Detailed description of SIP protocol

SIP protocol was made within IETF (Internet Engineering Task Force) – organization occupied with establishing Internet standards and general standards concerning TCP/IP protocols.

Such protocols as HTTP (Web) and SMTP (electronic mail) used in today’s most popular IP-services were taken here as the base. Ideologically SIP is based on the same approach as HTTP: request-reply. All SIP messages are textual and they can be read with eyes, but return codes – such as in HTTP therefore some of them will seem well known not only for the network administrators but also for many advanced Internet users (404 – Not Found, 200 – OK).
Substantially that SIP though it may be used in IP-telephony is not a protocol for transferring voice data – it is not attached to a certain kind data transmission. The name SIP deciphers itself as Session Initiation Protocol. This means that SIP ensures initiating, controlling and closing the sessions of information exchange, and the very transfer information could be anything: speech (as in the case of IP-telephony), music, video and, for example, text (the protocol allows organizing the sessions of collective work upon documents supported by MS Exchange and Lotus Notes).
The data type is determined by a separate protocol SDP (Session Description Protocol), which works in pair with SIP and has a wonderful opportunity to change the session parameters during the data exchange. The simplest example: two conversation partners are talking by IP-telephone and one wants to show a photograph to another – SDP will allow doing this within the framework of the real-time SIP session. Besides in principle nothing hinders the transfer to another terminal (for instance, if the conversation is made by mobile phone, you can switch unto computer or telephone with display and view the photograph there) during the action.

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RTP Delivers Multimedia Data In Real Time

A Real-time Transport Protocol (RTP) is a protocol standard defined by the Internet Engineering Task Force (IETF) that outlines a management system for programs using real-time transmission of data through mutlicast or unicast network services in multimedia situations. Originally designed by the IETF for videoconferencing for multiple participants, the protocol is largely used in Voice over Internet Protocol applications. Despite its name, the protocol cannot guarantee real-time delivery of data, however, the RTP does compensate for jitters and can detect when data arrives out of sequence, both issues being popular in VoIP communication. In IP telephony, RTP works with a signaling protocol, such as SIP or H.323, in order to set up connections in a network.

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