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
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