Linksys SPA922 Cisco Small Business Pro SPA and Wireless IP Phone Administrati - Page 90

SIP Over TCP, Configuring SIP, SPCP, and NAT, Configuring NAT Mapping with STUN

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5 88 Configuring SIP, SPCP, and NAT Session Initiation Protocol and Cisco IP Phones 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 111. 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 and Wireless IP Phone Administration Guide

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Configuring SIP, SPCP, and NAT
Session Initiation Protocol and Cisco IP Phones
88
Cisco SPA and Wireless IP Phone Administration Guide
5
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 111
.
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.
SIP UA
SIP UA
SIP Proxy
SIP Proxy
RTP
1
2
3
4
SIP Proxy