asterisk users - Feb 2016 - Asterisk 13 and WebRTC. Is wiki page still valid ?

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) ?
2. Do you have any pointer for WebRTC with Asterisk 13 and PJSIP ?

Regards

[1] https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5
[2] https://www.doubango.org/sipml5/
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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 
> <http://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 
> <http://123.123.123.123:8088/ws> with wss://123.123.123.123:8088/ws 
> <http://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."

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.
>
> 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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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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