Introduction to WebRTC for Real-Time Communication > 자유게시판

WebRTC enables peer-to-peer real-time communication in browsers and apps. getUserMedia accesses camera and microphone for video/audio. RTCPeerConnection establishes direct peer-to-peer connections. ICE (Interactive Connectivity Establishment) finds the best network path. STUN servers discover public IP address for connectivity. TURN servers relay media when direct connection fails. SDP (Session Description Protocol) describes media capabilities. Signaling is not defined by WebRTC and requires custom implementation. WebSocket is commonly used for signaling channel. DataChannel enables peer-to-peer data transfer. MediaStream synchronizes audio and video tracks. Bandwidth estimation adapts quality to network conditions. Codec support includes VP8, VP9, H.264, and Opus. Simulcast sends multiple quality layers for adaptive streaming. Screensharing captures application or browser window. WebRTC libraries like PeerJS simplify implementation. Recording MediaStream with . WebRTC requires HTTPS for browser access. Security includes encryption of all media streams. Applications include video conferencing, live streaming, and file sharing.