller and the recipient over the first communicative connection and executing speech recognition software to identify a plurality of conversation words delivered over the first communicative connection. By comparing the conversation words with a database of trigger words, a determination can be made as to whether the recipient is attempting to create an unauthorized call connection. Based on that comparison step, a detection response is executed. | 10-27-2011 |
| 20110261808 | Server Apparatus and DTMF Notification Method - According to one embodiment, a server apparatus includes a detector and a controller. The detector detects a dual tone multi frequency (DTMF) notification event sent from a communication partner server apparatus in a state that voice communication is performed between telephone terminals by way of a plurality of server apparatuses. The controller executes at least one of a change of a payload type included in a header of a received communication packet and a replacement of a payload data of the communication packet with a DTMF data, based on a detection result by the detector. | 10-27-2011 |
| 20110261809 | Method of and a Network Server and Mobile User Equipment for Providing Chat/VoIP Services in a Mobile Telecommunications Network - A method of and an application server ( | 10-27-2011 |
| 20110261810 | NOTIFICATION METHOD AND GATEWAY FOR ACCESSING A VOICE OVER IP NETWORK - A method is implemented by an access gateway to a voice over IP network, connected to a plurality of terminals associated with one and the same telephonic identifier in said network, and comprises, on detection by said gateway of a switch to a connected or disconnected state of a call established between an external terminal and a terminal of said plurality of terminals, transmitting to at least one other terminal of said plurality of terminals a message comprising information relating to said switch of said call to said connected or disconnected state. | 10-27-2011 |
| 20110268105 | SYSTEMS AND METHODS FOR PROVIDING TELEPHONY AND PRIVATE BRANCH EXCHANGE SERVICES VIA AN ETHERNET ADAPTER - The present application is directed towards systems and methods for providing telephony and private branch exchange services via a single device installed as an Ethernet adapter on a computing device. A device, based around a standard form factor such as a PCI card, with a CPU, operating system, and memory may be installed in a server or other computing device and utilize power from the computing device while operating independently. The device combines both Ethernet adapters, bridges, and switches, and circuit-switched telephone network and private branch exchange switches and ports to act as a bridge between packet-based and circuit-based networks. | 11-03-2011 |
| 20110268106 | CLEARINGHOUSE SERVER FOR INTERNET TELEPHONY AND MULTIMEDIA COMMUNICATIONS - A clearinghouse server for routing multi-media communications, including telephony calls, between a source device and a destination device via a distributed computer network, such as the global Internet. The clearinghouse server can authorize the completion of a communication from a source device to a destination device and collect usage-related information for the completed communication. In response to an authorization request issued by an enrolled source device, the clearinghouse server can identify one or more available destination devices available to accept a communication from an authorized source device. The clearinghouse server can provide a list of the identified destination devices, typically organized in a rank order, by sending an authorization response to the source device. In turn, the source device can use this list to select a destination device and contact that selected device via the computer network to complete the communication. | 11-03-2011 |
| 20110268107 | Registry Proxy Server Apparatus, Communication System, and Operation Mode Changing Method - According to one embodiment, a registry proxy server apparatus includes a connector, a register controller, a monitor and a change controller. The connector connects a plurality of IP telephone servers registering a plurality of IP telephone terminals and including a normal mode and an ecology mode. The register controller determines an arbitrary IP telephone server and allows the determined server to perform registry a IP telephone terminal as request source. The monitor monitors respective operational states of the plurality of IP telephone terminals and the plurality of IP telephone servers. The change controller changes an operation mode of at least one of the plurality of IP telephone servers based on a monitoring result obtained by the monitor. | 11-03-2011 |
| 20110268108 | Backplane Interface Adapter with Error Control and Redundant Fabric - A backplane interface adapter with error control and redundant fabric for a high-performance network switch. The error control may be provided by an administrative module that includes a level monitor, a stripe synchronization error detector, a flow controller, and a control character presence tracker. The redundant fabric transceiver of the backplane interface adapter improves the adapter's ability to properly and consistently receive narrow input cells carrying packets of data and output wide striped cells to a switching fabric. | 11-03-2011 |
| 20110268109 | COMMUNICATION TERMINAL DEVICE, COMMUNICATION SYSTEM, AND COMMUNICATION CONTROL METHOD - The first communication unit | 11-03-2011 |
| 20110268110 | Providing Packet-Based Multimedia Services via a Circuit Breaker - A packet-based multimedia service is provided to a terminal in a network. A packet signaling connection is established between the terminal and the network. Signaling information for the multimedia service is transferred via the packet signaling connection using Session Initiation Protocol (SIP) or a similar protocol. A circuit bearer connection is also established with the terminal. Data for the multimedia service is transferred via the circuit bearer connection. This allows the data to be carried across networks which do not support the required QoS functionality for the packet-based service, or which cannot efficiently carry packet-based data. The circuit bearer connection can be established by a network entity or by the terminal. The circuit bearer can be interworked to a packet-switched bearer at some point in the network, such as at a gateway, so as to provide a remote party with the appearance that a fully packet-switched connection is being used. | 11-03-2011 |
| 20110274103 | NETWORKING APPARATUS AND TELEPHONY SYSTEM - A networking apparatus including a PBX that relays between and manages IP and analog telephones that are assigned extension number information has: a VoIP conversion portion which converts an analog audio signal into an IP signal; an IP connection portion to which an IP telephone is connected; and an analog connection portion to which an analog telephone is connected. When a call originated from an IP or analog telephone is relayed and connected to an analog telephone, extension number information of the call originator is output to the analog telephone. | 11-10-2011 |
| 20110274104 | VIRTUAL AREA BASED TELEPHONY COMMUNICATIONS - A persistent virtual area that supports establishment of respective presences of communicants operating respective network nodes connected to the virtual area even after all network nodes have disconnected from the virtual area is maintained. A presence in the virtual area is established for a user of a Public Switched Telephone Network (PSTN) terminal device. Transmission of data associated with the virtual area to the PSTN terminal device. | 11-10-2011 |
| 20110280239 | COMMUNICATION SESSION HAND-OFF METHOD AND COMMUNICATION DEVICE - A communication session hand-off method of a communication device includes establishing a first communication session over a first network between a first communication interface of the communication device and a communication interface of another communication device, monitoring a predetermined condition of the communication device by the communication device, and based upon the predetermined condition, sending a request by the communication device to the another communication device for the another communication device to establish a second communication session over a second network between a second communication interface of the communication device and the communication interface of the another communication device. | 11-17-2011 |
| 20110286443 | SYSTEM, APPARATUS AND METHOD FOR ROAMING IN DECT-VOIP NETWORK - A system, an apparatus and related method for roaming in DECT-VoIP network are provided. The system for roaming in DECT-VoIP network includes at least a DECT-VoIP handset, and at least a DECT-VoIP apparatus. The system can be connected to the Internet through connecting the DECT-VoIP apparatus to appropriate internet access device, such as, ADSL modem or IP PBX so as to make and receive phone calls through VoIP. With presetting the VoIP client account information at the DECT-VoIP handset, the DECT-VoIP handset can roam from the coverage of a first DECT-VoIP apparatus previously registered with to the coverage of a second DECT-VoIP apparatus sharing the same ID with the first DECT-VoIP apparatus without interruption to the connection. | 11-24-2011 |
| 20110286444 | Method and Apparatus for Optimizing Response Time to Events in Queue - A system for optimizing response time to events or representations thereof waiting in a queue has a first server having access to the queue; a software application running on the first server; and a second server accessible from the first server, the second server containing rules governing the optimization. In a preferred embodiment, the software application at least periodically accesses the queue and parses certain ones of events or tokens in the queue and compares the parsed results against rules accessed from the second server in order to determine a measure of disposal time for each parsed event wherein if the determined measure is sufficiently low for one or more of the parsed events, those one or more events are modified to a reflect a higher priority state than originally assigned enabling faster treatment of those events resulting in relief from those events to the queue system load. | 11-24-2011 |
| 20110286445 | Method and apparatus for controllling telephone calls using a computer call assistant - Systems and methods for monitoring, making, managing and controlling telephone communications with a computer call assistant with an integrated voice/data communications system are disclosed. A call assistant computer application preferably runs on a personal computer (“PC”) coupled to the integrated system over a packet bus. The call assistant exchanges control and/or status packets with the integrated system preferably over a packet bus. The call assistant enables the user to make, receive and control telephone calls, monitor the status of the user's extension, voice mail, etc., and preferably operates with integrated systems capable of transmitting and receiving voice and data in multiple modes. In preferred embodiments, the computer call assistant operates with systems that are capable of multiple native mode voice and data transmissions and receptions with a communications system having a multi-bus structure, including, for example, a time division multiplexed (“TDM”) bus, a packet bus, and a control bus, and multi-protocol framing engines, preferably including subsystem functions such as PBX, voice mail, file server, web server, communications server, telephony server, LAN hub and data router. | 11-24-2011 |
| 20110286446 | Method and Apparatus for Use in an IP Multimedia - According to a first aspect of the present invention there is provided a method of routing a call from an IP Multimedia Subsystem network to a circuit switched network. The method comprises, at a circuit switched carrier select node of the IP Multimedia Subsystem network, receiving a session establishment request (S | 11-24-2011 |
| 20110292928 | METHOD, MODEM AND SERVER FOR BRIDGING TELEPHONE CALLS INTO INTERNET CALLS - The present invention relates to a method for bridging traffic in analogue channel into digital channel using Asymmetrical Digital Subscriber Line, said method comprises: step of PSTN network connecting, in which caller and callee ADSLs establish PSTN network connection using PSTN signaling in the analogue channel; step of discovering Internet call, in which the caller ADSL sends Internet call setup message to Internet call server, the caller ADSL and the callee ADSL make Internet call discovery procedure on the Internet and determine successful Internet call discovery; step of setting up Internet connection, in which the caller and callee ADSLs set up Internet connection in the digital channel by means of the successful Internet call discovery; step of bridging the PSTN network connection to the Internet, in which the caller and the callee ADSLs bridge the PSTN network connection to the Internet via the Internet connection which has been set up, and release the analogue channel. The invention further relates a modem and a Internet call server used in the method. | 12-01-2011 |
| 20110292929 | SIGNALLING MESSAGES IN A COMMUNICATIONS NETWORK NODE TO COMMUNICATE A CALLED ADDRESS STRING - Communications network node ( | 12-01-2011 |
| 20110292930 | Method And System For Communicating Across Telephone And Data Networks - A method and system for communicating across telephone and data networks are disclosed. According to one embodiment, a computer-implemented method, comprises receiving a first call from a first user phone that converts the first call from a format of a first local phone network to a first digital call. The first digital call is transmitted over a large area data network. The first digital call is converted to a format of a second local phone network to generate a second call. The second call is transmitted to a second user phone over the second local phone network. A real-time bi-directional voice communication session is established between the first user phone and the second user phone. | 12-01-2011 |
| 20110299522 | APPARATUS, METHOD AND SYSTEM FOR PROVIDING NEW COMMUNICATION SERVICES OVER EXISTING WIRING - The invention provides apparatus for providing a next-generation communication system over existing wiring. In one form the apparatus includes an input to receive broadband signals carrying next-generation communication data, a processor to extract the next-generation communication data from the broadband signals and a converter to convert the next-generation communication data into analogue telephone signals. The apparatus is arranged to output the analogue telephone signals at the input of the apparatus. Also described is a related method of providing a next-generation communication system over existing wiring. | 12-08-2011 |
| 20110299523 | Correlating Information Between Internet and Call Center Environments - Coordination of information at the network-based level between call centers connectable over a telecommunications network, such as a telephone network, and a packet network, creates improved integration of and bonding between a customer's interaction with a Web site and with a call center. Information about the customer and the customer's Web interaction are delivered to the call center agent along with the call, leading to increased productivity and efficiency in call handling and improved call routing. Calls may be routed to existing call centers based upon information from the Web experience, and information from the user's Web interaction is shared with the call center. Web interaction information is passed to existing call centers using known call center external control methods, such as DNIS signaling. Information about the Web experience may also be “whispered” to the call center agent, and an agent may “push” Web pages for review by the customer. | 12-08-2011 |
| 20110299524 | INTEGRATED INFORMATION COMMUNICATION SYSTEM - A communication system, for functioning without the use of dedicated lines or the Internet so as to ensure communication speed, communication quality, and communication trouble countermeasures, including a communication network and domain name server. The domain name server includes a domain name tree with a country number of a telephone number as a level 2 domain name of the domain name tree, and the domain name server receives, from a terminal, a telephone number of a destination terminal. Furthermore, based on the telephone number of the destination terminal, the domain name server (i) seeks out, in the domain name tree, an Integrated Information Communication System (ICS) user address of the destination terminal, and (ii) sends the ICS user address to the terminal, such that the communication system receives, from the terminal, the ICS user address as a destination address, and sends the ICS user frame to the destination terminal. | 12-08-2011 |
| 20110305238 | COMMUNICATION APPARATUS FOR HOSTED-PBX SERVICE - A communication apparatus is provided for relaying an IP telephony service provided by a PBX server to extension telephones. The communication apparatus includes: a memory unit which stores a user-ID/Subscriber definition table in which a user-ID preassigned to each of the extension telephones is associated with a user password and a subscriber number of the PBX server which is available by the user-ID; and a portal controller which has a Web server function for providing the PCs with a portal service, and processes portal data from the PCs and the PBX server. The portal controller determines, in accordance with a user-ID and a user password which are entered from the PC, a subscriber number to be used by the user-ID, records an access state to the portal service into a portal service access table, and associates the extension telephone that corresponds to the user-ID with the portal service. | 12-15-2011 |
| 20110310884 | TELEPHONY ENDPOINT ROUTING IN AN IP MULTIMEDIA SUBSYSTEM - The present invention provides a mechanism whereby a session established at a particular MGCF between a subscriber of a legacy network and a first user of a referral service can be replaced by another session to be established between the subscriber of the legacy network and a second user of the referral service without disturbing the sessions already established. Therefore, the present invention provides a new method and devices for generating an IMS Telephony Endpoint, which includes a first information field with a telephony Universal Resource Identifier identifying an originating or destination subscriber, as the case may be, and a second information field with information usable to identify the generating MGCF as the particular MGCF holding a session for such served subscriber. Such ITE is submitted towards the IMS and is included in a further referral service request involving such session so that the particular MGCF may be identified. | 12-22-2011 |
| 20110310885 | QUALITY OF SERVICE (QOS)-ENABLED VOICE-OVER-INTERNET PROTOCOL (VOIP) AND VIDEO TELEPHONY APPLICATIONS IN OPEN NETWORKS - A device defines a first bucket for general Internet protocol (IP) traffic provided to and from a user device associated with an open network, and defines a second bucket for quality of service (QoS)-based traffic provided to and from the user device. The device also assigns a first billing rate for the general IP traffic associated with the first bucket, and assigns a second billing rate to the QoS-based traffic associated with the second bucket, where the second billing rate is greater than the first billing rate. The device further associates the first billing rate and the second billing rate with a subscriber associated with the user device. | 12-22-2011 |
| 20110310886 | SYSTEM AND METHOD FOR TRANSFERRING A CALL BETWEEN ENDPOINTS IN A HYBRID PEER-TO-PEER NETWORK - An improved system and method are disclosed for peer-to-peer communications. In one example, the method enables an endpoint to move (e.g., transfer or forward) a call to another endpoint in a peer-to-peer environment. | 12-22-2011 |
| 20110310887 | CABLE MODEM AND METHOD OF SUPPORTING VARIOUS PACKET CABLE PROTOCOLS - A cable modem in communication with a trivial file transfer protocol (TFTP) server includes a default system and a backup system employing various packet cable protocols. The cable modem boots with the default system, obtains a vendor specific information (VSIF) from the TFTP server, and determines if the VSIF matches with the default system. The cable modem continuously boots with the default system if the VSIF matches with the default system. The cable modem configures the currently default system to be the backup system and configures the currently backup system to be the default system, and reboots with the newly configured default system if the VSIF mismatches with the default system. | 12-22-2011 |
| 20110310888 | Methods and Apparatuses for Handling Public Identities in an Internet Protocol Multimedia Subsystem Network - The present invention concern methods and apparatuses for handling public identities in an Internet Protocol Multimedia Subsystem, IMS, network. A Serving Call Session Control Function, S-CSCF, node receives a message including a public identity. If the message does include a profile key with a wildcarded identity, the S-CSCF uses the wildcarded identity received with the profile key to fetch a user/service profile related to the wildcarded identity. If the message does not include a profile key with a wildcarded identity, the S-CSCF | 12-22-2011 |
| 20110310889 | Methods and Apparatuses for Handling Public Identities in an Internet Protocol Multimedia Subsystem Network - The present invention concern methods and apparatuses for handling public identities in an Internet Protocol Multimedia Subsystem, IMS, network. A CSCF node receives information indicating a set of distinct public identities, within a range of a wildcarded public identity, which set of distinct public identities are not in the same Implicit Registration Set (IRS) as the wildcarded public identity. The information is received from a Home Subscriber Server (HSS) node. The CSCF node stores the information indicating the set of distinct public identities, which are not in the same IRS, in the CSCF node for allowing matching of an originating request to the information indicating the set of distinct public identities, which are not in the same IRS. Alternatively, the CSCF node forwards the information indicating the set of distinct public identities, which are not in the same IRS, to another CSCF node, for allowing matching of an originating request to the information indicating the set of distinct public identities, which are not in the same IRS. | 12-22-2011 |
| 20110310890 | COMMUNICATION DEVICE - A communication device configured to be connected with both a public switched telephone network and an IP network. The communication device may comprise an input allowing unit configured to allow a user to input specific identification information for the public switched telephone network, a judging unit configured to judge whether or not the communication device itself is in a specific state that is capable of executing a first communication process of communicating via the IP network using IP identification information for the IP network, and a communication unit configured to execute the first communication process, in a first case where the communication device is judged as being in the specific state, and execute a second communication process of communicating via the public switched telephone network, in a second case where the communication device is judged as not being in the specific state. | 12-22-2011 |
| 20110310891 | MERCHANT POWERED CLICK-TO-CALL METHOD - A method is disclosed for enhancing the predictability, scalability and cost effectiveness of online advertising with voice over IP connectivity and event tracking technologies. A service provider maintains a list of merchants who have offered to pay for customer VoIP calls to their establishment. The service provider maintains a real time connection with this merchant list and renders an advertisement in a distinguishing way in real time. A potential customer who views this advertisement on a web page may establish a VoIP call session with a merchant by selecting a free click-to-call link on the web page. When the customer places the call, the service provider pays for the call. Merchants in turn pay the service provider for displaying the ads that generated the calls on a price per call, price per impression or fixed fee basis. | 12-22-2011 |
| 20110317684 | SYSTEMS AND METHODS FOR TERMINATING COMMUNICATION REQUESTS - A IP telephony service allows customers to form user groups. Each user group can include multiple telephony devices that are associated with one or more users. One or more group identifiers would be associated with each user group. When an incoming communication is directed to a user group, a group identifier is used to retrieve a list of the members of the group, or a list of devices that correspond to the members of the user group. The communication is then sent to one or more members of the group, or to one or more of the devices that correspond to members of the user group. Handling preferences may determine how the incoming communication is delivered. In some instances, the incoming communication could be a telephone call. In other instances, the incoming communication could be a SMS message or an instant message. | 12-29-2011 |
| 20110317685 | CONSOLIDATED VOICEMAIL PLATFORM - A voicemail system for providing voicemail services to a secure facility. An embodiment of the voicemail system includes an internet router provided at a facility for communicating with a call processing center that is located outside the facility. A database at the call processing center stores voicemail messages, a call interface receives and stores voicemail messages for residents of the facility, a resident interface provides a plurality of residents of the facility with access to the stored voicemail messages via a telephone located at the facility, and a web server provides a plurality of authorized users access to the stored voicemail messages via a website. | 12-29-2011 |
| 20110317686 | SYSTEMS AND METHODS OF ESTABLISHING USER GROUPS IN AN INTERNET PROTOCOL ENVIRONMENT - A IP telephony service allows customers to form user groups. A user group is established by collecting a plurality of identifiers that are associated with members of a group, and associating the member identifiers with a group identifier. The member identifiers could be telephone numbers of telephony devices for each of the members, or device IDs of IP telephony devices for members of the group. The group identifier could be any type of identifier, and in some instances, the group identifier could be a telephone number. | 12-29-2011 |
| 20110317687 | SYSTEMS AND METHODS OF FORWARDING COMMUNICATION REQUESTS BASED ON HANDLING INSTRUCTIONS IN AN INTERNET PROTOCOL ENVIRONMENT - A IP telephony service allows customers to form user groups. Each user group can include multiple members, each of whom have their own telephony device. When a member of a user group sends an outgoing communication from one of the telephony devices associated with the user group, the service obtains communication handling instructions for the user group. The outgoing communication is then processed in accordance with the handling instructions. This could include sending copies of the outgoing communication to the telephony devices of other members of the user group. This could also include sending the outgoing communication with an origination identifier associated with the user group, rather than an origination identifier associated with the member's telephony device. | 12-29-2011 |
| 20110317688 | Dynamic Federations - Systems and methods of establishing IP telephony sessions between enterprises are disclosed. A first enterprise requests an association with a second enterprise. Both enterprises and the second enterprise belong to the same federation. The association request is accepted, to establish an association between the first and second enterprises. In response to the acceptance, a direct routed path is established between the first enterprise and the second enterprise. One of the associated enterprises requests activation of an IP telephony service. If the request to activate references the association, an IP telephony session is established using the direct routed path. | 12-29-2011 |
| 20110317689 | Service Path Routing Between Session Border Controllers - Systems and methods of establishing IP telephony sessions between enterprises are disclosed. A first enterprise requests an association with a second enterprise. Both enterprises and the second enterprise belong to the same federation. The association request is accepted, to establish an association between the first and second enterprises. In response to the acceptance, a direct routed path is established between the first enterprise and the second enterprise. One of the associated enterprises requests activation of an IP telephony service. If the request to activate references the association, an IP telephony session is established using the direct routed path. | 12-29-2011 |
| 20110317690 | SYSTEM AND METHOD FOR VOICE-ACTIVATED DIALING OVER IMPLICIT AND EXPLICIT NFA TRUNKS - A system and method for voice-activated dialing using implicit and explicit trunks including receiving a call from the user telephone and establishing a first connection. In response to establishing the first connection, a second connection may be established over the implicit trunk. In response to establishing the second connection, a third connection may be initiated. In response to the user telephone sending a keyword, the implicit trunk may be disconnected and the call may be connected via the explicit trunk. If a spoken number is received, then the spoken number may be translated into a computer readable telephone number. Alternatively, if a dialed telephone number is received from the user telephone, the telephone number may be used to route the call. In response to receiving the telephone number, the explicit trunk may be disconnected and the call from the user telephone may be routed to the received telephone number. | 12-29-2011 |
| 20120002663 | CONTROLLING TELEPHONE CALL PROCESSING USING GLOBAL SIGNALING CODES - In general, embodiments of the present invention involve attaching (e.g., pre-fixing) a Global Signaling Code (GSC) to a called party's telephone number thereby creating a modified Uniform Resource Indicator (URI). This modified URI is then sent in the “TO:” header of a SIP INVITE. The GSC will typically include a geographic indicator corresponding to a geographic location of a caller and a treatment indicator corresponding to a desired treatment of the call. The call will be routed based on the geographic indicator and treated according to the treatment indicator. Illustrative treatments for the call include (among others) voice mail avoidance, a preferred compression scheme for the call, etc. | 01-05-2012 |
| 20120002664 | SYSTEM AND METHOD FOR CALLING ADVERTISED TELEPHONE NUMBERS ON A COMPUTING DEVICE - Advertisers submit advertising campaigns to a call advertising system, each campaign including a telephone number associated with the advertiser and an amount per call that the advertiser is willing to pay to receive calls from users at the advertised telephone number. When an advertised telephone number is contained in content that is displayed on a computing device of a user, or when a user enters a telephone number that corresponds with the advertised telephone number, the computing device highlights the advertised telephone number to indicate to the user that the call can be made at a free or reduced rate to what a user would normally pay in order to make such a call. If the user initiates a call to the advertised telephone number, the call is routed to the advertiser and the advertiser is charged the amount that they agreed to pay to receive the call. | 01-05-2012 |
| 20120002665 | Telephone Exchange Apparatus and Telephone Terminal and a Control Method Used for a Telephone System - According to one embodiment, a telephone exchange apparatus includes a communication processor, a memory and a controller. The communication processor establishes a communication session between a plurality of telephone terminals on the private network and a telephone terminal on the global network through a common port specifying the private network. The memory stores a management table which associates a terminal ID specifying a telephone terminal to be connected, a session ID specifying a communication session, and an address and port ID specifying a network to be connected to the telephone terminal, for each session. The controller refers to the management table based on a session ID included in a communication packet, and sends instruction data to the communication processor to effect communication by the communication packet between telephone terminals establishing a communication session, based on a reference result of the management table. | 01-05-2012 |
| 20120002666 | Method for Extending Ethernet over Twisted Pair Conductors and to the Telephone Network and Plug-In Apparatus for Same Employing Standard Mechanics - An Ethernet extension device is provided for metro or last mile Ethernet service via twisted pairs as opposed to fiber optics. The Ethernet extension device is implemented as a plug-in extension for existing infrastructure (e.g., in a standard electrical wall box or Type-200™ Mechanics card) that employs lighting and power cross protection required by the telephone companies for Ethernet connectivity to the telephone network (e.g., for connection between a user's building and a telephone company building over existing outdoor telephone cables). | 01-05-2012 |
| 20120008619 | DIFFERENTIATION OF MULTIPLE MEDIA ENDPOINTS BEHIND AN ADDRESS TRANSLATION DEVICE - In one embodiment, two way communication between an IP phone behind a firewall and an IP phone behind a translation device is established. A network security device receives a remote packet from the translation device. The header of the remote packet includes the address of the translation device, and a payload of the remote packet includes an embedded remote address and the media port of the IP phone behind the translation device. A memory stores the media port matched with the address of the translation device. When the network security device receives a local packet from the IP phone behind the firewall destined for the IP phone behind the translation device, a controller rewrites the destination port of the local packet with the media port. | 01-12-2012 |
| 20120008620 | CONNECTION ARRANGEMENT - A plurality of inputs are configured to receive circuit switched traffic from a plurality of initiators. A plurality of outputs are configured to output said traffic to a network on chip. Each output is associated with a different quality of service traffic. A traffic controller directs the received circuit switched traffic to respective ones of the outputs in dependence on a quality of service associated with the traffic. | 01-12-2012 |
| 20120008621 | COMMUNICATING IN VOICE AND DATA COMMUNICATIONS SYSTEMS - A data and voice communication system includes communication between a line card and an accelerator card. Voice, data, and control traffic is received from the line card and is transmitted to the accelerator card via a physical link having separate voice, data, and control logical channels. The separate voice, data, and control logical channels are represented by labeled data packets. | 01-12-2012 |
| 20120014374 | Method, Device, and Computer Program Product for Adaptive Routing of Communications Across One or More Networks - Communications are adaptively routed across at least one network. Responsive to a dynamic user request for routing of communications at a particular service level, available routes within the network are determined for routing the communications. Route quality characteristics are determined for routes within the network for routing the communications. A route for routing the communications for the user is determined based on the available routes, the route quality characteristics, and the particular service level requested by the user. | 01-19-2012 |
| 20120014375 | Method for Telephone Connection Preservation - A method is provided in which a server is placed in the communication flow between a first communication terminal and a second communication terminal. The server executes connection preservation software which operates to keep the second communication terminal connected when the first communication terminal shuts down its communication software (e.g., VoIP application, etc.). Specifically, the connection preservation software transmits signaling that causes the connection to be transferred from the first communication terminal to a communication network node and vice versa. | 01-19-2012 |
| 20120014376 | METHOD AND SYSTEM FOR A GIGABIT ETHERNET IP TELEPHONE CHIP WITH 802.1P AND 802.1Q QUALITY OF SERVICE (QOS) FUNCTIONALITIES - A method for processing data may include receiving packetized data via at least one of a plurality of input ports in an Ethernet switch. Each of the plurality of input ports may be partitioned into a plurality of virtual local area network (VLAN) port domains with an assigned port domain identification (ID) for processing 802.1 Class of Service (CoS) priority and Quality of Service (QoS) packetized data. The Ethernet switch may be integrated within a single gigabit Ethernet IP telephone chip, the received packetized data having assigned at least one priority class. One or more bits in at least one of a plurality of registers in the Ethernet switch may be used to filter at least one ingress frame in the packetized data, based on at least one packet header attribute of the at least one ingress frame. | 01-19-2012 |
| 20120014377 | DISTRIBUTED PACKET-BASED TIMESTAMP ENGINE - A system handles timing information within a packet-switched network. The system classifies packets for processing depending on the packet type. After classification, a new timestamp value may be produced depending on the packet classification. The new timestamp value may use a timestamp value from the received packet, a value from a local clock, and an offset value. The timestamp value may be written into the packet, depending on the packet classification, and checksum-type fields may additionally be updated in the packet. In some embodiments, multiple physical layer circuits are integrated with a local clock circuit. | 01-19-2012 |
| 20120014378 | METHOD AND APPARATUS FOR ASSESSING VoIP PACKET PRODUCTION AND PACKET TRANSMISSION AND INDICATION AT THE END POINTS INVOLVED IN THE VoIP COMMUNICATION - The present invention relates to a method and a device for assessing and indicating the quality of VoIP calls, comprising the steps of end-point reception of the VoIP packets over an IP network link, end-point determination of the VoIP quality (QRX) of the received VoIP packet sequence, the VoIP quality (QTX) of the transmitted VoIP packet sequence, exchange of the quality information (QRX and QTX) between the end points, calulation of the difference (QRX-QTX) between the received VoIP quality and the VoIP quality transmitted by the other side; supply of the determined VoIP quality information to a quality indication; and end-point indication of the quality information in optical and/or acoustic form. | 01-19-2012 |
| 20120014379 | DIFFERENTIATED PRIORITY LEVEL COMMUNICATION - Provided are methods, apparatuses and systems for providing prioritized data distribution at a customer premise. A network access component may receive priority information from a trusted source, the priority information being indicative of an association between at least one identifier and a respective priority level. The network component may determine a particular identifier associated with data received from a communication entity. The network access component may determine a particular priority level associated with the data based on the particular identifier and the priority information. The network component may also prioritize at least a portion of the data on a basis of the particular priority level. | 01-19-2012 |
| 20120014380 | Method and Apparatus of Informing a Network of Change of User Equipment Capability - An embodiment method of informing a network of a change of user equipment capability includes receiving, by a network, a register request message carrying information of new user equipment capability to from a user equipment, analyzing, by the network, the register request message, and storing the information of new user equipment capability for reference by subsequent establishment of a session, stopping a current registration timer on the server side set for the user equipment, initiating a new registration timer on the server side for the user equipment, and sending a response message carrying information of the new registration timer on the server side to the user equipment so as to reset a registration timer on the user side based on information of the new registration timer on the server side in the response message. The capability change is informed to the network in time. | 01-19-2012 |
| 20120014381 | VOICE SERVICE IN EVOLVED PACKET SYSTEM - Methods and apparatus to manage voice service in evolved packet systems are disclosed. An example method in a user equipment (UE) with a first indicator related to voice services in an Evolved Packet System (EPS), the method includes receiving a Non Access Stratum (NAS) protocol response message with a second indicator and responsive to the first indicator and the second indicator, sending a notification to at least one of a user or an upper layer that a CS domain is not available. | 01-19-2012 |
| 20120014382 | SYSTEMS AND METHODS FOR TERMINATING COMMUNICATION REQUESTS - A IP telephony service allows customers to form user groups. Each user group can include multiple telephony devices that are associated with one or more users. One or more group identifiers would be associated with each user group. When an incoming communication is directed to a user group, a group identifier is used to retrieve a list of the members of the group, or a list of devices that correspond to the members of the user group. The communication is then sent to one or more members of the group, or to one or more of the devices that correspond to members of the user group. Handling preferences may determine how the incoming communication is delivered. In some instances, the incoming communication could be a telephone call. In other instances, the incoming communication could be a SMS message or an instant message. | 01-19-2012 |
| 20120014383 | Access Node Comprising VOIP Cards with Common IP/MAC Address - Disclosed are methods and arrangements in a VoIP access node for handling downstream and upstream RTP packets associated with an ongoing VoIP call. One common IP address can be assigned to the all of the RTP traffic of an access node instead of assigning one IP address per VoIP card, This may be done by identifying the destination VoIP card based on the destination UDP port number in a received RTP packet. The number of public IP addresses needed for VoIP service may be reduced without increasing the cost of the access node. | 01-19-2012 |
| 20120020350 | APPLICATION SERVICE INVOCATION BASED ON FILTER CRITERIA - An Internet Protocol Multimedia Subsystem (IMS) device includes a memory configured to store a subscriber profile, where the subscriber profile includes at least one criterion relating to an event that occurs after a session request has been forwarded to a terminating party. The IMS device further includes a processor configured to invoke at least one application service for a session based on the at least one criterion in the subscriber profile. | 01-26-2012 |
| 20120020351 | CALL BARRING - A application relates to call barring. Typically, PBX systems have the ability to bar calls originating from both an individual phone and group of phones. However, with nomadism/hot-desking in large organisations comprising typically many PBXs, often from different manufactures, a PBX base call barring solution is unworkable. Consequently, a computer telephony based solution has been chosen. The call barring is performed on the CTI server ( | 01-26-2012 |
| 20120027006 | SYSTEM AND METHOD FOR JOINT VOICE AND DATA TRANSMISSION - A method for multiplexing voice and data communication using a communications network is presented. In the method, at least one voice bit from a voice bit stream is retrieved, and at least one data bit from at least one data bit stream is retrieved. When the voice bit stream is not in a discontinuous transmission (DTX) period, the at least one voice bit and the at least one data bit are used to generate a modulated symbol. The method includes transmitting the modulated symbol using the communications network. | 02-02-2012 |
| 20120027007 | CABLE MODEM AND NETWORK REGISTRATION METHOD - A network registration method of a cable modem boots the cable modem with a default packet cable (PC) protocol, detects the network server through a network when the cable modem is booted, and determines whether the network server has been detected. When the network server has been detected, the method registers the cable modem on the network server through the network. Additionally, the method reboots the cable modem with a backup PC protocol upon the condition that the network server has not been detected. | 02-02-2012 |
| 20120027008 | Addressing Techniques For Voice Over Internet Protocol Router - An apparatus and method for increasing available ports on a voice router is provided. A first gateway and a second gateway are assigned a single port number for a data stream, the direction of packet flow is identified to determine a destination gateway. The destination gateway is one of the first and second gateways, depending on the direction of the packet flow. The packets are then forwarded to the destination gateway. The voice router can further consolidate RTCP streams from a plurality of gateways into a single port on the voice router. | 02-02-2012 |
| 20120027009 | COMMUNICATION DEVICE AND METHOD THEREOF - A communication device electronically connects with a phone and includes a control unit, a relay, a public switched telephone network (PSTN) interface, a voice over Internet Protocol (VoIP) network interface, and a ring signal detector. The relay is electronically connected with the control unit. The PSTN interface and the VoIP interface are electronically connected with the relay. The ring signal detector is electronically connected with the control unit and the PSTN interface. The control unit controls the relay to connect the phone with the PSTN interface when the ring signal detector detects a PSTN ring signal. In one embodiment, the control unit controls the relay to connect the phone with the VoIP interface. | 02-02-2012 |
| 20120027010 | ALARM SYSTEM IP NETWORK WITH PSTN OUTPUT - Alarm customers on VoIP may use an adapter for conversion to Internet Protocol (IP) signals or may have an alarm system that uses IP signals to transmit alarm signals over the Internet. IP signals from alarm customers may go to any monitoring center for alarm system monitoring. IP signals from alarm systems using IP conversion equipment can go only to monitoring centers with specialized receiving equipment specific to the type of transmitting equipment in use at the customer's premises. There is a pool of customers, whose dealers would convert to IP and stay with the current monitoring center if the center invested in receiving equipment. For the many small centers who will not or cannot invest in receiving equipment, the present invention will take IP signals from any or all brands of IP transmitting equipment, to a central server then retransmit to any center over POTS to the alarm monitoring center. Thus, an alarm monitoring center need not invest in a number of different IP monitoring systems in order to be IP monitoring compliant. | 02-02-2012 |
| 20120027011 | INTELLIGENT FORMATTING OF VOIP TELEPHONE NUMBERS - A system and method are disclosed for intelligent formatting of VoIP telephone numbers. The intelligent VoIP formatting system includes a presentation and user interface layer, an E.164 formatting engine, a location routine, and a database of telephone number data which are used to deconstruct input telephone numbers and reconstruct them as E.164 compliant telephone numbers. | 02-02-2012 |
| 20120033661 | DISTRIBUTED IP-PBX SIGNAL PROCESSING - Techniques are described by which an IP telephone system leverages the digital signal processing functions of end-user IP telephones by distributing signal processing tasks typically carried out by a centralized IP-PBX. The end-user IP telephones publicize their signal processing capabilities and availabilities to an IP-PBX, which maintains a resource capability mapping of the IP telephones. When the IP-PBX receive a bitstream for a communication session involving IP telephones and/or legacy phones of the IP telephone system, the IP-PBX determines the signal processing requirements for the bitstream, selects an available, capable IP telephone to perform the requirements, and distributes the bitstream to the selected IP telephone. The IP telephone performs the requisite signal processing and returns the processed bitstream to the IP-PBX, which forwards the processed bitstream to the destination endpoint for the communication session. | 02-09-2012 |
| 20120044930 | Device initiated DQoS system and method - A Data-Over-Cable Service Interface Specification (DOCSIS) cable modem system is coupled to: i) via a local area internet protocol (IP) network, a voice over internet protocol (VoIP) device operating Session Initiation Protocol (SIP) for signaling a VoIP media session; and ii) via a DOCSIS network, a cable modem termination system (CMTS) via a network. The cable modem system comprises instructions stored in a memory and executed by a processor. The instructions comprise: i) in response to receiving a frame via the local area IP network, determining if the frame is a Session Initiation Protocol (SIP) invite message signaling a VoIP session with a remote endpoint; and ii) in response to determining that the frame is a SIP invite message, generating a DOCSIS message to the CMTS to request an addition of reserved bandwidth on the DOCSIS network for the VoIP session. | 02-23-2012 |
| 20120044931 | Via Site for Managing Network Bandwidth - A system for using one or more via sites to manage network bandwidth, according to one embodiment of the present invention comprises a first call manager at a source site receiving an offer message to connect a call. The offer message includes an endpoint media settings list. The first call manager determines a first filtered media preferences list based on a source media settings list and the endpoint media settings list. The first call manager and transmits an invite message to a second call manager at a first via site. The invite message includes the first filtered media preferences list. The first call manager receives from the second call manager a call settings list that includes a description of the call settings negotiated between the source site, the first via site and a destination site. | 02-23-2012 |
| 20120057589 | IP Telephone Set and IP Telephone System - This IP telephone set includes a CPU acquiring a MAC address of a personal computer when the same is capable of communicating with the personal computer, while a server previously registers an extension number corresponding to the IP telephone set. The CPU controls the server to register an extension number corresponding to the MAC address of the personal computer in association with the IP telephone set, thereby rendering the IP telephone set usable as a telephone set corresponding to the IP telephone set itself and a telephone set of the personal computer. | 03-08-2012 |
| 20120069837 | SYSTEM AND METHOD FOR PROVIDING A FATE SHARING IDENTIFIER IN A NETWORK ENVIRONMENT - A method is provided in one example embodiment and includes receiving a first portion of an identifier associated with a communication session involving a first endpoint that generated the first portion of the identifier. The method also includes receiving a second portion of the identifier associated with the communication session involving a second endpoint that generated the second portion of the identifier. The method can further include communicating a reservation request associated with a network resource to be allocated for the communication session. The reservation request includes the first portion and the second portion of the identifier. | 03-22-2012 |
| 20120069838 | METHOD AND APPARATUS FOR A BLUETOOTH-ENABLED ETHERNET INTERFACE - In one embodiment, a method includes determining when a relay arrangement is available to pair with an endpoint. The relay arrangement is arranged to wirelessly communicate with the endpoint and to communicate over a wired network. The method also includes authenticating the endpoint with respect to the relay arrangement when the relay arrangement is available to pair with the endpoint, and pairing the endpoint with the relay arrangement if the endpoint is authenticated with respect to the relay arrangement. Pairing the endpoint with the relay arrangement includes the endpoint and the relay arrangement engaging in wireless communications, as well as the relay arrangement engaging in wired communications over the wired network. | 03-22-2012 |
| 20120069839 | METHOD AND SYSTEM FOR NETWORK SWITCH ELEMENT - Method and system for a network switch element is provided. The switch element includes a plurality of megaports, each megaport uniquely identified by a unique megaport address identifier for network addressing. Each megaport includes a plurality of operational ports, each operational port identified by a unique operational port address identifier. The switch element also includes a local crossbar for communication between the plurality of operational ports, and a shared logic module configured to provide common control of the plurality of operational ports within a megaport to allow operational ports to share resource of a single megaport to route network packets there between. The switch element also includes a global crossbar configured to allow communication between the megaports. | 03-22-2012 |
| 20120076128 | SYSTEM AND METHOD FOR ANONYMOUS WEB CALLING CHARGING ADVICE - Methods and systems are presented for anonymous web call processing, in which an application web server prompts a calling party to initiate a web call, queries a charging server for a tariff rate associated with the anonymous web call to the specified called party, renders the tariff rate to the calling party, and prompts the calling party to confirm initiation of the anonymous web call to the specified called party. | 03-29-2012 |
| 20120076129 | AUTOMATIC USER REDUNDANCY DETERMINATION - The embodiments presented herein provide an automated process for provisioning a user in a communication system. A session manager, which can be a server in the communication system that provides call connection and routing, may receive registration request from communication device (e.g. a cellular telephone, an IP-enabled phone, etc.). The session manager may determine one or more characteristics about the communication device and/or determine a load on one or more other session managers in a cluster of session managers. Based on both the communication device characteristics and/or the loads on the two or more session managers, the session manager can determine a set of session managers, which may include a primary session manager and a secondary session manager, which can manage the user data for the communication device. This session manager set information may then be sent to the communication device and to other session managers in the cluster. The set of session managers may then manage the user data for the communication device. | 03-29-2012 |
| 20120076130 | CABLE MODEM AND PRIORITY SETTING METHOD THEREOF - A cable modem and a priority setting method thereof receive signaling packets from telephone interfaces and local area network (LAN) interfaces, and preset the signaling packets from telephone interfaces as VoIP packets with a first priority and the signaling packets from the LAN interfaces as LAN packets with a second priority. The cable modem and the priority setting method pick out RTP packets from the VoIP and LAN packets, and classify the RTP packets as VoIP RTP packets and LAN RTP packets. The cable modem and the priority setting method determine whether all of the IP phones are on-hook, and update the LAN RTP packets from the second priority to the first priority in response to all of the IP phones being on-hook. The cable modem and the priority setting method allocate bandwidths for the signaling packets corresponding to the first priority and the second priority. | 03-29-2012 |
| 20120076131 | METHODS, SYSTEMS AND COMPUTER READABLE MEDIA FOR SUPPORTING A PLURALITY OF REAL-TIME TRANSPORT PROTOCOL (RTP) MULTIPLEXING ENABLEMENT METHODS IN A MEDIA GATEWAY - Methods, systems, and computer readable media for supporting a plurality of RTP multiplexing enablement methods in a media gateway are disclosed. According to one aspect, a system includes a media gateway controller configured to receive a signaling message to establish a call from a peer node in a first network, an a media gateway configured to perform a first and second RTP multiplexing enablement process on the media portion of the call, wherein the first RTP multiplexing enablement process includes sending a real-time transport control protocol (RTCP) Application packet to the peer node, refraining from performing RTP multiplexing on packets sent to the peer node until a reply RTCP Application packet is received from the peer node, and receiving RTP multiplexed packets from the peer node, and wherein the second RTP multiplexing enablement process includes obtaining an Internet protocol (IP) realm identifier that identifies the first network, accessing configuration data in the media gateway using the IP realm identifier to determine a predefined multiplexing enablement action for packets destined for the first network, and executing the determined predefined multiplexing enablement action on packets sent to the peer node in the first network. | 03-29-2012 |
| 20120076132 | SWITCH AND TELEPHONE SEARCHING METHOD - A switch includes a call record table recording at least one outgoing call. When the switch receives an incoming call, the switch searches the call record table to find a mapping outgoing call with a destination telephone number the same as a source destination telephone number of the incoming call. Then the switch obtains a communication line number of the mapping outgoing call in the call record table, and sends a ringing signal to a local telephone corresponding to the obtained communication line number. | 03-29-2012 |
| 20120076133 | System for Interconnecting Standard Telephony Communications Equipment to Internet - Apparatus include a communication predelivery receiver, configured to communicate with and relay both IP addressed overhead and IP addressed payload data from an originating communication device. The communication predelivery receiver includes a screener configured to, when the communication predelivery receiver is in communication with the originating communication device, receive certain data including first IP addressed data and second IP addressed data of corresponding first and second initiated communications. The communication predelivery receiver is configured and connected within a network so the first IP addressed data and the second IP addressed data are processed differently. For the first initiated communication, the first payload data is converted, after the point in time at which the screener received the first IP addressed data, from IP addressed first payload data to another network signal. For the second initiated communication, the second payload data is delivered without so converting the second payload data. | 03-29-2012 |
| 20120076134 | CITIZENS' EMERGENCY NETWORK SYSTEM AND METHOD - A system and method to provide an emergency communication network including a first responder device and at least one recipient by using a local agency communication node configured to relay communications using, in part, a VoIP interface to retransmit a communication received at a local agency as a ham communication received by at least one ham radio communication node which may be sent to and received by one or more recipients though a VoIP interface over the Internet. | 03-29-2012 |
| 20120076135 | IP TELEPHONE PROVISIONING SYSTEM AND METHODS - A method and apparatus relate to receiving and storing data through an interface for an Internet Protocol (IP) telephone database. The interface receives a request related to provisioning a function of an IP telephone. In response to receiving the request, data from the IP telephone database may be transmitted to the IP telephone to be used to provision the function. | 03-29-2012 |
| 20120076136 | METHODS AND APPARATUS TO PROVIDE A CALL-ASSOCIATED CONTENT SERVICE - Methods and apparatus to provide a call-associated content service to voice over Internet protocol (VoIP) devices are disclosed. An example method includes performing a telephone number database query at a call session control function server to obtain a first uniform resource identifier for a destination and to obtain a second uniform resource identifier for the destination, establishing a first communication session to the destination based on the first uniform resource identifier, and initiating a second communication session on behalf of a calling device to the destination based on the second uniform resource identifier, the destination to provide content associated with the first communication session via the second communication session. | 03-29-2012 |
| 20120076137 | Configuring Guest Users for a VoIP Device of a Primary User - A method to providing VoIP telephony to a plurality of users of a VoIP network includes authenticating and registering a primary user to a VoIP device to support call signaling associated with the primary user. A guest user subscriber of the VoIP network having a guest user directory number is authenticated in response to a password. The guest user directory number is mapped to the unique hardware identifier of the VoIP device, and a second set of VoIP parameters are downloaded from the VoIP network to the VoIP device in order to support call signaling associated with the guest user. VoIP calls are then within the VoIP network in response to the mappings of the primary user directory number and the guest user directory number to the same unique hardware identifier. | 03-29-2012 |
| 20120076138 | METHOD AND APPARATUS FOR ROUTING DATA - A method and apparatus for handling internet access telephone calls made via cable company telephone services. A head end data terminal receives cable signals and converts them into individual signals. An intelligent switch detects signals destined for an internet service provider and routes those signals on a separate path to the internet service provider. A central switch routes the other signals along a telephone network. A computer program can control the steps of receiving cable signals, converting them into voice band signals, routing the signals that are not for the intended recipient to a central switch, multiplexing the signals for the intended recipient together, and sending the multiplexed signals to the intended recipient. | 03-29-2012 |
| 20120082154 | DATA DRIVEN CONFIGURATION OF CALL MANAGEMENT APPLICATIONS - A call manager uses a call management application in conjunction with a live dial database to control routing of calls for managed devices. To generate the live dial database, the call management application accesses configured route patterns and enters these patterns into the live dial database. Upon identifying an expansion indicator in a configured route pattern, the call management application accesses dial plan data that includes multiple route pattern definitions that each define a pattern using one or more sub-strings and, for each sub-string, an associated tag. The call management application then enters patterns defined by the route pattern definitions into the live dial database based on various other criteria established for the configured route pattern having the expansion indicator. | 04-05-2012 |
| 20120087367 | Integrating Communications - A method of accessing a first communication system of a first communication provider via a packet-based network, the first communication system maintaining a first list of contacts being users of the first communication system. The method comprises establishing a contact-sharing channel with a second communication system of a second, partner communication provider, wherein the second communication system is accessible via an independently executable web-browser and the packet-based network, and wherein the second communication system maintains second group of contacts being users of the second communication system. The contact-sharing channel is used to fetch contact information of one or more of the second contacts, so as to display at least part of the fetched contact information in the client application and establish a communication based on at least part of the fetched contact information. | 04-12-2012 |
| 20120093145 | SUPPORTING A MULTIMEDIA APPLICATION BASED ON NETWORK ZONE RECOGNITION - In an embodiment, a user equipment (UE) determines that a current serving network is associated with a network support zone of a given type, wherein network support zones of different types are characterized by different levels of support for a multimedia client application configured to manage server-arbitrated communication sessions at the UE. The UE loads zone-specific network procedures for supporting the multimedia client application within the network support zone of the given type, and then executes the zone-specific network procedures at the UE. In another embodiment, an application server determines that the current serving network of the UE is associated with the network support zone of the given type. The application server selects zone-specific network parameters and/or features based on the determination, and then interacts with the multimedia client application within the network support zone of the given type with the selected zone-specific network parameters and/or features. | 04-19-2012 |
