The Basics of WebRTC: A Beginner's Overview

In developing applications requiring real-time media communication directly between browsers and devices, WebRTC (Web Real-Time Communication) is a crucial tool. Before 2011, achieving such functionality necessitated plugins or alternative methods, but now, WebRTC simplifies the process.

WebRTC enables audio and video streaming within web pages by facilitating direct peer-to-peer communication, eliminating the need for intermediaries. Although its internal workings involve C/C++, WebRTC offers a JavaScript API for seamless integration.

There are compelling reasons to opt for WebRTC in real-time communication scenarios:

1. Ease of integration: With a JavaScript API, WebRTC seamlessly integrates into applications.

2. Security: Before communication begins, all browsers must agree, ensuring safety.

3. Open-source support: Being a Google-maintained project, WebRTC benefits from a supportive community on platforms like GitHub and Stack Overflow.

4. Wide browser support: Modern browsers universally support WebRTC, ensuring broad accessibility for users.

5. Mobile compatibility: WebRTC extends its utility to mobile applications, with available SDKs for mobile and embedded environments.

6. Arbitrary data exchange: WebRTC facilitates not only media streaming but also arbitrary data exchange.

7. Low latency: WebRTC boasts sub-500 milliseconds of real-time latency, making it one of the fastest protocols available.

In essence, WebRTC streamlines real-time communication in web applications, offering simplicity, security, and speed without compromising on functionality.