Hi all! Some time ago Chromium browser sets rtcpMuxPolicy: required by default (soon on Chrome 58) It means that webRTC based clients not accepts a=rtcp:31757 And uses for RTP and RTCP multiplexing at one port
Main trouble that i found: calls between original SIP client and webRTC client (SIP client is initiator of call) When sip client sends invite it has a=rtcp:33445 Means it wants 2 different prots for RTCP and RTP As expected for this case webRTC client says 488 Not accessible here instead of 200 resonse I suppose rtpengine module should hept to handle it but i can not find any key how to do it I added form rtpengine_manage() rtcp-mux-offer and rtcp-mux-accept but it only adds "a=rtcp-mux" But not removes a=rtcp and ice cadidate prepeared for it. Suppose removing a=rtcp:12345 will gives just an issue for RTP session. Does rtpengine module have some keys for resole this issue?
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