Abstract: A method for supporting communications involves establishing a communications conference that includes multiple conference participants that each provide an audio stream. Audio data packets in the audio streams are identified, and audio information from selected audio data packets is mixed to generate a conference audio stream that is communicated to the conference participants. Enhanced media packets in at least one of the audio streams are identified, and the enhanced media packets are communicated to selected conference participants.

Abstract: A wireless communications device is a hub for a conference call between a user of the device, a first remote party, and a second remote party. The wireless communications device communicates with the first remote party over a first connection and with the second remote party over a second concurrently active connection. A circuit in the wireless communications device transcodes voice data received over the first and second connections so that the voice data may be shared between the parties.

Abstract: A network system integrated with SIP call server and SIP agent client. The network system is provided between a local device and a network for connecting with at least one remote device. The network system includes: at least one local connecting port for coupling with the local device; a remote connecting port for coupling with the network; and an SIP processing module for executing at least one SIP call server program and SIP agent client program. After the local and remote devices perform SIP registry on the SIP call server program, the local and remote devices can telecommunicate with each other by voice from a private SIP telecommunication network. The network system further includes an IP phone connecting port for coupling with at least one IP phone and a voice processing module for converting the voice signal of the IP phone into digital signal and converting the digital signal of the SIP processing module into voice signal.

Abstract: Systems and methods relate to setting-up a future telephone conference between a host party and at least one participant party. The host party dials into a system and enters the meeting date/time and meeting parties' telephone numbers. At the meeting time, the system retrieves the information from its database and dials the meeting participants.

Abstract: A peer to peer audio communication screen in system with adaptive bandwidth control. The peer-to-peer system can be used, for example, in classroom applications. More interactive peers are moved to a position where they will receive lower latency from the peer-to-peer system. This may be done by maintaining a score indicative of the amount of interactivity, and then using that score to move a peer either close to the source of audio packets, e.g., the core or further from that source.

Abstract: There is provided a method of a first mobile communication device for supporting a combinational circuit switched and packet switched session with other communication devices. During a combination session (100) between first and second mobile communication devices (110, 120), the first mobile communication device may forward or duplicate data to a third mobile communication device (210) without intervention from a remote server. There is also provided a method for supporting supplementary services in a combinational circuit switched and packet switched session between three or more mobile communication devices (110, 120, 210). Data is communicated between first and second mobile communication devices (110, 120) via a packet switched data session (140) while maintaining a circuit switched call session (130) between the first and second mobile communication devices.

Abstract: Video, voice and data are integrated using existing telephone connections by using a video network server to receive and transmit video signals to and from clients using telephone wires. Client frequency division multiplexing couplers, each associated with one of the clients, receive video and telephone signals from the client and frequency multiplex them onto the telephone wires for transmission. They also receive video signals from the server and frequency demultiplex them for transmission to the client. A video frequency division multiplexing coupler connected between the video network server and each of the client frequency division multiplexing couplers receive the video and telephone signals from the clients and transmit only the video signals to the video network server.

Abstract: 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.

Abstract: A communication unit comprises a command signal generator for generating a command signal. A transmitter transmits communication signals at one of two frequencies and for transmitting the command signal. A receiver receives communication signals at one of the two frequencies and for receiving the command signal. Upon initiation of the command signal, the command signal is transmitted for receipt by all active units in the system and the unit transmitting the command signal receives communication signals at a selected frequency of the two frequencies and transmits communication signals at another frequency. Upon reception of the command signal from another unit in the system, the unit receiving the command signal transmits a communication signal at the selected frequency of the two frequencies and receives communication signals at the another frequency.

Abstract: The present invention discloses a method for connecting one or more systems to a user for automatic determination of which system the user may use to participate in a call. A predefined list of video conferencing systems and/or locations for each user is predefined and arranged in a prioritized order. When scheduling a meeting and/or a video conference, the predefined user lists are taken into account when selecting end-points and other resources involved in the meeting/conference. The selection process may be controlled by predefined rules also taking various system availability, network resources and the routes required for connecting the other systems in the conference into account The present invention also introduces presence applications to facilitate scheduling and set up of ad hoc meetings/conferences.

