WebRTC Audio Call Application

This project is a WebRTC-based audio call application that enables peer-to-peer audio communication between devices. It leverages WebRTC for real-time communication and Socket.IO for signaling.

Features

• Establishes peer-to-peer audio calls using WebRTC.
• Supports multiple STUN and TURN servers for enhanced connectivity.
• Provides a simple UI for connecting, muting/unmuting audio, and ending calls.
• Automatically handles ICE candidates for seamless connection setup.
• Displays connection and signaling statuses for better feedback.
• Robust error handling and reconnection logic.

Technologies Used

• WebRTC: Real-time communication.
• Socket.IO: Signaling for WebRTC connections.
• HTML, CSS, JavaScript: Front-end development.

How It Works

1. Users connect to a signaling server (via Socket.IO).
2. The server assigns roles (caller or callee) and pairs peers.
3. SDP exchange and ICE candidate handling establish a WebRTC connection.
4. Users can start calls, mute/unmute audio, and hang up.

Getting Started

Prerequisites

• Node.js installed on your system.

Installation

1. Clone the repository:

   git clone https://github.com/<your-username>/<your-repo>.git
   cd <your-repo>