Web protocols and applications
Session Initiation Protocol (SIP)
Session Initiation Protocol (SIP) is becoming the most widely used call-signalling protocol in VoIP networks. SIP is an application layer protocol that is used to establish, modify, and terminate call sessions. It sets up the call (session) and closes it at the other end. It does not carry any voice or video data. This functionality is supplied by other protocols such as the Real-time Transport Protocol (RTP).
SIP was originally defined in RFC 2543 but that was obsoleted by RFC 3261.
SIP employs design elements similar to an HTTP request/response transaction.
Each SIP 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, providing a readable text-based format.
Some SIP commands include:
- REGISTER – used by a User Agent to identify its current IP address and the URLs for which it would like to receive calls.
- INVITE – used to establish a media session between User Agents.
- ACK – confirms reliable message exchanges.
- CANCEL – terminates a pending request.
- BYE – terminates a session between two users in a conference.
In this Course, you’ll further explore the web protocols that underpin the internet and the world wide web, and some of the applications they enable.
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