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
rtp.log
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