webRTC Zero to Hero

Syllabus

Let's dive deep into Phase 1: Foundations of WebRTC and make sure you understand every corner of it.

1. What is WebRTC?

History and Evolution

Pre-WebRTC Era:
Before WebRTC, real-time communication relied on Flash, Java applets, and third-party plugins. These solutions had security issues, high latency, and poor browser support.
Google’s Acquisition & Open Source Contribution:
- In 2010, Google acquired Global IP Solutions (GIPS), a company specializing in real-time voice and video processing.
- In 2011, Google open-sourced WebRTC and submitted it to the W3C (World Wide Web Consortium) and IETF (Internet Engineering Task Force) to standardize the technology.
Standardization & Widespread Adoption:
- WebRTC became a W3C Recommendation in 2021, making it an official web standard.
- Now, WebRTC is widely used in applications like Google Meet, Discord, WhatsApp Web, Microsoft Teams, and more.

Why WebRTC? Use Cases and Benefits

Key Features

Peer-to-Peer (P2P) Communication: Direct connection between users, reducing latency.
Real-Time Media Streaming: Supports live video/audio without extra plugins.
Data Transfer: Exchange arbitrary data (files, messages) via RTCDataChannel.
End-to-End Encryption (E2EE): Ensures secure communication by using DTLS and SRTP.
Cross-Platform Support: Works on browsers, mobile apps, and servers.

Use Cases

Use Case	How WebRTC Helps
Video Calls	Enables peer-to-peer video/audio chat applications.
Live Streaming	Can be used to broadcast events with low latency.
Screen Sharing	Share screens without third-party software using `getDisplayMedia()`.
Online Gaming	WebRTC's `RTCDataChannel` allows real-time game state sync.
IoT & Surveillance	Enables live camera streaming from IoT devices.

Comparison with Alternatives (Sockets, WebSockets, RTP, SIP)

Feature	WebRTC	WebSockets	RTP	SIP
Real-Time Audio/Video	✅ Yes	❌ No	✅ Yes	❌ No
Peer-to-Peer	✅ Yes	❌ No (Client-Server)	✅ Yes	❌ No
Text & Binary Data	✅ Yes	✅ Yes	❌ No	❌ No
Requires a Server	❌ No (after signaling)	✅ Yes	✅ Yes	✅ Yes
Encryption	✅ Yes (DTLS/SRTP)	✅ Yes (TLS/SSL)	❌ No	✅ Yes

Sockets/WebSockets → Good for bidirectional text communication but cannot handle media streams.
RTP (Real-Time Transport Protocol) → Used in VoIP and video conferencing but requires SIP for session management.
SIP (Session Initiation Protocol) → Handles call setup and teardown, often used with RTP but lacks native browser support.
WebRTC combines the best of these technologies, offering peer-to-peer, real-time communication with encryption.

2. How WebRTC Works (Under the Hood)

WebRTC enables direct communication between users. However, establishing this direct connection is not simple due to NAT (Network Address Translation) and Firewalls. Let’s break it down step by step.

Client-to-Client Communication

The three major steps to establishing a WebRTC connection:

Signaling → Exchanging connection details using a signaling server.
ICE Candidate Exchange → Finding the best route for peer-to-peer connection.
Direct Peer-to-Peer Communication → Once established, media/data flows directly.

💡 WebRTC does not handle signaling! You must implement it using WebSockets, Firebase, MQTT, or any other communication protocol.

Role of Browsers and APIs

WebRTC is supported natively in browsers like Chrome, Firefox, Safari, Edge. The three core WebRTC APIs:

API	Purpose
`getUserMedia()`	Captures audio/video from the user’s camera/microphone.
`RTCPeerConnection`	Establishes a peer-to-peer connection and handles ICE candidates.
`RTCDataChannel`	Enables peer-to-peer data transfer (text, files, game state sync).

Understanding SDP (Session Description Protocol)

What is SDP?

SDP is not a protocol but a format used to exchange connection information between peers.
It contains details like IP addresses, ports, media codecs, encryption methods.

💡 Analogy:
Imagine SDP as a handshake protocol where two users say:

Alice: "I support H.264 video and Opus audio, can you understand that?"
Bob: "Yes, I also support H.264 and Opus, let's communicate!"

Example SDP Message (Simplified)

v=0
o=- 123456789 2 IN IP4 192.168.1.5
s=WebRTC Call
c=IN IP4 192.168.1.5
t=0 0
m=audio 5004 RTP/AVP 96
a=rtpmap:96 opus/48000
m=video 5006 RTP/AVP 97
a=rtpmap:97 H264/90000

m=audio → Specifies an audio stream.
m=video → Specifies a video stream.
a=rtpmap → Defines codecs like Opus (audio) and H.264 (video).

ICE (Interactive Connectivity Establishment)

ICE is a framework used to find the best way for two peers to connect.

Why do we need ICE?
- Devices can be behind NAT or firewalls, making direct connection difficult.
- ICE gathers different network candidates and tests them.

STUN, TURN, and NAT Traversal

STUN (Session Traversal Utilities for NAT)

Used to discover a device's public IP.
Helps in most cases when NAT is not restrictive.
Example Public STUN Server:
```
stun:stun.l.google.com:19302
```

TURN (Traversal Using Relays around NAT)

When direct connection fails, TURN relays traffic through a server.
This is slower but necessary in strict NAT/firewall setups.

Running a TURN Server (Coturn)

coturn --realm=mywebrtcapp.com --user=username:password

💡 Analogy:

STUN is like asking a friend for their home address and then going directly.
TURN is like sending messages through a middleman because you can't reach your friend directly.

Final Thoughts

WebRTC is not just video calls; it’s a complete framework for real-time communication.
Understanding SDP, ICE, STUN, and TURN is essential for mastering WebRTC.
Next, we’ll move into Phase 2: Core WebRTC APIs to start coding WebRTC apps in React & Node.js.

What’s Next?

👉 Do you want to start with code examples for getUserMedia() and RTCPeerConnection, or do you have any questions so far? 🚀

EXTRA:

📌 WebRTC Jargon & How It Works (Easy to Understand) 🚀

Think of WebRTC like a phone call between browsers without needing a middleman (like Zoom or Skype). But since browsers don’t know each other, they need a way to find, connect, and talk securely. That’s what WebRTC does.

🔑 Key WebRTC Jargon (Must-Know Terms!)

1️⃣ Signaling

👉 A messenger that helps two people exchange info to start a call.
💡 WebRTC DOESN’T handle signaling itself! You need WebSockets, Firebase, or another service to send messages.

👀 Example of a Signaling Message:

{
  "type": "offer",
  "sdp": "v=0..."
}

(Browser A sends this to Browser B to start a call)

2️⃣ RTCPeerConnection (Peer-to-Peer Connection)

👉 A pipe between two browsers for sending audio, video, and data.
💡 This is the most important WebRTC API!

👀 How to create a connection:

const peerConnection = new RTCPeerConnection();

3️⃣ ICE (Interactive Connectivity Establishment)

👉 A way for two people to find the best way to connect (like checking multiple roads to see which one works).
💡 Uses STUN/TURN servers to get around firewalls & NATs.

👀 How ICE candidates look:

{
  "candidate": "candidate:842163049 1 udp 1677729535 192.168.1.1 4000 typ host"
}

4️⃣ STUN & TURN Servers

🔹 STUN (Session Traversal Utilities for NAT)
👉 A helper server that tells you your public IP so others can connect to you.
💡 Fast & Free, but fails if strict firewalls block traffic.

🔹 TURN (Traversal Using Relays around NAT)
👉 A relay server that sends your media through itself if direct connection fails.
💡 Slower & Expensive, but always works.

👀 Google’s Free STUN Server:

const config = { iceServers: [{ urls: "stun:stun.l.google.com:19302" }] };

5️⃣ SDP (Session Description Protocol)

👉 A set of rules that tells two browsers how to talk (like picking a language before chatting).
💡 Contains IP, Port, Audio/Video Codecs, and other connection details.

👀 Example of an SDP Offer:

{
  "type": "offer",
  "sdp": "v=0\no=- 4611732547713599112 2 IN IP4 127.0.0.1..."
}

6️⃣ Tracks & Streams

🔹 MediaStream → The whole video/audio stream.
🔹 Track → A single audio or video inside the stream.

