Cisco 521SG - Unified IP Phone VoIP Administration Manual page 100

Configuring SIP, SPCP, and NAT
SIP and Cisco IP Phones
Cisco Small Business SPA300 Series, SPA500 Series, and WIP310 IP Phone Administration Guide
SIP UA
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RTP
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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 real-
time delivery of data. RTP provides mechanisms for the sending and receiving
applications to support streaming data. Typically, RTP runs on top of UDP. See
Mapping with STUN.
SIP Proxy
SIP Proxy
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SIP Proxy
NAT
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