WebRTC & Video Streaming
Quick Reference: WebSockets | HTTP
Quick Reference
| Technology | Use Case | Latency | Complexity |
|---|---|---|---|
| WebRTC | P2P video/audio | Very low | High |
| HLS | On-demand video | Higher | Low |
| DASH | Adaptive streaming | Medium | Medium |
| RTMP | Live streaming | Low | Medium |
Clear Definition
WebRTC (Web Real-Time Communication) enables peer-to-peer audio, video, and data communication directly in browsers without plugins. Video Streaming architectures deliver video content efficiently to users.
š” Key Insight: WebRTC enables low-latency P2P communication. Video streaming uses adaptive bitrate streaming for quality optimization.
Core Concepts
WebRTC
- P2P Communication: Direct browser-to-browser connection
- Low Latency: Real-time audio/video
- NAT Traversal: STUN/TURN servers for connectivity
- Media Streams: Audio/video/data streams
Video Streaming
- Adaptive Bitrate: Adjust quality based on bandwidth
- CDN Distribution: Serve from edge locations
- Protocols: HLS, DASH, RTMP
- Encoding: Multiple quality levels
Use Cases
WebRTC
- Video Calls: Zoom, Google Meet
- Screen Sharing: Remote collaboration
- Gaming: Real-time multiplayer
- File Transfer: P2P file sharing
Video Streaming
- On-Demand: Netflix, YouTube
- Live Streaming: Twitch, live events
- Video Conferencing: Large meetings
Best Practices
- Adaptive Streaming: Use adaptive bitrate for video
- CDN: Distribute video through CDN
- Encoding: Pre-encode multiple quality levels
- Monitoring: Track quality metrics
Quick Reference Summary
WebRTC: P2P real-time communication for low-latency audio/video.
Video Streaming: Adaptive bitrate streaming for efficient video delivery.
Key: WebRTC for real-time, streaming protocols for on-demand/live video.
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