| 20120093146 | CALLER INFORMATION PROVISION - By the present invention are disclosed a method for providing and a method for obtaining voice call related information from a first party device ( | 04-19-2012 |
| 20120093147 | METHOD FOR PROVIDING INTERNET SERVICES TO A TELEPHONE USER - Method and system are disclosed for providing a telephone user with a capability to use Internet-based applications. The method comprises the following steps: At a first server, receiving an indication transmitted from the telephone denoting a request to be connected to an Internet-based application residing at a second server. Establishing a communication path that extends between the telephone and the second server via said first server. At the first server, creating a virtual client entity specific to that telephone and the Internet-based application to be used. The virtual client allows communication between the telephone device and that Internet-based application. The virtual client entity is maintained for the duration of a communication session that is about to take place between the telephone user and the Internet-based application. Thereby providing the telephone user the capability to use the Internet-based application, and enable exchanging communications between the second server and the telephone. | 04-19-2012 |
| 20120093148 | Method and Device for Triggering Nested Service - The present invention discloses a method for triggering a nesting service, which relates to the technique of broadband service triggering and service nesting. The present invention is proposed for solving the problem that the existing nesting service occupies more system bandwidths, wherein the method includes: when receiving a call request of a nesting service in a normal service flow, determining a service key of the nesting service according to a called number of the nesting service, taking the service key as a call access object, converting the call request of the nesting service into a call access message in a service platform where a normal service is located, and triggering the nesting service in the service platform. The present invention further discloses an apparatus for implementing the previously mentioned method. The present invention reduced the bandwidth resources consumed by the nesting service, and the scheme is simple and practical. | 04-19-2012 |
| 20120099584 | HOME GATEWAY AND TUNER SHARING METHOD - A home gateway transmits data on the Internet to a plurality of customer premise equipments (CPEs) via a cable modem termination system (CMTS) which connect a plurality of other home gateways. The home gateway distributes a plurality of tuners to the plurality of CPEs according to request packets sent by the plurality of CPEs. The home gateway transmits broadcast packets to the other home gateways to inquire whether one of the other home gateways has an unoccupied tuner or not upon the condition that there are no unoccupied tuners of the home gateway, receives data from the Internet via the unoccupied tuner, and transmits the data to one of the CPEs not distributed the tuner. | 04-26-2012 |
| 20120106540 | Secure Traffic Separation and Management Method - The present invention is a method for securing internet communications between various voice over IP (VoIP) applications. The method enables VoIP Devices to operate within multiple IP networks which are physically connected to the VoIP Device in a manner that ensures inbound and outbound network traffic separation from other connected IP networks based on applicable Security Classifications of the VoIP Device and/or VoIP Device user. | 05-03-2012 |
| 20120106541 | IP PHONE AND DESKTOP SHARING METHOD - An Internet protocol (IP) phone establishes a voice communication with a remote IP phone, and establishes a desktop sharing connection with the remote IP phone upon a sharing key being pressed. The IP phone captures local desktop display data of a local computer, and transmits the local desktop display data to the remote IP phone over the Internet. The IP phone receives remote desktop display data of a remote computer from the remote IP phone over the Internet, and outputs the remote desktop display data to the local computer to display the remote desktop display data on the local computer. | 05-03-2012 |
| 20120106542 | COMMUNICATION TERMINAL THAT PERFORMS NETWORK PACKET COMMUNICATION USING SIP SERVERS, CONTROL METHOD FOR THE COMMUNICATION TERMINAL, AND STORAGE MEDIUM - A communication terminal that determines the types of SIP servers, and stores SIP server information in association with the determined types so that SIP communication can be normally performed. The communication terminal carries out network packet communication using the SIP servers. SIP server information on the SIP servers is acquired, and types of the SIP servers are determined based on the acquired SIP server information. The SIP server information is stored in association with the determined types of the SIP servers. | 05-03-2012 |
| 20120106543 | SYSTEM, METHOD, AND COMPUTER PROGRAM PRODUCT FOR CONNECTING OR COUPLING ANALOG AUDIO TONE BASED COMMUNICATIONS SYSTEMS OVER A PACKET DATA NETWORK - An automated telecommunications system includes a first system operable to receive PSTN compatible audio frequency signals, to decode and interpret said incoming signals according to the message format and a modem protocol being used, and transmit digital messages to a second system over a packet data network. Said second system receives and interprets digital messages incoming from the first system, encodes and regenerate outgoing audio frequency signals. The system may be bi-directional and operate over a packet based data network, such as for example an Internet protocol (IP) based network, a satellite based network, or an IP based cable or wireless network. The functionality of said first and second systems may be combined at a single location and operate with a VoIP network to allow modem signals to pass across the VoIP system. | 05-03-2012 |
| 20120113977 | VPN DEVICE AND VPN NETWORKING METHOD - A VPN device capable of eliminating situations where cross calls occur is provided. The VPN device includes: an identification information acquisition unit that acquires first identification information which is identification information of a communication terminal ( | 05-10-2012 |
| 20120113978 | Converged Voice Mail Services - Novel tools and techniques for providing enhanced services to telephone subscribers with multiple telephone lines, which can include one or more traditional telephone lines, wireless telephone lines, Internet protocol-based telephone lines, and/or the like. For example, certain tools can provide a subscriber with a unified voice mailbox for a plurality of the subscriber's telephone lines. In some cases, the subscriber does not need to check multiple voice mailboxes (each associated with one of the subscriber's telephone numbers) in order to ensure that the subscriber has received all messages intended for that subscriber. Instead, any incoming call on any of the subscriber's lines can be routed to a converged voice mailbox for easy retrieval by the subscriber of all messages from one location. Some tools employ a VoIP switch to handle routing of the incoming call to the correct voice mailbox. | 05-10-2012 |
| 20120113979 | Slow Adaptation of Modulation and Coding for Packet Transmission - Systems and methods for performing MCS adaptation are provided. In some cases, the network performs MCS adaptation based on received NACKs. In other cases, the mobile station determines an MCS based on channel quality measurements, and feeds back the MCS adaptation decision to the network. In either case, NACK-only feedback may be implemented to reduce interference. | 05-10-2012 |
| 20120113980 | COMPUTER, INTERNET AND TELECOMMUNICATIONS BASED NETWORK - A method and apparatus for a computer and telecommunication network which can receive, send and manage information from or to a subscriber of the network, based on the subscriber's configuration. The network is made up of at least one cluster containing voice servers which allow for telephony, speech recognition, text-to-speech and conferencing functions, and is accessible by the subscriber through standard telephone connections or through internet connections. The network also utilizes a database and file server allowing the subscriber to maintain and manage certain contact lists and administrative information. A web server is also connected to the cluster thereby allowing access to all functions through internet connections. | 05-10-2012 |
| 20120113981 | PROVIDING REAL-TIME VOICE COMMUNICATION BETWEEN DEVICES CONNECTED TO AN INTERNET PROTOCOL NETWORK AND DEVICES CONNECTED TO A PUBLIC SWITCHED TELEPHONE NETWORK - Systems, methods, and apparatus for providing real-time voice communication between devices connected to an Internet Protocol (IP) network and devices connected to a public switched telephone network (PSTN). In one implementation, the system includes a computer-controlled switch for connection to a local PSTN, for receiving calls from the IP network and the PSTN, and for routing calls to the PSTN and the IP network; and gate interface circuitry connected to the switch and adapted for connection to the IP network. The gate interface circuitry includes gateway circuitry for interfacing the IP network and PSTN voice circuits, and gatekeeper circuitry for performing address translation, admission control, bandwidth management, and zone management. The switch stores at least one PSTN destination address and at least one IP network destination address for each subscriber. The switch routes an incoming call to any one of the destination addresses stored for the subscriber. | 05-10-2012 |
| 20120120946 | Telephone Exchange Apparatus, Telephone Exchange Apparatus Control Method, and Telephone System - According to one embodiment, a telephone exchange apparatus includes a determining module, a memory and a controller. The determining module determines whether an IP address is transferred through a router, by information notified from the plurality of telephone terminals, when a telephone terminal requests registration. The memory stores a management table associating the terminal ID, the results of determination by the determining module, and the IP address. The controller refers to the management table registered in the memory, and executes one of a first process of establishing connection for communication in a peer-to-peer form between first and second telephone terminals, and a second process of relaying connection by using a relay function, based on a reference results of the management table. | 05-17-2012 |
| 20120127986 | Converged Voice Services - Novel tools and techniques for providing a subscriber with converged voice services, in which calls to one of the subscriber's telephone numbers is routed to some or all of the subscriber's telephone numbers (e.g., sequentially or simultaneously, depending on system configuration and/or subscriber preference). Some implementations employ a VoIP switch to handle call distribution among the subscriber's various phone lines, even if some (or all) of the subscribers telephone lines are not VoIP lines. In such implementations, upon receiving an incoming call directed to one of the subscriber's lines, a service switching point will route the call to a VoIP switch. Instructed by an application server, the VoIP switch will set up call legs (e.g., via the PSTN) to one or more of the subscribers' telephone numbers. Optionally, when one of the call legs is answered, the application server will instruct the VoIP switch to disconnect the other call legs. | 05-24-2012 |
| 20120127987 | PACKET ROUTE MANAGEMENT DEVICE, VoIP SYSTEM AND METHOD OF CONTROLLING VoIP VOICE CALL QUALITY - A packet route management device, a voice over Internet protocol (VoIP) system, and a method of controlling voice call quality. The packet route management device may manage a packet route in an effort to control VoIP voice call based on bandwidth information of packet route devices. Accordingly, the packet route devices may be allowed to process VoIP packets in real time, and thus voice quality can be maintained at a constant level without packet delay or packet loss. | 05-24-2012 |
| 20120127988 | System and method for encoding telephone call data using varying codec algorithms - A method and system for transmitting a call over a packet switched network. A gateway server receives a telephone call and converts analog voice signals associated with the telephone call to a stream of digital data. A first part of the digital data is processed using a first codec algorithm and transmitted over the packet switched network. A change in network conditions is detected. A second part of the digital data is processed using a second codec algorithm and transmitted over the packet switched network. | 05-24-2012 |
| 20120127989 | METHODS AND SYSTEMS TO COMMUNICATE MEDIA DATA ACROSS DIFFERENT NETWORKS - Example methods and apparatus to communicate media across different networks are disclosed. A disclosed example method involves determining, with a processor, whether a second network device in a second network has a capability associated with a first descriptor in a first data packet from a first network device in a first network, communicating, with the processor, a second descriptor indicative of whether the second network device has the capability to the first network device via a second data packet, and when the second network device has the capability, receiving, via the second network device, data from a communication source that bypasses the first network device and communicates the data to the second network device. | 05-24-2012 |
| 20120127990 | Multiple Carrier Gateway System, Method and Apparatus - A system, method and apparatus adapted to provide multiple voice communication carriers access to a data network via a single gateway. | 05-24-2012 |
| 20120127991 | METHOD OF SELECTING A NETWORK RESOURCE - A method and apparatus are provided for selecting a network resource, in which a “controlled-entity” device connected to an IP network ( | 05-24-2012 |
| 20120140763 | UNIFIED COMMUNICATIONS IP PHONE USING AN INFORMATION HANDLING SYSTEM HOST - A VoIP telephone provides basic VoIP capabilities, such as those defined by SIP, with an onboard communications engine that presents a user interface at a display. More complex communications capabilities are provided by interfacing the telephone with an information handling system and driving the user interface of the telephone with a UC application executing on the information handling system. For example, the telephone has an LCD that presents a number pad or basic telephone number list for interaction with the communications engine; however, UC application drives a user interface with greater processing power and storage available from the information handling system, such as an address book or communication actions not supported by the communications engine. | 06-07-2012 |
| 20120140764 | METHOD AND APPARATUS FOR CONFIGURING IP MULTIMEDIA SUBSYSTEM NETWORK ELEMENTS - A system that incorporates teachings of the present disclosure may include, for example, a method for receiving initial filter criteria from a home subscriber server, transmitting information obtained from the initial filter criteria to a domain name system, receiving a multicast IP address from the domain name system, and transmitting a message to a plurality of IP multimedia subsystem network elements according to the multicast IP address. The message can be used for configuring the plurality of IP multimedia subsystem network elements. Other embodiments are disclosed. | 06-07-2012 |
| 20120140765 | METHOD FOR CONTROLLING IP PHONE AND APPLICATION PROGRAM FOR THE SAME - A method for controlling an IP phone and an application program for the same are disclosed. The application program is installed in a computer for connecting the computer and a proximal IP phone via a wired or wireless network. A subscriber sends data and commands to the proximal IP phone via application programs in the computer for controlling the proximal IP phone. The computer establishes connection with a remote phone via an expanded function of the proximal IP phone controlled by the application program. | 06-07-2012 |
| 20120140766 | Method for Accessing Service Resource Items That are For Use in a Telecommunications System - Service resource items for use in call setup in a telephone system are held on servers that are connected to a computer network which is logically distinct from the telephone system infrastructure; this computer network may, for example, make use of the Internet. Each service item is locatable on the network at a corresponding URI and is associated with a particular telephone number. A mapping is provided between telephone numbers and the URIs of associated service resource items. When it is desired to access a service resource item associated with a particular telephone number, this mapping is used to retrieve the corresponding URI which is then used to access the desired service resource item. | 06-07-2012 |
| 20120140767 | SITUATIONAL AWARENESS INTEGRATED NETWORK AND SYSTEM FOR TACTICAL INFORMATION AND COMMUNICATIONS - A system for enabling interoperability among various kinds of communications equipment and information transmission formats on the battlefield or during tactical missions. The system includes a multi-message translator (MMT) for translating a source text message having a first set of word fields defined according to a source message format, into a sink message having a second set of word fields defined according to a sink message format. The system also includes a voice bridging gateway (VBG) for bridging multiple voice communication networks having associated transmission protocols that are incompatible with one another. Dismounted soldiers obtain location based services including geo-referenced maps, and tactical communications including voice and text messaging, using smart phones or other lightweight COTS client devices that link through a personal networking node (PNN) server to one or both of the MMT and the VBG. | 06-07-2012 |
| 20120147881 | METHOD AND APPARATUS FOR PROCESSING MULTIPLE INCOMING CALLS IN A SINGLE DEVICE - A method is disclosed for processing multiple, simultaneous incoming calls directed to a single communication device. The method includes receiving, by a proxy server associated with a called communication device, an invitation for each incoming call directed to the called communication device. The method also includes determining, by the proxy server upon receipt of each invitation, the availability of the called communication device by suspending transmission of each invitation and by sending to the called communication device an associated call initiation message for each invitation. The associated call initiation message is free of a Session Description Payload (SDP). The method further includes receiving, by the proxy server, at least one answer message from the called communication device, each answer message being associated with a selected incoming call. | 06-14-2012 |
| 20120155453 | METHODS AND APPARATUS RELATED TO A SWITCH FABRIC SYSTEM HAVING A MULTI-HOP DISTRIBUTED CONTROL PLANE AND A SINGLE-HOP DATA PLANE - In some embodiments, an apparatus includes a compute device to communicate with a network control entity at each access switch from a set of access switches that define a portion of a data plane having a switch fabric coupling as hierarchical peers each access switch from the set of access switches. The compute device is operable to define a portion of a control plane that includes the network control entities from the set of access switches such that the compute device is hierarchically removed from the network control entities from the set of access switches. The compute device is operable to receive forwarding-state information from a first access switch from the set of access switches. The compute device to send the forwarding-state information to a second access switch from the set of access switches. | 06-21-2012 |
| 20120155454 | MOBILE PHONE DOCKING STATION FOR VoIP - A phone docking station includes a docking port configured to physically dock with a mobile phone. The docking port further includes a charging port configured to connect with the docked mobile phone and to supply a charging current to a battery of the mobile phone, and a Universal Serial Bus (USB) port. The phone docking station further includes a system configured to carry VoIP traffic from an external network through the USB port to the mobile phone, and from the mobile phone through the USB port to the external network; and enable, via the USB port, media streaming from the phone docking station to the mobile phone, and from the mobile phone to the phone docking station. | 06-21-2012 |
| 20120155455 | CALL ANALYSIS - A computer having a processor and a memory is configured to obtain a first set of data from a media gateway, the first set of data including a first trunk identifier and an identifier for a point code associated with the trunk. The computer is further configured to obtain a second set of data related to the media gateway from a call database, the second set of data including a second trunk identifier, and a release code. The computer is further configured to join the first set of data to the second set of data by matching the first trunk identifier to the second trunk identifier. The computer is further configured to determine a number of times that the release code corresponds to the point code. | 06-21-2012 |
| 20120155456 | SOFTSWITCH USAGE ANALYSIS - A first set of data is received from a media gateway, the first set of data including an identifier for a trunk and a point code associated with the trunk. A second set of data is received from the media gateway, the second set of data including the trunk identifier, an identifier for a link associated with the trunk identifier, and a utilization value associated with the link. In a computing device having a processor and a memory, a data record is generated from the first set of data and the second set of data, the data record including a utilization value associated with the point code. | 06-21-2012 |
| 20120155457 | MULTI-SESSION TRANSFER METHOD, CALL CONTROL DEVICE, SERVICE CONTINUITY AND CONTINUITY APPLICATION SERVER - A multi-session transfer method, a call control device, and a Service Continuity and Continuity Application Server are disclosed. In the embodiments of the present invention, in a multi-session cross-network transfer process, if a second session to be transferred includes video media, a Mobile Switching Center (MSC) server judges capabilities of a current network. If the current network is incapable of transmitting video media, the MSC server sends a request for transferring voice media of the second session to be transferred. The SCC AS receives the transfer request, and converts the second session to be transferred into voice session for transferring or releases the second session to be transferred, and therefore, the problem in the prior art of incapability of transferring the session is avoided, and the cross-network multi-session service transfer is improved. | 06-21-2012 |
| 20120163368 | Integrating a Trigger Button Module into a Mass Audio Notification System - An IP-enabled speaker with a trigger button module. Activation of the button on the trigger button module activates an audio link between a user of the speaker and a user of a destination communications device. | 06-28-2012 |
| 20120163369 | LOW LATENCY CALL TRANSFER - A system is configured to receive, from a user device, voice traffic to be sent to another user device; determine whether the voice traffic is local traffic based on whether an eNodeB, associated with the user device, and another eNodeB, associated with the other user device, are served by a particular network device; forward the voice traffic, as non-local traffic, to another network device, associated with a packet data network, for processing when the eNodeB or the other eNodeB are not served by the particular network device; process the voice traffic as local traffic when the eNodeB and the other eNodeB are served by the particular network device; and forward, to the other user device via the other eNodeB, the voice traffic, as local traffic, where the forwarding is performed in a manner that does not include routing the voice traffic via the other network device. | 06-28-2012 |
| 20120163370 | VOIP PHONE READINESS ALERTING - Readiness alerts are provided for callers using Voice Over Internet Protocol (“VoIP”) phones. When the call is placed on hold by the called party, for example to play a stream of music or recorded messages, the caller no longer has to remain attentive in order to detect when the call is answered by a human being. Instead, an agent monitors for the call to be answered. When this is detected, a readiness alert or alerts is/are sent to the calling party. In one approach, a visual message is displayed on a computing device of the calling party. In addition or instead, an audible message is rendered from the computing device. In another approach, the call is automatically forwarded to a mobile device or alternatively, to a different phone, at which the calling party is reachable. | 06-28-2012 |
| 20120163371 | Telephone System, Call Control Apparatus and Communication Connection Method - According to one embodiment, a telephone system includes a transmitter, a detector, a notification module and a controller. The transmitter transmits an outgoing request from a first server apparatus to a second server apparatus. The detector detects a response from the second terminal apparatus, at the second server apparatus. The notification module analyzes media information received at the second server apparatus from the second terminal apparatus and required to achieve peer-to-peer communication, and notifies the media information to the first server apparatus, when the detector detects the response from the second terminal apparatus. The controller causes the first server apparatus to perform the peer-to-peer communication between the first terminal apparatus and the second terminal apparatus based on the media information notified from the second server apparatus. | 06-28-2012 |
| 20120163372 | Method, Apparatus And System For Updating Location Information Of An IP Address - A method, apparatus and system for updating location information of an IP address are disclosed. The method includes: receiving an incoming call; searching a pre-stored location database for a location corresponding to the incoming call; searching a pre-stored log database for an IP address corresponding to the incoming call; and updating location information of the IP address stored in an IP database by the location searched out. The present disclosure may be applied when a service provider updates the location information of the IP address in the IP database. Thus, the problem that the location information in the IP database cannot be updated accurately in the prior art is solved. | 06-28-2012 |
| 20120170571 | SYSTEM AND METHOD FOR DYNAMIC TEMPLATE UPDATING FOR COMPRESSED MESSAGES - A method for updating a message template based on receiving a plurality of common deltas. At a first device, the method detects a first compressed message that includes a first delta and that references a first message template and stores an indication of having received the first delta. The first delta is combinable with the first message template to reconstruct a decompressed message. The first device receives at least one subsequent compressed message that includes the first delta and that references the first message template, and stores an indication of each subsequent time the first delta is received. Based on the stored indications, the first device determines that the first delta was received in accordance with at least one predetermined threshold factor, and in response thereto, initiating a procedure to generate a superseding message template, which updates the first message template by including the first delta. | 07-05-2012 |
| 20120170572 | Method for Enhancing Phone Conversations - A method for enhancing voice conversations comprises providing a server interacting with two or more communication devices which carry out a voice communication, said communication devices being provided with software to simultaneously establish a TCP/IP connection between them, which is mediated by said server. | 07-05-2012 |
| 20120170573 | Methods And Systems For Load Balancing Call Sessions Over A Dual Ring Internet Protocol (IP) Network | 07-05-2012 |
| 20120170574 | UNINTERRUPTED TRANSMISSION OF INTERNET PROTOCOL TRANSMISSIONS DURING ENDPOINT CHANGES - A method apparatus and computer readable medium for facilitating uninterrupted transmission of internet protocol (IP) transmissions containing real time transport protocol (RTP) data during endpoint changes. When an IP transmission is received at the caller RTP port or the callee RTP port, a call record having a caller RTP port identifier or a callee RTP port identifier matching a destination port identifier in the IP transmission is located and when the destination port identifier in the IP transmission matches the caller RTP port identifier of the record, a source IP address identifier and source port identifier from the IP transmission are set as the caller IP address identifier and caller port identifier respectively of the record when the caller IP address identifier and caller port identifier do not match the source IP address identifier and source port identifier respectively and a received SSRC identifier in the IP transmission matches the caller SSRC identifier. When the destination port identifier in the IP transmission matches the callee RTP port identifier of the record, the source IP address identifier and source port identifier from the IP transmission are set as the callee IP address identifier and callee port identifier respectively of the record when the callee IP address identifier and callee port identifier do not match the source IP address identifier and source port identifier respectively and the received SSRC identifier in the IP transmission matches the callee SSRC identifier. | 07-05-2012 |
| 20120177029 | Call processing telecommunication system and methods thereof in a WIFI network - Global identifiers can be dialed from anywhere in the world. The present application relates to a call processing telecommunication system and methods thereof in a wireless network for connecting communication devices. The integrated system can resolve issues with global identification using internal identifiers. In one illustrative embodiment, the global identifiers can be registered and mapped to an internal number such that the global numbers can be used to contact individuals. When a call is placed by a communication device, the internal identifier of the calling device can be tied back to the global identifier, which then provides access to the IP address of the VoIP client to complete the call. For incoming calls, the global identifiers can be translated into an internal identifier to route the call through the network. | 07-12-2012 |
| 20120177030 | PEER-TO-PEER, INTERNET PROTOCOL TELEPHONE SYSTEM WITH PROXY INTERFACE FOR CONFIGURATION DATA - Various embodiments of the invention provide a Peer-to-Peer (P | 07-12-2012 |
| 20120177031 | PEER-TO-PEER, INTERNET PROTOCOL TELEPHONE SYSTEM WITH AUTO-ATTENDANT - Various embodiments of the invention provide a Peer-to-Peer (P2P, Internet Protocol (IP) telephone system. The telephone system includes a plurality of terminals coupled together via an IP network. The terminals cooperate with one another to provide telephony features without a dedicated central controller such as a PBX and/or a KSU controller. The terminals may store system-wide configuration data and files referenced by the system-wide configuration data. The terminals may further determine whether the system-wide configuration data references a file that is not stored in the terminal, and requesting the file from another terminal of the telephone system in response to determining the terminal does not have a copy of the file. | 07-12-2012 |
| 20120177032 | PEER-TO-PEER, INTERNET PROTOCOL TELEPHONE SYSTEM WITH SYSTEM-WIDE CONFIGURATION DATA - Various embodiments of the invention provide a Peer-to-Peer (P2P, Internet Protocol (IP) telephone system. The telephone system includes a plurality of terminals coupled together via an IP network. The terminals cooperate with one another to provide telephony features without a dedicated central controller such as a PBX and/or a KSU controller. The terminals further cooperate with one another to maintain system-wide configuration data for the telephone system. | 07-12-2012 |
| 20120177033 | PROVIDING EFFECTIVE ADVERTISING VIA SYNCHRONIZED TELEPHONE AND DATA STREAMS - Information, such as advertising, is presented to VoIP users via a combination of telephone and data streams during session initiation via an IP phone or a personal computer. A proxy server that receives the call request coordinates with a media server to transmit targeted advertising to the caller during the post dial delay interval. After the advertisement concludes, the call continues normally. | 07-12-2012 |
| 20120182987 | XML BASED TRANSACTION DETAIL RECORDS - The present invention is directed to a method for managing transactions in a telecommunications network. The method includes creating an XML transaction detail file. At least one transaction detail record is stored in the XML transaction detail file in response to a telecommunications transaction. The at least one transaction detail record includes transaction data corresponding to the telecommunications transaction. | 07-19-2012 |
| 20120182988 | Communication System and Method - There is provided a packet-based communication system for conducting voice or video calls over a packet-based network. A client application on a first terminal is configured to determine an availability of one or more other, second user terminals installed with other instances of the client application, and to present a near-end user with an option to select one of said second user terminals for use by the near-end user in conjunction with the first user terminal when conducting a voice or video call with a far-end user via the client instances and packet-based communication system; thereby enabling the voice or video call to be conducted using the second terminal to consume or generate at least one stream of the call whilst the first user terminal concurrently generates or consumes at least another stream of the call or controls the call. | 07-19-2012 |
| 20120188999 | Personal Computer and Mobile Phone Communications through Peer-to-Peer Connection - An information handling system includes a network interface, a radio frequency interface, and a processor. The processor is configured to receive a call initiation signal from a remote system through the network interface, establish a connection with a portable communication device through the radio frequency interface, and accept a call from the remote system. Additionally, the processor is further configured to route audio signals related to the call between the portable communication device and the remote system. | 07-26-2012 |
| 20120189000 | MANAGING TELEPHONY SERVICES USING MULTIPLE USERS WITHIN A TELEPHONY CONTROL POINT IN A HOME NETWORK - A method for operating a telephony server in a home network is provided, comprising: receiving a first registration action from a first telephony control point in the home network, the first registration action including an identification of a first user within a first telephony control point; supplying a first authentication identifier to the first telephony control point, wherein the first authentication identifier is bound to both the first telephony control point and the first user; receiving an invoked action granting a telephony action from the first telephony control point, wherein the invoked action includes an authentication identifier; determining if the authentication identifier received in the invoked action matches the authentication identifier bound to the user which the telephony action involves; and executing the telephony action if the authentication information received in the invoked action matches the authentication identifier bound to the user which the telephony action involves. | 07-26-2012 |
| 20120189001 | VOICE OVER IP (VOIP) NETWORK INFRASTRUCTURE COMPONENTS AND METHOD - A voice over Internet protocol communication system and method provides infrastructure components as intermediaries between networks, the components include multi-protocol session controllers and a multi-protocol signaling switch as well as a management system. The session controllers process calls and participate in the calls that flow through it. The session controllers process calls that are either at the edge of the network or at the core of the voice over Internet protocol network. The session controllers associate calls with one another in call peers for incoming calls as ingress call peers and for outgoing calls as egress call peers. A centralized database of call routing policies is provided to the session controllers. The session controllers provide cost management, topology hiding, and inter-working, or conversion, of calls from SIP networks to H.323 networks for both voice and video. | 07-26-2012 |
| 20120189002 | Protocol Converting 9-1-1 Emergency Messaging Center - Delivery of Next Generation 9-1-1 emergency services to an Internet Protocol (IP) Public Safety Answering Point (PSAP) is enabled using an existing or legacy selective router and Automatic Location Information (ALI) database. Using a protocol converting 9-1-1 messaging center according to the principles of the present invention, existing or legacy selective router equipment can provide Enhanced 9-1-1 (E911) over time division multiplex (TDM) circuits to non-IP capable PSAPs. | 07-26-2012 |
| 20120195303 | SYSTEM AND METHOD FOR INTEGRATING CONVENTIONAL ENTERPRISE COMMUNICATION SYSTEMS INTO AN IP MULTIMEDIA SUBSYSTEM-BASED SERVICES ENVIRONMENT - A system and method for integrating conventional enterprise communication systems into an IMS-based services environment are disclosed. In one embodiment, when a call request is received for a call between first and second local users of a PBX, the PBX determines whether a class of service for either the first or second local user is for Exposed Local Call (ELC) processing. If so, then call information is routed through a Session Initiation Protocol (SIP) core network that is coupled to an Internet Protocol Multimedia Subsystem (IMS) server to provide IMS services to the call. | 08-02-2012 |
| 20120195304 | VOIP CONTENT DELIVERY AND CONTROL MANAGER - A method and system of delivering and managing audio and video content, comprising: receiving, over at least one computer, at least one audio and/or video file to be played using at least one VOIP audio and video PBX telephone system; managing remotely, using at least one computer application, the at least one audio and/or video file; and distributing the at least one audio and/or video file to the at least one VOIP audio and/or video PBX telephone system. | 08-02-2012 |
| 20120195305 | AGGREGATING ENDPOINT CAPABILITIES FOR A USER - A method and system for aggregating capabilities from multiple endpoints associated with a user are provided. The system aggregates the capabilities of the endpoints associated with a user into an aggregate view of available modes of communication for reaching the user. Then, the system publishes the aggregate view so that other users who want to send communications to the user will know the modes of communication available for that user. In addition, the system may designate certain modes of communication as preferred or as capable of reaching the user. | 08-02-2012 |
| 20120195306 | Method and System for Providing to a Second Party, Computer-Network Related Information about a First Party - A method and system for providing computer-network related information about a second party. First, the second party receives a telephone number of a first party. The second party's customer premises equipment (CPE) or elements in a telephone network then use the telephone number to index a database, which contains combinations of telephone numbers and computer-network addresses. Once the first party's computer-network address is retrieved, first-party-customized information present at the computer-network location specified by the computer-network address can be sent to and displayed on the second party's CPE. Also, the first party's telephone number can be sent to an application in the computer network, causing the first-party-customized information to be automatically displayed on the second party's CPE. | 08-02-2012 |
| 20120195307 | Method and System for Providing to a Second Party, Computer-Network Related Information about a First Party - A method and system for providing computer-network related information about a second party. First, the second party receives a telephone number of a first party. The second party's customer premises equipment (CPE) or elements in a telephone network then use the telephone number to index a database, which contains combinations of telephone numbers and computer-network addresses. Once the first party's computer-network address is retrieved, computer network information present at the computer-network location specified by the computer-network address can be sent to and displayed on the second party's CPE. Also, the first party's telephone number can be sent to an application in the computer network, causing the first-party-customized information to be automatically displayed on the second party's CPE. | 08-02-2012 |
| 20120201238 | METHOD FOR PROCESSING TELEPHONY SESSIONS OF A NETWORK - A method for processing telephony sessions of a network including at least one application server and a call router, the method including the steps of assigning a primary Uniform Resource Identifier (URI) and at least a secondary URI to an application; mapping a telephony session to the primary URI; communicating with the application server designated by the primary URI using an application layer protocol; receiving telephony instructions from the application server and processing the telephony instructions with the call router; detecting an application event; and upon detecting the application event, communicating with the application server designated by the secondary URI. | 08-09-2012 |
| 20120201239 | Multi-Line Telephone Calling - Network architectures, methods, and operations for routing telephone calls are disclosed. A customer premise includes a conventional POTS dedicated communication line to a central office switch and a broadband connection to a data network. The central office switch and the data network are connected by a communication link. A first telephone call originated at the customer premise may be connected via the dedicated communication link. A second or subsequent call originated at the customer premise may be connected via the data network. Optionally, the same call identifier may be assigned to the first and second or subsequent calls. | 08-09-2012 |
| 20120207147 | Communication System and Method - There is provided an instance of a client application enabling a first user terminal to access a packet-based communication system to conduct voice or video calls over a packet-based network. The client application is configured to receive an input from one or more audio and/or video input transducers of the first terminal, and to operate in conjunction with one or more other instances of the client application executed on one or more respective second terminals so as to participate in an analysis of the one or more inputs in relation to an input from one or more audio and/or video input transducers of the one or more second terminals; thereby enabling selection of one of the first and second terminals for use by a near-end user in conducting a call with a far-end user of a third user terminal via the respective client instance and packet-based communication system. | 08-16-2012 |
| 20120207148 | Using a Common Media Gateway Node and a Coordinated Codec by an Originating and a Terminating Call Control Node - A method for a call control node (MSC-S | 08-16-2012 |
| 20120207149 | SESSION INITIATION PROTOCOL (SIP) MESSAGE INCORPORATING A NUMBER OF PREDETERMINED ADDRESS HEADERS HAVING PREDETERMINED ADDRESS INFORMATION - A Session Initiation Protocol (SIP) message adapted for communication between a plurality of network elements located on a multi-media services provider system to form a multi-media communication path between at least a first communication device and at least a second communication device of a plurality of communication devices. The SIP message includes a header region and a plurality of header fields located on the header region. The plurality of header fields include predetermined address information associated with at least one network element of the plurality of network elements located on the multi-media services provider system. | 08-16-2012 |
| 20120207150 | INPUT/OUTPUT END POINT CONTROL FOR INTEGRATED TELECOMMUNICATIONS NETWORK SYSTEMS VOIP - Disclosed are systems and methods that enable expanded functionality for call control monitoring and control of external electrical/electromechanical devices using a telecommunication system. | 08-16-2012 |
| 20120213219 | IN-VOICEMAIL-SESSION CALL TRANSFERS - In one embodiment, a voicemail system stores a voicemail message left by a first caller for a second caller. The voicemail system facilitates the transfer of a call out of a playback of the voicemail to the first caller and generates a credential that is coupled with the call. If the first caller is not available for the call, the second caller is transferred to the voicemail mailbox of the first caller where second caller records a second voicemail, which is placed in the mailbox of the first caller. The call may be transferred back to the mailbox of the first caller in response to a transfer code entered by the second caller. The voicemail system verifies the second caller based on the credential and returns the second caller to the appropriate location in the queue of voicemails. | 08-23-2012 |
| 20120218989 | GATEWAY UNIT, COMMUNICATION SYSTEM AND COMMUNICATION METHOD - A gateway unit which relays communication between a call control network and a terminal having no call control function. The gateway unit includes: a session control section which substitutively performs call control processing in call control communication performed by the terminal by way of the call control network and reports a band required for the call control communication to the call control network; and a band adjustment section which determines an updated value of a band of the call control communication based on a communication packet in the call control communication, received from the terminal. The session control section reports the updated value to the call control network as the band required by the call control communication. | 08-30-2012 |
| 20120230323 | Smart Routing for Voice Over Internet Protocol - In a VoIP application where network packets must travel beyond the local internet provider, the network packet may be rerouted to avoid the open internet. The originating endpoint is provisioned to transmit network packets to the VoIP provider. The originating local internet provider receives the network packet from the originating endpoint, and transmits the network packet to the VoIP provider via a direct hand-off. The VoIP provider then transmits the network packet to the destination local internet provider via a direct hand-off. Finally, the destination local internet provider delivers the network packet to the destination end point. Transmission through the open internet is reserved as a back-up process, should transmission through the VoIP provider fail. | 09-13-2012 |
| 20120230324 | Method and Apparatus for Identifying VoIP Traffic - The present invention discloses a method for identifying Voice over Internet Protocol (VoIP) traffic. Based on a Media Gateway Control Protocol (MGCP), the method includes: identifying a packet related to a control connection by matching keywords according to an identification rule of the MGCP; extracting media connection negotiation information for establishing a media connection from the packet related to the control connection; matching the media connection negotiation information according to a transaction Identification (ID) in the media connection negotiation information; and identifying MGCP based VoIP traffic according to matching results. The present invention further discloses an apparatus for identifying VoIP traffic based on the MGCP. In the condition of using the MGCP and performing a plurality of media connection negotiations on one control connection simultaneously, the present invention can identify the MGCP based VoIP traffic without any particular requirement on the deployed positions of deep packet inspection devices. | 09-13-2012 |
| 20120236844 | ENABLING QUALITY VOICE COMMUNICATIONS FROM WEB PAGE CALL CONTROL - A system and method of bypassing the regulated portion of the Public Switching Telephone Network (PSTN) to establish carrier-grade voice transmissions and/or IP data communications between an Internet Calling Person having a first telephone and a first PC coupled to a first Local Service Access Provider (LSAP) and an Internet Called Party having a second telephone and a second PC coupled to a second different Local Service Access Provider (LSAP). | 09-20-2012 |
| 20120236845 | Voice and Data Exchange Over a Packet Based Network - A signal processing system which discriminates between voice signals and data signals modulated by a voiceband carrier. The signal processing system includes a voice exchange, a data exchange and a call discriminator. The voice exchange is capable of exchanging voice signals between a switched circuit network and a packet based network. The signal processing system also includes a data exchange capable of exchanging data signals modulated by a voiceband carrier on the switched circuit network with unmodulated data signal packets on the packet based network. The data exchange is performed by demodulating data signals from the switched circuit network for transmission on the packet based network, and modulating data signal packets from the packet based network for transmission on the switched circuit network. The call discriminator is used to selectively enable the voice exchange and data exchange. | 09-20-2012 |
| 20120236846 | METHODS AND APPARATUS TO ENABLE CALL COMPLETION IN INTERNET PROTOCOL COMMUNICATION NETWORKS - Methods and apparatus to enable call completion in Internet protocol communication networks are disclosed. Example methods disclosed herein to process a voice over Internet protocol call include sending a first query to a telephone number mapping server to obtain a session initiation protocol uniform resource identifier associated with a destination device with which the voice over Internet protocol call is to be established. Some such example methods further include, in response to the voice over Internet protocol call failing to be established using the session initiation protocol uniform resource identifier obtained from the first query, sending a second query to the telephone number mapping server to obtain a numeric Internet protocol address associated with the destination device, and using the numeric Internet protocol address obtained from the second query to establish the voice over Internet protocol call with the destination device. | 09-20-2012 |
| 20120236847 | SYSTEMS AND METHODS FOR INITIALIZING CABLE MODEMS - A system includes a first device and a second device. The first device is configured to transmit a discover message on a first upstream channel, where the discover message includes information representing capabilities of the first device. The second device is configured to receive the discover message from the first device and determine whether to switch the first device to a second upstream channel based on the capabilities information in the discover message. The second device makes the determination before a registration of the first device. The second device transmits a message to the first device instructing the first device to switch to the second upstream channel based on a result of the determination. | 09-20-2012 |
| 20120243530 | USING PSTN REACHABILITY TO VERIFY VOIP CALL ROUTING INFORMATION - A system for verifying VoIP call routing information. In particular implementations, a method includes verifying one or more Voice-over-Internet-Protocol (VoIP) call agents for respective destination telephone numbers based on demonstrated knowledge of previous public switched telephone network (PSTN) calls to the respective destination telephone numbers; receiving a call initiation message identifying a destination telephone number; and conditionally initiating a call over a VoIP network to a target VoIP call agent, or over a circuit switched network, based on whether the target VoIP call agent has been verified for the destination telephone number identified in the call initiation message. | 09-27-2012 |
| 20120243531 | Establishing a Communication Session - Parallel communication sessions can be established. A system receives an email communication from a user computer, and sends a reply to the user computer, wherein the reply includes a link allowing the user computer to establish a connection to a server; and wherein the link further includes an identifier. The system then receives from the user computer a connection request based on the link, said connection request including the identifier and specifying a first phone number associated with the user computer. The system then invokes an application for establishing a first voice connection with a first terminal identified by said first phone number, establishing a second voice connection with a second terminal identified by a second phone number indicated as associated with a computer terminal, and interconnecting the first and second voice connections to establish a voice call between the first and second terminals. The system also establishes a data connection with the user computer in response to the connection request, and establishes a data connection with the computer terminal, such that the user computer and the computer terminal can share a multimedia session, such that the voice call and the multimedia session are each associated with the identifier. | 09-27-2012 |
| 20120250675 | IMS APPLICATION SEQUENCING OPTIMIZER - An application optimizer is disclosed that is configured to optimize the execution of applications in an application sequence. Specifically, two or more applications in an application sequence may include proxy applications that are configured to embed commands into a call-setup message and the application optimizer is configured to parse and execute the commands embedded in the call-setup message as a single Back-to-Back User Agent. | 10-04-2012 |
| 20120250676 | VOIP GATEWAY AND METHOD FOR SETTING UP SPEECH COMMUNCIAITON THEREOF - A Voice over Internet Protocol (VoIP) gateway includes a virtual Session Initiation Protocol (SIP) proxy server, a management and monitoring module, and a virtual SIP phone. The virtual SIP proxy server is configured for registering the at least one of a number of local audio terminals and assigning an internet protocol (IP) address to each of them. When the local phone terminal dials the VoIP phone or receives a call from the VoIP, the management and monitoring module sets up a communication between the local phone terminal and the VoIP phone. | 10-04-2012 |
| 20120250677 | TELEPHONE EMERGENCY RESPONSE SYSTEMS AND METHODS - An emergency system includes a gateway and a database in communication with the gateway. The database comprises first routing information for establishing a first communication link between a communication device and a controller, second routing information for establishing a second communication link between the gateway and the controller, wherein the second routing information is correlated to the first routing information; and location data associated with the communication device. Upon receiving identification information related to the communication device, the gateway retrieves the first routing information from the database and provides the first routing information to a switch. The gateway retrieves the second routing information from the database and uses the second routing information to establish a second communication link with the controller, and the gateway retrieves the location data from the database. | 10-04-2012 |
| 20120250678 | METHOD OF SCHEDULING TRANSMISSION IN A COMMUNICATION NETWORK, CORRESPONDING COMMUNICATION NODE AND COMPUTER PROGRAM PRODUCT - Data streams are transmitted from a node towards a receiver in a communication network in the form of data packets for playout via a reproduction buffer at the receiver. The data packets are arranged in a scheduling queue and dropped from the scheduling queue if their sojourn time in the queue exceeds a given drop deadline. The reproduction buffer is emulated at the node in order to determine respective playout values for the data packets which are indicative of expected playout instants for the data packets by the reproduction buffer at the receiver. The drop deadlines are assigned to the data packets as a function of the respective playout values determined via the reproduction buffer as emulated at the node. | 10-04-2012 |
| 20120257615 | Self-Contained Security System Including Voice and Video Calls Via the Internet - A security system is provided. The security system includes a control panel located within a secured space, a plurality of sensors coupled to the control panel that detect security events within the secured space, a user interface within the secured area coupled to the control panel, a camera located in the user interface that collects video images in the secured space surrounding the user interface upon activation of one of the plurality of sensors, a sound transducer located in the user interface, a programmed processor within one of the user interface and the control panel that supports VoIP calls between a called or calling party and the user interface and a control button on the user interface that activates a VoIP connection with a called or calling party through the user interface and control panel. | 10-11-2012 |
| 20120257616 | Single-Rotator Latent Space Switch with an External Controller - A latent space switch based on a single rotator and an array of memory devices is disclosed. The switch interfaces with external nodes through a set of access ports. The rotator has a set of inlets and a set of outlets with each inlet connecting to each outlet during a time frame organized into time slots. During each time slot, an inlet alternately connects to an access port and a memory device while a transposed outlet of the inlet alternately connects to the same memory device and another access port. Multiple temporal multiplexers submit upstream control messages from the access ports to a multi-port master controller. Multiple temporal demultiplexers distribute downstream control messages sent from the master controller to the access ports. | 10-11-2012 |
| 20120257617 | SYSTEM AND METHOD FOR ROUTING COMMUNICATIONS BETWEEN PACKET NETWORKS BASED ON REAL TIME PRICING - The disclosed embodiments include a method for communicating data packets over packet networks owned by different communications carriers. The method includes monitoring, throughout scheduled times of the day, pricing information of the different communications carriers for communicating the data packets over the packet networks owned by the different communications carriers. The method determines over which of the packet networks owned by the different communications carriers to communicate data packets based on the pricing information. | 10-11-2012 |
| 20120263168 | Communication System and Method - A system comprising: one or more first user terminals each for a respective user, each of the first user terminals comprising a first interface to a service run over a communication network; an authorisation mechanism arranged to conditionally verify the users of the first user terminals as being licensed to use the communication service based on a first indication received via the first interface; and one or more second user terminals each for a respective user, each of the second user terminals comprising a second, alternative interface to the communication service; wherein the second interface is not operable to provide the first indication to the authorisation mechanism; and wherein the authorisation mechanism is configured to conditionally verify the users of the second user terminals as being licensed to use the communication service based on a second indication received via the second interface. | 10-18-2012 |
| 20120263169 | METHOD FOR APPLYING INTERNET TELEPHONE SERVICE BY CONVENTIONAL TELEPHONE WITH INTERNET TELEPHONY GATEWAY - The present invention provides a method for applying internet telephone service system by conventional telephone with internet telephony gateway. By message processors receiving and sending message and processing message between conventional telephone and internet telephone service system, and by a command set built-in second message processor, the user is capable of having internet telephone service system on a conventional telephone. In practice, the user can press the keys on the conventional telephone to select an account displayed on Caller ID display module thereon and then make a call to the account; additional system information such as unanswered internet calls or remaining credits of the internet telephone service system can also be displayed thereon. | 10-18-2012 |
| 20120263170 | SYSTEM AND METHOD FOR SWITCHING BETWEEN PUBLIC SWITCHED TELEPHONE NETWORKS AND VOICE OVER INTERNET PROTOCOL NETWORKS - A location customer premise equipment (CPE), the location CPE in electronic communication with a location telephone, a remote CPE and a remote telephone. The location CPE includes a first processor. The first processor receives a switch signal from the location telephone, and sends a corresponding first request signal to the remote CPE, and then the first processor receives a corresponding first response signal and a phone number of an unused network of the remote telephone from the remote CPE, and dials the phone number of an unused network to establish an unused network communication link with the remote CPE and cuts off a used network communication link. | 10-18-2012 |
| 20120263171 | SESSION PROCESSING METHOD, DEVICE, AND COMMUNICATION SYSTEM - Embodiments of the present invention disclose a session processing method, device, and communication system. The session processing method includes: when a first User Equipment (UE) is to set up an IP Multimedia Subsystem (IMS) session with a second UE, triggering setup of an unstructured supplementary service data (USSD) transaction between a Circuit Switched (CS) network and the first UE; and sending, by using the USSD transaction, an I1 protocol message used to bear IMS session control signaling. With the technical solutions of the present invention, when a UE is to set up an IMS session, a USSD transaction is set up between the UE and the CS network; I1 protocol messages exchanged between the UE and the Service Centralization and Continuity (SCC) Application Server (AS) (SCC-AS) are sent reliably by using the USSD transaction, which facilitates reliable control over IMS services. | 10-18-2012 |
| 20120263172 | METHOD AND SYSTEM FOR ROUTING AND SECURITY FOR TELEPHONE CALLS OVER A PACKET-SWITCHED NETWORK - A server, upon receiving a request to complete a call over a packet-switched network, looks up an address on the packet-switched network on a local table. If the local table does not contain a matching address, the server asks a routing server to identify a receiving server address. The routing server identifies a suitable address and sends a message to the originating server containing the address. The receiving server then receives, via the packet-switched network, a message requesting a telephone connection to a second telephone device. Before completing the connection to the second telephone device, information in the message representing the address of the device on the packet-switched network that originated the message is identified and the address information is compared to selected predetermined addresses for permitted origins of the message. | 10-18-2012 |
| 20120263173 | Message Handling in a Communications Network - A method and apparatus for handling a Session Initiation Protocol message in a communications network. When a network node receives a Session Initiation Protocol message, which comprises Request-URI header, the node rewrites the Request-URI header in the SIP message, and adds information to the SIP message useable by a remote node to determine the current target address of the message. The SIP message is then sent to a further node. In this way, the remote node that receives the message can determine the current target in the SIP message, even if the target has been re-written in the Request-URI as the result of, for example, a translation or re-routing operation. | 10-18-2012 |
| 20120263174 | SS7 MAP/LG+ TO SIP BASED CALL SIGNALING CONVERSION GATEWAY FOR WIRELESS VOIP - An SS7-based call protocol conversion gateway that translates between circuit-switched SS7 protocols and session initiation protocol (SIP) oriented protocol, allowing an E911 call initiated over a switched network to be routed by a VoIP network. The SS7-based call protocol conversion gateway provides a PSAP with MSAG quality (street address) information about a VoIP dual mode phone user without the need for a wireless carrier to invest in building out an entire VoIP core. Thus, wireless carriers may continue signaling the way they are today, i.e., using the J-STD-036 standard for CDMA and GSM in North America, yet see benefits of a VoIP network core, i.e., provision of MSAG quality location data to a PSAP. | 10-18-2012 |
| 20120263175 | SYSTEM AND METHOD FOR ALLOCATING SESSION INITIATION PROTOCOL (SIP) IDENTIFICATIONS (IDs) TO USER AGENTS - A communications system includes a Session Initiation Protocol (SIP) user agent. A server communicates with the SIP user agent and allocates an SIP ID for the user agent for subsequent communications using SIP. A database can be associated with the server and contain data relating to free SIP ID's that can be allocated to the SIP user agent and allocated SIP ID's. | 10-18-2012 |
