[SR-Users] [rtpengine] No media from WebRTC UA

[Alex Balashov]

Hello Alexey,

I am uncertain as to whether rtpengine_offer()/answer() support 
pseudovariable arguments. But if they do, you'll need to wrap them in a 
string literal:

    rtpengine_offer("$var(rtpp_flags)");

If they don't support PV arguments at all, you may be stuck with having 
to provide a static argument:

    rtpengine_offer("trust-address symmetric replace-origin 
replace-session-connection ICE=force RTP/SAVPF");

-- Alex

On 05/16/2014 02:45 AM, Alexey Rybalko wrote:

> Hello!
>
> During a call from classical SIP softphone to WebRTC there's no media
> from the browser (Mozilla, the same result is for Chrome). In case of a
> call from the browser to the softphone there's media flow from both sides.
>
> The snippets from kamailio.cfg related to the problem case
> (SIP-->WebRTC)  are below.
>
> OFFER:
> $var(rtpp_flags) = "trust-address symmetric replace-origin
> replace-session-connection";
> $var(rtpp_flags) = $var(rtpp_flags) + " ICE=force";
> $var(rtpp_flags) = $var(rtpp_flags) + " RTP/SAVPF";
> rtpengine_offer($var(rtpp_flags));
>
> ANSWER:
> $var(rtpp_flags) = "trust-address symmetric replace-origin
> replace-session-connection";
> $var(rtpp_flags) = $var(rtpp_flags) + " ICE=remove";
> $var(rtpp_flags) = $var(rtpp_flags) + " RTP/AVP";
>
> rtp.log is attached.
>
> Any help on this issue would be very appreciated.
>
>
>
> with best regards,
> Alexey
>
>
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-- 
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United States
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/