Sip Over Tcp - Cisco WIP310 - iPhone Wireless VoIP Phone Administration Manual

Configuring SIP, SPCP, and NAT
Session Initiation Protocol and Cisco IP Phones
Cisco SPA 500 Series and WIP310 IP Phone Administration Guide
SIP UA
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RTP
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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
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.
SIP Proxy
SIP Proxy
"Configuring NAT Mapping with STUN" section
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SIP Proxy
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