Asterisk: The Future of Telephony, 2nd Edition

Configuring SIP Telephones

The Session Initiation Protocol (SIP),^[57] commonly used in VoIP phones (either hard phones, or softphones), takes care of the setup and teardown of calls, along with any changes during a call such as call transfers. The purpose of SIP is to help two endpoints talk to each other (if possible, directly to each other). The SIP protocol is simply a signaling protocol, which means that its purpose is only to get the two endpoints talking to each other, and not to deal with the media of the call (your voice). Rather, your voice is carried using another protocol called the Real-Time Transport Protocol (RTP; RFC 3550) to transfer media directly between the two endpoints.

Note

We use the term media to refer to the data transferred between endpoints and used to reconstruct your voice at the other end. It may also refer to music or prompts from the PBX.

In the world of SIP, we call our endpoints user agents, of which there are two types: client and server. The client is the endpoint that generates the request, and the server processes the request and generates a response. When an endpoint wishes to place a call to another endpoint (such as our softphone calling another softphone), we generate our request and send this to a SIP proxy.^[58] A proxy server will take the request, determine where the request is destined for, and forward it on. Once the two user agents have negotiated a successful call setup, the media is transported via the RTP protocol and sent directly ...