Hi All,
We are trying to setup WebRTC with Kamailio as sip registrar and
dispatcher. We have succeed for outgoing calls but cannot attend incoming
calls to the Kamailio. It would be a great help if someone can provide a
sample config for webrtc + kamailio or blog for the same.
Thanking you all
--
Adesh Pandey
Sr. Software Developer
[image: phone]
+91 92129 92129 Ext: 56
[image: moible]
+91 8527384897
[image: Profile Pic]
[image: Facebook] <https://www.facebook.com/VoiceTree?_rdr>[image: Twitter]
<https://twitter.com/MyOperator>[image: Twitter]
<https://www.youtube.com/user/MyOperatorCo>[image: Linkedin]
<https://www.linkedin.com/company/voicetree-technologies>[image: Google+]
<https://plus.google.com/+MyoperatorCo/posts>
[image: Logo] <http://myoperator.co/>