Introducing "NAMO" Real-Time Speech AI Model: On-Device & Hybrid Cloud 📢PRESS RELEASE

WebRTC Video Call Tutorial: Build Real-Time Video Chat with JavaScript

A step-by-step guide to building a WebRTC video call application, covering essential concepts like signaling, peer connection setup, and media stream handling.

WebRTC Video Call Tutorial: Build Real-Time Video Chat with JavaScript

WebRTC (Web Real-Time Communication) enables real-time audio and video communication directly between browsers and devices. This tutorial provides a comprehensive guide to building a simple WebRTC video call application, covering the fundamental concepts and implementation details.

Introduction to WebRTC Video Calls

What is WebRTC?

WebRTC is a free, open-source project that provides browsers and mobile applications with real-time communication (RTC) capabilities via simple APIs. It allows audio and video communication to work inside web pages by allowing direct peer-to-peer communication, eliminating the need for plugins or native apps for RTC.

Why use WebRTC?

WebRTC simplifies the development of real-time communication applications. It eliminates the complexities of dealing with low-level networking protocols and codecs. Its browser support makes it accessible to a wide audience without requiring users to install additional software.

Benefits of WebRTC

  • Open Source and Free: WebRTC is free to use and modify, promoting innovation and collaboration.
  • Cross-Platform Compatibility: Works across major browsers and platforms, ensuring broad accessibility.
  • Peer-to-Peer Communication: Enables direct communication between users, reducing latency and server load.
  • Secure: WebRTC incorporates security features like encryption to protect user privacy.
  • Real-time communication: Reduce delay for more natural interaction

Prerequisites for this tutorial

Before starting this tutorial, you should have a basic understanding of:
  • HTML, CSS, and JavaScript
  • Node.js and npm (Node Package Manager)
  • Socket.IO (or another real-time communication library)

Setting up your Development Environment

Choosing your Development Tools

For this tutorial, we'll use the following tools:
  • Code Editor: Visual Studio Code (or any editor of your choice)
  • Node.js: A JavaScript runtime environment
  • npm: Node Package Manager (comes with Node.js)
  • Browser: Chrome, Firefox, or Safari (with WebRTC support)

Installing Necessary Packages

First, create a new directory for your project and navigate into it:

bash

1mkdir webrtc-video-call
2cd webrtc-video-call
3
Initialize a new Node.js project:

bash

1npm init -y
2
Install the necessary packages:

bash

1npm install socket.io express
2
  • socket.io: For real-time communication (signaling).
  • express: For creating a simple web server.

Setting up your Project Structure

Create the following files and directories:
1webrtc-video-call/
2├── index.html
3├── script.js
4├── server.js
5└── node_modules/
6
  • index.html: The main HTML file for the video call interface.
  • script.js: JavaScript file for WebRTC logic.
  • server.js: Node.js server for handling signaling.

Building the Basic WebRTC Application

Setting up Signaling

Signaling is the process of exchanging metadata between peers to establish a WebRTC connection. We'll use Socket.IO for real-time signaling.
Server-side (server.js):

javascript

1const express = require('express');
2const http = require('http');
3const socketIO = require('socket.io');
4
5const app = express();
6const server = http.createServer(app);
7const io = socketIO(server);
8
9const port = process.env.PORT || 3000;
10
11app.use(express.static('public'));
12
13io.on('connection', (socket) => {
14  console.log('User connected:', socket.id);
15
16  socket.on('joinRoom', (room) => {
17    socket.join(room);
18    console.log(`User ${socket.id} joined room ${room}`);
19    socket.to(room).emit('userJoined', socket.id);
20  });
21
22  socket.on('offer', (offer, room) => {
23    socket.to(room).emit('offer', offer, socket.id);
24  });
25
26  socket.on('answer', (answer, room) => {
27    socket.to(room).emit('answer', answer, socket.id);
28  });
29
30  socket.on('iceCandidate', (iceCandidate, room) => {
31    socket.to(room).emit('iceCandidate', iceCandidate, socket.id);
32  });
33
34  socket.on('disconnect', () => {
35    console.log('User disconnected:', socket.id);
36  });
37});
38
39server.listen(port, () => {
40  console.log(`Server running on port ${port}`);
41});
42
Client-side (script.js):

javascript

1const socket = io('http://localhost:3000'); // Replace with your server URL
2const room = 'myRoom'; // Replace with your room name
3let peerConnection;
4let localStream;
5
6socket.on('connect', () => {
7  console.log('Connected to signaling server');
8  socket.emit('joinRoom', room);
9});
10
11socket.on('userJoined', (userId) => {
12  console.log(`User ${userId} joined the room`);
13  createOffer(userId);
14});
15
16

Creating a Peer Connection

The RTCPeerConnection interface represents a WebRTC connection between two peers. It provides methods to connect to a remote peer, maintain and monitor the connection, and close the connection once it’s no longer needed.

javascript

1async function createPeerConnection(remoteUserId) {
2  peerConnection = new RTCPeerConnection({
3    iceServers: [
4      { urls: 'stun:stun.l.google.com:19302' },
5      { urls: 'stun:stun1.l.google.com:19302' },
6    ],
7  });
8
9  peerConnection.onicecandidate = (event) => {
10    if (event.candidate) {
11      console.log('ICE candidate:', event.candidate);
12      socket.emit('iceCandidate', event.candidate, room);
13    }
14  };
15
16  peerConnection.ontrack = (event) => {
17    console.log('Track event:', event);
18    const remoteVideo = document.getElementById('remoteVideo');
19    remoteVideo.srcObject = event.streams[0];
20  };
21
22  localStream.getTracks().forEach((track) => {
23    peerConnection.addTrack(track, localStream);
24  });
25
26  return peerConnection;
27}
28
29

