Asterisk 16 - WebRTC

Asterisk Asterisk WebRTC
1

Hi there

I configured a Asterisk with this instructions, but I have no audio and get this console error:

XXXXXX*CLI> originate PJSIP/100 application MusicOnHold ,10
[Feb  1 08:45:42]     -- Called 100
[Feb  1 08:45:42]     -- PJSIP/100-00000003 is ringing
[Feb  1 08:45:42]     -- PJSIP/100-00000003 is ringing
[Feb  1 08:45:42]     -- PJSIP/100-00000003 answered
[Feb  1 08:45:42]        > Launching MusicOnHold(,10) on PJSIP/100-00000003
[Feb  1 08:45:42]     -- Started music on hold, class 'default', on channel 'PJSIP/100-00000003'
[Feb  1 08:45:42]        > 0x7fd48401c770 -- Strict RTP learning after remote address set to: 192.168.XXX.XXX:44171
[Feb  1 08:45:42] ERROR[8159]: pjproject: <?>: 	   icess0x7fd484033d78 ......Error sending STUN request: Invalid argument
[Feb  1 08:45:42]        > 0x7fd48401c770 -- Strict RTP learning after ICE completion
[Feb  1 08:45:52]     -- Removed contact 'sip:qo840heo@183.183.XXX.XXX:40892;transport=ws' from AOR '100' due to transport shutdown
[Feb  1 08:45:52]   == Contact 100/sip:qo840heo@183.183.XXX.XXX:40892;transport=ws has been deleted
[Feb  1 08:45:52]     -- Stopped music on hold on PJSIP/100-00000003

What’s wrong?

2

I have a blog post[1] that talks about the different parts to investigate to see what may be going on, I’d suggest taking a look at it. Ultimately, though, WebRTC involves a lot of moving pieces so when something doesn’t work - it can involve Asterisk, the different technologies, the browser (a browser upgrade can break WebRTC).

[1] https://blogs.asterisk.org/2018/03/21/webrtc-asterisk-goes-wrong/

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