WebRTC Expert Feature

May 08, 2014

Voxbone Offers Easy WebRTC Option

Today, Voxbone introduced WebRTC into its global SIP, IP and local number network.  Voxbone delivers a global IP backbone and SIP infrastructure and dial in numbers in over 50 countries worldwide. Currently, these are delivered to the customer over SIP trunks. Voxbone just added WebRTC as a new end user access method to the network.

The implementation enables customers to use WebRTC with the Voxbone network -- all they have to do is put a hyperlink on a webpage with the right embedded URL indexes. That link sends the user to a Voxbone server that does all the WebRTC work. Voxbone gets the media, and then builds a peer connection through the Voxbone IP backbone to a media server that does the transcoding and protocol matching so the call “magically” arrives on your SIP trunk as a SIP session. In other words, they deal with all of the issues and you get instant WebRTC into your existing SIP environment. 

As Voxbone already has a very large market presence in both the global conferencing and cloud contact center markets, this is an obvious advantage for their users. Voxbone does charge for the WebRTC sessions with this approach, delivering them as part of the SIP trunks. The SIP trunks can be time multiplexed across the globe and across access in over 50 countries, the utilization levels can be high and there are not minute charges, just trunk capacity. WebRTC can be a convenience, or more importantly, it can be used to avoid even the local phone charges that exist in many areas.  And as users in different geographies and time zones connect, they are using the same capacity and the cost does not change.  This makes the solutions ideal for large deployments.

For contact centers, the WebRTC links can match either local phone number or other contexts so when they arrive at the contact center they get the proper routing and skills-based treatment, so even though they are SIP and do not require any new programming, they can enhance the value of customer service.  As an example, language specificity can be implemented by matching the WebRTC sessions to SIP sessions and the local numbers of the language region. From a conferencing perspective, WebRTC enhances the current Voxbone offer. The new solution enables rapid integration of WebRTC with the local dial numbers across the 50 supported countries and the rest of the world where local numbers are not available yet in the Voxbone solution.  For conference operators struggling with service options in underserved telephony areas, WebRTC is an ideal option.

The Voxbone offer removes a critical complexity barrier to WebRTC deployments in the enterprise space.  WebRTC is actually a simple API and works well, however, there are critical system level deployment requirements that must be implemented to have a business level deployable solution. TURN/STUN/ICE capabilities are critical to extend services to users behind firewalls, the WebRTC to SIP transcoding media services are required for easy integration into existing SIP environments, and security, especially as services penetrate the enterprise edge, are critical. With this service, Voxbone has built all of those into the offer. In its cloud delivered service, Voxbone has integrated the TIURN services, the WebRTC to SIP transcoding and security through carrier grade SBCs. In fact, as WebRTC is delivered through a SIP trunk, most organizations will have simple integration.

The overall quality of the service is enhanced as Voxbone will choose an IP path from the end user to the closest Voxbone POP, which exist in every continent and most of the 52 countries. The WebRTC IP traffic is carried on the real-time optimized Voxbone backbone and then delivered through the SIP trunk from a POP close to the SIP trunk termination location. This eliminates many quality issues that could emerge in international IP peering on the open Internet.

This seems to be a great way for enterprises to get their feet wet with WebRTC without having to develop a complete WebRTC deployment. As the cost per session is rolled into the SIP trunk costs, the costs are based purely on the number of active users, so if users switch to WebRTC there is no additional cost. If the usage goes up, the costs will only grow with actual use, which generally translates to other additional revenue or other positive business factors.  As of now, Voxbone has deployed the solution to 10 beta customers. I look forward to hearing more from Voxbone at the Global WebRTC Conference and Expo in Atlanta, June 19-21.

Want to learn more about the latest in WebRTC? Be sure to attend WebRTC Conference & Expo, June 17-19 in Atlanta, Ga. Stay in touch with everything happening at the event -- follow us on Twitter.

Edited by Rachel Ramsey


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