Web Real-Time Communications (WebRTC) is now an official World Wide Web Consortium (W3C) and Internet Engineering Task Force (IETF) standard. WebRTC is a platform that provides real-time communication and collaboration services such as audio and video calling to browsers, mobile apps, and desktop apps. According to the organizations, this is especially important as the COVID-19 pandemic continues and businesses remain remote.

“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 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.”

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WebRTC provides a JavaScript API for web real-time communications and a suite of communication protocols. It eliminates the need to install plugins or download additional applications for audio/video communication. Google first brought up the idea to create WebRTC in 2009 as an alternative to Adobe Flash and apps that didn’t work in the browser. “The previous generation of browser-based products were built on top of licensed proprietary technology. Various products were built with this technology, including Hangouts. Google then acquired the companies it had been licensing the technology from and made it available as the open source WebRTC project,” Huib Kleinhout, product manager at Google, wrote in a post. From there, the standardization of WebRTC kicked off with other industry leaders including Mozilla, Microsoft, Cisco and Ericsson collaborating on it.

The organizations see the WebRTC framework being used to provide tele-education, tele-health, entertainment, gaming, professional and workforce collaboration solutions.

Beyond video conferences and collaboration systems in the web browser, the W3C WebRTC working group is working on providing end-to-end encryption for server-mediated videoconferencing; live video processing of audio and video feeds; and Internet of Things use cases. The W3C is also working on WebTransport and Web Codec to bring low-latency streaming to the broader media and entertainment ecosystem.

The IETF on WebTransport and Ingest Signaling over HTTPS working groups are currently working on new protocols to support the WebTransport API, and HTTPBIS for WebRTC-based audiovisual sessions between broadcasting tools and networks.

“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.”