WebRTC World News Feed

October 22, 2013

Flowroute Focuses on SIP Trunking to Enable WebRTC

We use the Web to communicate in many different ways, including email, chat, video conferencing and voice. WebRTC sets out to take that one step further, making communications in a browser faster, easier and with better quality. It’s not just hype, either – there are projected to be more than three billion WebRTC-capable devices by 2016, ready to transform the communications landscape.

Flowroute is a technology company that is innovating in the telecom space. It is a carrier-grade VoIP provider offering SIP trunking and DIDs, and looks to develop software in non-traditional ways. I caught up with Sean Hsieh, chief marketing officer and founder of Flowroute, to discuss WebRTC, the relationship between SIP trunking and WebRTC, and what the company has been up to in the space.

You may remember Flowroute from when FreeSWITCH began using its SIP trunking service to enable interoperability between WebRTC endpoints and the PSTN. The company also has its own project with enabling WebRTC with its PiBX, which the company touts as “unified communications in a byte-sized PBX.” Basically, the idea behind PiBX is imagining what the desktop of tomorrow’s call center agent looks like – there will probably no longer be a physical phone, everything will be accessed on the computer. The screen will hold all relevant data needed to improve the customer experience – ability to call out, CRM data and any other information needed on-demand.

“WebRTC is just huge right now, so I think a lot of players are doing what they think is right in order to grow their market and customer base. We’re focused on SIP trunking. We’ve taken a different approach to telecom and are just trying to provide the best SIP trunking that exists in the market by doing it in non-traditional methods,” Hsieh said. “We’re looking for ways to offer WebRTC connectivity to our customers and we’re figuring out a way to do that in line with our overall vision as a SIP trunking company.”

When it comes to SIP trunking and WebRTC, most providers will say, “Connect with us, send us your audio and we’ll take that audio and in turn send it in toward the PTSN.” On the other hand, Flowroute says, “Send us the signaling but in terms of audio, forget about us – we’re not going to step in the way of your audio – send it directly with our peering carrier and we’ll get it connected so the call audio is transmitted more directly.”

Flowroute’s approach to handling audio is one of its differentiating factors in the industry, in addition to its defining characteristic of being a tech company innovating in the telecom space.

“With WebRTC, you do go with that traditional model of, ‘Send us the audio and we’ll push it around from there.’ So we’re looking at different ways to circumvent that, or how we can build a network where it doesn’t create a bottleneck anywhere,” he said.

Utilizing SIP trunking is a way to maximize the capabilities of WebRTC. It enables web developers to bring in other data and functionality they wouldn’t traditionally be able to with a SIP client on just a computer or phone.

“SIP trunking is at the center of the technology convergence web,” a Flowroute blog post said. “As communication technologies emerge and evolve, SIP stands as a link to unify and optimize your suite of communications platforms.”

The post continues to explain that a strong SIP trunk should work in the background, seamlessly connecting calls where they need to go and staying out of your thoughts so you can devote them where they’re needed most. This is what Flowroute hopes to establish with pairing SIP connections with WebRTC.

The Future of WebRTC

Call centers are going to be one of the first areas to realize the benefits of and implement WebRTC, according to Hsieh. They’re always looking for cheaper and newer ways to implement more seats and better systems to improve the customer experience, and WebRTC gives it to them almost for free – it requires no new equipment, for starters. The ability to click to call and immediately speak to agent creates a new layer of interaction, helping redefine the customer experience. Retail and ecommerce sites are other areas Hsieh believes can really benefit from WebRTC.

WebRTC has not reached mainstream adoption – yet. Browser compatibility still has to grow and improve, and people have to break out of their usage paradigms to realize how they can implement WebRTC day-to-day. Hsieh believes the next generation of tech users will be much more inclined to choose WebRTC as preferred method of contact, whether it’s for personal communications or getting in contact with an organization.

“I’m most excited to see how this will affect the mobile front. I think a lot of the mobile operating systems will be last to adopt WebRTC as a technology just because we’re playing with it on the desktop front right now,” Hsieh said. “We’ll see a huge change when people start using it on their mobile devices and don’t really realize that it’s WebRTC technology that is bridging these gaps.”

There will be many discussions related to SIP and WebRTC at the upcoming WebRTC Conference and Expo, happening Nov. 19-21 in Santa Clara, Calif. One session in particular, “SIP & WebRTC – Working Together?” will feature speakers from Sonus Networks, Constellation Research and AudioCodes to discuss the challenges and methods of integrating WebRTC into SIP and mapping out the integrations strategy between WebRTC and SIP.

Visit www.webrtc.world.com/conference to learn more. 

Edited by Stefania Viscusi
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