Understanding WebRTC Architecture
WebRTC (Web Real-Time Communication) enables peer-to-peer audio, video, and data sharing between browsers and mobile apps. The architecture consists of three main components: signaling for session establishment, STUN servers for NAT traversal, and TURN servers as relay fallback when direct connection fails.
Key WebRTC APIs include getUserMedia for capturing media, RTCPeerConnection for peer-to-peer communication, and RTCDataChannel for arbitrary data transfer. Understanding these components is essential for building robust video calling applications.
Signaling Server Setup
Signaling is not part of WebRTC spec, so you must implement it yourself. Use WebSocket for real-time bidirectional communication. Popular choices include Socket.IO, Firebase Realtime Database, or custom WebSocket servers. The signaling server exchanges SDP (Session Description Protocol) offers and answers, along with ICE (Interactive Connectivity Establishment) candidates between peers.
A reliable signaling server is critical - connection failures often stem from signaling issues, not WebRTC itself.
STUN and TURN Servers
STUN (Session Traversal Utilities for NAT) servers help clients discover their public IP address and port. Use Google's free STUN servers for development, but deploy your own for production. TURN (Traversal Using Relays around NAT) servers relay traffic when peer-to-peer connection fails, consuming significant bandwidth. Budget for TURN server costs as 10-20% of calls may require relay.
iOS Implementation
Use Google's WebRTC framework for iOS. Add the pod to your project, configure RTCPeerConnectionFactory, handle camera and microphone permissions, implement RTCVideoRenderer for displaying video, and manage CallKit integration for native calling experience. Handle background mode and interruptions properly for production apps.
Android Implementation
Android WebRTC implementation uses the same Google library. Initialize PeerConnectionFactory on a background thread, request CAMERA and RECORD_AUDIO permissions, create local and remote video tracks, use SurfaceViewRenderer for video display, and handle lifecycle events carefully to prevent memory leaks.
Handling Network Changes
Mobile networks are unstable. Implement ICE restart when network changes, handle cellular to WiFi transitions gracefully, adjust video quality based on bandwidth, implement reconnection logic with exponential backoff, and show connection quality indicators to users.
Optimization and Quality
Optimize for mobile constraints by starting with lower resolution and scaling up, using VP8 or VP9 codec for better compression, implementing simulcast for group calls, monitoring packet loss and adjusting bitrate, and providing audio-only fallback option for poor connections.