Asterisk SIP not-proxy

Why is Asterisk not a SIP Proxy?

Asterisk is *not* a SIP proxy. A SIP proxy handles call control on behalf of other user agents (UA) and usually does not maintain state during a call and therefore is never the endpoint of a call.

Asterisk, as a server, is a SIP registrar and location server and also acts as a useragent endpoint (softphone).

If it is 'controlling' or relaying a call from a SIP phone to another SIP phone, it simply acts as an endpoint UA to the originating call leg and then creates a new call to the receiving phone. Therefore, it stays "in the middle of the call," maintaining state and controlling, and optionally bridging, each remote endpoint. The audio channels (RTP) may go directly from phone to phone or may go through Asterisk's media bridge.

Asterisk can thus be described best as a "back-to-back user agent" (B2BUA), which is also consistent with the use of the term "PBX". Because of this architecture, fairly simple SIP functions, such as REFER (transfer) involve more aspects of the Asterisk core. On the other hand, the architecture provides additional power and flexibility, because each call leg can just as easily be replaced with a different technology channel (ZAP, H323, MGCP, etc) and, thus, Asterisk becomes a powerful media gateway.

Open Source SIP Proxys

Excellent open source SIP Proxys are available on the Internet. Check

See also

Created by: oej, Last modification: Thu 18 of Nov, 2010 (15:28 UTC) by rwolpov
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