WebRTC Handshake – Offer/Answer Flow

This project demonstrates the core WebRTC connection flow between two browsers using the Offer–Answer model and a signaling server.

The goal of this project is to clearly explain and implement how two peers establish a WebRTC connection by exchanging session descriptions.

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🚀 What This Project Covers

• Creating an RTCPeerConnection
• Generating an Offer and Answer
• Setting Local Description and Remote Description
• Exchanging SDP data using a signaling server
• Understanding the complete WebRTC handshake lifecycle

This project focuses on connection logic rather than media streaming to keep the fundamentals clear.

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🔁 WebRTC Connection Flow

1. Browser A creates an RTCPeerConnection
2. Browser A creates an Offer
3. Browser A sets the Local Description
4. The Offer is sent to Browser B via a signaling server
5. Browser B receives the Offer
6. Browser B sets the Remote Description
7. Browser B creates an Answer
8. Browser B sets the Local Description
9. The Answer is sent back to Browser A via the signaling server
10. Browser A receives the Answer and sets the Remote Description

Once this process is complete, the peer-to-peer connection is established.

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🧠 Why This Project?

• Learn WebRTC fundamentals
• Understand the difference between signaling and peer connections
• Prepare for frontend and full-stack interviews
• Serve as a foundation for building:
  • Video calling applications
  • Audio chat apps
  • Screen sharing tools
  • Real-time collaboration platforms

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🛠️ Tech Stack

• JavaScript
• WebRTC APIs
• Signaling Server (WebSocket / Mocked signaling)
• HTML & CSS

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📌 Future Enhancements

• ICE candidate exchange
• Audio and video streaming
• Multi-peer support
• UI visualization for each signaling step