webRTC with sipml5 using asterisk

When I call using webRTC from the browser.
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 .
What can I do? I used freepbx version 13 and asterisk 13.12.
Please help me.!!!

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.

[1] http://blogs.asterisk.org/2017/04/26/rtcp-mux-webrtc/

Thank you so much for your reply jcolp.

Now,I’m using the asterisk 13.12.1 .How can I solve this problem?Please help me!!!

You need to upgrade to at least 13.15, preferably 13.16.

Thanx for your reply pay123!!!

My WebRTC seems to be everything OK when I test with Chrome!But no audio on both side.How can I do for that.
Help me please!!!

WebRTC is still changing and I have bumped into various problems due to changes in Chrome/FireFox.
It is so annoying that after 3 years of using WebRTC I am still not using it in a production environment.

Last time I had no sound, I had to enable RTCP MUX in every peer connection as follows.

Don’t forget to check if there are error messages in the browser console log.
In Chrome you can open the development tools by pressing SHIFT+CTRL+I

In the developer console,

Failed to set ICE servers:SyntaxError: Unexpected token :
tsk_utils_log_error @ tsk_utils.js?svn=241:128

Not implemented
tsk_utils_log_error @tsk_utils.js?svn=241:128

Failed to find transport
tsk_utils_log_error @tsk_utils.js?svn=241:128
tsip_transport_layer.send @tsip_transport_layer.js?svn=241:54

Failed to set ICE servers indicates you have set something incorrectly.
When this fails in a NATted environment, the RTP data will be send to the incorrect location, resulting in no sound.

Are you using the sipml5 demo or your own modified code?
I recommend you use the sipml5 demo and make as litle changes as possible, to avoid misconfiguration.
At least try by not setting the ICE servers as this will result in the defaults to be used and those should be fine.
Programatically this would be set to ‘[]’ (empty square brackets)