September 2005
Intermediate to advanced
408 pages
16h 20m
English
Content preview from Asterisk: The Future of TelephonyBecome an O’Reilly member and get unlimited access to this title plus top books and audiobooks from O’Reilly and nearly 200 top publishers, thousands of courses curated by job role, 150+ live events each month,
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Configuring SIP
The Session Initiation Protocol (SIP), often used in VoIP phones (either hard phones or soft phones), takes care of the setup and teardown of calls, along with any renegotiations during a call. Basically, it helps two endpoints talk to each other (if possible, directly to each other). SIP does not carry media; rather, it uses the Real-time Transport Protocol (RTP) to transfer the media[34] directly between phone A and phone B once the call has been set up.
SIP and RTP
SIP is an application-layer signaling protocol that uses the well-known port 5060 for communications. SIP can be transported with either the UDP or TCP transport-layer protocols. Asterisk does not currently have a TCP implementation for transporting SIP messages, but it is possible that future versions may support it (and patches to the code base are gladly accepted). SIP is used to “establish, modify, and terminate multimedia sessions such as Internet telephony calls.”[35] SIP does not transport media between endpoints.
RTP is used to transmit media (i.e., voice) between endpoints. RTP uses high-numbered, unprivileged ports in Asterisk (10,000 through 20,000, by default).
A common topology to illustrate SIP and RTP, commonly referred to as the “SIP trapezoid,” is shown in Figure 4-2. When Alice wants to call Bob, Alice’s phone contacts her proxy server, and the proxy tries to find Bob (often connecting through his proxy). Once the phones have started the call, they communicate directly with each other ...
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