Asterisk Error Log :
Show in the asterisk CLI >>
"WARNING[2982][C-00000004]: res_rtp_asterisk.c:773 ast_rtp_ice_start: No RTCP candidates; skipping ICE checklist (0x7f6db831fbe8)
-- Channel SIP/100-0000000a joined 'simple_bridge' basic-bridge <745e6ab1-d793-4894-bf06-823a0a9a78bc>
-- Channel SIP/600-00000009 joined 'simple_bridge' basic-bridge <745e6ab1-d793-4894-bf06-823a0a9a78bc>
[2017-06-13 04:42:21] NOTICE[1766]: chan_sip.c:29402 check_rtp_timeout: Disconnecting call 'SIP/600-00000009' for lack of RTP activity in 31 seconds
-- Channel SIP/600-00000009 left 'simple_bridge' basic-bridge <745e6ab1-d793-4894-bf06-823a0a9a78bc>
== Spawn extension (macro-dial-one, s, 51) exited non-zero on 'SIP/600-00000009' in macro 'dial-one'
== Spawn extension (macro-exten-vm, s, 16) exited non-zero on 'SIP/600-00000009' in macro 'exten-vm'
== Spawn extension (ext-local, 100, 2) exited non-zero on 'SIP/600-00000009'
-- Executing [h@ext-local:1] Macro("SIP/600-00000009", "hangupcall,") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/600-00000009", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] ExecIf("SIP/600-00000009", "0?Set(CDR(recordingfile)=)") in new stack
-- Executing [s@macro-hangupcall:4] Hangup("SIP/600-00000009", "") in new stack" warning appeared and hangup .
Solution :
When using WebRTC you have to ensure you are using the most recent version of Asterisk and also keep up to date on any changes in the browser. In this case the browser is no longer doing optional RTCP-MUX resulting in problems. We have a blog post[1] which talks about this and how recent versions of Asterisk added support.
rtcp-mux in Asterisk
To get around this problem, the Asterisk team decided to add support for rtcp-mux into Asterisk before it became too late. I added support for rtcp-mux for chan_pjsip, and Sean Bright added rtcp-mux for chan_sip. The feature is available starting in Asterisk 13.15.0 and Asterisk 14.4.0. For those of you still on older versions, you may want to start upgrading soon if you plan to interoperate with Chrome.
chan_pjsip
For chan_pjsip, you can enable rtcp-mux on an endpoint by setting rtcp_mux=yes
chan_sip
For chan_sip, you can enable rtcp-mux by setting rtcp_mux=yes on a peer, user or friend. You also have the option of setting rtcp_mux=yes in the general section to apply it globally.
Detail :
http://blogs.asterisk.org/2017/04/26/rtcp-mux-webrtc/
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