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WebRTC primer: Using Web browsers for calls and video conferencing

This primer explores what WebRTC is, how it's being used, its potential impact on the network and standards under development.

In its simplest form, Web Real-Time Communications (WebRTC) allows you to use a Web browser as either a telephone to talk, text or chat; or as a video endpoint. The technology accomplishes this by adding a snippet of JavaScript code to Web browsers.

WebRTC is already fully enabled in the Google Chrome and Mozilla Firefox Web browsers, although these are prestandard implementations today. Other companies are also embracing it, most notably TenHands for video conferencing through Facebook, and AddLive for simple video conferencing.

Enterprise WebRTC opportunities: Direct customer engagement

One of the most promising aspects of WebRTC is customer engagement on websites. Companies running websites often look for ways to engage visitors through pop-up chat windows and other methods. With WebRTC, visitors can click a button to call or video chat with a call center agent immediately -- no more waiting.

Potential impact on the network

From a Web developer's perspective, being able to add call or chat features to the front end of the application means that someone sitting at their desk can call a co-worker with a browser-to-browser connection. The call no longer needs to route through the phone system, so calls could require fewer hops; they also lack the quality control or security policies normally associated with enterprise voice calls.

From a network perspective, this might sound scary. "This is one of the most interesting considerations for the enterprise: potential battles between those running networks and those running applications," said Irwin Lazar, vice president and service director for Nemertes Research.

WebRTC will lead to  cool and interesting experimentation and innovation, Lazar said. "But we may see a bit of a battle between those who want to control it and those who want to leverage it to its fullest potential," he added.

If companies decide to use WebRTC as a centralized initiative, and as all of these WebRTC-enabled applications pop up within other applications, it may become difficult to control quality, performance and the impact on the network.

WebRTC standards

Two groups, the Web Real-Time Communications Working Group and the Internet Engineering Task Force (IEFT), are currently working on standards for WebRTC.

The Web Real-Time Communications Group is focusing on how to embed WebRTC into a Web browser, what the job description should look like and how to define the different capabilities.

IEFT is defining the back-end details necessary for all WebRTC-enabled Web browsers to be able to talk to one another -- such as codecs, security mechanisms for encryption, standards to bring a third person into a conversation and how to establish participants' identities.

"The biggest outstanding issue in the standards world right now is that no one has agreed on which video codec to use," Lazar said.

Video calls made through WebRTC need to be encapsulated, and so far there's no consensus about which video codec to use for this encapsulation.

"There's a very well established codec, H.264, which nearly every video conferencing system in the world uses. But using it requires a license. If you embed this capability into the Web browser, who pays that license fee? Today, video conferencing companies pay and pass the cost along," Lazar said.

Google created its own video codec, VP8, but hasn't released all of the details about how it works. And Microsoft has already said it has no plans to use VP8 on Internet Explorer.

Moving forward with WebRTC

Awareness of WebRTC will expand in 2013, and many more applications should go mainstream. Right now, only about 15 companies are delivering small-scale WebRTC applications.

"All of the larger unified communications vendors -- Avaya, Cisco and Microsoft -- are working on WebRTC within their platforms, which they'll likely roll out by the end of this year," Lazar said. "Nailing down standards for the one video codec should help speed things along, but a lot could depend on whether or not Apple makes a move to embrace the technology."

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This could have a great deal of promise for remote pairing initiatives. We have tools like that already do much of this, but the voice and video over the browser are interesting wrinkles. I'm placing my bets on an open codec for video transmission, as I don't see it getting a big adopt without it. as to the QoS issues, on campus it would probably be very minimal, going through the open internet is another story. Reminds me a lot of the early days of VoIP :).
Web based UC can take many forms. Besides WebRTC's various incarnations (including at least 2 from Cisco: Spark and Remote Expert), good BYO video chat tools include Google Hangouts, Skype, Vidyo,, WebEx, and others. Add legacy standards-based videoconferencing from room systems and laptop apps like Jabber-Video. Forget trying to manage and secure user endpoints. Enabling all comers allows extranet collaboration which creates value. Create the server infrastructure that authenticates, transcodes, and bridges in a secure DMZ, on your network or in a cloud-based network operations center. Security is inherent (when engineered properly) because communications apps stay connected and only pass defined media types, not permitting hacker re-tries.