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WebRTC

WebRTC (Web Real-Time Communication) is an open-source project and set of APIs that enables real-time audio, video, and data communication directly between web browsers and mobile applications without requiring plugins or third-party software. It provides peer-to-peer connectivity with built-in encryption, NAT traversal mechanisms, and media processing capabilities. This technology forms the foundation for implementing full video calls, video conferencing, and other real-time communication features on the web.

Also known as: Web Real-Time Communication, WebRTC API, Real-Time Communication, Peer-to-Peer Video, Browser Video Calls
🧊Why learn WebRTC?

Developers should learn WebRTC when building applications that require real-time communication features like video calls, voice chats, screen sharing, or file transfers, especially in scenarios where low latency and direct peer-to-peer connections are critical, such as telehealth platforms, online education tools, or collaborative work applications. It's essential for creating browser-based communication systems that work across different devices and networks without relying on external plugins, making it ideal for modern web and mobile apps that prioritize user privacy and performance.

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