Technology: WebRTC to Enable Peer-to-Peer Browser Communications

WebRTC is a framework for real-time communication (RTC) between browsers that requires no special plug-in to work.

The intent of WebRTC is to be a standard capability built into all browsers, whether on mobile or desktop.  The WebRTC API is open source and free, and it’s goal is to be more efficient than any other existing real-time communication technology.  Conferencing and chat are two of the main target types of applications for WebRTC.  The WebRTC API provides basic components for enabling high-quality audio and video communications across the network between two computers.  It exposes network, video, and audio capabilities via JavaScript, allowing developers easy access to include those features within their own browser-based applications.

The WebRTC API is based on work originally done at Ericsson Labs and later donated to the World Wide Web Consortium (W3C) standards group.  While real time communication between peer computers isn’t new, RTC technology is used for example by applications like Skype and Google Talk, current uses require the download and install of a special app or plugin.

On the downside for WebRTC, at least in the short term, early mobile adopters of WebRTC will be hit by performance issues of the WebRTC video codec VP8.  This codec is relatively new and no chipsets currently support it, which means that mobile devices will be burdened with encoding and decoding communications in software.  If WebRTC catches on though, it’s not likely that there won’t be chipsets available.

Google, Ericsson, and Mozilla are all contributors to the proposed WebRTC standard.  Microsoft is holding back and proposed a competing standard to the WebRTC on August 8th, 2012 called CU-RTC-WEB — this is despite rumors that Microsoft will use WebRTC as the foundation of a browser-based Skype application.  The Microsoft proposal is a much more low-level API and relegates much of the work to Javascript rather than building the functionality directly within the browser.   While Apple has much at stake in the adoption of a WebRTC-like specification, they aren’t expected to support the WebRTC standard any time soon.

That’s not stopping Google and Mozilla from moving forward with it.

Users of Google Chrome now have WebRTC available to them.  Google WebRTC blog recently announced WebRTC availability in release 23 of Chrome, saying that “It’s the biggest milestone yet… web developers can now offer Chrome users the ability to have live, high quality audio and video communication as part of their web experience.”

And so now do Firefox users.  The FireFox blog announces that WebRTC is available for desktop browser users found now in the nightly build of FireFox.

WebRTC has been called disruptive and revolutionary.  But others just see it as one small evolutionary step in communications technology.

Russell Bennett, Principal of UC Insights, a consulting firm that specializes in Unified Communications, told Dave Michels of UCS that “There are only about four wideband codec’s available in the world right now. One’s GIPS, which was bought by Google, one is Microsoft’s Real Time Audio, there was another one, a third-party codec was bought by Skype which they now call Silk, and also there’s a Russian version by a company called Spirit DSP. All except Spirit are now owned by the big vendors. And Google is proposing their implementation as a standard or an ‘open standard’ which kind of sounds like Cisco and its tele-presence standard. It’s just proprietary technology being offered on a non-Royalty basis or a zero Royalty basis. This is an attempt to bank some kind of business leverage.”