Handling Media Streams

To capture audio and video from the user's device, we use the getUserMedia API. Then, we attach the mediastream to our local video element.

javascript

1async function startVideo() {
2  try {
3    localStream = await navigator.mediaDevices.getUserMedia({
4      video: true,
5      audio: true,
6    });
7
8    const localVideo = document.getElementById('localVideo');
9    localVideo.srcObject = localStream;
10  } catch (error) {
11    console.error('Error accessing media devices:', error);
12  }
13}
14
15startVideo();
16
17

ICE Candidate Exchange

ICE (Interactive Connectivity Establishment) is a framework used to discover the best way to connect peers. ICE candidates are potential connection paths, including direct connections, connections through STUN servers, and connections through TURN servers.

javascript

1socket.on('iceCandidate', async (candidate, userId) => {
2    console.log("Received ICE candidate from " + userId);
3    try {
4        await peerConnection.addIceCandidate(new RTCIceCandidate(candidate));
5    } catch (e) {
6        console.error("Error adding received ice candidate", e);
7    }
8});
9
10async function createOffer(remoteUserId) {
11    const pc = await createPeerConnection(remoteUserId);
12    peerConnection = pc;
13    const offer = await peerConnection.createOffer();
14    await peerConnection.setLocalDescription(offer);
15
16    socket.emit("offer", offer, room);
17}
18
19socket.on("offer", async (offer, userId) => {
20    console.log("Received offer from " + userId);
21    const pc = await createPeerConnection(userId);
22    peerConnection = pc;
23    await peerConnection.setRemoteDescription(new RTCSessionDescription(offer));
24    const answer = await peerConnection.createAnswer();
25    await peerConnection.setLocalDescription(answer);
26
27    socket.emit("answer", answer, room);
28});
29
30
31socket.on("answer", async (answer, userId) => {
32    console.log("Received answer from " + userId);
33    await peerConnection.setRemoteDescription(new RTCSessionDescription(answer));
34});
35

Advanced WebRTC Techniques

Handling Multiple Peers

To handle multiple peers, you'll need to manage multiple RTCPeerConnection objects and associate each connection with a specific user. You can use a data structure (e.g., an object or map) to store the peer connections, indexed by user ID or socket ID. The signaling server needs to route offers, answers, and ICE candidates to the correct peer connections.

Implementing STUN and TURN Servers

STUN (Session Traversal Utilities for NAT) and TURN (Traversal Using Relays around NAT) servers are used to help peers behind NAT (Network Address Translation) firewalls to establish connections. STUN servers allow peers to discover their public IP address and port, while TURN servers relay traffic between peers when a direct connection is not possible.
To use STUN and TURN servers, you need to configure the iceServers property of the RTCPeerConnection object. You can use publicly available STUN servers or set up your own TURN server.

javascript

1const peerConnection = new RTCPeerConnection({
2    iceServers: [
3        { urls: 'stun:stun.l.google.com:19302' },
4        { urls: 'stun1.l.google.com:19302' },
5        // Add TURN servers here if needed
6        // { urls: 'turn:yourturnserver.com:3478', username: 'yourusername', credential: 'yourpassword' }
7    ],
8});
9

Improving Video Quality

Here are some techniques for improving video quality:
  • Constraints: Use getUserMedia constraints to specify the desired video resolution, frame rate, and other parameters. For example:

javascript

1const constraints = {
2    video: { width: { ideal: 1280 }, height: { ideal: 720 } },
3    audio: true,
4};
5
6localStream = await navigator.mediaDevices.getUserMedia(constraints);
7
  • Bandwidth Management: Implement bandwidth estimation and adaptation techniques to adjust the video bitrate based on network conditions.
  • Codec Selection: Use a video codec that is well-suited for real-time communication, such as VP8 or H.264.

Deploying your WebRTC Application

Choosing a Hosting Provider

You can deploy your WebRTC application to various hosting providers, such as:
  • Heroku: A popular platform for deploying Node.js applications.
  • Netlify: A platform for deploying static websites and single-page applications.
  • AWS: Amazon Web Services offers a wide range of hosting services.

Deployment Steps

  1. Prepare your application for deployment:
    • Make sure your code is clean and well-tested.
    • Specify the dependencies in your package.json file.
  2. Configure your hosting environment:
    • Set up a Node.js environment on your chosen hosting provider.
    • Configure environment variables (e.g., port number, API keys).
  3. Deploy your application:
    • Use the hosting provider's deployment tools to upload your code.
    • Start the application server.
  4. Configure SSL/TLS:
    • Obtain an SSL/TLS certificate for your domain.
    • Configure your web server to use the certificate.

Troubleshooting Common WebRTC Issues

  • No audio or video:
    • Check if the user has granted permission to access the camera and microphone.
    • Verify that the correct media devices are selected.
    • Check the browser's console for error messages.
  • Connection issues:
    • Check the network connectivity.
    • Ensure that STUN and TURN servers are configured correctly.
    • Check the browser's console for ICE errors.
  • Audio or video quality issues:
    • Check the bandwidth and network conditions.
    • Adjust the video resolution and frame rate.
    • Experiment with different video codecs.

Conclusion and Further Exploration

This tutorial provided a comprehensive introduction to building WebRTC video call applications. You've learned about the fundamental concepts of WebRTC, including signaling, peer connection establishment, and media stream handling.
To further explore WebRTC, consider the following resources:

Get 10,000 Free Minutes Every Months

No credit card required to start.

Want to level-up your learning? Subscribe now

Subscribe to our newsletter for more tech based insights

FAQ