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.