<div dir="ltr">Thank you much for yor reply.<br><div class="gmail_extra"><br><div class="gmail_quote">2016-02-18 13:30 GMT+01:00 Simon Hohberg <span dir="ltr"><<a href="mailto:simon.hohberg@mcs-datalabs.com" target="_blank">simon.hohberg@mcs-datalabs.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
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Hi Oliver,<span class=""><br>
<br>
<div>On 02/18/2016 12:10 PM, Olivier wrote:<br>
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<div>Hello,<br>
<br>
</div>
I'm trying to have my first calls with WebRTC.<br>
</div>
My server has asterisk 13.7.0.<br>
<br>
</div>
I'm following the instructions from the wiki [1].<br>
</div>
So I'm using [2] live demo from a Chrome navigator (v48) on
Debian Jessie station.<br>
<br>
</div>
Whenever I type something like <a>ws://</a><a href="http://123.123.123.123:8088/ws" target="_blank">123.123.123.123:8088/ws</a>
in Expert Mode form (see [1]), I'm getting this error :<br>
</div>
<b>2:SecurityError:
Failed to construct 'WebSocket': An insecure WebSocket
connection may not be initiated from a page loaded over HTTPS.</b><br>
<div>If I replace <a>ws://</a><a href="http://123.123.123.123:8088/ws" target="_blank">123.123.123.123:8088/ws</a>
with <a>wss://</a><a href="http://123.123.123.123:8088/ws" target="_blank">123.123.123.123:8088/ws</a>,
this error message becomes with<br>
<label align="center"><i>Disconnected:<span> </span><b>Failed to connet to the server</b></i></label><br>
<br>
My questions are:<br>
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<div>1. Is wss now required by sipml5 live demo (implying wiki
page is not up-to-date) ?<br>
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</blockquote></span>
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. <br>
See here:
<a href="https://developers.google.com/web/updates/2015/10/chrome-47-webrtc?hl=en" target="_blank">https://developers.google.com/web/updates/2015/10/chrome-47-webrtc?hl=en</a>.
It says: "
Starting with Chrome 47, getUserMedia() requests are only allowed
from secure origins: HTTPS or localhost."<br></div></blockquote><div><br></div><div>Is it implied here that both HTTPS and WSS must also come from the same server (Same Origin Policy) ?<br></div><div><br>Then, can I also install my own WebRTC demo page on my own private Asterisk server and access this demo page through HTTPS ?<br></div><div>If I'm not mistaken, this should fulfill all requirements.<br></div><div> <br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div bgcolor="#FFFFFF" text="#000000">
<br>
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.<span class=""><br>
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<div>2. Do you have any pointer for WebRTC with Asterisk 13 and
PJSIP ?<br>
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</blockquote></span>
Unfortunately, there is not much documentation about this, as far as
I can tell.<span class=""><br></span></div></blockquote><div><br></div><div>I didn't find any. <br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div bgcolor="#FFFFFF" text="#000000"><span class="">
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<div><br>
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<div>Regards<br>
</div>
<div>
<div>
<div><br>
[1] <a href="https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5" target="_blank">https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5</a><br>
[2] <a href="https://www.doubango.org/sipml5/" target="_blank">https://www.doubango.org/sipml5/</a><br>
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Regards,<br>
<br>
Simon<br>
</div>
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