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Brent Kelly
Brent Kelly is president and principal analyst at KelCor, Inc., where he provides strategy and counsel to key client types...
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Brent Kelly | May 27, 2013 |

 
   

Preparing for Disruption with WebRTC

Preparing for Disruption with WebRTC It is important to understand what WebRTC can do for you, but it is equally important to understand what WebRTC may do to you.

It is important to understand what WebRTC can do for you, but it is equally important to understand what WebRTC may do to you.

Introduction
WebRTC is an emerging standard that enables real-time voice, video and data sharing in a Web browser without the need for browser plugins. Potentially billions of devices supporting a browser--PCs, laptops, smartphones, tablets and a host of new devices--from a variety of manufacturers will be real-time communications-enabled. Whereas browsers have typically interacted only with one or more Web servers, WebRTC allows browsers to exchange media and data with one another directly and in a secure manner.

Although third-party programs like Skype have been around for a long time, and some browser-based plugins have been available for limited communications interactions, the implications WebRTC brings to organizations of all types and sizes are enormous. Ubiquitous voice, video, and data for gaming, customer service, communications and personal and group engagement opens a new world of possibilities for innovation and disruption.

The transformative power behind WebRTC is that ordinary Web developers using just JavaScript Application Programming Interfaces (APIs) can craft fully functioning voice, video and data collaboration applications or embed these capabilities within other applications with just a few lines of code.

A WebRTC Primer
WebRTC (Web real-time communications) is an effort to create an open framework for embedding real-time communications capabilities into Web browsers. WebRTC allows HTML5 Web programmers, with no telecommunications skills and using simple Javascript APIs, to surface real-time audio and video functionality in Web servers and in browser-based applications running on computers, laptops, tablets and smartphones without the need for browser plugins or third-party applications.

Two standards bodies involved in creating the WebRTC standards include: the World Wide Web Consortium (W3C) and the Internet Engineering Task Force (IETF). The W3C is tasked with creating the Web APIs used in WebRTC while the IETF focuses on the underlying communications and data transfer protocols. Together, both groups collaborate on WebRTC specifications.

Powered by a Triangular P2P Architecture
The WebRTC architecture involves Web servers and browser clients. The Web server "serves up" Web applications with embedded Javascript, and the browser clients (PCs, tablets, smartphones) run the Javascript application. Traditionally, Web browsers have communicated only with Web servers. What is unique about WebRTC is that the Web application can now enable peer-to-peer (P2P) communications between two browser clients (See Figure 1).

Figure 1. WebRTC's Triangle Architecture (Adapted from "WebRTC: APIs and RTCWeb Protocols of the HTML5 Real-Time Web", Johnson, Alan B. and Daniel C. Burnett, First Edition, September 2012, Digital Codex LLC)

While the control data flows between the browser client and the Web server, the audio and video streams flow directly between the browsers. Directly transmitting media between browsers is very useful because voice and video are very sensitive to network latency and jitter, and the direct transmission eliminates additional paths for traffic to travel, on which it could encounter additional impairments.

WebRTC enables point-to-point browser communications as well as multipoint communications sessions. In a multipoint session, each browser sends and receives audio, video and data streams to and from every other browser in the session in a fully meshed configuration (see Figure 2).

Figure 2. Fully Meshed Peer Connections in WebRTC Multi-Point Communications Sessions

Keep in mind, WebRTC will not scale particularly well in many-to-many situations due to the processing power and network bandwidth required for all of the individual peer-to-peer connections that must be established. Consequently, audio and video bridging infrastructure may be required for large meetings with numerous endpoints.

The good news is that the majority of multipoint audio or video meetings typically involve only three or four endpoints. But these have typically been room or group endpoints. WebRTC will enable individuals to meet in multipoint video conferences, and recent data indicates that the number of endpoints participating in such conferences is increasing because people no longer congregate in three to four conference rooms for video meetings.

Next page: WebRTC Requires Directory Services





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