WebRTC (Web Real-Time Communication) is an open-source technology that enables real-time audio, video calls, and data transfer on web browsers and mobile apps without the need for plugins. The WebRTC connection establishment process involves multiple steps, including connecting to signaling servers, exchanging Offer/Answer, ICE candidate exchange, and establishing data channels. The core components of WebRTC include Signaling Servers for session establishment, RTCPeerConnection for managing connections, Offer and Answer for session descriptions, and Channels for data transfer. ICE (Interactive Connectivity Establishment) plays a vital role in resolving network connectivity issues by facilitating NAT traversal, optimal path selection, and connection recovery. ICE candidates, STUN and TURN servers are essential parts of the ICE workflow. With ICE, WebRTC can function effectively in complex network environments, ensuring successful peer-to-peer communication. The example of a WebSocket-based signaling server for WebRTC-based peer-to-peer chat demonstrates the practical implementation of WebRTC technology.