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WebRTC: Marking a milestone in real-time communications

27 Jan 2021

The publication of the standards that provide a foundation for Web Real-Time Communications (WebRTC) marks a milestone in the development of conferencing services used by billions of people around the world.

More than a decade ago, when rich web applications were in their infancy, engineers from across the web and real-time communications industries came together to tackle a challenging problem: could modern voice and video over IP technology be brought to the ubiquitous platform of the Web?

The task was daunting. Real-time communications involved complicated protocol mechanics and network address translation (NAT) traversal machinery, while the Web lacked the APIs and security model needed to safely effectuate two-way real-time communications. But the idea of being able to make a video call in your browser at the click of a button presented nearly limitless possibilities for collaboration, connection, and productivity. 

That idea has become a reality for billions of users around the world thanks to years of intensive work to standardize WebRTC in the IETF and the World Wide Web Consortium (W3C). Last week, the IETF published a set of 50 specifications (comprising the bulk of RFCs published in January) that define the core WebRTC protocol stack together with several other protocols that use WebRTC building blocks. Earlier this week, the W3C published WebRTC 1.0, the web APIs that makes browser-to-browser calls possible. Even prior to the finalization of these specifications—years prior, in fact—WebRTC technologies were being deployed and used as part of most modern services that use voice or video, including many that do not involve web browsers. The availability of WebRTC code, APIs, and standards has made it simple to add real-time communications functionality to any application. And that widespread availability has been a true lifeline during the COVID-19 pandemic.  

There is already work underway to extend WebRTC. The IETF WebTransport (WEBTRANS) work is aiming to build out additional web support for a variety of transport properties. The WebRTC Ingest Signaling over HTTPS (WISH) work is focusing on the development of a protocol to support one-way WebRTC-based audiovisual sessions between broadcasting tools and real-time media broadcast networks. Similar work to expand the use cases of WebRTC is ongoing in the W3C.

Finishing the core WebRTC standards required tremendous effort from dozens of IETF and W3C participants over many years. The end result is a hugely popular technology suite that fulfills the Internet’s central promise—connecting people—on a global scale every day. It will be exciting to see what the future holds as the IETF community continues to build on this success.

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