摘要:
A method for combining Internet protocols in a Differentiated Services model environment is described. The Session Initiation Protocol (SIP) and Common Open Policy Service (COPS) are combined together to provide methods of setting up a session and tearing down a session, while maintaining Authentication, Authorization, and Accounting (AAA) policies. The Open Settlement Policy (OSP) is also combined with SIP and COPS. This combination provides for an interchange of parameters between session setup, teardown, authorization, policy, Quality of Service (QoS), and usage reporting
摘要:
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.
摘要:
Methods are disclosed for providing prepaid telephony service via an Internet protocol (IP) network system. A first method provides controlling at least one media agent or call routing station/switch of an IP network system for allowing and/or blocking call media streams from traversing through the media agent. A second method provides directing all signaling messages transmitted by a signaling agent or station and all media packets transmitting voice and data communications through at least one common device within the IP network system. The methods further provide for continuously monitoring a subscriber's account balance and terminating the prepaid telephony service if the account balance is less than a predetermined amount.
摘要:
A system and method for associating a handoff address to a communication session includes determining one or more communication sessions are established in a first network. Each of one or more handoff addresses is associated with each of the one or more communication sessions. Associating each of the one or more handoff addresses is according to a temporal order of establishment of the one or more communication sessions.
摘要:
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.
摘要:
A system and method for providing an addition to the Session Initiation Protocol is disclosed. The addition is a new field header, preferably entitled “Feature”, that is added to the REGISTER message. This field would contain control information for various feature services, like the Do Not Disturb feature and other services provided by traditional PBX systems.
摘要:
Gathering location information from a first wireless network to determine whether to anchor a communication session in a second wireless network, a mobile node capable of communicating with both the first wireless network and the second wireless network, includes receiving location information from the first wireless network as the mobile node moves through one or more service areas of the first wireless network. It is determined whether a triggering event occurs. If the triggering event occurs, the location information is stored to determine whether to anchor a session in the second wireless network.
摘要:
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.
摘要:
A virtual private network includes an internet protocol (IP) network and a public switched telephone network (PSTN). An enterprise gateway is operably connected to the IP network. The enterprise gateway is operably connected to a switch of the PSTN through a direct access line (DAL). The set-up signaling for virtual private network calls and the calls themselves are transported across the internet protocol network and the public switched telephone network through the direct access line.
摘要:
An approach for processing voice calls over a packet switched network as to efficiently utilize the functionalities of a Voice Response Unit (VRU). According to one embodiment, a call originator, acting as a User Agent Client in accordance with the Session Initiation Protocol (SIP), issues messages to establish a first call-leg with the VRU. The VRU performs digit collection to obtain information to authenticate the call originator and to authorize the voice call. Based upon the issued messages from the call originator, the VRU establishes a second call-leg with the call terminator. The VRU is released from the voice call after binding the call-legs to connect the call originator to the call terminator.