• Created by Aman Yadav

Roadmap

1. The Fundamentals

• What is WebRTC? A clear definition and why it's a game-changer for communication.
• WebRTC vs. WebSockets: Understanding the key difference: Peer-to-Peer (P2P) vs. Client-Server.
• The Three Core APIs: A high-level look at the main building blocks of any WebRTC application:
  1. getUserMedia(): Accessing cameras and microphones.
  2. RTCPeerConnection: Establishing the direct connection.
  3. RTCDataChannel: Sending arbitrary data (like text chat).

2. The Connection Lifecycle (The Tricky Part)

• Signaling: The "out-of-band" process of coordinating the connection. We'll see why we still need a server (like our WebSocket server!) to get things started.
• SDP (Session Description Protocol): The "what." How peers describe their media capabilities (e.g., "I can handle H.264 video and Opus audio").
• ICE (Interactive Connectivity Establishment): The "how." The process of finding the best possible network path between two peers.
• STUN & TURN Servers: Understanding the helper servers that are essential for navigating network complexities like NATs and firewalls.

3. Client-Side Implementation (JavaScript)

• Getting Local Media: Using getUserMedia to capture video and audio from your devices.
• Creating a Peer Connection: Setting up the RTCPeerConnection object.
• The Offer/Answer Flow: The step-by-step dance of creating an "offer" (from the caller) and an "answer" (from the callee) using SDP.
• Exchanging ICE Candidates: How peers share potential network paths to find one that works.

4. Building a Simple Application

• A Two-Person Video Chat: We'll walk through the complete code flow, combining signaling (with WebSockets) and WebRTC to build a basic video call application.
• Data Channels in Action: Adding a simple text chat alongside the video call using RTCDataChannel.