WebRTC 技术概览

WebRTC (Web Real-Time Communication) lets browsers and mobile apps talk directly to each other - no intermediate server for the actual audio, video, or data.

"Real-time" means things happen with no perceptible delay: instant interactions, low latency, and immediate feedback - like a face-to-face chat over the network.

The "Web" in WebRTC means it runs straight in Chrome, Firefox, Safari, and Edge - no plugins. The "RTC" enables live audio, video, and data between devices.

WebRTC's standout feature is peer-to-peer connections: two devices talk directly. No middleman means lower latency and better privacy once connected.

WebRTC isn't just video. It handles voice calls, live video, and direct data transfer - text, files, anything - between peers.

WebRTC is an open standard with encryption on by default and built-in network adaptability, so it connects peers even through tricky network setups.

Before WebRTC, live video meant clunky plugins. Now you add rich, real-time communication with plain JavaScript APIs - making the web far more dynamic.

WebRTC works across all major browsers on desktop and mobile - Chrome, Firefox, Safari, Edge, Opera - so your app reaches a huge audience with no compatibility pain.

The high-level flow: signaling (peers swap setup info via a server), connection (find the best direct route), then streaming. The server only sets up the call.

Let's see if you've grasped the main idea!

Recap: WebRTC enables real-time audio, video, and data in the browser via peer-to-peer links. It's an open standard with wide support. Next: signaling.