[asterisk-users] webrtc no audio

[Joshua Colp]

Marek Cervenka wrote:
> hello,
>
> i'm facing strange problem
>
> asterisk13.5 + chan_sip wss transport + SIPML5 1.5.230
> person1 to person3 are behind different NATs
> audio devices double checked
>
> call from person1(chrome) to person2(chrome) works
> call from person1(chrome) to person 3(chrome) - no audio on both side
> (RTP flowing only in one direction)
> call from person2(chrome) to person 3(chrome) - no audio on both side
> (RTP flowing only in one direction)
> BUT
> call from person2(chrome) to person 3(Jitsi sip client) - works!
>
> any tips howto find the problem?

You would need to look at the ICE negotiation to see if it tried and 
failed. After that would be looking at the DTLS negotiation. Asterisk 
console output could provide some information.

-- 
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