WebRTC (Web Real-Time Communication) is an open-source project and an HTML5 specification that enables browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple JavaScript APIs.

It allows for direct peer-to-peer (P2P) communication for streaming audio, video, and arbitrary data between browsers without the need for plugins or third-party software.

1. Why WebRTC?

Before WebRTC, achieving real-time audio/video communication in a browser typically required:

• Proprietary Plugins: Think Adobe Flash Player or custom browser extensions. These often had security concerns, were not universally available, and created a fragmented user experience.
• Heavy Server-Side Relaying: Media would often be routed through a central server, increasing latency and server costs.

WebRTC brings many benefits to modern applications:

• Real-Time Communication: Enable voice and video calls, live streaming, and interactive applications directly in the browser.
• Peer-to-Peer Connection: Direct communication between clients reduces the reliance on centralized servers, lowering latency and cost.
• Cross-Platform Compatibility: Supported by modern browsers (Chrome, Firefox, Edge, Safari) and mobile platforms, making it highly accessible.
• Open Standards: Being an open standard means continuous improvements and broad industry support.

2. Key Components of WebRTC