Configuring SIP, SPCP, and NAT
Session Initiation Protocol and Cisco IP Phones
Cisco Small Business SPA300 Series, SPA500 Series, and WIP310 IP Phone Administration Guide 99
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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).
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 125.
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.
SIP UA
SIP UA
SIP Proxy
SIP Proxy
RTP
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SIP Proxy
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