| 20120269185 | SYSTEM AND METHOD FOR COMPUTER BASED COLLABORATION INITIATED VIA A VOICE CALL - A method and infrastructure for online collaboration, cloud computing in which collaboration sessions, known as peering sessions, are triggered by telephone calls. Other systems and methods are disclosed. | 10-25-2012 |
| 20120269186 | HIERARCHICAL DATA COLLECTION NETWORK SUPPORTING PACKETIZED VOICE COMMUNICATIONS AMONG WIRELESS TERMINALS AND TELEPHONES - A packet-based, hierarchical communication system, arranged in a spanning tree configuration, is described in which wired and wireless communication networks exhibiting substantially different characteristics are employed in an overall scheme to link portable or mobile computing devices. The network accommodates real time voice transmission both through dedicated, scheduled bandwidth and through a packet-based routing within the confines and constraints of a data network. Conversion and call processing circuitry is also disclosed which enables access devices and personal computers to adapt voice information between analog voice stream and digital voice packet formats as proves necessary. Routing pathways include wireless spanning tree networks, wide area networks, telephone switching networks, internet, etc., in a manner virtually transparent to the user. A voice session and associate call setup simulates that of conventional telephone switching network, providing well-understood functionality common to any mobile, remote or stationary terminal, phone, computer, etc. | 10-25-2012 |
| 20120269187 | Methods, Apparatus and Computer Program Products for Associating Local Telephone Numbers with Emergency Phone Calls in a Packet Switched Telephone Systems - A packet switched telephone system includes a packet switched routing apparatus. The packet switched routing apparatus selectively associates a local telephone number with a phone call based on a called telephone number, and routes the phone call based on the called telephone number. The local telephone number may be substituted for a calling telephone number when the called telephone number corresponds to a predefined number, such as an emergency number. When the called telephone number corresponds to an emergency number, the phone call may be routed with the substituted local telephone number to a Public Safety Access Point (PSAP) that services the local area of the subscriber. | 10-25-2012 |
| 20120269188 | SYSTEM AND METHOD FOR DYNAMIC CALL ROUTING - A telecommunications system is disclosed which may include a VOIP (Voice Over Internet Protocol) network having a plurality of network elements; at least one carrier data center; communication links enabling communication between the network elements and the at least one carrier data center, wherein the carrier data center is operable to receive a query describing a communication session active at a given one of the network elements, over one of the communication links, and to generate a routing table in response to the query. | 10-25-2012 |
| 20120275450 | Obtaining Services Through a Local Network - A local network gateway may exchange signaling data for a voice communication session with an external network. The gateway may also exchange internal signaling information relating to that voice communication session across logical ports within the local network gateway using a local signaling protocol. A voice communication session can be established between the local gateway and an external network and bridged to local communication sessions between the local gateway and end points. | 11-01-2012 |
| 20120275451 | METHODS AND APPARATUS TO PERFORM CALL SCREENING IN A VOICE OVER INTERNET PROTOCOL (VOIP) NETWORK - Methods and apparatus to perform call screening in a voice over Internet protocol (VoIP) network are disclosed. Disclosed example methods include receiving a call screening termination message at a feature server of an Internet protocol multimedia subsystem, initiating a communication session between a first user device operable with the Internet protocol multimedia subsystem and a second user device operable with the Internet protocol multimedia subsystem in response to the call screening termination message, the first and second user devices being involved in a call screening session, and releasing a signaling resource of a messaging server, the signaling resource having been used in the call screening session. | 11-01-2012 |
| 20120275452 | SYSTEM AND METHOD FOR INSTANT VoIP MESSAGING - Methods, systems and programs for instant voice messaging over a packet-switched network are provided. A method for instant voice messaging may comprise receiving an instant voice message having one or more recipients, delivering the instant voice message to the one or more recipients over a packet-switched network, temporarily storing the instant voice message if a recipient is unavailable; and delivering the stored instant voice message to the recipient once the recipient becomes available. | 11-01-2012 |
| 20120281690 | INSERTING OUT-OF-BAND DATA INTO IN-BAND DATA STREAMS - A computer-implemented method for inserting an out-of-band signaling packet into a real-time protocol (RTP) stream is provided. The method includes receiving the out-of-band signaling packet intended for transmission to a user device and forming a synthesized packet based on payload information from the out-of-band signaling packet and header information stored in a data structure describing the RTP stream. The method also includes inserting the synthesized packet into the RTP stream. The method further includes receiving an RTP packet intended for transmission to the user device via the RTP stream, analyzing an insertion sequence number and an insertion flag maintained in the data structure, and discarding or forwarding the RTP packet via the RTP stream based on the analyzing. | 11-08-2012 |
| 20120281691 | METHOD AND APPARATUS IN WHICH CALL SIGNALING MESSAGES BYPASS IN-TRANSPARENT SWITCHING NODES OR NETWORKS - A method is disclosed for initiating a call in a communication network between a first signaling entity and a second signaling entity each connected to a respective node and the nodes connected to a network wherein communications between the signaling entities and the nodes are conducted using a first protocol and communications are carried over the network using a second protocol. Call setup information is also exchanged between the first node and the second node over a separate connection. | 11-08-2012 |
| 20120281692 | NETWORK SWITCHING SYSTEM WITH ASYNCHRONOUS AND ISOCHRONOUS INTERFACE - To provide a switching system with telephone switching function mainly on the basis of hardware processing by using isochronous channel which is a real time communication channel. The switching system comprises a gateway node connected with ISDN (Integrated Services Digital Network) and PSTN (Public Switched Telephone Network), and one or more extension nodes, and a serial bus such as IEEE 1394 bus. The gateway node transforms data rate of outside line into data rate of extension node, and the other way around, and secure a seamless communication channel. Concretely, the gateway node secures an isochronous channel, according to a request from the extension nodes or the outside line, and executes switching such as transfer or reservation. A resource manager holds a table for managing the gateway node and extension node. | 11-08-2012 |
| 20120287923 | RESIDENTIAL GATEWAY FOR VOICE OVER INTERNET PROTOCOL COMMUNICATIONS - A method and system are provided for preventing data loss in a VoIP system. In particular, during a VoIP call, it is determined whether incoming ringing on a POTS line causes an unacceptable level of signal loss or errors. If so, for subsequent VoIP calls, the CO handling calls to the POTS line is instructed to either answer each call with a busy signal or automatically forward calls to the POTS line to the VoIP line or other selected telephone. Calling returns to normal upon ending of the VoIP call. In this manner, incoming ringing on the POTS line does not result in call dropping or lengthy retraining processes. | 11-15-2012 |
| 20120287924 | EFFICIENT ADDRESS CACHING FOR PACKET TELEPHONY SERVICES - A method for telephony includes receiving at an Internet telephony service provider a subscriber request to place a call to a telephone number. A cache associated with the internet telephony service provider is queried to check if the cache holds a record for the telephone number. If the cache holds the record, the record is obtained. If the cache does not hold the record, a request is sent to a database server that maintains a database of records associating endpoint user terminal telephone numbers of subscribers with respective packet network addresses of the endpoint user terminal. The call is placed to the endpoint user terminal telephone number via a public switched telephone network whilst the request is sent to the database server to retrieve the packet network address of the endpoint user terminal to which calls to the telephone number should be placed. | 11-15-2012 |
| 20120287925 | POTS EXTENDER FOR VOICE FALLBACK IN A SUBSCRIBER LINE - A full services access multiplexer is described. A master DSL modem is coupled to a conductor pair. A POTS extender is coupled to the conductor pair and may sense the operation of a fallback or other signal on the conductor pair. A suppression signal may be transmitted to the master DSL modem upon occurrence of the fallback. The suppression signal may travel over a control circuit. Traffic over a backplane or other network segment may be uninterrupted to an Integrated Access Device by handling signals inbound and outbound to the backplane via packet assembler and disassembler (PAD). The PAD may transmit a data stream to vocoder and received a data stream from vocoder for injection onto the backplane. | 11-15-2012 |
| 20120294302 | METHOD AND APPARATUS FOR MANAGING CALLS - A system that incorporates teachings of the present disclosure may include, for example, one or more components for receiving a call request at a first server from a first end user device, transmitting the call request from a first server to an intermediate server for establishing a voice or video call over an IP multimedia subsystem between the first end user device and a second end user device via a second server, and routing the voice or video call request from the first server to the second server using a second IP address when an undesired condition is determined to be associated with the IP multimedia subsystem. The second IP address can be obtained from a group of IP addresses stored in a memory of the first server and the group of IP addresses can be associated with other servers. Other embodiments are disclosed. | 11-22-2012 |
| 20120294303 | WIDE AREA COMMUNICATION NETWORKING - A communications network is disclosed and includes a broadband communication line having a first derived voice channel and a second derived voice channel, wherein the first and second derived voice channels are established as a function of an available bandwidth associated with the broadband communication line. The communication network further includes a residential gateway in communication with the broadband communication line. The residential gateway includes a switch, a network interface device in communication with the switch, and wherein the switch is configured to select at least one of the first or second derived voice channels for voice communication over the broadband communication line as a function of the available bandwidth. | 11-22-2012 |
| 20120294304 | VOICE-OVER-IP HYBRID DIGITAL LOOP CARRIER - Certain exemplary embodiments can comprise a method of use comprising: for a call between a local IP network and a remote non-IP network, converting between IP packets and PCM robbed bit signaling via a VoIP channelized router; providing the PCM robbed bit signaling to a TDM switch via the VoIP channelized router; and/or converting between IP packets and GR303 call reference values via the VoIP channelized router. | 11-22-2012 |
| 20120300767 | Method and apparatus for voice traffic management in a data network - Method and apparatus for voice traffic management in a data network includes establishing a default maximum bandwidth setting at a LAN egress port when voice-type traffic is not present in a LAN portion of the data network, detecting voice-type traffic, reducing the bandwidth setting at the LAN egress port to effect a change in a rate of non voice type traffic and monitoring non voice type traffic and voice quality statistics to determine if the rate of non voice type traffic entering the data network has changed. Once the desired change has occurred, performing a linear increase of the bandwidth setting at the LAN egress port to a first value while monitoring voice quality statistics, determining if voice quality has degraded during increase of the bandwidth setting and repeating the last two steps if voice quality has not degraded. | 11-29-2012 |
| 20120300768 | PROVIDING TELECOMMUNICATION SERVICES BASED ON AN E.164 NUMBER MAPPING (ENUM) REQUEST - Data associated with an E.164 number mapping (ENUM) request can be received from an element of a telecommunications network. Logical trunk-group information and gateway information can be associated with the ENUM request. A call is managed by the computing element through a packet-based network based at least in part on the logical trunk-group information and the gateway information. | 11-29-2012 |
| 20120300769 | Real-Time VoIP Transmission Quality Predictor and Quality-Driven De-Jitter Buffer - Voice over Internet Protocol (VoIP) transmission quality predictor working in real-time provides feedback information regarding transmission impairments, transmission quality and end-user satisfaction to quality enhancement mechanisms along the transmission path. Quality enhancement mechanisms use this feedback information in the process of tuning their control parameters. The transmission quality predictor calculates the transmission rating factor R at each of the outputs based on the information regarding voice codec, packet loss, and mouth-to-ear delay. Information regarding voice codec and packet loss is determined from VoIP packet headers. Information regarding mouth-to-ear delay is determined from an additional time stamp that is inserted into each RTP packet header by the VoIP sender. Finally, the play-out delay of received VoIP packets for the actual time window is set equal to the play-out buffer output in the prior time window which gave the highest R-factor value. | 11-29-2012 |
| 20120300770 | METHOD AND APPARATUS FOR PROVIDING DISASTER RECOVERY USING NETWORK PEERING ARRANGEMENTS - The present invention enables network providers to create peering arrangements with other providers that allow them to fail over to other networks in the event of a site failure. This invention would lower the cost to provide site diversity within a provider's network by allowing cost sharing between the provider's network and other networks. For example, when an Application Server (AS) in a network fails, the network provider can send a call to a partner's network and uses an AS in the partner's network to process the call request. | 11-29-2012 |
| 20120314698 | LOCAL TERMINAL DEVICE AND METHOD FOR EXCHANGING VOIP SIGNALING - A local terminal device receives a local mail address of a local user, and converts the local mail address into a local domain name. The local terminal device transmits the local domain name and a local Internet protocol (IP) address to a dynamic domain name serve (DDNS) server to register with the DDNS server. The local terminal device receives a remote mail address of a remote user, and converts the remote mail address into a remote domain name. The local terminal device queries the DDNS server for a remote IP address corresponding to the remote domain name. The local terminal device exchanges VoIP signaling with a remote terminal device according to the remote IP address. | 12-13-2012 |
| 20120314699 | SYSTEM AND METHOD FOR LOCATION MANAGEMENT AND EMERGENCY SUPPORT FOR A VOICE OVER INTERNET PROTOCOL DEVICE - In a particular embodiment a system and method for processing a call in a Voice over internet protocol (VoIP) network are disclosed. The method includes receiving the call associated with a private identifier (PRID) at a server, classifying a call location based on the PRID and sending the call from the server to a Public Safety Answering Point (PSAP) with a call back number associated with the PRID. The system includes a first server interface for receiving a message containing a PRID associated with the call, a second server interface to access the data base for searching for the PRID in the data structure; a third server interface to receive an output from the database indicating whether a PRID has been found in the data structure; and a fourth server interface to send the call to a PSAP. | 12-13-2012 |
| 20120314700 | Methods and Apparatus for Enhancing the Scalability of IMS in VoIP Service Deployment - Methods for Enhancing the Scalability of IMS in VoIP Service Deployment lower the number of messages transmitted between functions of an IMS network. The number of messages transmitted between functions of an IMS network are lowered by storing and utilizing predetermined configuration information pertaining to the calling and called parties including the media and codecs the parties support. The predetermined configuration information, which may be based on a prior peering business agreement, supports the implementation of a one round procedure for establishing an IMS communication session. | 12-13-2012 |
| 20120314701 | System and Method of Providing Communication Service Using a Private Packet Network Backbone Exchange - Systems and methods for providing communication service using a private packet network backbone exchange (PPNBE) are disclosed. The private packet network backbone exchange may include a logical call control entity, a logical routing database, one or more PPNBE gateways, and a private packet network. The PPNBE may provide “one-hop” call connection between a call-originating entity and a call-terminating entity without traversal of any other exchanges. The PPNBE may simultaneously support both one-hop local and long distance calls. The PPNBE architecture is easily scalable and incorporated into existing communication networks for on-net and off-net service. Methods and systems for providing access tandem communication service using a PPNBE are also disclosed. | 12-13-2012 |
| 20120314702 | METHOD AND APPARATUS FOR PROGRAMMING SESSION INITIATION PROTOCOL BACK-TO-BACK USER AGENTS - In one embodiment, the present disclosure is a method and apparatus for programming session initiation protocol back-to-back user agents. In one embodiment, a method for programming a telecommunication feature as a session initiation protocol back-to-back user agent includes receiving source code defining the feature, the source code using at least one abstraction that hides session initiation protocol signaling details required by the feature and generating executable code that causes a session initiation protocol server to execute the feature, in accordance with the abstractions. | 12-13-2012 |
| 20120320903 | System and Method for Device Specific Customer Support - A method of routing a voice communication from an information handling system to one of a plurality of queues includes automatically obtaining an identifier from the information handling system that uniquely identifies the information handling system, and transmitting the identifier so that the voice communication can be routed to one of the queues. | 12-20-2012 |
| 20120320904 | Customer Support System and Method Therefor - A method of routing a voice communication from a web page to one of a plurality of queues includes determining whether a user has previously visited the web page. If the user has not previously visited the web page, the web page is displayed without a button to initiate the voice communication. If the user has previously visited the web page, the button is displayed. | 12-20-2012 |
| 20120320905 | System and Method for Routing Customer Support Softphone Call - A method of routing a voice communication from a web page to one of a plurality of queues includes determining a plurality of variables for routing the voice communication, and assigning a default value to each variable for the web page so that the voice communication can be routed to one of the queues regardless of whether the first value has been ascertained. | 12-20-2012 |
| 20120320906 | SYSTEM FOR COMMUNICATING BETWEEN INTERNET PROTOCOL MULTIMEDIA SUBSYSTEM NETWORKS - A method that incorporates teachings of the present disclosure may include, for example, receiving an assignment to provide communication services to a communication device, supplying a first telephone number mapping system of a first internet protocol multimedia subsystem communication system with contact information of the communication device and a serving call session control function operating in the first internet protocol multimedia subsystem communication system, supplying a second telephone number mapping system of a second internet protocol multimedia subsystem communication system with contact information of the communication device and the serving call session control function, and receiving a session initiation protocol INVITE from an originating serving call session control function of the second internet protocol multimedia subsystem communication system for establishing communications with the communication device. Additional embodiments are disclosed. | 12-20-2012 |
| 20120320907 | Systems and Methods of Providing Multi-Homed Tandem Access - Systems and methods for providing multi-homed tandem access in a communication system are disclosed. The disclosure may include a private packet network backbone exchange (PPNBE) in connection with a set of access tandems and with a call destination such as an end-user or a communications service. A set of LRNs may be homed across the set of access tandems, with each of the set of LRNs mapped to one or more TNs corresponding to the call destination. An originating party may use any of the TNs to reach the call destination. A plurality of originating calls each including one of the TNs may be received at the PPNBE from any of the access tandems and routed to the call destination. Thus, the present disclosure provides greater call capacity than available access tandem architectures as well as optimizes a maximum number of call paths to a particular call destination. | 12-20-2012 |
| 20120320908 | SYSTEM AND METHOD FOR ENABLING DTMF DETECTION IN A VOIP NETWORK - A method, mobile terminal, and system for selectively establishing an outgoing caller ID on a mobile terminal served by a wireless network, for identifying a line called on a mobile terminal, and for directing a call from a mobile terminal to a network subscriber based on accessed information of the subscriber in the subscriber's network. | 12-20-2012 |
| 20120327929 | Connection Switching Device and Telephone Device - A connection switching device for a telephone device supporting a PSTN function and a VoIP function and taking the PSTN function as a default voice communication function is disclosed. The connection switching device includes a dual tone multi-frequency (DTMF) receiver for determining whether a phone number inputted to the telephone device includes a specified trigger code, and a control unit coupled to the DTMF receiver for switching the default voice communication function from the PSTN function to the VoIP function when a decision result of the DTMF receiver indicates that the phone number includes the specified trigger code. | 12-27-2012 |
| 20130003718 | SELF-FORMING VOIP NETWORK - A self-forming VoIP connection capability is described that may be superimposed over wired networks, wireless networks, or combinations thereof. As described herein, a local network cluster forms while isolated from a conventional SIP server, or alternately may exist as a cluster of network nodes and clients that later becomes isolated from a conventional SIP server by a break in the network. Either way, each network node thus enabled with distributed SIP registry functionality according to this invention independently constructs a local SIP registry and SIP server capability within that node. Subsequently, while isolated from a conventional SIP server, VoIP conversations among client devices connected to nodes within an isolated cluster will continue, and nodes and clients may join or leave an isolated cluster with conversations able to be initiated or continued while a node has network connectivity to the cluster. | 01-03-2013 |
| 20130003719 | ENHANCED EMERGENCY SERVICES FOR FIXED WIRELESS CUSTOMER PREMISES EQUIPMENT - A gateway device, provided in a customer premises, receives a call from a user device, and detects dialed information associated with the call. The gateway device identifies the call as an emergency call based on the dialed information, and terminates all other calls communicated by the gateway device except for the emergency call. The gateway device notifies an outdoor broadband unit, associated with the customer premises, about the emergency call. | 01-03-2013 |
| 20130003720 | SYSTEM FOR AD-HOC COMMUNICATION SESSIONS - In one implementation, a guest device on an ad hoc network is permitted to initiate a communication session through a packet switched network depending on the destination endpoint of the communication session. A network device maintains a list of approved destination endpoints, which may be identified by telephone numbers, addresses, or uniform resource identifiers. The approved destination endpoints correspond to services that are offered to users of guest devices, such as voicemail, videoconferencing, or customer service. The network device receives a request for a communication session from a guest device, and the request includes data indicative of a destination endpoint. The network device compares the data indicative of the destination endpoint to the list of approved destination endpoints. If there is a match, the request is forwarded to a next hop router. If there is no match the request is dropped or returned to the guest device. | 01-03-2013 |
| 20130003721 | VOIP SERVER AND METHOD FOR MANAGING GEOGRAPHICAL INFORMATION - A method for managing geographical information of a plurality of Voice over Internet Protocol (VoIP) terminal devices using a VoIP server receives location information of the VoIP terminal devices from the VoIP terminal devices, stores geographical information of the VoIP terminal devices into a data sheet stored in a storage device of the VoIP server. The method detects a query from one of the VoIP terminal devices, retrieves the geographical information from the data sheet according to the query, and obtains an electronic map from the storage device. The method further populates the retrieved geographical information on the electronic map, and sends the electronic map to the VoIP terminal device corresponding to the location information. | 01-03-2013 |
| 20130003722 | VOICE COMMUNICATION OF DIGITS - The present invention relates to SIP networks and, more particularly, to digit collection in SIP networks. The SIP user communicates the digits to a Media Server through voice/speech. The Media Server collects the digits and checks to determine if the digits satisfy required Dual Tone Multi Frequency rules. The Media Server plays a prompt message to the SIP user to indicate start of session to collect the digits and the SEP user says character(s) to indicate that the SIP user has completed saying the digits. | 01-03-2013 |
| 20130003723 | Virtual PBX based on Feature Server Modules - A virtual private branch exchange is formed by a plurality of interconnected feature server modules, each having an integral feature server that is configured and operates independently of the other feature server modules. Within a virtual private branch exchange, the feature server modules may be logically arranged in a hierarchy having at least a main feature server module and one or more subordinate feature server modules. A particular feature server module may operate in multiple virtual private branch exchanges, and may have a distinct set of rules for handling calls originating in different virtual private branch exchanges. | 01-03-2013 |
| 20130003724 | METHOD AND APPARATUS FOR PROVIDING SHARED SERVICES - The present invention enables an overlay capability to be invoked on network systems and elements that are designed to support multiple customer bases. Depending on the registered identification of the user, screens and other user interfaces that provide access to functions can be overlaid on the network component and segmented along customer classifications. | 01-03-2013 |
| 20130010782 | METHOD AND APPARATUS FOR PERSISTENT ANCHORING OF INTERNET PROTOCOL DEVICES - In one embodiment, a method includes detecting a presence of a first endpoint on a port, and determining when the first endpoint is of a first type after detecting the presence of the first endpoint on the port. The port is associated with a switch that is part of a network and has a fixed address. The switch is arranged to support the first type. The method also includes mapping the first endpoint to the fixed address when it is determined that the endpoint is of the first type, wherein mapping the endpoint to the fixed address includes identifying the first endpoint by the fixed address within the network. | 01-10-2013 |
| 20130010783 | METHOD AND APPARATUS FOR PROVIDING ACCESS AND EGRESS UNIFORM RESOURCE IDENTIFIERS FOR ROUTING - A method and apparatus for providing routing of calls in a packet network, using one or more criteria extracted from signaling information to determine the routing for the calls are disclosed. The routing criteria extracted from signaling messages comprises at least one of: an access Uniform Resource Identifier, a destination phone number, a destination URI host, a calling party number, a calling party URI host, an incoming IP address, or a requested codec. An access URI and the egress URI are used to enhance routing decisions in a VoIP network. The egress URI can be used to specify egress route selections from the egress point of a VoIP network. The access URI can be used to influence the routing decisions within the VoIP network as well as the routing decisions with regard to egress routes from the egress point of the VoIP network. | 01-10-2013 |
| 20130010784 | METHOD AND APPARATUS FOR RE-ORIGINATING CALLS - A method and apparatus for enabling a subscriber who is originating a call to a called party endpoint to specify call handling treatments when a busy or a no answer network condition is encountered when calling the called party endpoint with no subscribed network based voice mail service are disclosed. The subscriber, for example, can re-originate the call that is originally directed to a home phone number of the called party to a cellular phone number of the called party instead upon encountering a busy or no answer network condition. | 01-10-2013 |
| 20130010785 | Bearer Path Optimization - Call control for originating and terminating calls in a visited circuit-switched subsystem (CS) or home multimedia subsystem (MS) as well as transferring calls between the visited CS and the home MS may be anchored at a continuity control function (CCF) in the home MS. Call signaling for the call may be passed through the CCF. When the user element is homed to the home MS and served by the visited CS, the bearer path for the call is established based on the relative proximity of the home MS and the visited CS. When a local MS is more proximate to the visited CS, the bearer path may be routed through a gateway in the local MS, instead of through the gateway in the home MS. When the home MS is sufficiently proximate to the visited CS, the bearer path is routed through the gateway in the home MS. | 01-10-2013 |
| 20130016715 | NETWORK TELEPHONY APPLIANCE AND SYSTEM FOR INTER/INTRANET TELEPHONY - A network appliance ( | 01-17-2013 |
| 20130016716 | APPARATUS FOR REMOTELY REBOOTING VoIP COMMUNICATION DEVICES AND AN ASSOCIATED METHOD AND COMPUTER PROGRAM PRODUCT - An apparatus is provided for remotely rebooting Voice over Internet Protocol (VoIP) communication devices. In general, the apparatus remotely selects VoIP communication devices connected to a network, reboots the selected devices, and evaluates the status of each device. A processor allows a user to select VoIP communication devices connected to the network and receives inputs pertaining to the reboot operation, including a time input and search criteria. The processor communicates with the designated VoIP communication devices over a packet-switching network to instruct the devices to reboot and monitors each device. In this way, multiple VoIP communication devices may be rebooted from a remote location, and problems or issues that arise during the reboot process may be identified and addressed. An associated method and computer program product are also provided for remotely rebooting VoIP communication devices. | 01-17-2013 |
| 20130022036 | SYSTEM AND METHOD OF PROVIDING A HIGH-QUALITY VOICE NETWORK ARCHITECTURE - Providing high quality voice/sound communications over a local loop of a telephone network is disclosed. The method includes receiving a voice signal, digitizing the voice signal into a high quality voice signal, utilizing sampling rates and/or sizes above the threshold, negotiating voice processing characteristics between a customer premises equipment and a network element, receiving speech at a customer premises equipment according to the negotiation, converting the received speech into high bandwidth signal and transmitting the high bandwidth signal to a telephone local loop, transmitting the high bandwidth signal froth the local loop to wideband node that packetizes the high bandwidth signal for transmission to a packet network and receiving the packetized signal from the packet network at a switch that switches between an on-network or off-network status. A voice over IP platform can route packetized signals from the packet network to the telephone network or another packet network. | 01-24-2013 |
| 20130022037 | METHOD, TERMINAL DEVICE, AND SYSTEM FOR ESTABLISHING A COMMUNICATION BETWEEN A FIRST PARTY AND A SECOND PARTY - A method for establishing a communication between a first party and a second party includes: receiving, by the second party, a communication request (CR) and identification information of the first party from the first party using a first signalization standard and using public identification information of the second party; and establishing direct communication between the first and the second party using a second signalization standard and using the identification information of the first party. The first signalization standard corresponds to a Public Switched Telephone Network (PSTN) or Public Land Mobile Network (PLMN) service or to Internet Protocol. The second signalization standard corresponds to Internet Protocol. | 01-24-2013 |
| 20130022038 | ENHANCED TELEPHONY COMPUTER USER INTERFACE ALLOWING USER INTERACTION AND CONTROL OF A TELEPHONE USING A PERSONAL COMPUTER - Enhanced telephony computer user interfaces seamlessly integrate and leverage the features of personal computers and telephones. The manner in which media is presented at a computing system can also be modified automatically in response to detected telephone operations. These modifications can include pausing media in response to a detected telephone call and/or adjusting a volume of the media presentation. The media presentation/volume can also be resumed/restored upon detecting that the telephone call has terminated. | 01-24-2013 |
| 20130022039 | Circuit-Switched and Multimedia Subsystem Voice Continuity with Bearer Path Interruption - The present invention maintains calls during transfers between different types of subsystems, even when there is a bearer path interruption at the user element due to the original connection from the transferring-out subsystem being dropped before the new connection in the transferring-in subsystem is established. The call signaling leg toward the remote endpoint of the remote party is held, while the call signaling leg toward the user element is moved from the transferring-out subsystem to the transferring-in subsystem and a new bearer path is established via the transferring-in subsystem. During transfers with a bearer path interruption, a portion of the bearer path leading to the remote endpoint may be connected to a media resource function, which will provide an announcement to the remote party. Once the user element is accessible in the transferring-in subsystem, the bearer path is further transferred to the user element via the transferring-in subsystem. | 01-24-2013 |
| 20130028250 | SYSTEMS AND METHODS OF PROVIDING COMMUNICATIONS SERVICES - An IP telephony system allows users of the IP telephony system to register extension telephony devices with the IP telephony system. An extension telephony device is one that is provided with service by a separate telephony service provider. Once an extension telephony device is registered, a user can obtain communications services from the IP telephony system using the extension telephony device. A extension telephony device may be tied to a user's main telephony services account with the IP telephony system such that when the user obtains communications services from the IP telephony system using an extension telephony device, the user will be billed for those communications services through the user's main account. | 01-31-2013 |
| 20130028251 | SYSTEM AND METHOD FOR PROCESSING TELEPHONY SESSIONS - In one embodiment, the method of processing telephony sessions includes: communicating with an application server using an application layer protocol; processing telephony instructions with a call router; and creating call router resources accessible through a call router Application Programming Interface (API). In another embodiment, the system for processing telephony sessions includes: a call router, a URI for an application server, a telephony instruction executed by the call router, and a call router API resource. | 01-31-2013 |
| 20130028252 | COMPUTER TELEPHONY - Computer-telephony events are initiated by a user operating a computer terminal to control, through a computer-telephony controller, operation of a communications terminal. The computer-telephony controller receives from a first computer terminal a request for a first computer-telephony event; in which the request for the first computer-telephony event comprises a label value. The computer-telephony controller refers to a mapping between the label value and an identifier of a communications terminal; and initiates the requested first computer-telephony event controlling operation of the communications terminal. The computer-telephony controller then receives, from second computer terminal | 01-31-2013 |
| 20130034093 | Policy Rule Management For QoS Provisioning - Described herein is a policy-based Internet Protocol (IP) network wherein the Quality of Service (QoS) provisioning across various network devices is managed by policy processing via a user interface including a graphic user interface. The user interface incorporates information made available by a server, such as lightweight directory access protocol (LDAP) server, having a repository, and thereby allows for a consistent set up voice-over IP devices, video devices and network data devices with minimal entries by the user. Further, the user interfaces allows for efficient policy creation and editing. | 02-07-2013 |
| 20130039360 | HYBRID UNIFIED COMMUNICATIONS DEPLOYMENT BETWEEN CLOUD AND ON-PREMISE - A hybrid Unified Communications (UC) telephony deployment includes users of a tenant that are hosted between a UC cloud deployment and a UC on-premise deployment that offers PSTN connectivity for the users. An identity of a tenant and its' users are maintained consistently between the on-premise and cloud based UC deployment (e.g. telephone numbers, dialing preferences, voice mail . . . ). Each user of the tenant can register with the UC service from one or more locations (e.g. on-premise, off-premise . . . ) whether or not they are hosted by the on-premise deployment or the cloud based UC deployment. Functionality of the UC deployment may also be maintained at one or more locations within the hybrid UC deployment. For example, some services (e.g. voicemail, or other services) can be hosted in the cloud while the remaining services are hosted on-premise. Different Telco providers may be chosen by the tenant to provide PSTN services for one or more users of the tenant. | 02-14-2013 |
| 20130039361 | User Equipment and Method for Executing a Service - User equipment and method for executing an application, which uses application data, and which is executed in an Application Server in a VoIP based telecommunications network. The method comprises providing a user equipment including a database having the data stored therein, wherein the user equipment further includes an Application Server. The method further comprises invoking the SIP-AS included in the user equipment by a network node of the telecommunications network, providing, within the user equipment, the data to the Application Server included in the user equipment, executing the application, using the data, by the Application Server included in the user equipment, and communicating a result of executing the application from the Application Server included in the user equipment to the network. | 02-14-2013 |
| 20130039362 | INLINE POWER SYSTEM AND METHOD FOR NETWORK COMMUNICATIONS - A system and method for coupling a communications device to a primary communications network having a first communications format and to a secondary communications network having a second communications format. The system and method can comprise a first port configured for connecting to the communications device, a second port configured for connecting to the primary communications network and facilitating the communication of the data between the primary communications network and the communications device through the first port. | 02-14-2013 |
| 20130044746 | PRIVATE IP COMMUNICATION NETWORK ARCHITECTURE - A disclosed Internet Linked Network Architecture delivers telecommunication type services across a network utilizing digital technology. The unique breadth and flexibility of telecommunication services offered by the Internet Linked Network Architecture flow directly from the network over which they are delivered and the underlying design principles and architectural decisions employed during its creation. The present invention supports current telecommunication and voice over IP standards and applications. This new network not only replaces the telecommunication network presently in place, but it also offers a more feature rich and cost effective alternative. For example, traditional telecommunication switches are more expensive, less reliable and slower than the faster digital data switches utilized in the present invention. Furthermore, the programmable nature of the digital devices comprising the present invention allows the new network to be built with a scalable and extensible architecture, providing the flexibility necessary to incorporate new or future digital enhancements. | 02-21-2013 |
| 20130044747 | NETWORKS COMMUNICATIONS BANDWIDTH MANAGER CONTROL SYSTEM PROVIDING FOR ASSOCIATED COMMUNICATIONS PROCESSING RESOURCES - Systems and methods for network communications bandwidth management and in particular to unified bandwidth manager that interfaces with and hierarchically manages a plurality of service-specific bandwidth reservation and session management systems. By utilizing a novel bandwidth management system, a better purpose specific bandwidth reservation system may thereby be achieved. | 02-21-2013 |
| 20130058325 | METHOD AND APPARATUS FOR PROVIDING RINGING TIMEOUT DISCONNECT SUPERVISION IN REMOTE TELEPHONE EXTENSIONS USING VOICE OVER PACKET-DATA-NETWORK SYSTEMS (VOPS) - A Multiservice Access Concentrator (MAC) provides a time limit for a first ringing voltage signal in response to an attempted call. The call is attempted via a voice over packet-data-network system (VOPS), wherein the VOPS comprises voice over Internet Protocol (IP), voice over Frame Relay, voice over Asynchronous Transfer Mode (ATM), and voice over High-level Data Link Control (HDLC) network systems. Generation of the first ringing voltage signal is terminated upon expiration of the time limit. A control message is transmitted to terminate the attempted call, wherein the control message is transmitted via the VOPS. | 03-07-2013 |
| 20130058326 | INTERNET TELEPHONY UNIT AND SOFTWARE FOR ENABLING INTERNET TELEPHONE ACCESS FROM TRADITIONAL TELEPHONE INTERFACE - Automatic selection and establishment of a communications connection between a telephone device to a receiver device, including entering an address of a receiver device into the telephone device for initiating the communications connection to the receiver device, and automatically selecting a communications network for establishing the communications connection to the receiver device, and selecting the communications network from an internet-based network, a hybrid telephone/internet network, and a telephone network. Automatically determine network access capabilities of the receiver device based on the address of the receiver device, and automatically evaluate the cost of establishing a communications connection for each of the communications networks which the receiver device is capable of accessing. The communications network with the lowest cost is selected. | 03-07-2013 |
| 20130058327 | METHOD FOR PROVIDING EARLY-MEDIA SERVICE BASED ON SESSION INITIATION PROTOCOL - Disclosed is a method of providing an early media service based on a session initiation protocol (SIP). More particularly, the present disclosure relates to a method of providing an early media service based on SIP, wherein an application server and a media server can provide early media of a multimedia form, such as images, moving images and the like, as well as audios, by using an early session or a regular session. | 03-07-2013 |
| 20130058328 | TELECOMMUNICATION AND MULTIMEDIA MANAGEMENT METHOD AND APPARATUS - A telecommunication and multimedia management apparatus and method that supports voice and other media communications and that enables users to: (i) participate in multiple conversation modes, including live phone calls, conference calls, instant voice messaging or tactical communications; (ii) review the messages of conversations in either a live mode or a time-shifted mode and to seamlessly transition back and forth between the two modes; (iii) participate in multiple conversations either concurrently or simultaneously; (iv) archive the messages of conversations for later review or processing; and (v) persistently store media either created or received on the communication devices of users. The latter feature enables users to generate or review media when either disconnected from the network or network conditions are poor and to optimize the delivery of media over the network based on network conditions and the intention of the users participating in conversations. | 03-07-2013 |
| 20130064240 | Systems and methods for multiple mode voice and data communications using intelligenty bridged TDM and packet buses - In a communications system, a first packet network is provided. Packetized data is transferred between the system and one or more packet-based devices. A TDM network is provided, and data is transmitted in frames having slots. Data transmitted via the TDM network includes data for voice communications for telephony devices. The TDM network is selectively coupled to the first packet network and a WAN. A processor and a control bus interface circuit control transfer of packetized data and transmittal of data for voice communications. The processor controls processing of packetized data and data'for voice communications. A switch/multiplexer selectively controls providing data to/from particular slots. The processor selectively controls voice communications from telephony devices over the TDM network and packet-based communications over the packet network. Voice communications that stay in a circuit-switched form occur over the TDM network and the WAN. | 03-14-2013 |
| 20130064241 | AGILE NETWORK PROTOCOL FOR SECURE COMMUNICATIONS USING SECURE DOMAIN NAMES - A network device comprises a storage device storing an application program for a secure communications service, and at least one processor configured to execute the application program for the secure communications service so as to enable the network device to send a request to look up a network address of a second device based on an identifier associated with the second device, receive an indication that the second device is available for the secure communications service, the indication including the requested network address and provisioning information for a secure communication link, connect to the second device over the secure communication link, using the received network address of the second device and the provisioning information for the secure communication link, and communicate at least one of video data and audio data with the second device using the secure communications service via the secure communication link. | 03-14-2013 |
| 20130064242 | METHODS AND APPARATUS TO MEASURE MARKET SHARE FOR VOICE OVER INTERNET PROTOCOL CARRIERS - Methods and apparatus to measure market share for VoIP carriers is disclosed. An example method includes querying a plurality of VoIP carrier servers to determine the VoIP carrier server that owns the telephone subscriber number (SN), in response to the querying, receiving a plurality of messages operable to determine whether the telephone SN is found within any one of the plurality of VoIP carrier servers, when the received plurality of messages is at least one of inconclusive or when the telephone SN is not found within any one of the plurality of VoIP carrier servers, placing a first partial call to the telephone SN from a first VoIP number within a first VoIP carrier network, receiving a first signal from the first VoIP carrier network, and based on the first received signal, determining whether the telephone SN belongs to the first VoIP carrier network. | 03-14-2013 |
| 20130070755 | SYSTEMS AND METHODS OF ROUTING IP TELEPHONY DATA PACKET COMMUNICATIONS - Systems and methods of selecting a media path for data packets bearing the media of a telephone call to traverse during a voice over Internet protocol telephone call include testing the call quality of multiple potential media paths immediately before a call is setup. The potential media path with the highest call quality is used as the initial media path for the call. If the call quality of the media path currently in use declines below a threshold value during a call, potential alternate media paths are identified and tested for call quality. If one of the alternate media paths has better call quality than the meida path presently in use, the call is switched to the alternate media path with the highest call quality. | 03-21-2013 |
| 20130070756 | Method, System and Software for Establishing a Communication Channel Over a Communications Network - The establishment of a VoIP connection between first and second telecommunication devices ( | 03-21-2013 |
| 20130070757 | VOICE OVER DATA TELECOMMUNICATIONS NETWORK ARCHITECTURE - The present invention describes a system and method for communicating voice and data over a packet-switched network that is adapted to coexist and communicate with a legacy PSTN. The system permits packet switching of voice calls and data calls through a data network from and to any of a LEC, a customer facility or a direct IP connection on the data network. The system includes soft switch sites, gateway sites, a data network, a provisioning component, a network event component and a network management component. The system interfaces with customer facilities (e.g., a PBX), carrier facilities (e.g., a LEC) and legacy signaling networks (e.g., SS7) to handle calls between any combination of on-network and off-network callers. | 03-21-2013 |
| 20130070758 | Systems and Methods for Transmitting Subject Line Messages - A method includes receiving a call setup signaling message at a subject line messaging application server. The call setup signaling message includes a subject header. The subject header includes a message identifier that corresponds to a subject line message selected by a caller device. The method includes replacing the message identifier with the subject line message to form a modified call setup signaling message when a removal determination indicates that the subject header should remain. The method also includes sending the modified call setup signaling message to a called party device via a server. | 03-21-2013 |
| 20130070759 | INSTANT INTERNET BROWSER BASED VoIP SYSTEM - The present invention is an instant Internet browser based VoIP system with a VoIP client in the form of temporary VoIP applets that can start in a Web browser and can establish an instant peer-to-peer connection with another web-based or hardware embedded/installed VoIP client using session initiation protocol (SIP) and real-time transport protocol (RTP) audio streaming. The applet is a small file that is easily loaded onto a user's browser and uses application program interfaces (APIs) that require no additional libraries. The applet is written in JAVA, although other programming languages may also be used to write the applet. | 03-21-2013 |
| 20130077617 | SYSTEM AND METHOD FOR SPLIT SIP - Disclosed herein are systems, methods, and non-transitory computer-readable storage media for splitting SIP back-to-back user agents and converting SIP communications between a back-to-back user agent server and a back-to-back user agent client to HTTP requests while preserving SIP headers. The back-to-back user agent server receives a SIP invite from a caller and converts the SIP invite to an HTTP request, wherein headers from the SIP invite are preserved in the HTTP request. The server transmits the HTTP request to a user agent client via a wide area network connection and receives, from the user agent client, an HTTP response to the HTTP request. The server converts the HTTP response to a SIP response, and transmits the SIP response to the caller, wherein the SIP response contains instructions for establishing SIP communications between the caller and a callee via a network. | 03-28-2013 |
| 20130077618 | EXPEDITIOUS RESOURCE RESERVATION PROTOCOL - Techniques are provided for making reservations between a calling device and one or more called devices using the Expeditious Resource Reservation Protocol (E-RSVP). An example technique involves sending an invite message from the calling device to one or more called devices over a data network. In response to the invite message, each of the called devices transmits a resource reservation message to the calling device. The resource reservation messages may be, for example, a Resource Reservation Protocol (RSVP) message. Based on the information in the received resource reservation messages, the calling device establishes reservations with each of the called devices. | 03-28-2013 |
| 20130077619 | COMMUNICATION APPARATUS AND COMMUNICATION SYSTEM - A communication apparatus includes: a capability-information generating section which generates a first capability information including a first receivable-period information in relation to a first receivable period; a first transmission section which transmits the first capability information to the communication-destination device via the IP network; a first receiving section which receives a first data transmitted from the communication-destination device via the IP network during the first receivable period; a storage section in which a second data to be transmitted to the communication-destination device is stored; a second receiving section which receives a second capability information including a second receivable-period information in relation to a second receivable period; and a second transmission section which transmits a second data stored in the storage section to the communication-destination device during the second receivable period based on the second capability information. | 03-28-2013 |
| 20130077620 | SYSTEM AND METHOD FOR INDICATING CIRCUIT SWITCHED ACCESS AT IMS REGISTRATION - In IP Multimedia Subsystem (IMS) IMS Control Channel Protocol (ICCP) is used between a user equipment (UE) and IMS Control Channel Function (ICCF) and Session Initiated Protocol (SIP) interface (between to ICCF, Call Session Control Function and Application Server) to support the indication of Circuit Switched (CS) access in header information. The indication can be used by an S-CSCF or AS for different purposes such as routing decision, charging, and presence info. | 03-28-2013 |
| 20130083792 | VOICE OVER INTERNET PROTOCOL SESSION IDENTIFIERS FOR VOICE OVER INTERNET PROTOCOL CALLS - Embodiments of the present disclosure describe methods, apparatuses, and systems for voice session identifiers to facilitate voice over Internet protocol calls. | 04-04-2013 |
| 20130089086 | REGIONAL INDEPENDENT TANDEM TELEPHONE SWITCH - Implementations of the present disclosure involve an apparatus and/or method for a regional independent tandem switch of a telecommunications network. The tandem switch processes communications between a long distance carrier and a local exchange carrier through a voice over IP (VOIP) network. By utilizing the VOIP network, the regionally independent tandem may process long distance communications to and from any point in the network, regardless of the physical proximity of the tandem to the originating/destination communication device. The regionally independent nature of the tandem also allows for flexibility in communication routing through the network, load balancing between the network tandem switches and reduction of needed components of the network for proper processing of the long distance communications. | 04-11-2013 |
| 20130089087 | IMS and Method of Multiple S-CSCF Operation in Support of Single PUID - A method for providing multimedia services to subscriber user equipment (UE) within an IP multimedia subsystem network (IMS) may include configuring the IMS to enable a single UE to fork register and cooperate with multiple serving-call session control functions (S-CSCFs). After obtaining IP connectivity, the single UE signals to register with the IMS and the IMS determines whether the UE is configured to fork register with multiple S-CSCFs. If the UE is configured, the IMS determines which S-CSCFs are eligible for the UE registration and fork registers the UE to multiple S-CSCFs of the eligible S-CSCFs. Consequently, incoming and outgoing calls to/from the UE are routed by the IMS to any of the multiple registered S-CSCFs. | 04-11-2013 |
| 20130094494 | Method And Apparatus For Interworking Voice And Multimedia Services Between CSI Terminal And IMS Terminal - A method and apparatus is provided for communication between a first terminal capable of using both a Circuit Switched (CS) call and an Internet Protocol Multimedia Subsystem (IMS) session, and a second terminal capable of using the IMS session, in a communication system supporting a Combined CS call and IMS session (CSI service). If the second terminal sends a Session Initiation Protocol (SIP) request (INVITE) message to originate a voice service or a multimedia service including the voice service with the first terminal, a CSI Application Server (AS) of an IMS domain managing the first terminal separates a voice service-related component included in the request message from a multimedia service-related component, and generates and sends first and second request messages to the first terminal. Upon receipt of first and second response messages corresponding to the first and second request messages from the first terminal, the CSI AS generates a combined response message and sends the combined response message to the second terminal. | 04-18-2013 |
| 20130094495 | SESSION TRANSFER METHOD, APPLICATION SERVER, AND COMMUNICATIONS SYSTEM - Embodiments of the present invention relate to the field of communications and provide a session transfer method, an application server, and a communications system, which implement a cross-network session transfer between a CS domain and an IMS domain. The method includes: after receiving a session transfer request sent by user B or user C in the IMS domain, sending a release message to the user B currently in a call to terminate a session between user A in the CS domain and the user B, and negotiating media information of the user A and media information of the user C to establish a session between the user A and the user C. The embodiments of the present invention apply to a cross-network session between the CS domain and the IMS domain. | 04-18-2013 |
| 20130094496 | METHOD AND APPARATUS FOR ENABLING PHONE NUMBER DIALING USING EMAIL ADDRESSES - The invention comprises a method and apparatus for enabling a subscriber to establish a call to at least one destination phone number using an email address. Specifically, the method comprises receiving an incoming communication request from the subscriber for establishing the call (where the incoming communication request comprises an email address), determining at least one destination phone number associated with the email address, and processing the call using at least one of the at least one destination phone number associated with the email address. | 04-18-2013 |
| 20130094497 | METHOD AND APPARATUS FOR SENDING ALERTS TO INTERNET PROTOCOL PHONES - The present invention enables an alert message and the display of calling party identity on all on-hook phones associated with an extension sharing the same phone number, when one phone is off-hook and in use. In one exemplary embodiment, this capability enables all other members of a household to receive information regarding an incoming call even when one phone is in use by another member. | 04-18-2013 |
| 20130100950 | METHOD, USER EQUIPMENT AND SERVER FOR MULTIMEDIA SESSION TRANSFER - The present invention discloses a method, User Equipment (UE), and server for multimedia session transfer, and relates to a mobile communication technology, and in particular, to a technology for transferring multimedia sessions from a Circuit Switched (CS) network to a Packet Switched (PS) network. The method includes: receiving a session transfer request sent by a UE, where the session transfer request carries a static Session Transfer Identifier (STI); executing a procedure for transferring the active CS session according to the CS session transfer request, sending the dynamic STI corresponding to the held CS session to the UE, and receiving the request for transferring the held CS session and executing a procedure for transferring the held CS session. Further, a UE and a server are provided. | 04-25-2013 |
| 20130107872 | Processor-memory module performance acceleration in fabric-backplane enterprise servers | 05-02-2013 |
| 20130107873 | System for Interconnecting Standard Telephony Communications Equipment to Internet | 05-02-2013 |
| 20130107874 | INTELLIGENT END USER DEVICES FOR CLEARINGHOUSE SERVICES IN AN INTERNET TELEPHONY SYSTEM | 05-02-2013 |
| 20130107875 | METHOD AND SYSTEM FOR A COMMUNICATION SESSION INITIALIZATION IN A TELECOMMUNICATION NETWORK | 05-02-2013 |
| 20130107876 | System and Method for Originating a Call via a Circuit-Switched Network from a User Equipment Device | 05-02-2013 |
| 20130107877 | IP TELEPHONE SYSTEM AND METHOD | 05-02-2013 |
| 20130114589 | SYSTEM AND METHOD FOR CONFIGURING AN IP TELEPHONY DEVICE - A system and method for IP telephony are disclosed. The system includes an IP telephone (IPT) and a Service Gateway (SG). The SG receives an identifier, e.g., a vendor class identifier, included in a DHCP discover message from the IP telephone and determines if the identifier is valid. If so, the SG issues a DHCP offer comprising DHCP lease information to the IP telephone, including a range of port numbers assigned to the IP telephone based on the identifier, where the range of port numbers comprises ports which are not reserved for use by other IP protocols. The DHCP lease information includes information indicating operational software for the IP telephone which the IP telephone executes to enable IP communications. The SG mediates IP communications between the IP telephone and an IP device, where the IP telephone uses at least a subset of the range of port numbers to send or receive IP communications. | 05-09-2013 |
| 20130114590 | SYSTEMS AND METHODS OF PROVIDING COMMUNICATIONS SERVICES - An IP telephony system allows a user to register a telephony device that receives its native telephony service from a different telephony service provider as an extension telephone. The user can then place calls through the IP telephony system using the extension telephone. Such calls may or may not be established using the extension telephone's native telephony service provider. | 05-09-2013 |
| 20130114591 | System for transporting ethernet frames over very high speed digital subscriber lines - An apparatus for and method of encapsulating Ethernet frames over a Very high speed Digital Subscriber Line (VDSL) transport facility. The VDSL frames are transmitted over a point to point VDSL link where they are subsequently extracted and forwarded as standard Ethernet frames. The VDSL facility transport system comprises an Ethernet to VDSL Customer Premises Equipment (CPE) coupled to a DSL Access Multiplexor (DSLAM) over a VDSL transport facility. The Ethernet to VDSL CPE functions to receive a 10BaseT Ethernet signal and encapsulate the Ethernet frame into a VDSL frame for transmission over the VDSL facility. The DSLAM is adapted to receive VDSL frames, extract Ethernet frames therefrom and generate and output a standard Ethernet signal. Ethernet frames are encapsulated within VDSL frames and transmitted on the wire pair without regard to the state of the SOC signals. | 05-09-2013 |
| 20130114592 | IMS CALL ROUTING USING TEL-URIs - The present invention proposes a specific handling of tel URIs in an IMS terminating network so as to enable routing of calls using telephone numbers (and not SIP URIs with embedded telephone numbers) as identifiers of the target users of those calls. Specifically, the present invention introduces a conversion module which is located within the IMS terminating network and is capable of converting SIP URIs with embedded telephone numbers into equivalent tel URIs which are then used by a terminating I-CSCF and S-CSCF to query the SLF and/or HSS so that, they can route the calls to the target users. | 05-09-2013 |
