| Class / Patent application number | Description | Number of patent applications / Date published |
|---|
| 370263000 | Conferee signals combined or distributed via time channels | 12 |
| 20100157852 | Systems and methods for multiple mode voice and data communications using intelligenty bridged TDM and packet buses and methods for performing telephony and data functions using the same - Systems and methods by which voice/data communications may occur in multiple modes/protocols are disclosed. In particular, systems and methods are provided for multiple native mode/protocol voice and data transmissions and receptions with a computing system having a multi-bus structure, including, for example, a TDM bus and a packet bus, and multi-protocol framing engines. Such systems preferably include subsystem functions such as PBX, voice mail and other telephony functions, LAN hub and data router. In preferred embodiments, a TDM bus and a packet bus are intelligently bridged and managed, thereby enabling such multiple mode/protocol voice and data transmissions to be intelligently managed and controlled with a single, integrated system. A computer or other processor includes a local area network controller, which provides routing and hub(s) for one or more packet networks. The computer also is coupled to a buffer/framer, which serves to frame/deframe data to/from the computer from TDM bus. The buffer/framer includes a plurality of framer/deframer engines, supporting, for example, ATM and HDLC framing/deframing. The buffer/framer is coupled to the TDM bus by way of a switch/multiplexer, which includes the capability to intelligently map data traffic between the buffer/framer and the TDM bus to various slots of the TDM frames. Preferably, a DSP pool is coupled to buffer/framer in a manner to provide various signal processing and telecommunications support, such as dial tone generation, DTMF detection and the like. The TDM bus is coupled to a various line/station cards, serving to interface the TDM bus with telephone, facsimiles and other telecommunication devices, and also with a various digital and/or analog WAN network services. | 06-24-2010 |
| 20110141951 | METHODS AND APPARATUS FOR TIMESLOT TELECONFERENCING - Methods and apparatus are provided for a timeslot teleconference. A meeting invite is created for a timeslot teleconference on a computer. Timeslots are created for the timeslot teleconference in the meeting invite on the computer. Participants are designated for the timeslots in the meeting invite on the computer, and the timeslots are different time periods during the timeslot teleconference, which can have variable start and end times, as determined by a host. Reach numbers are added for each of the participants in the meeting invite on the computer, and the reach numbers are contact numbers to call the participants for their respective timeslots. The meeting invite is transmitted to participants on the computer. To participate in the timeslot teleconference, the timeslot teleconference is configured for the reach numbers to be called for the participants to correspond to their respective timeslots and/or the participants to call in for their respective timeslots. | 06-16-2011 |
| 20130034026 | VOICE BAND SWITCHING UNIT - A method of dynamically switching voice channels between sampling rates in a telecommunication system including a plurality of conferenced channels that have a telephony side, a network side, and a conferencing unit therebetween. During conferencing the presence of a switching stimuli is detected, and at least a first affected voice channel affected by the switching stimuli is identified from the plurality of conferenced voice channels. Responsive to the switching stimuli, based on a telephony side sampling rate of the first affected voice channel, a network side sampling rate of the first affected voice channel, and respective sampling rates of other conferenced voice channels, it is determined whether or not to switch a sampling rate for at least the first affected voice channel using a switching criteria. Provided the determining decides to switch, the sampling rate is switched for at least the first affected voice channel. | 02-07-2013 |
| 20130329610 | Combined In-Ear Speaker and Microphone for Radio Communication - A multivoice radio system allows users to speak and listen, at the same time, to others using multivoice radios without using a base station. Several embodiments of the present invention are directed toward combining a multivoice radio with push-to-talk (PTT), long-range radio. Other embodiments combine two or more multivoice systems and/or use an extender to increase a distance that multivoice radios can communicate with each other. Combining multivoice systems, in some embodiments, creates a voice and/or data mesh network. Switch-to-talk (STT) functionality can be added to systems to increase a number of users and promote radio discipline. Additionally, a combined in-ear microphone and speaker is disclosed. Further embodiments are directed to combining duplex radios, such as mobile phones, to multivoice systems, and/or PTT systems. | 12-12-2013 |
| 20140247756 | Method, Device, and Conference System for Processing Media Data Packet - A method, a device, and a conference system for processing a media data packet are provided. The method includes: establishing with a conference proxy server (CPS), a virtual public participant (VPP) channel corresponding to a conference identity; when there are media data packets with same content in a media data packet corresponding to each conference member identity, acquiring a media data packet with a highest frequency of same content among the media data packets as a public media data packet, and forwarding the public media data packet to the CPS through the VPP channel; acquiring a conference member identity that currently does not multiplex the VPP channel, generating a real-time transport protocol control protocol (RTCP) multiplexing start command corresponding to the conference member identity, and delivering the RTCP multiplexing start command to the CPS, so that the CPS starts to forward the public media data packet to a user equipment corresponding to the conference member identity. | 09-04-2014 |
