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Hello<br>
<br>
<br>
setting "nat=no" or omitting "nat=" in peer definition does not help
either. Still no audio.<br>
<br>
Why do you think this is a NAT issue ? IP and port information in
SDP-body is correct.<br>
<br>
<br>
<br>
<br>
Kind regards.<br>
<br>
<br>
<div class="moz-cite-prefix">On 12-08-16 09:25, Антон Сацкий wrote:<br>
</div>
<blockquote
cite="mid:CAFgS45vP0sAZ3PHVFX=T2TFajugw03GQJwPnujgRttSywAerbQ@mail.gmail.com"
type="cite">
<p dir="ltr">Try delete nat from 770000wrtc settings ice should do
the same</p>
<div class="gmail_extra"><br>
<div class="gmail_quote">On Aug 11, 2016 10:00 PM, "Jonas
Kellens" <<a moz-do-not-send="true"
href="mailto:jonas.kellens@telenet.be" target="_blank">jonas.kellens@telenet.be</a>>
wrote:<br type="attribution">
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex">On
11-08-16 18:03, Matt Fredrickson wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex">
On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens <<a
moz-do-not-send="true"
href="mailto:jonas.kellens@telenet.be" target="_blank">jonas.kellens@telenet.be</a>>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex">
My main reason not to upgrade to Ast 13 is because I'm
afraid of losing<br>
functionality as there are certain functions
deprecated/replaced. This can<br>
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<br>
and NAT is working (not causing problems) why should Ast
13 bring me audio<br>
and Ast 12 don't ??<br>
</blockquote>
If you want to minimize grief, start with 13 - WebRTC has
been a<br>
moving target for the last 5 years, it is not an old,
mature standard<br>
like ISDN or SIP. If you find interop problems in an
older version of<br>
Asterisk with WebRTC, it's likely that it has been fixed
in 13, and if<br>
it hasn't the most likely place to obtain the fix will be
in 13.<br>
<br>
After you get the WebRTC part working, then you can move
back the<br>
versions of Asterisk you're using to see if it still
works.<br>
<br>
As far as ICE not working goes, if the browser you're
talking to is<br>
not on the same network as the Asterisk server, it's
*possible* you<br>
might need a true TURN server as well, instead of just an
ICE server.<br>
<br>
Matthew Fredrickson<br>
<br>
</blockquote>
<br>
Matthew<br>
<br>
when I set the following in rtp.conf :<br>
<br>
turnaddr=<a moz-do-not-send="true"
href="http://192.158.29.39:3478?transport=udp"
rel="noreferrer" target="_blank">192.158.29.39:3478?tr<wbr>ansport=udp</a><br>
turnusername=28224511:13793308<wbr>08<br>
turnpassword=JZEOEt2V3Qb0y27GR<wbr>ntt2u2PAYA<br>
<br>
<br>
then Asterisk 12 gets really slow and sometimes
unresponsive. Calls result in 480 request timeout (possibly
due to the freeze of Asterisk).<br>
<br>
So this is also no solution.<br>
<br>
Can not even test if it brings me some audio in my webRTC
calls.<br>
<br>
<br>
(putting the above lines back in comment resolves the issue
of Asterisk freeze. This is all EXTREMELY BUGGY !)<br>
<br>
<br>
Asterisk 13 here I come (with very high expectations).<br>
<br>
<br>
Kind regards.<br>
<br>
<br>
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</blockquote>
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