👀 Example:

navigator.mediaDevices.getUserMedia({ video: true, audio: true })
  .then(stream => {
    peerConnection.addTrack(stream.getTracks()[0], stream);
  });

7️⃣ RTCDataChannel

👉 A fast way to send text, files, and game data directly between peers.
💡 Think of it like WebSockets but faster!

👀 Example:

const dataChannel = peerConnection.createDataChannel("chat");
dataChannel.send("Hello, WebRTC!");

📌 WebRTC Flow (How It Works Step-by-Step!)

1️⃣ User requests video/audio → (getUserMedia())
2️⃣ Peers exchange signaling messages (via WebSockets, Firebase, etc.)
3️⃣ Offer & Answer are exchanged (using SDP)
4️⃣ ICE Candidates are exchanged (STUN/TURN helps find the best route)
5️⃣ RTCPeerConnection is established (Media/Data start flowing)
6️⃣ Connection stays active (Handles network changes & packet loss)
7️⃣ Call Ends (Connection is closed)

🔥 This is the foundation of WebRTC. Master these and you're already ahead of most developers! 🚀

Let's dive into Phase 2: Core WebRTC APIs and Internals in detail. We'll cover each API with its internal working, usage, and practical examples to ensure you master WebRTC at a deep level.

3. Media Capture (`getUserMedia()` API)

How Browsers Capture Video/Audio

The getUserMedia() API allows a web app to access the user's microphone and camera. It returns a MediaStream, which contains audio/video tracks.

💡 Under the Hood:

The browser asks for user permission before accessing media.
Once granted, it creates tracks (video/audio) and sends them as a stream.
The stream can be played in a <video> element, processed, or sent over WebRTC.

Constraints and Permissions

Basic Usage

navigator.mediaDevices.getUserMedia({ video: true, audio: true })
  .then((stream) => {
    document.getElementById("videoElement").srcObject = stream;
  })
  .catch((error) => console.error("Error accessing media:", error));

Custom Constraints

const constraints = {
  video: {
    width: { min: 640, ideal: 1280, max: 1920 },
    height: { min: 360, ideal: 720, max: 1080 },
    frameRate: { max: 30 },
    facingMode: "user", // 'environment' for back camera
  },
  audio: {
    echoCancellation: true,
    noiseSuppression: true,
    autoGainControl: true,
  }
};

navigator.mediaDevices.getUserMedia(constraints)
  .then((stream) => { /* Use stream */ })
  .catch((error) => console.error("Error:", error));

Constraint	Description
`width`, `height`	Defines resolution (min, max, ideal).
`frameRate`	Max FPS (useful for smoother streaming).
`facingMode`	`"user"` (front camera), `"environment"` (back).
`echoCancellation`, `noiseSuppression`	Improves audio quality.

Processing Streams (Filters, Effects)

1️⃣ Applying CSS Filters to Video

<video id="videoElement" autoplay style="filter: grayscale(100%)"></video>

2️⃣ Applying a WebGL Filter (Green Screen)

const video = document.getElementById("videoElement");
const canvas = document.createElement("canvas");
const ctx = canvas.getContext("2d");

navigator.mediaDevices.getUserMedia({ video: true }).then((stream) => {
  video.srcObject = stream;
  requestAnimationFrame(updateCanvas);
});

function updateCanvas() {
  ctx.drawImage(video, 0, 0, canvas.width, canvas.height);
  let frame = ctx.getImageData(0, 0, canvas.width, canvas.height);
  
  for (let i = 0; i < frame.data.length; i += 4) {
    let green = frame.data[i + 1]; // Green channel
    if (green > 100) { frame.data[i + 3] = 0; } // Make green pixels transparent
  }
  
  ctx.putImageData(frame, 0, 0);
  requestAnimationFrame(updateCanvas);
}

🔹 This removes the green screen (chroma keying).

4. Peer Connection (`RTCPeerConnection`)

Creating a Connection

The RTCPeerConnection API is used to establish a secure peer-to-peer connection.

Basic Flow

1️⃣ Caller creates an RTCPeerConnection instance.
2️⃣ Caller creates an SDP offer and sends it to the callee.
3️⃣ Callee receives the offer, creates an SDP answer, and sends it back.
4️⃣ Both exchange ICE candidates to establish a connection.

Basic Example (Caller)

const peerConnection = new RTCPeerConnection();

// Add local stream to connection
navigator.mediaDevices.getUserMedia({ video: true, audio: true })
  .then((stream) => {
    stream.getTracks().forEach(track => peerConnection.addTrack(track, stream));
  });

// Create and send SDP Offer
peerConnection.createOffer().then((offer) => {
  return peerConnection.setLocalDescription(offer);
}).then(() => {
  sendSignalingData({ type: "offer", sdp: peerConnection.localDescription });
});

Basic Example (Callee)

const peerConnection = new RTCPeerConnection();

receiveSignalingData((data) => {
  if (data.type === "offer") {
    peerConnection.setRemoteDescription(new RTCSessionDescription(data.sdp));

    peerConnection.createAnswer().then((answer) => {
      return peerConnection.setLocalDescription(answer);
    }).then(() => {
      sendSignalingData({ type: "answer", sdp: peerConnection.localDescription });
    });
  }
});

Handling ICE Candidates

WebRTC needs ICE candidates to establish a stable connection.

peerConnection.onicecandidate = (event) => {
  if (event.candidate) {
    sendSignalingData({ type: "candidate", candidate: event.candidate });
  }
};

On the receiver side:

receiveSignalingData((data) => {
  if (data.type === "candidate") {
    peerConnection.addIceCandidate(new RTCIceCandidate(data.candidate));
  }
});

Understanding SRTP (Secure Real-Time Transport Protocol)

WebRTC encrypts media streams using DTLS-SRTP.
SRTP secures audio/video streams, preventing eavesdropping.
Encryption keys are exchanged via DTLS (Datagram Transport Layer Security).

5. Data Channels (`RTCDataChannel`)

WebRTC supports peer-to-peer data transfer (text, files, game data).

Sending Text and Binary Data

Sender

const dataChannel = peerConnection.createDataChannel("chat");

// Send messages
dataChannel.send("Hello, WebRTC!");

Receiver

peerConnection.ondatachannel = (event) => {
  const dataChannel = event.channel;

  dataChannel.onmessage = (msgEvent) => {
    console.log("Received:", msgEvent.data);
  };
};

Reliability and Ordering

Mode	Reliable?	Ordered?	Use Case
`reliable: true`	✅ Yes	✅ Yes	File transfer, messaging
`reliable: false`	❌ No	❌ No	Live gaming, real-time telemetry
`ordered: false`	✅ Yes	❌ No	VoIP, video frames

const dataChannel = peerConnection.createDataChannel("game", {
  ordered: false, // Packet loss is acceptable
  maxRetransmits: 3 // Retransmit up to 3 times
});

Use Cases (Gaming, Messaging)

1️⃣ Low-latency multiplayer games (e.g., real-time state updates).
2️⃣ Decentralized chat apps (without a central server).
3️⃣ Live file sharing (P2P).

Final Thoughts

You’ve now mastered WebRTC’s core APIs 🚀.
Next, we’ll move into Phase 3: Advanced WebRTC Features (Screen Sharing, Simulcast, Mesh vs. SFU, Recording).

Now, let’s move into Phase 3: WebRTC Signaling & Communication, where we cover how peers discover and connect to each other.

6. Understanding Signaling Mechanisms

WebRTC provides peer-to-peer communication, but signaling (coordination) happens outside WebRTC.

Why Signaling?

Since WebRTC does not handle signaling, we need a mechanism to:
✅ Exchange Session Descriptions (SDP) – Helps peers agree on audio/video formats.
✅ Send ICE Candidates – Finds the best network path for communication.
✅ Detect Disconnections – Handles reconnections when needed.

💡 Think of Signaling as the "handshake" before direct communication begins.

Implementing Signaling with WebSockets

🔹 WebSockets provide real-time, bidirectional communication over TCP.

