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.

Links

• WebRTC: https://webrtc.org
• Janus-Gateway: https://janus.conf.meetecho.com/
• NethServer VoIP PBX: http://docs.nethserver.org/en/v7/freepbx.html
• NethServer community: https://community.nethserver.org/
• NethServer repo:
  • https://github.com/NethServer/nethserver-freepbx
  • https://github.com/NethServer/freepbx
  • https://github.com/NethServer/janus-gateway
  • https://github.com/NethServer/nethserver-janus
• NethVoice (Enterprise NethServer VoIP PBX): https://nethvoice.it