To no avail, still no sound flowing even though rtcpMuxPolicy:“negotiate” is set.
FireFox works just fine.
WebRTC Chrome 57.0.2987.133 (64-bit), no sound in IVR using Doubango SIPml5 v2.1.3 Asterisk 13.15.0.
WebRTC FireFox ESR 52.0(64-bit), sound OK in IVR using Doubango SIPml5 v2.1.3 Asterisk 13.15.0.
Doubango SIPml5 v2.1.3 sets rtcpMuxPolicy:“negotiate”, so that should have fixed the problem, but still no sound in Chrome.
This can be fixed in Asterisk by enabling RTCP MUX. It’s available as of 13.15.0:
- The RTP layer of Asterisk now has support for RFC 5761: "Multiplexing RTP
Data and Control Packets on a Single Port." For the PJSIP channel driver,
chan_pjsip, you can set "rtcp_mux = yes" on a PJSIP endpoint in pjsip.conf
to enable the feature. For chan_sip you can set "rtcp_mux = yes" either
globally or on a per-peer basis in sip.conf.
The sample configuration file does not have the complete option documentation, there’s a few things missing from there. We’ll add a link to the wiki in the future which is automatically kept up to date.
The RTCP MUX feature itself is available in 13.15.0 on both chan_sip and chan_pjsip. It can be enabled by setting “rtcp_mux” to yes on either a PJSIP endpoint or SIP peer. If that isn’t working then you’ll need to be more specific about what exactly is happening and what you are seeing.