[asterisk-users] Asterisk 13 and WebRTC

[Annus Fictus]

Hello,

I mean a working configuration (SIP o PJSIP) without patches or code 
corrections.

Thank you

Regards


El 09/09/2016 a las 03:47, marek cervenka escribió:
> using in production
>
> last asterisk 13 + pjsip bundled + pjsip patch for RTP/SAVPF (search 
> pjsip conf) + sipml5 version from roginvs
>
> https://github.com/DoubangoTelecom/sipml5/pull/238
>
>
> Dne 08/09/2016 v 23:36 Annus Fictus napsal(a):
>> Hello list,
>>
>> before to lost my time, I'd like know if someone have a WebRTC 
>> working configuration on Asterisk 13.11.0 SIP or PJSIP channel.
>>
>> Thank you
>>
>> Regards
>>
>>
>>
>
>