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Alan Quayle
Alan Quayle has 22 years of experience in the telecommunication industry, focused on developing profitable new businesses in service providers,...
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Alan Quayle | June 13, 2012 |

 
   

The Rise of WebRTC: The Rules of Communications are About to be Rewritten

The Rise of WebRTC: The Rules of Communications are About to be Rewritten Ubiquitous WebRTC deployment would make real-time communications pervasive on the Internet.

Ubiquitous WebRTC deployment would make real-time communications pervasive on the Internet.

I recently made my first WebRTC (Web Real Time Communications) free voice call using the global calling service frisB from the latest Chrome browser, and then went on to explore a number of services available using WebRTC. I soon realized we are about to witness an explosion of innovations enabled by WebRTC. Before we review some of the innovations, a brief review of WebRTC is provided.

WebRTC is a HTML5 standard being drafted by the World Wide Web Consortium (W3C) and the Internet Engineering Task Force (IETF). The W3C is working on standardizing the interaction with HTML while the IETF is working on standardizing the underlying protocols. There is also a WebRTC project from Google that in June 2011 open sourced under a royalty free BSD (Berkeley Software Distribution)-style license a framework that has accelerated browser support of WebRTC. For the sake of simplicity I will refer to all these complementary endeavors as WebRTC, which embeds real-time communications into any browser.

WebRTC includes iLBC (Internet Low Bitrate Codec), iSAC (Internet Speech and Audio Coder), G.711, and G.722 codecs for audio and VP8 for video. These codecs include capabilities such as packet loss concealment and echo cancellation so they can robustly cope with a lack of guaranteed quality of service. WebRTC enables applications such as voice calls, video chat, file sharing, messaging, white-boarding, gaming, human/computer interaction, etc. without any client or plug-in download to run from a browser using simple HTML and JavaScript APIs. Ubiquitous WebRTC deployment would make real-time communications pervasive on the Internet.

The main goal of WebRTC is not interoperability with legacy systems--that's up to the legacy systems to implement. It's to open communications to new use cases and to web developers. Imposing the complexity of SIP to web developers would have made it very hard to attract their interest. With the WebRTC spec, a great 1:1 video chat experience can be built with under 100 lines of JavaScript code (see apprtc.appspot.com).

On a couple of practical issues, WebRTC includes and abstracts key NAT (Network Address Translator) and firewall traversal technology such as STUN (Simple Traversal of User datagram protocol through Network address translators), ICE (Interactive Connectivity Establishment), TURN (Traversal Using Relay NAT), RTP-over-TCP (Real-time Transport Protocol over Transmission Control Protocol) and support for proxies, enabling sessions to work like Skype. It also abstracts signaling by offering a signaling state machine that maps directly to PeerConnection. The PeerConnection API is what browser vendors will implement and expose WebRTC to web application developers. Web developers can choose the communication signaling protocol depending on their usage scenario (for example, but not limited to: SIP, XMPP/Jingle, etc.). Essentially any browser becomes a SIP endpoint, a telephone, an "open" Skype-like client, an endpoint for any real-time communication and control.

On the current status of browser implementations:

* Google Chrome: integrated WebRTC into its developer channel in January 2012, allowing any website to take advantage of WebRTC. The Google Talk plugin is a complex piece of software. So Google is taking it one step at a time and not making promises about the migration of when Google Talk plugin will be migrated to the WebRTC framework.

* Mozilla Firefox: Mozilla integrated WebRTC into its Firefox alpha in early 2012, which gave the browser the ability to perform audio mixing on a media stream. In April 2012, Mozilla released a demo of WebRTC video calling that ran inside the Firefox browser.

* Internet Explorer: Microsoft has started work on implementation of WebRTC.

* By the end of this year we'll see Chrome and Firefox running WebRTC--that's over 50% of the desktop market. The other half of the market (mainly IE) will take a little longer, likely to the end of 2013. In the latter part of 2013, we may see WebRTC on the browsers of some tablets and possibly smartphones. But as discussed later in this article, most tablet and smartphone users appear comfortable in downloading applications (clients) to their phone.



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