| 20190150113 | CONFERENCE AUDIO MANAGEMENT | 05-16-2019 |
| 370265000 | Particular technique for combining diverse information types | 3 |
| 20080212499 | WEB AND MULTI-MEDIA CONFERENCE - Methods, systems, and machine-readable media are disclosed for providing multi-media conferencing. According to one embodiment, a method for providing a multi-media conference can comprise receiving a request to participate in the conference from a first client via a signaling protocol. For example, the signaling protocol comprises Session Initiation Protocol (SIP) or another signaling protocol. The first client and a media server adapted to support the conference can be invited to join the conference via third-party call control of the signaling protocol. The method can also include assigning a conference identifier to the conference. In such a case, inviting the first client and the media server can comprise sending invitation messages to each of the first client and the media server wherein the invitation messages include the conference identifier. | 09-04-2008 |
| 20080239998 | SIP COMMUNICATION SYSTEM, CALL CONTROL SERVER AND SIP COMMUNICATION METHOD - There is provided a SIP communication system or the like which, by performing SIP communication between a terminal and a call control server, performs position registration or calling control of the terminal and provides one or multiple supplementary services for an incoming/outgoing call in accordance with subscriber information managed by a home subscriber server, wherein the call control server is provided with supplementary service function sections for providing the multiple supplementary services, respectively, and a service management section for causing the services of the multiple supplementary service function sections to be distributed and executed while managing conflicts. | 10-02-2008 |
| 20150078210 | Method, Device, and System for Implementing Multi-Party Conference Call by Using Set Top Box - A method, device, and system for implementing a multi-party conference call by using a set-top box. The method includes: in a conferencing state, the set-top box sending a network server that manages the conference a request message for adding another specified set-top box into the conference where the set-top box is located; and after the set-top box receives an acknowledgment message fed back by the network server for indicating that the specified set-top box has joined in the conference, distributing audio and/or video signals to be transmitted by the set-top box to other members of the conference through the network server. | 03-19-2015 |
| 370266000 | Using summation of conferee signals | 3 |
| 370267000 | Digital summation | 2 |
| 20080304429 | Method of transmitting data in a communication system - A method of transmitting a combined audio signal to at least one of a plurality of participants in a communication event comprising; receiving an audio signal from each of said plurality of participants together with audio activity information associated with each of said received audio signals; analysing a measure of audio activity for each received audio signal based on the audio activity information associated with each of said received signals, wherein said measure of audio activity allows audio signals comprising audio activity to be compared; selecting a set of audio signals from said received audio signals based on the analysed measure of audio activity for each signal; decoding said set of audio signals; and combining said set of audio signals to generate said combined audio signal to be transmitted to said at least one of said plurality of participants. | 12-11-2008 |
| 20100054160 | AUXILIARY ANTENNA SYSTEMS AND METHODS FOR WIRELESS COMMUNICATIONS SYSTEM BASE STATIONS - Auxiliary antenna systems can be used to increase a link margin of a wireless communications system that includes a base station. The auxiliary antenna system is positioned within a service region of the base station and spaced apart from the base station. Related systems, methods and components are described. | 03-04-2010 |
| 370269000 | Including cancellation of certain signals | 1 |
| 20080310328 | CLIENT-SIDE ECHO CANCELLATION FOR MULTI-PARTY AUDIO CONFERENCING - A “Client-Side Echo Canceller” provides a unique system and method for reducing Multipoint Control Unit (MCU) computational overhead in a multi-point audio conference. In general, the local audio input signal of each client is transmitted in real-time to the MCU. The MCU then combines the audio input signals of all clients to create a single composite signal that is transmitted back to all clients in real-time. Each client then locally processes the composite signal to remove each client's local contribution to the composite signal prior to local playback in order to eliminate a local echo of each client's local audio input. In various embodiments, local cancellation of the local audio input from the composite signal is performed on either a time domain or a transform domain representation of the composite signal. Further, since each client receives the same signal, MCU transmission bandwidth can be reduced via multicast transmissions. | 12-18-2008 |
