| 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 tha |
