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Wiki HomeWhy 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- SIP Express Router: SER is used by many SIP providers standalone or in conjunction with Asterisk
- Vovida.org
- sipX from Sipfoundry is a native SIP proxy but also a complete SIP PBX
- OpenSER - scalable and robust SIP server with TLS support
See also
- Asterisk at large: Combining SER and Asterisk
- Asterisk cmd SIPRedirect: Sending out a 302 temporarily moved with Asterisk 1.2.x
- Realtime Integration Of Asterisk With OpenSER - step by step integration of Asterisk with OpenSER to provide media services (voicemail service with MWI, conferencing, IVR, ...) to large number of VoIP subscribers
- http://lists.digium.com/pipermail/asterisk-users/2007-July/191607.html: Attempt on asterisk-user Mailinglist to explain Asterisk vs. SIP-Proxy
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RTP channels with DTMF detection not through asterisk?
However, if the "t" option is added in the dial comment, like exten => 1001,1,Dial(SIP/1001,60,tr), huge RTP traffic flooring the Asterisk server and debases the voice quality very much.
The DTMF mode is out of band.(dtmfmode=info)
Is there solution to make RTP transfering directly between the 2 end points even when call transfer is required?
Or how can I do to improve the performance?
silence voice for 1 second through Asterisk
for your understanding call senario is like below.
IP Phone -->Proxy Server-->Asterisk-->Audiocodes trunk gateway -->PSTN call: 1second silence voico between IP phone and PSTN
IP Phone -->Proxy Server-->Audiocodes trunk gateway--> PSTN Call: it is OK. no 1second silence voice between IP Phone and PSTN
why I need Asterisk is because of CDR. So I test it above.
silence voice for 1 second through Asterisk
for your understanding call senario is like below.
IP Phone -->Proxy Server-->Asterisk-->Audiocodes trunk gateway -->PSTN call: 1second silence voico between IP phone and PSTN
IP Phone -->Proxy Server-->Audiocodes trunk gateway--> PSTN Call: it is OK. no 1second silence voice between IP Phone and PSTN
why I need Asterisk is because of CDR. So I test it above.
How many calls through Asterisk?
regards
Rolf (-at- rowi.net)
rtp channels not through asterisk?