Client-Side WebSocket Code

const socket = new WebSocket("wss://your-server.com");

socket.onopen = () => console.log("Connected to signaling server");

socket.onmessage = (message) => {
  const data = JSON.parse(message.data);
  if (data.type === "offer") {
    handleOffer(data.offer);
  } else if (data.type === "answer") {
    handleAnswer(data.answer);
  } else if (data.type === "candidate") {
    handleNewICECandidate(data.candidate);
  }
};

function sendMessage(message) {
  socket.send(JSON.stringify(message));
}

Alternative Signaling Approaches

🔸 REST API – Works but not real-time. Requires polling.
🔸 Firebase Realtime Database – Good for small projects.
🔸 MQTT (Message Queue Telemetry Transport) – Used in IoT but not optimal for WebRTC.

Example of using Firebase as a signaling server:

import { ref, set, onValue } from "firebase/database";

const signalingRef = ref(database, "signaling");

function sendSignalingMessage(type, data) {
  set(signalingRef, { type, data });
}

onValue(signalingRef, (snapshot) => {
  const message = snapshot.val();
  if (message?.type === "offer") handleOffer(message.data);
});

📌 WebSockets are the most recommended option for production apps.

7. Setting Up a Signaling Server (Node.js + WebSockets)

Let's build a real-time signaling server using WebSockets and Node.js.

Step 1: Install Dependencies

npm init -y
npm install ws express

Step 2: Create the WebSocket Server (`server.js`)

const WebSocket = require("ws");
const server = new WebSocket.Server({ port: 8080 });

let clients = [];

server.on("connection", (socket) => {
  clients.push(socket);

  socket.on("message", (message) => {
    clients.forEach((client) => {
      if (client !== socket && client.readyState === WebSocket.OPEN) {
        client.send(message);
      }
    });
  });

  socket.on("close", () => {
    clients = clients.filter((client) => client !== socket);
  });
});

console.log("WebSocket signaling server running on ws://localhost:8080");

🔹 This broadcasts signaling messages to all connected clients.

Step 3: Exchanging SDP Messages

📌 The WebRTC flow works like this:
1️⃣ Caller sends an SDP offer via WebSockets.
2️⃣ Callee responds with an SDP answer.
3️⃣ Both peers exchange ICE candidates.

Client-Side Code (Caller)

const peer = new RTCPeerConnection();

peer.onicecandidate = (event) => {
  if (event.candidate) {
    sendMessage({ type: "candidate", candidate: event.candidate });
  }
};

// Create an offer
peer.createOffer().then((offer) => {
  return peer.setLocalDescription(offer);
}).then(() => {
  sendMessage({ type: "offer", offer: peer.localDescription });
});

Client-Side Code (Callee)

socket.onmessage = (message) => {
  const data = JSON.parse(message.data);

  if (data.type === "offer") {
    peer.setRemoteDescription(new RTCSessionDescription(data.offer));

    peer.createAnswer().then((answer) => {
      return peer.setLocalDescription(answer);
    }).then(() => {
      sendMessage({ type: "answer", answer: peer.localDescription });
    });
  } else if (data.type === "candidate") {
    peer.addIceCandidate(new RTCIceCandidate(data.candidate));
  }
};

📌 Once ICE candidates are exchanged, WebRTC will connect the peers directly! 🎉

8. Connecting Peers via STUN & TURN

💡 STUN and TURN help bypass network restrictions (NAT, Firewalls).

Server Type	Function
STUN (Session Traversal Utilities for NAT)	Finds public IP address, helps NAT traversal.
TURN (Traversal Using Relays around NAT)	Relays traffic when direct connection fails.

🔹 STUN is lightweight but doesn’t work for restrictive firewalls (Carrier-Grade NAT).
🔹 TURN is slower but guarantees connectivity by relaying media.

Implementing a Public STUN Server

WebRTC provides Google’s free STUN servers:

const peer = new RTCPeerConnection({
  iceServers: [{ urls: "stun:stun.l.google.com:19302" }]
});

Setting Up a Private TURN Server

🔹 For production, set up a TURN server (e.g., Coturn).

Step 1: Install Coturn

sudo apt update && sudo apt install coturn

Step 2: Configure Coturn (`/etc/turnserver.conf`)

lt-cred-mech
use-auth-secret
static-auth-secret=your_secret_key
realm=yourdomain.com
listening-port=3478

Step 3: Restart and Enable Coturn

sudo systemctl restart coturn
sudo systemctl enable coturn

Step 4: Use Private TURN Server in WebRTC

const peer = new RTCPeerConnection({
  iceServers: [
    { urls: "stun:stun.l.google.com:19302" }, // Public STUN
    { urls: "turn:your-turn-server.com", username: "user", credential: "pass" } // Private TURN
  ]
});

📌 TURN servers ensure connectivity but require hosting and authentication.

Summary of Phase 3

✅ Built a WebSocket-based Signaling Server.
✅ Implemented SDP and ICE Candidate Exchange.
✅ Understood STUN/TURN for NAT Traversal.
✅ Set up a Private TURN Server for Production.

Next: Phase 4 - Advanced WebRTC Features 🚀

🔹 Screen Sharing
🔹 Recording WebRTC Streams
🔹 SFU vs. Mesh Networks

Now, let's dive into Phase 4: Advanced WebRTC Features 🚀

9. Media Processing & Manipulation

Applying Filters (Blur, Effects)

🔹 WebRTC allows real-time video processing using Canvas & WebGL.

Example: Apply a Blur Effect Using Canvas

const video = document.querySelector("video");
const canvas = document.createElement("canvas");
const ctx = canvas.getContext("2d");

function applyBlurEffect() {
  canvas.width = video.videoWidth;
  canvas.height = video.videoHeight;

  ctx.filter = "blur(5px)";
  ctx.drawImage(video, 0, 0, canvas.width, canvas.height);
  
  requestAnimationFrame(applyBlurEffect);
}

video.addEventListener("play", applyBlurEffect);

💡 Use WebGL shaders for more advanced effects like background removal.

Screen Sharing (getDisplayMedia API)

🔹 WebRTC provides getDisplayMedia() to share the screen.

Example: Capture and Stream the Screen

async function startScreenShare() {
  try {
    const stream = await navigator.mediaDevices.getDisplayMedia({ video: true });
    document.querySelector("video").srcObject = stream;
  } catch (error) {
    console.error("Screen share error:", error);
  }
}

📌 For security reasons, browsers require user permission for screen sharing.

Recording Streams (MediaRecorder API)

🔹 WebRTC allows recording both local and remote streams.

Example: Record and Download Video

let recorder;
let recordedChunks = [];

function startRecording(stream) {
  recorder = new MediaRecorder(stream);

  recorder.ondataavailable = (event) => recordedChunks.push(event.data);
  
  recorder.onstop = () => {
    const blob = new Blob(recordedChunks, { type: "video/webm" });
    const url = URL.createObjectURL(blob);
    const a = document.createElement("a");
    a.href = url;
    a.download = "recorded-video.webm";
    a.click();
  };

  recorder.start();
}

📌 Use WebRTC recording for saving meetings, lectures, or gameplay.

10. Multi-Peer Connections (SFU & Mesh Networks)

Group Video Calls (Mesh vs. SFU vs. MCU)

🔹 Three architectures for multi-party WebRTC calls:

Architecture	How it Works	Pros	Cons
Mesh	Each peer connects directly to every other peer.	Simple, no servers.	High bandwidth usage.
SFU (Selective Forwarding Unit)	A central server forwards streams to all participants.	Efficient bandwidth, low latency.	Requires a media server.
MCU (Multipoint Control Unit)	Server mixes video streams and sends a single combined stream to users.	Low CPU usage for clients.	High server costs.

Implementing SFU with Mediasoup

🔹 Mediasoup is a popular SFU for WebRTC-based applications.

Step 1: Install Mediasoup

npm install mediasoup

Step 2: Create an SFU Server

const mediasoup = require("mediasoup");

async function createSFU() {
  const worker = await mediasoup.createWorker();
  const router = await worker.createRouter({
    mediaCodecs: [{ kind: "video", mimeType: "video/VP8" }],
  });

  console.log("SFU is running...");
}

createSFU();

📌 SFU servers relay media, reducing peer-to-peer bandwidth consumption.

