1855 results about "Session Initiation Protocol" patented technology

A method and system to provide push-to-talk from one user to another in a wireless packet data telecommunications network is described. The system may include: a wireless communication network including push-to-talk (PTT) functionality, with a Session Initiation Protocol (SIP) Proxy Server; a SIP Registrar and Location Server operable to store contact addresses of active mobile devices; a Realtime Transport Protocol (RTP) Media Gateway (PTT Server) operable to function as a call endpoint for each of a plurality of mobile devices wherein the plurality of mobile devices are segmented into membership groups, the PTT Server further operable to multicast a communication from one member of the group to the other members of the group; and an Internet Protocol (IP) network interconnecting the SIP Proxy server, the SIP Registrar and Location Server, and the PTT Server.

A method, apparatus, and computer program product for flexibly assigning Session Initiation Protocol (SIP) Endpoints to any part of a communication resource is disclosed. The present invention removes the device location and behavior requirements from the application, thus providing a generic interface for robust application development.

Upper layer applications discover, establish and manage the QoS connectivity through a SIP (session initiation protocol) that serves as a unified interface. In one implementation, this is IP Multimedia Subsystem-centric, further supporting applications that utilize SIP for session control. This capability extends to the data card universe allowing UMTS data card vendors to establish concurrent QoS-based sessions using multiple primary PDP (packet data protocol) contexts based on a set of SIP triggers, further allowing applications running on a computing system to transparently utilize the established pipes based on the individual QoS requirements.

A system and method for providing Value-Added Services (VAS) in an integrated telecommunications network having a packet-switched network portion (PSN) operable with Session Initiation Protocol (SIP). The integrated telecommunications network includes a SIPext SSP server, a trigger server, and a service node having a Service Logic Program (SLP) that is operable with Intelligent Network Application Protocol (INAP). The SIPext SSP and service nodes are provided with the capability to communicate using SIP-compliant messaging. New header fields are provided that specify operations to be performed by the service node with respect to a service. INAP service parametric data is also provided in the header fields in a sequential form. When a call is received in the SIPext SSP server for a user having a subscription for a VAS, it queries the user profile stored in the trigger server. If the user is subscribed for a service, a SIP request message is formulated based on the user profile, wherein appropriate headers are populated with relevant parametric information and call context data. The service node launches the SLP based on the information provided in the request message and sends a SIP response message to the SIPext SSP server with an instruction concerning the provisioning of the VAS. The SIPext SSP server, thereafter, takes an appropriate action based on the response message and any parametric information contained therein.

A network processing device is signaled to establish a media path with another network processing device in a packet switched network with Session Initiated Protocol (SIP), International Telecommunication Union (ITU) standard H.323, MGCP, Megaco, or some other signaling protocol. The network processing device receives a request to send information to an observer device that is not directly in the media path. The network processing device then sends information about the media path to the observer device.

Systems, methods, and machine-readable media are disclosed for providing customization to routing of messages in a system where routing is based on stored addresses, such as, for example, a Session Initiation Protocol (SIP) system. In one embodiment, a method of providing customized routing of messages can comprise receiving a message to a target endpoint at a system for signaling between endpoints. The system can have one or more stored addresses (Target SIP URIs) for each of a plurality of endpoints. One or more target Sip URIs for the target endpoint can be determined from the stored target SIP URIs. One or more policies can be applied to the one or more target SIP URIs for the target endpoint. The one or more policies can comprise a set of one or more conditions and one or more actions associated with each condition.

Systems, methods, and machine-readable media are disclosed for switching a media streaming session between a plurality of networks. In one embodiment, a method of switching networks in a media streaming session can comprise detecting a plurality of networks available for communication of streaming data. The plurality of networks can include a first network providing a first streaming session and a second network. A determination can be made as to whether to switch the first streaming session from the first network. Determining to switch the first streaming session from the first network to the second network can be based on detecting a loss of the first network, detecting a lower cost alternative to the first network, etc. In response to determining to switch the first streaming session from the first network, the first streaming session can be switched from the first network to the second network.

