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.
Where should the data go?
Use WebSockets when your server owns the state or needs to process every message. Use WebRTC when peers should exchange files, audio, video, or live updates with each other.
You can use both. Keep server connections for stored state and permission checks, and use WebRTC for a call or a file transfer between users.
Comparison
| Need | WebRTC | WebSockets |
|---|---|---|
| Data path | Peer to peer, directly or through an encrypted TURN relay | Browser to server to browser |
| Best for | Media and data exchanged between online peers | Updates distributed from a server |
| Backend needs | Signaling; TURN for some networks; storage if needed | Always-on socket service |
| Scaling pressure | Peer bandwidth and connection count; TURN traffic when relayed | Server connections, processing, and outgoing bandwidth |
| Media support | Built for audio, video, screen share, and data channels | General data transport; no built-in media negotiation |
Where Trystero fits
Trystero handles WebRTC discovery and connection setup. It gives you a room API for messages, requests, files, and media, using an existing signaling service or your own relay.
A room does not provide persistence or resolve conflicting edits. Keep those responsibilities in your app. Also plan for peer bandwidth and connection counts as rooms grow, especially for video.