| 20130121332 | SYSTEM AND METHOD FOR FACILITATING VOIP COMMUNICATIONS - A method for facilitating VoIP communication between VoIP providers. First and second VoIP service providers are registered with a VoIP communication system, which generates a table of registered VoIP subscribers from the first and second VoIP service providers. The information from the table is applied to a call generated from a subscriber on the first VoIP service provider, such that if the desired party is a subscriber to the second VoIP service provider as noted in the table, the call generated from the subscriber on the first VoIP service provider is sent to the desired party as a packet switched call. | 05-16-2013 |
| 20130121333 | METHODS AND APPARATUS TO CONTROL A FLASH CROWD EVENT IN A VOICE OVER INTERNET PROTOCOL (VOIP) NETWORK - Methods and apparatus to control a flash crowd event in a voice over Internet protocol (VoIP) network are disclosed. An example method comprises receiving at a VoIP border element a VoIP registration response message having a field representing a priority assigned to a VoIP endpoint, receiving a message from the VoIP endpoint at the VoIP border element, detecting whether a network congestion condition exists, and placing the message received from the VoIP endpoint into one of a plurality of queues based on the priority when the congestion condition is detected. | 05-16-2013 |
| 20130121334 | METHOD FOR REQUESTING DOMAIN TRANSFER AND TERMINAL AND SERVER THEREOF - A method, terminal and server for controlling a domain transfer operation, are discussed. According to an embodiment, the method includes determining, by a terminal, whether a session is transferred from a first domain to a second domain, wherein the determining is based on an operator policy and radio conditions, wherein the operator policy is received by the terminal from a network, and wherein the operator policy includes at least one of first information indicating an operator's preferred domain and second information indicating whether to initiate the domain transfer in a short time. | 05-16-2013 |
| 20130128879 | Local Identity Based On Called Number - Methods, devices, and storage media provide for receiving a message pertaining to a telephone call set-up; identifying in the message a called telephone number; selecting a calling telephone number based on the called telephone number; replacing a calling telephone number included in the message with the selected calling telephone number; and transmitting the message that includes the selected calling telephone number. | 05-23-2013 |
| 20130128880 | System and Method for Dynamic Telephony Resource Allocation Between Premise and Hosted Facilities - A population of networked Application Gateway Centers or voice centers provides telephony resources. The telephony application for a call number is typically created by a user in XML (Extended Markup Language) with predefined telephony XML tags and deployed on a website. A voice center provides facility for retrieving the associated XML application from its website and processing the call accordingly. The individual voice centers are either operated at a hosted facility or at a customer's premise. Provisioning Management Servers help to allocate telephony resources among the voice centers. This is accomplished by suitably updating a voice center directory. In this way, the original capacity at a premise, predetermined by the hardware installed, can be adjusted up or down. If the premise is under capacity, it can be supplemented by that from a hosted facility. If the premise has surplus capacity, it can be reallocated for use by others outside the premise. | 05-23-2013 |
| 20130128881 | Method and System of Voice Carry Over for Instant Messaging Relay Services - A method of assisting communication for a user is provided. The method includes receiving an IM message including a request for a voice carry over from the user, and transmitting to the user an invitation to join a first voice connection. The method further includes initiating the first voice connection with the user, and initiating a second voice connection with a recipient. Additionally, the method includes communicating to the recipient a first voice communication from the user over the first and second voice connections, and communicating to the user a response IM message including a transcribed version of a second voice communication from the recipient. An apparatus for assisting communication for a user is provided. A computer-readable medium having stored thereon computer-executable instructions is provided. The computer-executable instructions cause a processor to perform a method when executed. | 05-23-2013 |
| 20130128882 | SYSTEM AND METHOD FOR PROCESSING TELEPHONY SESSIONS - In one embodiment, the method of processing telephony sessions includes: communicating with an application server using an application layer protocol; processing telephony instructions with a call router; and creating call router resources accessible through a call router Application Programming Interface (API). In another embodiment, the system for processing telephony sessions includes: a call router, a URI for an application server, a telephony instruction executed by the call router, and a call router API resource. | 05-23-2013 |
| 20130128883 | SYSTEM AND METHOD FOR PROCESSING TELEPHONY SESSIONS - In one embodiment, the method of processing telephony sessions includes: communicating with an application server using an application layer protocol; processing telephony instructions with a call router; and creating call router resources accessible through a call router Application Programming Interface (API). In another embodiment, the system for processing telephony sessions includes: a call router, a URI for an application server, a telephony instruction executed by the call router, and a call router API resource. | 05-23-2013 |
| 20130136120 | Client Routing in a Peer-to-Peer Overlay Network - A method of client routing in a peer-to-peer (“P2P”) overlay network is provided. In one embodiment, the method of client routing in a P2P overlay network comprises requesting communication with a client by a first peer using the P2P overlay network, wherein said first peer is directed to a second peer to which said client is registered in the P2P overlay network; determining that said client is not attached to said second peer in the P2P overlay network and said client has access to another network; providing said second peer with said client's location in the P2P overlay network using said other network, wherein said client's location is associated with a third peer to which said client is attached and not registered in the P2P overlay network; forwarding said client's location from said second peer to said first peer using the P2P overlay network, and using said client's location to communicate with said client by said first peer using the P2P overlay network. | 05-30-2013 |
| 20130142192 | VOICE COMMUNICATION APPARATUS FOR INTERMITTENTLY DISCARDING PACKETS - In a voice communication device for use on an IP network, a transmitter encodes voice information and assembles RTP packets with the encoded voice information inserted in a payload of the RTP packets to transmit a stream of the assembled RTP packets. RPT packets may intermittently be discarded at a rate of discarding from a stream of the RTP packets to be transmitted. The remaining packets will be transmitted to a destination. | 06-06-2013 |
| 20130142193 | USING INDIRECT COMMUNICATION TO PROVIDE A SOLUTION TO USE INTERNATIONAL DIALING CONVENTION AND INCORPORATING PHONE NUMBERS FOR NON-PHONE DEVICES - An indirect communication system and a method of indirect communication include a mobile phone as either a calling device or receiving device. The mobile phone calls another device to set up a prospective communication, and then uses Voice over Internet Protocol (VoIP) to communicate with the other device over the Internet. The receiving device receives a generated signal notifying the receiving device of a proposed communication with the calling device. A server sets up a meeting point channel after the calling device has connected to the server. The server receives outgoing VoIP packets from the calling device and redirects the outgoing VoIP packets to the receiving device. | 06-06-2013 |
| 20130142194 | METHODS, SYSTEMS, AND COMPUTER PROGRAM PRODUCTS FOR PROVIDING INTRA-CARRIER IP-BASED CONNECTIONS USING A COMMON TELEPHONE NUMBER MAPPING ARCHITECTURE - Internet protocol (IP) based calls from a first terminal in an IP based communications system are routed to a second terminal in another communications system. In response to a call setup request at a common communications core that is common to both the IP based communications system and the other communications system, a query is transmitted to a private telephone number mapping database that contains routing information for terminals in both the IP based communications system and the other communications system requesting routing information for the second terminal. Routing information for the call setup request is received from the private telephone number mapping database for routing the call. | 06-06-2013 |
| 20130148646 | SYSTEMS AND METHODS OF PROVIDING COMMUNICATIONS SERVICES - An IP telephony system allows a calling party to provide a message that is played to the called party before the called party is connected to the calling party. The message can provide information about the call that helps the called party decide whether to answer the call. In some instances, information provided by the calling party can be used by the IP telephony system to automatically determine how to the handle the call. | 06-13-2013 |
| 20130148647 | ARCHITECTURES FOR CLEARING AND SETTLEMENT SERVICES BETWEEN INTERNET TELEPHONY CLEARINGHOUSES - A system for routing voice telephone calls over IP networks as opposed to traditional switched circuit networks. The voice communications during the telephone call are packaged as digital data and access the Internet through gateways. The system supports the linking of a source gateway in a first clearinghouse to a destination gateway in a second clearinghouse. The system further supports the selection of a destination gateway based on factors such as cost, speed of routing, and transmission quality of the voice data The components of the system are arranged so as to minimize the number of signals sent between clearinghouses in identifying the optimal destination gateway. | 06-13-2013 |
| 20130148648 | Dynamic Application Integration Associated with Hosted VoIP PBX Using Client-Side Integration Proxy - A system for dynamically integrating and synchronizing a plurality of software applications of an end user as part of a telephonic communication between the end user and a third party managed and handled by a hosted VoIP PBX includes a client side integration proxy having (i) an API for communicating and exchanging data with the software applications and (ii) memory cache dedicated to storing information about the telephonic communication handled and managed by a hosted VoIP PBX, wherein the client side integration proxy assigns a record in the memory cache for storing information about the telephonic communication and enables the software applications periodically to access and dynamically update, modify, or add to the record as the information about the telephonic communication is updated or changed by other software applications and by the end user. | 06-13-2013 |
| 20130148649 | SYSTEM AND METHODS TO ROUTE CALLS OVER A VOICE AND DATA NETWORK - Systems and methods to route a call over a voice and data network (VDN) are provided. A particular method includes receiving a call from a calling device at a telecommunications gateway (TCG). Authentication data is received via the call. A determination is made whether a user account associated with the authentication data is authorized to route calls via the TCG to a VDN. When the user account is authorized to route calls via the TCG to the VDN, call data received at the TCG via the call is converted into a format compatible with the VDN. The converted call data is sent via the VDN to a destination device. | 06-13-2013 |
| 20130148650 | COMMUNICATION NETWORK SYSTEM, CALLING TERMINAL AND VOICE CALL ESTABLISHING METHOD THEREOF - A communication network system, a calling terminal and a voice call establishing method thereof are provided. The communication network system comprises a called terminal, the calling terminal and a session initiation protocol (SIP) server. The calling terminal generates and transmits an invite message including IPv4 connection information and IPv6 connection information of the calling terminal. The SIP server is communicatively connected to the calling terminal and the called terminal. The SIP server receives the invite message from the calling terminal and forwards the invite message to the called terminal. The called terminal establishes a voice call with the calling terminal according to one of the IPv4 connection information and the IPv6 connection information of the calling terminal. | 06-13-2013 |
| 20130148651 | INTEGRATED CIRCUITS, SYSTEMS, APPARATUS, PACKETS AND PROCESSES UTILIZING PATH DIVERSITY FOR MEDIA OVER PACKET APPLICATIONS - In one form of the invention, a process of sending real-time information from a sender computer ( | 06-13-2013 |
| 20130148652 | Method, Computer-Readable Medium, and Apparatus for Providing Different Services to Different Users of an Aggregate Endpoint in an Internet Protocol Multimedia Subsystem (IMS) Network - Different services are provided to different users or groups of users of an aggregate endpoint in an internet protocol multimedia subsystem (IMS) network. The different users or groups of users are differentiated based on service profiles designated by different PUIDs for the different users or groups of users of each of the users or groups of users. Different services for transmission and/or receipt of packets for the different users or groups of users of the aggregate endpoint are provided, depending upon the differentiation between the different users or groups of users. | 06-13-2013 |
| 20130156024 | Voice-Over-IP Enabled Chat - A network-based system and method for providing anonymous voice communications using the telephone network and data communications links under the direction of a Call Broker and associated network elements. A user (the call initiator) present in a text chat room session establishes a data connection to Call Broker and, after qualifying for access (e.g., using credit card information) and providing a callback number, receives voice session information and participant access codes for each desired participant in a voice call. The initiator causes session information and participant codes to be passed to one or more selected chat participants in the current text chat room. When a selected participant uses the received session information, and enters the received participant code and a callback number, the Call Broker in cooperation with a Network Adjunct Processor (NAP) completes voice links to the initiator and the selected participant(s). | 06-20-2013 |
| 20130156025 | EXCHANGE AND USE OF GLOBALLY UNIQUE DEVICE IDENTIFIERS FOR CIRCUIT-SWITCHED AND PACKET SWITCHED INTEGRATION - According to one aspect, a system and method of exchanging GRUUs (Globally Routed User Agent URI (Uniform Resource Identifier)) between a first telephony-enabled device and a second telephony enabled device using a circuit-switched message is provided. Once exchanged, the telephony enabled devices can exchange SIP (session initiated protocol) communications routed by the GRUUs. Any one of the telephony-enabled devices can add a media component to the SIP communications. According to another aspect, a system and method of generating GRUUs is provided. According to another aspect, a system and method of handing off communications to a packet switched network from a circuit switched network is provided. | 06-20-2013 |
| 20130156026 | QUANTUM AND PROMISCUOUS USER AGENTS - A call processing system includes a call processing server. The call processing server processes calls for an internal network that employs SIP features and functions. The call processing server can receive calls from or send calls to one or more external communication endpoints that are not part of the internal network. However, the call processing server can associate a floating user agent with the communication from the external communication endpoint and lock the floating user agent to a gateway. After locking onto a gateway and initiating the call, the floating user agent can then publish call event status and receive SIP primitives similar to other SIP-enabled devices. | 06-20-2013 |
| 20130156027 | FACILITY MANAGEMENT PLATFORM FOR A HYBRID COAXIAL/TWISTED PAIR LOCAL LOOP NETWORK SERVICE ARCHITECTURE - The present invention provides a facility management platform to monitor and view the status of a plurality of individually addressable downstream devices including, but not limited to, addressable terminals, IRG's, settops, cable modems, taps, nodes, and/or hubs at a network control center. The FMP may display problems at these downstream devices, for example, power loss, and/or may automatically notify the appropriate companies and/or personnel to correct the problem. | 06-20-2013 |
| 20130163579 | Adaptive Subscriber Buffering Policy with Persistent Delay Detection for Live Audio Streams - Methods, systems, and apparatus, including computer programs encoded on computer storage media, for an adaptive subscriber buffering policy with persistent delay detection for live audio streams. In one aspect, a method includes decoding frames of multimedia data received from a first network; storing the decoded frames of multimedia data in a buffer; monitoring the buffer to determine a level of delay; and providing an output, based on the monitoring of the buffer, to cause a reduction in the level of delay during retrieval and encoding of the stored frames of multimedia data. | 06-27-2013 |
| 20130163580 | CLIENT-SERVER ARCHITECTURE FOR AUDIO-VIDEO COMMUNICATIONS - A method enabling VoIP communication sessions between a VoIP based client application and a non-VoIP standards based client application. The method includes providing a server on a digital communications network that includes runs or provides media proxy. The media proxy receives a media packet from the first communications application formatted according to a first protocol. The method includes performing packet translation on the media packet to generate a media packet that is formatted according to a second protocol that differs from the first protocol but that is used by a second communications application. The method includes transporting the translated media packet to the second communications application over the network. The packet translation includes translating the protocol while simply copying the audio-video data or payload from the original message. The communications session includes performing communications session setup between the two communications applications by signaling between these two applications. | 06-27-2013 |
| 20130163581 | SYSTEMS AND METHODS OF INTEGRATING CALL DETAIL RECORD INFORMATION FROM MULTIPLE SOURCES - A system for establishing Internet protocol based telephony communications between a calling telephony device and a called telephony device includes multiple logical elements that help to setup and carry the telephony communication. Multiple ones of the logical elements create call detail records that contain data about the telephony communication. Information contained in the call detail records is used to determine which call detail records relate to the same telephony communication. Once the call detail records relating to the same telephony communication are identified, a global identification number can be written into each of the call detail records to tie them together, or information drawn from the multiple call detail records can be used to create a single consolidated call detail record. | 06-27-2013 |
| 20130163582 | SYSTEMS AND METHODS OF MANAGING COMMUNICATION REQUESTS IN A VOIP COMMUNICATION SYSTEM - An Internet protocol (IP) telephony system includes elements that prevent more than a specified number of simultaneous calls to be carried by the IP telephony system under a single user account. Multiple copies of a call session database are maintained in different geographical areas, and the multiple copies of the call session database are frequently and rapidly synchronized. The call session database is consulted before a new call is setup to determine if the user account that is to be used for the new call already is being used for a maximum number of simultaneous calls. If so, the new call setup request is denied. | 06-27-2013 |
| 20130163583 | SYSTEMS AND METHODS FOR COMMUNICATION SETUP VIA RECONCILIATION OF INTERNET PROTOCOL ADDRESSES - Systems and methods for forwarding data packets to facilitate an IP telephony communication make use of a media relay to accomplish the forwarding actions. The media relay is configured receive setup signaling indicating a first acceptable originating IP address for received data packets. The media relay also allows the acceptable originating IP address to change once after the initial call setup has occurred. Also, if a REINVITE action occurs during a call, the media relay will allow the acceptable originating IP address to change once after the REINVITE signaling has been completed. | 06-27-2013 |
| 20130163584 | MEDIA IDENTIFICATION, CLASSIFICATION, FORWARDING, AND MANAGEMENT FOR VOICE AND VIDEO COMMUNICATIONS - A system processes media, such as voice and video, in a scalable and secure manner. The system can process voice and video for a large quantity of users. For example, the system can enable large quantities of simultaneous phone conversations over an IP network. The IP network can carry voice, video, and other data concurrently. The system identifies which packets carry voice data, which packets carry video, and which packets carry other kinds of data. The system scales both with the quantity of users and in terms of network topology. Multiple digital signal processors (DSPs) can be controlled by and connected to a switching device via an Ethernet network. One or more DSPs connected to the switching device interact with the switching device as separate IP devices, in that each such DSP may have its own separate IP address to which IP packets may be addressed. | 06-27-2013 |
| 20130163585 | TELEPHONE SYSTEM, SERVER APPARATUS, AND CONTROL METHOD USED IN THE SERVER APPARATUS - According to one embodiment, a server apparatus includes a memory, a determination module, a storage controller and a service controller. The memory stores in an authentication management table associating identification information with first authentication information and second authentication information. The determination module determines whether registration is authenticated based on the authentication management table. The storage controller stores the identification information, the attribute information, and the IP addresses in a registration management table. The service controller performs control regarding use of the communication services. | 06-27-2013 |
| 20130163586 | METHOD AND APPARATUS FOR COMPLETING A CIRCUIT SWITCHED SERVICE CALL IN AN INTERNET PROTOCOL NETWORK - A method and an apparatus for processing a session request in an Internet Protocol network are disclosed. For example, the method receives a session request, and queries an tElephone NUmbering Mapping (ENUM) server for a called party of the session request. The method determines if at least one Naming Authority Pointer (NAPTR) resource record associated with the called party is received from the ENUM server, and forwards the session request to a circuit switched network if the at least one NAPTR resource record is not received from the ENUM server. The method determines a Session Description Protocol (SDP) value of the session request if the at least one NAPTR resource record is received from the ENUM server, and processes the session request in accordance with the SDP value if the at least one NAPTR resource record is received from the ENUM server. | 06-27-2013 |
| 20130163587 | VOICE OVER INTERNET PROTOCOL REAL TIME PROTOCOL ROUTING - A method for call signaling and media flow in a network including receiving call signaling information from an originating Voice over Internet Protocol (VoIP) endpoint relaying the call signaling information to a destination VoIP endpoint, directing the originating VoIP endpoint to use a RTP media proxy and receiving a stream of media to the RTP media proxy from the originating VoIP endpoint. | 06-27-2013 |
| 20130163588 | METHOD OF TRANSMITTING DATA IN A COMMUNICATION SYSTEM - A method of receiving at a terminal a first signal transmitted via a communication network, said method comprising the steps of; receiving at the terminal the first signal comprising a plurality of data elements; analysing characteristics of the first signal; receiving from a user of the terminal a second signal to be transmitted from the terminal; analysing characteristics of the second signal to detect audio activity in the second signal; and applying a delay between receiving at the terminal and outputting from the terminal at least one of said plurality of data elements; and adjusting the delay based on the analysed characteristics of the first signal and on the detection of audio activity in the second signal. | 06-27-2013 |
| 20130163589 | Solutions for Voice Over Internet Protocol (VoIP) 911 Location Services - An E-9-1-1 voice-over-IP (VoIP) solution is provided wherein a 911 call from a mobile VoIP device is routed directly to the correct Public Safety Answer Point (PSAP) via dedicated trunks, together with correct location information and call-back number. VoIP gateways are implemented locally, at least one per LATA, and accept VoIP packetized data inbound, and convert it to standard wireline voice calls. Calls are routed to an IP address at the VoIP gateway, which then egresses the call to a voice port at a selective router. Mid-call updating of location of a moving VoIP terminal is provided to a PSAP. The location of the VoIP is validated using HTTP based protocol by pushing location information to a VoIP location server, and comparing it against a geographic location database to confirm that a contained street address is valid. | 06-27-2013 |
| 20130163590 | METHOD OF PROCESSING SIP MESSAGES - An SIP message processing method performed by a node of a telecommunications network, said network having a home gateway connected to an IMS network core via an access connection, at least a first SIP terminal and a second SIP terminal locally connected to said home gateway, said first SIP terminal being suitable for setting up a first SIP session to the IMS network core by passing via said access connection, said second SIP terminal being suitable for setting up a second SIP session to the IMS network core by passing via said access connection, said processing method including a step of obtaining a bandwidth of said access connection, said processing method being characterized in that it comprises: a step of receiving a first SIP message concerning the first SIP session; a step of determining the bandwidth in use by said first SIP session as a function of said first SIP message; a step of receiving a second SIP message concerning the second SIP session; a step of determining a bandwidth authorized for said second SIP session as a function of the second SIP message, of the bandwidth of the access connection, and of the bandwidth in use by the first SIP session; and a step of sending a third SIP message to said first terminal, said second terminal, or said IMS network core, said third message being determined as a function at least of said bandwidth authorized for said second SIP session, said third SIP message being for influencing a selection of bandwidth for the first SIP session or for the second SIP session. | 06-27-2013 |
| 20130170487 | SYSTEM AND METHOD FOR ESTABLISHING A VOICE OVER IP SESSION - A method, computer program product, and computer system for establishing a Voice over IP (VoIP) session. One or more computing devices initiate the VoIP session between a plurality of devices. A first communication channel is established as an active channel for a first computing device of the plurality of devices, where the active channel is fully enabled for use by the first computing device. A second communication channel is established as a passive channel for the first computing device while maintaining the active channel, where the passive channel is at least partially enabled for use by the first computing device. | 07-04-2013 |
| 20130170488 | INTEGRATED ACCESS DEVICE AND ASSOCIATED MODEM BOX AND SPLITTING MODULE - The present invention concerns a modem box, a splitting module, and an integrated access device. | 07-04-2013 |
| 20130170489 | Methods, Systems, and Products for Security Systems - Methods, systems, and products notify of alarms in security systems. An alarm message is received from a security system associated with a network address. The network address is associated to a notification address. A Voice-over Internet Protocol call to the notification address is initiated over a data network to alert of an alarm from a security system. | 07-04-2013 |
| 20130177011 | Communication Session Processing - Measures for use in processing communication sessions in a telecommunications network are provided. Each communication session has a signalling path spanning a plurality of devices including one or more intermediate network devices and at least two endpoint devices, the signalling path comprising a plurality of signalling segments, each segment being between two devices in the plurality of devices. A first signalling message, comprising a first identifier associated with the communication session, is received via a first signalling segment for a communication session. At least part of the first identifier is transformed using a deterministic encryption algorithm to generate a second identifier. A second signalling message, comprising the second identifier, is transmitted via a second signalling segment for the communication session to associate the second identifier with the communication session. | 07-11-2013 |
| 20130177012 | Techniques For Providing Multimedia Communication Services To A Subscriber - A technique for providing multimedia communication services to a subscriber includes receiving a communication query for the subscriber, the communication query having an associated requested communication mode. The technique also includes servicing the communication query for the subscriber using the requested communication mode when the requested communication mode corresponds to one of one or more selected communication modes. | 07-11-2013 |
| 20130177013 | METHODS AND APPARATUS TO FORM SECURE CROSS-VIRTUAL PRIVATE NETWORK COMMUNICATION SESSIONS - Example methods and apparatus to form secure cross-VPN (virtual private network) communication sessions in multiprotocol label switching (MPLS)-based networks are disclosed. An example method comprises receiving a communication session initiation request from a first device associated with a first MPLS-based VPN, determining whether the communication session initiation request is directed to a second device associated with a second MPLS-based VPN, sending a cross-VPN link setup request to a route reflector to enable a cross-VPN communication path over which the first and second devices are permitted to communicate when the communication session initiation request is directed to the second device associated with the second VPN, and facilitating a communication session between the first and second devices via the communication path enabled by the route reflector. | 07-11-2013 |
| 20130177014 | SYSTEMS AND METHODS OF MITIGATING PHANTOM CALL TRAFFIC - Systems and methods for mitigating phantom call traffic in a communication system are disclosed. A call may be received at a communications exchange. A jurisdiction of the call may be determined in real-time, and egress signaling information and a route of the call to a terminating local exchange may be determined based on the jurisdiction. At least a portion of the egress signaling information may be provided to the terminating exchange in call signaling and/or in call detail billing records so that the terminating exchange is enabled to correctly charge for call termination. | 07-11-2013 |
| 20130182700 | SYSTEMS AND METHODS FOR NETWORK MONITORING AND TESTING USING A GENERIC DATA MEDIATION PLATFORM - Embodiments are directed to systems and methods for network monitoring and testing using a generic data mediation platform between one or more probes and one or more dashboards. The generic data mediation platform performs data correlation, filtering enrichment and aggregation of events obtained from monitored networks. | 07-18-2013 |
| 20130182701 | Communications Links - A system including a plurality of remotely configurable telephone network communication terminals ( | 07-18-2013 |
| 20130182702 | METHODS, SYSTEMS, AND COMPUTER PROGRAM PRODUCTS FOR PROVIDING INTER-CARRIER IP-BASED CONNECTIONS USING A COMMON TELEPHONE NUMBER MAPPING ARCHITECTURE - A system includes a network entry point configured to receive and process internet protocol (IP) based connection requests originating from outside the communications network, and a telephone number mapping (ENUM) database configured to store a uniform resource identifier (URI) of the network entry point and to respond to a request for routing information for a subscriber terminal within the communications network received from a remote entity outside the communications network by providing a naming authority pointer resource (NAPTR) record including the URI of the network entry point to the remote entity in response to the request for routing information. | 07-18-2013 |
| 20130182703 | SYSTEM AND METHOD FOR PROVIDING AUTOMATIC DETERMINATION OF A CALL TYPE IN TELEPHONY SERVICES OVER A DATA NETWORK - A system and method for providing automatic determination of a call type in a voice over Internet Protocol data network is provided. A 10-digit telephone number for making a telephone call is entered and a look up of the telephone number is performed in a database comprising at least known long distance numbers. If no match exists, the telephone number is transmitted over the data network to a local Call Management Server. If the local call is not successful, the number is automatically transmitted with a pre-pended “I” to a Long Distance Call Management Server. Advantageously, a VoIP system is provided for automatically determining that a number is a long distance call, upon which it will automatically pre-pend a leading “1” to the number and re-transmit the resultant 11 digit number to the appropriate Long Distance Call Management Server. | 07-18-2013 |
| 20130188633 | Service Based Release of a Subscriber Registrar Server from a Signalling Path in an Internet Protocol Communication Network. - A method of and servers for establishing a signalling path ( | 07-25-2013 |
| 20130195102 | ROUTING CALLS WITHOUT TOLL FREE CHARGES - A method may include receiving a call placed to a local telephone number and identifying a toll free number associated with the local telephone number. The method may also include determining whether the toll free number is associated with a first call center for a customer using voice over Internet protocol (VoIP) trunking to route calls to the first call center, and identifying a dialed number identification service (DNIS) number associated with the toll free number. The method may further include forwarding the call and the DNIS number to the first call center via a VoIP trunking connection, in response to determining that the toll free number is associated with the first call center. | 08-01-2013 |
| 20130195103 | METHOD AND APPARATUS FOR PROCESSING VOIP DATA - A system and method for processing Voice over Internet Protocol (VoIP) data including determining whether received audio data is VoIP data, transferring, when received audio data is VoIP data, the received VoIP data to a first path, and transferring, when received audio data is not VoIP data, the received audio data to a second path. The system and method can process, when received audio data is VoIP data, the received VoIP data via a VoIP data processing path including a voice engine for VoIP, instead of an audio data processing path, irrespective of types of mobile devices and types of applications. | 08-01-2013 |
| 20130208716 | TERMINAL, METHOD AND SYSTEM FOR PERFORMING COMBINATION SERVICE USING TERMINAL CAPABILITY VERSION - A terminal, method and system for providing a CS service, a SIP-based service, or a CSI service, are provided. According to an embodiment, the terminal includes a controller to receive a terminal capability version of at least one target terminal, to compare the received terminal capability version with a previously stored terminal capability version of the at least one target terminal, and to determine whether to request for terminal capability information of the at least one target terminal based on the comparison result, wherein the terminal capability version identifies a version of capabilities of the at least one target terminal. | 08-15-2013 |
| 20130208717 | Presence Based Telephony Call Signaling - Methods and systems for presence based telephony call signaling are presented. An incoming call is received at a computer, where the computer includes a computer loudspeaker and computer display. A headset donned state or a headset doffed state is identified for a wireless headset, where the wireless headset includes a headset speaker and headset output user interface. A proximity between the wireless headset and the computer is determined. An incoming call notification is output to the headset speaker, the headset output user interface, the computer loudspeaker, or the computer display responsive to identifying the headset donned state or headset doffed state and determining the headset proximity. | 08-15-2013 |
| 20130215881 | METHOD AND SYSTEM FOR ROUTING CALLS OVER A PACKET SWITCHED COMPUTER NETWORK - The present invention describes how a trusted network routing authority, such as a VoIP inter-exchange carrier or clearinghouse can provide routing and secure access control across multiple network domains with a single routing and admission request. This technology can improve network efficiency and quality of service when an Internet Protocol (IP) communication transaction, such as a Voice over IP (VoIP), must be routed across multiple devices or administrative domains. This technology defines the technique of performing multiple route look-ups at the source of the call path to determine all possible routes across intermediate domains to the final destination. The VoIP inter-exchange carrier or clearinghouse then provides routing and access permission tokens for the entire call path to the call source. | 08-22-2013 |
| 20130215882 | METHOD AND APPARATUS FOR ENABLING REGISTRATION OF AGGREGATE END POINT DEVICES THROUGH PROVISIONING - A method and apparatus for enabling registration of an Aggregate End Point (AEP) device that is incapable of supporting a Session Initiation Protocol (SIP) based Internet Protocol Multimedia Subsystem (IMS) registration are disclosed. The method performs a static registration of the AEP device in a plurality of network elements associated with an Internet Protocol Multimedia Subsystem (IMS) network by provisioning. The method then processes an originating call request or a terminating call request associated with the AEP device using the static registration. | 08-22-2013 |
| 20130215883 | System and Method of Presenting Caller Identification Information at a Voice Over Internet Protocol Communication Device - A method includes receiving a call from communication device associated with a caller. The call is directed to a second communication device associated with a callee. The method includes determining whether a caller profile associated with the caller is available. The method includes searching at least two networks in response to determining that the caller profile is not available. The method includes creating the caller profile based on the search of the at least two networks and populating a caller identification portal with at least one of a plurality identification characteristics included in the caller profile. | 08-22-2013 |
| 20130223431 | Processing Requests - Measures for processing requests in a telecommunications network are provided. A blacklist for determining routeing attempts to be conducted during a default routeing attempt procedure is maintained. A request comprising an identifier for a given destination for the request is received and, in response to performing a destination address lookup for the identifier, a set of one or more destination addresses for routeing the request to is received. One or more destination addresses in the set are compared to the list of destinations on the blacklist. If the comparison indicates that at least one of the destination addresses in the set is not present on the blacklist, the request is processed according to a default routeing attempt procedure. If the comparison indicates that all of the destination addresses in the set are present on the blacklist, the request is processed according to an alternative routeing attempt procedure. | 08-29-2013 |
| 20130223432 | Communication System - A method of controlling a communication session is provided. The communication session is established in a telecommunications network between a first communication client of a plurality of communication clients on one or more communication devices associated with a first party and a communication device associated with a second party, the established communication session comprising at least a first communication session leg established between the first communication client and an intermediate node in the telecommunications network. The method includes detecting a loss in connectivity associated with the first communication session leg, and in response to detecting the loss in connectivity, initiating set up of a second communication session leg between an intermediate node in the telecommunications network and at least a second communication client of the plurality of communication clients on one or more communication devices associated with the first party. | 08-29-2013 |
| 20130223433 | Reducing Broadcast of SS7 Network Management Messages by SIGTRAN Relay Node - The application relates to the transport of SS7 signaling over an Internet Protocol based network and in particular to the routing of SS7 Signaling Network Management SSNM Messages such as Destination Unavailable DUNA and Signalling Congestion SCON by a SIGTRAN relay node. The SSNM messages are checked against a so-called “validation rule”, which either specifies a list of allowed affected point codes or a threshold rate of SSNM messages with the same affected point code. If the SSNM message is “valid” it is sent to a so-called “broadcast domain” which comprise nodes such as signaling endpoints or component of one or more application servers. | 08-29-2013 |
| 20130223434 | METHOD AND SYSTEM FOR COORDINATING DATA AND VOICE COMMUNICATIONS VIA CUSTOMER CONTACT CHANNEL CHANGING SYSTEM - This invention (The Customer Contact Channel Changer) enables the integration of different Customer Contact Channels such as live call centre ACD (Automatic Call Distribution) agents, ADSI (Analog Display Services Interface) enhanced IVR (Interactive Voice Response) systems and WWW (World Wide Web) servers. The world wide web servers are used to allow customers with computer equipment to access information from an organizations databases in a self service mode. Frequently these customers have questions best answered by human ACD agents. With this invention the connection between the customer with the question and the agent with the answer is done quickly and efficiently with both parties sharing screens of common information. Also control is retained by the customer to make the call happen when they want it. | 08-29-2013 |
| 20130223435 | System and Method of Routing Voice Communications Via Peering Networks - A method of routing voice communications is disclosed and includes receiving network event data at a telephone number mapping (ENUM) server of an originating network from a plurality of peering border elements associated with a plurality of peering communication networks. The network event data indicates a health status of each of the peering communication networks. The method also includes determining a network weight related to each of the peering communication networks based on the network event data, wherein each network weight indicates a proportion of calls that are to be routed from the originating network via the related peering communication network. The method further includes dynamically assigning a plurality of preference field values to session initiation protocol (SIP) addresses of the peering border elements within a plurality of naming authority pointer (NAPTR) records, based on the network weights. | 08-29-2013 |
| 20130223436 | SYSTEMS AND METHODS OF PROVIDING COMMUNICATIONS ON A SOFTWARE PLATFORM - A computer implemented method of delivering a message is disclosed. The method includes generating a call activation interface on a software platform. The call activation interface is associated with at least one discrete item from a plurality of discrete items on the software platform. The method further includes presenting a list of a plurality of contacts in response to a selection of the call activation interface, receiving at least one contact from the list, and generating a call setup request directed to the at least one contact. The call setup request includes an announcement message that includes information associated with the discrete item. | 08-29-2013 |
| 20130223437 | Connection Control with B2BUA Located Behind NAT Gateway - There is proposed a mechanism for a connection control conducted in a communication network (such as IMS) when a back to back user agent (B2BUA) and network address translation function are involved in the establishment of the connection. When a control network element, such as a P-CSCF, receives a signaling message related to the establishment of the communication connection, via a communication leg coming from a network address translation device, it is determined whether address information contained in an SDP element of the signaling message matches with preset address information allocated to a border gateway function or BGF. When no matching is determined, normal processing like an initiation of a latching procedure at an own BGF is conducted. Otherwise, if a matching address information is determined, i.e. a mirrored SDP is deemed to be present, a latching processing at an own BGF is inhibited and the received BGF's address information are used in the connection establishment procedure. This processing is conducted at all session ends coming from the B2BUA, so that a communication connection with media flow can be established through pinholes of the BGF(s). | 08-29-2013 |
| 20130230042 | SYSTEM AND METHOD FOR SESSION INITIATION PROTOCOL HEADER MODIFICATION - A method for modifying the contents of session initiation protocol (SIP) messages is presented. The method includes receiving a SIP message. The SIP message may include a set of message header fields. The method includes receiving an application policy. The application policy may specify how to modify the SIP message based on a characteristic of the SIP message. Alternatively, the application policy may be retrieved from a database such as one provided by a home subscriber server (HSS) or an application server. The method includes using the application policy to modify the SIP message resulting in a modified message, and sending the modified message. | 09-05-2013 |
| 20130230043 | SYSTEMS, PROCESSES AND INTEGRATED CIRCUITS FOR RATE AND/OR DIVERSITY ADAPTATION FOR PACKET COMMUNICATIONS - A media over packet networking appliance provides a network interface, a voice transducer, and at least one integrated circuit assembly coupling the voice transducer to the network interface. The at least one integrated circuit assembly provides media over packet transmissions and holds bits defining reconstruction of a packet stream having a primary stage and a secondary stage. The secondary stage has one or more of linear predictive coding parameters, long term prediction lags, parity check, and adaptive and fixed codebook gains. The packet stream has an instance of single packet loss, and the reconstruction includes receiving a packet sequence represented by P(n)P(n−1)′, [Lost Packet], P(n+2)P(n+1)′, and P(n+3)P(n+2)′, obtaining as information from the secondary stage one or more of the linear predictive coding parameters, long term prediction lags, parity check, and adaptive and fixed codebook gains, and performing an excitation reconstruction utilizing said packet sequence thus received. | 09-05-2013 |
| 20130235866 | Efficient and on Demand Convergence of Audio and Non-Audio Portions of a Communication Session for Phones - In one embodiment, source data for a communication session may be split into an audio portion for transmission on a phone channel and a non-audio portion for transmission on a data channel. A server and a phone may accordingly establish an audio portion of a communication session on the phone channel. In response to a trigger, the server may provide a push notification on the data channel to the phone, where the push notification is associated with an application executing on the phone that is configured to participate in the non-audio portion of the communication session on the data channel with the server. Upon obtaining the push notification on the data channel during the audio portion on the phone channel, the application may correspondingly activate on the phone to participate in the non-audio portion of the communication session during the phone's participation in the audio portion (e.g., merging the portions). | 09-12-2013 |
| 20130235867 | HYBRID TYPE TELEPHONY SYSTEM - A hybrid type telephony system capable of establishing a connection between conventional type telephone sets contained in an exchange unit and LAN type telephone sets contained in an IP network, the system comprising: a gateway circuit connected between the exchange unit and the IP network and performing voice data format conversion, and a central control unit connected to the LAN of the. IP network for establishing a communication path to the exchange unit via a control bus, controlling switching of IP packets of the IP network, managing IP address information of the LAN type telephone sets and the gateway circuit via the LAN, and controlling connection between the LAN type telephone sets and connection between the LAN type telephone sets and the gateway circuit. | 09-12-2013 |
| 20130242978 | SYSTEM PREVENTING DOUBLE DIGIT DETECTION CAUSED BY IN-BAND DUAL-TONE MULTI-FREQUENCY SIGNALING AND METHODS THEREOF - A T2P (TDM to packet) delay buffer is provided. The delay buffer can prevent double digit detections caused by in-band DTMF leak when out-of-band DTMF is used. The T2P delay buffer is initialized with an audio pattern that represents silence in a configurable amount of delay. When a DTMF digit is detected, the system can stop taking the voice payload from the T2P delay buffer and start injecting RFC4733 RTP packets into the RTP stream at a pre-configured rate. The RFC4733 DTMF RTP packets continue to be injected into the RTP stream until the DTMF digit stops. Once the end of the DTMF digit is detected, the content of the T2P delay buffer can be discarded and the T2P delay buffer is reinitialized with an audio pattern that represents silence in a configurable amount of delay. After the T2P delay buffer is reinitialized, the voice packetization can be continued. | 09-19-2013 |
| 20130242979 | METHOD OF REAL-TIME VOIP CALL - The invention discloses a method of real-time VOIP call, characterized by comprising the following steps: a) a user dials a landline phone and establishes a connection with a service terminal; b) the service terminal resolves a further dialing of the user, corresponds dialing information with an addressing address stored in the service terminal of the calling user, wherein the addressing address comprise a destination user name, a separator and a destination address; c) the service terminal of the calling user resolves information of the destination user name and sends it to a service terminal of the destination address; d) the service terminal of the destination address resolves the received information of the destination user name, and finds the information of the destination user name stored in the service terminal of the destination address, and then the service terminal of the destination address further finds a phone number of the destination user internally stored in the service terminal of the destination address; e) the service terminal of the destination address initiates a call request to the destination user; f) a calling connection is established. This method can achieve video conference and fax in the same way as the existing internet phone, and moreover, it is completely free of charge. | 09-19-2013 |
| 20130242980 | DATA STREAM CLASSIFICATION - Systems, methods, and other embodiments associated with data stream classification are described. One example method includes identifying packets associated with the data stream. The example method may also include updating a set of characterization data associated with the data stream based on information associated with a packet. The example method may also include assigning a data stream classifier to the data stream by comparing characterization data to identification data upon determining that the set of characterization data indicates that the data stream is able to be classified. The example method may also include providing a signal associated with the data stream classifier. | 09-19-2013 |
| 20130242981 | METHOD AND APPARATUS FOR ENABLING PEER-TO-PEER COMMUNICATION BETWEEN ENDPOINTS ON A PER CALL BASIS - A method and apparatus for enabling a user to signal to the network that a call to be initiated or a call that is in progress needs to occur in a peer-to-peer relationship with the terminating endpoint. The network will then remove itself from the call signaling and media path and direct the signaling and media communication to occur directly between the two endpoints. | 09-19-2013 |
| 20130242982 | METHOD AND SYSTEM FOR CUSTOMER SELECTED DIRECT DIALED VOICE-OVER-INTERNET PROTOCOL (VOIP) - A Voice-over-Internet protocol (VOIP) communications network system that enables direct-dialed (single-stage) access to the Internet Protocol network from the circuit-switched network. Specifically, the VOIP network system includes a VOIP service implemented on a communications system which, after a customer number has been registered for the service, automatically recognizes calls from the registered customer's telephone number and determines if the call can be routed as a VOIP call over the IP network. In embodiments of the present invention, the customer can register for the VOIP service by selecting both a provider and a calling plan or by only selecting a provider. The system can be implemented to handle intra-state, inter-state and international voice-band calls (for example, regular telephone calls, facsimile transmissions and modem initiated calls) using standard circuit-switched telephone lines, cable, twisted pair, digital subscriber line and wireless. | 09-19-2013 |
| 20130250937 | Method for Converging Telephone Number and IP Address - A method for converging telephone numbers and Idata addresses follows the steps of (a) accessing from a first memory location of a communication appliance one of an E.164 telephone number or an IPv6 address; (b) using the E.164 criteria of [Country Code-Identification Code-Subscriber Number 1-Subscriber Number 2-Extension-Ext2-Ext3] for a telephone number, converting that number to an IPv6 hexadecimal notation IP address in the format [::::[]:[:[]], and converting in the reverse for an IP address to a telephone number; (c) storing the telephone number or address in a separate memory location of the appliance; and (d) depending on the nature of a communication session initiated by a user, retrieving the appropriate IP address or telephone number as the destination for the communication. | 09-26-2013 |
| 20130250938 | SYSTEMS, PROCESSES AND INTEGRATED CIRCUITS FOR RATE AND/OR DIVERSITY ADAPTATION FOR PACKET COMMUNICATIONS - Packets of real-time information are sent with a source rate greater than zero kilobits per second, and a time or path or combined time/path diversity rate initially being zero kilobits per second. This results in a quality of service QoS, optionally measured at the sender or the receiver. When the QoS is on an unacceptable side of a threshold of acceptability, the sender sends diversity packets at an increased rate. Increasing the diversity rate while either reducing or maintaining the overall transmission rate is new. CELP-based multiple-description data partitioning sends the base or important information plus a subset of fixed excitation in one packet and sends the base or important information plus the complementary subset of fixed excitation in another packet. Reconstruction produces acceptable quality when only one of the two packets is received and better quality when both packets are received. Reconstruction provides for single and multiple lost packets. | 09-26-2013 |
| 20130250939 | Authentication Tokens for Use in Voice Over Internet Protocol Methods - Setup of a Voice over Internet Protocol (VoIP) call is initiated and an authentication token is received for the VoIP call that is set up, that indicates that the VoIP call is authorized. The authentication token is inserted into packets for the VoIP call. The packets, including the authentication token therein, are transmitted into an IP network. The authentication token may be placed in an IP version 6 (IPv6) flowID field. | 09-26-2013 |
| 20130250940 | SYSTEM AND METHOD FOR PROCESSING A PLURALITY OF REQUESTS FOR A PLURALITY OF MULTI-MEDIA SERVICES - A system and method for processing a plurality of requests for a plurality of multi-media services received at a Private Service Exchange (PSX) defined on the system from a plurality of IP-communication devices. The system further includes a media server (MS) coupled to the PSX and to at least one IP Service Control Point (IP-SCP), which is operative to process the plurality of requests for the plurality of multi-media services. The IP-SCP further selectively directs the requests to the media server, which operates to form a preliminary multi-media communication path with a calling communication device. The MS further operates to play a plurality of announcements to the calling communication device over the preliminary multi-media communication path, as well as to collect caller-entered data from the calling communication device over the preliminary multi-media communication path. | 09-26-2013 |
| 20130250941 | METHOD AND APPARATUS FOR PROCESSING A CALL TO AN AGGREGATE ENDPOINT DEVICE - A method and an apparatus for processing a call to an aggregate endpoint device over a network are disclosed. For example, the method receives a session request by an application server, wherein a route header for the session request comprises an aggregate endpoint identifier, and determines at least one Public User Identity (PUID) of the aggregate endpoint device in accordance with the aggregate endpoint identifier. The method obtains a Serving-Call Session Control Function Fully Qualified Domain Name (S-CSCF FDQN) of a Serving-Call Session Control Function (S-CSCF) that performs a termination processing for the aggregate endpoint device, and forwards the session request to the S-CSCF with a route header that comprises the PUID of the aggregate endpoint device. | 09-26-2013 |
| 20130250942 | Traffic Routing Across and Between Networks - A method of routing signalling plane traffic through or out of an IP Multimedia Subsystem, IMS, network. The method comprises maintaining within user profile data stored in a subscriber database of the IMS network, routing information defining signalling plane routing profiles for a user associated with the user profile data. Said routing information is transferred from the subscriber database to a Call Session Control Function, CSCF, of the IMS network at or following registration of said user to the IMS network. The method then comprises handling a Session Initiation Protocol, SIP, request for said user at the CSCF including adding said routing information into the request and forwarding the SIP request to a next hop node, and, at said next hop node or at a further downstream node, making a routing decision for the SIP request or for related signalling on the basis of the routing information contained within the SIP request. | 09-26-2013 |
| 20130250943 | INFORMATION PROCESSOR, INFORMATION PROCESSING METHOD AND NON-TRANSITORY STORAGE MEDIUM STORING INFORMATION PROCESSING PROGRAM - Provided is an information processor that receives a request for initiating a call between a phone on the calling side and a phone on the called side and performs call transmission with respect to the phone on the calling side and the phone on the called side, the information processor including: a registration unit that registers a list of a plurality of IP address conversion servers performing IP address conversion needed for a call between the phone on the calling side and the phone on the called side; a selection unit that selects, on the basis of the location of the phone on the calling side or the phone on the called side, the IP address conversion server to be used from the list that is registered in the registration unit; and a call execution unit that executes, using the selected IP address conversion server, a call while performing IP address conversion for voice packets exchanged between the phone on the calling side and the phone on the called side. | 09-26-2013 |
| 20130259026 | SYSTEM AND METHOD TO INFLUENCE SIP ROUTING BY SEQUENCED APPLICATIONS - System and method to influence routing of a call by a sequenced application from among a plurality of sequenced applications, the method including: receiving a header for the call, the header comprising at least one directive from one or more of the plurality of sequenced applications; arbitrating conflicts from among the at least one directive in the header, in order to determine a set of sequenced application headers to at least partially execute; and at least partially executing the set of sequenced application headers. | 10-03-2013 |
| 20130259027 | VOICE SERVICE DISCOVERY - In one implementation, voice service discovery may include a voice service discovery protocol (VSDP) and a VSDP device, which receives voice virtual local area network data and generates a packet including root voice service configuration data. The VSDP device may be in an enabled state, a listening state, or a disabled state. Receiving voice service configuration data may initiate the enabled state in the VSDP device, and the packet may be transmitted to a remote device. Voice service discovery may include advertising a voice service to a plurality of voice over internet protocol endpoints according to the root voice service configuration data. | 10-03-2013 |
| 20130259028 | TELEPHONY INTEGRATED COMMUNICATION SYSTEM AND METHOD - A system and method of telephony integrated network communications over a computerized network. The system includes a registration module, a verification module, an account generation module, a transaction module, a security module, and a unification module, including a processor, in communication with the account generation module and configured to unify a received telephony number with a network service thereby generating a unified phone number. The method includes managing registration of a user, verifying registration data, generating an account associated with registration data, unifying a received telephony number with a network service thereby generating a unified phone number, and managing a transaction associated with an account having a unified phone number. | 10-03-2013 |
| 20130259029 | SYSTEM THAT ENABLES A CALLING PARTY TO COMMUNICATE WITH A CALLED PARTY OVER A COMMUNICATIONS NETWORK USING AN INTERNET CONVERSION DEVICE - A proprietary internet converter (PIC) is disclosed, which allows a calling party end-user device with internet access such as a mobile telephone, to initiate voice communication with a called party VoIP (Voice Over Internet protocol) end-user device. The ID (Internet Device with a built-in PIC) converts the protocols used by the calling party end-user device so that the switch that routes calls to the called party VoIP end-user device understands instructions sent from the calling party end-user device. The switch has a call forwarding function. The calling party gives the calling party user name (e.g. ISP user name/contact or VoIP user name/contact) to the PIC over the internet. The PIC then sets call forwarding function on the switch, for that particular calling party, so that an incoming call from the calling party is automatically forwarded to the ISP user or VoIP user defined by the calling party. | 10-03-2013 |
| 20130259030 | SYSTEM AND METHOD FOR PROVIDING AN INDICATION OF CERTAINTY OF LOCATION OF ORIGIN OF AN INTERNET PROTOCOL EMERGENCY CALL - A method for indicating certainty of origin of an IP emergency call includes: (a) entering a registered address related with an assigned telephone number for a VoIP calling instrument in a provisioning system database; (b) when the call is initiated with a VoIP Service Provider (VSP) unit having an assigned internet address; in no particular order: (1) looking up the assigned internet address in a first database to ascertain a first data element relating to a VSP unit geographic region; and (2) looking up the assigned telephone number in a second database to ascertain a second data element relating to a registration geographic region containing the registered address; (c) comparing the first and second data elements; (d) if the data elements match, delivering the call; and (e) if the data elements do not match: (1) setting a flag presenting an alert to the call taker; and (2) delivering the call. | 10-03-2013 |
