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SIP Phone WebRTC

This is a WebRTC SIP Phone that can be easily integrated into your web application to make audio and video calls.

It uses Janus-Gateway produced by Meetecho.

How to use

  1. clone the repository "git clone https://github.com/alepolidori/janus-webrtc-phone.git"
  2. run "npm install"
  3. open "index.html" into your web browser

Functions

  1. Initialize Janus stack
  2. Register an extension
  3. Make an audio/video call
  4. Answer a call
  5. Hangup a call

Requirements

Since it is a client demo, it requires a VoIP PBX backend to interact with other extensions. You can easily install one of your own using NethServer VoIP PBX: follow this guide.

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