Building Real-Time Apps: WebSockets, WebRTC & Live Data

“Make it real-time” is one of those requests that sounds simple and hides a dozen decisions. A live map of trucks, a chat thread, a collaborative document, a video call, a dashboard that updates itself. These all feel similar to a user, but under the hood they use very different tools. Pick the wrong one and you'll either over-engineer or hit a wall. Here's how to choose.

The three tools, and when each fits

Server-Sent Events (SSE) are the simplest. The server pushes a one-way stream of updates to the browser over plain HTTP. If data only needs to flow down. A live price ticker, a notification feed, a status dashboard. SSE is wonderfully low-effort and reconnects on its own.

WebSockets open a persistent two-way channel between client and server. Reach for them when both sides need to talk continuously: chat, multiplayer state, live cursors, presence (“who's online”), or anything where the client sends frequent updates too. This is the workhorse of most real-time features.

WebRTC is the heavy machinery, built for audio, video, and direct peer-to-peer connections between users' devices. It's what powers video calls. It's powerful but genuinely complex. Only reach for it when you actually need live media or true device-to-device data.

Data flows down only → SSE. Both directions → WebSockets. Live audio/video or peer-to-peer → WebRTC. Most apps need the middle one.

The detail that surprises teams: WebRTC still needs servers

People assume “peer-to-peer” means no infrastructure. Not quite. Before two devices can connect directly, they need a signaling server to exchange connection details, plus STUN/TURN servers to punch through firewalls and home routers. When a direct path can't be established. And on restrictive networks it often can't. Traffic falls back to relaying through a TURN server, which costs bandwidth. Budget for that; it's the line item most first-time WebRTC projects forget.

The hard part isn't opening the connection

Getting a single WebSocket working is an afternoon. Making it reliable at scale is the real work. Connections drop on flaky mobile networks, so you need automatic reconnect and a way to catch up on missed messages. One server can only hold so many live connections, so scaling out means a way to broadcast a message to users spread across many servers. Usually a pub/sub layer like Redis sitting behind them.

// A connection that survives the real world: reconnect + resync
socket.on("disconnect", () => scheduleReconnect());
socket.on("connect", () => socket.emit("resume", { lastSeenId }));
// server replays anything newer than lastSeenId, then resumes live

For most teams, a managed real-time platform or a battle-tested library handles reconnection, scaling, and presence so you don't rebuild it from scratch. That's usually the right call unless real-time is your core product.

The bottom line

Match the tool to the data flow. SSE for one-way, WebSockets for two-way, WebRTC for live media. And then spend your energy on the unglamorous parts: reconnection, catching up after a drop, and scaling across servers. That's what separates a demo from a feature people rely on. We've built live tracking and real-time tools that hold up in production. tell us what you're building, or browse our work.