I’ve been thinking a lot about video calls lately. Not just the ones we use every day, but what it takes to build them from the ground up. The magic of seeing and hearing someone in real-time, with no plugins, is powered by a technology called WebRTC. But moving from a simple demo to a system that can handle dozens of people in a meeting is a significant challenge. Today, I want to walk through how to build that system, piece by piece.
Why does this matter? Because real-time communication is no longer a luxury; it’s how we work, learn, and connect. Understanding the tools that make it possible is incredibly empowering for a developer.
Let’s start with the basics. WebRTC lets browsers talk directly to each other. For a one-on-one call, this direct connection is perfect. But have you ever wondered what happens when you add a third person, or a tenth? The direct approach quickly falls apart. Each device would need to send its video to every other device, consuming massive amounts of upload bandwidth. This is where our journey from a simple peer-to-peer chat to a scalable conference room begins.
For the simple case, we can use a library like PeerJS. It handles the complex handshake needed to connect two browsers. You need a signaling server to help them find each other, but once connected, the data flows directly between them.
Here’s a glimpse of setting up a PeerJS server in Node.js:
const { ExpressPeerServer } = require('peer');
const express = require('express');
const app = express();
const server = require('http').createServer(app);
const peerServer = ExpressPeerServer(server, {
path: '/peerjs',
debug: true,
});
app.use('/peer', peerServer);
server.listen(3000);On the client side, the code to start a video call is surprisingly straightforward. You get access to the user’s camera, create a Peer object, and then you can either call another peer or answer an incoming call.
// Get the user's media stream
const stream = await navigator.mediaDevices.getUserMedia({ video: true, audio: true });
const peer = new Peer();
// To call another user
const call = peer.call('other-peer-id', stream);
call.on('stream', (remoteStream) => {
// Show the remote video
videoElement.srcObject = remoteStream;
});
// To answer a call
peer.on('call', (incomingCall) => {
incomingCall.answer(stream);
incomingCall.on('stream', (remoteStream) => {
// Show the remote video
});
});This works beautifully for two people. But what’s the breaking point? When do you need a more powerful architecture? The answer is when you want a true multi-person conference. This is where we introduce a media server.
Think of a media server as a smart traffic cop. Instead of everyone sending video to everyone else, each participant sends one stream to the server. The server then decides who needs which stream and sends it to them. This saves a huge amount of bandwidth on each user’s device. The most efficient pattern for this is called a Selective Forwarding Unit, or SFU.
For building an SFU in Node.js, Mediasoup is a fantastic choice. It’s powerful, well-documented, and used in production by large companies. It manages the complexities of routing audio and video between many participants.
Setting up Mediasoup involves creating a “worker” process that handles the media and a “router” for each conference room. Here is a simplified version of how you might start a Mediasoup worker:
const mediasoup = require('mediasoup');
const config = require('./config');
async function createWorker() {
const worker = await mediasoup.createWorker({
logLevel: 'warn',
rtcMinPort: 10000,
rtcMaxPort: 10100,
});
worker.on('died', () => {
console.error('Mediasoup worker died, exiting in 2 seconds...');
setTimeout(() => process.exit(1), 2000);
});
return worker;
}Each participant connects to the router and produces their media stream. The router then creates consumers for the streams that participant should see and hear. The signaling to coordinate all this—telling the client how to connect to the SFU, which streams are available—is handled through WebSockets using a library like Socket.io.
So, how does a client connect to this Mediasoup SFU? The process is more involved than PeerJS but follows a clear sequence: get device capabilities, create a transport for sending media, and then start producing your video.
// Client-side steps to join a Mediasoup room
// 1. Get router capabilities from server via socket
// 2. Create a WebRTC transport
const transport = await device.createSendTransport({
id: serverTransportId,
iceParameters: serverIceParams,
iceCandidates: serverIceCandidates,
dtlsParameters: serverDtlsParams,
});
// 3. Connect the transport and start producing video
transport.on('connect', async ({ dtlsParameters }, callback, errback) => {
// Signal server to connect transport with these parameters
await socket.emit('connect-transport', { dtlsParameters });
callback();
});
const stream = await navigator.mediaDevices.getUserMedia({ video: true });
const videoTrack = stream.getVideoTracks()[0];
const producer = await transport.produce({ track: videoTrack });This architecture allows you to host conferences with hundreds of users. The server handles the heavy lifting of mixing and routing, and each client only deals with a few streams.
But a real production system needs more. What happens when a user’s internet connection is poor? The SFU can adapt by sending them a lower-resolution video. How do people join a specific meeting? You need a room system with unique IDs. What about security and authentication? You must ensure only authorized users can produce video.
You also need TURN servers. WebRTC tries to make a direct connection, but sometimes a user is behind a restrictive firewall or NAT. A TURN server acts as a relay, bouncing the video traffic through it to ensure the connection works. It’s less efficient but crucial for reliability.
Building the full application means combining these pieces: a signaling server (Socket.io), a media server (Mediasoup), a room management layer, and a TURN server like Coturn. You’ll also want to add features like screen sharing, which is just another media track, and maybe even recording, which involves piping media streams to a file on the server.
The journey from a simple two-person call to a robust video conference platform is challenging but deeply rewarding. You’re not just building a feature; you’re building infrastructure that enables human connection. You learn about network protocols, real-time data handling, and scalable system design.
What problem will you solve with this knowledge? A new virtual classroom, a telehealth platform, or a better way for remote teams to collaborate? The tools are now in your hands.
I hope this guide helps you start building. If you found this walk-through useful, please share it with a fellow developer who might be curious about real-time video. I’d love to hear about your projects or answer any questions in the comments below. Let’s keep the conversation going.
As a best-selling author, I invite you to explore my books on Amazon. Don’t forget to follow me on Medium and show your support. Thank you! Your support means the world!
101 Books
101 Books is an AI-driven publishing company co-founded by author Aarav Joshi. By leveraging advanced AI technology, we keep our publishing costs incredibly low—some books are priced as low as $4—making quality knowledge accessible to everyone.
Check out our book Golang Clean Code available on Amazon.
Stay tuned for updates and exciting news. When shopping for books, search for Aarav Joshi to find more of our titles. Use the provided link to enjoy special discounts!
📘 Checkout my latest ebook for free on my channel!
Be sure to like, share, comment, and subscribe to the channel!
Our Creations
Be sure to check out our creations:
Investor Central | Investor Central Spanish | Investor Central German | Smart Living | Epochs & Echoes | Puzzling Mysteries | Hindutva | Elite Dev | JS Schools
We are on Medium
Tech Koala Insights | Epochs & Echoes World | Investor Central Medium | Puzzling Mysteries Medium | Science & Epochs Medium | Modern Hindutva
