Live Streaming WebRTC

WebRTC is a browser-native API for real-time peer-to-peer (P2P) communication: video, audio, and data. Unlike traditional streaming (RTMP, HLS), WebRTC achieves sub-second latency by avoiding media servers (when possible). Two browsers connect directly, exchange video/audio via an encrypted tunnel. For larger audiences, WebRTC flows through a media server (Janus, Kurento) which relays to viewers. WebRTC handles codec negotiation, bandwidth adaptation, and packet loss recovery automatically. Live streaming with traditional protocols (RTMP, HLS) introduces 3-30 second delays due to encoding, buffering, and edge distribution. For interactive applications (gaming tournaments, live auctions, telehealth, remote collaboration), that delay is unacceptable. WebRTC enables true real-time. Building WebRTC infrastructure is complex (signaling, STUN/TURN, codec management) but the payoff is immense: sub-second latency at scale. The skill is scarce and well-compensated.