WEBRTC-to-SIP - Setup for a WEBRTC client and Kamailio server to call SIP clients

  •        481

How to setup Kamailio + RTPEngine + TURN server to enable calling between WEBRTC client and legacy SIP clients. This setup will bridge SRTP --> RTP and ICE --> nonICE to make a WEBRTC client (SIPJs) be able to call legacy SIP clients. This setup is for Debian 9 Stretch for all servers.

https://github.com/havfo/WEBRTC-to-SIP

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