| 20130259031 | METHOD AND APPARATUS FOR DISTRIBUTED COMPOSITIONAL CONTROL OF END-TO-END MEDIA IN IP NETWORKS - A method and an apparatus for performing a distributed control of end-to-end media on packet networks such as Voice over Internet Protocol and Service over Internet Protocol networks are disclosed. The method first receives a request from a first media endpoint device for opening at least one media channel to a second media endpoint device wherein said request contains a descriptor of said first media endpoint device. The method then updates one or more slot states and link states in response to said request and records the current state of each slot for supporting said media channel. The method also records the most recently received descriptor of said media endpoint device as a most recent descriptor for said slot supporting said media channel. The method executes one or more link objects in response to said request for controlling said at least one media channel. | 10-03-2013 |
| 20130259032 | Blending Telephony Services in an Internet Protocol Multimedia Subsystem - An Internet protocol Multimedia Subsystem (IMS) gateway application server includes an originating application server module adapted to invoke call control services in response to requests initiated by a voice over Internet Protocol (IP) (VoIP) client associated with a communication device such as an IP telephone. Disclosed gateway application servers include a proxy server module adapted to notify the communication client of session control messages intended for the communication device. | 10-03-2013 |
| 20130266002 | COMBOPHONE WITH QoS ON CABLE ACCESS - A method of providing QoS to a session from a client to a first network includes providing data packets from the client to be conveyed in a session from the client to a first network, inserting each of the data packets into an encapsulating packet, and transmitting the encapsulating packets through the second network to the first network, forming a tunnel through the second network. The method includes receiving the encapsulating packets at a terminating device in the first network. The terminating device removes the encapsulating headers to recover the data packets. The method includes determining an association between the packet headers of the data packets and the encapsulating headers, identifying data packets requiring QoS, and using the association to identify encapsulating packets corresponding data packets requiring Quality of Service. The method includes applying QoS to the encapsulating packets, corresponding to the session of data packets requiring the QoS, being conveyed through the tunnel. | 10-10-2013 |
| 20130266003 | METHOD AND APPARATUS IN GATEWAY FOR TRANSFERRING SWITCHED DIALED DIGITS TO SWITCHED PARTY GATEWAY - In one embodiment, the method in a gateway for transferring switched dialed digits to a switched party gateway includes A. judging whether switched dialed digits from a user match against a table of overlap dialed numbers; B. if the switched dialed digits match against the table of overlap dialed numbers, then transmitting the switched dialed digits in match to the switched party gateway; and C. if a subsequent switched dialed digit from the user is received, then transmitting the subsequent switched dialed digit to the switched party gateway. | 10-10-2013 |
| 20130272295 | Legacy to Cloud Telephone System - A virtual private branch exchange (VPBX) has an internet-connected server with at least one digital processor coupled to a data repository, and software executing on the one or more processors from a non-transitory medium. The VPBX maintains configuration data for individual legacy PBX systems, including telephony services enabled by the software to be provided to each legacy PBX system, receives packet-data protocol telephone calls uniquely digitally associated with the individual legacy systems, forwards those calls via a gateway to PSTN destinations, or via the Internet to IP destinations, and provides additional telephony services associated at the VPBX with each legacy system. | 10-17-2013 |
| 20130272296 | SELF-FORMING VOIP NETWORK - A self-forming VoIP connection capability is described that may be superimposed over wired networks, wireless networks, or combinations thereof. As described herein, a local network cluster forms while isolated from a conventional SIP server, or alternately may exist as a cluster of network nodes and clients that later becomes isolated from a conventional SIP server by a break in the network. Either way, each network node thus enabled with distributed SIP registry functionality according to this invention independently constructs a local SIP registry and SIP server capability within that node. Subsequently, while isolated from a conventional SIP server, VoIP conversations among client devices connected to nodes within an isolated cluster will continue, and nodes and clients may join or leave an isolated cluster with conversations able to be initiated or continued while a node has network connectivity to the cluster. | 10-17-2013 |
| 20130272297 | METHODS AND SYSTEMS FOR MANAGING A CALL SESSION - Methods and systems are provided for managing call sessions on public and private networks. The methods and systems operate to receive and send voice over internet protocol (VoIP) communications using a network, such as an IP network. The methods and systems also operate to receive and send emergency information over IP and other data networks. Based on certain criteria, the methods and systems determine whether to transfer a VoIP communication and/or emergency information to another entity associated with the IP network. | 10-17-2013 |
| 20130272298 | VOICE OVER NETWORK (VoN)/VOICE OVER INTERNET PROTOCOL (VoIP) ARCHITECT HAVING HOTLINE AND OPTIONAL TIE LINE - Voice service over a next generation network is provided using Advanced Intelligent Network solutions. According to an exemplary embodiment, a Voice over Network system includes a communications device having a directory communication address in communication with a telecommunications network, means for decoding the directory communications address to identify a voice over internet protocol service feature of the communications address, and means for establishing an internet protocol telephony communications connection of the communications device with a called party's communications address via a VoN hotline. According to further exemplary embodiments, the hotline may include a media gateway, an application server, a feature server, and means for communicating among the media gateway, the application server, and the feature server. | 10-17-2013 |
| 20130279494 | MEDIA PLANE OPTIMIZATION FOR VOICE OVER LTE - Methods and apparatus are disclosed for defining an optimized media path. In one exemplary method, a proxy session controller registers, for a plurality of realms, a plurality of Border Gateway Functions (BGFs) residing on a plurality of network nodes, with the registration including registering a preferred BGF colocated with a data network gateway on a single network node for a selected set of the realms. A request message is received from a user terminal, and if a realm associated with the request message is included in the selected set of realms, the controller assigns the preferred BGF to the user terminal to provide an optimized media path that includes the user terminal, the preferred BGF, and the data network gateway. | 10-24-2013 |
| 20130279495 | SYSTEMS AND METHODS OF PROVIDING COMMUNICATIONS SERVICES - An IP telephony system allows users of the IP telephony system to register extension telephony devices with the IP telephony system. An extension telephony device is one that is provided with service by a separate telephony service provider. Once an extension telephony device is registered, a user can obtain communications services from the IP telephony system using the extension telephony device. An extension telephony device may be tied to a user's main telephony services account with the IP telephony system such that when the user obtains communications services from the IP telephony system using an extension telephony device, the user will be billed for those communications services through the user's main account. | 10-24-2013 |
| 20130279496 | COMMUNICATION APPARATUS AND A COMMUNICATION PROTOCOL SWITCHING METHOD - According to one embodiment, an communication apparatus includes a processor and a controller. The processor receives a communication signal from the communication ports or the communication network, and executes communication processing based on a first program corresponding to a communication protocol of a communication port being connected. The controller switches from the first program set in the processor to a second program corresponding to a communication protocol different from that of the first program, based on a predetermined condition. | 10-24-2013 |
| 20130279497 | METHOD FOR CORRELATING MESSAGES ACROSS MULTIPLE PROTOCOLS IN A TELECOMMUNICATION NETWORK - A method for correlating a plurality of messages conforming to different protocols and associated with a common communication call in a communication network. The method comprising extracting an information from a message, extracting a protocol specific identifier, forming a plurality of protocol specific queues, forming a correlation ID using the information extracted from the message and the protocol specific identifier of the message, such that the messages associated with the same communication call have same correlation ID and forming a call specific queue comprising of messages extracted from the plurality of protocol specific queues associated with the same common communication call and having same correlation ID. The method further includes processing of messages from the call specific queue with FIFO criterion. | 10-24-2013 |
| 20130279498 | METHOD FOR DIALING FROM INTERNET EXTENSION TO CONVENTIONAL EXTENSION - A system for dialing from Internet extension of other SIP proxy server to conventional extension is disclosed. A VoIP gateway or an IP auto attendant is used for dialing from Internet extension to conventional extension. The phone number of the Private Branch Exchange and the voice guidance are not needed. The calling number of SIP message is interpreted directly and converted into DTMF (Dual-tone multi-frequency) messages for dialing into a conventional extension. | 10-24-2013 |
| 20130279499 | COMMUNICATIONS NETWORK ROUTER AND SYSTEM - A communications network router is described for secure routing of calls between domains of different classification levels. The network router includes a first communications port arranged to receive a first communications channel of a first domain, a second communications port arranged to receive a second communications channel of a second domain, a switch apparatus arranged selectively to connect the first port to the second port to establish a communications link between the first and second domains. The first domain has a different level of classification to the second domain. | 10-24-2013 |
| 20130287016 | METHOD AND USER TERMINAL FOR SUPPORTING PROVISION OF CAPABILITIES - A method in a first user terminal ( | 10-31-2013 |
| 20130294439 | SYSTEM AND METHOD FOR BYPASSING DATA FROM EGRESS FACILITIES - An open architecture platform bypasses data from the facilities of a telecommunications carrier, e.g. an incumbent local exchange carrier, by distinguishing between voice and data traffic, and handling voice and data traffic separately. An SS7 gateway receives and transmits SS7 signaling messages with the platform. When signaling for a call arrives, the SS7 gateway informs a control server on the platform. The control server manages the platform resources, including the SS7 gateway, tandem network access servers (NASs) and modem NASs. A tandem NAS receives the call over bearer channels. The control server determines whether the incoming call is voice traffic or data traffic, by the dialed number, and instructs the tandem NAS how to handle the call. Voiced traffic is transmitted to a switch for transmission from the platform. Data traffic is terminated at a modem NAS, where it is converted into a form suitable for a data network, such as a private data network or an Internet services provider (ISP). The converted data is sent by routers to the data network. The data network need not convert the data, as the function has already been provided by the platform. In lieu of a conversion, the modems can create a tunnel (a virtual private network) between a remote server and the data network. | 11-07-2013 |
| 20130294440 | SYSTEM AND METHOD FOR TRACKING COMMUNICATIONS RESOURCES - A system for tracking communications resources. The system includes a first database configured to store data indicative of usage of numeric identifiers assigned to a service provider. The system also includes a computing device in communication with multiple databases via a network. The multiple databases including numeric identifier usage information associated with multiple service providers. The computing device executes software to query the first database and the multiple databases to retrieve information associated with the service provider to determine current utilization of numeric identifiers by the service provider and to determine months to exhaust of numeric identifiers, and automatically generate a report utilizing the current utilization and months to exhaust in response to determining the current utilization and months to exhaust. | 11-07-2013 |
| 20130294441 | SYSTEMS, PROCESSES AND INTEGRATED CIRCUITS FOR IMPROVED PACKET SCHEDULING OF MEDIA OVER PACKET - A method of processing first and second record packets of real-time information includes computing for each packet a deadline interval and ordering processing of the packets according to the respective deadline intervals. A single-chip integrated circuit has a processor circuit and embedded electronic instructions forming an egress packet control establishing an egress scheduling list structure and operations in the processor circuit that extract a packet deadline intervals, place packets in the egress scheduling list according to deadline intervals; and embed a decoder that decodes the packets according to a priority depending to their deadline intervals. | 11-07-2013 |
| 20130294442 | VOICE OVER INTERNET PROTOCOL MULTI-ROUTING WITH PACKET INTERLEAVING - A method and system for processing data packets is described within. The method executed by the system includes the steps of receiving a first data packet, determining if the first data packet is a first expected data packet, determining if the first data packet is a next expected date packet, storing the first data patent if the first data packet is the next expected data packet and waiting a period of time for a second data packet. | 11-07-2013 |
| 20130294443 | NETWORKING BETWEEN VOIP -AND PSTN- CALLS - Programmatically reversing numerical line identity presented at a communications services gateway into named IP Telephony users with “prior association”, delivers dynamic “reverse address resolution” switching connections from ground to cloud, permitting any conventional telephone to dial and connect to any associated IP Telephony endpoint in the world, without changes to the conventional telephone. Reversing line identity into associated named users bridges both the addressability and economic divide between mass conventional “paying” (mobile and fixed) and “free” IP Telephony networks. A system for supporting communications between a user on an IP-addressed-communications-device and a telephony subscriber device, the telephony subscriber device having a corresponding telephone number, includes: one or more service nodes configured to: receive from the user the telephone number of the telephony subscriber device and create an association from the telephone number to the user, wherein the association allows the telephony subscriber device to connect to the user. | 11-07-2013 |
| 20130301637 | HOME GATEWAY WITH STANDBY STATE SUPPORT - A home gateway with standby state support is described, which is adapted to connect to a voice service. The gateway comprises a connector for a telephone, a detection circuit for determining a user request to make a telephone call, a processor configured to initiate connection to a voice service, a messaging service module for signaling a message to the user that voice service is being connected, and a transfer service module for, upon connection to the voice service, transferring the telephone call to the voice service. | 11-14-2013 |
| 20130301638 | Common Routing - In an embodiment, call routing from a customer to a destination is provided by intercepting a call setup message sent from a customer switch intended to signal a switch to perform a call routing function. In response to the call setup message being intercepted, a routing engine is queried with the destination of the call for a specific route over which to carry the call to the destination. The call setup message is modified to include the specific route. The switch is directed with the modified call setup message to use the specific route to carry the call from the customer to the destination. | 11-14-2013 |
| 20130308627 | APPARATUS FOR REDUCING NETWORK TRAFFIC IN A COMMUNICATION SYSTEM - A system that incorporates teachings of the present disclosure may include, for example, an application server that includes a memory storing computer instructions, and a processor coupled to the memory. The processor responsive to executing the computer instructions can perform operations including transmitting to Home Subscriber Server (HSS) a User Data Request (UDR) command without initiating a third party registration process, where the UDR command includes a request for dynamic device information associated with a communication device, and receiving from the HSS a User Data Answer (UDA) command comprising the dynamic device information associated with the communication device. Other embodiments are disclosed. | 11-21-2013 |
| 20130308628 | NAT TRAVERSAL FOR VOIP - A method of communication between users' electronic communication devices connected to a network via NAT devices, comprising: sending a call request to a signaling server, locating a relay server IP address, sending the call request and the relay server IP address to the receiving device, sending the relay server IP address to the calling device, starting communication via the relay server and following said communication start: identifying and reporting by the devices' public and private addresses, establishing connectivity between the devices and continuing the communication in a peer-to-peer mode. | 11-21-2013 |
| 20130308629 | TELECOM INFORMATION FOR WEB SERVICES THAT ARE PROVIDED BY A TELECOM NETWORK - Systems and methods for identifying telecom information for web services that will be provided by a telecom network. In one embodiment, a telecom data element is implemented in a telecom network. When a web service request is initiated by an application and sent to a web service gateway, the web service gateway sends a query to the telecom data element. The telecom data element receives the query, and identifies telecom information that relates to the web service that will be provided by the telecom network. For example, the telecom information may comprise policy rules, context information (e.g., location, presence, etc.), and subscriber profiles. After collecting the telecom information, the telecom data element transmits a response to the web service gateway that includes the telecom information. The web service gateway may then process the telecom information to determine how to handle the service request for the web service. | 11-21-2013 |
| 20130308630 | Dynamic Application Integration Associated with Telephonic Communications Through Hosted VoIP PBX Using Client-Side Integration Proxy - A system for collecting information associated with a telephonic communication made through a VoIP system by dynamically integrating a plurality of end user software applications including a client side integration proxy in electronic communication with a hosted VoIP PBX. Software executing on the client side integration proxy retrieves data related to a requested previous telephonic communication from a data store, assigns a portion of a memory cache for storing the retrieved data about the previous telephonic communication, enables one or more of the plurality of end user software applications to access the data about the previous telephonic communication, enables one or more of the plurality of end user software applications to update, modify, or add to the data about the previous telephonic communication, and retrieves and presents the supplemented data about the previous telephonic communication to the end user. | 11-21-2013 |
| 20130308631 | SYSTEM AND METHOD TO INITIATE A PRESENCE DRIVEN PEER TO PEER COMMUNICATIONS SESSION ON NON-IMS AND IMS NETWORKS - An architecture and method is provided for call routing using both IMS and non-IMS frameworks. The method includes receiving presence information of a third party from a non-IP Multimedia Subsystem (IMS) network device. The method further includes routing the third party to at least one callee designated device based on configurable preferences provided by the callee and correlated to presence information using an IMS compliant component. The method additionally includes providing a charging record for the routing on an IMS complaint charging platform. | 11-21-2013 |
| 20130308632 | ROUTING TERMINATING CALLS - A subscriber server routes a terminating call in a network that includes a circuit switched (CS) network, a packet switched (PS) network, and an IP Multimedia Subsystem (IMS). The server receives a request for routing information in relation to the terminating call from a CS node in the CS network. A determination is made at server based on whether a UE associated with the terminating call is registered in the IMS and the UE has access to the PS network. The server instructs the CS node to route the terminating call in the CS network when the UE is not registered in the IMS or when the UE does not have access to the PS network. The server instructs the CS node to send the terminating call to the IMS for handling when the UE is registered in the IMS and when the UE has access to the PS network. | 11-21-2013 |
| 20130308633 | Determining A Location Address For Shared Data - A method and apparatus for determining a location address for shared packet switched data. The data is to be accessible by a user of a first device and a user of a second device. A symmetric function is applied to a known identifier associated with the first device and a known identifier associated with the second device. The result of the function can then be used as an address for data to be accessible by users of both the first device and the second device, the data being stored on a remote server at a location defined by the address. The method ensures that the same address is calculated regardless of which identifier is used first, and that both devices can calculate the address without requiring any further signalling or capability/discovery mechanism. | 11-21-2013 |
| 20130315227 | TELEPHONY - In one embodiment of an improvement to telephony, a solution to the problem of communicating to “the many” is made by enabling telecommunications service providers to: accept digital dialog as well as conventional dialog, enable augmented phone service to be added to conventional phone services, handle non-calls in addition to calls, and turn content into content-of-interest. | 11-28-2013 |
| 20130315228 | SYSTEM AND METHOD OF COMMUNICATION IN AN IP MULTIMEDIA SUBSYSTEM NETWORK - A system and method of communication in an IMS network is disclosed. An apparatus that incorporates teachings of the present disclosure may include, for example, a call processing server having a controller element that receives from a terminal device a calling ID for establishing communications with a called party, submits to a telephone number mapping (ENUM) server a query corresponding to the calling ID, receives from the ENUM server a plurality of communication identifiers retrieved from a Naming Authority Pointer record according to a grade of service (GoS) of the called party, and selects according to the GoS of the called party a communication identifier from the plurality of communication identifiers to establish communications with the called party. Additional embodiments are disclosed. | 11-28-2013 |
| 20130315229 | CACHING OF ANNOUNCEMENTS AT THE EDGE OF A PACKET SWITCHED TELECOMMUNICATION NETWORK - A method and an access server operating in a packet switched telecommunication network, such as a Voice over Internet Protocol (VoIP) network, for distributing an announcement to user equipment. The method comprises receiving an announcement from a media source, receiving a caching indication from the media source to allow caching of the announcement and sending the received announcement to the user equipment. The access server can cache the announcement in a caching unit associated with the access server on receipt of the caching indication and the announcement. Furthermore a method and a media source cooperating with the access server for generating an announcement and a caching indication to allow the access server to cache the announcement. | 11-28-2013 |
| 20130315230 | METHOD AND APPARATUS FOR PROVIDING A USER WITH CHARGING-RELATED VOICE SERVICE - The present invention provides a ubiquitous solution for providing charge-related voice service such as voice announcement and interactive voice response in an IP multimedia subsystem. In particularly, an online charging system ( | 11-28-2013 |
| 20130315231 | METHOD FOR INTEGRATING FUNCTIONS OF A TELECOMMUNICATIONS NETWORK IN A DATA NETWORK - In a method for integrating functions of a telecommunications network (TN) in a data network (DN), a switching System (PBX) of the telecommunications network is connected to an IM Server (XS) of the data network via a device (CCGW), with the aid of which the IM Server is rendered capable of providing Computer telephony integration Services of the switching System (PBX) to a Communications subscriber of the data network. | 11-28-2013 |
| 20130322427 | CORE NETWORK ARCHITECTURE - A network includes at least two core local area network (LAN) fabrics, each including a first core switch cluster deployed at a first sub-core and a second core switch cluster deployed at a second sub-core different from the first sub-core. The network also includes a multi-port link aggregation group to link the first core switch cluster and the second core switch cluster. | 12-05-2013 |
| 20130322428 | APPARATUS AND METHODS FOR ORIGINATION OF VOICE AND MESSAGING COMMUNICATION IN A NETWORK - A method that incorporates teachings of the subject disclosure may include, for example, receiving a query from a call session server for a first pointer associated with a telephone number of a terminating device of a requested communication session, transmitting to the call session server the first pointer including a session initiation protocol uniform resource identifier associated with the terminating device to initiate an internet protocol communication session, receiving a notification from the call session server responsive to the call session server failing to initiate the internet protocol communication session, and transmitting to the call session server a second pointer including a telephone protocol uniform resource identifier for originating a circuit-switched communication session responsive to receiving the notification. Other embodiments are disclosed. | 12-05-2013 |
| 20130322429 | METHOD AND APPARATUS FOR MANIPULATING AVPS IN A DIAMETER ROUTING AGENT - Various exemplary embodiments relate to a method and related network node including one or more of the following: receiving a Diameter message at the DRA from an origin device; establishing a message context object in response to receiving the Diameter message, wherein the message context object includes a first collection of child objects; evaluating at least one rule, including: modifying, based on a first instruction, a first index value associated with the first collection of child objects, accessing, based on a second instruction and from the first collection of child objects, a child object corresponding to the first index value; and transmitting a message based on the evaluation of the at least one rule. | 12-05-2013 |
| 20130322430 | DIAMETER ROUTING AGENT LOAD BALANCING - Various exemplary embodiments relate to a method and related network node including one or more of the following: receiving a Diameter message at the DRA from an origin device; encountering an instruction to perform load balancing; locating an applicable load balancing pool of a plurality of load balancing pools for the Diameter message; identifying a pool host from the applicable load balancing pool to receive the Diameter message; modifying a destination address of the Diameter message to include an address of the identified pool host; and transmitting the Diameter message based on the modified destination address. | 12-05-2013 |
| 20130322431 | Controller For The Intelligent Interconnection Of Two Communication Networks, And Method Of Use For Same - A caller ID based call routing feature. A processing system in the public switched telephone network (PSTN) receives first identifying information for identifying the source of a telephone call and associates additional information stored in a memory with the first identifying information. The additional information may then be transmitted to the subscriber via the Internet for display. Another feature is a branch calling feature where the subscriber may program a processing system within the PSTN to forward an incoming call to two or more end units (e.g., telephones) simultaneously. If the call at an end unit is answered, answer supervision signaling is transmitted back to the processing system which then terminates all other calls. The processing system then connects the calling party to the subscriber. The branch calling may be made for any combination of local, long distance, and cellular telephone numbers. | 12-05-2013 |
| 20130322432 | SYSTEM AND METHOD FOR ESTABLISHING A CALL BEING RECEIVED BY A TRUNK ON A PACKET NETWORK - A method for establishing a call over a packet network may include receiving a call request via an originating trunk on a packet network from an originating call device. Status of an originating trunk and terminating segment may be determined. If the status of the originating trunk and terminating segment are within a first range, the call between the originating and terminating call device over the determined transmission path at a first data rate may be established. Otherwise, if the status of the originating trunk or terminating segment is within a second range, a determination as to whether the originating trunk and terminating segment can operate at a lower data rate may be made, and, if so, the call may be established over the transmission path between the originating call device and terminating call device at the lower data rate. | 12-05-2013 |
| 20130322433 | METHOD FOR SETTING UP A COMMUNICATION LINK - In a method for setting up a communication link between a first telephony terminal (PA) and a second telephony terminal (PB) in a communication network which transports data packets, in particular on the Internet, with the aid of at least one signalling Server (SA, SB), in particular with the aid of an SIP Server, the first telephony terminal informs a first signalling Server that a call is intended to be made to the second telephony terminal. The first signalling Server which has been informed or a second signalling Server which has been informed by this first signalling Server recognizes that the call is intended to be made with a particular quality of Service and sets up a communication link between the first telephony terminal and the second telephony terminal, which link corresponds to this quality of Service. | 12-05-2013 |
| 20130329722 | PRODUCING ROUTING MESSAGES FOR VOICE OVER IP COMMUNICATIONS - A process and apparatus to facilitate communication between callers and callees in a system comprising a plurality of nodes with which callers and callees are associated is disclosed. In response to initiation of a call by a calling subscriber, a caller identifier and a callee identifier are received. Call classification criteria associated with the caller identifier are used to classify the call as a public network call or a private network call. A routing message identifying an address, on the private network, associated with the callee is produced when the call is classified as a private network call and a routing message identifying a gateway to the public network is produced when the call is classified as a public network call. | 12-12-2013 |
| 20130329723 | METHOD AND APPARATUS FOR PROVIDING CALL ROUTING IN A NETWORK - A method and an apparatus for providing call routing in a network are disclosed. For example, the method receives a signaling message for a call, and determines if the signaling message contains information for determining if routing of the call requires an ENUM (tElephone Numbering Mapping) query. The method then processes the call by bypassing the ENUM query if the signaling message contains the information. | 12-12-2013 |
| 20130329724 | TELEPHONE SYSTEM, SERVER APPARATUS AND CONTROL METHOD - According to one embodiment, a telephone system includes a plurality of communication terminals and a server apparatus. The server apparatus includes a memory, a determination module, and a controller. The memory stores in a management table terminal IDs in association with terminal type identifying information representing whether the communication terminals are SIP terminals or non-SIP terminals. The determination module determines whether or not a third communication terminal is an SIP terminal based on the management table. The controller causes a display of the third communication terminal to show the terminal ID of the second communication terminal as caller's information. | 12-12-2013 |
| 20130336308 | Call Invites - A network node, computer program product and method establishing a call between a caller and a callee over a network. Multiple versions of a call invite are sent for establishing the call between a caller client of the caller and one or more callee clients implemented at one or more callee terminals of the callee. The multiple versions of the call invite are sent over a plurality of different delivery mechanisms. One of the delivery mechanisms comprises a push notification on a push channel. | 12-19-2013 |
| 20130336309 | Notification of Communication Events - An apparatus, computer program product and method, the apparatus comprising: processing apparatus configured to generate a push notification relating to a communication from an originating endpoint intended for a destination endpoint, the communication to be conducted over a packet-based network; and transceiver apparatus arranged to send the push notification to the destination endpoint. The processing apparatus is configured to generate the push notification with a payload comprising an indication of a language to be used by the destination endpoint to output a user notification notifying a destination user regarding the communication. | 12-19-2013 |
| 20130336310 | Notification of Communication Events - A network element of a communication provider comprises transceiver apparatus arranged to receive a request message from an originating endpoint via a packet-based communication network; and processing apparatus configured to generate, in response to the request message from the originating endpoint, a push notification relating to a communication from the originating endpoint intended for a destination endpoint, the communication to be conducted over the packet-based network. The transceiver apparatus is arranged to send the push notification to the destination endpoint over the packet-based network. The processing apparatus is further configured to generate the push notification with a payload comprising an indication of an image representing an originating user, to be output by the destination endpoint in a user notification notifying a destination user regarding the communication. At least the indication of the image is determined and inserted into the payload of the push notification at the network element. | 12-19-2013 |
| 20130336311 | Notification of Communication Events - A network element of a communication provider arranged to receive a call invite from an originating end-user terminal inviting a destination end-user terminal to a proposed session for conducting a voice or video call over a packet-based network, in response to generate a push notification, and to send the push notification to the destination end-user terminal. The processing apparatus is configured to generate the push notification with a payload comprising call signalling information enabling a response regarding the proposed session to be formulated by the destination end-user terminal and returned to the originating end-user terminal, the call signalling information comprising at least (i) an indication that a session between end-user terminals is sought, and (ii) an identifier for responding to the originating end-user terminal. | 12-19-2013 |
| 20130336312 | COMMUNICATION SYSTEM, DATACENTER APPARATUS, AND CONTROL METHOD USED IN DATACENTER APPARATUS - According to one embodiment, a communication system includes at least one user apparatus and a datacenter apparatus. The datacenter apparatus includes a processor, a memory and a controller. The processor includes a plurality of containers required to execute a plurality of communication functions associated with the exchange processing between the communication terminals or between the communication terminal and the communication line. The memory stores a user ID used to identify the user apparatus. The controller provides a communication service using at least one of the plurality of containers when a use request of the communication function is received from the user apparatus. | 12-19-2013 |
| 20130336313 | METHOD AND APPARATUS FOR CONFIGURING IP MULTIMEDIA SUBSYSTEM NETWORK ELEMENTS - A system that incorporates teachings of the present disclosure may include, for example, a method for receiving initial filter criteria from a home subscriber server, transmitting information obtained from the initial filter criteria to a domain name system, receiving a multicast IP address from the domain name system, and transmitting a message to a plurality of IP multimedia subsystem network elements according to the multicast IP address. The message can be used for configuring the plurality of IP multimedia subsystem network elements. Other embodiments are disclosed. | 12-19-2013 |
| 20130336314 | METHOD FOR COMPLETING INTERNET TELEPHONY CALLS - A call between a calling party and a called party, one or both of whom may be subscribers to Internet Telephony (IT) services, commences upon the receipt of a call dialed by the calling party to the Plain Old Telephony Service (POTS) number associated with the calling party. A first hub receives the call and routes it to the called party if that party is not an IT services subscriber that is currently on line. If the called party is an IT services subscriber that is on-line, the call is received at an Internet Services Provider serving the called party. The ISP converts the call to an IT format if the call is not already in that format and thereafter delivers the call to the called party. | 12-19-2013 |
| 20130343373 | VOICE-OVER-INTERNET PROTOCOL (VOIP) APPLICATION PLATFORM - A computer-implemented system is provided that facilitates implementation of a voice over IP (VOIP) application. The system includes a host system and a user interface (UI) host process residing on the host system. The system also includes an agent host process residing on the host system which is being configured to process a VOIP call received by one or more VOIP applications executable on the host system. The agent host process running as a foreground or background process for the duration of the VOIP call to (i) communicate with a VOIP server associated with the VOIP application, (ii) capture content from at least one input device associated with the host system and (iii) render content on an output device associated with the host system when an instance of the UI host process operates in the foreground. | 12-26-2013 |
| 20130343374 | NETWORK MARKETING AND ANALYSIS TOOL - Implementations described and claimed herein provide systems and methods for differentiating a portion of network traffic having an Internet Protocol-based attribute with reasonable certainty. In one implementation, data corresponding to a delivery of network traffic across a communications network is received. A first filter is applied to obtain a first subset of the data based on one or more characteristics of originating access traffic. The first subset includes network traffic known to originate with the Internet Protocol-based attribute. A second filter is applied to data excluded from the first subset based on one or more characteristics of terminating access traffic to obtain a second subset of the data. The second subset includes network traffic known to terminate with the Internet Protocol-based attribute. The first subset is correlated with the second subset to identify the portion of network traffic having the Internet Protocol-based attribute. | 12-26-2013 |
| 20130343375 | SYSTEM AND METHOD FOR TRANSMITTING AND RECEIVING SESSION INITIATION PROTOCOL MESSAGES - Provided are a system and method for transmitting and receiving Session Initiation Protocol (SIP) messages. The system includes user equipment (UE) including an SIP client configured to generate an SIP message, and an external transport configured to receive the SIP message from the SIP client, generate a packet by combining a tunneling header with the received SIP message, and transmit the generated packet to an SIP broker server, and the SIP broker server configured to receive the packet from the external transport in the UE, remove the tunneling header from the packet, and transmit the SIP message from which the tunneling header has been removed to a communication counterpart. | 12-26-2013 |
| 20130343376 | DIRECTORY NUMBER MOBILITY UTILIZING DYNAMIC NETWORK DISTRIBUTED DIAL-PEER UPDATES - Methods, logic, apparatus, and systems are provided to support cross cluster directory number (DN) extension mobility (EM) using dynamic network distributed dial-peer updates in a communication networks, which includes a plurality of clusters or systems and each of the plurality of clusters including a call control agent (CCA). Identification data corresponding to an identity of an associated user is received into a first cluster of a multiple cluster telecommunication network. A directory number and associated first telecommunication device corresponding to the user are registered with a first call control agent of the first cluster in accordance with received identification data. Registration data corresponding to the registered directory number is communicated to at least a second cluster of the telecommunications network. An incoming connection request associated with the registered directory number is routed directly to the first CCA without redirection to any other CCAs within the multiple cluster telecommunication network. | 12-26-2013 |
| 20140003420 | Service Controlling in a Service Provisioning System | 01-02-2014 |
| 20140016634 | SYSTEMS AND METHODS FOR LOCATION MANAGEMENT AND EMERGENCY SUPPORT FOR A VOICE OVER INTERNET PROTOCOL DEVICE - An example method stores a nomadic service designator and an operating mode designator in association with a public user identifier. The nomadic service designator indicates whether an IP device is allowed to access VoIP services from different network locations. The public user identifier facilitates establishing a call with the IP device. The operating mode designator indicates when the IP device is in a suspended operating mode and an unrestricted mode. The suspended operating mode restricts the IP device to a subset of communication services associated with a service subscription of the IP device, and to a 911 service. The unrestricted operating mode is based on a registered geographic location associated with the IP device being a current geographic location of the IP device, and is based on a service provider being able to provide an E911 service including a location-identification service at the current geographic location of the IP device. | 01-16-2014 |
| 20140023064 | SYSTEMS, METHODS, AND MEDIA FOR CONNECTING EMERGENCY COMMUNICATIONS - Systems, methods, and media for connecting emergency communications are provided. For example, the methods can include: receiving an emergency communication at a particular public safety answering point from a caller directed to the particular public safety answering point by a location-to-service system; creating a conference on a conference system in response to the particular public safety answering point accepting the emergency communication; selecting a particular call-taker of a plurality of call-takers; sending the particular call-taker an invitation to accept the emergency communication; in response to receiving an indication that the particular call-taker has accepted the emergency communication, connecting the selected call-taker to the conference; connecting the caller to the conference; determining the location of the caller; querying the location-to-service system to identify at least one particular emergency responder of a plurality of emergency responders; and connecting at least one of the at least one particular emergency responder to the conference. | 01-23-2014 |
| 20140023065 | METHOD AND APPARATUS FOR PROVIDING A WIDE AREA NETWORK INFRASTRUCTURE - A method and apparatus for providing a wide area network infrastructure for providing services on IP networks such as Voice over Internet Protocol (VoIP) and Service over Internet Protocol (SoIP) networks are disclosed. For example, an enterprise customer may subscribe to a service for obtaining a reliable wide area network infrastructure for communicating among two or more customer locations. The network service provider creates a virtual private network in the public domain and another virtual private network in the private domain to interconnect the customer locations. It then connects each customer edge router to two provider edge routers one in each domain and/or instances of provider edge functionality in each domain. Routes are then advertised via two control planes to both virtual private networks. | 01-23-2014 |
| 20140023066 | ACCESS GATEWAY MANAGEMENT SYSTEM - An Access Gateway Management System (AGMS) allows telephone operating companies to transition their existing wireline customers over to Voice over the Internet Protocol (VoIP) technology without having to invest in new workflow processes, systems, or maintenance facilities by adapting the Operational Support Systems interfaces currently employed for managing legacy circuit-switched switching systems to manage Line Access Gateways (LAGs), which are the generic line termination systems employed in VoIP infrastructure. The AGMS also configures and adapts metallic loop test systems currently deployed for the purpose of routine maintenance and troubleshooting of subscriber lines terminating directly or indirectly (through access systems) on existing switching systems to continue to provide this functionality when the lines terminate on LAGs. Synchronization of the subtended LAGs is coordinated with the legacy network by the AGMS. | 01-23-2014 |
| 20140023067 | Telephone Call Processing Method and Apparatus - Methods, apparatus and computer program products for processing signaling information for telephone call attempts in a packet-based telephony service. An overload protection node is introduced to protect a signaling node from an overload of signaling information for telephone call attempts. A characteristic of signaling information for telephone call attempts in the packet-based telephony service is monitored by the overload protection node and on the basis of the monitored characteristic, the overload protection node processes signaling information for telephone call attempts according to one of a number of different modes of operation. A mode of operation may involve transmitting signaling information for a call attempt to the signaling node or selecting one or more call attempts at the overload protection node for the purpose of reducing overload in the signaling node. | 01-23-2014 |
| 20140029605 | SYSTEMS AND METHODS FOR COMMUNICATING A STREAM OF DATA PACKETS VIA MULTIPLE COMMUNICATIONS CHANNELS - Systems and methods of preventing an Internet service provider from identifying a stream of data packets as carrying a voice over Internet protocol telephony communication can make use of encryption techniques to prevent the Internet service provider from examining the content of the data packets. Also, multiple communications channels may be established between a telephony device and elements of an IP telephony system. A stream of data packets bearing the media of an IP telephony communication is then separated into sub-streams, and each sub-stream is sent through a different one of the communications channels. This prevents an Internet service provider from identifying a stream of data packets as bearing the media of an IP telephony communication based on a pattern in the data traffic. | 01-30-2014 |
| 20140029606 | SYSTEMS AND METHODS FOR COMMUNICATING A STREAM OF DATA PACKETS VIA MULTIPLE COMMUNICATIONS CHANNELS - Systems and methods of preventing an Internet service provider from identifying a stream of data packets as carrying a voice over Internet protocol telephony communication can make use of encryption techniques to prevent the Internet service provider from examining the content of the data packets. Also, multiple communications channels may be established between a telephony device and elements of an IP telephony system. A stream of data packets bearing the media of an IP telephony communication is then separated into sub-streams, and each sub-stream is sent through a different one of the communications channels. This prevents an Internet service provider from identifying a stream of data packets as bearing the media of an IP telephony communication based on a pattern in the data traffic. | 01-30-2014 |
| 20140029607 | VoIP Phone Authentication - Described are computer-based methods and apparatuses, including computer program products, for voice over internet protocol (VoIP) phone authentication. In some examples, the method includes receiving an authentication request from a computing device; authenticating the computing device for access to a network based on the authentication request; determining if a VoIP endpoint device is associated with a network address associated with the authentication request; and authenticating the VoIP endpoint device if the VoIP endpoint device is associated with the network address. | 01-30-2014 |
| 20140036906 | TAGGING VoIP ORIGINATED TRAFFIC - A network node receives a Session Initiation Protocol (SIP) or H.323 signaling packet associated with data traffic, and determines if the data traffic originated from a source node or source network as Voice over Internet Protocol (VoIP) traffic. The network node tags a header of the SIP or H.323 signaling packet with a tag that identifies the data traffic as a VoIP originated call based on the determination, and sends the tagged SIP or H.323 signaling packet towards a destination. The network node further sends data identifying the data traffic as a VoIP originated call to an administrative system for at least one of call billing, rating, settlement or taxation purposes. | 02-06-2014 |
| 20140036907 | METHOD AND SYSTEM FOR COMMUNICATING ACROSS TELEPHONE AND DATA NETWORKS - A method and system for communicating across telephone and data networks are disclosed. According to one embodiment, a computer-implemented method, comprises receiving a first call from a first user phone that converts the first call from a format of a first local phone network to a first digital call. The first digital call is transmitted over a large area data network. The first digital call is converted to a format of a second local phone network to generate a second call. The second call is transmitted to a second user phone over the second local phone network. A real-time bi-directional voice communication session is established between the first user phone and the second user phone. | 02-06-2014 |
| 20140044122 | SYSTEMS AND METHODS OF MAKING A CALL - Systems and methods of making calls are provided. A particular method includes receiving input indicating a destination address at a mobile communication device. The method also includes determining whether the destination address is of a predetermined type. The method further includes initiating a call to a communication bridge via a mobile communication network when the destination address is of the predetermined type. The method also includes sending an instruction to the communication bridge to initiate a communication connection to the destination address. | 02-13-2014 |
| 20140044123 | SYSTEM AND METHOD FOR REAL TIME COMMUNICATING WITH A CLIENT APPLICATION - A system and method for communicating with a client application that can include establishing a client signaling communication channel with a first client application; receiving a communication request from the first client application through the client signaling communication channel, wherein the communication request contains at least an authentication token and a specified communication destination; verifying the authentication token; if the authentication token is verified, at the system bridge, establishing a signaling communication channel with the communication destination and a second media communication channel with the specified communication destination; at the system bridge, establishing a first media communication channel with the client application; and merging the first media communication channel with the second media communication channel. | 02-13-2014 |
| 20140044124 | Communication Networks in Which an Application Server and Multiple Directory Numbers are Used to Provide Internet Protocol Like Features to Time Division Multiplexed Phone Lines - A communication network includes a switching system, an application server, and a softswitch that communicatively couples the application server to the switching system. The switching system is configured to detect a first call to a primary directory number associated with a phone line and to forward the first call to the application server. The application server is configured to instruct the softswitch to generate a second call to a RingMaster directory number associated with the phone line. And the softswitch is configured to bridge the first and second calls responsive to detection of a communication path completion to the phone line. | 02-13-2014 |
| 20140044125 | Outbound Communication Session Establishment on a Telecommunications Network - Disclosed are techniques for establishing a communication session in a call server between communication devices. A call server receives a communication session establishment message from a communication device that has an associated VoIP telephone number. The message is indicative of the communication device wanting to place a call to a target communication device that has an associated target telephone number. The communication session establishment message includes the VoIP telephone number and the target telephone number. The call server extracts the VoIP telephone number and establishes a first communication link with the VoIP device. The call server then extracts the target telephone number and establishes a second communication link with the target communication device. The call server may then join the first and second communication links to establish a communication session between the communication devices. | 02-13-2014 |
| 20140050214 | GATEWAY AND METHOD FOR ESTABLISHING VOICE COMMUNICATION OVER NETWORK USING THE GATEWAY - A gateway connects to a communication device, and communicates with a voice communication server through a network. When the communication device requests to establish a voice communication with a third-party communication terminal, the gateway parses the request to calculate a network bandwidth sufficient to establish the voice communication, and then requests the voice communication server to allocate the required network bandwidth for establishing the voice communication. Then voice data streaming sent from the communication device and the third-party communication terminal are respectively processed to generate RTP packets. The RTP packets are transmitted between the communication device and the third-party communication terminal, so as to realize the voice communication between the communication device and third-party communication terminal. | 02-20-2014 |
| 20140050215 | METHOD AND APPARATUS FOR VIRTUALIZING PRIVATE BRANCH EXCHANGE - The present invention relates to a method and an apparatus for virtualizing a private branch exchange by unifying resources of the private branch exchange into a virtual server thereby to form an internet-based (cloud) environment and using redistribution of the unified resources. A virtual system includes a virtual server configured to register terminals included in physical local system, which performs management to provide registration and services with terminal equipments included in the physical local system, and to redistribute the registered terminal equipments; and at least one virtual local system formed by redistributing the terminal equipments registered in the virtual server. | 02-20-2014 |
| 20140056295 | INTEGRATED INFORMATION COMMUNICATION SYSTEM - A communication system, for functioning without the use of dedicated lines or the Internet so as to ensure communication speed, communication quality, and communication trouble countermeasures, including a communication network and domain name server. The domain name server includes a domain name tree with a country number of a telephone number as a level 2 domain name of the domain name tree, and the domain name server receives, from a terminal, a telephone number of a destination terminal. Furthermore, based on the telephone number of the destination terminal, the domain name server (i) seeks out, in the domain name tree, an Integrated Information Communication System (ICS) user address of the destination terminal, and (ii) sends the ICS user address to the terminal, such that the communication system receives, from the terminal, the ICS user address as a destination address, and sends the ICS user frame to the destination terminal. | 02-27-2014 |
| 20140056296 | Bandwidth Management and Codec Negotiation Based on WAN Topology - A system for bandwidth management and codec negotiation, according to one embodiment of the present invention comprises: a configuration storage module having supported codecs storage, codec lists and preferred site settings storage, and a call manager having an extension module, a trunk module, a location service engine, a codec manager, a bandwidth manager, and a media manager. The codec manager and the bandwidth manager used for negotiating a codec for a call between two endpoints. The present invention also includes a number of methods including a method for negotiating a codec for a call, a method for managing bandwidth for a call, a method for adding a description of a new codec supported by an endpoint, a method for adding an identifier of a supported codec to a codec list and a method for editing code site codec settings. | 02-27-2014 |
| 20140056297 | METHOD AND APPARATUS FOR PROVIDING A RELIABLE VOICE EXTENSIBLE MARKUP LANGUAGE SERVICE - A method and apparatus for providing a reliable Voice Extensible Markup Language (VXML) over packet networks such as Voice over Internet Protocol (VoIP) and Service over Internet Protocol (SoIP) network are disclosed. For example, a service provider may utilize a plurality of content servers that can be accessed by at least one telephony browser. The telephony browser can reach the content browsers directly as well as through a shared server that may load balance among the content servers. When a request for a VXML content, e.g., a VXML application, is received, the telephony browser sends the request to the shared server. If the request fails or a response is not received prior to expiration of a predetermined time interval, then the telephony browser sends a second request directly to one of the content servers that is capable of providing the requested content. | 02-27-2014 |
| 20140064266 | COMMUNICATION METHOD AND SYSTEM THEREOF - Disclosed are a communication method and a communication system. The method includes the following steps: a first communication device sends a communication request signal and a second identification code corresponding to a second communication device to a first switch device via a first exchange device by means of telephone connection; the first switch device sends the communication request signal and the second identification code to a second switch device via a cloud server by means of network connection; the second switch device sends the communication request signal to the second communication device via a second exchange device by means of telephone connection; when the second communication device confirms the communication request signal, a communication connection is set up between the first communication device and the second communication device via the first switch device, the cloud server and the second switch device. | 03-06-2014 |
| 20140064267 | Modem With Voice Processing Capability - A network gateway is configured to facilitate on line and off line bi-directional communication between a number of near end data and telephony devices with far end data termination devices via a hybrid fiber coaxial network and a cable modem termination system. The described network gateway combines a QAM receiver, a transmitter, a DOCSIS MAC, a CPU, a voice and audio processor, a voice synchronizer, an Ethernet MAC, and a USB controller to provide high performance and robust operation. | 03-06-2014 |
| 20140064268 | SYSTEM AND METHOD FOR ROUTING CALLS ASSOCIATED WITH PRIVATE DIALING PLANS - Methods and systems for routing a call including a destination number associated with a PDP including a routing engine operable to route the call to a PDP call resolution server, and a first switch operable to receive an egress path identifier and a PDP telephone number from the PDP call resolution server, the egress path identifier identifying an egress path for routing the call to a destination endpoint associated with the destination number, and the PDP telephone number identifying a selected PDP destination endpoint and a second switch operable to receive the call based on the egress path identifier and route the call to the selected PDP destination endpoint using the PDP telephone number. | 03-06-2014 |
| 20140071978 | VOICE ENERGY COLLISON BACK-OFF - A device may perform audio signal collision detection configured to receive first audio information from a first device and receive second audio information from a second device over a communications network. The device may be further configured to identify a collision of the first audio information and the second audio information, and to address the collision by adjusting at least one of an audio attenuation level and a communications latency of the second audio information responsive to identification of the collision, and store the operational parameters to an adjustment profile system. | 03-13-2014 |
| 20140071979 | APPARATUS AND METHODS FOR A SCALABLE COMMUNICATIONS NETWORK - A method that incorporates teachings of the subject disclosure may include, for example, transmitting a first request for a name authority pointer to a first in-region name server of a plurality of in-region name servers of a first geographic region responsive to determining that a telephone number of a call is located in the first geographic region, transmitting a second request for the name authority pointer to an out-of-region name server associated with a second geographic region responsive to determining that the telephone number is located in the second geographic region, and receiving the name authority pointer from at least one of the in-region name server or the out-of-region server. Other embodiments are disclosed. | 03-13-2014 |
| 20140071980 | SYSTEM AND METHOD FOR MANAGING TRAFFIC BURSTS ON A PER TENANT BASIS - A system that supports multiple contact centers includes a communications network that is coupled between a private network (e.g. MPLS network) and a remote computing environment (e.g. cloud environment). A server system in the remote computing environment monitors health of different network segments (e.g. bandwidth of the connection between the communications network and the remote computing environment, bandwidth of a link used by a tenant to access the private network, etc.). When it is determined that quality of service for voice conversations for one or more contact centers is at risk due to a health status parameter of a network segment reaching a threshold, an appropriate system reaction is triggered. The system reaction may be to offload future calls to a peer remote computing environment to service future calls. The system reaction may also be to cancel outbound campaigns, provide pre-determined “sorry” messages, and the like. | 03-13-2014 |
| 20140079054 | CALLER-CALLEE ASSOCIATION OF A PLURALITY OF NETWORKED DEVICES - The present disclosure generally relates to systems and methods for establishing and maintaining communication between two or more communication devices coupled to communication networks. Some specific aspects relate to communication between a plurality of communication devices each of which is coupled to a respective network. Other aspects relate to establishing such communication by way of contact lists maintained and facilitated on systems coupled to the networks. Users of multiple communication networks, such as VoIP, PSTN and wireless, employ multiple communication devices to communicate with their contacts. For example, a VoIP enabled computer is necessary to access contacts on a VoIP network and a mobile or cellular telephone is used to access contacts on wireless and PSTN networks. A contact list, stored on one communication device, in some instances, cannot be accessed from another communication device. For example, a contact list stored in a VoIP enabled computer cannot be accessed from PSTN or wireless phone devices. Various embodiments described herein provide a convenient solution that can integrate contacts stored on different communication devices and make them accessible from a single device. | 03-20-2014 |
