Web Real-time Communications (WebRTC) recently became a recommendation of the World Wide Web Consortium (W3C) and a standard of the Internet Engineering Task Force (IETF). This is an important milestone in WebRTC’s long journey, which began in 2011 when Google open-source key communications technologies and Ericsson implemented the ConnectionPeer API. The new standard will continue to evolve as the WebRTC Working group works to integrate new use cases (real-time processing of audio and video feeds, Internet of Things use cases, etc.).

Ericsson commented on the adoption of the new standard:

“Ericsson, which has a long history of building person-to-person communication systems, was involved in the early development of WebRTC with a view to shaping the technology for use in different mobile and fixed environments. The news that WebRTC is now an official global standard means that everyone can use it to build and deliver different communication solutions from a stable base. Now, we are taking the next step and using WebRTC in 5G networks as well.”

WebRTC emerged at a time when real-time communications (RTC) was complex and expensive, and its audio and video technologies either had to be licensed or developed in-house. Websites that use RTC (such as Skype, Facebook, Google Hangouts) usually require downloading, installing, updating plug-ins or native applications – and sometimes troubleshooting and user support. WebRTC seeks to implement open standards for real-time, plug-in free video, audio and data communications.

After Google acquired GIPS, a provider of real-time voice and video processing software for IP networks, it opened source key RTC technologies (such as echo cancellation) in May 2011. WebRTC adoption was not immediate due to discussion of the content and scope of the specification and insufficient support from major browser and communication providers. While Chrome, Firefox, and Opera all supported WebRTC early on, Microsoft rolled out support for a competing set of real-time communication apis in 2015. Apple officially added support for WebRTC with Safari 11 in 2017.

Today, WebRTC is supported by 95% of all major browsers used by Web users. The W3C highlighted the expanded adoption of the new standard: “2020 has shown how important WebRTC has become in today’s world of travel and physical contact restrictions […] .”

More and more organizations are using WebRTC for training, interviewing, strategic planning, or in place of face-to-face meetings and other social interactions — not only in place of face-to-face meetings, but now in place of in-office interactions as well. Use WebRTC for training in areas such as healthcare and security. Schools and universities are already turning to virtual learning platforms. Cloud gaming and social networks use real-time streaming and interactive live streaming. Entertainment companies are trying to figure out how to engage audiences remotely. Sports are trying to recreate the in-stadium experience with WebRTC. Family and friends use products based on WebRTC technology every day.

The W3C also mentions addressing emerging use cases through future improvements and additions to the standard: audio, real-time processing of funny Hat video feeds, file sharing, the Internet of Things, machine learning, virtual reality gaming, untrusted JavaScript cloud conferencing, and more. Follow the example of a machine learning algorithm (RAISR) that produces a high-quality version of a low-resolution image:

Top: Original image, bottom: RAIRR super resolution 2x.

WebRTC is an open framework for the Web that enables real-time communication in a browser. It includes the basic building blocks for high-quality communication on the Web, such as networking, audio, and video components used in voice and video chat applications. These components, when implemented in a browser, can be accessed through JavaScript apis, making it easy for developers to implement their own RTC Web applications. WebRTC’s work is being standardized at the W3C API level and the IETF protocol level.

Read more: EasyRTC Video conferencing cloud service

EasyRTC is a global real-time audio development platform developed by TSINGSEE Qingxi Video team based on the open source framework WebRTC technology, supporting one-to-one and one-to-many video calls.

EasyRTC has MCU and SFU architecture, no need to install client and plug-in, pure H5 online video conference system, support wechat small program, H5 page, APP, PC client and other access methods, greatly meet the needs of voice and video social, online education and training, video conference and telemedicine and other scenarios.

With the rapid development of mobile Internet, AI, 5G and other emerging technologies, combined with WebRTC technology, more application scenarios will be derived in the future, changing the way of life such as clothing, food, housing and transportation of human beings.