11. Optimizing WebRTC for Performance

Bandwidth Estimation & Adaptation

🔹 WebRTC dynamically adjusts quality based on network conditions.

How to Set Bandwidth Limits

const sender = peerConnection.getSenders().find((s) => s.track.kind === "video");

const params = sender.getParameters();
params.encodings[0].maxBitrate = 500000; // 500 kbps
sender.setParameters(params);

📌 This prevents WebRTC from using too much bandwidth.

Handling High Latency & Packet Loss

🔹 WebRTC uses Forward Error Correction (FEC) and Retransmissions (RTX) to recover lost packets.

Enable RED and ULPFEC (Error Correction)

const peer = new RTCPeerConnection({
  iceServers: [{ urls: "stun:stun.l.google.com:19302" }],
  encodedInsertableStreams: true,
});

📌 This improves video quality in poor network conditions.

Summary of Phase 4

✅ Added real-time filters to WebRTC streams.
✅ Implemented screen sharing & media recording.
✅ Explored Mesh, SFU, and MCU for multi-peer calls.
✅ Optimized WebRTC performance with bandwidth control & error correction.

Next: Phase 5 - Building a Production-Ready WebRTC App 🚀

🔹 Deploying a WebRTC App
🔹 Security Best Practices
🔹 Integrating WebRTC with React

Now, let's tackle Phase 5: Security, Deployment & Scaling 🔐🚀

12. Security Best Practices in WebRTC

🔹 Encryption in WebRTC (DTLS, SRTP)

🔐 WebRTC encrypts all media and data by default using:
✅ DTLS (Datagram Transport Layer Security) → Encrypts signaling
✅ SRTP (Secure Real-time Transport Protocol) → Encrypts media streams

How to Ensure Encryption is Enabled

const peer = new RTCPeerConnection({
  iceServers: [{ urls: "stun:stun.l.google.com:19302" }],
  certificates: [RTCCertificate.generateCertificate({ name: "ECDSA", namedCurve: "P-256" })]
});

📌 This ensures the connection uses a secure DTLS certificate.

🔹 Secure Signaling

Even though WebRTC encrypts media, signaling is not encrypted by default! 🚨
Solution:
✅ Use WSS (WebSocket Secure) instead of WS
✅ Encrypt SDP messages
✅ Prevent unauthorized access with authentication

Example: Secure WebSocket Signaling

const wss = new WebSocket("wss://your-secure-server.com");
wss.onmessage = (message) => {
  const data = JSON.parse(message.data);
  if (data.sdp) {
    peerConnection.setRemoteDescription(new RTCSessionDescription(data.sdp));
  }
};

📌 Never expose raw SDP or ICE candidates to untrusted users.

🔹 Preventing IP Leaks

By default, WebRTC can expose your private IP address due to ICE candidate discovery.
Solution:
✅ Disable non-essential ICE candidates
✅ Use a TURN server to mask the real IP

How to Block Local IP Leaks

const peer = new RTCPeerConnection({
  iceServers: [{ urls: "turn:your-turn-server.com", username: "user", credential: "pass" }],
  iceTransportPolicy: "relay"  // Forces WebRTC to use TURN
});

📌 This prevents WebRTC from revealing local network IP addresses.

13. Deploying a WebRTC Application

🔹 Hosting a WebRTC Server

To deploy a WebRTC app, you need:
✅ Frontend Hosting (Vercel, Netlify, S3, Firebase Hosting)
✅ Signaling Server (Node.js with WebSockets)
✅ TURN/STUN Servers (Coturn, Twilio, Google STUN)

Example: Deploying a WebRTC Signaling Server with PM2

npm install -g pm2
pm2 start server.js --name webrtc-signaling
pm2 save
pm2 startup

📌 PM2 keeps your signaling server running even after crashes.

🔹 Scaling WebRTC (Load Balancing, Cloud Solutions)

Challenges in Scaling WebRTC:
❌ Peer-to-peer connections don’t scale well
❌ Too many clients → High bandwidth usage
❌ TURN servers get overloaded

Solutions:
✅ Use an SFU (Selective Forwarding Unit) like Mediasoup
✅ Deploy TURN/STUN on AWS or DigitalOcean
✅ Use WebRTC Load Balancing with Cloudflare or Nginx

How to Load Balance a TURN Server

coturn -c /etc/turnserver.conf --min-port=49152 --max-port=65535

📌 Use multiple TURN servers to distribute the load.

14. Debugging & Troubleshooting WebRTC

🔹 Using Chrome’s WebRTC Internals

✅ Open chrome://webrtc-internals/
✅ Inspect ICE Candidates, SDP, and DTLS Handshakes
✅ Identify packet loss, latency, and jitter

Example: Debugging ICE Candidate Errors

chrome://webrtc-internals/

📌 If ICE fails, check if your TURN server is misconfigured.

🔹 Common Issues & Fixes

Issue	Cause	Fix
No audio/video	Media permissions blocked	Ensure `getUserMedia()` permissions are granted
ICE Connection Failed	Firewall blocking UDP	Use a TURN server
High latency in calls	Poor network conditions	Reduce bitrate & enable error correction
Video lagging/freezing	CPU overload	Optimize video encoding & resolution

📌 Summary of Phase 5

✅ WebRTC security: Encryption, Secure Signaling, IP Leak Prevention
✅ Deployed WebRTC with PM2, Coturn, and Cloudflare
✅ Used Chrome’s WebRTC Internals to debug errors
✅ Learned fixes for common WebRTC issues

Phase 6: Real-World Project & Beyond

Let's build a 1:1 Video Calling App using React (Frontend) + Node.js (Backend) + WebRTC + WebSockets from scratch. 🚀

We'll follow these steps:
✅ Set up a React frontend for video streaming 📹
✅ Build a Node.js WebSocket signaling server 🔄
✅ Connect two users with WebRTC peer-to-peer communication 🔗

📌 Step 1: Setup the React Frontend

Install Dependencies

npx create-react-app webrtc-app
cd webrtc-app
npm install socket.io-client

📌 Step 2: Implement Video Streaming in React

Create a WebRTCComponent.js file:

import React, { useEffect, useRef, useState } from "react";
import io from "socket.io-client";

const socket = io("http://localhost:5000"); // Connect to WebSocket Server

const WebRTCComponent = () => {
  const localVideoRef = useRef(null);
  const remoteVideoRef = useRef(null);
  const [peerConnection, setPeerConnection] = useState(null);

  useEffect(() => {
    // Get User Media
    navigator.mediaDevices.getUserMedia({ video: true, audio: true })
      .then((stream) => {
        localVideoRef.current.srcObject = stream;

        const pc = new RTCPeerConnection({
          iceServers: [{ urls: "stun:stun.l.google.com:19302" }]
        });

        // Add local stream to peer connection
        stream.getTracks().forEach((track) => pc.addTrack(track, stream));

        setPeerConnection(pc);

        // Handle ICE Candidate Exchange
        pc.onicecandidate = (event) => {
          if (event.candidate) {
            socket.emit("ice-candidate", event.candidate);
          }
        };

        // Receive Remote Stream
        pc.ontrack = (event) => {
          remoteVideoRef.current.srcObject = event.streams[0];
        };

        // Listen for SDP Offers
        socket.on("offer", async (offer) => {
          await pc.setRemoteDescription(new RTCSessionDescription(offer));
          const answer = await pc.createAnswer();
          await pc.setLocalDescription(answer);
          socket.emit("answer", answer);
        });

        // Listen for SDP Answers
        socket.on("answer", async (answer) => {
          await pc.setRemoteDescription(new RTCSessionDescription(answer));
        });

        // Listen for ICE Candidates
        socket.on("ice-candidate", async (candidate) => {
          try {
            await pc.addIceCandidate(new RTCIceCandidate(candidate));
          } catch (error) {
            console.error("Error adding ICE candidate:", error);
          }
        });

      })
      .catch((error) => console.error("Error accessing media devices:", error));
  }, []);

