Guide
When to use WebRTC vs WebSockets
WebSockets connect each browser to a server. WebRTC can connect browsers directly to each other. The right choice depends on where your realtime data should flow.
Short answer
Use WebSockets when the server is the source of truth or every message needs to pass through your backend. Use WebRTC when peers should exchange low-latency data, files, audio, or video directly.
Many production systems use both: WebSockets for accounts, persistence, matchmaking, moderation, and durable state; WebRTC for high-volume or low-latency peer-to-peer flows.
Comparison
| Need | WebRTC | WebSockets |
|---|---|---|
| Data path | Browser to browser after discovery | Browser to server to browser |
| Best for | Video, voice, files, presence, local multiplayer | Chat, feeds, dashboards, server-owned state |
| Backend needs | Discovery, optional TURN, optional persistence | Always-on socket service |
| Scaling pressure | Less app payload through your servers | Every realtime payload touches your servers |
| Media support | Built for audio, video, screen share, and data channels | Data only unless you add media infrastructure |
Where Trystero fits
Trystero is for the WebRTC side of the decision. It gives your JavaScript app rooms, peer events, actions, request/response calls, binary transfer, and media stream helpers without requiring a custom signaling server.
If your app already has WebSockets, you can keep them for durable server workflows and add Trystero for direct peer-to-peer features such as video chat, collaborative cursors, multiplayer state, or browser-to-browser file transfer.