A system is provide for extending services to cellular devices. The system includes a wireless gateway having a client side with an intelligent client agent and a server side with an intelligent server agent, a wireless device having a first mode operable in a cellular system and a second mode operable in a wireless local area network and including an intelligent client agent for determining operation mode and for communicating with the intelligent client agent of the wireless gateway and a communications protocol for communication between the wireless gateway intelligent client agent and the wireless device intelligent client agent to effect services. The services include one or more of voice, directory, presence, and media services. The voice service includes at least one of call waiting, call forwarding conferencing, and caller identification. The media service includes at least one of interactive voice response (IVR), text-to-speech, speech recognition, play, record. The directory service includes a public switched telephone network (PSTN) directory, a cellular network directory, an Internet directory and a private branch exchange (PBX) directory. The communications protocol includes encapsulation for at least one of directory, mobility and security services and other protocols. The other protocols include at least one of Internet protocol (IP), session initiation protocol (SIP), and lightweight directory access protocol (LDAP).

A system that incorporates teachings of the present disclosure may include, for example, a Unified Messaging System (UMS) having a Session Border Controller (SBC) to receive a Session Initiation Protocol Uniform Resource Identifier (SIP URI), detect from a host portion of the SIP URI that the SIP URI is associated with a failover communication session, update a user portion of the SIP URI to indicate a failure disposition, and route the updated SIP URI to a call agent of the UMS. Additional embodiments are disclosed.

A method and system for operating a standalone client in a dual-mode mobile communications device to hand off a call between first and second wireless networks operating under different communications protocols, using Session Initiation Protocol (SIP) and ISDN User Part (ISUP) signalling. The standalone client-side application, or handoff agent, directly controls handoff of the call to the second wireless network until at least one new media stream to connect the dual-mode mobile communications device to the second device is established over the second wireless network. The standalone client-side application solely originates and controls an exchange of messages at the application layer, the exchange of messages modifying the communications channel.

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

Systems, methods, and machine-readable media are disclosed for providing customization to routing of messages in a system where routing is based on stored addresses, such as, for example, a Session Initiation Protocol (SIP) system, to pass through a firewall. In one embodiment, a method of providing customized routing of messages to pass through a firewall can comprise receiving a message from an initiating agent to a target communication identifier at a system for signaling between endpoints. The system can have one or more stored network addresses for each of a plurality of communication identifiers. One or more network addresses for the target endpoint can be determined from the stored network addresses associated to the communication identifier. One or more policies can be applied to the one or more network addresses for the target communication identifier to route the message through the firewall.

A method and mechanism to allow a location server to initiate a user plane location service (e.g., SUPL defined by OMA) procedure to a user plane enabled device via Instant Messaging, or alternatively, via an existing SIP session if a multimedia session is already established. The location request is signaled to the user plane enabled device via a SIP INFO message. The location request uses SIP messaging to overcome firewall and other network security issues. Location using SUPL over SIP may be provided about a caller making an E911 emergency call. An Instant Message may be sent to the user plane enabled device, e.g., a VoIP wireless phone.

A soft switch 134 providing wireless PBX voice services to a local area network (WLAN) is used to extend PBX functionality to the cellular domain. A dual mode remote unit is capable of receiving signals both in the cellular system as well as the WLAN. The cellular system is comprised of a data-bearing path and a voice-bearing path. When the dual mode remote unit is within the WLAN, it communicates both voice over IP (VoIP) signaling as well as session initiation protocol (SIP) control signaling over the WLAN. When the remote unit is outside the WLAN, it communicates voice signaling over the voice-bearing path of the cellular network using a standard cellular voice channel. In parallel, it uses the data-bearing path of the cellular network to transmit SIP control signaling.

Methods, systems, and computer readable media system for collecting and distributing session initiation protocol (SIP) event watcher entity subscription information in a communications network are disclosed. According to one aspect, the method includes receiving, at a first notifier server from a watcher entity client, a first subscription request for SIP event information associated with one or more watched entity clients. The method also includes receiving, at a watcher information (WINFO) entity, SIP event subscription information that includes an identifier indicating that the first notifier server serves the watcher entity client, storing the SIP event subscription information in the WINFO entity, detecting the unavailability of the first notifier server. The method further includes sending an instruction message, based on the identifier in the stored SIP event subscription information, to direct the watcher entity to generate a second subscription request associated for the SIP event information.