  // Function to Start a Call
  const startCall = async () => {
    if (!peerConnection) return;
    const offer = await peerConnection.createOffer();
    await peerConnection.setLocalDescription(offer);
    socket.emit("offer", offer);
  };

  return (
    <div>
      <h1>WebRTC 1:1 Video Call</h1>
      <video ref={localVideoRef} autoPlay playsInline muted />
      <video ref={remoteVideoRef} autoPlay playsInline />
      <button onClick={startCall}>Start Call</button>
    </div>
  );
};

export default WebRTCComponent;

📌 Step 3: Setup the WebSocket Signaling Server

Install Dependencies

mkdir server
cd server
npm init -y
npm install express socket.io cors

📌 Step 4: Implement the WebSocket Server

Create server.js inside the server folder:

const express = require("express");
const http = require("http");
const { Server } = require("socket.io");
const cors = require("cors");

const app = express();
const server = http.createServer(app);
const io = new Server(server, {
  cors: { origin: "http://localhost:3000", methods: ["GET", "POST"] }
});

io.on("connection", (socket) => {
  console.log("New client connected:", socket.id);

  socket.on("offer", (offer) => {
    socket.broadcast.emit("offer", offer);
  });

  socket.on("answer", (answer) => {
    socket.broadcast.emit("answer", answer);
  });

  socket.on("ice-candidate", (candidate) => {
    socket.broadcast.emit("ice-candidate", candidate);
  });

  socket.on("disconnect", () => {
    console.log("Client disconnected");
  });
});

server.listen(5000, () => console.log("Server running on port 5000"));

📌 Step 5: Run the Application

Start the Server

cd server
node server.js

Start the React Frontend

cd webrtc-app
npm start

📌 Step 6: Test the WebRTC 1:1 Video Call

Open http://localhost:3000 in two browser tabs.
Click the "Start Call" button in one tab.
The other tab should receive the call and connect.

🔥 What’s Next?

✅ Basic 1:1 Video Calling is working! 🚀
➡️ Want to add Screen Sharing, Group Calls, or Recording? 🎥

📌 Implementing Group Video Calls with WebRTC + React + Node.js

We'll extend our 1:1 WebRTC setup to support multiple users using Mesh Topology (each peer connects to every other peer).

🌟 Steps to Implement Group Video Calling

Update the Frontend – Allow multiple video streams
Modify WebRTC Logic – Support multiple peer connections
Update the Signaling Server – Handle multiple users

📌 Step 1: Update the React Frontend

Install dependencies if not already installed

npm install socket.io-client

🔹 Create a New Component: `GroupCall.js`

This component manages multiple peer connections and displays multiple video streams.

import React, { useEffect, useRef, useState } from "react";
import io from "socket.io-client";

const socket = io("http://localhost:5000"); // Connect to WebSocket Server
const ICE_SERVERS = [{ urls: "stun:stun.l.google.com:19302" }];

const GroupCall = () => {
  const [peers, setPeers] = useState({});
  const [stream, setStream] = useState(null);
  const videoRefs = useRef({}); // Store video elements dynamically

  useEffect(() => {
    // Get User Media
    navigator.mediaDevices.getUserMedia({ video: true, audio: true })
      .then((mediaStream) => {
        setStream(mediaStream);
        socket.emit("join-room"); // Notify server that we joined
      })
      .catch((error) => console.error("Media access error:", error));

    // Handle New User Connection
    socket.on("user-joined", (userId) => {
      const peerConnection = createPeerConnection(userId);
      setPeers((prev) => ({ ...prev, [userId]: peerConnection }));
    });

    // Handle Incoming Offers
    socket.on("offer", async ({ senderId, offer }) => {
      const peerConnection = createPeerConnection(senderId);
      await peerConnection.setRemoteDescription(new RTCSessionDescription(offer));
      const answer = await peerConnection.createAnswer();
      await peerConnection.setLocalDescription(answer);
      socket.emit("answer", { receiverId: senderId, answer });
      setPeers((prev) => ({ ...prev, [senderId]: peerConnection }));
    });

    // Handle Incoming Answers
    socket.on("answer", ({ senderId, answer }) => {
      peers[senderId]?.setRemoteDescription(new RTCSessionDescription(answer));
    });

    // Handle Incoming ICE Candidates
    socket.on("ice-candidate", ({ senderId, candidate }) => {
      peers[senderId]?.addIceCandidate(new RTCIceCandidate(candidate));
    });

    // Handle User Disconnection
    socket.on("user-left", (userId) => {
      peers[userId]?.close();
      setPeers((prev) => {
        const updatedPeers = { ...prev };
        delete updatedPeers[userId];
        return updatedPeers;
      });
    });

    return () => {
      socket.off("user-joined");
      socket.off("offer");
      socket.off("answer");
      socket.off("ice-candidate");
      socket.off("user-left");
    };
  }, []);

  // Function to Create a New Peer Connection
  const createPeerConnection = (userId) => {
    const peerConnection = new RTCPeerConnection({ iceServers: ICE_SERVERS });

    // Add local stream to the peer connection
    if (stream) {
      stream.getTracks().forEach((track) => peerConnection.addTrack(track, stream));
    }

    // Handle ICE Candidates
    peerConnection.onicecandidate = (event) => {
      if (event.candidate) {
        socket.emit("ice-candidate", { receiverId: userId, candidate: event.candidate });
      }
    };

    // Handle Incoming Tracks
    peerConnection.ontrack = (event) => {
      videoRefs.current[userId].srcObject = event.streams[0];
    };

    return peerConnection;
  };

  // Function to Start a Call (Only Used by First User)
  const startCall = async () => {
    for (const userId in peers) {
      const offer = await peers[userId].createOffer();
      await peers[userId].setLocalDescription(offer);
      socket.emit("offer", { receiverId: userId, offer });
    }
  };

  return (
    <div>
      <h1>Group Video Call</h1>
      <video ref={(el) => (videoRefs.current["self"] = el)} autoPlay playsInline muted />
      {Object.keys(peers).map((userId) => (
        <video key={userId} ref={(el) => (videoRefs.current[userId] = el)} autoPlay playsInline />
      ))}
      <button onClick={startCall}>Start Call</button>
    </div>
  );
};

export default GroupCall;

📌 Step 2: Modify the WebSocket Signaling Server

Update server.js to handle multiple users.

const express = require("express");
const http = require("http");
const { Server } = require("socket.io");
const cors = require("cors");

const app = express();
const server = http.createServer(app);
const io = new Server(server, {
  cors: { origin: "http://localhost:3000", methods: ["GET", "POST"] }
});

let users = {}; // Store connected users

io.on("connection", (socket) => {
  console.log(`User connected: ${socket.id}`);

  socket.on("join-room", () => {
    users[socket.id] = socket;
    socket.broadcast.emit("user-joined", socket.id);
  });

  socket.on("offer", ({ receiverId, offer }) => {
    if (users[receiverId]) {
      users[receiverId].emit("offer", { senderId: socket.id, offer });
    }
  });

  socket.on("answer", ({ receiverId, answer }) => {
    if (users[receiverId]) {
      users[receiverId].emit("answer", { senderId: socket.id, answer });
    }
  });

  socket.on("ice-candidate", ({ receiverId, candidate }) => {
    if (users[receiverId]) {
      users[receiverId].emit("ice-candidate", { senderId: socket.id, candidate });
    }
  });

  socket.on("disconnect", () => {
    console.log(`User disconnected: ${socket.id}`);
    delete users[socket.id];
    socket.broadcast.emit("user-left", socket.id);
  });
});

server.listen(5000, () => console.log("Server running on port 5000"));

📌 Step 3: Run the Application

Start the Signaling Server:

cd server
node server.js

Start the React Frontend:

cd webrtc-app
npm start

🔥 How It Works

✅ User joins the room → Notifies others
✅ Each user creates a WebRTC connection with every other peer
✅ ICE candidates and SDP are exchanged via WebSockets
✅ Multiple video streams appear dynamically

✨ What’s Next?

Optimize Performance: Switch from Mesh Topology to SFU (Selective Forwarding Unit)
Enhance UI: Display user names, mute/unmute, UI improvements
Add Screen Sharing & Recording

🎉 Congratulations! You've successfully built a Real-Time Video Calling App!

