WebRTC Becomes Official Standard

February 1, 2021

The World Wide Web Consortium (W3C) and the Internet Engineering Task Force (IETF) noted that Web Real-Time Communications (WebRTC), which powers myriad services, is now an official standard, bringing audio and video communications anywhere on the Web.

WebRTC, comprised of a JavaScript API for Web Real-Time Communications and a suite of communications protocols, allows any connected device, on any network, to be a potential communication end-point, on the Web. WebRTC already serves as a cornerstone of online communication and collaboration services.

"Today’s landmark achievement is timely. Faced with a global pandemic of the COVID-19 coronavirus, the world has gone more and more virtual. It makes the Web even more crucial to society in information sharing, real-time communications, and entertainment," said Dr Jeff Jaffe, W3C CEO. “It is gratifying to see our technologies playing a key role in enabling such critical digital infrastructure. Combining the universal reach of the Web with the richness of live audio & video conversations has reshaped how the world communicates."

"Voice and video over IP revolutionized the way that people communicate around the world,” said IETF Chair Alissa Cooper. “Integrating these technologies into the Web platform has dramatically expanded their reach. Thanks to close collaboration between the IETF and W3C to standardize these technologies. WebRTC has enabled billions of people to connect and engage with each other during the COVID-19 pandemic, regardless of device or geography."

Live audio-video communication systems in any website, any web app

The WebRTC framework provides the building blocks from which web and app developers can seamlessly add video chat to a range of applications, including tele-education and tele-health, entertainment and gaming, professional and workformce collaboration.

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.

WebRTC is massively deployed as a communications platform and powers video conferences and collaboration systems across all major browsers, both on desktop and mobile. Billions of users can interact now that WebRTC makes live video chat easier than ever on the Web. And from startups to Web-scale companies, in commercial products and open source projects, WebRTC has vastly expanded the ability to deploy real-time interaction solutions to customers and users.

Real-world positive and timely impact

The year 2020 has shown both how critical WebRTC already is in a world where travel and physical contacts need to be limited, as well as the many improvements that can be brought to the technology to address new usages that have emerged.

Businesses and households are relying on WebRTC for a wealth of operations, increasing its adoption. Organizations are leveraging WebRTC to conduct training, interviews, strategic planning or as a substitute for in-person meetings to keep connected through happy hour and other social interactions - it is replacing not only in-person meetings, but it is now also replacing the human interactions inside offices. Domains such as healthcare and defense use WebRTC for training. Schools and universities have shifted to virtual learning platforms. Cloud gaming and social networks use live streaming and interactive broadcasts. Entertainment is trying to figure out how to bring the audience back to the studios by doing it remotely. Sports are trying to recreate the in-stadium experience using WebRTC. Families and friends make daily use of products that are built with WebRTC or parts of it.

The future of WebRTC is already in the works

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.

There is already work underway in the IETF WebTransport (WEBTRANS) and WebRTC Ingest Signaling over HTTPS (WISH) working groups that will build on, coordinate with and extend efforts of other IETF working groups. These include QUIC, to defin new protocols that support the development of the WebTransport API, and HTTPBIS, to specify 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 has started work on WebRTC Next Version Use Cases to map out WebRTC's future, notably:

  • End-to-end encryption in server-mediated videoconferencing
  • Live processing of audio and video feeds, including via Machine Learning
  • Internet of Things (e.g., an IoT sensor maintains a long-term connection and seeks to minimize power consumption)

The WebRTC Working Group is iterating on existing and new use cases, with a focus on understanding the full range of the needs and their priority. W3C recently started work on WebTransport and Web Codecs, which promises to bring the benefits of low-latency streaming to the broader media and entertainment ecosystem.

WebRTC joins the many W3C standards that define an Open Web Platform for application development with unprecedented potential to enable developers to build rich interactive experiences, powered by vast data stores, that are available on any device and environment.

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