8 expert Websockets Vs Sse Vs Webrtc answers researched from official documentation. Every answer cites authoritative sources you can verify.
WebSockets: bidirectional real-time (chat, collaborative editing, gaming). SSE: unidirectional server→client (live feeds, notifications, stock tickers). WebSockets require ws:// protocol upgrade, SSE uses standard HTTP. SSE simpler (auto-reconnect, EventSource API), WebSockets more flexible (binary data, lower latency). Choose SSE for: one-way updates, HTTP/2 multiplexing. Choose WebSockets for: duplex communication, sub-100ms latency.
WebRTC: peer-to-peer audio/video/data (video calls, gaming, file transfer, <50ms latency). WebSockets: client-server architecture (chat, notifications). WebRTC uses UDP (lossy, low latency), WebSockets use TCP (reliable, higher latency). WebRTC requires STUN/TURN servers for NAT traversal. Choose WebRTC for: media streaming, P2P data. Choose WebSockets for: server coordination, guaranteed delivery.
SSE advantages: (1) Auto-reconnect with Last-Event-ID (built-in resume), (2) Standard HTTP (works through proxies, CDNs), (3) Simpler API (EventSource vs WebSocket protocol), (4) HTTP/2 multiplexing (1 connection, many streams), (5) No custom server logic needed. Limitations: unidirectional only, text-based (no binary), 6 concurrent connections per domain in HTTP/1.1.
WebSocket advantages: (1) Bidirectional (client can push to server), (2) Binary data support (protobuf, msgpack), (3) Lower overhead (persistent connection, no HTTP headers per message), (4) Lower latency (10-50ms typical vs 50-200ms SSE), (5) No concurrent connection limit. Drawbacks: custom protocol, requires ws:// upgrade, manual reconnect logic, harder to proxy.
WebRTC data channels: P2P text/binary data transfer without server relay. Use cases: (1) Gaming (real-time game state sync, <20ms latency), (2) File transfer (browser-to-browser, no upload to server), (3) Collaborative editing (CRDTs, low latency), (4) IoT device control, (5) Screen sharing metadata. Supports: unreliable (UDP-like) and reliable (TCP-like) modes.
Typical latencies (2025): WebRTC data channels: 5-20ms (P2P, UDP), WebSockets: 10-50ms (TCP, server relay), SSE: 50-200ms (HTTP long-polling fallback for old browsers). WebRTC lowest for P2P, WebSockets for client-server real-time, SSE acceptable for non-interactive updates. Network conditions and server distance affect all.
2025 browser support: WebSockets (100% modern browsers, IE10+), SSE (95%+ modern, Safari 5+, no IE), WebRTC (99%+ modern, Chrome 23+, Safari 11+). All supported on mobile (iOS 11+, Android Chrome). SSE polyfills available for old browsers. WebRTC requires getUserMedia permissions for audio/video. All production-ready.
Decision matrix: (1) One-way server updates → SSE (simplest), (2) Bidirectional chat/notifications → WebSockets, (3) Audio/video calls → WebRTC (only option), (4) P2P file transfer → WebRTC data channels, (5) Real-time gaming → WebRTC (lowest latency) or WebSockets (simpler), (6) Live dashboard → SSE (auto-reconnect). Consider: latency requirements, infrastructure complexity, browser support.