In a first embodiment of the present invention, a method for providing single sign-on in a network having a HyperText Transfer Protocol (HTTP) portion and a Session Initiation Protocol (SIP) portion is provided, the method performed at a gateway and comprising: receiving an HTTP request for an assertion from a requester over the HTTP portion; generating a SIP request using the request for assertion; sending the SIP request to a SIP registrar over the SIP portion; receiving a SIP response including information regarding an assertion from the SIP registrar; and sending the information regarding the assertion in an HTTP response to the requester, such that the requester can use the information regarding the assertion in authenticating the requester to a web server.

A system is provided for establishing a communication session with a terminal (i.e., terminating SIP client). The system includes a network node (e.g., SIP proxy) located in a network across which an originating client is capable of communicating. The network node is capable of sending a trigger to the terminal independent of the network. The network node is also capable of receiving a registration message, in response to the trigger, from the terminal across the network. In this regard, the network node is capable of receiving the registration message to thereby identify the terminal across the network such that a communication session is capable of being established with the terminal based upon the identity of the terminal across the network.

A system and method of providing a subscriber service to service users in a telecommunications network. In networks utilizing Session Initiation Protocol (SIP) control signaling for call setup and control, the SIP REGISTER message is modified to indicate service capability information and optionally a traffic load indication for service providers. The REGISTER message is sent to a modified Presence and Instant Messaging (PIM) server that stores presence information and the service capability information for registered service providers. The PIM server then notifies subscribing service users of the identity of the service provider that is registered on the network. The PIM server may utilize the traffic load information to balance the traffic load between service providers by providing users with the identity of the service provider that is the most lightly loaded.

A system that incorporates teachings of the present disclosure may include, for example, a server having a controller to receive presence information from each presence source of a plurality of presence sources, wherein the presence information is associated with a user, receive a Session Initiation Protocol (SIP) Uniform Resource Identifier (URI) which identifies the user, wherein identification information associated with the user from each presence source is mapped to the SIP URI, select at least a portion of the received presence information to be utilized in determining a presence status of the user, determine the presence status of the user based on the selected presence information, and transmit the presence status and the SIP URI to a computing device operably coupled to an Internet Protocol Multimedia Subsystem (IMS) network. Other embodiments are disclosed.

The invention relates to a method for the provision of a user interactive dialogue UID prior to connection acceptance service for telecommunication terminals in a communication network, wherein a media gateway controller forms a service switching point SSP. Parameters and signaling information necessary for controlling the UID prior to connection acceptance service are transmitted by conversion to the session initiation protocol SIP or are transmitted there from in a standard signaling protocol.

A soft switch 134 providing wireless PBX voice services to a local area network (WLAN) is used to extend PBX functionality to the cellular domain. A dual mode remote unit is capable of receiving signals both in the cellular system as well as the WLAN. The cellular system is comprised of a data-bearing path and a voice-bearing path. When the dual mode remote unit is within the WLAN, it communicates both voice over IP (VoIP) signaling as well as session initiation protocol (SIP) control signaling over the WLAN. When the remote unit is outside the WLAN, it communicates voice signaling over the voice-bearing path of the cellular network using a standard cellular voice channel. In parallel, it uses the data-bearing path of the cellular network to transmit SIP control signaling.

The embodiments disclosed include a system and method for sharing services between service profiles on a single telecommunications device resulting in improved ease of access for a user who wishes to access services through different service profiles. The user can access services in multiple service profiles with a single device. In one embodiment, the techniques described below are enabled through a Session Initiation Protocol (“SIP”)-based next-generation network (“NGN”), such as the IP Multimedia Subsystem (“IMS”) architecture.

A software suite is disclosed for routing communication events over a data-packet-network using an IP session initiation and management protocol. The software suite comprises, a server application running on the network for computing and serving routing determinations per request, a session management application running on the network for initiating and managing routed and established session events, a parsing application running on the network for parsing request data received under session initiation protocol and a conversion application running on the network for converting data received under session initiation protocol into a routing request. All received communication requests for routing are in the form of the session initiation protocol wherein they are parsed and converted into routing requests processed by the server application and routed to determined destinations and wherein events are established as session events conducted under the session initiation and management protocol.

