Tutorials

Tutorials used:

WebRTC without setting up server: https://www.grafikart.fr/tutoriels/javascript/webrtc-864
WebRTC without any external libraries: https://websitebeaver.com/insanely-simple-webrtc-video-chat-using-firebase-with-codepen-demo
WebRTC with Firebase https://dev.to/rynobax_7/creating-a-multiplayer-game-with-webrtc

Other tutorials found

WebRTC with Pusher and Node: https://pusher.com/tutorials/webrtc-video-call-app-nodejs

Definition of STUN/SERVER

A STUN/TURN server is used for NAT traversal in VoIP. Whether you're at home behind a common router, at work behind an enterprise firewall, or traveling, chances are that you will be behind a NAT which must be traversed before making calls. [source: Viagénie]

If a STUN server doesn’t work, then WebRTC will try the next server, which is why you should add several. But using more than two STUN/TURN servers slows down discovery. STUN servers are cheaper than TURN servers, which is why Google and Firefox allow anyone to access their STUN servers for free. TURN servers are harder to find for free, but they do exist. [sources: websitebeaver, Bugzilla]

Servers used in this app

simple-peer provides by default two STUN server:

config: {
  iceServers: [
    { urls: 'stun:stun.l.google.com:19302' },
    { urls: 'stun:global.stun.twilio.com:3478?transport=udp' },
  ]
}

Mozilla provides another free STUN server : {urls: 'stun:stun.services.mozilla.com'}

Viagénie provides a free TURN server:

{urls: 'turn:numb.viagenie.ca','credential': '****','username': '****'}

So my final config is as follows:

config: {
  iceServers: [
    { urls: 'stun:stun.l.google.com:19302' },
    { urls: 'stun:global.stun.twilio.com:3478?transport=udp' },
    { urls: 'stun:stun.services.mozilla.com' },
    { urls: 'turn:numb.viagenie.ca', credential: '****', username: '****' },
  ]
}

Paid service provided by Twilio:

Pricing
FAQ

WebRTC in real world apps:

Google Allo
Goole Duo

codelastnight / webrtc-firebase