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

[Jonas Kellens]

Question : I noticed I received an error when installing pjproject 
--with-external-srtp

I do not seems to have the srtp capability.
(However I can easily install with "yum install libsrtp-devel")

Can this have anything to do with the no-audio-problems that I'm having ??



Kind regards.


On 12-08-16 15:02, Jonas Kellens wrote:
> 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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>>
>>
>
>
>

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