Video Conferencing System Design (Zoom, Google Meet)
Introduction
Video conferencing platforms enable real-time audio, video, and screen sharing for millions of users. These systems must deliver low latency, high quality, and reliability across the globe.
Problem Statement
How can we design a video conferencing system that supports real-time communication, scales to millions of users, and maintains high quality?
System Requirements
- Real-time audio and video streaming.
- Low latency and high reliability.
- Scalability for large meetings and webinars.
- Screen sharing and chat features.
- Security (encryption, authentication).
High-Level Design
The system consists of:
- Client Applications: Capture and render audio/video.
- Media Servers: Route, mix, and transcode streams.
- Signaling Servers: Manage session setup and control.
- TURN/STUN Servers: Handle NAT traversal.
- Database: Stores user, meeting, and metadata.
Key Components
- WebRTC: Standard for real-time communication in browsers.
- SFU/MCU: Selective Forwarding Unit (SFU) or Multipoint Control Unit (MCU) for efficient stream routing.
- Load Balancing: Distributes sessions across servers.
- Recording and Archiving: Stores meeting recordings.
Challenges
- Scalability: Handling thousands of concurrent streams.
- Network Variability: Adapting to changing bandwidth and latency.
- Synchronization: Keeping audio, video, and screen sharing in sync.
- Security: End-to-end encryption and access control.
Example Technologies
- WebRTC: Real-time media transport.
- Media Servers: Janus, Jitsi, Kurento.
- Signaling: WebSockets, gRPC.
Conclusion
Video conferencing at scale requires efficient media handling, robust signaling, and adaptive streaming. By leveraging WebRTC, scalable media servers, and secure protocols, you can deliver high-quality, reliable video communication.