[asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?
jonas.kellens at telenet.be
Fri Aug 12 08:02:03 CDT 2016
setting "nat=no" or omitting "nat=" in peer definition does not help
either. Still no audio.
Why do you think this is a NAT issue ? IP and port information in
SDP-body is correct.
On 12-08-16 09:25, Антон Сацкий wrote:
> Try delete nat from 770000wrtc settings ice should do the same
> On Aug 11, 2016 10:00 PM, "Jonas Kellens" <jonas.kellens at telenet.be
> <mailto:jonas.kellens at telenet.be>> wrote:
> On 11-08-16 18:03, Matt Fredrickson wrote:
> On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens
> <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>>
> 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 :-)
> I will do so if there is no other option.
> 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 ??
> If you want to minimize grief, start with 13 - WebRTC has been a
> moving target for the last 5 years, it is not an old, mature
> like ISDN or SIP. If you find interop problems in an older
> version of
> Asterisk with WebRTC, it's likely that it has been fixed in
> 13, and if
> it hasn't the most likely place to obtain the fix will be in 13.
> After you get the WebRTC part working, then you can move back the
> versions of Asterisk you're using to see if it still works.
> As far as ICE not working goes, if the browser you're talking
> to is
> not on the same network as the Asterisk server, it's
> *possible* you
> might need a true TURN server as well, instead of just an ICE
> Matthew Fredrickson
> when I set the following in rtp.conf :
> then Asterisk 12 gets really slow and sometimes unresponsive.
> Calls result in 480 request timeout (possibly due to the freeze of
> So this is also no solution.
> Can not even test if it brings me some audio in my webRTC calls.
> (putting the above lines back in comment resolves the issue of
> Asterisk freeze. This is all EXTREMELY BUGGY !)
> Asterisk 13 here I come (with very high expectations).
> Kind regards.
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