Cisco SPA508G Administration Guide - Page 100
Configuring SIP, SPCP, and NAT, Mapping with STUN, Real-time Protocol RTP.
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Configuring SIP, SPCP, and NAT SIP and Cisco IP Phones 4 SIP UA 2 4 SIP Proxy RTP 3 1 SIP Proxy SIP Proxy SIP UA In typical commercial IP telephony deployments, all calls go through a SIP proxy server. The requesting phone is called the SIP user agent server (UAS), while the receiving phone is called the user agent client (UAC). SIP message routing is dynamic. If a SIP proxy receives a request from a UAS for a connection but cannot locate the UAC, the proxy forwards the message to another SIP proxy in the network. When the UAC is located, the response is routed back to the UAS, and a direct peer-to-peer session is established between the two UAs. Voice traffic is transmitted between UAs over dynamically-assigned ports using Real-time Protocol (RTP). RTP transmits real-time data such as audio and video; it does not guarantee realtime delivery of data. RTP provides mechanisms for the sending and receiving applications to support streaming data. Typically, RTP runs on top of UDP. See NAT Mapping with STUN. Cisco Small Business SPA300 Series, SPA500 Series, and WIP310 IP Phone Administration Guide 99