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According to this announcement: "As of the most recent Chrome Canary build, the default RTCP multiplexing policy is "require", instead of "negotiate". This will affect the next Chrome release, M57."

I'm using sipml5 API to make webrtc calls (Back end is Asterisk). When I tried to make a call from webrtc extension I got this error in chrome console and there was no audio.

onSetRemoteDescriptionError

DOMException: Failed to set remote answer sdp: Session error code: ERROR_CONTENT. 
Session error description: rtcpMuxPolicy is 'require', but media description does not contain 'a=rtcp-mux'..

So how can I set rtcpMuxPolicy to "negotiate" in sipml5?

  • My chrome version: 58.0.3026.3 dev (64-bit)
  • SIPML5 API version: 2.0.3
  • Asterisk version: 13.11.0
Sibin John Mattappallil
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1 Answers1

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After digging a long time in internet I found following points,

Quoted from asterisk forum: https://issues.asterisk.org/jira/browse/ASTERISK-26732

Chrome 57 has a breaking change when it comes to interop with WebRTC gateways. They've changed their previous "negotiate", to "require" when it comes to rtcp-mux. Asterisk, as I understand it does not have rtcp multiplexing and so will break when it comes to compatibility with WebRTC across all versions of Asterisk that supports WebRTC.

Reference to rtcpMuxPolicy in chrome: https://www.chromestatus.com/feature/5654810086866944

The rtcpMuxPolicy is used by the application to specify its preferred policy regarding use of RTP/RTCP multiplexing. When the policy is "negotiate", the ICE candidates for both RTP and RTCP will be gathered. If the remote-endpoint is capable of multiplexing RTCP, multiplex RTCP on the RTP candidates. If it is not, use both the RTP and RTCP candidates separately.

According to the google group forum - https://groups.google.com/forum/#!topic/discuss-webrtc/eM57DEy89MY

As of the most recent Chrome Canary build, the default RTCP multiplexing policy is "require", instead of "negotiate". This will affect the next Chrome release, M57. This means that offer/answer negotiation with an endpoint that doesn't support RTCP multiplexing will fail, resulting in the error: "ERROR_CONTENT. Session error description: Failed to setup RTCP mux filter." We can probably make this error more descriptive, but the bottom line is that setRemoteDescription will fail if the SDP does not contain "a=rtcp-mux". For any application that doesn't yet support RTCP multiplexing, you can get the old behavior by explicitly setting the RTCRtpMuxPolicy to "negotiate" in the RTCConfiguration. For example: pc = new RTCPeerConnection({rtcpMuxPolicy: "negotiate"})

In short,

  • In previous versions of chrome rtcp multiplxing set to 'negotiate'
  • Starting from version 57 chrome changed rtcp multiplxing to 'require'
  • Asterisk, as I understand it does not support rtcp multiplexing.
  • webrtc allow RTCRtpMuxPolicy flag options "negotiate" and "require"
  • In Sipml5 API 2.0.3 , as I understood there is no option to set RTCRtpMuxPolicy.

Solution: I updated sipml5 api to 2.1.3 from 2.0.3. Now error changed to a warning.

[Deprecation] The rtcpMuxPolicy option is being considered for removal and may be removed no earlier than M60, around August 2017. If you depend on it, please see https://www.chromestatus.com/features/5654810086866944 for more details.

Now everything is working fine.

Sibin John Mattappallil
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