[SR-Users] rtcp-mux is missing in the OK for WebRTC-Calls

Richard Fuchs rfuchs at sipwise.com
Fri Jul 10 16:26:38 CEST 2020


On 10/07/2020 04.59, Benjamin Flügel | vio:networks wrote:
> Hey guys,
>
> I'm trying to configure a Kamailio to work with a browser softphone based on SIPJS using WebRTC.
> So far it works great on Firefox but have a specific problem with chrome, when I want to make call from the softphone to another extension.
> After anwsering the call Chrome/the softphone sends a BYE immediately, because this line "a=rtcp-mux" is missing in the OK.
>
> The Kamailio is a proxy. Behind the Kamailio there is an Asterisk, which is responsible for the pbx-features.
>
>
> Those are my rtpengine Flags for the Invite:
>
> rtpengine_manage: replace-origin replace-session-connection trust-address via-branch=extra rtcp-mux-demux DTLS=off SDES-on ICE=remove RTP/AVP
>
>
> And those are the flags for the response, in this case the OK:
>
> rtpengine_manage: replace-origin replace-session-connection rtcp-mux-offer rtcp-mux-accept generate-mid DTLS=off SDES-on ICE=force RTP/SAVPF direction=internal direction=external loop-protect
>
> It seems that the Kamailio ignores ths "rtcp-mux-offer rtcp-mux-accept" in the response. Can you help me get it to work?

You don't need to provide some of these options in your answer (neither 
rtcp-mux nor the direction nor the protocol - the direction should be 
specified in the offer). You should also provide the same via-branch 
option in your answer as you did in the offer, especially if this is a 
branched offer. In particular if this is a branched offer and the 
via-branches weren't given correctly, then that would explain the 
missing rtcp-mux.

Cheers




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