WebRTC: Real-Time Communication di Browser

WebRTC (Web Real-Time Communication) adalah open-source project enabling real-time communication via simple APIs. Developed oleh Google, standardized by W3C dan IETF. Built into browsers, no plugins needed. Use cases: video conferencing, voice calls, screen sharing, file transfer, live streaming, gaming. Core APIs: MediaStream (getUserMedia untuk access camera/microphone), RTCPeerConnection (peer-to-peer audio/video), RTCDataChannel (arbitrary data transfer). Architecture: peer-to-peer connection, signaling server for connection setup, STUN/TURN servers untuk NAT traversal. Connection flow: get user media, create peer connection, create offer (SDP), exchange via signaling server, create answer, exchange ICE candidates, connection established. Signaling: not part of WebRTC spec, commonly use WebSockets, implementations dengan Socket.io, any bidirectional protocol works. SDP (Session Description Protocol): describes media capabilities, codecs, formats. ICE (Interactive Connectivity Establishment): find best path between peers, STUN server untuk discover public IP, TURN server untuk relay jika direct connection impossible. Media handling: constraints (resolution, frame rate), track control (mute, disable), multiple streams, screen sharing. Data channels: reliable/unreliable transfer, ordered/unordered, low-latency, use cases: chat, file transfer, gaming data. Codecs: VP8, VP9, H.264 untuk video; Opus untuk audio, browser support varies, negotiate via SDP. Quality: bandwidth estimation, adaptive bitrate, congestion control, network statistics via getStats(). Security: mandatory encryption (DTLS for data channels, SRTP for media), end-to-end encryption possible. Browser support: Chrome, Firefox, Safari, Edge all support, mobile browsers supported. Challenges: NAT traversal (TURN servers expensive), codec compatibility, cross-browser differences, scaling (mesh topology doesn't scale beyond 4-5 peers). Architectures: Mesh (peer-to-peer, simple but doesn't scale), SFU (Selective Forwarding Unit, server forwards streams, scalable), MCU (Multipoint Control Unit, server mixes streams, highest server load). Signaling servers: Socket.io, Firebase, PeerJS (abstracts signaling), simple-peer library. Media servers: Janus, Kurento, Jitsi, Mediasoup (SFU), commercial solutions (Vonage, Twilio, Agora). Libraries: simple-peer (simplified API), PeerJS (easy P2P), adapter.js (browser compatibility). Recording: MediaRecorder API, server-side recording. Screen sharing: getDisplayMedia, electron integration untuk desktop apps. Testing: test on different browsers/devices, network conditions simulation, automated testing tricky. Production considerations: TURN server costs (bandwidth intensive), scaling strategy, monitoring, fallback mechanisms. Use cases: Zoom/Google Meet competitors, customer support, telemedicine, online education, social apps. Future: WebRTC NV (Next Version) improvements, better codec support, insertable streams for ML processing. Learning curve: moderate, signaling most confusing part. WebRTC democratizes real-time communication, enables building Zoom-like apps with open-source technology. Privacy-focused alternative to proprietary solutions.