RFC 8825 - WebRTC Overview
Web Real-Time Communication architecture and protocols
What is WebRTC?
WebRTC (Web Real-Time Communication) is a collection of standards, protocols, and APIs that enable real-time peer-to-peer communication of audio, video, and data directly between web browsers and mobile applications.
Key Components
Media Capture
Access to camera and microphone through getUserMedia API.
Peer Connection
Direct peer-to-peer communication without server intermediaries.
Data Channels
Bidirectional data communication between peers.
NAT Traversal
ICE, STUN, and TURN protocols for connectivity through firewalls.
Architecture Overview
Signaling
Session Description Protocol (SDP) for negotiating media capabilities and network information.
Media Transport
Real-time Transport Protocol (RTP) for audio and video delivery.
Security
Mandatory encryption using DTLS and SRTP for secure communication.
Common Use Cases
- Video Conferencing: Browser-based video calls without plugins
- Voice over IP: Real-time audio communication
- Screen Sharing: Desktop and application sharing
- File Transfer: Direct peer-to-peer file sharing
- Gaming: Low-latency multiplayer gaming
- IoT Communication: Device-to-device communication