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My main reason not to upgrade to Ast 13 is because I'm afraid of
losing functionality as there are certain functions
deprecated/replaced. This can also cause headache :-)<br>
<br>
I will do so if there is no other option.<br>
<br>
But still, I don't see why Ast 13 would differ so much in this case
? If ICE and NAT is working (not causing problems) why should Ast 13
bring me audio and Ast 12 don't ??<br>
<br>
<br>
<br>
I indeed use SIPML5 demo as quick test-case. So do many tutorials on
the web.<br>
<br>
Self-signed certificates should be OK as long as they are imported
in the browser. Never knew this could cause audio problems ?<br>
<br>
<br>
<br>
<br>
Kind regards.<br>
<br>
<br>
<br>
<div class="moz-cite-prefix">On 11-08-16 16:25, Jonathan H wrote:<br>
</div>
<blockquote
cite="mid:CAEebyNXSayymqT2g0O=drwDt=35hjFDcMVsX21TRsqGMKEQCog@mail.gmail.com"
type="cite">
<div dir="ltr">I'm genuinely fascinated why you are insisting on
using a version of Asterisk almost 3 years old, for which EOL
support ended last year.
<div><br>
</div>
<div>Is there any particular reason you cannot or will not use
the current version as others have suggested?</div>
<div><br>
</div>
<div>Also, I see you are using Doubango and WebRTC, but in the
logs, I see WS and WSS.</div>
<div><br>
</div>
<div>You NEED to be using 100% WSS otherwise you've not got a
hope in hell of anything working with WEBRTC.</div>
<div>Check the console of the web browser you are trying to make
the call from (CTRL-SHIFT-I in Chrome on Windows, for
example).</div>
<div><br>
Also, you'll need to be using valid certificates - self-signed
certificates won't work for any current implementation of
WebRTC that I know of, certainly not if anything involves
current versions of Chrome or Firefox. That said, LetsEncrypt
certs work fine for this, so no need to spend out on one.</div>
<div><br>
</div>
<div>Switch to Asterisk 13.10 and save yourself a whole lotta
headache.</div>
<div>
<div class="gmail_extra"><br>
<div class="gmail_quote">On 11 August 2016 at 15:09, Jonas
Kellens <span dir="ltr"><<a moz-do-not-send="true"
href="mailto:jonas.kellens@telenet.be" target="_blank">jonas.kellens@telenet.be</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex">Hello<br>
<br>
Using Asterisk 12.8.2.<br>
<div>
<div><br>
</div>
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</blockquote>
<div><br>
</div>
<div> </div>
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex">
<div>
<div>
On 10-08-16 22:03, Matt Fredrickson wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex">
My suggestion is to verify and debug against
Asterisk 13 first, and<br>
then you can try backing down versions, rather
than reverse. WebRTC<br>
is a rapidly moving target, and has required
ongoing changes that may<br>
not have made it into older and feature frozen
versions of Asterisk.</blockquote>
</div>
</div>
</blockquote>
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