asterisk users - Aug 2016 - Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

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> 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
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.

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> 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>
>> 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
> 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.
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20160812/1355bbf8/attachment.html>

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, ????? ?????? 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.
>
>
>     -- 
>     _____________________________________________________________________
>     -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>     New to Asterisk? Join us for a live introductory webinar every Thurs:
>     http://www.asterisk.org/hello
>
>     asterisk-users mailing list
>     To UNSUBSCRIBE or update options visit:
>     http://lists.digium.com/mailman/listinfo/asterisk-users
>     <http://lists.digium.com/mailman/listinfo/asterisk-users>
>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20160812/a3c8a774/attachment.html>