What is SIP ? Session Initiation Protocol	Presented by:	Copyright 2000©
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SIP provides the basic elements of telephony: call setup and termination, call
configuration, and data transfer. This is accomplished using SIP for call setup
and termination portion, SDP to describe call configuration, and RTP for data
transfer. RTCP is also used for data stream management.

SIP can run over any datagram or stream protocol such as UDP 10 , TCP, ATM,
and frame relay. SIP is commonly run over TCP/IP because of inexpensive
widespread connectivity, directory services, naming services, and a widely
known development environment.

The audio and video data streams are transported using the Real-time
Transport Protocol 11 (RTP) over UDP. SIP call signaling messages can be
carried over UDP or TCP, with UDP being the preferred method because of its
better performance and scaleability. One important consideration when using
SIP over UDP is that the entire message should fit within a single packet. If a
SIP message is fragmented into multiple datagrams, the probability of losing
the entire message increases with the number of fragments. When SIP
messages are being transmitted over a WAN, the retransmissions that result
due to lost fragments can seriously degrade call signaling performance. The
default port for SIP is 5060 although any available user port may be used. The
port to be used for RTP/RTCP is specified in SIP call signaling messages.

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