Web Real-Time Communications

Web Real-Time Communications

WebRTC, otherwise known as Web Real-Time Communications is an open-source project – promoted by Google, Mozilla and others – enables plugin-free Real Time Communications via Javascript API’s. It facilitates browser to browser applications for voice calling, video chat and file sharing. The supported codec for WebRTC is currently VP8. WebRTC uses a server called Web Conferencing Server that in conjunction with a STUN Server  it is required to provide the initial page and synchronise the connections between two WebRTC endpoints.

Why is WebRTC important?

The WebRTC project is incredibly important as it marks the first time that a powerful real-time communications (RTC) standard has been open sourced for public consumption. It opens the door for a new wave of RTC web applications that will change the way we communicate today.

Significantly better video quality	WebRTC video quality is noticeably better than Flash.
Up to 6x faster connection times	Using JavaScript WebSockets, also an HTML5 standard, improves session connection times and accelerates delivery of other Gothuo events.
Reduced audio/video latency	WebRTC offers significant improvements in latency through WebRTC, enabling more natural and effortless conversations.
Freedom from Flash	With WebRTC and JavaScript WebSockets, you no longer need to rely on Flash for browser-based RTC.
Native HTML5 elements	Customize the look and feel and work with video like you would any other element on a web page with the new video tag in HTML5

Here's the video from the WebRTC session at Google I/O 2013, which gives a good look at the current state of the WebRTC universe.

WebRTC  Will Explore Changes in Enterprise Communications

• Enterprise UC
• Contact Centers/Customer Service
• Adding Video to Web Apps
• Vertical Applications
• Healthcare
• Financial Services
• Retail