[OpenSIPS-Devel] WebRtc and opensips

[Răzvan Crainea]

Any chance you could provide a pcap capture of the call?

Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 08/20/2015 01:45 PM, Герман Чичикин wrote:
> Hi, All!
>
> Having trouble with testing webrtc with a rtpengin. I use opensips.cfg 
> from opensips.org (Documentation -> Tutorials -> WebSocket Transport 
> using OpenSIPS), opensips v2.1, clients - sipml5, jssip demo webphones 
> or my own script in various combinations. Clients register without 
> problems. When a client calls another, the callee joyfully calls. But 
> when I pick up the phone call is dropped, session ends and no any 
> media. The caller receives a "bye" from opensips.
> Anybody can tell in what direction to dig?
>
> Best regards, Herman.
>
>
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> Devel at lists.opensips.org
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