[asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

[Jonas Kellens]

Hello


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.




Kind regards.


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>>
>         wrote:
>
>             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
>         standard
>         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
>         server.
>
>         Matthew Fredrickson
>
>
>     Matthew
>
>     when I set the following in rtp.conf :
>
>     turnaddr=192.158.29.39:3478?transport=udp
>     <http://192.158.29.39:3478?transport=udp>
>     turnusername=28224511:1379330808
>     turnpassword=JZEOEt2V3Qb0y27GRntt2u2PAYA
>
>
>     then Asterisk 12 gets really slow and sometimes unresponsive.
>     Calls result in 480 request timeout (possibly due to the freeze of
>     Asterisk).
>
>     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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