WebRTC WebRTC (Web Real-Time Communication) is an open source technology that enables real-time video and audio streaming via a web browser. Encompassing HTML5 application programming interfaces (APIs), streaming protocols and standards, WebRTC eliminates the need for extra plugins or software to enable bi-directional streaming between browsers with latency low enough to resemble in-person communication and is commonly used for video conferencing. WebRTC latency is under 500ms end-to-end and ensures reliability over bad network conditions with an adaptive network encoding technique called simulcast. Through simulcasting, a cascade of streams with different bitrates and quality are created, ensuring that video contribution is not hindered. Unlike adaptive bitrate streaming, the [...]
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