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/>