WebRTC (Web Real-Time Communication) is an API that can be used by video-chat, voice-calling, and P2P-file-sharing Web apps. We can use STUN(Session Traversal Utilities for NAT) or TURN(Traversal Using Relays Around NATs) servers for relaying the traffic between peers.
- We can use public STUN/TURN servers or setup own using COTURN server.
iceConfig: {
iceServers: [
{
urls: [
'stun:stun1.l.google.com:19302',
'stun:stun2.l.google.com:19302',
],
},
],
iceCandidatePoolSize: 10,
}