📌 Implementing Live Streaming with WebRTC

Live streaming with WebRTC involves one broadcaster (host) and multiple viewers (audience). Unlike peer-to-peer (P2P) calls, a more efficient approach is to use Selective Forwarding Units (SFU) instead of Mesh Topology, allowing a single broadcaster to stream to multiple viewers.

🌟 Architecture for WebRTC Live Streaming

Broadcaster captures media using getUserMedia() and sends it to an SFU/WebRTC server.
Server (SFU) forwards the stream to multiple viewers using WebRTC.
Viewers receive the stream via RTCPeerConnection and display it.

📌 Why SFU and Not Mesh?

Mesh: High bandwidth usage, not scalable.
SFU: Efficient, single upstream from broadcaster, multiple downstreams to viewers.

📌 Step 1: Install Dependencies

npm install socket.io socket.io-client express cors

📌 Step 2: Create the WebRTC Server (SFU)

📌 server.js (Signaling + Media Relay)

const express = require("express");
const http = require("http");
const { Server } = require("socket.io");
const cors = require("cors");

const app = express();
app.use(cors());

const server = http.createServer(app);
const io = new Server(server, {
  cors: { origin: "http://localhost:3000", methods: ["GET", "POST"] }
});

let broadcaster = null;
let viewers = {};

io.on("connection", (socket) => {
  console.log("User connected:", socket.id);

  socket.on("broadcaster", () => {
    broadcaster = socket.id;
    socket.broadcast.emit("broadcaster");
  });

  socket.on("watcher", () => {
    if (broadcaster) {
      viewers[socket.id] = socket;
      socket.emit("broadcaster", broadcaster);
    }
  });

  socket.on("offer", ({ receiverId, offer }) => {
    if (viewers[receiverId]) {
      viewers[receiverId].emit("offer", { senderId: socket.id, offer });
    }
  });

  socket.on("answer", ({ receiverId, answer }) => {
    if (broadcaster === receiverId) {
      io.to(receiverId).emit("answer", { senderId: socket.id, answer });
    }
  });

  socket.on("ice-candidate", ({ receiverId, candidate }) => {
    io.to(receiverId).emit("ice-candidate", { senderId: socket.id, candidate });
  });

  socket.on("disconnect", () => {
    if (broadcaster === socket.id) {
      broadcaster = null;
      io.emit("broadcaster-disconnected");
    } else {
      delete viewers[socket.id];
    }
  });
});

server.listen(5000, () => console.log("Server running on port 5000"));

📌 Step 3: Create Broadcaster Component in React

📌 Broadcaster.js

import React, { useEffect, useRef, useState } from "react";
import io from "socket.io-client";

const socket = io("http://localhost:5000");

const Broadcaster = () => {
  const videoRef = useRef(null);
  const [peerConnections, setPeerConnections] = useState({});
  const ICE_SERVERS = [{ urls: "stun:stun.l.google.com:19302" }];

  useEffect(() => {
    navigator.mediaDevices.getUserMedia({ video: true, audio: true }).then((stream) => {
      videoRef.current.srcObject = stream;
      socket.emit("broadcaster");

      socket.on("watcher", (viewerId) => {
        const peerConnection = new RTCPeerConnection({ iceServers: ICE_SERVERS });

        stream.getTracks().forEach((track) => peerConnection.addTrack(track, stream));

        peerConnection.onicecandidate = (event) => {
          if (event.candidate) {
            socket.emit("ice-candidate", { receiverId: viewerId, candidate: event.candidate });
          }
        };

        peerConnection.createOffer().then((offer) => {
          peerConnection.setLocalDescription(offer);
          socket.emit("offer", { receiverId: viewerId, offer });
        });

        setPeerConnections((prev) => ({ ...prev, [viewerId]: peerConnection }));
      });

      socket.on("answer", ({ senderId, answer }) => {
        peerConnections[senderId]?.setRemoteDescription(new RTCSessionDescription(answer));
      });

      socket.on("ice-candidate", ({ senderId, candidate }) => {
        peerConnections[senderId]?.addIceCandidate(new RTCIceCandidate(candidate));
      });

      socket.on("broadcaster-disconnected", () => {
        Object.values(peerConnections).forEach((pc) => pc.close());
        setPeerConnections({});
      });
    });

    return () => socket.disconnect();
  }, []);

  return (
    <div>
      <h1>Live Stream (Broadcaster)</h1>
      <video ref={videoRef} autoPlay playsInline muted />
    </div>
  );
};

export default Broadcaster;

📌 Step 4: Create Viewer Component in React

📌 Viewer.js

import React, { useEffect, useRef, useState } from "react";
import io from "socket.io-client";

const socket = io("http://localhost:5000");

const Viewer = () => {
  const videoRef = useRef(null);
  const [peerConnection, setPeerConnection] = useState(null);
  const ICE_SERVERS = [{ urls: "stun:stun.l.google.com:19302" }];

  useEffect(() => {
    socket.emit("watcher");

    socket.on("broadcaster", (broadcasterId) => {
      const peerConnection = new RTCPeerConnection({ iceServers: ICE_SERVERS });

      peerConnection.onicecandidate = (event) => {
        if (event.candidate) {
          socket.emit("ice-candidate", { receiverId: broadcasterId, candidate: event.candidate });
        }
      };

      peerConnection.ontrack = (event) => {
        videoRef.current.srcObject = event.streams[0];
      };

      socket.on("offer", async ({ senderId, offer }) => {
        await peerConnection.setRemoteDescription(new RTCSessionDescription(offer));
        const answer = await peerConnection.createAnswer();
        await peerConnection.setLocalDescription(answer);
        socket.emit("answer", { receiverId: senderId, answer });
      });

      socket.on("ice-candidate", ({ senderId, candidate }) => {
        peerConnection.addIceCandidate(new RTCIceCandidate(candidate));
      });

      setPeerConnection(peerConnection);
    });

    return () => socket.disconnect();
  }, []);

  return (
    <div>
      <h1>Live Stream (Viewer)</h1>
      <video ref={videoRef} autoPlay playsInline />
    </div>
  );
};

export default Viewer;

📌 Step 5: Run the Application

Start the WebRTC Server
```
node server.js
```
Start the React App
```
npm start
```
Open two browser tabs:
- One for the Broadcaster (Live Streamer)
- One for the Viewer

🎯 How It Works

✅ Broadcaster captures video/audio using getUserMedia()
✅ Signaling via WebSockets to connect to viewers
✅ WebRTC Peer Connection established
✅ Viewers receive and display the live stream

🚀 What's Next?

Recording the Stream (Save video files on the server)
Adding Chat with RTCDataChannel
Optimizing for Low Latency

📌 WebRTC in Mobile Apps (React Native + Expo)

WebRTC isn't just for web browsers—it works on mobile apps too! In this section, we'll explore how to use WebRTC in React Native (Expo) to build a real-time video calling app.

📌 Step 1: Understanding WebRTC in React Native

Unlike web browsers, where WebRTC is built-in, React Native requires external libraries to support WebRTC.

📌 Key Libraries:

react-native-webrtc - Provides WebRTC functionality in React Native.
socket.io-client - Used for signaling (SDP exchange).

📌 WebRTC Workflow in React Native

Get User Media → Access camera & microphone using WebRTC API.
Establish Connection → Use WebSockets for signaling.
Send/Receive Video & Audio → Use RTCPeerConnection.
Handle ICE Candidates → Ensure NAT traversal with STUN/TURN.