| 20140086235 | METHODS AND SYSTEMS FOR CONTROLLING SETUP OF CALLS THROUGH COMMUNICATION SYSTEMS - A method by at least one network node is disclosed for controlling setup of calls through a communication system. Information is received for a call request that comprises a network address of an origination device of the call request and a virtual identifier associated with a destination device to which the incoming call is directed. A user call profile is retrieved from among a plurality of user call profiles in a user call profile repository using the virtual identifier to identify the user call profile. The user call profile includes a plurality of rules defined by the user for controlling setup of calls to the destination device. Setup of a call path between the origination device and the destination device is controlled responsive to the user call profile. | 03-27-2014 |
| 20140086236 | APPLICATION OF A NON-SECURE WARNING TONE TO A PACKETISED VOICE SIGNAL - A method is disclosed of applying a non-secure warning tone to a packetized voice signal which includes receiving a voice signal containing a sequence of voice samples; providing a non-secure warning tone signal containing a plurality of tone samples; and modifying the voice signal by selectively including tone samples in the sequence of voice samples. | 03-27-2014 |
| 20140092896 | METHOD AND APPARATUS FOR PROVIDING ACCESS TO REAL TIME CONTROL PROTOCOL INFORMATION FOR IMPROVED MEDIA QUALITY CONTROL - Various embodiments provide methods and systems operable to provide access to real time control protocol (RTCP) information for improved media quality control. An example embodiment includes a message processor to receive a message, the message including information indicative of an RTP port identifier, and to add to the received message information indicative of an auxiliary RTCP port identifier; and a message communication component to communicate the information indicative of an auxiliary RTCP port identifier to a node. | 04-03-2014 |
| 20140092897 | COMPUTER SYSTEM, A TELECOMMUNICATION DEVICE AND A TELECOMMUNICATION NETWORK - A telecommunication network | 04-03-2014 |
| 20140098808 | Methods, Systems, and Computer Program Products for Providing Intra-Carrier IP-Based Connections Using a Common Telephone Number Mapping Architecture - Internet protocol (IP) based calls from a first terminal in an IP based communications system are routed to a second terminal in another communications system. In response to a call setup request at a common communications core that is common to both the IP based communications system and the other communications system, a query is transmitted to a private telephone number mapping database that contains routing information for terminals in both the IP based communications system and the other communications system requesting routing information for the second terminal. Routing information for the call setup request is received from the private telephone number mapping database for routing the call. | 04-10-2014 |
| 20140098809 | SYSTEM AND METHOD FOR PROCESSING MEDIA REQUESTS DURING TELEPHONY SESSIONS - In a preferred embodiment, the method of caching media used in a telephony application includes: receiving a media request; sending the media request to a media layer using HTTP; the a media layer performing the steps of checking in a cache for the media resource; processing the media request within a media processing server; and storing the processed media in the cache as a telephony compatible resource specified by a persistent address. The system of the preferred embodiment includes a call router and a media layer composed of a cache and media processing server. | 04-10-2014 |
| 20140105207 | NETWORK ELEMENT INDEPENDENT VOIP CALL PERSISTENCY - A method includes receiving a signaling protocol message associated with a voice over Internet Protocol (VoIP) call. The method includes identifying a From value of the signaling protocol message. The From value includes at least one symbol. The method includes converting each symbol of the From value to a corresponding American Standard Code for Information Exchange (ASCII) decimal value. A To value of the at least one signaling protocol message is identified. The To value includes at least one symbol. The method includes converting each symbol of the To value to a corresponding ASCII decimal value. The ASCII decimal value of the From value is compared to the ASCII decimal value of the To value to determine a larger integer and a smaller integer. The method includes concatenating the larger integer and the smaller integer to form a remote service identifier based on a predetermined sequence. | 04-17-2014 |
| 20140112333 | Calling an Unready Terminal - A voice or video call is to be established between a caller and a callee based on a call flow that involves a call establishment request and a corresponding call acceptance response. A first call establishment request is sent to a called terminal (of the callee) that is unready to accept the call upon receiving this first call establishment request. Once the called terminal is ready to accept the call, instead of the call acceptance response, a reverse call establishment request for the call is received back from the called terminal. The reverse call establishment request is automatically accepted on behalf of the caller on condition that the reverse call establishment request was received back from the called terminal within a certain time limit. If so, the call is accepted by sending an instance of the call acceptance response to called terminal. | 04-24-2014 |
| 20140112334 | DEVICE, SYSTEM, AND METHOD OF CONVERSATION PROXY - Device, system, and method of conversation proxy. A method of communication includes: instructing a first network element, which intends to send conversation packets to a second network element via a first route that excludes a Session Border Controller (SBC) server, to send the conversation packets to the second network element via a second route that includes the SBC server. | 04-24-2014 |
| 20140112335 | NETWORK TELEPHONY SYSTEM - The present invention includes a network telephone having a microphone coupled to provide voice data to a network, a speaker coupled to facilitate listening to voice data from the network, a dialing device coupled to facilitate routing of voice data upon the network, a first port configured to facilitate communication with a first network device, a second port configured to facilitate communication with a second network device and a prioritization circuit coupled to apply prioritization to voice data provided by the microphone. | 04-24-2014 |
| 20140112336 | TELEPHONY USAGE DERIVED PRESENCE INFORMATION - The present invention relates to a mechanism for providing state information, which bears on the presence of a telephone user, to a presence system. The state information is derived by monitoring events relating to telephony use. Once derived, the state information is directly or indirectly sent to a presence service, which provides presence information to applications requiring such information about the telephone user. The state information preferably bears on the presence, absence, or availability of the telephone user based on their interaction with a telephony device or function. In one embodiment, a telephony switching system is configured to monitor events associated with a telephony device or function and send messages to a presence service over a packet-switched network when the state of the telephony device or function changes. In another embodiment, an IP telephone system is configured to provide state information to the presence service. | 04-24-2014 |
| 20140112337 | MULTI-CHASSIS CASCADING APPARATUS - Embodiments of the present invention relate to the communications field, and provide a multi-chassis cascading apparatus. The apparatus includes a line card chassis LCC, where a fabric interface chip FIC and a switch element SE 1/3 are deployed in each line card chassis LCC; the fabric interface chip FIC is connected to the switch element SE 1/3 that is located in the same line card chassis LCC as the fabric interface chip FIC is; and a switch element SE 2 is deployed in each line card chassis LCC; the switch element SE 1/3 is connected to the switch element SE 2 that is located in the same line card chassis LCC as the switch element SE 1/3 is; and the switch element SE 1/3 is connected to the switch element SE 2 that is located in another line card chassis LCC. | 04-24-2014 |
| 20140119363 | Waved Time Multiplexing - Technologies generally described herein relate to waved time multiplexing. In some examples, a command flit can be transmitted from a sender node of a network-on-chip (“NOC”) to a destination node of the NOC via an intermediate node along a circuit-switched path. The command flit can include an interval period and a release duration. When the command flit has been transmitted, one or more data flits can be transmitted from the sender node to the destination node via the intermediate node along the circuit-switched path. The sender node, the destination node, and the intermediate node can be configured to reserve router resources of the sender node, the destination node, and the intermediate node respectively for circuit-switched traffic during a use duration of the interval period and to release the router resources for packet-switched traffic during the release duration in a waved time multiplex arrangement. | 05-01-2014 |
| 20140119364 | METHOD FOR PLACING CALL IN VOICE CALL CONTINUITY AND TERMINAL AND SERVER THEREOF - A method, server and terminal for providing a Voice Call Continuity (VCC) service, are discussed. According to an embodiment, the terminal includes a storage unit to store operator policy information and user preference information, the user preference information including domain selection information specifying a user-preferred domain to be used when originating the outgoing call from the terminal, the operator policy information including domain selection information specifying a network-preferred domain to be used when originating the outgoing call from the terminal; and a controller to perform a domain selection for the outgoing call based on the user preference information or the operator policy information. | 05-01-2014 |
| 20140119365 | Integration of Voice Chat Services - A communication system provides a user with the ability to redirect telephone calls to a voice chat account and vice versa. In one example, a voice chat gateway may receive communication requests and determine whether communications directed to a first party is to be redirected. If so, the voice chat gateway may reroute the communication request to an appropriate destination. For instance, a user may request that all communications such as telephone calls be rerouted as a voice chat to the user's voice chat account. Alternatively, a user may request that all communications including voice chats be redirected to a telephone number. Users may further be allowed to call a voice chat account through a telephone network and initiate a voice chat with a telephone number through a data network. | 05-01-2014 |
| 20140126569 | SYSTEMS AND METHODS FOR EXCHANGING CALL ROUTING POLICIES FOR VOICE OVER IP CALLS - An emergency call handling system can provide emergency call routing and processing that compliments or modifies the routing and processing provided by conventional enhanced 911 (E-911) and next generation 911 (NG-911) systems. Both the routing and processing can be based on rule sets detailed in emergency call handling profiles (ECHPs). Any entity (e.g., SIP servers, switches, terminals, etc.) within the system can process and route the emergency call by executing the rule sets within the ECHPs associated with the call. The ECHPs are delivered to the entities within the system by value or by reference. A SIP server, e.g., may execute all or a subset of the rule set, and may communicate with an application server to execute other subsets of the rule set. | 05-08-2014 |
| 20140126570 | Connecting a PBX to an IMS-Network - A border gateway ( | 05-08-2014 |
| 20140126571 | TELEPHONE EXCHANGE APPARATUS, TELEPHONE EXCHANGE APPARATUS CONTROL METHOD, AND TELEPHONE SYSTEM - According to one embodiment, a telephone exchange apparatus includes a determining module, a memory and a controller. The determining module determines whether an IP address is transferred through a router, by information notified from the plurality of telephone terminals, when a telephone terminal requests registration. The memory stores a management table associating the terminal ID, the results of determination by the determining module, and the IP address. The controller refers to the management table registered in the memory, and executes one of a first process of establishing connection for communication in a peer-to-peer form between first and second telephone terminals, and a second process of relaying connection by using a relay function, based on a reference results of the management table. | 05-08-2014 |
| 20140133481 | Web Telephone with Integrated Voice and Data - Techniques are provided for more efficient communication between two or more entities via a web telephone with integrated voice and data. A phone call may be made from a web page by a calling party and the calling party's related information may be transmitted to a called party concurrently. In addition, a conference call between several participants can be made by sending a message to each participant directing to a web page for joining a conference call and exchanging texts. Further, a secured web page can display various information of and sent by calling parties during the voice communication. | 05-15-2014 |
| 20140133482 | METHOD AND SYSTEM FOR A MULTITENANCY TELEPHONE NETWORK - A method and system for operating a multitenancy telephony system including a call queue that stores call requests received from a plurality of users; an expandable and contractible telephony resource cluster that establishes call sessions for call requests; a analysis system that calculates capacity requirements of the system; a resource allocator that manages the scaling and operation of the telephony resource cluster; and a plurality of telephony network channels that are used as telephony communication channels for call sessions. | 05-15-2014 |
| 20140140339 | VOICE OVER INTERNET PROTOCOL SYSTEM AND METHOD - A voice over internet protocol (VOIP) system and method are provided. The VOIP method includes steps of generating a phone number for a first terminal device, and a domain name corresponding to that phone number, transmitting the domain name and an IP address to a DDNS server for registration, applying the same procedure to second and third terminal devices, generating a name of a group for the first, second, and third terminal devices, generating a domain name corresponding to the name of the group, and transmitting the domain name and the IP address for registration, and acquiring the domain name for the required name of the group, thus allowing calls to be made and available within the group. | 05-22-2014 |
| 20140140340 | Method and System for Providing a Setup Timer in a SIP-Based Network - A system and method for providing a setup timer in a SIP-based network including initiating a session by transmitting one or more messages to a first user. The system and method also comprises starting a first timer upon transmitting the one or more messages, wherein the first timer is configured to expire after a first predetermined time period. The system and method further comprises starting a second timer upon transmitting the one or more messages, wherein the second timer is configured to expire after a second predetermined time period. The system and method furthermore comprises transmitting one or more instructions upon expiration of at least one of the first timer and the second timer, and taking one or more actions based at least in part on the one or more instructions. | 05-22-2014 |
| 20140146812 | METHOD AND APPARATUS FOR REGISTERING COMMUNICATION DEVICES IN A COMMUNICATION SYSTEM - A system that incorporates the subject disclosure may include, for example, a method for receiving and granting, by a first session border controller, registration requests from a plurality of communication devices, where the plurality of communication devices are assigned to a second session border controller as a primary source for communication services, and where the registration requests are received responsive to the second session border controller being unable to provide services to the plurality of communication devices. The method can further include receiving, by the first session border controller, a re-registration request from one of the plurality of communication devices, and transmitting, by the first session border controller, a message to the one of the plurality of communication devices responsive to receiving the re-registration request to cause the one of the plurality of communication device to register with the second session border controller. Other embodiments are disclosed. | 05-29-2014 |
| 20140146813 | METHOD AND APPARATUS FOR PROVISIONING A SCALABLE COMMUNICATIONS NETWORK - A method that incorporates teachings of the subject disclosure may include, for example, determining at a first directory server of a first regional call processing system whether a new name authority pointer associated with a telephone number is within a first geographic region of the first regional call processing system, transmitting the new name authority pointer to a first name server of the first regional call processing system for provisioning the name authority pointer to the first name server responsive to determining that the telephone number is located within the first geographic region, and transmitting the new name authority pointer to a second directory server for provisioning the new name authority pointer to a second name server of a second regional call processing system responsive to determining that the telephone number is not located within the first geographic region. Other embodiments are disclosed. | 05-29-2014 |
| 20140153562 | SYSTEMS AND METHODS OF ROUTING IP TELEPHONY DATA PACKET COMMUNCIATIONS - Systems and methods performed by a telephony device allow the telephony device to test the quality of multiple potential paths which can be used to conduct a telephony communication. By testing the conditions that presently exist, the telephony device can choose the path that is presently offering the best quality. A telephony communication may be setup over an initial path, and then subsequent testing may determine that it is best to switch to an alternate path offering better call quality. The initial path used for the telephony communication may be the one that offers the fastest initial connection. When multiple potential paths exist, the telephony communication may be conducted over a first path while keep alive messages are communicated over a second path so that the telephony communication can be quickly switched to the second path. | 06-05-2014 |
| 20140153563 | Network Node and Method of Routing Messages in an IP-Based Signaling Network - A method of routing messages in an IP-based signaling network is provided. Further, a network node for performing such a routing of messages in an IP-based signaling network is provided. The network node comprises at least two interfaces and a processing unit configured to route an incoming message received at a first of the at least two interfaces to a second interface of the at least two interfaces. Further, a relay network comprising plural of such network nodes as relay nodes is provided. | 06-05-2014 |
| 20140153564 | METHOD AND APPARATUS FOR PROVIDING ENHANCED SERVICES LOCAL ROUTING - A method and apparatus for enabling a call originated in the VoIP network to be routed from the egress of the VoIP network to the terminating PSTN network using an egress route, such as an appropriate egress access trunk, that is in the same Local Calling Area (LCA) of the called party number are disclosed. This allows the call to be completed to the called party without paying access charge. For example, the method assigns a Billing Telephone Number (BTN) or a Charge Number (CgN) to an egress route between a communication network and a Public Switched Telephone Network (PSTN) network. The method then routes a call originating from the communication network and terminating to the PSTN network using the egress route without incurring an access charge. | 06-05-2014 |
| 20140153565 | METHOD FOR PROCESSING TELEPHONY SESSIONS OF A NETWORK - A method for processing telephony sessions of a network including at least one application server and a call router, the method including the steps of assigning a primary Uniform Resource Identifier (URI) and at least a secondary URI to an application; mapping a telephony session to the primary URI; communicating with the application server designated by the primary URI using an application layer protocol; receiving telephony instructions from the application server and processing the telephony instructions with the call router; detecting an application event; and upon detecting the application event, communicating with the application server designated by the secondary URI. | 06-05-2014 |
| 20140153566 | SYSTEM AND METHOD FOR MANAGING LATENCY IN A DISTRIBUTED TELEPHONY NETWORK - A system and method of preferred embodiments include at a signaling gateway of a first region, receiving a communication invitation of a first endpoint from a communication provider; signaling the communication invitation to a communication-processing server in a second region; in response to communication processing of the communication-processing server, dynamically directing signaling and media of the communication according to processing instructions and resources available in at least the first and two regions; wherein dynamically directing signaling and media communication of the communication comprises selectively routing media communication exclusively through communication resources of the first region if resources are available in the first region or selectively routing media communication between the first endpoint, the gateway, and at least the communication-processing server if media resources are not in the first region. | 06-05-2014 |
| 20140161119 | INLINE POWER SYSTEM AND METHOD FOR NETWORK COMMUNICATIONS - A system and method for coupling a communications device to a primary communications network having a first communications format and to a secondary communications network having a second communications format. The system and method can comprise a first port configured for connecting to the communications device, a second port configured for connecting to the primary communications network and facilitating the communication of the data between the primary communications network and the communications device through the first port. | 06-12-2014 |
| 20140161120 | PROCESSING OF CALL DATA RECORDS - Processing of call data records is disclosed. Data packets captured from a monitored telecommunications network are obtained, decoded, and sessions are identified. The decoded data packets and the identified sessions are filtered with a dynamic filter determining adjustable conditions for the protocol attributes and the session attributes such that the decoded data packets and the identified sessions fulfilling the adjustable conditions are kept in order to get attribute-filtered decoded data packets and attribute-filtered identified sessions. Dynamic call data records are generated such that their structure and contents are determined dynamically on the basis of the attribute-filtered decoded data packets and the attribute-filtered identified sessions. The dynamic call data records are stored. | 06-12-2014 |
| 20140161121 | Method, System and Device for Authenticating IP Phone and Negotiating Voice Domain - A method for authenticating an IP phone and negotiating a voice domain includes receiving an authentication request packet sent by an IP Phone, encapsulating a user name and password of the IP Phone in a RADIUS request packet, and sending the RADIUS packet encapsulating the user name and password of the IP Phone to a RADIUS server. If a result of the authentication performed by the RADIUS server on the IP Phone is that the authentication succeeds, sending a Voice VLAN value to the IP Phone through an extensible authentication protocol EAP extension packet. In the present application, dynamic security authentication and negotiation functions between a client and a server, and rapid deployment of an internal network of an enterprise may be implemented. | 06-12-2014 |
| 20140169363 | INTELLIGENT SOFTPHONE INTERFACE - An interface unit ( | 06-19-2014 |
| 20140169364 | TELEPHONY TERMINAL - Methods and apparatus implementing a telephony terminal for connecting a telephone to a data network. In one implementation, a telephony system includes: a phone connection for connecting to a telephone; a network connection for connecting to a network; and a controller connected to said phone connection and to said network connection; wherein said controller provides a phone service for processing information for said phone connection, said controller provides a network service for processing information for said network connection, and said controller provides a network voice service for converting information to and from a network voice format. | 06-19-2014 |
| 20140177627 | Enabling Quality Voice Communications From Web Page Call Control - A system and method of bypassing the regulated portion of the Public Switching Telephone Network (PSTN) to establish carrier-grade voice transmissions and/or IP data communications between an Internet Calling Person having a first telephone and a first PC coupled to a first Local Service Access Provider (LSAP) and an Internet Called Party having a second telephone and a second PC coupled to a second different Local Service Access Provider (LSAP). | 06-26-2014 |
| 20140177628 | METHOD, APPARATUS AND SYSTEM FOR IMPLEMENTING LOGIN OF IP TELEPHONE NUMBER - Embodiments of the present invention disclose a method, an apparatus and a system for implementing login of an Internet Protocol (IP) telephone number. The method includes: after receiving a neighbor discovery protocol message of a data link layer sent by a connected IP telephone, obtaining, by a communication client, when determining that the communication client has used a communication account to log in to a communication server, from the communication server, an IP telephone number associated with the communication account, and sending the IP telephone number to the IP telephone, so that the IP telephone uses the IP telephone number to execute a login operation, thereby solving the problem in the prior art that the user operation is complex because a manual input manner needs to be used in logging in by using both the communication account and the IP telephone number. | 06-26-2014 |
| 20140185608 | SYSTEMS AND METHODS FOR CONNECTING TELEPHONY COMMUNICATIONS - Systems and methods performed by an IP telephony system are designed to determine when two parties to a recently terminated telephony communication are simultaneously calling each other in an attempt to re-establish a telephony communication. When the IP telephony system determines that this situation is occurring, the IP telephony system acts to connect the two parties, rather than have both of them see their new call setup attempt fail because the other party's telephony device is indicated to be busy. | 07-03-2014 |
| 20140185609 | SYSTEMS AND METHODS FOR PROVIDING INFORMATION IN A CONTACT LIST - Systems and method are provided for deriving contact information for one or more contacts of a contact list. This information is used to modify how contacts on the contact list are displayed. The modification can include changing an order in which the contacts are presented on a contact list. | 07-03-2014 |
| 20140185610 | SELECTIVELY PATCHING ERASURES IN CIRCIUT-SWITCHED CALLS WHOSE FRAME ERASURE RATE RISES ABOVE A THRESHOLD BY ESTABLISHING AND SYNCHRONIZING A VOIP STREAM - The disclosure is related to selectively patching frame erasures in a first stream. A receiver receives the first stream, receives a second stream corresponding to the first stream, detects a missing frame in the first stream, and attempts to replace the missing frame in the first stream with a corresponding frame from the second stream. | 07-03-2014 |
| 20140192799 | MULTI-MEDIA DATA RATE ALLOCATION METHOD AND VOICE OVER IP DATA RATE ALLOCATION METHOD - A multimedia bitrate adaption method for individual users wherein only a set of predetermined bitrates may be selected as the multimedia bitrate of the user. The predetermined bitrates are quantized into exponentially distributed levels, and the users' experiences corresponding to the predetermined bitrates are with the same intervals, thereby increasing the service capacity and improving user satisfaction. | 07-10-2014 |
| 20140192800 | PHONE APPLIANCE WITH DISPLAY SCREEN AND METHODS OF USING THE SAME - A phone appliance and method of use are provided where the phone appliance can be used to make VoIP communications calls. In a preferred embodiment, the phone appliance includes an RF connection for connecting to a computer or other computing device for facilitating the placement of the VoIP communications calls. The phone appliance further includes a display or portal for depicting advertisements provided by various advertisers. The advertisements provided can be used to defray all or part of the cost associated with making VoIP communications calls. The portal can also be used to communicate with businesses for ordering products, such as ordering a pizza, and to perform various services, such as purchasing stocks. In an exemplary system, the phone appliance is used to transmit to a control center information related to the user of the phone appliance, such as interests and buying habits, and queries for receiving additional information for various advertised products and services. The control center transmits the queries to the appropriate vendors for providing the user with additional information. Other functions and features are provided to the phone appliance, such as being able to download e-mail messages stored within or received by the computer. | 07-10-2014 |
| 20140198785 | APPARATUS, METHOD AND SYSTEM FOR PROVIDING NEW COMMUNICATION SERVICES OVER EXISTING WIRING - The invention provides apparatus for providing a next-generation communication system over existing wiring. In one form the apparatus includes an input to receive broadband signals carrying next-generation communication data, a processor to extract the next-generation communication data from the broadband signals and a converter to convert the next-generation communication data into analogue telephone signals. The apparatus is arranged to output the analogue telephone signals at the input of the apparatus. Also described is a related method of providing a next-generation communication system over existing wiring. | 07-17-2014 |
| 20140198786 | MANAGING NETWORK BANDWIDTH - A system for using one or more via sites to manage network bandwidth, according to one embodiment of the present invention comprises a first call manager at a source site receiving an offer message to connect a call. The offer message includes an endpoint media settings list. The first call manager determines a first filtered media preferences list based on a source media settings list and the endpoint media settings list. The first call manager and transmits an invite message to a second call manager at a first via site. The invite message includes the first filtered media preferences list. The first call manager receives from the second call manager a call settings list that includes a description of the call settings negotiated between the source site, the first via site and a destination site. | 07-17-2014 |
| 20140198787 | FORCED HOLD CALL HANDLING IN A VOP ENVIRONMENT - The present invention provides a technique for providing a forced hold service such as is used for an emergency services call, which is supported at least in part over a packet network. The forced hold service acts to effectively hold a connection for the call with a called party, even when the caller takes an action that would normally end a call, such as going on hook, pressing end, or the like. When the caller takes an action that would normally end the call, the forced hold service allows the caller to automatically reconnect to the emergency services provider over the held connection upon going offhook, pressing send, or the like. Alternatively, the emergency services provider can effectively re-engage the call wherein the caller is reconnected over the held connection upon going offhook, pressing send, or the like. | 07-17-2014 |
| 20140204935 | System and Method for Instant VoIP Messaging - Methods, systems and programs for instant voice messaging over a packet-switched network are provided. A method for instant voice messaging may comprise receiving an instant voice message having one or more recipients, delivering the instant voice message to the one or more recipients over a packet-switched network, temporarily storing the instant voice message if a recipient is unavailable; and delivering the stored instant voice message to the recipient once the recipient becomes available. | 07-24-2014 |
| 20140211783 | SYSTEMS AND METHODS FOR INTEGRATING ROUTE AND RANK INFORMATION INTO CALL DETAIL RECORDS - The present technology is directed to systems and methods for integrating route and rank information into call detail records. The system receives information relating to a communication that is established between a first communication device and a second communication device. The information includes routing information for the established communication that includes at least a route identifier and a rank identifier. The system records the received route identifier and the rank identifier in a call detail record that is generated for the established communication. The route and rank information may be analyzed to for various purposes including troubleshooting and quality improvement. | 07-31-2014 |
| 20140211784 | SYSTEMS AND METHODS FOR INTEGRATING ROUTE AND RANK INFORMATION INTO CALL DETAIL RECORDS - The present technology is directed to systems and methods for integrating route and rank information into call detail records. The system receives information relating to a communication that is established between a first communication device and a second communication device. The information includes routing information for the established communication that includes at least a route identifier and a rank identifier. The system records the received route identifier and the rank identifier in a call detail record that is generated for the established communication. The route and rank information may be analyzed to for various purposes including troubleshooting and quality improvement. | 07-31-2014 |
| 20140211785 | SYSTEMS AND METHODS FOR INTEGRATING ROUTE AND RANK INFORMATION INTO CALL DETAIL RECORDS - The present technology is directed to systems and methods for integrating route and rank information into call detail records. The system receives information relating to a communication that is established between a first communication device and a second communication device. The information includes routing information for the established communication that includes at least a route identifier and a rank identifier. The system records the received route identifier and the rank identifier in a call detail record that is generated for the established communication. The route and rank information may be analyzed to for various purposes including troubleshooting and quality improvement. | 07-31-2014 |
| 20140211786 | Method and Apparatus for Creating and Distributing Cost Telephony-Switching Functionality within an IP Network - A system for providing and managing IP telephone calls establishes separate and distinct call legs between IP-capable appliances and routers and between routers, and creates calls, changes calls, and manages telephony functions by joining and disjoining calls legs. In some instances one or more call legs disjoined from an active call are maintained as established to be joined later to other call legs to create other active calls. By managing EP calls as separate and distinct legs functions of intelligent, connection-oriented telephony networks may be simulated in IP telephony systems. The management is provided by software running on processors coupled to routers in the IP network. | 07-31-2014 |
| 20140211787 | Method, Network Node and Application Service for Making Available Call Detail Records in an IP Multimedia Subsystem Type Network - The present invention relates to a method of operating a network node (e.g. an MGCF) in a telecommunications network system during a call, wherein the call has been established through a packet switched and a circuit switched part of the network. The method comprises a step of the network node receiving an indication of the call being terminated. The network node also receives call detail data from a switching network node in the circuit switched part of the network, and includes the received call detail data in a call termination message for making the call detail data available in the packet switched part of the network. Then, the network node transmits the termination message to a call controlling SIP-AS in the packet switched part of the network. The present invention further relates to methods of operating a call controlling SIP-AS and an extended services SIP-AS. | 07-31-2014 |
| 20140211788 | METHOD AND A SYSTEM TO DISCOVER AND ESTABLISH AN ENRICHED COMMUNICATION CHANNEL IN A VOICE CALL - A method and a system to discover and establish an enriched communication channel in a voice call. | 07-31-2014 |
| 20140211789 | PROVIDING REAL-TIME VOICE COMMUNICATION BETWEEN DEVICES CONNECTED TO AN INTERNET PROTOCOL NETWORK AND DEVICES CONNECTED TO A PUBLIC SWITCHED TELEPHONE NETWORK - Systems, methods, and apparatus for providing real-time voice communication between devices connected to an Internet Protocol (IP) network and devices connected to a public switched telephone network (PSTN). In one implementation, the system includes a computer-controlled switch for connection to a local PSTN, for receiving calls from the IP network and the PSTN, and for routing calls to the PSTN and the IP network; and gate interface circuitry connected to the switch and adapted for connection to the IP network. The gate interface circuitry includes gateway circuitry for interfacing the IP network and PSTN voice circuits, and gatekeeper circuitry for performing address translation, admission control, bandwidth management, and zone management. The switch stores at least one PSTN destination address and at least one IP network destination address for each subscriber. The switch routes an incoming call to any one of the destination addresses stored for the subscriber. | 07-31-2014 |
| 20140211790 | MULTI-MODE ENDPOINT IN A COMMUNICATION NETWORK SYSTEM AND METHODS THEREOF - A method, apparatus, and communication network system that allows an endpoint to be simultaneously registered with more than one communications server is described. In one embodiment, the communication network system includes a network, a plurality of communications servers that are coupled to the network, and a plurality of endpoints coupled to the network. Each endpoint is capable of being simultaneously registered with more than one communications server. A communication method for an endpoint involves registering a first logical line of the endpoint with a first communications server, and registering a second logical line of the endpoint with a second communications server. Consequently, flexibility is obtained by allowing an endpoint to choose the registering communications server for each logical line of the endpoint. | 07-31-2014 |
| 20140211791 | ENHANCED TELEPHONY COMPUTER USER INTERFACE ALLOWING USER INTERACTION AND CONTROL OF A TELEPHONE USING A PERSONAL COMPUTER - Enhanced telephony computer user interfaces seamlessly integrate and leverage the features of personal computers and telephones. The manner in which media is presented at a computing system can also be modified automatically in response to detected telephone operations. These modifications can include pausing media in response to a detected telephone call and/or adjusting a volume of the media presentation. The media presentation/volume can also be resumed/restored upon detecting that the telephone call has terminated. | 07-31-2014 |
| 20140219272 | ORIGINATOR MOBILE DEVICE ASSISTED VOICE CALL TECHNOLOGY SELECTION - A smart Voice Over LTE (VoLTE) application for allowing a wireless mobile device to select an appropriate access technology for establishing a voice call with a target mobile device, based on the capabilities of the target mobile device. Selection on the client side allows interoperability of a VoLTE wireless mobile device on a circuit switched network without requiring use of a gateway between the circuit switched and VoLTE networks. If the target mobile device is only configured for legacy circuit switched network calls, the wireless mobile device need not begin the call connection on the VoLTE network and instead may establish the call on the circuit switched network from the beginning. | 08-07-2014 |
| 20140219273 | APPARATUS AND METHODS FOR ORIGINATION OF VOICE AND MESSAGING COMMUNICATION IN A NETWORK - A method that incorporates teachings of the subject disclosure may include, for example, receiving a query from a call session server for a first pointer associated with a telephone number of a terminating device of a requested communication session, transmitting to the call session server the first pointer including a session initiation protocol uniform resource identifier associated with the terminating device to initiate an internet protocol communication session, receiving a notification from the call session server responsive to the call session server failing to initiate the internet protocol communication session, and transmitting to the call session server a second pointer including a telephone protocol uniform resource identifier for originating a circuit-switched communication session responsive to receiving the notification. Other embodiments are disclosed. | 08-07-2014 |
| 20140219274 | PEER-TO-PEER, INTERNET PROTOCOL TELEPHONE SYSTEM WITH PROXY INTERFACE FOR CONFIGURATION DATA - Various embodiments provide a Peer-to-Peer (P2P, Internet Protocol (IP) telephone system. The telephone system includes a plurality of terminals coupled together via an IP network. The terminals cooperate with one another to provide telephony features without a dedicated central controller such as a PBX and/or a KSU controller. The terminals may further receive requests for configuration data residing on other terminals, relay the requests to such other terminals to obtain the request configuration, and return the requested configuration data to the requesting device. | 08-07-2014 |
| 20140233558 | CALL ROUTING AND REAL-TIME MONITORING - A method and system for providing call routing analytics. A virtual session initiation protocol switch is provided and hosted in an Internet cloud-based environment. The switch streams live call detail records to a computer system having a processor configured to process all of the subscriber's call records to monitor route performance for the subscriber. Real-time route performance data is transmitted to the subscriber for display at a subscriber computer. The subscriber can then alter a routing of at least a portion of the call utilizing the switch in response to the real-time route performance data to increase quality of signaling and business performance. | 08-21-2014 |
| 20140233559 | METHOD, APPARATUS, AND SYSTEM FOR ESTABLISHING VOICE COMMUNICATION - Methods, apparatus and systems for establishing voice communication are provided herein. An exemplary method can be implemented by an electronic device. A request initiated by a user can be received. The request can include one of a voice-communication request, a voice-communication-invitation request, a temporary-voice-communication request, a temporary-voice-communication-invitation request, an exclusive-voice-communication request, and an exclusive-voice-communication-invitation request. According to the request, a voice-room number can be obtained from a first server or a second server, without using third-party communication software. The method further includes starting a voice client and entering a voice room directly using the voice client, according to the voice-room number. | 08-21-2014 |
| 20140233560 | CALL ROUTING USING INFORMATION IN SESSION INITIATION PROTOCOL MESSAGES - A system and method for efficient and accurate establishment of SIP sessions between calling and called end-points. A calling endpoint creates a SIP INVITE request including a header portion and a body portion. The header portion includes standard routing information and the body portion includes additional information about the user and relevant to caller's intent. A SIP server receives a SIP INVITE request and determines an address of the called end-point as a function of the standard routing information contained in the header and the additional caller information contained in the body. The SIP server routes the session to the determined address. | 08-21-2014 |
| 20140241339 | TRAVERSAL METHOD FOR ICMP-SENSITIVE NAT - In SIP network environment, a general NAT traversal method will become invalid when an NAT with ICMP (Internet Control Message Protocol) is met. The present invention provides four sessions for SIP, i.e. Login Session, Port Prediction Session, Synchronization Session and Media Session, and the SIP network environment includes a first Internet telephone, a second Internet telephone, a first symmetric NAT, a second symmetric NAT and an SIP proxy server. The first symmetric NAT and the second symmetric NAT are ICMP-sensitive. In the Synchronization Session, the first Internet telephone and the second Internet telephone are designed to transmit packets synchronously to avoid port locking. | 08-28-2014 |
| 20140241340 | SYSTEM AND METHOD FOR SOFTWARE TURRET PHONE CAPABILITIES - Disclosed herein are systems, methods, and non-transitory computer-readable storage media for software turret phone functionalities. A system can display, via a graphical user interface, graphical elements grouped according to a common attribute. Each of the graphical elements can represent an open communication line connecting a first device associated with a first user with a second respective device associated with a second respective user. The system can receive a selection of a graphical element of the graphical elements from the first user, the selection indicating a selected user associated with a respective device. Based on the selection, the system can manipulate a corresponding open communication line connecting the first device associated with the first user with the respective device associated with the selected user. | 08-28-2014 |
| 20140241341 | REGISTRATION OF SIP-BASED COMMUNICATIONS IN A HOSTED VOIP NETWORK - Aspects of the present disclosure involve systems, methods, computer program products, and the like, for implementing a registrar component or functionality in a telecommunications network. In one implementation, the registrar functionality is handled at a Session Border Controller (SBC) or Network Address Translation (NAT) Traversal Manager (NTM) device of the network to alleviate an application server of the network from performing the registration function. | 08-28-2014 |
| 20140241342 | EMERGENCY SERVICES FOR PACKET NETWORKS - The present invention provides a technique for facilitating emergency services via packet networks. Emergency service providers will implement emergency proxies to ensure that proper call setup requests for emergency services are forwarded to the appropriate entities, even if those entities are in overload conditions. The emergency proxies may authenticate and filter call setup requests to ensure that only proper call setup requests are forwarded to help prevent such overload conditions. The emergency proxies may operate solely in a packet network, as well as at the interface between a packet network and a circuit-switched network to assist in call setup requests originating from either the packet network or the circuit-switched network. | 08-28-2014 |
| 20140241343 | Method and Device for Starting Limited-Time Licenses for Telecommunication Systems in a Controlled Manner - The invention relates to a method and a device for starting limited-time licenses for a telecommunication system (OSO MX′) in a controlled manner, said system comprising a number of IP terminals connected thereto, preferably IP telephone devices. The licenses acquired by the respective customer for the telecommunication system (OSO MX′) are downloaded from a central license server (CLS′) on the internet. A date and a time, and thus an activation time period, are configured via a browser dialog using the web-based management component (WBM′) which is connected to the internet for the purpose of an activation the first time the telecommunication system (OSO MX′) is started, whereby the maximally allowed use time for using the licenses at no charge is set. After the maximally allowed use time expires, the use of those licenses for which no right of use has been acquired is prohibited. | 08-28-2014 |
| 20140254584 | NOTIFICATION OF TOO MANY "NO ANSWER" OF FORWARDED TO NUMBER - Method and system for notifying subscribers of calls missed by forwarded to number. The present invention relates to services provided by interne protocol multimedia subsystem and more specifically to a system and method of notifying a subscriber about the performance of selected forwarded number. When an call is forwarded to a forward to identity the application server increments a counter in case the call is not answered by the forward to identity. Once this counter reaches a certain threshold, the subscriber is sent a notification of the number of calls missed by selected forward to identity. | 09-11-2014 |
| 20140254585 | TELEPHONE OUTLET WITH PACKET TELEPHONY ADAPTER, AND A NETWORK USING SAME - An outlet for a Local Area Network (LAN), containing an integrated adapter that converts VoIP to and from analog telephony, and a standard telephone jack (e.g. RJ-11 in North America) for connecting an ordinary analog (POTS) telephone set. Such an outlet allows using analog telephone sets in a VoIP environment, eliminating the need for an IP telephone set or external adapter. The outlet may also include a hub that allows connecting both an analog telephone set via an adapter, as well as retaining the data network connection, which may be accessed by a network jack. The invention may also be applied to a telephone line-based data networking system. In such an environment, the data networking circuitry as well as the VoIP/POTS adapters are integrated into a telephone outlet, providing for regular analog service, VoIP telephony service using an analog telephone set, and data networking as well. In such a configuration, the outlet requires two standard telephone jacks and a data-networking jack. Outlets according to the invention can be used to retrofit existing LAN and in-building telephone wiring, as well as original equipment in new installation. | 09-11-2014 |
| 20140254586 | CALL SETUP USING VOICE OVER THE INTERNET PROTOCOL (VOIP) - A method of performing call setup in a system comprises an origination telephony network, a termination telephony network and a packet switched data network interconnecting therebetween comprises steps of implementing call setup across the two telephony networks by SS7 protocol and implementing call setup within the packet switched data network by H.323 protocol. In particular, the call setup in the data network is not started until information of the resources status in the termination telephony network is available. A novel gateway is provided to implement the method, which comprises both SS7 capabilities and H.323 functionalities. | 09-11-2014 |
| 20140254587 | System and Method for Conveying End-to-End Call Status - A system that incorporates teachings of the present disclosure may include, for example, receiving a session initiation protocol (SIP) subscribe message over a primary call leg requesting outdial event notification on a secondary call leg. Receiving of the call is responsive to a unified messaging system placing a call outside of its local access and transport area. A call is placed on the secondary call leg responsive to receiving outdial information on the primary call leg. Alternative outdial information directed to the calling party is received responsive to the call on the secondary leg being unanswered. Another call is placed on the secondary leg responsive to receiving the alternative outdial information without requiring receipt of another SIP subscribe message over the primary call leg. A calling card server performs SIP messaging associated with one of the SIP notify message or the SIP subscribe message. Other embodiments are disclosed. | 09-11-2014 |
| 20140269674 | METHOD AND APPARATUS FOR RAPID SETUP OF A TELEPHONY COMMUNICATION USING MULTIPLE COMMUNICATION CHANNELS - A first telephony device sets up a first communication channel through an Internet protocol (IP) network for conducting an IP based telephony communication with a second telephony device. The first communication channel includes one or more media relays. The first telephony device then begins to conduct the telephony communication with the second telephony device over the first communication channel. While the initial stages of the telephony communication are ongoing, the first telephony device sets up a second communication channel with the second telephony device that does not utilize media relays. The telephony communication is then switched to the second communication channel. Proceeding in this fashion ensures that a communication channel can be rapidly established between the first and second telephony devices so that the telephony communication can quickly commence. | 09-18-2014 |
| 20140269675 | APPARATUS AND METHODS FOR CONDUCTING COMMUNICATIONS WITH A TELEPHONY DEVICE THAT IS ASSIGNED MULTIPLE IDENTIFIERS ASSOCIATED WITH DIFFERENT GEOGRAPHICAL REGIONS - A telephony device is assigned two telephone numbers, a first telephone number from a first country and a second telephone number from a second country. When a user places an outgoing call to a telephone number in the first country, the caller ID information indicates that the call is originating from the first telephone number associated with the first country. Also, the user is charged only the local termination rates for calls in the first country, regardless of where the telephony device is located when the call is placed. When the user places an outgoing call to a telephone number in the second country, the caller ID information indicates that the call originated from the second telephone number, associated with the second country. Also, the user is charged only the standard local termination rates for calls in the second country, regardless of where the telephony device is located when the call is placed. | 09-18-2014 |
| 20140269676 | SYSTEMS AND METHODS FOR RAPID SETUP OF TELEPHONY COMMUNICATIONS - When an incoming communication is directed to a telephony device that is capable of conducting an IP based communication via an IP telephony system and a cellular-based communication via a mobile telephony service provider, the communication is initially setup as a cellular-based communication via the mobile telephony service provider. At the same time, an IP-based communication channel is setup between the telephony device and an IP telephony system. Once the IB-based communication channel is available, the communication is transitioned from the cellular-based communication channel to the IP based communication channel. | 09-18-2014 |
| 20140269677 | SYSTEMS AND METHODS FOR TRANSITIONING A TELEPHONY COMMUNICATION BETWEEN CONNECTION PATHS TO PRESERVE COMMUNICATION QUALITY - An IP telephony communication being conducted by a user telephony device is transitioned from an IP based communications path to a cellular based voice or video communications path if a quality of the telephony communication falls below a threshold level. The user telephony device and/or elements of an IP telephony system handling the call could detect when the quality is below the threshold level. The communication might also be transitioned to a cellular based voice or video communications channel if the strength of a wireless connection between the user telephony device and a wireless access point providing access to a data network falls below a threshold level. | 09-18-2014 |
| 20140269678 | METHOD FOR PROVIDING AN APPLICATION SERVICE, INCLUDING A MANAGED TRANSLATION SERVICE - The present invention provides a method for providing an application service, such as a managed translation service. In an embodiment, a voice call is established between a source station and a destination station. A decision may be made as to whether language translation is desired, such as from an indicator specifying commencement of a translation session. If the translation service is commenced, then the call is directed to a gateway for transmission over a data network to a translation application, which may be a cloud-based application. The translation may occur in real time or near real time. Translated speech may be displayed to the calling party and/or the called party. | 09-18-2014 |
| 20140269679 | ENRICHING TRAFFIC DATA IN A TELEPHONE NETWORK - A processor is configured to receive call records from telecommunication network and load the call records into a telephone number (TN) list. Local Routing Number (LRN) data identifying ported numbers are obtained from a reference database and applied to the call records to enrich the call records with ported information. Also, Local Exchange Routing Guide (LERG) data associated with the LRN data are obtained from the reference database and applied to the call records to enrich the call records with LERG information. Based on the LERG information, an originating carrier from a routing carrier for the call records is distinguished. The originating carrier is a carrier originating a call and passing the call to the routing carrier for delivery to its final destination. The ownership to the call records are assigned based on the originating carrier. | 09-18-2014 |
| 20140269680 | METHOD FOR BIDIRECTIONAL DATA TRANSMISSION VIA A PACKET-ORIENTED NETWORK DEVICE - A telecommunication system for bidirectional data transmission of a data set between a data transmission device and a data reception device via at least one packet-oriented network device, which includes encapsulation of the data set to enable a connection-oriented data transmission of the data set; connection-oriented transmission of the encapsulated data set by means of at least one mobile telephone from the data transmission device to a base station of a mobile telephone network; evaluation of the data encapsulation protocol in the base station for an unpacking of the data set to enable a packet-oriented data transmission of the data set; and packet-oriented transmission of the data set from the base station to the data reception device. | 09-18-2014 |
| 20140269681 | Method And System For Call Routing - In an embodiment, a method and corresponding apparatus of managing call routing includes sending a first message by a session border controller (SBC) to a routing engine, the first message including event information indicative of an event related to a call, the event being associated with a second message received by the SBC; receiving a response message including call managing information related to the call, the call managing information being determined based on at least part of the event information, the call managing information to be returned to the routing engine in a subsequent message related to the call; and maintaining the call managing information received in the response message, the call managing information to be returned to the routing engine in a subsequent message related to the call. | 09-18-2014 |
| 20140269682 | Voice Over IP (VoIP) Network Infrastructure Components and Method - A voice over Internet protocol communication system and method provides infrastructure components as intermediaries between networks, the components include multi-protocol session controllers and a multi-protocol signaling switch as well as a management system. The session controllers process calls and participate in the calls that flow through it. The session controllers process calls that are either at the edge of the network or at the core of the voice over Internet protocol network. The session controllers associate calls with one another in call peers for incoming calls as ingress call peers and for outgoing calls as egress call peers. A centralized database of call routing policies is provided to the session controllers. The session controllers provide cost management, topology hiding, and inter-working, or conversion, of calls from SIP networks to H.323 networks for both voice and video. | 09-18-2014 |
| 20140286331 | MULTI-TRAVERSAL METHOD FOR NAT IN BREAK-IN - In SIP network environment, a general traversal method for a port restricted NAT will become invalid when other users break in. The present invention provides four sessions for SIP, i.e. Login Session, Port Prediction Session, Multi-Traversal Session and Media Session, and the | 09-25-2014 |
| 20140286332 | CIRCUIT-SWITCHED AND MULTIMEDIA SUBSYSTEM VOICE CONTINUITY - The present invention moves service control, including call control, for a user element from a cellular network to a multimedia subsystem (MS), such as the Internet Protocol (IP) Multimedia Subsystem (IMS). Call control is provided by the MS irrespective of whether the user element is using cellular or WLAN access for the call. Call control for originating and terminating calls in the CS or MS as well as transferring calls between a circuit-switched subsystem (CS) and MS is anchored at a continuity control function (CCF) in the MS. All call signaling for the call is passed through the CCF. The CCF is a service provided in the user element's home MS and anchors the user element's active CS calls and MS sessions to enable active roaming across the CS and MS. | 09-25-2014 |
| 20140286333 | METHOD AND SYSTEM FOR LOCAL CALLING VIA WEBPAGE - The present disclosure provides a method for local calling via a webpage, including that: a local calling operation instruction is serialized, the serialized local calling operation instruction is filled into a webpage as a hyperlink character string, and the webpage is sent to a server or a local operating system; and a browser acquires the webpage from the local operating system or the server according to a model of the operating system local to the browser local to the browser, de-serializes the hyperlink character string in the webpage, and requests, according to an instantiated local calling operation instruction obtained through the de-serialization, the operating system local to the browser to execute a local calling operation indicated by the instantiated local calling operation instruction. The present disclosure further provides a system for local calling via a webpage. With the method and system provided in the present disclosure, a new local calling function may be added without updating any browser code, thereby improving universality of local calling. | 09-25-2014 |
| 20140293994 | Network System for Configurable Delivery of Combined Power and Data Signals Over Twisted Pair Wiring - A telecommunications panel and associated system are disclosed. In one example, the panel includes a panel housing having a first side and a second side, one or more data connectors on the first side, and a power input signal connector on the first side. The panel includes one or more combined power output and data signal connectors on the second side, each of the combined power output and data signal connectors configured to electrically connect to a twisted pair cable and including a plurality of twisted pairs each having first and second wire contacts. The one or more twisted pairs are configured to carry a power signal as a direct current voltage difference between the first and second wire contacts, and the remaining twisted pairs from the plurality of twisted pairs are configured to carry differential data signals. The telecommunications panel is configured to selectably allow pairs of the remaining twisted pairs from the plurality of twisted pairs to cooperate to carry a power signal. | 10-02-2014 |
| 20140293995 | COMMUNICATION MANAGEMENT SYSTEM, COMMUNICATION MANAGEMENT METHOD, AND COMPUTER PROGRAM PRODUCT - A communication management system includes a storage unit, a receiving unit, and a first transmitting unit. The storage unit stores therein: an address of a first terminal that establishes a first session with a relaying apparatus; an address of a conversion system that establishes a second session with the relaying apparatus, the conversion system mutually converting a first communication scheme and a second communication scheme; information for identifying the second terminal that establishes a third session with the conversion system; and a host name of another communication management system managing an address of the second terminal. When the receiving unit receives, from the first terminal, a request to start communication between the first terminal and the second terminal, the first transmitting unit transmits, to the relaying apparatus, the address of the first terminal, the address of the conversion system, the terminal identifying information, and the host name. | 10-02-2014 |
| 20140293996 | Method and System for Locating a Voice over Internet Protocol (VOIP) Device Connected to a Network - A method and system for locating a device connected to a network by determining a current network address for the device and comparing the current network address to a network address in a user profile. If the network addresses match, the device is located based on a physical address associated with the network address in the user profile. | 10-02-2014 |
