Web-based real time communications (WebRTC) is widely coming into its own as a whole new paradigm when it comes to communications in general, offering users and developers alike a host of new options and new possibilities in most every field from conferencing to gaming. But some are starting to suggest that WebRTC may be more than just a clever new trick for developers to use. Some are suggesting that WebRTC may well be the new gold standard when it comes to cloud-based communications, and those people—like Orange Labs' Jamil Chawki—are building a pretty impressive case around the idea.
Chawki recently detailed how WebRTC was rapidly gaining ground in the field, describing how the variety of applications that could fall under the banner of cloud technologies—like services (SaaS) and infrastructure (IaaS)—could get a little extra boost by bringing in communications as a service (CaaS) to the mix, which isn't a bad way to talk about WebRTC as a whole. Since WebRTC can provide real-time data processing to enable both audio and video communications, it's in line to work well with the other cloud technologies as a supplement, as well as as a standalone communications service. That's enough to get Chawki, among others, thinking like this could be the new standard.
Chawki went on to describe how a new standard in CaaS had been in demand for over a year, and the Internet Engineering Task Force (IETF) and the World Wide Web Consortium (W3C) got to work so doing. The actual workings behind WebRTC can actually be distilled down into simpler terms, involving a WebRTC client and a pair of APIs known as Media Stream and Peer Connection. The WebRTC client uses a combination of HTML5 and JavaScript to process the call between the browser and the server. Then the media is transmitted via Peer Connection, and Media Stream steps in to manage the various interfaces that go into making a WebRTC call, be said interfaces just a microphone and speakers, or stepping up to a webcam or beyond.
But what really makes WebRTC a good candidate for a new standard, at least as far as Chawki and the like are concerned, is that there are beta versions of WebRTC already available for use. Naturally, there's still some room to grow here—Chawki notes that several elements have yet to be finalized in WebRTC, like NAT traversal and coding functions—but with several Web browsers already offering at least some version of WebRTC, it's already setting itself up to be a solid new standard.
A standard, at least in the conventional sense, needs to be widely used, and there's little doubt that WebRTC—once all the kinks have been fully ironed out—is likely to prove a standard on several fronts, making setting up conferencing or even head-to-head gaming as simple a process as opening up a Web browser and turning on speakers and the like. With that kind of ease of use behind it, and that kind of utility as well on hand, it's not out of line to suggest that many developers, and many users, will turn to WebRTC tools on one frame or another.
Calling WebRTC a new standard may be a bit premature, but if the path of technology continues in the direction it's already looking to go, then this new technology may well ultimately prove a standard by the time it's all said and done.
Edited by
Alisen Downey
