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?