[asterisk-users] Asterisk 13 and WebRTC. Is wiki page still valid ?

[Olivier]

Thank you much for yor reply.

2016-02-18 13:30 GMT+01:00 Simon Hohberg <simon.hohberg at mcs-datalabs.com>:

> Hi Oliver,
>
> On 02/18/2016 12:10 PM, Olivier wrote:
>
> Hello,
>
> I'm trying to have my first calls with WebRTC.
> My server has asterisk 13.7.0.
>
> I'm following the instructions from the wiki [1].
> So I'm using [2] live demo from a Chrome navigator (v48) on Debian Jessie
> station.
>
> Whenever I type something like ws://123.123.123.123:8088/ws in Expert
> Mode form (see [1]), I'm getting this error :
> *2:SecurityError: Failed to construct 'WebSocket': An insecure WebSocket
> connection may not be initiated from a page loaded over HTTPS.*
> If I replace ws://123.123.123.123:8088/ws with wss://
> 123.123.123.123:8088/ws, this error message becomes with
> *Disconnected: Failed to connet to the server*
>
> My questions are:
> 1. Is wss now required by sipml5 live demo (implying wiki page is not
> up-to-date) ?
>
> Yes, like the error says, you have to use wss on pages served via https.
> Furthermore, Chrome requires the use of https when you want to use
> getUserMedia.
> See here:
> https://developers.google.com/web/updates/2015/10/chrome-47-webrtc?hl=en.
> It says: " Starting with Chrome 47, getUserMedia() requests are only
> allowed from secure origins: HTTPS or localhost."
>

Is it implied here that both HTTPS and WSS must also come from the same
server (Same Origin Policy) ?

Then, can I also install my own WebRTC demo page on my own private
Asterisk server and access this demo page through HTTPS ?
If I'm not mistaken, this should fulfill all requirements.


>
> The solution for development is, to host the webrtc client locally, so
> that you load the page from localhost. In that case getUserMedia is allowed
> with http, too (as the quote says). That means you have to download the
> dubango client and run a webserver on your dev machine.
>
> 2. Do you have any pointer for WebRTC with Asterisk 13 and PJSIP ?
>
> Unfortunately, there is not much documentation about this, as far as I can
> tell.
>

I didn't find any.

>
>
> Regards
>
> [1]
> https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5
> [2] https://www.doubango.org/sipml5/
>
>
>
>
> Regards,
>
> Simon
>
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