| 20140293997 | Method, Apparatus, and System for Implementing VOIP Call in Cloud Computing Environment - A method for implementing a VOIP call in a cloud computing environment and relates to the VOIP call field. By using an RDP proxy to implement bidirectional transmission of voice streams between a cloud desktop client and a communication peer end, and further implement a VOIP call, a communication delay and load of a cloud desktop virtual machine are reduced. The method is used for a VOIP call in a cloud computing environment. | 10-02-2014 |
| 20140301386 | METHODS AND SYSTEMS FOR PROVIDING AND PLAYING VIDEOS HAVING MULTIPLE TRACKS OF TIMED TEXT OVER A NETWORK - The present invention relates to video provided over one or more networks. Methods and systems for providing, playing, and/or editing video having multiple tracks of timed text are provided in different embodiments of the present invention. | 10-09-2014 |
| 20140307730 | VOICE OVER INTERNET INTEGRATION - An Internet telephony system (ITS) comprising an enclosure, an interface to a packet switched computer network; a communications processor, configured to implement a voice over Internet Protocol communication, to execute at least a client portion of a telephony control application supporting at least a coordinated delivery of data presented to the user through a user interface and a conversation using the voice over Internet Protocol communication; and to execute a web browser; and at least one manual control input. | 10-16-2014 |
| 20140307731 | CALL ROUTING - An inbound traffic allocation module is configured to store, in a database, data received from a plurality of site-services, and to determine a route capacity based at least in part on the received data, data received from each of the site-services including data related to at least one of a health and a busyness of the site-service. A traffic manager module is configured to retrieve the data from the site-services and to provide the data to the inbound traffic allocation module. A service selection engine module is configured to receive a request to route a call, and to route the call to one of the site-services based at least in part on the route capacity associated with the site-service. | 10-16-2014 |
| 20140314074 | WEB SERVICES INTERFACE - A Call Session Control Function (CSCF) entity in an IP Multimedia Subsystem (IMS) network comprises a first interface for interfacing with other entities and uses signalling in an Extensible Markup Language (XML) format. The other entities that the CSCF interfaces with can be located outside the IMS network, such as servers supporting third party IT or web-based applications, or within the IMS network. The CSCF directs XML based service requests by filtering XML messages received via the first interface. | 10-23-2014 |
| 20140321453 | METHOD AND SYSTEM FOR ROUTING MEDIA CALLS OVER REAL TIME PACKET SWITCHED CONNECTION - A method for routing media calls over a real time packet switch connection includes providing a session controller for connecting to a network. The method further includes providing a signaling switch for connecting to the session controller. The method further includes controlling call routing in the network with the session controller where the call routing control includes identifiers for elements in at least two layers of a seven layer model. The call routing control includes a preference for a codec for the call. The call routing is carried out taking into consideration a mean opinion score qualifier from previous calls having a same source and destination. | 10-30-2014 |
| 20140321454 | Converged Voice Services - Novel tools and techniques for providing a subscriber with converged voice services, in which calls to one of the subscriber's telephone numbers is routed to some or all of the subscriber's telephone numbers (e.g., sequentially or simultaneously, depending on system configuration and/or subscriber preference). Some implementations employ a VoIP switch to handle call distribution among the subscriber's various phone lines, even if some (or all) of the subscribers telephone lines are not VoIP lines. In such implementations, upon receiving an incoming call directed to one of the subscriber's lines, a service switching point will route the call to a VoIP switch. Instructed by an application server, the VoIP switch will set up call legs (e.g., via the PSTN) to one or more of the subscribers' telephone numbers. Optionally, when one of the call legs is answered, the application server will instruct the VoIP switch to disconnect the other call legs. | 10-30-2014 |
| 20140321455 | METHOD TO PROCESS A CALL REQUEST - Establishing a communication session in a packet-based network. A communication session request is received from an originating device. The communication session request includes a destination address. A communication session is established with the originating device. The communication session includes a communication session identifier. Based on the destination address, a first media path is set up between the originating device and a first destination device, the first media path not including a communication session controller. The first media path between the originating device and the first destination device is taken down while maintaining the communication session with the originating device. After taking down the first media path, a second media path in the communication session is set up using the communication session identifier. The second media path extends between the originating device and a second destination device, the second media path not including the communication session controller. | 10-30-2014 |
| 20140321456 | PACKET-SWITCHED CORE NETWORK ARCHITECTURE FOR VOICE SERVICES ON SECOND- AND THIRD-GENERATION WIRELESS ACCESS NETWORKS - Communications systems and methods with an evolved packet-switched core network architecture to enable voice services on second- and third-generation wireless access networks. The systems and methods permit unmodified 2G and 3G mobile devices to conduct voice calls using conventional circuit-switched user-plane and control-plane protocols at the air interface while the voice calls are switched at the back-end using a packet-switched core network. The system may include a translation module at a controller component that is configured to provide both user-plane and control-plane translation functions between an unmodified 2G/3G mobile device that utilizes circuit-switched protocols for a voice call and the packet-switched core network that utilizes packet-switched protocols to switch the voice call. | 10-30-2014 |
| 20140321457 | PACKET HANDLER FOR HIGH SPEED DATA NETWORKS - An improved packet handler for VoIP cable modems and other high-speed digital devices includes a direct communication link via hardware among internal processing components. Incoming and outgoing digital information packets are filtered into MAC packets, voice PDU packets, and non-voice PDU packets, such that priority can be given to relaying voice packets and minimizing potential voice delay within the cable network. Hardware components, including specialized logic circuitry, modify voice packets to an appropriate signal form for subsequent signal processing or signal transmission. Proprietary bus communication protocols can also be provided to facilitate relay of packets between a central processing unit (CPU) and a digital signal processor (DSP) within a VoIP cable modem. Line cards including subscriber line interface circuit (SLIC) and subscriber line audio processing circuit (SLAC) components provide analog-to-digital (A/D) and digital-to-analog (D/A) conversion functionality. | 10-30-2014 |
| 20140334479 | Routing Technique - A first communication device sends a call request to a second communication device. The call request comprises a source address associated with the first device and a destination address associated with the second communication device. A communication system modifies the call request by replacing or augmenting the source address with a dynamic address and adding a key associated with the source address. The modified request is sent to the second device. | 11-13-2014 |
| 20140334480 | METHOD AND APPARATUS FOR PROVIDING NETWORK BASED SERVICES TO NON-REGISTERING ENDPOINTS - Many of the current IMS standards and enriched services were originally designed for the individual subscribers that are serviced by the wireless network. However, the IMS standards do not fully address the problem of providing the IMS enriched services and features to business PBX customers or wholesale customers that do not directly register to the IMS network. The present invention discloses a method for providing IMS enriched services and features to business PBX customers or wholesale customers through the use of a static provisioning and registration method. | 11-13-2014 |
| 20140334481 | SCALABLE NAT TRAVERSAL - A system and method for traversing a firewall for a voice-over-IP session or other communication session uses four main components: a relay agent, and NAT | 11-13-2014 |
| 20140334482 | METHOD FOR CONVERGING TELEPHONE NUMBER AND IP ADDRESS - A method for converging telephone numbers and Idata addresses follows the steps of (a) accessing from a first memory location of a communication appliance one of an E.164 telephone number or an IPv6 address; (b) using the E.164 criteria of [Country Code—Identification Code—Subscriber Number 1—Subscriber Number 2—Extension—Ext2—Ext3] for a telephone number, converting that number to an IPv6 hexadecimal notation IP address in the format [::::[]:[:[]], and converting in the reverse for an IP address to a telephone number; (c) storing the telephone number or address in a separate memory location of the appliance; and (d) depending on the nature of a communication session initiated by a user, retrieving the appropriate IP address or telephone number as the destination for the communication. | 11-13-2014 |
| 20140334483 | CALL SERVER SELECTION - In a click-to-call communication environment, the present invention is employed to select an appropriate call server to use when establishing a call between two endpoints. A computing terminal provides a request to initiate a call between the two endpoints. The request is passed to a service node directly or through any number of intermediate nodes, such as a web server. The request may identify a source and a destination for the call. The service node will select a call server to use for establishing the call between the two endpoints based on the destination for the call, and send instructions to the call server to initiate the call. In response, the call server will initiate the call between the two endpoints. | 11-13-2014 |
| 20140334484 | SYSTEM, DEVICE, AND METHOD OF VOICE-OVER-IP COMMUNICATION - The present invention includes devices, systems, and methods of Voice-over-Internet Protocol (VoIP) communication. For example, a method includes: receiving a data stream comprising a set of VoIP packets; and modifying a Real Time Protocol (RTP) header of at least one of said VoIP packets to modify a jitter buffer delay of said data stream. Optionally, the method includes decreasing the jitter buffer delay by: dropping at least one packet from said data stream; and decreasing a sequence number and a timestamp value in an RTP header of at least one additional packet subsequent to said at least one packet. Optionally, the method includes increasing the jitter buffer delay by: identifying a pair of consecutive packets in the incoming data stream, the pair of consecutive packets having consecutive sequence numbers; and increasing a sequence number in an RTP header of at least a latter packet in said pair of consecutive packets. | 11-13-2014 |
| 20140341210 | DELIVERING CORRECT NUMBER INFORMATION IN A PRIVATE SIP NETEWORK - A computer device may include logic configured to receive a first Session Initiation Protocol (SIP) message from a telephone device; select a first back-to-back user agent (B2BUA) and a first screened telephone number (STN) associated with the first B2BUA; generate a second SIP message that includes a first destination address of the first B2BUA, and the first STN; and send the second SIP message to the first B2BUA. The logic may further determine that an acknowledgement response has not been received from the first B2BUA; select a second B2BUA, and a second STN associated with the second destination B2BUA, in response to determining that the acknowledgement response has not been received; generate a third SIP message that includes a second destination address of the second B2BUA, and the second STN; and send the third SIP message to the second B2BUA. | 11-20-2014 |
| 20140341211 | METHOD, APPARATUS, AND COMMUNICATION SYSTEM FOR ALLOCATING AND MANAGING VOICE CHANNELS - Methods, apparatus and communication systems for allocating and managing voice channels are provided. After receiving first voice data, a voice server can search for a voice channel corresponding to a first user of the first voice data. When successfully finding the first voice channel corresponding to the first user, the voice server can write the first voice data to the first voice channel. When successfully allocating a second voice channel that is currently already-timed-out to the first user, the voice server can write the first voice data to the second voice channel, and mark the second voice channel as assigned to the first user. Utilizing efficiency of the voice channels is therefore improved and deploying of hardware resources can be reduced. | 11-20-2014 |
| 20140341212 | SYSTEMS AND METHODS OF PROVIDING COMMUNICATIONS SERVICES - An IP telephony system allows a user to register a telephony device that receives its native telephony service from a different telephony service provider as an extension telephone. The user can then place calls through the IP telephony system using the extension telephone. Such calls may or may not be established using the extension telephone's native telephony service provider. | 11-20-2014 |
| 20140341213 | SYSTEMS AND METHODS OF PROVIDING COMMUNICATIONS SERVICES - An IP telephony system allows a user to register a telephony device that receives its native telephony service from a different telephony service provider as an extension telephone. The user can then place calls through the IP telephony system using the extension telephone. Such calls may or may not be established using the extension telephone's native telephony service provider. | 11-20-2014 |
| 20140341214 | METHOD AND APPARATUS FOR DETECTING ONE OR MORE PREDETERMINED TONES TRANSMITTED OVER A COMMUNICATION NETWORK - A gateway includes a network interface and an apparatus for detecting predetermined tones The apparatus includes an input to receive a signal transmitted over the network interface, a frequency divider to divide the signal into two different components, each component being associated with a different frequency sub band, wherein each frequency sub band is selected to include a predetermined frequency of a predetermined tone, a frequency discriminator to determine frequencies of tones in the components, and a decision logic block to provide an indication that a first predetermined tone has been detected when a first determined frequency of a first tone in a first component corresponds to a first predetermined frequency of the first tone. | 11-20-2014 |
| 20140341215 | METHOD AND APPARATUS FOR CALL PROCESSING FOR SIP AND ISUP INTERWORKING - A system that incorporates teachings of the present disclosure may include, for example, a server having a controller to adjust a call processing logic for Session Initiated Protocol to Integrated Services Digital Network User Part (ISUP) calls based at least in part on interworking profiles assigned to ISUP trunk groups supporting the calls. Additional embodiments are disclosed. | 11-20-2014 |
| 20140348156 | OPTIMIZING ROUTE SELECTION BASED ON TRANSCODING - Disclosed is a method of selecting a route for data transmission. The method comprises: receiving an ingress codec profile and an identification of a destination; determining whether there is a matching codec pair that includes a codec that is identified in both the ingress codec profile and one or more egress codec profile; identifying one or more compatible codec pairs between the codecs listed in the ingress codec profile and the codecs listed in the egress codec profiles associated with egress trunk groups; prioritizing a list of codec pairs, each codec pair in the list of codec pairs comprising a codec identified in the ingress codec profile and a codec identified in an egress codec profile; and, selecting one trunk group from the egress trunk group list, the selected trunk group having a codec profile that comprises a codec from a codec pair in the prioritized list of codec pairs. | 11-27-2014 |
| 20140348157 | SYSTEM AND METHOD FOR WEB TELEPHONE SERVICES - A web telephone service system comprises a client web page adapted to be displayed on a screen of a computer, the client web page including an element associated with the web telephone service and embedded software code, a client web server hosting the web page and an authentication key, an application server adapted to authenticate the client using the authentication key and determining a client telephone number associated with the client, a media server adapted to translate IP traffic to and from a real-time protocol traffic, a media gateway in communication with a private branch exchange adapted to associate the client telephone number with a dial plan, a SIP trunk adapted to translate VOIP traffic to and from POTS traffic, and whereby a user clicking on the visual element is operable to cause a voice communication line to be automatically established between the computer and a client POTS telephone device. | 11-27-2014 |
| 20140348158 | PROVISIONING VPN PHONES - Methods, systems and computer readable media for provisioning VPN phones are disclosed. | 11-27-2014 |
| 20140348159 | EFFICIENT ADDRESS CACHING FOR PACKET TELEPHONY SERVICES - A method for telephony includes receiving at an Internet telephony service provider a subscriber request to place a call to a telephone number. A cache associated with the internet telephony service provider is queried to check if the cache holds a record for the telephone number. If the cache holds the record, the record is obtained. If the cache does not hold the record, a request is sent to a database server that maintains a database of records associating endpoint user terminal telephone numbers of subscribers with respective packet network addresses of the endpoint user terminal. The call is placed to the endpoint user terminal telephone number via a public switched telephone network whilst the request is sent to the database server to retrieve the packet network address of the endpoint user terminal to which calls to the telephone number should be placed. | 11-27-2014 |
| 20140355599 | SYSTEM FOR COMMUNICATING BETWEEN INTERNET PROTOCOL MULTIMEDIA SUBSYSTEM NETWORKS - A method that incorporates teachings of the present disclosure may include, for example, receiving an assignment to provide communication services to a communication device, supplying a first telephone number mapping system of a first internet protocol multimedia subsystem communication system with contact information of the communication device and a serving call session control function operating in the first internet protocol multimedia subsystem communication system, supplying a second telephone number mapping system of a second internet protocol multimedia subsystem communication system with contact information of the communication device and the serving call session control function, and receiving a session initiation protocol INVITE from an originating serving call session control function of the second internet protocol multimedia subsystem communication system for establishing communications with the communication device. Additional embodiments are disclosed. | 12-04-2014 |
| 20140355600 | SYSTEM AND METHOD FOR PROCESSING TELEPHONY SESSIONS - In one embodiment, the method of processing telephony sessions includes: communicating with an application server using an application layer protocol; processing telephony instructions with a call router; and creating call router resources accessible through a call router Application Programming Interface (API). In another embodiment, the system for processing telephony sessions includes: a call router, a URI for an application server, a telephony instruction executed by the call router, and a call router API resource. | 12-04-2014 |
| 20140362850 | Efficient Transmission of Voice Data Between Voice Gateways in Packet-Switched Networks - A voice gateway receives packets transmitting voice data between two voice gateways, each received packet includes header data and voice media payload, means for establishing a voice trunk between the two voice gateways based on the header data of the received packets. The voice media payloads of the received packets are extracted and combined into a single packet, the single packet including for a voice call a dedicated channel data of the established voice trunk comprising an identifier of the respective voice call and the voice media payload of the respective voice call. A single set of header data is added to the single packet, the single set of header data includes base information for synchronizing the flow of data between the two voice gateways. The single packet is transmitted to the receiving voice gateway via the voice trunk. | 12-11-2014 |
| 20140362851 | Method, Communication System and Communication Terminal for the Transmission of Data - Disclosed is a method, a communication system, and a communication device for transmitting data to a first subscriber, within the framework of a connection signaling from a first primary service communication device of the first subscriber to a second primary service communication device, a primary address information message associated with the first primary service communication device and a secondary address information message associated with a first secondary service communication device of the first subscriber is transmitted to the second primary service communication device. The transmitted address information messages are identified and stored via the primary service communication device. For the transmission of data to be transmitted to the first subscriber, the stored secondary address information message is transferred from the second primary service communication device to a second secondary service communication device, and is transmitted based on the transferred secondary address information message during transmission to the first secondary service communication device. | 12-11-2014 |
| 20140369343 | METHODS, SYSTEMS, AND COMPUTER READABLE MEDIA FOR ASSIGNING SEPARATE DEDICATED BEARERS FOR AUDIO AND VIDEO STREAMS IN A TEST SIMULATION ENVIRONMENT - Methods, systems, and computer readable media for assigning separate dedicated bearers for separate audio and video streams in a test simulation environment are disclosed. In one embodiment, the method includes generating dedicated bearer information associated with each of a dedicated audio bearer and a dedicated video bearer to be used for a communication session and providing the dedicated bearer information to a simulation device. The method further includes, from the simulation device, using the dedicated bearer information to negotiate establishment of the dedicated audio bearer and the dedicated video bearer between the simulation device and a system under test (SUT) and sending simulated traffic data to the SUT over the dedicated audio bearer and the dedicated video bearer in accordance with a mapping based on the dedicated bearer information and identifiers included in the simulated traffic. | 12-18-2014 |
| 20140369344 | Computing Latency Introduced by Media Transcoding Operations - Systems and methods for computing latency introduced by media transcoding operations are described. In some embodiments, a method may include receiving incoming Real-Time Protocol (RTP) packets, each having time of arrival and a payload encoded with a first codec and receiving outgoing RTP packets, each having a time of transmission and a payload encoded with a second codec. The method may also include calculating a latency associated with a transcoding of at least one of the incoming RTP packets into at least one corresponding one of the outgoing RTP packets based upon a difference between the time of transmission of the at least one corresponding one of the outgoing RTP packets and the time of arrival of the at least one of the incoming RTP packets. In some cases, the incoming and outgoing RTP packets may be Voice-over-Internet Protocol (VoIP) packets. | 12-18-2014 |
| 20140376541 | DUAL-TONE MULTI-FREQUENCY (DTMF) PROGRAMMING OF AN AUTO-DIALER - In one or more embodiments, a device is configured to automatically contact a Voice-over-Internet Protocol (VoIP) service provider using a communication system that is not native to the VoIP service provider. The device can receive and/or intercept a representation of a first address that is not directed to the VoIP service provider, and generate a representation of a second address effective to contact the VoIP service provider using the representation of the second address. Some embodiments provide an ability to program and/or query a device using telecommunication signaling. In some cases, a service provider can remotely manage firmware and/or software updates to the device using the telecommunication signaling. Alternately or additionally, an end user can manually program information into the device through the telecommunication signaling, such as a predefined address associated with the service provider. | 12-25-2014 |
| 20140376542 | AUTO-DIALER MANAGEMENT THROUGH FXO INTERFACE - In one or more embodiments, a device is configured to automatically contact a Voice-over-Internet Protocol (VoIP) service provider using a communication system that is not native to the VoIP service provider. The device can receive and/or intercept a representation of a first address that is not directed to the VoIP service provider, and generate a representation of a second address effective to contact the VoIP service provider using the representation of the second address. Some embodiments provide an ability to program and/or query a device using telecommunication signaling. In some cases, a service provider can remotely manage firmware and/or software updates to the device using the telecommunication signaling. Alternately or additionally, an end user can manually program information into the device through the telecommunication signaling, such as a predefined address associated with the service provider. | 12-25-2014 |
| 20140376543 | SYSTEM AND METHOD FOR PROVIDING A COMMUNICATION ENDPOINT INFORMATION SERVICE - A system and method for providing a telephony endpoint information service at a communication platform includes obtaining information of a first endpoint through a set of information collection processes; storing the obtained information in an endpoint repository; receiving an endpoint query request of a communication event, wherein the endpoint query request specifies at least a first endpoint; accessing endpoint information for the first endpoint; and augmenting the communication event according to the accessed endpoint information. | 12-25-2014 |
| 20140376544 | METHOD AND SYSTEM FOR CONVERGING CALL - A method for converging a call is disclosed. A non IP-Multimedia-Subsystem (IMS) network is provided with a non IMS network interworking device, by which the non IMS network is converged with a Voice over Internet Protocol (VoIP) network. The method further includes: when the non IMS network determines that a callee receiving a call request is a convergence call user, the non IMS network forwards the call request to the VoIP network; and the VoIP network initiates a call to the callee. A system for converging a call which implements the aforementioned method is further disclosed. The disclosure takes full advantage of the service characteristic of the convergence network, so as to supply users with convenient service access implementation and reduce the service charge. | 12-25-2014 |
| 20140376545 | Implementing a High Quality VOIP Device - A method is provided for Voice over Internet Protocol (VoIP) devices to communicate over an Internet Protocol (IP) network. The method includes synchronizing the VoIP devices using one or more dual-tone multi-frequency (DTMF) codes over a telephone network, retransmissions of voice packets in bursts, retransmissions of voice packets following a time lag, adjusting the number of retransmissions based on quality of service, retransmission of a missing voice packet identified in a list received from a peer device, discarding low energy voice frames in a jitter buffer to prevent overflow, stopping playout at a low energy voice frame when the jitter buffer is below a minimum buffer size, and selective transmission and retransmission of voice packets based on their energy levels. | 12-25-2014 |
| 20150009986 | SYSTEMS AND METHODS FOR IP AND VOIP DEVICE LOCATION DETERMINATION - A method and system for precise position determination of general Internet Protocol (IP) network-connected devices. A method enables use of remote intelligence located at strategic network points to distribute relevant assistance data to IP devices with embedded receivers. Assistance is tailored to provide physical timing, frequency and real time signal status data using general broadband communication protocols. Relevant assistance data enables several complementary forms of signal processing gain critical to acquire and measure weakened or distorted in-building Global Navigation Satellite Services (GNSS) signals and to ultimately extract corresponding pseudo-range time components. A method to assemble sets of GNSS measurements that are observed over long periods of time while using standard satellite navigation methods, and once compiled, convert using standard methods each pseudo-range into usable path distances used to calculate a precise geographic position to a known degree of accuracy. | 01-08-2015 |
| 20150009987 | Call Handling for IMS registered user - The present invention proposes a solution for providing IMS services to users having circuit-switched controlled terminals. In particular, it is proposed, in order to allow IMS to take the full call and service control, to combine circuit-switched and packet-based multimedia functionality in a new node type called Mobile Access Gateway Control Function (MAGCF). In particular the present invention provides a method for ensuring that the MAGCF node acts as a roaming anchor point in order to enforce the handling of originating and terminating calls in the IMS. | 01-08-2015 |
| 20150016446 | SIP SIGNALLING - The present invention relates to signalling between entities in a SIP protocol communication in which the transport protocol can be changed during the call to provide improved call reliability. | 01-15-2015 |
| 20150016447 | Reverse Context System - One aspect relates to indicating at least partially relative to an at least one receiving communicating device a called entity information at least partially describing an at least one desired contacting or forwarding entity that an at least one contacting communicating device is attempting to contact. Another aspect can relate to transferring from at least one desired contacting or forwarding entity a called entity information at least partially describing the at least one desired contacting or forwarding entity which an at least one contacting communicating device is attempting to contact. Yet another aspect can relate to filtering out communications having at least one desired uniform resource identifiers that do not contain a prescribed called entity information matching a prescribed pattern. | 01-15-2015 |
| 20150016448 | Session Establishment in an IP Multimedia Subsystem Network - According to a first aspect of the present invention there is provided a method of handling an IP Multimedia Subsystem, IMS, session establishment request received from a calling party at a Multimedia Telephone Application Server, MMTel AS, within an IMS network from a Serving Call Session Control Function, S-CSCF, over an IMS Service Control, ISC, interface. The method comprises, within the MMTel AS: a) determining that said request is an originating call case request; b) establishing a first Multimedia Telephone, MMTel, service engine instance within the MMTel AS and using that first MMTel service engine instance to handle the request according to an originating call case; c) determining whether or not a called party associated with said request is served by said MMTel AS and, if so, establishing a second MMTel service engine instance within the MMTel AS, passing the request to said second MMTel service engine instance, and using the second MMTel service engine instance to handle the request according to a terminating call case; and d) forwarding said IMS session establishment request over said ISC interface to said S-CSCF, or to another S-CSCF serving the called party. | 01-15-2015 |
| 20150023344 | METHOD AND SYSTEM FOR MANAGING TELEPHONY SERVICES IN A UNIVERSAL PLUG AND PLAY HOME NETWORK ENVIRONMENT - The present invention provides a method and system for managing telephony services in a Universal Plug and Play (UPnP) home network environment. In one embodiment, a method includes creating one or more profiles associated with at least one Telephony Control Point (TelCP) in an UPnP home network environment, each of the one or more profiles includes one or more service settings associated with telephony services. The method also includes storing the one or more profiles associated with the at least one TelCP in a service settings database. Additionally, the method includes setting one of the one or more profiles as an active profile for the at least one TelCP, and providing telephony services to the at least one TelCP according to the one or more service settings associated with the active profile. | 01-22-2015 |
| 20150023345 | EXAMPLE-BASED AUDIO INPAINTING - A method for packet loss concealment, comprising: continuously receiving a digital audio stream; extracting audio features from the digital audio stream while the digital audio stream is unharmed; and upon detecting a gap in the digital audio stream, filling the gap with one or more previous segments of the digital audio stream, wherein said filling is based on a matching of the one or more of the extracted audio features with one or more audio features adjacent to the gap. | 01-22-2015 |
| 20150030016 | MEDIA SESSIONS - Measures for enabling media bypass of one or more session border controllers (SBCs) in a telecommunications network which includes a plurality of SBCs. An SBC receives an inbound offer message requesting setup of a media session between an originating endpoint device and a terminating endpoint device. The SBC transmits an outbound offer message to an ensuing SBC, wherein the originating endpoint device address comprised in the inbound offer message is re-written with an SBC address in the outbound offer message. The SBC receives an inbound answer message including a terminating endpoint device address. In response to receipt of the inbound answer message, the SBC transmits an outbound answer message to the originating endpoint device without re-writing the terminating endpoint device address, whereby the SBC and the ensuing SBC are bypassed in the media path for the media session. | 01-29-2015 |
| 20150030017 | VOICE COMMUNICATION METHOD AND APPARATUS AND METHOD AND APPARATUS FOR OPERATING JITTER BUFFER - Voice communication method and apparatus and method and apparatus for operating jitter buffer are described. Audio blocks are acquired in sequence. Each of the audio blocks includes one or more audio frames. Voice activity detection is performed on the audio blocks. In response to deciding voice onset for a present one of the audio blocks, a subsequence of the sequence of the acquired audio blocks is retrieved. The subsequence precedes the present audio block immediately. The subsequence has a predetermined length and non-voice is decided for each audio block in the subsequence. The present audio block and the audio blocks in the subsequence are transmitted to a receiving party. The audio blocks in the subsequence are identified as reprocessed audio blocks. In response to deciding non-voice for the present audio block, the present audio block is cached. | 01-29-2015 |
| 20150030018 | MULTI-CHANNEL MULTI-ACCESS VOICE OVER IP INTERCOMMUNICATION SYSTEMS AND METHODS - The present invention provides systems and methods employing Voice over Internet Protocol (VoIP) technology to provide multi-channel, multi-access voice communication capabilities. | 01-29-2015 |
| 20150036678 | VOIP CLIENT CONTROL VIA IN-BAND VIDEO SIGNALLING - The present document relates to telecommunication networks. In particular, the present document relates to the provision of network interaction services within a telecommunication network. A method for enabling interaction services with a network comprising a network server ( | 02-05-2015 |
| 20150036679 | METHODS AND APPARATUSES FOR TRANSMITTING AND RECEIVING AUDIO SIGNALS - Methods and corresponding apparatuses for transmitting and receiving audio signals are described. A transformation is performed on the audio signals in units of frame in order to obtain transformed audio data of each frame, said transformed audio data consisting of multiple signal components in the frequency domain. These signal components of each frame are distributed into multiple adjacent packets in order to generate packets in which signal components distributed from multiple frames are interleaved. Subsequently, the generated packets are transmitted. Accordingly, in case that packet loss occurs during transmission, the audio signals can be recovered based on the received signal components without consuming additional bandwidth. Therefore, robustness against packet loss can be achieved with little overhead. | 02-05-2015 |
| 20150036680 | ACCESS GATEWAY MANAGEMENT SYSTEM - An Access Gateway Management System (AGMS) allows telephone operating companies to transition their existing wireline customers over to Voice over the Internet Protocol (VoIP) technology without having to invest in new workflow processes, systems, or maintenance facilities by adapting the Operational Support Systems interfaces currently employed for managing legacy circuit-switched switching systems to manage Line Access Gateways (LAGs), which are the generic line termination systems employed in VoIP infrastructure. The AGMS also configures and adapts metallic loop test systems currently deployed for the purpose of routine maintenance and troubleshooting of subscriber lines terminating directly or indirectly (through access systems) on existing switching systems to continue to provide this functionality when the lines terminate on LAGs. Synchronization of the subtended LAGs is coordinated with the legacy network by the AGMS. | 02-05-2015 |
| 20150043571 | ECHO CANCELLER FOR VOIP NETWORKS - An echo canceller for an IP network includes an adaptive filter that models the echo path and generates an estimate of the echo signal from a receiving input signal. The echo canceller subtracts the estimate of the echo signal from a sending input signal to generate a sending output signal with reduced echo. Variation in the echo delay is detected. A delay circuit compensates for the changes in the echo delay to provide proper time-alignment between the estimate of the echo signal and the sending input signal so that the echo signal will be more effectively cancelled. | 02-12-2015 |
| 20150043572 | Efficient Allocation And Usage of Communication Channels for Text Streams - A communication system capable of allocating resources to provide text services to IP telephone users upon receiving requests for text services during a telephone call or during the establishment of the telephone call. The system uses available resources for text services efficiently in that it allocates the resources upon receiving a request for text services and de-allocates the resources upon termination of the telephone call or upon receiving a request for termination of text services during the telephone call. A party to the telephone call can make a request for text services or make a request to terminate text services during the telephone call by activating a button on its IP phone. | 02-12-2015 |
| 20150049754 | TELEPHONY ENABLED INFORMATION EXCHANGE - Systems and methods for IMS telephony enabled information exchange are disclosed. In some implementations, a transfer request to transfer a file from a first user to a second user is received at a IMS gateway and during an ongoing telephone call established between a first device associated with the first user and a second device associated with the second user. The second user on the ongoing telephone call is identified at the IMS gateway. Upon identifying the second user and responsive to the transfer request, the file is transferred from the first user to the second user. | 02-19-2015 |
| 20150049755 | Power Management in an Internet Protocol (IP) Telephone - Power management is provided in an Internet protocol (IP) telephone and system to provide energy savings during times that the IP telephone is not in use or use is not expected. A low-power operating mode disables at least a portion of the IP telephone. The low-power operating mode may be initiated by a command received by the IP telephone from the IP telephone controller according to a schedule, which may be modified locally by the user to individualize the user's schedule. The low-power operating mode may alternatively be activated manually by a user pressing a special key, sequence or combination. The low-power operating mode is canceled upon an indication that a user either is or should be present at the IP telephone. | 02-19-2015 |
| 20150049756 | POTS Telephony over High Speed Data Networks - Novel tools and techniques are provided for delivering plain old telephone service (“POTS”) telephony over high speed data networks. In particular, various embodiments provide tools and techniques for concurrent transmission of POTS voice signals and data signals over the same wire(s) of high-speed data lines or data cables. Various systems and methods might, in some instances, utilize upbanding or rebanding of the POTS voice band to a higher frequency band above the data stream band spectrum for transport of voice concurrent with data over the same wire(s) in the cable. The system might comprise interface devices at either end of a cable segment, one interface device to reband the voice signal and to combine the voice signal with the data signal for each dual-transport wire in the cable, and another interface device at the other end to separate the voice signal from the data signal. | 02-19-2015 |
| 20150049757 | Method for Transmitting and Receiving of an Information-Signal Via a Network, Transmitter and Receiver for Application of Method and Splitter Unit for Application Within the Network - An information-signal (e.g., video-stream of certain quality) (SB1.1, SB1.2, SB1.3, . . . ) is split into two or more (Multicast-) sub-data-streams and transmitted via different channels (CH1,CH2). Thereby, on switching over of two information-signals, seamlessly switch over to another information-signal (e.g., from SD to HD quality) is enabled (in particular at the GOP-boundary in case of video). | 02-19-2015 |
| 20150055646 | METHOD AND DEVICE FOR MAKING AVAILABLE AT LEAST ONE COMMUNICATION DATUM - A method and a device are provided for making available at least one communication datum retrieved during consultation of a multimedia stream on a first terminal. The method includes a step of reception of a request for obtaining the at least one communication datum, a step of extraction of the at least one communication datum from the multimedia stream, a step of transmission to at least one second terminal of the at least one communication datum extracted. The communication datum transmitted thus makes it possible to implement on the second terminal an application making it possible to establish a multimedia communication between the second terminal and a remote device. | 02-26-2015 |
| 20150055647 | System and Method for Registering an IP Telephone - A system and method for establishing connection of an IP telephone to a network may include, in response to receiving a registration request from an IP telephone, generating a command to cause network access devices to ping the IP telephone. The command may be communicated to the network access devices. Ping information may be received in response to the network access devices pinging the IP telephone. A network access device may be selected that has the highest quality network access path to the IP telephone. In response to selecting the network access device that has the highest quality network access path, a network address of the selected network access device may be communicated to a network device to enable the IP telephone to communicate with the selected network access device. Credentials may be communicated to the IP telephone to register with the selected network access device. | 02-26-2015 |
| 20150055648 | GENERATING A COMFORT INDICATOR AT AN ORIGINATING TERMINAL - A call request is sent to establish a telephony session over an Internet Protocol (IP) network between an originating terminal and a destination device. A message responsive to the call request is received from a node connected to the IP network. In response to receiving the message, local generation of a comfort indicator at the 5 originating terminal is performed. | 02-26-2015 |
| 20150063345 | IP MULTIMEDIA SUBSYSTEM SUPPORT FOR PRIVATE BRANCH EXCHANGES - According to an aspect of the present invention there is provided a method of providing support for multiple links between a Private Branch Exchange (PBX) and an IP Multimedia Subsystem (IMS). The method comprises, at an IMS Application Server (AS), configuring the AS with capacity information relating to each of the multiple links, receiving SIP call control messages relating to calls involving the PBX and thereby maintaining usage information for each of the multiple links, the usage information indicating which of the multiple links is being used for each ongoing call involving the PBX. The method further comprises using the capacity information and the usage information relating to each of the multiple links to select one of the multiple links to carry an incoming call terminating at the PBX, and modifying SIP call control messages relating to the incoming call to ensure that the incoming call is carried by the selected link. | 03-05-2015 |
| 20150063346 | CONVERGED MEDIA PACKET GATEWAY FOR A NOVEL LTE DATA AND VOICE CORE NETWORK ARCHITECTURE - An Evolved Packet Core network comprises a converged media packet gateway control element including a GTP-C function of SGW, PGW, and P-CSCF and ATCF functions. The network further comprises at least one converged media packet gateway bearer element disposed remotely from the converged media packet gateway control element, in communication and under the management of the converged media packet gateway control element. The at least one converged media packet gateway bearer element includes SGW and PGW bearer plane terminating S1-U, and ATGW and IMS-AGW NAT functions. | 03-05-2015 |
| 20150063347 | SYSTEMS AND METHODS OF IMPROVING THE QUALITY OF VOIP COMMUNICATIONS - Methods of addressing problems in a voice over Internet protocol (VOIP) telephony system include collecting data on network events, analyzing the data, and taking corrective action when possible. If an IP telephony device is registering with the VOIP telephony system more frequently than necessary, which can indicate the IP telephony device is unnecessarily jumping between proxy services, the IP telephony device is instructed to re-initialize itself. If an IP telephony device sends two successive stay alive registration messages to a proxy server from different ports of a router, which can indicate that a router pinhole is closing between stay alive messages, then the IP telephony device is instructed to send stay alive registration messages more frequently. If data packet statistics indicate that an IP telephony device is experiencing a jitter problem, the IP telephony device is instructed to increase the size of a data buffer for incoming data packets. | 03-05-2015 |
| 20150078370 | SYSTEMS AND METHODS OF ASSIGNING AND USING VIRTUAL TELEPHONE NUMBERS - A single virtual telephone number that is used to route telephony communications is assigned to first and second users if the first and second users rarely, if ever, communicate with the same party. When an incoming call is received on the virtual telephone number, telephony information for the first and second users is reviewed to determine if the calling party is one that has communicated with either of the first and second users in the past. If so, the telephony communication is routed to the user who has communicated with the calling party in the past. | 03-19-2015 |
| 20150078371 | SYSTEMS AND METHODS OF ASSIGNING AND USING VIRTUAL TELEPHONE NUMBERS - A single virtual telephone number that is used to route telephony communications is assigned to first and second users if the first and second users rarely, if ever, communicate with the same party. When an incoming call is received on the virtual telephone number, telephony information for the first and second users is reviewed to determine if the calling party is one that has communicated with either of the first and second users in the past. If so, the telephony communication is routed to the user who has communicated with the calling party in the past. | 03-19-2015 |
| 20150078372 | Voice Data Transmission With Adaptive Redundancy - Voice data transmission with adaptive redundancy creates a voice data packet by packetizing the voice data payload and a number of redundant payloads selected from a set of previous voice data payloads. The voice data from the voice data payload is analysed to determine whether it is a critical or non-critical payload by classifying the received voice data as voiced or unvoiced. If at least a portion of the voice data is classified as unvoiced, the voice data payload is determined to be a critical payload. If it is a critical payload, then the voice data payload is added to the set of previous voice data payloads for inclusion as a redundant payload in subsequent voice data packets. The voice data packet is then forwarded for transmission over the network. | 03-19-2015 |
| 20150078373 | System, Method, and Apparatus for User-Initiated Provisioning of a Communication Device - An embodiment of a method and apparatus for provisioning of a communication device includes receiving a registration request from a first communication device. The registration request includes an address associated with the first communication device. The method further includes registering the first communication device in response to receiving the registration request, placing a call request to the first communication device, and establishing a call session with the first communication device. The method further includes prompting a user of the first communication device for a user identifier, and receiving a user identifier from the user of the first communication device. The method still further includes sending one or more configuration parameters associated with the user identifier to the first communication device. The one or more configuration parameters are operable to configure the first communication device. | 03-19-2015 |
| 20150078374 | Enabling Ad-Hoc Data Communication Over Established Mobile Voice Communications - In one embodiment, a first PC may receive a trigger to establish a data communication session with a second PC over an established voice call between first and second phones over a WAN. In response, the first PC may discover the first phone as an authorized personal area network (PAN) device, and may establish a first PAN communication session between the first PC and the first phone. A request may then be transmitted to the second phone over the established voice call to establish the data communication session between the first and second PCs, and in response, the second phone may discover the second PC as an authorized PAN device from the second phone. A second PAN communication session may thus be established between the second phone and the second PC, and data may be exchanged between the PCs using the PAN communication sessions and the established voice call. | 03-19-2015 |
| 20150085855 | METHOD AND SYSTEM FOR MANAGING THE COMMUNICATION BETWEEN TWO USERS - The present invention relates to a method for managing the communication between a calling user ( | 03-26-2015 |
| 20150085856 | REGISTERING A DEVICE WITH A VoIP CORE NETWORK - Disclosed are a management method and server suitable for managing a request issued by a device on a VoIP core network for the purpose of registering a current address of contact of the device. The management method comprises, on receiving the request, obtaining a number of addresses of contact registered on the core network in association with the public identifier of the device. When that number has reached a maximum registration capacity defined for the public identifier, an interrogation message is sent to the addresses of contact registered on the core network in association with the public identifier if, at the end of a predetermined time period, at least one of the addresses of contact has not responded to said interrogation message or is declared as being inactive in response to the interrogation message, accepting the request. Otherwise the request is rejected. | 03-26-2015 |
| 20150085857 | Method and Apparatus for Enabling Subscriber Lines to Join DSL Vectoring System, and DSL Vectoring System - The present invention discloses a method and an apparatus for enabling subscriber lines to join a DSL vectoring system the DSL vectoring system and can shorten overall joining time of subscriber lines in a DSL system. The method includes grouping subscriber lines into at least two groups according to time when the subscriber lines request to go online, where the at least two groups of subscriber lines that are obtained after the grouping include a first group of subscriber lines and a second group of subscriber lines; starting a joining process for all the subscriber lines in the first group of subscriber lines; and during the joining process of the first group of subscriber lines, putting all the subscriber lines in the second group of subscriber lines into a joining process. | 03-26-2015 |
| 20150092769 | Protocol Translations for Internet Services - An Internet protocol Multimedia Subsystem (IMS) gateway application server includes an originating application server module adapted to invoke call control services in response to requests initiated by a voice over Internet Protocol (IP) (VoIP) client associated with a communication device such as an IP telephone. Disclosed gateway application servers include a proxy server module adapted to notify the communication client of session control messages intended for the communication device. | 04-02-2015 |
| 20150098462 | METHOD AND APPARATUS FOR INITIATING COMMUNICATION SESSIONS - An aspect of the subject disclosure may include, for example, receiving a request from a communication device to initiate a communication session in a packet-switched network, obtaining a first name authority pointer record responsive to determining that there is an undesirable operational state in the packet-switched network, wherein the first name authority pointer record comprises a commented out record, obtaining a second name authority pointer record responsive to determining that there is a desirable operation state in the packet-switched network, wherein the second name authority pointer record comprises a record, and initiating the communication session according to one of the first name authority pointer record or the second name authority pointer record. Other embodiments are disclosed. | 04-09-2015 |
| 20150098463 | TELECOMMUNICATION SYSTEM - A telecommunication system comprising a computer system comprising a store of representations of telecommunication actions and a store of representations of telecommunication rules. A URL is associated with each of the telecommunication actions. The computer system is configured to, at least in part, command the telecommunication actions. Each telecommunication action is carried out, at least in part, in a manner dependent on at least one of the telecommunication rules. In response to an electronic device accessing a URL associated with one of the telecommunication actions, the telecommunication action is carried out, at least in part, in a manner dependent on at least one of the telecommunication rules. The telecommunication system is configured to change the unique URL to a new unique URL in response to a telecommunication action and associate a different telecommunication action with the new unique URL and/or change a telecommunication action associated with the unique URL. | 04-09-2015 |
| 20150103820 | PEER-TO-PEER INTERNET PROTOCOL TELEPHONE SYSTEM WITH SYSTEM-WIDE CONFIGURATION DATA - Various embodiments of the invention provide a Peer-to-Peer (P2P, Internet Protocol (IP) telephone system. The telephone system includes a plurality of terminals coupled together via an IP network. The terminals cooperate with one another to provide telephony features without a dedicated central controller such as a PBX and/or a KSU controller. The terminals further cooperate with one another to maintain system-wide configuration data for the telephone system. | 04-16-2015 |
| 20150110103 | Media Playout for VOIP Applications - In overview, the various embodiments include methods performed by a receiver device processor for implementing one or more buffer space management strategies to free up space in the buffer in response to determining that the buffer is full. In the various embodiments, the receiver device processor may analyze media packets stored in the buffer based on various criteria to identify media packets that may be discarded without significantly impacting playout sound quality to free space in the buffer for incoming packets. | 04-23-2015 |
| 20150117439 | SYSTEMS AND METHODS FOR CONTROLLING TELEPHONY COMMUNICATIONS - A telephony communication system prevents an incoming telephony communication directed to a user from being completed if the user is not authorized to communicate with the calling party. Likewise, a telephony system prevents a user from completing an outgoing telephony communication directed to a called party if the user the user is not authorized to communicate with the called party. The telephony communication system also electronically monitors a telephony communication between first and second parties and censors portions of the telephony communication to terminates the telephony communication if predetermined keywords appear in the communication. | 04-30-2015 |
| 20150117440 | OPTIMIZING CALL BEARER PATH USING SESSION INITIATION PROTOCOL PROXY - A system that incorporates the subject disclosure may include, for example, a serving device receiving a first message from a gateway device coupled to a communication network, the first message comprising a first session descriptor protocol relating to a port of the network connecting to equipment of a first subscriber of the network to originate a call to a mobile communication device of a second subscriber of the network, whereby the serving device and the gateway device comprise nodes on a signaling path for the call. The serving device sends a second message to the gateway device, the second message comprising a second session descriptor protocol relating to the serving device, for delivery to the network to facilitate a bearer path between the network and the serving device, wherein the gateway device is not a node of the bearer path. Other embodiments are disclosed. | 04-30-2015 |
| 20150117441 | SYSTEM AND METHOD FOR CELL PHONE TO CELL PHONE SIGNAL TRANSMISSION VIA THE INTERNET - A telephone system is provided for transmitting telephone signals between first and second mobile stations. The system includes a first internet protocol interface configured to receive an incoming cell phone signal generated by the first mobile station and to transmit the phone signal to the internet. A second internet protocol interface is configured to receive the phone signal sent through the internet by the first internet protocol interface and to transmit the phone signal to the second mobile station, such that users of the first and second mobile stations can engage in a conversation where the phone signals are communicated over substantial distances through the internet. | 04-30-2015 |
| 20150117442 | SYSTEMS AND METHODS FOR REDUCING SIGNALLING IN AN INTERNET PROTOCOL TELEPHONY SYSTEM - A telephony communication setup request sent from a telephony device to an element of an IP telephony system includes a first encrypted code that is generated using one or more data items that are specific to the telephony device. The element of the IP telephony system receiving the setup request obtains the same data items locally and creates a second encrypted code. If the second code matches the first encrypted code, the telephony device and/or the setup request are authenticated, and the element of the IP telephony system proceeds to setup the requested telephony communication. | 04-30-2015 |
| 20150117443 | INTERNET SWITCH BOX, SYSTEM AND METHOD FOR INTERNET TELEPHONY - An Internet switch box connects between a telephone set and a public switched telephone network (PSTN) line, the latter of which is used both for PSTN telephone conversations and for connection to an Internet service provider (ISP). The switch box contains hardware and embedded software for establishing a connection to an ISP and for Internet telephony. When two users, each having an Internet switch box connected to the telephone set, wish to have an Internet telephony conversation, one calls the other over the PSTN. When they agree to an Internet telephony conversation, they signal their Internet switch boxes, by pressing either buttons on the switch boxes or certain keys on the telephone keypads, to switch to Internet telephony. The switch boxes disconnect the PSTN call and connect to their ISPs. Once the switch boxes are on the Internet, they contact each other through a server which supplies Internet protocol (IP) addresses of switch boxes, and the users continue their conversation by Internet telephony. The users can also prearrange to call each other solely by Internet telephony, in which case they do not need to talk to each other over the PSTN. | 04-30-2015 |
| 20150117444 | CALL HANDLING USING IP MULTIMEDIA SUBSYSTEM - A network device receives, via a user interface, a selection to call a contact's mobile device. The mobile device detects a feature, on the contact's mobile device, that supports presenting a priority call notification and initiates, via a first communication protocol, a pending call to the contact's mobile device. The mobile device presents, prior to the pending call being connected and based on the detecting the feature set, a user interface to solicit a priority indicator option. The mobile device receives, via the user interface, a selection of the priority indicator option and sends, via a second communication protocol, a priority indication flag for the pending call. | 04-30-2015 |
| 20150124802 | METHODS, SYSTEMS, AND COMPUTER READABLE MEDIA FOR INTELLIGENT OPTIMIZATION OF DIGITAL SIGNAL PROCESSOR (DSP) RESOURCE UTILIZATION IN A MEDIA GATEWAY - The subject matter described herein includes methods, systems, and computer readable media for intelligent optimization of digital signal processor (DSP) resource utilization in a media gateway. In one method, it is determined in a media gateway whether predetermined conditions exist for DSP-less IP-IP switching for a call. In response to determining that the predetermined conditions exist, DSP-less IP-IP switching is implemented for the call in the media gateway. After implementing the DSP-less IP-IP switching for the call, it is determined whether a predetermined event occurs that requires insertion of DSP resources during the call. In response to determining that the predetermined event occurs, the DSP resources are inserted into the call during the call. | 05-07-2015 |
| 20150124803 | METHOD AND APPARATUS FOR PROCESSING VoIP DATA - A system and method for processing Voice over Internet Protocol (VoIP) data including determining whether received audio data is VoIP data, transferring, when received audio data is VoIP data, the received VoIP data to a first path, and transferring, when received audio data is not VoIP data, the received audio data to a second path. The system and method can process, when received audio data is VoIP data, the received VoIP data via a VoIP data processing path including a voice engine for VoIP, instead of an audio data processing path, irrespective of types of mobile devices and types of applications. | 05-07-2015 |
| 20150124804 | SS7 ISUP to SIP Based Call Signaling Conversion Gateway for Wireless VoIP E911 - An SS7-based call protocol conversion gateway that translates between circuit-switched SS7 protocols and session initiation protocol (SIP) oriented protocol, allowing an E911 call initiated over a switched network to be routed by a VoIP network. The SS7-based call protocol conversion gateway provides a PSAP with MSAG quality (street address) information about a VoIP dual mode phone user without the need for a wireless carrier to invest in building out an entire VoIP core. Thus, wireless carriers may continue signaling the way they are today, i.e., using the J-STD-036 standard for CDMA and GSM in North America, yet see benefits of a VoIP network core, i.e., provision of MSAG quality location data to a PSAP. | 05-07-2015 |
| 20150131648 | Internet Protocol Communication Accessibility Improvement - Methods, devices, and systems for improving the accessibility of a target computing device configured to use IP communications software. In various embodiments, a server associated with a VOIP application may perform operations to determine the likelihood that the target computing device will be called via the application. The server may calculate the likelihood based on evaluations of past usage information, such as historical call logs, as well as activity information, such as location information and user interface inputs reported by caller computing devices. The server may further calculate a confidence as to whether the target computing device is accessible via the application. For example, the server may evaluate activity information to determine whether IP address and registration information is valid. When there is no confidence in accessibility, the server may transmit messages to the target computing device, such as push notifications using out-of-band transmissions with commands for refreshing a registration. | 05-14-2015 |
