AudioCodes Edit Series

WebRTC – a web technology that would extend the native capabilities of web browsers to enable voice, video and data sharing without separate client or plug-in software – has been getting plenty of attention lately. Evolving under the IETF and W3C standards process, much of the technology leadership comes from a team of developers at Google (News - Alert) who see it as a game-changing communication technology. Innovative web-based applications include collaboration systems (like Google Hangouts), distance learning, gaming and more.

I see the contact center as likely benefiting the most from WebRTC. Initially offering click-to-call features for web users, WebRTC will put a zero-install experience on virtually every consumer desktop, tablet and mobile – making it easy to ask questions, get support, or otherwise interact with the contact center. 

Not a replacement for existing contact centers, it’s widely understood that WebRTC will be an evolution in the contact center – adding the capability to existing contact center operations and having web surfers call using their browsers, following the same call routine/queuing that today’s telephone callers do. This allows agents to handle callers from either the web or PSTN interchangeably – Nirvana for the contact center manager.

However, this comes with a couple technical hurdles. One immediate challenge is codec support.  For a number of very valid network performance reasons, WebRTC defines the Opus High Definition codec as a means of transporting voice over the Internet. With enhanced packet loss concealment, error correction and outstanding voice quality, Opus was designed from the ground up to deal with common network errors that occur when communicating over the open Internet.

So, how will a contact center support Opus? 

There are well-understood approaches to adding WebRTC and Opus to a contact center.

SBCs, Gateways and Transcoding – The first approach is using WebRTC-enabled SBCs to terminate the sessions and media streams from the web, transcoding the sessions to a format that the existing contact center can handle (usually G.711). Seen as the least disruptive approach, it may suffice for certain applications or as an interim bridge technology, but transcoding does not scale very well and adds latency to web-based calls that could reduce customer satisfaction.

Opus End-to-End – a second, and more popular, approach is to facilitate use of the Opus codec all the way to the agent, allowing the endpoint device to support both G.711 calls from the PSTN and Opus from web callers. Far more efficient than transcoding, support for Opus at the agent makes high-definition calling a reality, improving the user experience and satisfaction.

Realizing that no two contact centers are the same and existing investments will drive much of the decision-making, our strategy at AudioCodes (News - Alert) is to support both approaches – starting with supporting Opus on our contact center-ready 400HD IP Phones. In addition, AudioCodes is also planning support for Opus in our SBC products, facilitating transcoding when appropriate.

With these important technologies in place, we see WebRTC and the contact center as a natural fit – like chocolate and peanut butter, the sum will surely be sweeter than the parts.