📌 Step 2: Install Dependencies

Run the following commands in your React Native (Expo) project:

npx expo install react-native-webrtc socket.io-client

📌 Step 3: Set Up the WebRTC Signaling Server

📌 server.js (Node.js WebSockets for signaling)

const express = require("express");
const http = require("http");
const { Server } = require("socket.io");
const cors = require("cors");

const app = express();
app.use(cors());

const server = http.createServer(app);
const io = new Server(server, {
  cors: { origin: "*", methods: ["GET", "POST"] }
});

let users = {};

io.on("connection", (socket) => {
  console.log("User connected:", socket.id);
  
  socket.on("join", (room) => {
    users[socket.id] = room;
    socket.join(room);
    socket.to(room).emit("new-user", socket.id);
  });

  socket.on("offer", ({ receiverId, offer }) => {
    io.to(receiverId).emit("offer", { senderId: socket.id, offer });
  });

  socket.on("answer", ({ receiverId, answer }) => {
    io.to(receiverId).emit("answer", { senderId: socket.id, answer });
  });

  socket.on("ice-candidate", ({ receiverId, candidate }) => {
    io.to(receiverId).emit("ice-candidate", { senderId: socket.id, candidate });
  });

  socket.on("disconnect", () => {
    socket.broadcast.emit("user-disconnected", socket.id);
    delete users[socket.id];
  });
});

server.listen(5000, () => console.log("Signaling server running on port 5000"));

📌 Step 4: Implement WebRTC in React Native (Expo)

📌 VideoCall.js

import React, { useEffect, useRef, useState } from "react";
import { View, Text, TouchableOpacity } from "react-native";
import { RTCPeerConnection, RTCView, mediaDevices, RTCIceCandidate, RTCSessionDescription } from "react-native-webrtc";
import io from "socket.io-client";

const socket = io("http://localhost:5000");

const ICE_SERVERS = [{ urls: "stun:stun.l.google.com:19302" }];

const VideoCall = () => {
  const [localStream, setLocalStream] = useState(null);
  const [remoteStream, setRemoteStream] = useState(null);
  const peerConnection = useRef(null);

  useEffect(() => {
    startCall();
    
    socket.on("new-user", async (userId) => {
      const offer = await createOffer();
      socket.emit("offer", { receiverId: userId, offer });
    });

    socket.on("offer", async ({ senderId, offer }) => {
      await peerConnection.current.setRemoteDescription(new RTCSessionDescription(offer));
      const answer = await peerConnection.current.createAnswer();
      await peerConnection.current.setLocalDescription(answer);
      socket.emit("answer", { receiverId: senderId, answer });
    });

    socket.on("answer", async ({ senderId, answer }) => {
      await peerConnection.current.setRemoteDescription(new RTCSessionDescription(answer));
    });

    socket.on("ice-candidate", ({ senderId, candidate }) => {
      peerConnection.current.addIceCandidate(new RTCIceCandidate(candidate));
    });

    return () => {
      socket.disconnect();
      if (peerConnection.current) peerConnection.current.close();
    };
  }, []);

  const startCall = async () => {
    peerConnection.current = new RTCPeerConnection({ iceServers: ICE_SERVERS });

    const stream = await mediaDevices.getUserMedia({ video: true, audio: true });
    setLocalStream(stream);
    stream.getTracks().forEach((track) => peerConnection.current.addTrack(track, stream));

    peerConnection.current.ontrack = (event) => {
      setRemoteStream(event.streams[0]);
    };

    peerConnection.current.onicecandidate = (event) => {
      if (event.candidate) {
        socket.emit("ice-candidate", { candidate: event.candidate });
      }
    };

    socket.emit("join", "room1");
  };

  const createOffer = async () => {
    const offer = await peerConnection.current.createOffer();
    await peerConnection.current.setLocalDescription(offer);
    return offer;
  };

  return (
    <View style={{ flex: 1, justifyContent: "center", alignItems: "center" }}>
      <Text>React Native WebRTC Video Call</Text>
      {localStream && <RTCView streamURL={localStream.toURL()} style={{ width: 200, height: 200 }} />}
      {remoteStream && <RTCView streamURL={remoteStream.toURL()} style={{ width: 200, height: 200, marginTop: 20 }} />}
      <TouchableOpacity onPress={startCall} style={{ marginTop: 20, padding: 10, backgroundColor: "blue" }}>
        <Text style={{ color: "white" }}>Start Call</Text>
      </TouchableOpacity>
    </View>
  );
};

export default VideoCall;

📌 Step 5: Run the Application

Start the Signaling Server
```
node server.js
```
Start Expo Project
```
npx expo start
```
Test on Two Devices (or Two Emulator Windows)
- Open the app on two different phones or emulators.
- Start a video call and see the WebRTC connection in action!

🚀 Features Implemented

✅ Real-time Video Calling using WebRTC
✅ Signaling with WebSockets
✅ STUN Server for NAT Traversal
✅ Mobile-Friendly UI using Expo

🎯 Next Steps

🔹 Implement Call End Functionality
🔹 Add Mute/Unmute & Camera Toggle
🔹 Use a TURN Server for Better Connectivity
🔹 Enable Group Video Calls with SFU (Mediasoup)

🔥 Server-Side WebRTC with Node.js & FFmpeg (Headless WebRTC) 🚀

WebRTC isn't just for browsers—it can run on the server too! This is useful for:
✅ Recording Calls on the Server
✅ Live Streaming to YouTube/Twitch
✅ Processing Media (e.g., Adding Watermarks, Transcoding)

📌 How Does Server-Side WebRTC Work?

Since WebRTC needs a media source, we replace the browser's camera/microphone with:

FFmpeg → Simulates a media source (e.g., video file, RTSP stream).
Puppeteer (Headless Chrome) → Can act as a hidden WebRTC client (optional).
GStreamer/Mediasoup → Server-side SFU for media routing.

📌 Step 1: Install Dependencies

First, install WebRTC & media processing tools:

npm install express socket.io wrtc fluent-ffmpeg
sudo apt install ffmpeg  # For Linux/macOS

📌 Step 2: Setup the WebRTC Signaling Server

📌 server.js (Node.js WebRTC Signaling Server)

const express = require("express");
const http = require("http");
const { Server } = require("socket.io");
const wrtc = require("wrtc");

const app = express();
const server = http.createServer(app);
const io = new Server(server);

let clients = {};

io.on("connection", (socket) => {
  console.log("Client connected:", socket.id);

  socket.on("offer", async ({ senderId, offer }) => {
    console.log("Received Offer:", offer);
    const peerConnection = new wrtc.RTCPeerConnection();
    clients[senderId] = peerConnection;

    peerConnection.onicecandidate = (event) => {
      if (event.candidate) {
        socket.emit("ice-candidate", { candidate: event.candidate });
      }
    };

    peerConnection.ontrack = (event) => {
      console.log("Received Media Stream!");
      recordStream(event.streams[0]);
    };

    await peerConnection.setRemoteDescription(new wrtc.RTCSessionDescription(offer));
    const answer = await peerConnection.createAnswer();
    await peerConnection.setLocalDescription(answer);
    
    socket.emit("answer", { answer });
  });

  socket.on("ice-candidate", ({ candidate }) => {
    clients[socket.id]?.addIceCandidate(new wrtc.RTCIceCandidate(candidate));
  });

  socket.on("disconnect", () => {
    console.log("Client disconnected:", socket.id);
    delete clients[socket.id];
  });
});

server.listen(5000, () => console.log("WebRTC Server running on port 5000"));

📌 Step 3: Capture and Record WebRTC Streams using FFmpeg

📌 recordStream.js

const ffmpeg = require("fluent-ffmpeg");
const fs = require("fs");

function recordStream(mediaStream) {
  console.log("Recording WebRTC stream...");

  const videoFile = `./recordings/recording_${Date.now()}.mp4`;
  const ffmpegProcess = ffmpeg()
    .input(mediaStream)
    .inputFormat("rtsp")
    .output(videoFile)
    .videoCodec("libx264")
    .audioCodec("aac")
    .on("end", () => console.log(`Recording saved: ${videoFile}`))
    .on("error", (err) => console.error("FFmpeg Error:", err))
    .run();
}

module.exports = { recordStream };

📌 Step 4: Send Video from FFmpeg as a WebRTC Stream

If we want WebRTC to receive a video file instead of a real webcam, we can use FFmpeg:

ffmpeg -re -i sample.mp4 -f rtp rtp://localhost:5004

This simulates a live video feed that WebRTC can consume.

