It is a 3GPP (Third Generation Partnership Project) signaling protocol.
It is one of the major signaling protocols used in Voice over IP (VoIP).
SIP was developed by IETF MMUSIC Working Group in an initiative to create a more robust standard compared to the H.
323, which is a current widely used standard for voice sessions.
The need was to cater to the increasing demand put by services such as video conferencing, unified messaging, and voice chatting, which are now being implemented through Internet.
SIP is based on Hypertext Transfer Protocol (HTTP).
It basically deals with embedding call setup and signaling features in networking elements such as user agents and proxy servers.
The signaling and call setup can be used by IP-based communication system to support the call processing functions and features present in the Public Switched Telephone Network (PSTN).
SIP enables a VoIP application to have the same kind of high quality and reliability that traditional telephone systems can deliver.
SIP adds quality and reliability to VoIP, which until recently has been associated with the term "low cost alternative".
SIP handles the signaling part of a communication session.
It serves as a carrier for the Session Description Protocol (SDP).
SDP handles the media portion of the session.
The transmission of voice and video content are done by the Real-time Transport Protocol (RTP).
A SIP session thus involves packet streams of RTP.
SIP is a part of the protocols involved in a multimedia session.
The major advantage of SIP is in its support for both IP and conventional telephone communication.
Communication service providers can leverage on this protocol to interconnect conventional telephone and IP communication services.
This especially sounds good for the vendors of VoIP telephony, wherein both the domains are involved.
Owing to SIP's roots in HTTP, voice based application can also be seamlessly integrated with web services.
SIP is scalable, easy to implement, and requires less setup time than its predecessor protocols.
Being text based, it is easy to program.
It is a peer-to-peer protocol, requiring no implementation in the network level.
The logic is implemented at the communication endpoints, which may be in hardware or software level.
Real time sessions are established when and where required.
Since SIP can be used to modify any session in progress, a normal telephone call session can be converted into a multi-party videoconference.
Users can join in the session no matter what kind of terminal he is using or where he is located.
The other person may be logged on to Internet through a PC, or may be traveling with a cell phone.
SIP holds lot of promise in today and tomorrow's communication world.
It has been established as a standard for call control and signaling on 3G cell phone networks by the Third Generation Partnership Project (3GPP).
This means that all multimedia and IP voice call signaling will be done through SIP.
New services involving fixed network IP services can thus introduced with ease.