S2E2: WebRTC In The Cloud 

Thank you for your support. Getting back into the swing of things, so more video's coming soon!

Glad it was helpful!

My pleasure 😊

You're welcome!

My pleasure 😊

I would check the Music on hold with another phone, best is with a hardware phone. It can often be because the transcoding doesn't work. it would be opus to say ulaw or GSM. github.com/InnovateAsterisk/Browser-Phone/issues

@@InnovateAsterisk Thanks

Yes, the actual HTML and javascript can be hosted on any secure server, even here: www.innovateasterisk.com/phone/ for free

Yes, actually I have been starting up a little business that will essentially offer Browser Phone as a SaaS. Its called: Siperb (www.siperb.com/). Part of the roadmap for the development will be to bring the code over to a Node.js module. We will also release as mobile (ios and Android) and web. It will still be WebRTC based, and compatible with ReactNative. The future for the Service will be more around Storage, Transcoding, Analysys, Translation, and AI or Machine Learning.

Yes, this video shows the cloud way - but it would be possible to do the same on a local network.

The web socket connection should be persistent, and should not disconnect unless there is a network issue. The reason the peer becomes unreachable just after registration, is because Asterisk sends an OPTIONS packet a few seconds after registration (if the peer is set to qualify). If the option packet fails to reply (from the browser), the peer is marked as unreachable. In most cases this is because the contact is not re-written. You can confirm this by debugging sip messages in the asterisk CLI. It's just been too many years since I have worked with chan_sip to remember the solution. I would recommend pjsip - i have a video for WebRTC on PJSIP

The first place to check for issues is in the Developer Tools - and then if you still get stuck, hop over to the GitHub issues page, and submit an issue.

DNS its a fairly large topic, but the basics are that SSL certificates can only be issued to domains or sub-domains, and domains can only be stored in something called a Domain Name Server (DNS). This is software that says 'ok you entered www.google.com, this refers to the following server ip address.' The same goes for the location of the Asterisk server; when the user enters www.innovateasterisk.com its pointing the user to my server ip address. Now the important thing here is that only I can make changes to the `innovateasterisk.com` domain, one of the entries I have added was `www`, and I can add more - by this I can say i'm in control of the domain. Nobody else can add these entires, much the same as only you are in control of your own email, facebook account, etc, etc. As you saw in the video at a point i had to rush off, and make a change to the DNS - this proves that I am in fact in control (can make changes) to this domain.

I'm not familiar with Freepbx, but the Asterisk 13 using chan_sip is covered in this video: ru-vid.com/video/%D0%B2%D0%B8%D0%B4%D0%B5%D0%BE-mS28vfT8wJ8.html

Please post technical questions to the Github issues page: github.com/InnovateAsterisk/Browser-Phone/issues