This vendor-written tech primer has been edited by Network World to eliminate product promotion, but readers should note it will likely favor the submitter's approach.

The International Telecommunication Union predicts Web Real Time Communications (WebRTC) will be installed in more than four billion devices by the end of 2016. Indeed, the promise of WebRTC is the mass adoption of voice, video and file collaboration, but the question is, how does it fit within existing communication systems?

Drafted by W3C with protocol work done by IETF, WebRTC simplifies the complex world of real-time communication. Although WebRTC isn't confined solely to web applications, embedding real time communications directly into Web browsers has been the focus for most of the industry. After all, the promise of free high quality video and audio in one of the most deployed application in the world (Web browser) is exciting. As browsers work to integrate and refine the technology (Firefox and Chrome support it today), WebRTC holds the promise of instant connectivity through familiar interfaces with little or no additional software.

WebRTC defines browser APIs together with a collection of communication processes and protocols. From a development perspective, core functions are encapsulated into three main JavaScript APIs: getUserMedia, RTCPeerConnection and RTCDataChannel. These APIs are incorporated into browsers that support WebRTC, hence a web developer who has JavaScript programming experience can bring an interactive video collaboration experience to the web. The following diagram illustrates the architecture:  

Roadblocks to scalability

Like any new technology, WebRTC isn't without issues. Traditionally, video and other collaboration systems have used conferencing bridges to link a certain number of participants to a call. WebRTC technology works by changing the structure of the connections between parties.

WebRTC allows a mesh-based technology to enable users to send and receive streams to and from each other. This is not new in the video world, as technologies exist that accomplish this today. Each stream operates independently, which reduces the strains of conferencing applications (as bandwidth doesn't aggregate to a single choke point) unless, of course, bandwidth inefficiencies come into play.

With this mesh technology, WebRTC can, in theory, accommodate an infinite number of participants on a call. But in practice, the more parties that join a call, the more bandwidth that call will take. Bandwidth inefficiencies can mount quickly, as each device connected to the call receives and transmits multiple transmissions. If bandwidth falters, quality suffers and the entire call can fail.

On these more complex calls, signaling factors in as well. In the past, Session Initiation Protocol (SIP) has provided a way to register users and identify them uniquely, as well as to manage call notifications and modifications. WebRTC in its infancy does not include a concrete means of signaling, leaving some basic call functionality up in the air. Without protocols for connecting, disconnecting and identification, disorder can ensue.

Working WebRTC into an SIP world

If WebRTC exists on an island as a solution separate and apart from existing video, voice and file sharing technology, it won't live up to its potential. These issues of scalability and operation will override its accessibility. Additionally, organizations across the board have already invested heavily in collaboration systems, including everything from hardware and special conference rooms to software and maintenance. They aren't going to want to abandon these investments. They will, however, seek to use them alongside WebRTC.

Developers of traditional collaboration technology are in a great position to integrate WebRTC, making it interoperable with their existing offerings. Doing so not only keeps traditional systems relevant but also enables them to solve the problems WebRTC presents. For instance, if WebRTC users can participate in calls hosted by SIP systems, they can dynamically connect to the video core rather than create a mesh amongst themselves. Enabling the users to have the best of all worlds, mesh configuration when it makes sense (limited number of users, bandwidth is sufficient) and hub-and-spoke through the video core when needed.

Integration also lets WebRTC users benefit from some of the core value-add capabilities of existing collaboration systems. Features like automatic muting of callers in noisy locations (when they're not speaking, of course) become all the more critical with the more democratized WebRTC: if anyone in any office or coffee shop anywhere can connect via a browser, quality will suffer without these state-of-the-art controls that are native to high-end collaboration technology.

Delivering on the promise of collaboration

On the surface WebRTC may seem to address video collaboration in the browser, but it's more than that; it has the potential to bring consumers of this technology into your collaboration framework. For WebRTC to be universally successful, a methodology that embraces a hybrid approach and delivers a quality experience for all call participants will win out. Until all browsers can support WebRTC and can do so with this requisite quality, it makes sense to maintain plug-in alternatives and to use these alternatives to fill the gaps in WebRTC functionality.

WebRTC can live up to its hype. It can bring seamless video, voice and file sharing collaboration to the masses and it can do so quickly. It has to move forward together with existing collaboration infrastructures. There's a lot already in place for quality, cutting-edge communication and there's no reason to leave all of that development behind. Integrate it and bring it along to the browser level, and WebRTC will certainly earn its day in the sun.

Ashan Willy, is the SVP, Worldwide Product Management & Systems Engineers, Polycom.