login | register
Sat 04 of Jul, 2009 [01:27 UTC]

voip-info.org

Discuss [5] History

Asterisk SIP not-proxy

Created by: oej,Last modification on Thu 19 of Jun, 2008 [23:42 UTC] by bhakimi

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




Comments

Comments Filter
222

333RTP channels with DTMF detection not through asterisk?

by hhwang, Tuesday 18 of July, 2006 [10:16:56 UTC]
In my case, Asterisk is used to acheive call transfer and call parking.
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?
222

333silence voice for 1 second through Asterisk

by pyoungmu, Monday 17 of July, 2006 [13:04:19 UTC]
I interoperate with Asterisk and Audiocodes trunk(pstn)gateway. I faced problem with 1 second silence voice IP to PSTN calls. after 1 second everything is fine. What do I check Asterisk? please help me out.
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.


222

333silence voice for 1 second through Asterisk

by pyoungmu, Monday 17 of July, 2006 [13:03:49 UTC]
I interoperate with Asterisk and Audiocodes trunk(pstn)gateway. I faced problem with 1 second silence voice IP to PSTN calls. after 1 second everything is fine. What do I check Asterisk? please help me out.
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.


222

333How many calls through Asterisk?

by rowitech, Wednesday 09 of March, 2005 [14:44:29 UTC]
I understand that Asterisk is not acting like a SIP proxy. But due to NAT I cannot let my phones all around the globe streaming the voice directly to each other (may I be wrong?). So wouldn't it be a good idea just to use Asterisk clusters instead of SER especially if you have much features which need to go through asterisk before the call?

regards
Rolf (-at- rowi.net)
222

333rtp channels not through asterisk?

by , Thursday 23 of December, 2004 [22:38:53 UTC]
at the moment all my media is passing through asterisk. what should I do to make audi channels to go directly from phone to phone?