Linksys WIP310 SPA500 Series and WIP310 IP Phone Administration Guide - Page 92
SIP Over TCP, Configuring SIP, SPCP, and NAT, Configuring NAT Mapping with STUN
UPC - 745883580927
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Configuring SIP, SPCP, and NAT Session Initiation Protocol and Cisco IP Phones 4 SIP UA 2 4 SIP Proxy RTP 3 1 SIP Proxy SIP Proxy SIP UA 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). The Internet protocol RTP transmits real-time data such as audio and video; it does not guarantee real-time delivery of data. RTP provides mechanisms for the sending and receiving applications to support streaming data. Typically, RTP runs on top of the UDP protocol. See "Configuring NAT Mapping with STUN" section on page 106. SIP Over TCP To guarantee state-oriented communications, Cisco IP phones can use TCP as the transport protocol for SIP. This protocol is "guaranteed delivery", which assures that lost packets are retransmitted. TCP also guarantees that the SIP packages are received in the same order that they were sent. TCP overcomes the problem with UDP ports being blocked by corporate firewalls. With TCP, new ports do not need to be opened or packets dropped, because TCP is already in use for basic activities such as Internet browsing or e-commerce. Cisco SPA 500 Series and WIP310 IP Phone Administration Guide 81