Feat/video calling web rtc

[muntaxir4]

🚀 Initial PR: WebRTC Video Calling with WebSocket Signaling

📝 Description

This PR lays the groundwork for real-time video calling using WebRTC, integrated with a WebSocket-based signaling server for peer-to-peer connection negotiation. The signaling flow handles offer/answer exchange and ICE candidate distribution between clients. This marks the initial step in adding robust, scalable, and low-latency video communication support to the platform.

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🔧 Changes Made

Backend (FastAPI):

• 🔌 WebSocket endpoint: route for offer, answer, and ICE candidate exchange.
• 🧠 Connection manager: Maintains in-memory peer sessions and mapping via user IDs or room IDs.
• 📡 Broadcast logic: Routes signals from caller to callee and vice versa via WebSocket.

Frontend:

• 🎥 Basic WebRTC logic using RTCPeerConnection.
• 🖥️ Media capture setup using navigator.mediaDevices.getUserMedia.
• 🧭 Socket signaling client:
  • Connects to backend WebSocket.
  • Sends offers/answers and ICE candidates.
  • Handles remote stream and peer connection.

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📌 Next Steps (Future PRs)

• Add STUN/TURN server support for better NAT traversal.
• Improve room management and signaling error handling.
• UI/UX enhancements for call controls (mute, end call, etc.).
• Add user presence indication and busy signal logic.