Abstract: Assigning inbound ringing ISDN channels to a particular endpoint's call uses a framing listening technique to distinguish between H.221 framing, master bonding channel framing and slave bonding channel framing. If the channel is receiving Master Bonding Channel framing, the MCU detects that this is a new call from a unique far endpoint and may then begin a process to address this new call. If the channel is receiving Slave Bonding Channel framing, the MCU then transmits a multi-frame pattern and looks for a Physical Video Unit Identifier the MCU sent to this far end via the Master Channel upon initial negotiation. If the channel is receiving H.221 framing, a H.243 terminal address assigned using a Terminal Indicate Assignment (TIA) message in the initial channel is used. The far endpoint may then send this address in its additional channels through the use of a Terminal Indicate Additional Channel X (TIX) message.

Abstract: A method for sending IP packets from a base station to a mobile station during transition of the mobile station from a dormant state to an active state is provided. The method includes receiving a lagging feature request from the mobile station. A physical layer of a traffic channel is established between the base station and the mobile station. A network layer of the traffic channel is established between the base station and the mobile station. After the physical layer of the traffic channel is established and while the network layer of the traffic channel is being established, the IP packets are packaged and sent to the mobile station in a particular format.

Abstract: In a method for determining at least one user terminal for a telecommunication conference session with a multiplicity of telecommunication conference session user terminals, wherein a predefined telecommunication conference session participation condition is allocated to the at least one telecommunication conference session user terminal, the predefined telecommunication conference session participation condition may be checked by involving two telecommunication conference session servers. A first telecommunication conference session server receives an enquiry for determining at least one user terminal for a telecommunication conference session and a second telecommunication conference session server determines condition-related information for determining the user terminal which meets the telecommunication conference session participation condition.

Abstract: A method for minimizing the number of transcoders needed to support group calls includes a first step of receiving a request from a call initiator to initiate a group call with a plurality of call targets. Next, a listing of call initiator vocoders supported by the call initiator is received. Then, a call invitation is sent to each of a plurality of networks associated with the call targets. Network responses are received from each of the plurality of networks. The network responses comprise one or more network supported vocoders. The responses from all of the plurality of networks are collected. Then, the amount of transcoder needed to support the group call is optimized by minimizing the number of call initiator vocoders and network supported vocoder pairs.

Abstract: At least one condition concerning a first user (102) within a communication network is obtained. Presence information of a second user (114) within the communication network is determined based upon the at least one condition of the first user (102). The presence information of the second user is then projected to the first user (102).

Abstract: In a method for the conditional setting-up of a telecommunication conference session with a multiplicity of telecommunication conference session user terminals, wherein a group of user terminals having at least one telecommunication conference session user terminal is identified by means of a telecommunication conference session user group identification, a predefined telecommunication conference session set-up condition is checked by involving two telecommunication conference session servers. A first telecommunication conference session server receives an enquiry for the conditional setting-up of a telecommunication conference session and a second telecommunication conference session server determines condition-related information for checking the predefined telecommunication conference session set-up condition.

Abstract: The present invention provides methods of inter-technology handoff of a multi-mode mobile unit in a packet-switched communication system. One embodiment of the method includes establishing a call from a first application layer client in the first mobile unit using a first wireless communication link that operates according to a first wireless access technology. The method also includes sending, in response to detecting changing conditions of the first wireless communication link, a request to form a conference call including the first application layer client and inviting a second application layer client in the first mobile unit to join the conference call. The second application layer client uses a second wireless communication link that operates according to a second wireless access technology. The method further includes handing off the call from the first application layer client to the second application layer client.

Abstract: A method for transmitting data from a participant device in a session in an Internet Protocol (IP) system (100) is provided. The method comprises sending (405) one or more a first Real-time Transport Protocol (RTP) packet of the session to a first address. The method further comprises sending (410) one or more a first Real-time Transport Control Protocol (RTCP) packet of the session to a second address. The participant device sends the one or more a first RTP packet and the one or more a first RTCP packet.