A system and method of providing a subscriber service to service users in a telecommunications network. In networks utilizing Session Initiation Protocol (SIP) control signaling for call setup and control, the SIP REGISTER message is modified to indicate service capability information and optionally a traffic load indication for service providers. The REGISTER message is sent to a modified Presence and Instant Messaging (PIM) server that stores presence information and the service capability information for registered service providers. The PIM server then notifies subscribing service users of the identity of the service provider that is registered on the network. The PIM server may utilize the traffic load information to balance the traffic load between service providers by providing users with the identity of the service provider that is the most lightly loaded.

A method, computer readable medium having computer executable instructions, and a computer readable medium having stored thereon a data structure for signing and validating Session Initiation Protocol (“SIP”) routing headers are disclosed. A SIP node may receive a SIP request including a message header. A signature based upon at least a portion of the message header and a SIP node header entry may be generated. The signature may then be inserted into the SIP node header entry.

A subscriber of a communications network is enabled to top-up an account for a multimedia service provided on the communications network while the service is being provided. Real-time prepaid charging may be applied to a multimedia service being provided to a subscriber of a communications network. Both a count and a time period may be determined from an account balance, and the number of information units exchanged during the service and the duration of the service may be compared against the count and time period, respectively. Further processing may be defined to be performed if either and / or both thresholds are reached. A subscriber of a communications network may be notified of a threshold amount of service being reached while the service is being provided using the same session as is being used to provide the service (in-band) or using a different session (out-of-band). Further, a subscriber may be notified of a threshold being reached using a Short Message Service (SMS) by sending an SMS message to the subscriber. A subscriber of a communications network may be enabled to top-up a service account while the service is being provided using the same session as is being used to provide the service (in-band) or using a different session (out-of-band). To notify the client that a threshold for a service has been reached and / or to enable the client to top-up, either in-band or out-of-band sessions may be used based on any of a number of factors, including a characteristic of the session in progress implementing the service, the capabilities of the user terminal, the capabilities of the application and application server providing the service, or any combination thereof. A session for providing a multimedia service may be implemented using a multimedia control protocol capable of controlling (e.g., initiating, maintaining and terminating) a session that includes the exchange of multimedia content, including audio, video, data or any combination thereof. For example, such protocol may be the Session Initiation Protocol (SIP). Service content transmitted from a server to a subscriber on the subscriber's user terminal may be buffered (e.g., cached) in a buffer while the subscriber is provided the opportunity to top-up the account. While the subscriber is being enabled to top-up a prepaid account for a service, a different billing method, for example, postpaid charging, may be applied for the service until the subscriber tops-up or until the opportunity to top-up has elapsed.

A telecommunications network uses a supervisory service broker in a packet messaging environment, particularly Session Initiation Protocol (SIP) messaging, to changeably allocate and to account for the usage of network resources by users. The SIP INVITE and other messages are passed through the supervisory broker, monitored for accounting and potentially redirected to change allocations of resources. The resources can generally comprise applications and applications servers such as media storage and playback, voicemail and directory services, subnet messaging and instant messaging, voice-over-Internet, wireless linkages and others, and can be logically grouped in packages that are allocable. Users such as telecom carriers and others can contract for usage and also can contract to provide resources for use by others, in an unlimited way or according to terms, and for appropriate credit.

A method, network entity and test controller for establishing a Session Initiation Protocol (SIP) session between a plurality of SIP-based network entities and performing test measurements over the SIP session. The test controller instructs the network entities to establish the SIP session and specifies which test measurements are to be performed. The SIP session is established between the network entities, the test measurements are carried out, and the results are reported back to the test controller by at least one of the network entities, which may be terminals or packet data nodes such as a CDMA2000 Packet data Switched Node (PDSN) or Base Station Controller (BSC). The network entities may comprise SIP user agents and test modules in the forms of software and / or hardware.

A system, method, and apparatus are directed towards managing a Voice over IP (VOIP) messages over a network, employing the Real-time Transport Protocol (RTP) and Session Initiation Protocol (SIP) over the Transmission Control Protocol (TCP). The VOIP messages are sent by a source device to a destination device through a relay server. The relay server may throttle the VOIP messages employing buffer management. When the buffer is substantially full, the relay server will drop packets from the source device. Indication of the lost packets may be provided to the source device through a Real-time Transport Control Protocol (RTCP) report. The source device may then employ the RTCP report to modify a type of codec employed, and thereby adjust a rate of flow of VOIP packets sent towards the destination device. Additionally, the relay server may provide port translation services for RTP / RTCP packets between the source and destination devices.