The Rise Of WebRTC Technology

A WebRTC fact:

In the summer of 2010, at IETF 78, Google’s nascent WebRTC team with engineers from Microsoft, Apple, Mozilla, Skype, Ericsson, etc, decided to gauge the interest in building an RTC platform for the web. A one-day workshop was held to understand how such a standard should be written and defined. This was followed by intense activity in the W3C (World Wide Web Consortium) and the IETF which culminated in the formation of two working groups in May 2011: the IETF’s RTCWeb5 and the W3C’s WebRTC.


An important development in Chrome 89 (Released on March 9, 2021)

Google along with other members founded AOMedia (Alliance for Open Media). Google has been actively defining the AV1 video bitstream for the RTC use case. With AV1 becoming a standard, the video codec is being integrated into WebRTC. 89th Chrome version ships an AV1 software encoder providing AV1-to-web applications for RTC. AV1 delivers 30 to 50 percent extra bit rate savings at the same quality compared with VP9. It also offers another level of bandwidth efficiency and quality for video-calling services. AV1 is critical in facilitating RTC services to scale with higher-quality video experiences in the future.


WebRTC is not about voice and video communication anymore. Gaming, low-latency video streaming, AR/VR (augmented reality/virtual reality), mixed-reality services, etc. are up-and-coming use cases where WebRTC technology is getting used to its complete potential. 

Such use cases push the latency barrier. The need for further transport protocol optimizations rises. Standardization covering this need is WebTransport that focuses on optimizing for super-low-latency client-server media streaming via the QUIC protocol.

Owing to so many new and varied use cases of the technology, WebRTC standardization is evolving into something called WebRTC NV (Next Version). NV is not a completely new API but it allows access to the low-level media pipeline inside PeerConnections. Media is accessible using the Streams and WebCodecs APIs. Insertable Streams APIs are there to provide the foundation for full E2EE (end-to-end encryption) multiparty conferencing in browsers.

WebRTC for mobile devices

It started through the native integration into mobile social media, messaging, and video calling apps. With 5G network technology around the corner, video calling will become even more of a commodity. Moreso, WebRTC’s open architecture allows interesting innovations using machine learning, artificial intelligence, etc. to enhance the call quality and hide the effects of network disruptions.

Started as a way to bring audio and video to the web, WebRTC technology has now expanded into more use cases than initially imagined— video calling, AR/VR experiences, cloud-based gaming, scalable live streaming services, point-to-point video chat, multiuser conversations.

WebRTC has grown from enabling useful experiences to being essential in allowing enterprises, brands, and other users to continue work and education, and keep vital human contact during a pandemic. 


WebRTC has come a long way starting from that one-day workshop in 2010. Almost all modern services using voice or video communication are based on the WebRTC. 

2020 was a year where WebRTC proved to be a blessing. With a global pandemic going on, people across the globe opted for Real-time Communication solutions to work, educate, and connect with loved ones. WebRTC came to be one of the most important technologies for voice and video communication. 

WebRTC allowed for an ecosystem of interoperable communications apps to shine. At, we have created many custom WebRTC applications and solutions that serve numerous business needs.

The WebRTC success story would not have been possible without the open-source community. The ecosystem comprises code contributors, testers, bug filers, and corporate partners and we are happy to work with this revolutionary technology. If you need a WebRTC application for your business, get in touch with our experts here