Abstract: Certain exemplary embodiments provide a method, comprising: receiving, at a subscriber interface line card, an analog signal from a POTS subscriber loop circuit; quantizing the analog signal into a plurality of digital samples; encoding the plurality of digital samples via codec instructions running on a digital signal processor installed on the subscriber interface line card; and converting, via conversion instructions running on the digital signal processor, the encoded plurality of digital samples to a plurality of VoATM packets.

Abstract: A meeting aid (10) configured to facilitate the identification of persons in attendance at a meeting, said meeting aid being one of a set of such meeting aids each for allocation to a said person in attendance at a meeting, the meeting aid comprising: a data input device (60) configured to allow the person to whom the meeting aid has been allocated to input information identifying themselves; communication means (65) for receiving identifying information from one or more of the other meeting aids of said set of meeting aids; a processor (70) configured to process and save received identifying information and identifying information inputted by the person to whom the meeting aid has been allocated, said saved identifying information providing the person to whom the meeting aid has been allocated with a list comprising at least a portion of the identifying information associated with those persons to whom each meeting aid of the set of meeting aids has been allocated; and display means (50) responsive to said p

Abstract: An embodiment of a conference bridge adapted for voice-over-IP applications has a modular, scalable architecture and establishes and internally relocates conference calls using SIP protocol commands for efficient processing. Further embodiments of the conference bridge support a plurality of conference access numbers. Dialed Number Identification Service groups may be programmed to selectively vary functions of the conference bridge depending on the DNIS identification of the called number.

Abstract: Disclosed are systems and methods that provide centralized or nodal inmate management and telephone call processing capabilities to controlled environment facilities. An inmate management and call processing system serves a plurality of facilities and includes an inmate information database. The database is shared across the facilities and contains inmate records that may be accessed and modified by each facility as the inmate is transferred among those facilities. Each inmate record may include, among other information, contact information of third parties whom may be notified of the inmate's arrest and/or subsequent transfers of the inmate to different facilities. The inmate management and telephone call processing system may use third party contact information to establish accounts used to charge calls or transactions made by an inmate while residing at the facility.

Abstract: A method for communicating data during an audio conference is provided. Digital data is received from a data source. The digital data is then modulated onto a carrier signal. The modulated carrier signal is subsequently combined with an audio signal of the audio conference. The combined signal is communicated to at least one remote communication device. Upon receipt by the at least one remote communication device, the combined signal is separated into the audio signal and the modulated carrier signal. The modulated carrier signal is then demodulated to extract the digital data. Both the digital data and the audio signal are output by the at least one remote communication device.

Abstract: Method and apparatus for call interworking between user equipments (UEs) located in domain using different communication protocols. The methods and apparatus receive a call request for establishing a call between a first UE and a second UE, the call request including a list of addresses corresponding to the second UE. A list of list of addresses corresponding to the first UE is then retrieved and a forking operation is performed the received list of addresses corresponding to the second UE. Then a determination is made, based on a set of rules, on which address to be used from the list of addresses corresponding to the first UE and which address to be used from the list of addresses corresponding to the second UE for establishing the call between the first UE and the second UE.

Abstract: Certain exemplary embodiments provide a method, comprising: receiving, at a subscriber interface line card, an analog signal from a POTS subscriber loop circuit; quantizing the analog signal into a plurality of digital samples; encoding, via high-quality audio codec instructions running on a digital signal processor installed on the subscriber interface line card, the plurality of digital samples; and converting, via conversion instructions running on the digital signal processor, the encoded plurality of digital samples into a plurality of VoATM packets.

Abstract: A network based communication system allows a communication connection that carries voice data using packetized IP telephony protocol to be established via a wireless local area network access point by sending a user address to a remote database server. The database server converts the user address to an Internet address and informs the wireless handset of an active Internet address to be used to complete the call. In some embodiments the user address is a telephone number that can also be used to reach the user using the public switched telephone network. Other types of media are also supported like digital pictures, digital video, data conference sessions, and application sharing sessions.

