The World Wide Web Consortium (W3C) and the Internet Engineering Task Force (IETF) have announced that Web Real-Time Communications (WebRTC) is now an official web standard.

WebRTC is made up of a JavaScript API for Web Real-Time Communications and a suite of communications protocols. It is designed to allow any connected device, on any network, to be a potential communication end-point, on the Web.

WebRTC is already heavily used in online communication and collaboration services, providing a way to add peer-to-peer audio and video chat to projects.

The W3C says that:

"with the foundations standardized and deployed as a royalty-free feature in Web browsers and other devices and platforms, setting up a secure audio-video communication system with WebRTC has become a built-in capability, eliminating the need to install plugins or download separate applications."

With the use of WebRTC expanding beyond the initial core design to power video conferences and collaboration systems in web browsers and other ecosystems (e.g., native apps), more features and more optimizations are now needed.

The developers of WebRTC are already working on extras for it, with work underway in the IETF WebTransport (WEBTRANS) and WebRTC Ingest Signaling over HTTPS (WISH) working groups. Other IETF working groups are also working on new protocols that support the development of the WebTransport API, and the specification of a simple, extensible, HTTPS-based signaling protocol to establish one-way WebRTC-based audiovisual sessions between broadcasting tools and real-time media broadcast networks.

The W3C WebRTC Working Group says that they envisage the next version of WebRTC will be used for tasks including end-to-end encryption in server-mediated videoconferencing; live processing of audio and video feeds, including via machine learning; and the Internet of Things (e.g., an IoT sensor maintains a long-term connection and seeks to minimize power consumption).

More Information

WebRTC

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