| 20150131649 | CONTENT DELIVERY SYSTEM - The audio outputs | 05-14-2015 |
| 20150131650 | Internet Protocol Communication Accessibility Improvement - Methods, devices, non-transitory processor-readable instructions, and systems for a VOIP application server associated with a VOIP application to improve performance of a target computing device for IP communications via the VOIP application. An embodiment method may include determining whether the target computing device is likely to be called using the VOIP application during a contact period, and directing the target computing device to adjust a performance setting for receiving an IP communication in response to determining a likelihood the device will be called during the contact period. When a call is likely, the performance setting may be raised via transmitting dummy traffic to target computing device, activating a quality-of-service on an Rx interface corresponding to the VOIP application and the target computing device, and/or transmitting a message directing the target computing device to utilize an aggressive slot cycle index setting or an aggressive discontinuous reception setting. | 05-14-2015 |
| 20150131651 | SYSTEM AND METHOD FOR CLIENT COMMUNICATION IN A DISTRIBUTED TELEPHONY NETWORK - A system and method for regional routing of internet protocol based real-time communication that includes registering a set of client application endpoint routes, comprising registering at least a first client gateway route of a first endpoint in a first region; receiving a communication invitation of the first endpoint; processing a set of communication instructions associated with the communication invitation and identifying a set of communication resources and at least a second endpoint; querying the client application endpoint routes and identifying a client gateway route of the second endpoint; and dynamically directing signaling path and media path of the communication according to the regional availability of the communication resources, the client gateway route of the first endpoint, and client gateway instance route of the second endpoint. | 05-14-2015 |
| 20150131652 | EXPEDITED RESOURCE NEGOTIATION IN SIP - A method of expediting resource negotiation in a modified Session Initiation Protocol (SIP) reduces the number of messages exchanged for resource negotiation, thereby reducing the latencies involved in session setup. The method entails sending an INVITE message having a modified SIP header containing an indication that the originator's terminal seeks a fast session setup. The INVITE message further contains a list of all codecs available at the originator's terminal and how many each type of media component are required. These codecs can be provided in an order of preference. The answerer selects the codecs for the requested media types from the list of available codecs without engaging in a back-and-forth resource negotiation for the codecs. The result is that the session can be set up with fewer messages which provides quicker session setup than in the prior art. | 05-14-2015 |
| 20150146715 | Communication System Architecture - Disclosed is a first call controller instance of a communication system configured to access a first failure-tolerant region of computer storage to access a call state, the first call controller instance being assigned to so access the call state responsive to a first instruction received via a network. At least part of the call state is replicated in a second failure-tolerant region of the computer storage so that a second call controller instance of the communication system can access the at least part of the call state, the second call controller instance being assigned to so access the at least part of the call state responsive to a second instruction received via the network. | 05-28-2015 |
| 20150146716 | Communication System Architecture - Disclosed is a communication system—for effecting communication events between a computer system, comprising first and second computer devices, and additional endpoint(s) connected via a communication network—comprising processing units, each having access to computer storage holding executable code modules for managing a communication event configured to implement a media modality controller configured to manage media modality of an established communication event and a call controller configured to establish the communication event. An instance of the media modality controller is assigned responsive to an instruction initiated to the media controller by the call controller to convey media modality control signals of the communication event to a media agent on the first device without accessing a call agent on the second device. The initiation of the instruction by the call controller is responsive to an instruction received via the network from the call agent on the second device. | 05-28-2015 |
| 20150295956 | SYSTEMS AND METHODS OF DISTRIBUTED SILO SIGNALING - The embodiments described herein recite a telephone communication system used for handling information such as messages, typically voice mail messages, and, more particularly, is directed to a system that provides distributed session initiation protocol (SIP) silos. Distributed SIP silos (DSS) is a Communications Application Platform (CAP) feature that maintains the site's call capacity even when a signaling server fails. DSS uses multiple non-redundant signaling servers to provide SIP signaling for the same set of media ports. Because there are multiple signaling servers providing signaling for the same set of ports, the failure of one signaling server only terminates the calls it was actively processing and once those calls have been cleaned up, all the available (non-suspended) ports in the configuration are available to the remaining signaling servers. | 10-15-2015 |
| 20150295979 | SYSTEMS AND METHODS OF DISTRIBUTED SILO SIGNALING - The embodiments described herein recite a telephone communication system used for handling information such as messages, typically voice mail messages, and, more particularly, is directed to a system that provides distributed session initiation protocol (SIP) silos. Distributed SIP silos (DSS) is a Communications Application Platform (CAP) feature that maintains the site's call capacity even when a signaling server fails. DSS uses multiple non-redundant signaling servers to provide sip signaling for the same set of media ports. Because there are multiple signaling servers providing signaling for the same set of ports, the failure of one signaling server only terminates the calls it was actively processing and once those calls have been cleaned up, all the available (non-suspended) ports in the configuration are available to the remaining signaling servers. | 10-15-2015 |
| 20150295981 | SYSTEMS AND METHODS OF DISTRIBUTED SILO SIGNALING - The embodiments described herein recite a telephone communication system used for handling information such as messages, typically voice mail messages, and, more particularly, is directed to a system that provides distributed session initiation protocol (SIP) silos. Distributed SIP silos (DSS) is a Communications Application Platform (CAP) feature that maintains the site's call capacity even when a signaling server fails. DSS uses multiple non-redundant signaling servers to provide sip signaling for the same set of media ports. Because there are multiple signaling servers providing signaling for the same set of ports, the failure of one signaling server only terminates the calls it was actively processing and once those calls have been cleaned up, all the available (non-suspended) ports in the configuration are available to the remaining signaling servers. | 10-15-2015 |
| 20150296087 | PRODUCT SERVICE SYSTEM AND METHOD - An appliance is configured to establish a communications link through the appliance between a user and a service representative as part of installing, diagnosing and servicing the appliance. | 10-15-2015 |
| 20150304170 | EXCHANGE AND USE OF GLOBALLY UNIQUE DEVICE IDENTIFIERS FOR CIRCUIT-SWITCHED AND PACKET SWITCHED INTEGRATION - According to one aspect, a system and method of exchanging GRUUs (Globally Routed User Agent URI (Uniform Resource Identifier)) between a first telephony-enabled device and a second telephony enabled device using a circuit-switched message is provided. Once exchanged, the telephony enabled devices can exchange SIP (session initiated protocol) communications routed by the GRUUs. Any one of the telephony-enabled devices can add a media component to the SIP communications. According to another aspect, a system and method of generating GRUUs is provided. According to another aspect, a system and method of handing off communications to a packet switched network from a circuit switched network is provided. | 10-22-2015 |
| 20150304276 | NETWORK ADDRESS TRANSLATION TRAVERSAL SYSTEM AND METHOD FOR REAL-TIME COMMUNICATIONS - A network address translation traversal system and method for real-time communications are provided. The network address translation traversal system includes a user terminal equipment, a network address translation device, a signaling control system and a peer terminal. The user terminal equipment is in a private network, while the signaling control system and the peer terminal are in a public network. The signaling control system is configured to instruct the network address translation device to create a network address translation mapping for a real-time communication connection. The peer terminal is configured to create the real-time communication connection with the user terminal equipment via the network address translation device directly according to the network address translation mapping. | 10-22-2015 |
| 20150304504 | ALTERNATE ROUTING OF COMMUNICATIONS IN A PACKET-BASED NETWORK - A method for performing alternate and therefore least cost routing in distributed H.323 Voice over IP (VoIP) networks is provided. With this method, the VoIP network consists of a hierarchy of gatekeeper (GK) functions to provide alternate routing, network element redundancy, and scalability. The alternate routing function is performed by a directory gatekeeper with route selection advancing from a first route to a second route by either of two conditions: (1) there are no resources available to terminate the call in the first zone; and (2) a lack of response to the directory GK request for such resources. | 10-22-2015 |
| 20150312357 | SYSTEMS AND METHODS FOR LOCATION MANAGEMENT AND EMERGENCY SUPPORT FOR A VOICE OVER INTERNET PROTOCOL DEVICE - An example apparatus includes a memory and a call session controller. The memory is to store a first public user identifier in association with a first nomadic service designator and a first operating mode designator, and to store a second public user identifier in association with a second nomadic service designator and a second operating mode designator. The call session controller is to determine whether a first call using the first public user identifier is eligible to be established based on the first nomadic service designator and the first operating mode designator, and determine whether a second call using the second public user identifier is eligible to be established based on the second nomadic service designator and based on the second operating mode designator. | 10-29-2015 |
| 20150319063 | DYNAMICALLY ASSOCIATING A DATACENTER WITH A NETWORK DEVICE - The present application details exemplary methods and systems for monitoring and analyzing network characteristics between the network device and a plurality of datacenters. The network device dynamically maps to the datacenter that associates with a superior available network connection. Further, the network device may dynamically map to different datacenters based on various network characteristics between the network device and available connections between each datacenter. | 11-05-2015 |
| 20150319196 | Method and System for Transfer of Call Control - According to an aspect of the present disclosure, a method is disclosed for a SIPIMR node to withdraw from a SIP session set-up loop by replying a redirect message to the SIP Invite sending node. The redirect message contains information for setting up alternative SIP session, said information being available in the SIP Invite receiving node. This information is usable by the SIPIMS node for establishing an alternative SIP session as well as information needed by other nodes and applications further on in the SIP session set-up loop. The SIPIMS node is specially adapted to retrieve the information from the redirect message for establishing a new SIP session based on and containing the information from the redirect message. Several implementations are given for use in an IP Multimedia Subsystem of a telecommunication network. | 11-05-2015 |
| 20150319212 | Media Controller - A data processing device comprising: a jitter buffer for receiving data packets; a media decoder configured to decode the data packets so as to form a stream of media frames, each frame comprising a plurality of samples; a media consumer having an input buffer for receiving the stream of media frames and being configured to play media frames from the input buffer according to a first frame rate; a buffer interface configured to monitor the input buffer so as to detect when the number of samples at the input buffer of the media consumer falls below a predetermined level and, in response, generate a play-out request; and a media controller configured to, responsive to each of the generated play-out requests, play-out one or more data packets to the media decoder so as to cause media frames of the stream to be delivered into the input buffer at a rate commensurate with the first frame rate. | 11-05-2015 |
| 20150319310 | Converged Voice Services - Novel tools and techniques for providing a subscriber with converged voice services, in which calls to one of the subscriber's telephone numbers is routed to some or all of the subscriber's telephone numbers (e.g., sequentially or simultaneously, depending on system configuration and/or subscriber preference). Some implementations employ a VoIP switch to handle call distribution among the subscriber's various phone lines, even if some (or all) of the subscribers telephone lines are not VoIP lines. In such implementations, upon receiving an incoming call directed to one of the subscriber's lines, a service switching point will route the call to a VoIP switch. Instructed by an application server, the VoIP switch will set up call legs (e.g., via the PSTN) to one or more of the subscribers' telephone numbers. Optionally, when one of the call legs is answered, the application server will instruct the VoIP switch to disconnect the other call legs. | 11-05-2015 |
| 20150319311 | Method, Communication System and Communication Terminal for the Transmission of Data - Disclosed is a method, a communication system, and a communication device for transmitting data to a first subscriber, within the framework of a connection signaling from a first primary service communication device of the first subscriber to a second primary service communication device, a primary address information message associated with the first primary service communication device and a secondary address information message associated with a first secondary service communication device of the first subscriber is transmitted to the second primary service communication device. The transmitted address information messages are identified and stored via the primary service communication device. For the transmission of data to be transmitted to the first subscriber, the stored secondary address information message is transferred from the second primary service communication device to a second secondary service communication device, and is transmitted based on the transferred secondary address information message during transmission to the first secondary service communication device. | 11-05-2015 |
| 20150326731 | Notification of Communication Events - A network element of a communication provider comprises transceiver apparatus arranged to receive a request message from an originating endpoint via a packet-based communication network; and processing apparatus configured to generate, in response to the request message from the originating endpoint, a push notification relating to a communication from the originating endpoint intended for a destination endpoint, the communication to be conducted over the packet-based network. The transceiver apparatus is arranged to send the push notification to the destination endpoint over the packet-based network. The processing apparatus is further configured to generate the push notification with a payload comprising an indication of an image representing an originating user, to be output by the destination endpoint in a user notification notifying a destination user regarding the communication. At least the indication of the image is determined and inserted into the payload of the push notification at the network element. | 11-12-2015 |
| 20150326732 | System and Method for Re-Routing Calls - The disclosed embodiments include a system, computer program product, and method for routing a call over a packet network. A call request may be received from a calling party to call a called party at a network address. At least one potential call path over a packet network may be determined to connect the calling party to the called party at the network address. Network performance information associated with each potential call path may be accessed and a determination may be made that each of the call paths are impaired or congested. In response to determining that each of the call paths are impaired or congested, the call may be routed over a call path other than one of the at least one potential call paths to enable the calling party to communicate with the called party. | 11-12-2015 |
| 20150326734 | SBC FOR CLOUD ENVIRONMENT AND METHOD FOR OPERATING SBC - A resource monitoring method comprises the steps of: generating process information including a process identifier, process user, process name, CPU usage, and IO usage; determining at least one first process having the same process user and process name from among a plurality of processes currently being executed; generating first group process information including a group process identifier, total CPU usage, and total IO usage of the determined at least one first process; and monitoring resources of the computing device in units of the generated first group process information. | 11-12-2015 |
| 20150341394 | METHOD AND SYSTEM TO ENHANCE PERFORMANCE OF A SESSION INITIATION PROTOCOL NETWORK AND ITS ELEMENTS - In accordance with at least one embodiment of the present invention, a communication apparatus includes a communication unit configured to send and receive messages on a network. Each message has a message header configured to include a plurality of header fields, where the message header includes a suppression header field that indicates a request to suppress at least one header field in at least one subsequently sent or received message. The communication apparatus is configured to establish a suppressed header field message exchange session with a peer on the network so that at least one subsequent message exchanged between the communication apparatus and the peer is free of at least one suppressible header field. | 11-26-2015 |
| 20150341395 | Method And System For Location-Based Communication - A method and system for location-based communication. A selection of a source transceiver from a plurality of available transceivers associated with a source may be received. A telephony session may be initiated with the selected source transceiver. A mobile target transceiver may be geographically located. A result of the geographic location regarding the located mobile target transceiver may be provided. The located mobile target transceiver may be connected to the telephony session. | 11-26-2015 |
| 20150350449 | SYSTEMS AND METHODS FOR CONNECTING TELEPHONY COMMUNICATIONS - Systems and methods performed by an IP telephony system are designed to determine when two parties to a recently terminated telephony communication are simultaneously calling each other in an attempt to re-establish a telephony communication. When the IP telephony system determines that this situation is occurring, the IP telephony system acts to connect the two parties, rather than have both of them see their new call setup attempt fail because the other party's telephony device is indicated to be busy. | 12-03-2015 |
| 20150358362 | DYNAMIC APPLICATION INTEGRATION ASSOCIATED WITH TELEPHONIC COMMUNICATIONS THROUGH HOSTED VoIP PBX USING CLIENT-SIDE INTEGRATION PROXY - A system for collecting information associated with a telephonic communication made through a VoIP system by dynamically integrating a plurality of end user software applications including a client side integration proxy in electronic communication with a hosted VoIP PBX. The client side integration proxy includes its own API for communicating and exchanging data with a plurality of end user software applications. Software executing on the client side integration proxy receives a request from the end user to retrieve information about a previous telephonic communication, retrieves data related to the previous telephonic communication from a data store, assigns a portion of a memory cache for storing the retrieved data about the previous telephonic communication, enables one or more of the plurality of end user software applications to access the data about the previous telephonic communication currently stored in the assigned portion of the memory cache, enables one or more of the plurality of end user software applications, in response to the data about the previous telephonic communication accessed from the assigned portion of the memory cache, to update, modify, or add to the data about the previous telephonic communication currently stored in the assigned portion of the memory cache based on data relevant to the previous telephonic communication obtainable or generated by the one or more of the plurality of end user software applications, and retrieves and presents the supplemented data about the previous telephonic communication to the end user. | 12-10-2015 |
| 20150373061 | Managing Voice over Internet Protocol (VoIP) Communications - The disclosed embodiments include a computer implemented method for managing network communications. In one embodiment, the method includes gathering, using performance information packet (PIP) data packets, network performance information from a communications network that includes network performance information from a set of egress points between the communications network and an outside network. The method selects a network connection including an egress point and an egress packet path within the communications network to the egress point offering the best quality of service between the communications network and an outside network based on the network performance information. The method then establishes the network connection between the communications network and the outside network for routing communications. | 12-24-2015 |
| 20150373194 | METHOD AND APPARATUS FOR NOTIFICATION OF MULTIPLE TELEPHONY DEVICES - The Internet Protocol telephony system comprises a registration unit; a signal handling interface; a rules engine; and a call processing interface. The registration unit is configured to store, in association with a customer record, a notification list of one or more telephony devices that that are to be notified upon receipt by the Internet Protocol telephony system of a communication addressed to a customer identifier corresponding to the customer record. The signal handling interface is configured to receive a signal originated by a telephony device. The rules engine is configured to consult the notification list and apply rules logic to determine in what order the one or more telephony devices on the notification list are to be notified. The call processing interface is configured to route notifications to the one or more telephony devices. | 12-24-2015 |
| 20150381666 | VOICE COMMUNICATION SYSTEM AND SERVICE WITHIN A MULTI-PROTOCOL NETWORK - A voice communication system ( | 12-31-2015 |
| 20150381815 | SYSTEM AND METHOD FOR COMMUNICATIONS IN A MULTI-PLATFORM ENVIRONMENT - A radio networking system includes at least a first communication port operable to facilitate communication between the radio networking system and a first communication platform. The system further includes at least a second communication port operable to facilitate communication between the radio networking system and a second communication platform. Additionally, a call control software module is operable to automatically and intelligently switch an incoming call from the first communication platform to the second communication platform which is controlled by a processor operable to execute the call control software module according to the information contained in a system configuration database. The call control software module further manages the configuration of call connections and conferencing services and supports real time user control of desired communication services. | 12-31-2015 |
| 20160006877 | VOICE MESSAGE TRANSMISSION AND PLAYBACK - A system and method is provided for receiving a voice message, selecting from multiple destinations, transmitting the voice message to a destination and subsequently playing back the voice message after it has been received by the destination. The system has a plurality of access devices which may be coupled to each other over a network such as the Internet. These access devices may include computers, workstations, and the like. Access devices may include a voice decoding device for converting a voice message received over a network into a voice signal for playback to a telephone device and may include a voice encoding device for converting a voice signal received into a voice message. | 01-07-2016 |
| 20160014164 | Mediation Of A Combined Asynchronous And Synchronous Communication Session | 01-14-2016 |
| 20160014165 | Mediation Of A Combined Asynchronous And Synchronous Communication Session | 01-14-2016 |
| 20160014232 | Voice Over Internet Protocol (VoIP) Mobility Detection | 01-14-2016 |
| 20160020890 | APPARATUS, SYSTEM AND METHOD OF USER-EQUIPMENT (UE) CENTRIC TRAFFIC ROUTING - Some demonstrative embodiments include devices, systems of User Equipment (UE) centric access network selection. For example, a node B may transmit to a User Equipment (UE) a cellular communication message over a cellular communication medium, the message including a value of a predefined parameter, which is based on a cellular network load of a cellular network. | 01-21-2016 |
| 20160021163 | SELECTIVE TRANSCODING - The disclosure is related to selective transcoding performed between a calling party and a called party. Such transcoding method may include receiving a call connection request message from a calling user equipment, determining whether transcoding is required for communication between the calling user equipment and a called user equipment based on the received call connection request message, and initiating the transcoding when the transcoding is determined as required. | 01-21-2016 |
| 20160021257 | CALL CONTROL METHOD BASED ON APPLICATION PRIORITY - A call control method is provided. The call control method includes broadcasting a message with a base station, wherein the message includes a plurality of check values respectively corresponding to different types of applications; receiving the message with a terminal device; selecting, with the terminal device, the check value corresponding to a specific application initiated with the terminal device; performing, with the terminal device, a persistence test to initiate a call to the base station for the specific application according to the selected check value; and requesting a MAC layer of the terminal device to initiate an access procedure of the call when the persistence test is passed. | 01-21-2016 |
| 20160028778 | CLASS 4 LONG DISTANCE SOFTSWITCH NETWORK WITH INTEGRATED CLASS 5 APPLICATION SERVICES - A telecommunication system including a class 4 long distance softswitch network with one or more a core routing engines and one or more class 5 application servers. The class 4 long distance softswitch network further includes at least one edge device, which may be in the form of a session border controller or media gateway, with at least one connection, including PRI, SS7 and TDM connections, to at least one customer premise equipment of at least one retail customer, which may be an enterprise customer. The class 5 application server is configured to provide the customer with class 5 services within the class 4 network. | 01-28-2016 |
| 20160036867 | METHOD AND APPARATUS FOR ENABLING REGISTRATION OF AGGREGATE END POINT DEVICES THROUGH PROVISIONING - A method and apparatus for enabling registration of an Aggregate End Point (AEP) device that is incapable of supporting a Session Initiation Protocol (SIP) based Internet Protocol Multimedia Subsystem (IMS) registration are disclosed. The method performs a static registration of the AEP device in a plurality of network elements associated with an Internet Protocol Multimedia Subsystem (IMS) network by provisioning. The method then processes an originating call request or a terminating call request associated with the AEP device using the static registration. | 02-04-2016 |
| 20160036868 | SYSTEMS AND METHODS FOR INGRESS CALL FILTERING - An ingress call filter system enables real-time or near real-time efficiencies of an inter-carrier switch. The ingress call filter determines if a received call to a called party is likely to fail. If the call is likely to fail, the ingress call filter returns an indication that the call should be filtered or rejected. If the call in not likely to fail, the ingress call filter returns an indication that a call router should attempt to establish the call. Such techniques mitigate penalties assessed to the inter-carrier network for incomplete or failed calls in real time or near real time. | 02-04-2016 |
| 20160036990 | SELF-HEALING INTER-CARRIER NETWORK SWITCH - A vendor evaluator system enables real-time or near real-time efficiencies of an inter-carrier switch. At a nominal periodic rate (e.g., every fifteen minutes or less), the vendor evaluator system updates a performance map of vendor exchanges to which calls may be routed from the switch. Specifically, in real-time or in near real-time, an updated performance score is generated for each vendor exchange and/or for each NPA-NXX code that is connected to the switch based on a weighted average of post dial delay, average call hold time, and attempt-seizure ratio. The score is compared against performance thresholds to determine whether or not the respective exchange and/or code should be included in a pool of acceptably-performing, candidate vendor exchanges from which a particular exchange is selected to service a call. Such techniques prevent poorly performing vendors from servicing calls routed by the inter-carrier network switch in real time or near real-time. | 02-04-2016 |
| 20160044052 | SYSTEM AND APPARATUS FOR ROGUE VOIP PHONE DETECTION AND MANAGING VOIP PHONE MOBILITY - A method and a system track network access information for authorized network devices. The access information facilitates tracking movement of the device throughout the network. In addition the access information can be used to detect when an unauthorized device attempts to access the network, posing as an authorized device | 02-11-2016 |
| 20160056976 | Integrating Communications - A method of accessing a first communication system of a first communication provider via a packet-based network, the first communication system maintaining a first list of contacts being users of the first communication system. The method comprises establishing a contact-sharing channel with a second communication system of a second, partner communication provider, wherein the second communication system is accessible via an independently executable web-browser and the packet-based network, and wherein the second communication system maintains second group of contacts being users of the second communication system. The contact-sharing channel is used to fetch contact information of one or more of the second contacts, so as to display at least part of the fetched contact information in the client application and establish a communication based on at least part of the fetched contact information. | 02-25-2016 |
| 20160057172 | METHOD AND APPARATUS FOR DISTRIBUTED COMPOSITIONAL CONTROL OF END-TO-END MEDIA IN IP NETWORKS - A method and an apparatus for performing a distributed control of end-to-end media on packet networks such as Voice over Internet Protocol and Service over Internet Protocol networks are disclosed. The method first receives a request from a first media endpoint device for opening at least one media channel to a second media endpoint device wherein said request contains a descriptor of said first media endpoint device. The method then updates one or more slot states and link states in response to said request and records the current state of each slot for supporting said media channel. The method also records the most recently received descriptor of said media endpoint device as a most recent descriptor for said slot supporting said media channel. The method executes one or more link objects in response to said request for controlling said at least one media channel. | 02-25-2016 |
| 20160057176 | CONTROLLING TELEPHONE CALL PROCESSING USING GLOBAL SIGNALING CODES - In general, embodiments of the present invention involve attaching (e.g., pre-fixing) a Global Signaling Code (GSC) to a called party's telephone number thereby creating a modified Uniform Resource Indicator (URI). This modified URI is then sent in the “TO:” header of a SIP INVITE. The GSC will typically include a geographic indicator corresponding to a geographic location of a caller and a treatment indicator corresponding to a desired treatment of the call. The call will be routed based on the geographic indicator and treated according to the treatment indicator. Illustrative treatments for the call include (among others) voice mail avoidance, a preferred compression scheme for the call, etc. | 02-25-2016 |
| 20160065624 | Method and Apparatus for Bidirectional Emulation of Telephonic Device Communication - The present systems and processes are directed to bridging telephone communications, such that a SIP communication server is able to communicate with the legacy PBX or key system interface for interworking connectivity of the devices. A bidirectional emulator module is provided permitting a communication session with legacy analog device and/or digital device. In certain embodiments, analog and digital telephones are registered with the emulator. An external telephone system such as a SIP communication server is also registered. A line dictionary contains the communication protocols of the analog and digital telephones and external telephone system. Outgoing line traffic and incoming line traffic are monitored in real-time, where the emulator module translates the user requests and line traffic according to the respective communication protocol data retrieved from the line dictionary. | 03-03-2016 |
| 20160065745 | METHOD AND APPARATUS FOR COMPLETING A CIRCUIT SWITCHED SERVICE CALL IN AN INTERNET PROTOCOL NETWORK - A method and an apparatus for completing a circuit switched service call in an Internet Protocol network are disclosed. For example, the method receives a session request and queries an tElephone NUmbering Mapping server for a called party, and determines if at least one Naming Authority Pointer resource record associated with the called party is received from the ENUM server. The method determines if a Session Initiation Protocol Universal Resource Identifier is returned, if the at least one NAPTR resource record associated with the called party is received, and processing the session request using the SIP URI, if the SIP URI is returned. The method determines if the processing of the session request using the SIP URI resulted in a successful call completion and processes the session request using a telephone URI, if the processing of the session request using the SIP URI resulted in an unsuccessful call completion. | 03-03-2016 |
| 20160065747 | A METHOD OF RESOLVING A PORTED TELEPHONE NUMBER INTO A NETWORK RESOURCE IDENTIFIER - A method of resolving by an original network/domain of a telephone number of a called party, belonging to a recipient network/domain. The method includes: a) a calling party, belonging to the original domain dispatches a message using the telephone number of the called party as routing identifier; b) a server of the original domain in charge of routing the message, or a DNS server, produces an interrogation key dependent on the telephone number and on a penultimate domain name; c) a DNS request is sent to a DNS server associated with the penultimate domain; d) the DNS server associated with the penultimate domain performs, by using the interrogation key, a search in a database ENUM associated therewith; e) the search provides at least one record containing a return network/domain including the recipient domain; and f) a DNS request is sent to the return domain. | 03-03-2016 |
| 20160072852 | Network Initiated CS Services During IMS Call - Methods, nodes, a terminal, a communication system and computer programs to be used in association with terminating a circuit switched signaling service to a terminal ( | 03-10-2016 |
| 20160072962 | ACR BUFFERING IN THE CLOUD - A network element of an Internet Protocol multimedia subsystem buffers network resource usage information in the cloud. After generating network resource usage information based on an observation of network resource usage, the network element transmits the network resource usage information to a cloud-based storage service for buffering. Once a network resource usage collection function is available, the network element retrieves the network resource usage information from the cloud-based storage service and transmits it to a charging collection function for generation of call detail records. | 03-10-2016 |
| 20160080220 | APPARATUS TO INDICATE TO A USER WHEN A VOIP COMMUNICATION SESSION IS ACTIVELY ESTABLISHED - A Voice over Internet Protocol (VoIP) detecting apparatus for detecting unintended VoIP traffic between a VoIP device and a VoIP network includes an interface configured to couple between the device and the VoIP network, a receiver configured to sense packets on the VoIP network, an indicator, and a processor coupled to a memory device. The processor is configured to receive a plurality of packets sensed by the receiver, determine a packet type for each of the plurality of packets sensed by the receiver, determine the presence of VoIP traffic based on the determined packet types, and activate the indicator based on the determination of the presence of VoIP traffic. | 03-17-2016 |
| 20160080288 | ROUTER FABRIC - A router fabric for switching real time broadcast video signals in a media processing network includes a logic device configured to route multiple channels of packetized video signals to another network device, a crossbar switch configured to be coupled to a plurality of input/output components and to switch video data of the multiple channels between the logic device and the plurality of input/output components in response to a control instruction, and a controller configured to map routing addresses for each video signal relative to the system clock, and to send the control instruction with the mapping to the crossbar switch and the logic device. | 03-17-2016 |
| 20160080429 | Routing of Sessions to Other Communication Networks - System, methods, nodes, and instruction set for routing a session invitation from a first network ( | 03-17-2016 |
| 20160080430 | VOICE TRANSMISSION TECHNOLOGY SELECTION - The present invention relates to a method of transmitting a signal by an MME (Mobility Management Entity) in a wireless network, the method comprising: receiving a tracking area update message from an UE (User Equipment); transmitting, on reception the tracking area update message, an Update Location message comprising an IMS voice over PS (Packet Switched) session supported indication to an HSS (Home Subscriber Server), wherein the IMS voice over PS session supported indication indicates whether or not an IMS Voice over PS Session is supported homogeneously in all tracking areas in the serving MME. | 03-17-2016 |
| 20160080435 | APPARATUS AND METHODS FOR ORIGINATION OF VOICE AND MESSAGING COMMUNICATION IN A NETWORK - A method that incorporates teachings of the subject disclosure may include, for example, receiving a query from a call session server for a first pointer associated with a telephone number of a terminating device of a requested communication session, transmitting to the call session server the first pointer including a session initiation protocol uniform resource identifier associated with the terminating device to initiate an internet protocol communication session, receiving a notification from the call session server responsive to the call session server failing to initiate the internet protocol communication session, and transmitting to the call session server a second pointer including a telephone protocol uniform resource identifier for originating a circuit-switched communication session responsive to receiving the notification. Other embodiments are disclosed. | 03-17-2016 |
| 20160088460 | CALL PROCESSING METHOD AND APPARATUS FOR USE IN LTE SYSTEM - A method for performing robust header compression (ROHC) of request for comments (RFC) 3095 over a packet data convergence protocol (PDCP) layer to compress and decompress voice over long term evolution (VoLTE) packets is provided. In the voice call operation, an evolved node B (eNB) receives an initialization and refresh (IR) packet transmitted by a user equipment (UE) and performs ROHC compression/decompression on the voice packet with a compressed header based on the information contained in the received IR packet. If the IR packet is lost, the eNB cannot establish the context for the voice packet, resulting in ROHC operation failure and a call drop problem. In order to maintain the voice call, the eNB sends the UE a static negative acknowledgement (SNACK) when a voice packet is received in the state where no context exists. | 03-24-2016 |
| 20160094437 | METHOD AND SYSTEMS FOR INTELLIGENT CALL ROUTING - Methods and systems for intelligent call routing are provided herein. In some embodiments, a method for intelligent call routing may include receiving a call request directed to a subscriber identifier associated with a plurality of devices, wherein the call request includes a caller identifier; determining one or more devices of the plurality of devices to which to route the call request based on a comparison of the caller identifier and address book information obtained from each of the plurality of devices; and routing the call to the one or more determined devices. | 03-31-2016 |
| 20160094482 | MINIMIZING NETWORK BANDWIDTH FOR VOICE SERVICES OVER TDM CES - A method and system are provided for reducing bandwidth usage in TDM CES systems conveying analog data, such as voice data. A transmitting router receiving TDM frames for packetization monitors the digitized analog data in the TDM frames. If the analog data has not changed beyond a configured threshold for a configured length of time, the transmitting router signals the receiving router at the far end of a TDM Pseudowire that no packets for the TDM Pseudowire will be sent. The transmitting router does not send any packets over the TDM Pseudowire, not even packets with empty payloads. The receiving router a receiving such a signal starts to generate its own packets for placing in its jitter buffer. Valid data already within the jitter buffer is played out to the access port, but once this runs out dummy packets placed in the jitter buffer by the receiving router are played out. In this way the jitter buffer maintains its fill level even when no packets are being sent across the TDM Pseudowire. The transmitting router continues to monitor the received digitized analog data, and only when it determines that the analog data has changed beyond the threshold does the transmitting router signal the receiving router, and begins sending packets once again. The method and system thereby reduce the bandwidth usage in TDM CES systems conveying analog data by refraining from needlessly sending packets over the TDM-Pseudowire when the analog data is not changing. | 03-31-2016 |
| 20160099979 | METHOD AND APPARATUS FOR RAPID SETUP OF A TELEPHONY COMMUNICATION USING MULTIPLE COMMUNICATION CHANNELS - A first telephony device sets up a first communication channel through an Internet protocol (IP) network for conducting an IP based telephony communication with a second telephony device. The first communication channel includes one or more media relays. The first telephony device then begins to conduct the telephony communication with the second telephony device over the first communication channel. While the initial stages of the telephony communication are ongoing, the first telephony device sets up a second communication channel with the second telephony device that does not utilize media relays. The telephony communication is then switched to the second communication channel. Proceeding in this fashion ensures that a communication channel can be rapidly established between the first and second telephony devices so that the telephony communication can quickly commence. | 04-07-2016 |
| 20160100062 | Continuation of vendor neutral VOIP interface devices and compatible portable phones - A Voice over Internet Protocol (VoIP) interface system for enabling communications between a plain old telephone service (POTS) device and a packet data network is disclosed. The VoIP interface system includes a POTS interface configured to communicate POTS voice telephone communications and a powered packet-based interface configured to communicate packet-based voice telephone communications with a host computer system that also provides power. The VoIP interface system also includes a memory system that includes a set of instructions, that direct the host system to prompt the user for a selection from a plurality of VoIP service providers, receive a selection identifying one of the plurality of VoIP service providers, identify profile information associated with the user and the selected one of the VoIP service providers, and populate one or more fields of a Session Initiation Protocol (SIP) module based on the profile information when executed. | 04-07-2016 |
| 20160105480 | SYSTEMS AND METHODS FOR HANDLING MULTIPLE CONCURRENT SESSION INITIATION PROTOCOL TRANSACTIONS - Systems and methods for handling the processing of multiple SIP transactions that have been requested at substantially the same time can involve establishing a priority order for processing the SIP transactions, and then individually processing the SIP transactions based on the established priority order. One or more SIP transactions having a lower priority can be held in a SIP processing queue of a software application until the processing of SIP transactions having a higher priority has been completed. Each time that the processing of a higher priority SIP transaction is completed, the next-highest priority SIP transaction in the queue is submitted for processing. Also, where possible, two or more SIP transactions in the queue may be consolidated into a single SIP transaction. | 04-14-2016 |
| 20160112466 | DISTRIBUTED CONNECTIVITY POLICY ENFORCEMENT WITH ICE - Instead of utilizing a centralized server or hardware (routers/gateways) to enforce connectivity policy restrictions, the policy connectivity restrictions for media session traffic are enforced by an endpoint that is involved in the media communication. Based on the policy requirements, the client enforces the policy restrictions by restricting the candidates that may be selected for the establishment of the media path. For example, the enforcement may result in the client selecting a path from available candidates that avoids congested Wide Area Network (WAN) links, avoiding a low bandwidth link, or possibly even failing the communication completely. The clients may also provide periodic updates to the policy server to allow tracking of the utilization of managed WAN links. | 04-21-2016 |
| 20160112576 | SYSTEMS AND METHODS OF MODIFYING DATA PACKETS USED IN IP TELEPHONY COMMUNICATIONS - Systems and methods performed by an IP telephony device or an element of an IP telephony system mask the data contained in data packets bearing the media of an IP telephony communication to prevent an Internet service provider from identifying the data packets as carrying the media of an IP telephony communication. The systems and methods can also modify the size of data packets and/or modify the data transfer rate of a stream of data packets bearing the media of an IP telephony communication to prevent an Internet service provider from identifying the stream of data packets as bearing the media of an IP telephony communication. | 04-21-2016 |
| 20160119233 | METHOD AND SYSTEMS FOR SEAMLESS MEDIA TRANSITION BETWEEN NETWORKS - Methods and systems for forwarding data packets containing media of an Internet protocol (IP) communication are provided herein. In some embodiments, a method for forwarding data packets containing media of an IP communication may include receiving IP communication setup signaling that includes an indication of a first originating IP address for data packets that are to be forwarded to a destination address; setting the first originating IP address as an authorized originating IP address; receiving a data packet from a second originating IP address directed to the destination address; and determining whether to forward the data packet from the second originating IP address to the destination address based on whether a first pre-defined time interval has elapsed since a last data packet was received from the first originating IP address. | 04-28-2016 |
| 20160127425 | REGISTRATION METHOD FOR MANAGING NAT SHUTDOWN - The present invention provides a registration method for managing NAT shutdown. In Internet communication field, a user must perform registrations intermittently to a server through NAT, and increase the time interval of registration step by step. But NAT itself will shutdown if no packet is passed through for a long period. The registration method of the present invention is to adjust the time interval of registration step by step so that the time interval of registration is slightly less than the shutdown time of NAT, and then fix the time interval of registration to assure that all of the Invite packet can pass through without blocking up by the shutdown of NAT. | 05-05-2016 |
| 20160134527 | SYSTEM AND METHOD FOR VIRTUAL NETWORK-BASED DISTRIBUTED MULTI-DOMAIN ROUTING CONTROL - A system for distributed multi-domain routing control includes two or more software-defined network (SDN) controllers and virtual routers. The two or more SDN controllers create virtual topology information by collecting information about switches under the control thereof, exchange network information about a different virtual network, as well as create a routing path from a source terminal to a destination cross switch or a routing path from the destination cross switch to a destination terminal based on location identification information of the destination terminal and/or the virtual topology information. Then, the virtual routers manage communication between the different virtual tenant networks or communication between each virtual tenant network and an external network. | 05-12-2016 |
| 20160134663 | SYSTEM AND METHOD FOR PROVIDING ENTERPRISE VOICE CALL CONTINUITY - An improved system and method are disclosed for providing voice call continuity in an enterprise network. For example, an enterprise public branch exchange (PBX) may be configured with a pilot number that is used to provide VCC services when called by a client. Digit collection via DMTF signaling or other means may be used to collect destination information from the client. The enterprise network may use the collected digits to establish a communication session with another device that corresponds to the destination information. | 05-12-2016 |
| 20160134665 | TELEPHONY APPLICATION PLATFORM - A hosted private branch exchange (PBX) platform includes associated application programming interfaces (APIs) that provide a range of integration points with the PBX platform that, in turn, enables the development of a broad range of applications that can customize and/or enhance the basic functionality of the underlying PBX platform. | 05-12-2016 |
| 20160142292 | Systems and Methods for Software Configurable Air Interface Adaptation - A base station may update a SoftAI profile to obtain an updated SoftAI profile specifying a new air interface configuration that was unknown to the base station prior to updating the SoftAI profile. The base station may receive SoftAI configuration information from a network controller, and update the SoftAI profile based on the SoftAI configuration information. The updated SoftAI profile may define a new combination of physical layer parameters, a new waveform, a new modulation coding scheme (MCS), or any other AI configuration parameter, or collection of AI configuration parameters. The SoftAI configuration information, or a separate network instruction, may also specify one or more conditions for using the new air interface configuration to communicate traffic over a wireless link between the base station and a wireless device. | 05-19-2016 |
| 20160142299 | ROUTING METHOD REDUCING THE CONSEQUENCES OF MICROLOOPS - The invention concerns a method for routing a packet in a packet switching network in disconnected routing mode, implemented by a router, comprising the following steps:—receiving (E | 05-19-2016 |
| 20160142446 | SYSTEM AND METHOD FOR INTEGRATING SESSION INITIATION PROTOCOL COMMUNICATION IN A TELECOMMUNICATIONS PLATFORM - A system and method for facilitating signaling and media communication at a communication platform that includes receiving a communication request to a resource, wherein the communication request specifies a destination endpoint; establishing signaling and media communication in a session with the destination endpoint of the communication request; registering a callback resource to a signaling event of the session; monitoring signaling messages of the session; detecting the signaling event in the signaling messages of the session; and triggering the callback resource upon detecting the signaling event. | 05-19-2016 |
| 20160142550 | SYSTEMS AND METHODS FOR CALL PROCESSING - The present invention provides flexible, user-definable call screening processes. The user can optionally define to which telecommunication terminals a screened call is to be broadcast to and under what conditions. An incoming call is forwarded to a call management system that asks the caller to leave a voice message. The call management system selectively couples the call to a POTS line or a VoIP-capable device so that the user can listen to the incoming message and thereby screen the incoming call. Based on the screening, the user can instruct the call management system to connect the caller to the user. | 05-19-2016 |
| 20160150089 | USER CONTROLLED CALL MANAGEMENT - A system and methods for call facilitation are provided. The system includes interfaces to call handling networks for receiving or transmitting information by voice, data, email or internet protocol, a storage means and a processing means. The storage means stores caller and/or callee associated information. And the processing means initiates and controls calls to one or more of the call handling networks utilizing associated information corresponding to the caller and the callee of the calls. The associated information includes identification information, calling rules, authentication information, and electronic addresses for each caller and/or callee. In addition, a method for facilitation of a call between a caller and a callee includes setting-up and/or controlling the call in response to information in one or more URL links accessed by the caller and/or the callee via their electronic addresses and/or online accounts. | 05-26-2016 |
| 20160150090 | Exchange And Use Of Globally Unique Device Identifiers For Circuit-Switched And Packet Switched - According to one aspect, a system and method of exchanging GRUUs (Globally Routed User Agent URI (Uniform Resource Identifier)) between a first telephony-enabled device and a second telephony enabled device using a circuit-switched message is provided. Once exchanged, the telephony enabled devices can exchange SIP (session initiated protocol) communications routed by the GRUUs. Any one of the telephony-enabled devices can add a media component to the SIP communications. According to another aspect, a system and method of generating GRUUs is provided. According to another aspect, a system and method of handing off communications to a packet switched network from a circuit switched network is provided. | 05-26-2016 |
| 20160156677 | Notification of Communication Events | 06-02-2016 |
| 20160156678 | METHOD, AND RELATED APPARATUS FOR RECOVERING CALLED SERVICE OF TERMINAL | 06-02-2016 |
| 20160156784 | DYNAMIC TELEPHONE NUMBER ALLOCATION MANAGEMENT | 06-02-2016 |
| 20160157288 | ENABLING COMBINATIONAL SERVICES IN A COMMUNICATIONS NETWORK | 06-02-2016 |
| 20160165043 | NETWORK-EXTENSIBLE AND CONTROLLABLE TELEPHONE - A system that includes a telephone adapted for connection to a data communications network. The telephone is capable of discovering other devices connected to the data communications network and using those devices to extend its own functionality. The telephone is also accessible and controllable by other devices on the data communications network. | 06-09-2016 |
| 20160165057 | CALLER-CALLEE ASSOCIATION OF A PLURALITY OF NETWORKED DEVICES - The present disclosure generally relates to systems and methods for establishing and maintaining communication between two or more communication devices coupled to communication networks. Some specific aspects relate to communication between a plurality of communication devices each of which is coupled to a respective network. Other aspects relate to establishing such communication by way of contact lists maintained and facilitated on systems coupled to the networks. Users of multiple communication networks, such as VoIP, PSTN and wireless, employ multiple communication devices to communicate with their contacts. For example, a VoIP enabled computer is necessary to access contacts on a VoIP network and a mobile or cellular telephone is used to access contacts on wireless and PSTN networks. A contact list, stored on one communication device, in some instances, cannot be accessed from another communication device. For example, a contact list stored in a VoIP enabled computer cannot be accessed from PSTN or wireless phone devices. Various embodiments described herein provide a convenient solution that can integrate contacts stored on different communication devices and make them accessible from a single device. | 06-09-2016 |
| 20160165067 | Application and platform to build enhanced data repositories for facilitating a merchant/service provider electronic exchange - A platform may include resources configured to receive information representing a phone call from a caller (e.g., a traveler) to a callee (e.g., a proprietor) and instead of immediately connecting the phone call may communicate a message to the caller telling the caller the call is being connected. Prior to connecting the phone call with the callee, the platform may connect with a proprietor device (e.g., a landline telephone or a smartphone) and communicate a message to the callee that an incoming phone call is from a potential customer and is being provided courtesy of a merchant electronic exchange. The message may further encourage the callee to join the merchant electronic exchange as a member and/or inform the callee of the number of leads it has provided to the callee. After communicating the message to the proprietor, the platform may connect the call between the caller and the callee. | 06-09-2016 |
| 20160173536 | SYSTEM AND METHOD FOR PROCESSING MEDIA REQUESTS DURING TELEPHONY SESSIONS | 06-16-2016 |
| 20160182256 | PIPELINED HYBRID PACKET/CIRCUIT-SWITCHED NETWORK-ON-CHIP | 06-23-2016 |
| 20160182365 | ROUTING MESSAGE DELIVERY METHOD APPLICABLE TO NETOWRK NODE AND NETWORK NODE USING THE SAME AND COMMUNICATION NETWORK USING THE SAME | 06-23-2016 |
| 20160182731 | Base Phone and Additional Phone Implementation, Answering, Calling, and Intercom Method, and IP Terminal | 06-23-2016 |
| 20160182732 | SYSTEMS AND METHODS FOR SETTING UP INTERNET PROTOCOL COMMUNICATIONS | 06-23-2016 |
| 20160191327 | Method, Device, System for Detecting Data Link, Controller and Gateway - Disclosed are a method, device, system for detecting a data link, controller, and gateway. The method comprises: an SDN controller sends a GTP request message to a first UGW, and instructs the first UGW to send the GTP request message to another GTP endpoint in a GTP user plane signaling format; the SDN controller receives a GTP response message from the first UGW, and detects a data link between the first UGW and the another GTP endpoint according to the GTP response message, the GTP response message being corresponding to the GTP request message. The disclosure solves the problem in the prior art that logic of a user plane and logic of a control plane are unclear during detection of a data link between GTP endpoints, thereby improving the clarity of the logic of the user plane and the clarity of the logic of the control plane. | 06-30-2016 |
| 20160191698 | METHODS AND APPARATUS FOR PROVISIONING AND USING IP CLIENTS WHICH CAN BE ARRANGED ACCORDING TO GROUPS AND SHARE A COMMON TELEPHONE NUMBER - Methods and apparatus for allowing users in a group, e.g., family members, to share a telephone number while using different devices, e.g., IP devices, are described. The methods and apparatus allow for users in a group to obtain individual presence information for members of the group by making a single request for presence information corresponding to members of the group. In some embodiments, in response to the single request, the requesting device will be supplied with individual presence information for each member of the group to which the request corresponds. The presence information may be received in individual communications, e.g., one individual message providing presence information on a user of the group. Alternatively presence information for the individual members of the group may be included in a single message that is sent to the requesting device. | 06-30-2016 |
| 20160191713 | VOIP Analog Telephone System - A multi-port VoIP telecommunications system that allows the user to gain access to telephone connectivity through the Internet by connecting directly to the Internet or by connecting to the Internet through the existing Internet connection of a computer or cell phone device. The present system includes an Ethernet port, a Wi-Fi receiver to facilitate the transmission and receipt of Internet protocol signals wirelessly, a USB plug connectable to the ATA, connectivity to a home monitoring network and connectivity to Bluetooth devices. | 06-30-2016 |
| 20160198380 | SIP HEADER TO INDICATE MOBILITY TRANSFER OPERATION | 07-07-2016 |
| 20160205145 | PEER-TO-PEER INTERNET PROTOCOL TELEPHONE SYSTEM WITH SYSTEM-WIDE CONFIGURATION DATA | 07-14-2016 |
| 20160205212 | Caching Of Announcements At The Edge Of A Packet Switched Telecommunication Network | 07-14-2016 |
| 20160205259 | COMMUNICATION SYSTEM AND METHOD FOR MOBILE DEVICES IN THE ABSENCE OF CELLULAR COVERAGE | 07-14-2016 |
| 20160205260 | Methods and Apparatus to Provide Extended Voice Over Internet Protocol (VoIP) Services | 07-14-2016 |
| 20160381218 | SYSTEMS AND METHODS FOR IP AND VOIP DEVICE LOCATION DETERMINATION - A method and system for precise position determination of general Internet Protocol (IP) network-connected devices. A method enables use of remote intelligence located at strategic network points to distribute relevant assistance data to IP devices with embedded receivers. Assistance is tailored to provide physical timing, frequency and real time signal status data using general broadband communication protocols. Relevant assistance data enables several complementary forms of signal processing gain critical to acquire and measure weakened or distorted in-building Global Navigation Satellite Services (GNSS) signals and to ultimately extract corresponding pseudo-range time components. A method to assemble sets of GNSS measurements that are observed over long periods of time while using standard satellite navigation methods, and once compiled, convert using standard methods each pseudo-range into usable path distances used to calculate a precise geographic position to a known degree of accuracy. | 12-29-2016 |
| 20160381230 | EXCHANGE AND USE OF GLOBALLY UNIQUE DEVICE IDENTIFIERS FOR CIRCUIT-SWITCHED AND PACKET SWITCHED INTEGRATION - According to one aspect, a system and method of exchanging GRUUs (Globally Routed User Agent URI (Uniform Resource Identifier)) between a first telephony-enabled device and a second telephony enabled device using a circuit-switched message is provided. Once exchanged, the telephony enabled devices can exchange SIP (session initiated protocol) communications routed by the GRUUs. Any one of the telephony-enabled devices can add a media component to the SIP communications. According to another aspect, a system and method of generating GRUUs is provided. According to another aspect, a system and method of handing off communications to a packet switched network from a circuit switched network is provided. | 12-29-2016 |
| 20180026711 | COMMUNICATION SYSTEM | 01-25-2018 |
| 20180026811 | EFFICIENT TRANSPORT OF ENCAPSULATED MEDIA TRAFFIC OVER RESTRICTIVE NETWORKS | 01-25-2018 |
| 20190149582 | SYSTEM AND METHOD FOR PROCESSING TELEPHONY SESSIONS | 05-16-2019 |