Abstract: A voice conference apparatus consists of at least one base conference block having three or more signal ports, each including an input for a digitalized voice signal and an output for a digitalized voice signal, wherein a base conference block is configured such that, at a signal port, optionally a terminal or another base conference block may be coupled up using a coupler for the realization of a voice conference apparatus extended with respect to the number of participants or in terms of space, and wherein a base conference block is configured to enable a voice conference among the terminals or base conference blocks coupled thereto, and a remote coupler configured to couple a base conference block with at least one other spatially remote base conference block in order to enable a spatially extended voice conference. A voice conference system may be composed of at least two spatially separate voice conference apparatuses and is structured in a decentralized and modular way and is flexibly extensible.

Abstract: In FDMA communications between two or more primary radios (41, 42; 141, 142), use is made of a quiescent period in the primary communication. The quiescent period is used by secondary radios (43, 44; 143, 144) to communicate with each other. The quiescent period may be a transmit interrupt (TI) frame, normally provided in a DIIS protocol in order to permit return signaling from a called radio (43, 44). The secondary radios (43, 44) may signal to each other during the TI frame. Alternatively, the quiescent period may be a gap in channel activity, in which case regulated access telegrams (96) are transmitted by either a calling primary device (141), or a “last to call” primary device (141), or a base station (141), the regulated access telegrams determining a reservation window period during which secondary devices (143, 144) may signal to one another.

Abstract: A system and method for cascading local conferences into a multi-site conference is disclosed. The local conferences are located in different sites where each site has endpoints served by a local multipoint control unit (MCU). An invitation to a multi-site conference is sent to participants at the different sites. The invitation includes a conference ID of the multi-site conference. The invited participants join the multi-site conference by calling a dial-in number of the local MCU associated with the caller's site. The caller gives the conference ID of the multi-site conference to the local MCU. The system searches the plurality of local MCUs to find those local conferences having the same conference ID conducted by different MCUs. For each found conference ID, the local MCUs are cascaded to establish a multi-site conference associated with the conference ID.

Abstract: To control a conference circuit between multiple terminal devices in a packet-oriented communication network, circuit-oriented conference control signaling is transmitted via a packet-oriented transport protocol between a first terminal device and a communication server of the communication network. Setting up of packet-oriented media channels between the terminal devices and a packet-oriented conference mixing device is initiated by the communication server depending on the conference control signaling.

Abstract: The invention relates to a method for managing a packet switched, centralized conference call between a plurality of terminals 13. In order to enable an enhancement of the user comfort, it is proposed that the method comprises at a conference call server 12 receiving data packets from all terminals 13. Based on these data packets, then at least one terminal 13 currently providing voice data is determined. In a next step, the data received in the data packets is mixed, and the mixed data is inserted into new data packets together with at least one identifier associated to one of the terminals 13 which were determined to provide voice data, such that the at least one identifier can be distinguished from any other information in the data packets. Finally, the new data packets are transmitted to terminals 13 participating in the conference call. The invention relates equally to a corresponding server and to a corresponding terminal.

Abstract: Methods and systems for utilizing a parallel conversational thread during a remote collaboration are described. In one embodiment, a managing entity is utilized to establish a first communications path for a remote collaboration of at least two parties. Then, a request for a parallel conversation is received from at least one of the at least two parties during the remote collaboration. The managing entity then establishes a second communications path for the parallel conversation between the at least two parties. Wherein, the enabling of the parallel conversation on the second communications path is performed without inhibiting the first communications path, such that a portion of a first party can have a parallel conversation with a portion of a second party on the second communications path while the first communications path for the remote collaboration remains active.

Abstract: Methods, systems, and computer-readable media for admitting conferees into a conference call. Methods to achieve the foregoing can comprise: receiving a request from the conferee for access to the conference call; obtaining data representing a unique identifier associated with the conferee; matching the data representing the unique identifier with data representing an identifier that was previously associated with the conferee; and connecting the conferee directly to the given conference call.

