WebRTC offers developers unprecedented capabilities for streaming voice and video data through the browser. The three WebRTC APIs - getUserMedia, RTCDataChannel, RTCPeerConnection - work together to capture, relay, and render the input that comes from a computer’s webcam and microphone from one browser to another. From its inception, WebRTC has been backed by some of the web’s leading development teams. But Google, more than any other company, has paved the way for WebRTC’s success. From its inception in the Chrome team’s lab to its current status as a game-changing technology, WebRTC has relied on Google’s backing to sustain its growth. Google and WebRTC, for better or worse, are inseparable entities at this point. This of course has ramifications for WebRTC’s codification at the W3C.
Google and WebRTC - A History
The history of Google and WebRTC is actually the story of several different companies becoming interworking subsidiaries of a larger project. Google used its finances, stature, and engineering prowess to produce WebRTC as the first true in-browser solution to real-time communications in 2011. The sudden renaissance of browser-based RTC in 2011 was spurred by a series of mergers. Google acquired On2, the creators of the VP8 video codec, and Global IP solutions, a company that was already licensing low level components for RTC. With these new resources, the WebRTC Chrome project was born. Google made the project open source, and its developers ultimately settled on G.711, OPUS, and VP8 as the required codecs for the WebRTC API. The first implementation of WebRTC was built by Ericsson in 2011. In November 2012, Chrome 23 became the first large scale browser to offer embedded WebRTC functionality right out of the box.
Google and WebRTC - Overarching Design
The decision to treat traditional data separately from audiovisual streams comes from Google’s desire to give RTCDataChannel bidirectional support like WebSockets. This allows data to arrive asynchronously, which leads to faster transfer rates and greater delivery efficiency. The unchosen signaling mechanism of RTCPeerConnection gives developers leeway in crafting the underlying architecture of the way an app can communicate across the Internet. OnSIP has a mature SIP platform that offers a reliable solution to developers who want to scale their WebRTC application, broker connections between endpoints, and traverse NAT and firewall settings that can disrupt real-time data sharing.
Google and WebRTC at the W3C
The World Wide Web Consortium (W3C) is the international standards foundation for the world wide web. The 385 member panel was founded in 1994 by Tim Berners-Lee, and the group is primarily responsible for codifying major advances in web technology. WebRTC has experienced significant adoption, and it is currently completely interoperable between Chrome and Firefox, but it is still awaiting official codification by the W3C. Official standardization will likely pave the way for other companies, such as Microsoft and Apple, to utilize the technology.