What is WebRTC? Definition, Use cases, How It Works

WebRTC, or Web Real-Time Communication, is a secure and open-source protocol designed for bi-directional, real-time communication in browsers and mobile applications, enabling ultra-low latency audio, video, and data exchanges. It supports both peer-to-peer and server-assisted topologies, making it ideal for real-time streaming, communications, and interactive applications. Key features include ultra-low latency, secure data transfer through encryption, scalability with the right platform, native browser support, and multi-track media and codec support. WebRTC can be used across various applications, such as remote production workflows, live sports streaming, surveillance, online auctions, and more, due to its sub-250 ms latency that feels instantaneous to users. Compared to other protocols like RTMP, HLS, and SRT, WebRTC remains a foundation for interactive streaming experiences due to its mature real-time stack and native browser adoption, despite the emergence of new protocols like WHIP, WHEP, MOQ, and HESP.

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