Bridging the gap between WebRTC, VoIP and phones

The WebRTC protocol has shown great promise for integrating telephony capabilities into modern enterprise applications. But challenges remain.

The WebRTC protocol promises to make it easier for enterprise developers to roll out applications that bridge call centers as well as voice notification and public switched telephone network (PSTN) services. Additionally, the WebRTC project provides browsers and mobile applications with real-time communications capabilities via simple APIs, and is a great choice for implementing the call center side of the apps. WebRTC is essentially a set of JavaScript APIs baked into browsers that let developers build applications in order to enable users to make browser-to-browser calls without external plug-ins.

"It's undeniable that the WebRTC protocol is disrupting the Voice over IP (VoIP) industry and redefining what communicating means," said Sacha Nacar, developer community manager at Voxbone, which sells local phone numbers around the world that link to SIP) endpoints. WebRTC enables secure real-time communications applications in the browser without requiring any plug-in or fancy equipment on the client side. In significant ways, WebRTC and SIP overlap in function. But to more fully understand the issue, developers need to consider other important parts of the communications process like signaling or being able to connect a wide range of devices.

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