<![CDATA[WebRTC - Does it work with the latest FreePBX?]]>Does anyone know how to get the WebRTC plugin working on the latest FreePBX? The instructions provided in the WebRTC Guide seem to be specific to IssabelPBX and don't line up completely with the options available on FreePBX.

I have the plugin installed and the extension configured as close as possible to the example in the guide. When I load up FOP 2, the extension appears to register, as shown in FOP2 as well as in the asterisk logs, however it is not able to send or receive calls in any way. It just says "Trying ##########" in the UI and nothing at all in the asterisk logs. I cannot even cancel the call in the WebRTC UI, it just stays kind of frozen until I close the FOP2 page entirely and reload.

]]>http://localhost:4567/topic/2898/webrtc-does-it-work-with-the-latest-freepbxRSS for NodeTue, 27 Oct 2026 04:22:24 GMTTue, 29 Sep 2020 17:48:44 GMT60<![CDATA[Reply to WebRTC - Does it work with the latest FreePBX? on Mon, 05 Oct 2020 11:48:05 GMT]]>

@Edi said:
do you have latest fop2

Yep, latest FOP2, latest FreePBX and latest versions of Chrome and Firefox.

]]>http://localhost:4567/post/12082http://localhost:4567/post/12082Mon, 05 Oct 2020 11:48:05 GMT<![CDATA[Reply to WebRTC - Does it work with the latest FreePBX? on Thu, 01 Oct 2020 19:10:33 GMT]]>do you have latest fop2

]]>http://localhost:4567/post/12081http://localhost:4567/post/12081Thu, 01 Oct 2020 19:10:33 GMT<![CDATA[Reply to WebRTC - Does it work with the latest FreePBX? on Wed, 30 Sep 2020 19:07:57 GMT]]>Final update for today:
It would seem as though it DOES work! Just not in Google Chrome. For the heck of it I just tried it in Firefox and it worked first try.
What would cause this to work in Firefox but not Chrome?

]]>http://localhost:4567/post/12080http://localhost:4567/post/12080Wed, 30 Sep 2020 19:07:57 GMT<![CDATA[Reply to WebRTC - Does it work with the latest FreePBX? on Wed, 30 Sep 2020 15:32:52 GMT]]>Another minor update:
I got calls going through now by delecting a different Media Encryption and transport setting, but one way audio only, whether it's in out outbound calls. I can hear everything via the WebRTC client, but the other side doesn't hear anything. RTP debug in asterisk shows the right IPs and ports, and everything is being passed by the firewall. My browser is configured to allow microphone access as well.

The settings within FreePBX I applied to the extension I am testing with are as follows, everything else is default settings:
Transport: 0.0.0.0-wss
Enable AVPF - Yes
Enable ICE Support - Yes
Enable rtcp Mux - Yes
Enable WebRTC defaults - Yes
Media Encryption - DTLS-SRTP (not recommended)
Enable DTLS - Yes
Use Certificate - I selected the real, not self-signed SSL cert configured for my server

So what am I missing?

]]>http://localhost:4567/post/12079http://localhost:4567/post/12079Wed, 30 Sep 2020 15:32:52 GMT<![CDATA[Reply to WebRTC - Does it work with the latest FreePBX? on Wed, 30 Sep 2020 12:19:47 GMT]]>Minor update:
I was loading FOP2 in HTTP but WebRTC needs HTTPS, so I fixed up the ssl configuration in fop2.cfg and reloaded in HTTPS. Now the WebRTC phone behaves slightly differently. Instead of getting stuck on "Trying ##########" it just immediately flips to Call Ended when trying to dial a number.
When receiving a call it rings, I see the callerID, but when answering it immediately flips to Call Ended as well.
So something is still not right... Any ideas?

]]>http://localhost:4567/post/12078http://localhost:4567/post/12078Wed, 30 Sep 2020 12:19:47 GMT