SIP Connection

SIP Connection

The Session Initiation Protocol (SIP) is a signaling communications protocol, widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks.

The protocol defines the messages that are sent between peers which govern establishment, termination, and other essential elements of a call. SIP can be used for creating, modifying and terminating two-party (unicast) or multiparty (multicast) sessions consisting of one or several media streams. Other SIP applications include video conferencing, streaming multimedia distribution, instant messaging, presence information, file transfer, fax over IP and online games.

Originally designed by Henning Schulzrinne and Mark Handley in 1996, SIP has been developed and standardized in RFC 3261 under the auspices of the Internet Engineering Task Force (IETF). It is an application layer protocol designed to be independent of the underlying transport layer; it can run on Transmission Control Protocol (TCP), User Datagram Protocol (UDP), or Stream Control Transmission Protocol (SCTP). It is a text-based protocol, incorporating many elements of the Hypertext Transfer Protocol (HTTP) and the Simple Mail Transfer Protocol (SMTP).

SIP works in conjunction with several other application layer protocols that identify and carry the session media. Media identification and negotiation is achieved with the Session Description Protocol (SDP). For the transmission of media streams (voice, video) SIP typically employs the Real-time Transport Protocol (RTP), which may be secured with the Secure Real-time Transport Protocol (SRTP). For secure transmissions of SIP messages the protocol may be encrypted with Transport Layer Security (TLS).