Yes it works in those versions which is why I said I would “announce” it when it’s ready. Asterisk 11.11 and 12.4.0 support it but only with DTLS enabled on said extensions and the right settings for either chan_sip or pjsip. I have completed the work on WebRTC for FreePBX for both chan_sip and pjsip but it is not officially released few reasons:
First being that the work I’ve done for WebRTC will only be available for FreePBX 12, I will not be downgrading it. That’s because I had to write a certificate manager for FreePBX because WebRTC now requires encrypted calls to be placed with certificates in place, which is great if you are using Firefox because now both firefox and chrome work (with stipulations), but again I only wrote that for FreePBX 12.
Secondly I wrote a whole new WebRTC interface for UCP which is only in 12, I am no longer working on the ARI counterpart.
Thirdly, what users (like yourself) don’t seem to understand is how bug laden WebRTC really is. People think it will just magically work, but I’m here to tell you that isn’t true.
First off, if you are using Chrome you won’t be able to make outbound calls outside of your pbx if you also want the ability to place calls on hold (see: 3530 - webrtc - Web-based real-time communication - Monorail)
If you decide to use FireFox instead then you won’t be able to place calls on hold at all (see: Add SIP hold / retrieve mechanics to the Session · Issue #46 · versatica/JsSIP · GitHub and 840728 - Implement advanced RTCPeerConnection session handling)
I am up in the air about what to do. If I release WebRTC to the general public I will be faced with these questions daily whereas people won’t understand how to use it. Otherwise I just remove the hold functionality until it works correctly in both browsers but that means you can’t place a call on hold, transfer or do anything cool. Perhaps I will release both but I am not sure. For certain I will remove the hold functionality from Firefox, if I remove the hold functionality from Chrome as well then the bug (mentioned above) won’t be a problem.
I honestly believe that the people who use or talk about WebRTC (the SIP spec) online right now aren’t trying to do fancy things like place calls on hold (yes that is a “fancy” thing in WebRTC world) or call waiting/swapping, they are just making a simple interface ring and saying “oh look it works, cool” or providing an interface for two people to talk to each other over a webpage without a pbx in the middle, not the things you’d expect in an enterprise phone system, but these are just my opinions.
Work on the FreePBX version of webrtc is ongoing, you can see the progress here: GitHub - FreePBX/webrtc: Module of FreePBX (WebRTC Phone) :: The WebRTC Module allows an Administrator to enable a "WebRTC phone" that can be attached to a user's extension which they can connect to through FreePBX User Control Panel, this WebRTC phone will then receive phone calls at the same time as the users extension using user and device mode behind the scenes. If you have User and Device Mode enabled any extension you enable the WebRTC Phone a duplicate extension of 99XXXX will be created (where XXXX is the original extension number), when the user then views the web interface of the WebRTC phone they will be connected to device 99XXXX which will receive calls from the original extension
My question to you is thus: What do you expect from FreePBX in terms of WebRTC? Are you building your own interface or using ours?
Here’s what it looks like in UCP right now (and yes it works, these are not mockups)
