Detailed description of SIP protocol

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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Return Call (*69)The function of the return call allows you to quickly and comfortably connect to the person, who has called you as the last. Simply dial *69 (USA) in order to find out the number and the call time, after that press the “1” key to make the call. In case the number is busy, the system will try to reach the line during 30 minutes and when the person will be available, you will receive a signal and be connected.

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