#javascript Audio/Video Streaming: #WebRTC APIs 101 explained
summary
A minimal video‑call app is built from scratch and demonstrably works across Chrome and Firefox using browser APIs and a tiny Node.js server: it captures camera/mic with getUserMedia, creates peer links with RTCPeerConnection (SDP offers/answers, ICE candidates, STUN for public IP, TURN as fallback), and exchanges signaling over a simple WebSocket signaling implementation over HTTPS. The demo runs live, STUN handles ~80–90% of cases, TURN relays when needed.
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chapters (8)
- 0:00Introduction to WebRTC
- 3:38WebRTC JavaScript APIs overview
- 5:45GetUserMedia API demo
- 9:15RTCPeerConnection explained
- 11:22Signaling and ICE candidates
- 13:08Live demo setup
- 22:38WebRTC in practice demo
- 24:21Protocols and resources








