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SIP is an application-layer control protocol that can establish, modify, and terminate
multimedia sessions such as Internet telephony calls. SIP can also invite participants
to already existing sessions, such as multicast conferences. Media can be added to
(and removed from) an existing session. SIP transparently supports name mapping and
redirection services, which supports personal mobility - users can maintain a single
externally visible identifier regardless of their network location.
SIP should be used in conjunction with other protocols in order to provide complete
services to the users. These architectures will include protocols such as the Real-time
Transport Protocol (RTP) for transporting real-time data and providing QoS feedback, the
Real-Time streaming protocol (RTSP) for controlling delivery of streaming media, the
Media Gateway Control Protocol (MEGACO) for controlling gateways to the Public
Switched Telephone Network (PSTN), and the Session Description Protocol (SDP) for
describing multimedia sessions.
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Core
| PSTN |
Extensions |
Minor Extensions
| Conferencing |
Call Control | Events
| QoS |
OAM 