Abstract: A system, apparatus, method, and computer program product for establishing secure videoconferences between multiple nodes communicating via two or more transport protocols in which at least one encryption protocol is used.

Abstract: Tools and techniques for determination and display of endpoint identities are described herein. The tools may provide machine-readable storage media containing machine-readable instructions for receiving responses to requests to open call sessions involving given endpoints, and for extracting human-understandable identification information from these responses. These instructions may also extract previously stored identification information associated with the endpoint, where the identification information was extracted from a previous response transmitted from the endpoint.

Abstract: Video, voice and data are integrated using existing telephone connections by using a video network server to receive and transmit video signals to and from clients using telephone wires. Client frequency division multiplexing couplers, each associated with one of the clients, receive video and telephone signals from the client and frequency multiplex them onto the telephone wires for transmission. They also receive video signals from the server and frequency demultiplex them for transmission to the client. A video frequency division multiplexing coupler connected between the video network server and each of the client frequency division multiplexing couplers receive the video and telephone signals from the clients and transmit only the video signals to the video network server.

Abstract: Connecting an endpoint to a conference call includes associating a caller identifier with a conference call identifier, where the caller identifier corresponds to an endpoint and the conference call identifier corresponds to a conference call. A call is received from the endpoint, and the caller identifier is determined to be associated with the conference call identifier. Connection of the endpoint to the conference call is initiated.

Abstract: A video conferencing terminal includes an encoder that receives a baseband audio signal and a baseband video signal and creates compressed audio packets and compressed audio-video packets. A network interface driver receives the compressed audio packets and the compressed audio-video packets, IP encapsulates the compressed audio packets and the compressed audio-video packets, transmits the IP packets onto an IGMP network. The network interface driver also requests audio packets and audio-video packets from the network and recovers compressed audio packets and compressed audio-video packets from the IP packets received from the network. A decoder receives compressed audio packets and compressed audio-video packets from the network interface driver and generates a baseband video signal and multiple baseband audio signals.

Abstract: When one of two IP telephones that are maintaining a call receives a TV camera connection request, the IP telephone obtains a URI of an IP TV camera service from a resource record of the opposing terminal. Then, the IP telephone delivers the URI of the IP TV camera service to an IP TV camera adaptor of its own IP telephone in order to make a video communication request. The resource record of the opposing terminal is associated with a telephone number of the opposing IP telephone, the record being obtained by connecting to a DNS server. The IP TV camera adaptor connects a call, via the IP telephone, to an IP TV camera adaptor of the opposing terminal, using the URI of the IP TV camera service of the opposing terminal and performs video image communication.

Abstract: Apparatus, and an associated method, for providing a Web-based directory with automatic dialing capabilities that may be used by the user of a standard wireline or cellular telephone to establish contact with entities having SIP phones or other IP-based communication devises. The Web-based directory includes not only the names of various entities, but information on how to establish voice contact with them as well such as IP addresses or other contact numbers. A Web-based directory user establishes an Internet connection and retrieves one or more Web pages of the Web-based directory, selecting from them one or more entities from among those listed.

Abstract: A method (30) schedules the utilization of the supplemental channel of a base station transceiver (20-25). This scheduling includes time slot assignment as well as data transfer rate per time slot. One method simply selects the next time slot with a maximum rate for the primary base station transceiver (130). The method then selects the same time slot for each of the secondary links with the secondary base stations (132). The data is then simply sent to each of the BTSs (20-25) for transmission to the mobile station (10). In another alternative, a request is made for a supplemental channel usage for the primary link (144). Then secondary links are selected for transmission to the mobile station (10) only if they provide additional diversity gain (148) and resources are available at the secondary link BTSs.

Abstract: A method and apparatus are provided for setting up a voice call between a client and one of a plurality of agents of an organization. The method includes the steps of detecting a contact request associated with the client determining a communication system address of the client based upon the detected contact request transferring a request to a router at a router Internet protocol address for a voice connection between the client at the determined address and one of the plurality of agents selecting an agent of the plurality of agents and forming a voice channel between the determined address of the client and an agent Internet protocol address of a terminal of the selected agent.

