Google building Skype-alike software into Chrome
Shortly after releasing software for audio and video chat as an open-source project called WebRTC as open-source software, Google is beginning to build it into its Chrome browser.
The real-time chat software originated from Google’s 2010 acquisition of Global IP Solutions (GIPS), a company specializing in Internet telephony and videoconferencing.
The obvious beneficiary for the project is Gmail, whose audio and video communications ability today requires use of a proprietary plug-in. Gmail chat is getting more important as Google’s VoIP (voice over Internet Protocol) efforts mature and integrate with the Google Voice service.
But Google has higher hopes that WebRTC will be used well beyond Gmail. Rather, it hopes WebRTC will become an incarnation of a nascent Web standard for videoconferencing and peer-to-peer communications and for the necessary underlying network communication protocols. In an introductory blog post, Google said it released the technology as open-source, royalty-free software and pledged to work with other browser makers Mozilla and Opera on the real-time chat project.
If Google and allies succeed in establishing the technology and building support into multiple browsers, that would mean anybody building a Web site or Web application could draw upon the communications technology. In other words, anyone could build a rival to services, such as Skype, with just a Web application.
Google is a prominent advocate of the idea of Web-based applications and the cloud computing approach it enables. Web apps more easily span different computing systems–not just Windows and
Mac OS X, but also a profusion of smartphones. Google therefore is working hard to try to catch Web apps up with what native apps can do already. Ultimately, the company stands to profit by encouraging more people to lead an online existence where they’re more likely to perform Google searches and perhaps pay for Google services such as Google Apps.
The WebRTC software soon will begin arriving in Chrome.
WebRTC uses two audio codecs from GIPS, iSAC for high-bandwidth connections and iLBC for narrowband connections. (A codec is software used to encode and decode streams of data such as audio and video.)
For video, WebRTC uses Google’s VP8 codec, another open-source, royalty-free technology the company acquired and released in an effort to lower barriers for new technology on the Web.
Originally published by CNET.