Chirag Desai wrote:
I have two accounts on Asterisk 13. One uses chan_sip and the other pjsip.
In my snom 760 the setup for these two accounts is identical.
When I call echo test from the account using chan_sip audio comes
through fine.
When I call echo test from the account using pjsip there is no audio.
With rtp set debug on, I can see that audio is being sent to the snom's
internal IP 192.168.0.x
I can add a stun server in the config for this account and RTP flows to
the Public IP and I get audio.
I was wondering why there is a difference between pjsip and chan_sip so
that one works without stun and the other requires it. Does anybody
know why? Maybe my settings are off in pjsip.
There should be nothing different, except for how you configure things.
What is the full PJSIP configuration? What is the environment where
Asterisk is running? Is ICE actually in use on the other side? What is
the full SIP trace?
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
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