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

[Alexey Rybalko]

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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