Session Initiation Protocol

Session Initiation Protocol, or SIP, is a signaling protocol, widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP). The protocol can be used for creating, modifying, and terminating two-party or multi-party sessions consisting of one or several media streams. The modification can involve changing addresses or ports, inviting more participants, or adding or deleting media streams. Other feasible application examples include video conferencing, streaming, multimedia distribution, instant messaging, presence information, file transfer, and online games. In November 2000, SIP was accepted as a 3GPP signaling protocol and permanent element of the IP Multimedia Subsystem architecture for IP-based streaming multimedia services in cellular systems.

The SIP protocol is an application layer protocol designed to be independent of the underlying transport layer; it can run on Transmission Control Protocol, User Datagram Protocol, or Stream Control Transmission Protocol. It is a text-based protocol, incorporating many elements of HTTP and SMTP. SIP employs design elements similar to the HTTP request/response transaction model. Each transaction consists of a client request that invokes a particular method or function on the server and at least one response. SIP reuses most of the header fields, encoding rules and status codes of HTTP and providing a readable text-based format.

SIP works in concert with several other protocols and is only involved in the signaling portion of a communication session. SIP is primarily used in setting up and tearing down voice or video calls. It is also found in messaging applications, such as instant messaging, subscription, and notification. The voice and video stream communications in SIP applications are carried over another application protocol, the Real-time Transport Protocol. Parameters for media streams are defined and negotiated using the Session Description Protocol which is transported in the SIP packet body.

A motivating goal for SIP was to provide a signaling and call setup protocol for IP-based communications that can support a superset of the call processing functions and features present in the public switched telephone network. SIP by itself does not define these features; rather, its focus is call-setup and signaling. However, it was designed to enable the construction of functionalities of network elements, designated proxy servers, and user agents. These are features that permit familiar telephone-like operations: dialing a number, causing a phone to ring, and hearing ringback tones or a busy signal.

Although several other VoIP signaling protocols exist, SIP is distinguished by its proponents for having roots in the IP community rather than the telecommunications industry. SIP has been standardized and governed primarily by the IETF, while other protocols have traditionally been associated with the International Telecommunication Union.

Related links

• HTTP GET - Plum IVR Glossary
• HTTP POST - Plum IVR Glossary
• Application - Plum IVR Glossary