SIP

The Session Initiation Protocol (SIP) is an IETF-defined 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 (unicast) or multiparty (multicast) sessions consisting of one or several media streams. The modification can involve changing addresses or ports, inviting more participants, and adding or deleting media streams. Other feasible application examples include video conferencing, streaming multimedia distribution, instant messaging, presence information, file transfer and online games.

SIP was originally designed by Henning Schulzrinne and Mark Handley starting in 1996. The latest version of the specification is RFC 3261 from the IETF Network Working Group. In November 2000, SIP was accepted as a 3GPP signaling protocol and permanent element of the IP Multimedia Subsystem (IMS) 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 (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).

A SIP (Session Initiation Protocol) connection is a service offered by many ITSP (Internet Telephony Service Providers) that connects a company's PBX to the existing telephone system infrastructure (PSTN) via Internet using the SIP VoIP standard.

Using a SIP connection may simplify administration for the organisation as the SIP connection typically will use the same Internet connection that is used for normal data.

This removes the need to also have a BRI/PRI installed as well, although sharing the same bearer circuit for calls and data raises its own challenges in maintaining call quality.

If the call traffic runs on the same connection with other traffic like Email or Web, voice and even signalling packets may be dropped and the voice stream can get interrupted. To avoid this, many companies split voice and data up into two separate internet connections to solve this problem, so that the resource conflict on the Internet access side is avoided. Other devices perform traffic shaping in order to avoid this resource conflict, but they still depend on the merit of the service provider not to drop packets from the Internet to the PBX.



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