The ubiquity of the internet has strongly influenced modern communication systems, with VoIP (Voice over Internet Protocol) haven hitherto exerted enormous pressure on traditional communication players (Telcos), and the competition is even now fiercest as the new technology, WebRTC is gaining traction on the web.
WebRTC (Web-based Real-time Communication) is an open-source project (of course, free) that enables browser-to-browser applications with real-time communication capabilities through simplified JavaScript APIs without plug-ins.
The prototype comprises codes that Google open sourced in May 2011, and followed by an ongoing work to standardized the relevant protocols in the IETF and browsers APIs in the W3C. The implementation revolves on works on HTML5 and related technologies supported by the Web Hypertext Application Technology Working Group (WHATWG). The project is supported by Google, Mozilla and Opera, increasingly spreading amongst other major web technology vendors.
Google is equally responsible for the key voice and video codecs WebRTC currently support - VP8 (video), iSAC (wide-band links) and iLBC (narrower connections).
What does all these mean? It won't be long after the point of aligning to standards for developers to start building applications into web-pages for voice and video calls. Albeit, such capabilities are possible already, but are specific to vendors implementing them. What WebRTC does is open the technology to all, not just to vendors that can afford to license a proprietary system.
Now, the revolutionary drive behind WebRTC is that ordinary Web developers using just JavaScript APIs can build fully functioning voice, video and data collaboration applications or have same embed within other applications or website with just a few lines of code.
The new technology, WebRTC will make voice and video call enabled web pages pervasive, with no plug-ins to download and install that may not be compatible with all the browsers you use, on the desktop or on your mobile devices. And that may ultimately be the future of voice and video calls for all.
The ubiquity of the internet has strongly influenced modern communication systems, with VoIP (Voice over Internet Protocol) haven hitherto exerted enormous pressure on traditional communication players (Telcos), and the competition is even now fiercest as the new technology, WebRTC is gaining traction on the web.
WebRTC (Web-based Real-time Communication) is an open-source project (of course, free) that enables browser-to-browser applications with real-time communication capabilities through simplified JavaScript APIs without plug-ins.
The prototype comprises codes that Google open sourced in May 2011, and followed by an ongoing work to standardized the relevant protocols in the IETF and browsers APIs in the W3C. The implementation revolves on works on HTML5 and related technologies supported by the Web Hypertext Application Technology Working Group (WHATWG). The project is supported by Google, Mozilla and Opera, increasingly spreading amongst other major web technology vendors.
Google is equally responsible for the key voice and video codecs WebRTC currently support - VP8 (video), iSAC (wide-band links) and iLBC (narrower connections).
What does all these mean? It won't be long after the point of aligning to standards for developers to start building applications into web-pages for voice and video calls. Albeit, such capabilities are possible already, but are specific to vendors implementing them. What WebRTC does is open the technology to all, not just to vendors that can afford to license a proprietary system.
Now, the revolutionary drive behind WebRTC is that ordinary Web developers using just JavaScript APIs can build fully functioning voice, video and data collaboration applications or have same embed within other applications or website with just a few lines of code.
The new technology, WebRTC will make voice and video call enabled web pages pervasive, with no plug-ins to download and install that may not be compatible with all the browsers you use, on the desktop or on your mobile devices. And that may ultimately be the future of voice and video calls for all.
WebRTC (Web-based Real-time Communication) is an open-source project (of course, free) that enables browser-to-browser applications with real-time communication capabilities through simplified JavaScript APIs without plug-ins.
The prototype comprises codes that Google open sourced in May 2011, and followed by an ongoing work to standardized the relevant protocols in the IETF and browsers APIs in the W3C. The implementation revolves on works on HTML5 and related technologies supported by the Web Hypertext Application Technology Working Group (WHATWG). The project is supported by Google, Mozilla and Opera, increasingly spreading amongst other major web technology vendors.
Google is equally responsible for the key voice and video codecs WebRTC currently support - VP8 (video), iSAC (wide-band links) and iLBC (narrower connections).
What does all these mean? It won't be long after the point of aligning to standards for developers to start building applications into web-pages for voice and video calls. Albeit, such capabilities are possible already, but are specific to vendors implementing them. What WebRTC does is open the technology to all, not just to vendors that can afford to license a proprietary system.
Now, the revolutionary drive behind WebRTC is that ordinary Web developers using just JavaScript APIs can build fully functioning voice, video and data collaboration applications or have same embed within other applications or website with just a few lines of code.
The new technology, WebRTC will make voice and video call enabled web pages pervasive, with no plug-ins to download and install that may not be compatible with all the browsers you use, on the desktop or on your mobile devices. And that may ultimately be the future of voice and video calls for all.