📌 Step 5: Live Stream WebRTC to YouTube/Twitch

📌 liveStream.js

const { spawn } = require("child_process");

function streamToYouTube(mediaStream, streamKey) {
  const ffmpegProcess = spawn("ffmpeg", [
    "-i", mediaStream,
    "-f", "flv",
    `rtmp://a.rtmp.youtube.com/live2/${streamKey}`
  ]);

  ffmpegProcess.stderr.on("data", (data) => console.log("FFmpeg:", data.toString()));
}

module.exports = { streamToYouTube };

Now you can broadcast WebRTC calls live on YouTube! 🎥

📌 Step 6: Run the WebRTC Server

node server.js

Then test it with a WebRTC client like React, React Native, or even OBS Studio.

🚀 Features Implemented

✅ Server-Side WebRTC Peer Connection
✅ WebRTC Stream Recording using FFmpeg
✅ Live Streaming WebRTC to YouTube/Twitch
✅ Handling WebRTC ICE Candidates in Node.js

🎯 Next Steps

🔹 Add TURN Server for NAT Traversal
🔹 Multi-Client Video Conferencing (SFU)
🔹 Transcode WebRTC Streams for Different Devices

🔥 Now you have a fully working WebRTC server with media recording!

🔥 Integrating WebRTC with AI/ML (Real-Time Video Processing) 🚀

WebRTC can be enhanced with AI/ML for:
✅ Face Detection & Recognition (OpenCV, TensorFlow.js)
✅ Background Blur & Virtual Backgrounds (BodyPix, MediaPipe)
✅ Speech-to-Text & Translation (Whisper, Deepgram)
✅ Gesture Recognition & Eye Tracking

📌 Step 1: Install Dependencies

npm install express socket.io wrtc @tensorflow/tfjs @tensorflow-models/body-pix
npm install fluent-ffmpeg opencv4nodejs

wrtc → WebRTC in Node.js
@tensorflow/tfjs & @tensorflow-models/body-pix → AI for background removal
opencv4nodejs → Face detection & object tracking

📌 Step 2: Setup WebRTC with AI Video Processing

📌 server.js (WebRTC Server with AI Processing)

const express = require("express");
const http = require("http");
const { Server } = require("socket.io");
const wrtc = require("wrtc");
const cv = require("opencv4nodejs");
const { applyAIProcessing } = require("./aiProcessing");

const app = express();
const server = http.createServer(app);
const io = new Server(server);

let clients = {};

io.on("connection", (socket) => {
  console.log("Client connected:", socket.id);

  socket.on("offer", async ({ senderId, offer }) => {
    console.log("Received Offer:", offer);
    const peerConnection = new wrtc.RTCPeerConnection();
    clients[senderId] = peerConnection;

    peerConnection.onicecandidate = (event) => {
      if (event.candidate) {
        socket.emit("ice-candidate", { candidate: event.candidate });
      }
    };

    peerConnection.ontrack = (event) => {
      console.log("Received Media Stream!");
      applyAIProcessing(event.streams[0]); // 🔥 Apply AI processing here
    };

    await peerConnection.setRemoteDescription(new wrtc.RTCSessionDescription(offer));
    const answer = await peerConnection.createAnswer();
    await peerConnection.setLocalDescription(answer);

    socket.emit("answer", { answer });
  });

  socket.on("ice-candidate", ({ candidate }) => {
    clients[socket.id]?.addIceCandidate(new wrtc.RTCIceCandidate(candidate));
  });

  socket.on("disconnect", () => {
    console.log("Client disconnected:", socket.id);
    delete clients[socket.id];
  });
});

server.listen(5000, () => console.log("WebRTC AI Server running on port 5000"));

📌 Step 3: Implement AI Processing on Video Streams

📌 aiProcessing.js (Apply AI on WebRTC Video)

const tf = require("@tensorflow/tfjs-node");
const bodyPix = require("@tensorflow-models/body-pix");
const cv = require("opencv4nodejs");

// Load AI models
async function loadModels() {
  console.log("Loading AI Models...");
  return await bodyPix.load();
}

async function applyAIProcessing(mediaStream) {
  const model = await loadModels();

  console.log("Applying AI Processing on WebRTC Stream...");

  mediaStream.on("data", async (frameBuffer) => {
    let frame = cv.imdecode(frameBuffer);
    
    // Convert to Tensor & Apply BodyPix (Background Blur)
    const tensor = tf.node.decodeImage(frameBuffer);
    const segmentation = await model.segmentPerson(tensor);
    
    // Apply Blur Effect on Background
    if (segmentation) {
      frame = frame.blur(new cv.Size(15, 15));
    }

    console.log("Processed Frame with AI!");
  });
}

module.exports = { applyAIProcessing };

📌 Step 4: Run the WebRTC AI Server

node server.js

🔥 Features Implemented

✅ Face & Object Detection with OpenCV
✅ Background Blur with TensorFlow.js
✅ Real-Time Video Processing on WebRTC Streams

🎯 Next Steps

🔹 Real-Time Hand Gesture Recognition (MediaPipe)
🔹 Live Captioning with AI Speech Recognition
🔹 Emotion Detection & Sentiment Analysis

🔥 Now you have WebRTC integrated with AI/ML!

Name		Name	Last commit message	Last commit date
Latest commit History 4 Commits
client		client
server		server
README.md		README.md
image.png		image.png
syllabus.md		syllabus.md

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webRTC Zero to Hero

Syllabus

1. What is WebRTC?

History and Evolution

Why WebRTC? Use Cases and Benefits

Key Features

Use Cases

Comparison with Alternatives (Sockets, WebSockets, RTP, SIP)

2. How WebRTC Works (Under the Hood)

Client-to-Client Communication

Role of Browsers and APIs

Understanding SDP (Session Description Protocol)

What is SDP?

Example SDP Message (Simplified)

ICE (Interactive Connectivity Establishment)

STUN, TURN, and NAT Traversal

STUN (Session Traversal Utilities for NAT)

TURN (Traversal Using Relays around NAT)

Final Thoughts

What’s Next?

EXTRA:

📌 WebRTC Jargon & How It Works (Easy to Understand) 🚀

🔑 Key WebRTC Jargon (Must-Know Terms!)

1️⃣ Signaling

2️⃣ RTCPeerConnection (Peer-to-Peer Connection)

3️⃣ ICE (Interactive Connectivity Establishment)

4️⃣ STUN & TURN Servers

5️⃣ SDP (Session Description Protocol)

6️⃣ Tracks & Streams

7️⃣ RTCDataChannel

📌 WebRTC Flow (How It Works Step-by-Step!)

3. Media Capture (getUserMedia() API)

How Browsers Capture Video/Audio

Constraints and Permissions

Basic Usage

Custom Constraints

Processing Streams (Filters, Effects)

4. Peer Connection (RTCPeerConnection)

Creating a Connection

Basic Flow

Basic Example (Caller)

Basic Example (Callee)

Handling ICE Candidates

Understanding SRTP (Secure Real-Time Transport Protocol)

5. Data Channels (RTCDataChannel)

Sending Text and Binary Data

Sender

Receiver

Reliability and Ordering

Use Cases (Gaming, Messaging)

Final Thoughts

6. Understanding Signaling Mechanisms

Why Signaling?

Implementing Signaling with WebSockets

Client-Side WebSocket Code

Alternative Signaling Approaches

7. Setting Up a Signaling Server (Node.js + WebSockets)

Step 1: Install Dependencies

Step 2: Create the WebSocket Server (server.js)

Step 3: Exchanging SDP Messages

Client-Side Code (Caller)

Client-Side Code (Callee)

8. Connecting Peers via STUN & TURN

Implementing a Public STUN Server

Setting Up a Private TURN Server

Step 1: Install Coturn

Step 2: Configure Coturn (/etc/turnserver.conf)

Step 3: Restart and Enable Coturn

Step 4: Use Private TURN Server in WebRTC

Summary of Phase 3

Next: Phase 4 - Advanced WebRTC Features 🚀

9. Media Processing & Manipulation

Applying Filters (Blur, Effects)

Example: Apply a Blur Effect Using Canvas

Screen Sharing (getDisplayMedia API)

3. Media Capture (`getUserMedia()` API)

4. Peer Connection (`RTCPeerConnection`)

5. Data Channels (`RTCDataChannel`)

Step 2: Create the WebSocket Server (`server.js`)

Step 2: Configure Coturn (`/etc/turnserver.conf`)

🔹 Create a New Component: `GroupCall.js`