WebRTC is a free, open-source project that enables real-time communication capabilities, such as audio and video conferencing, peer-to-peer file sharing, and screen sharing, directly in web browsers and mobile applications. It provides APIs for establishing secure and low-latency communication channels between browsers and devices using standard web technologies. Readers can learn how WebRTC facilitates real-time communication and collaboration in web applications, enabling developers to build interactive and engaging experiences with audio and video capabilities without requiring plugins or third-party software.

WebRTC

Probing WebRTC Bandwidth Probing – why and how in gcc

VoRS (VoRS)

Cloudflare Calls: millions of cascading trees all the way down

pojntfx/weron: Overlay networks based on WebRTC.

WebRTC Signaling Explained: Making Video Calls Seamless

WebSockets vs Server-Sent-Events vs Long-Polling vs WebRTC vs WebTransport

WebRTC. How to establish a p2p connection between browsers

End-to-End Encryption in WebRTC… 4 Years Later

Live Streaming from Unity - Real-Time Playback (Part 7)

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