Abstract: A system and method for handling multimedia conference calls in a telecommunication network. To determine the capabilities of the other terminals in a group, a first terminal sends a query message to a group server. The group server sends further query messages to the other terminals and collects their responses containing their capabilities. The group server analyzes the capabilities to determine a common set of capabilities. The group server then sends the common set of capabilities in a response to the first terminal.

Abstract: In a call between terminals in different FWs, tunnel devices T1, T3 mutually conduct call control and speech packet transmission by way of a relay device T2 outside of the FWs, thereby enabling a call through transmissions of speech packets and the like between the terminals even in the case of Symmetric NAT. A speech packet transmission path for RTP and the like between terminals is set as a directly routed path (direct path) rather than by way of a tunnel device thereby eliminating bottleneck in the tunnel. The IP telephone technology can be provided for transmitting speech packet by way of a tunnel even in the case of Symmetric NAT.

Abstract: A method and system for reducing latency in initiation of a real-time data session between wireless communication stations. In order to initiate a session with one or more target parties, an originating station will send a voice call origination message to a network switch, providing in the origination message (i) a session-initiation feature code and (ii) a target code indicative of the one or more target parties. The switch will then detect the session-initiation feature code as an intelligent-network trigger and responsively signal to a service controller, which will in turn signal to a communication server. The communication server will then invite the originating station and each of the one or more target parties to participate in the session, setting up a conference leg with each and bridging the legs together to allow the session to proceed.

Abstract: A method and system for scheduling and conducting videoconferences that ensure that the necessary bandwidth for the videoconference will be available and that select the least cost equipment to carry the videoconference. An embodiment of the present invention includes the steps of determining for a schedule request by a user if the necessary bandwidth to conduct a videoconference will be available. Also, the embodiment measures the performance of equipment and systems used to conduct the videoconference and determines if the equipment and systems are performing at a level necessary to conduct the videoconference. Alternative embodiments may include selecting multipoint endpoints to support the videoconference, scheduling videoconferences over two independent networks, emulating routers to ensure that the least cost route for the videoconference is selected, and responding to a scheduling request by informing the user to reschedule the videoconference when the network cannot support the videoconference.

Abstract: A method and system for location based subject matter teleconferencing is disclosed. A user makes a request to join a teleconference based on the user's location and subject matter. A conference-call-setup-system (“CCSS”) receives the request. The CCSS determines the location of the user and identifies the teleconference that matches the request. Additionally, the CCSS may use the time of the request when identifying the teleconference that matches the request. The CCSS then arranges for a conference bridge to join the user to the teleconference. As a result, the user can communicate with other users in the same general location who want to discuss the same subject matter.

Abstract: In one embodiment, a method includes receiving data that indicates a maximum number of end nodes with which to set up real-time data sessions that is less than a conference number of nodes whose data is mixed in a single real-time data conference. Real-time data sessions are established with a connected set of a connected number of nodes not greater than the maximum number. A mix of the real-time data received from the connected set is presented to a user of the first node. Local real-time data is received from a user. The received real-time data is distributed by mixing and sending. This includes, for each individual node in the connected set, mixing received local real-time data with real-time data received from other nodes in the connected set different from the individual node to form individual mixed real-time data, and sending the individual mixed real-time data to the individual node.

Abstract: The present invention discloses a solution for performing a sharing an ongoing data session between a first user and a group of at least one additional user in a telecommunication system. The basic idea of this invention is to provide a network integration node with a shared streaming session logic, replication logic and group membership management functions of a conference node, like for example the “Push to-talk over Cellular (PoC)” application. Using this functionality the conference system is to be included into the ongoing streaming session, shall invite additional members and forward the streaming traffic to them as well. This results into a shared streaming service using unicast communication resources.

Abstract: A system and method for providing additional information to conference participants with regard to a predefined conference event such as scheduled conference. In various embodiments, outgoing conference indications include an identifier that indicates to a recipient that the conference invitation is being sent due to the occurrence of a scheduled event and/or due to the occurrence of other conditions being met.