Abstract: In order to facilitate a handoff of a communication session between a legacy circuit switched network and a packet data network such as a 3GPP LTE, 3GPP2 UMB, or WiMAX network, which communication session is anchored in an IMS Voice Call Continuity Application Server (VCC AS), a Circuit Switched Proxy (CS Proxy) is provided in the packet data network that communicates with a Mobile Switching Center (MSC) of the circuit switched network. The CS Proxy allows the packet data network to communicate with the VCC AS, the MSC to communicate with the packet data network, and the MSC to communicate with the VCC AS via the packet data network, thereby allowing an exchange of handoff-related information between the packet data network and the VCC AS, the packet data network and the circuit switched network, and the circuit switched network and the VCC AS via the packet data network.

Abstract: One or more gateways (107 and 108) as serve to interconnect the discrete call facilitation platforms (403 and 404) of a given communication network with an external communication network (109) as provided with the routing addresses of such discrete call facilitation platforms. The discrete call facilitation platforms, however, are not provisioned in a reciprocal fashion. Instead, the discrete call facilitation platforms are only provided with a subset of available gateway addresses. In a preferred approach, this is accompanied by less-than-fully-meshed connectivity between the gateways and the discrete call facilitation platforms.

Abstract: A method for routing bearer traffic (70?) between 3G networks (32, 42) and 2G networks (12, 22) and vice versa uses voice signaling gateways (VSGW)(14, 24, 34, 44). The method determines a VSGW nearest the calling party (11). The nearest VSGW directly routes (86) the bearer traffic through the inter-network (50) to the called party (31) and avoids the “tromboning” effect of always sending bearer traffic to the called party's home location.

Abstract: A communication system provides in-band speaker arbitration in a multi-participant communication session by use of RTP floor control messages that include a speaker arbitration command embedded in a data packet header extension.

Abstract: A communication system provides for an Internet Protocol Multimedia Subsystem (IMS)-based three way call. A mobile station (MS) that receives an invitation to participate in a call determines to switch the call to a three way call. The MS conveys a request to an IMS network to switch the call to a conference call. An IMS network server receives the request, determines a unique identifier for the conference call, and provides the conference call identifier to the MS via one or more of a Session Initiation Protocol Refer message, Message message, and Notify message. In response to receiving the conference call identifier, the MS dials into the conference call and requests that the call originator also dial in. The IMS network server then requests that the call originator dial in, and the call originator does so in response to the request.

Abstract: In order to facilitate a handoff of a communication session between a legacy circuit switched network and a packet data network such as a 3GPP LTE, 3GPP2 UMB, or WiMAX network, which communication session is anchored in an IMS Voice Call Continuity Application Server (VCC AS), a Circuit Switched Proxy (CS Proxy) is provided in the packet data network that communicates with a Mobile Switching Center (MSC) of the circuit switched network. The CS Proxy allows the packet data network to communicate with the VCC AS, the MSC to communicate with the packet data network, and the MSC to communicate with the VCC AS via the packet data network, thereby allowing an exchange of handoff-related information between the packet data network and the VCC AS, the packet data network and the circuit switched network, and the circuit switched network and the VCC AS via the packet data network.

Abstract: A session initiation protocol (SIP) based communication system registers a number of different device types (10, 50) in a database (30). Each type of communication device is registered including a communication device description, internet protocol address, priority, registration time, and expiration time (80). When a call is received by the network (20), the communication device at which to reach the user is first selected by priority (116) and then by registration time of the communication device (122).

Abstract: Transcoding needs and selection are performed by a central call control function (17, 35). Each transcoder (18, 23, 34, 36) registers with the appropriate call control (17, 35). The call control then determines whether a vocoder is required at all. If vocoding is required, call control (17, 35) selects the appropriate transcoder (18, 34) and inserts the appropriate transcoder into the bearer traffic stream (16).

Abstract: A third generation (3G) gateway provides the facilities for extending a number of internet protocols between a second generation (2G) network and a 3G network. The various messages of a session initiation protocol (SIP) are extended to provide user identification and server identification subfields. The SIP message extensions facilitate the communication of 2G and 3G networks for the purposes of implementing push-to-talk and dispatch service features.

Abstract: The voice signaling gateway (10) converts signaling and bearer traffic information between iDEN network (20) and various other networks (30–60). The voice signaling gateway includes a signaling portion (14) and a bearer traffic portion (12). The voice signaling gateway interfaces with any networks which support session initiation protocol (SIP). Further, the voice signaling gateway is capable of. interworking with both 3G and 2G type networks.

Abstract: The voice signaling gateway (10) provides group call and call back services among iDEN network (20) users (21) and a number of other various networks (30-60) users (31-62). The voice signaling gateway has packet data interface (17) and web interface (16) In an alternate embodiment of the invention, an integrated applications server (70) is added to the voice signaling gateway (100) to provide group call services. The integrated applications server has the functions for supporting group calls which were within the voice signaling gateway (10) and forms a stand-alone special feature system.

Abstract: The voice signaling gateway (10) provides group call and call back services among iDEN network (20) users (21) and a number of other various networks (30-60) users (31-62). The voice signaling gateway has packet data interface (17) and web interface (16) In an alternate embodiment of the invention, an integrated applications server (70) is added to the voice signaling gateway (100) to provide group call services. The integrated applications server has the functions for supporting group calls which were within the voice signaling gateway (10) and forms a stand-alone special feature system.

Abstract: The voice signaling gateway (10) converts signaling and bearer traffic information between iDEN network (20) and various other networks (30-60). The voice signaling gateway includes a signaling portion (14) and a bearer traffic portion (12). The voice signaling gateway interfaces with any networks which support session initiation protocol (SIP). Further, the voice signaling gateway is capable of. interworking with both 3G and 2G type networks.

Abstract: The present application describes various embodiments that address the need in wireless environments and within the SIP framework to efficiently maintain SIP contact addresses. It introduces the concept of a SIP proxy user agent (UA) (e.g., 123 or 124) to serve as a gateway between a SIP core network and a SIP-unaware mobile (101). A new message is described, a combined registration and event subscription message, which is used by SIP proxy UAs to both register a new contact address for a mobile and to subscribe to the mobile's contact information. By updating and subscribing to the contact information, a SIP proxy UA is notified of subsequent changes to a mobile's contact information. Registration and deregistration of contact addresses can thus be effected in a manner that allows SIP contact addresses to be more efficiently maintained.

Abstract: A communication system provides in-band speaker arbitration in a multi-participant communication session by use of RTP floor control messages that include a speaker arbitration command embedded in a data packet header extension.