摘要:
A method and apparatus for providing routing of calls in a packet network, using one or more criteria extracted from signaling information to determine the routing for the calls are disclosed. The routing criteria extracted from signaling messages comprises at least one of: an access Uniform Resource Identifier, a destination phone number, a destination URI host, a calling party number, a calling party URI host, an incoming IP address, or a requested codec. An access URI and the egress URI are used to enhance routing decisions in a VoIP network. The egress URI can be used to specify egress route selections from the egress point of a VoIP network. The access URI can be used to influence the routing decisions within the VoIP network as well as the routing decisions with regard to egress routes from the egress point of the VoIP network.
摘要:
A method and apparatus for providing routing of calls in a packet network, e.g., a Voice over Internet Protocol (IP) network, using one or more criteria extracted from signaling information to determine the routing for the calls are disclosed. In one embodiment, the routing criteria extracted from signaling messages comprises at least one of: an access Uniform Resource Identifier, a destination phone number, a destination URI host, a calling party number, a calling party URI host, an incoming IP address, or a requested codec. An access URI and the egress URI are used to enhance routing decisions in a VoIP network. For instance, the egress URI can be used to specify egress route selections from the egress point of a VoIP network. The access URI can be used to influence the routing decisions within the VoIP network as well as the routing decisions with regard to egress routes from the egress point of the VoIP network.
摘要:
A method and apparatus for providing routing of calls in a packet network, e.g., a Voice over Internet Protocol (IP) network, using one or more criteria extracted from signaling information to determine the routing for the calls are disclosed. In one embodiment, the routing criteria extracted from signaling messages comprises at least one of: an access Uniform Resource Identifier, a destination phone number, a destination URI host, a calling party number, a calling party URI host, an incoming IP address, or a requested codec. An access URI and the egress URI are used to enhance routing decisions in a VoIP network. For instance, the egress URI can be used to specify egress route selections from the egress point of a VoIP network. The access URI can be used to influence the routing decisions within the VoIP network as well as the routing decisions with regard to egress routes from the egress point of the VoIP network.
摘要:
A method and apparatus for providing routing of calls in a packet network, e.g., a Voice over Internet Protocol (IP) network, using one or more criteria extracted from signaling information to determine the routing for the calls are disclosed. In one embodiment, the routing criteria extracted from signaling messages comprises at least one of: an access Uniform Resource Identifier, a destination phone number, a destination URI host, a calling party number, a calling party URI host, an incoming IP address, or a requested codec. An access URI and the egress URI are used to enhance routing decisions in a VoIP network. For instance, the egress URI can be used to specify egress route selections from the egress point of a VoIP network. The access URI can be used to influence the routing decisions within the VoIP network as well as the routing decisions with regard to egress routes from the egress point of the VoIP network.
摘要:
A method and apparatus for providing routing of calls in a packet network, using one or more criteria extracted from signaling information to determine the routing for the calls are disclosed. The routing criteria extracted from signaling messages comprises at least one of: an access Uniform Resource Identifier, a destination phone number, a destination URI host, a calling party number, a calling party URI host, an incoming IP address, or a requested codec. An access URI and the egress URI are used to enhance routing decisions in a VoIP network. The egress URI can be used to specify egress route selections from the egress point of a VoIP network. The access URI can be used to influence the routing decisions within the VoIP network as well as the routing decisions with regard to egress routes from the egress point of the VoIP network.
摘要:
A method and apparatus for enabling a network provider, in concert with IP technology and protocols, to provide the ability to offer a simple pre-answer or post-answer call redirection, such as call transfer, to customers with IP endpoints is disclosed. The present invention allows call transfers to be initiated from an IP endpoint but processed in the packet network, e.g., the VoIP network instead of being processed by the endpoint. When a redirecting party (RP) receives a call from a calling party (CP), the RP simply sends a VoIP signaling message to the network to initiate a call transfer to redirect the call from the CP to a TP instead and the network will complete the call transfer on behalf of the RP.
摘要:
A method for supporting advanced features in a core SIP network when the calling party is in a network operating with the H.323 protocol is disclosed. Specifically, after processing the calls by collecting information associated with the advanced features, a REFER message is sent by an application server to an ingress border element associated with the calling party. The ingress border element then sends an H.450.2 FACILITY message containing the information in the SIP REFER message to an application gateway associated with the calling party. Upon receiving a SETUP message from the application gateway, the ingress border element then translates that SETUP message into a SIP INVITE message to establish the call.
摘要:
Methods of controlling media server resources in a VoIP network are disclosed. In an embodiment, an IP node provides a service request. An application server receives the service request and sends a request for media server resources to a media server resource broker. The media server resource broker determines that the request should be handled by a first media server. The media server resource broker queries the first media server to obtain an IP address and port number for use in establishing a call between the IP node and the first media server. The media server resource broker then provides a signal to the IP Node so that it can establishing the call with the appropriate port on the media server. In an embodiment, the media server resource broker updates a database module that tracks assignment levels of the first media server so as to reflect the most recent request, thus decreasing the number of ports available for the first media server. When the call is complete, the media server resource broker, being in the signal path, can update the assignment level of the first media server.
摘要:
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. Upon detecting the request to execute the post answer call redirecting and/or teleconferencing in the first communication link, the IP-based portion of the multi-media provider systems responds by forming the multi-media communication path between at least first, second and third communication devices.
摘要:
A method and apparatus for enabling a network provider, in concert with IP technology and protocols, to provide the ability to offer a simple pre-answer or post-answer call redirection, such as call transfer, to customers with IP endpoints is disclosed. The present invention allows call transfers to be initiated from an IP endpoint but processed in the packet network, e.g., the VoIP network instead of being processed by the endpoint. When a redirecting party (RP) receives a call from a calling party (CP), the RP simply sends a VoIP signaling message to the network to initiate a call transfer to redirect the call from the CP to a TP instead and the network will complete the call transfer on behalf of the RP.