Sep 14

How Will WebRTC Revolutionize VoIP Service?

VoIP web

Web Real-Time Communications (WebRTC) is a technology that the World Wide Web Consortium (W3C) supports to enable communications in real time, such as instant messaging, voice, video, and data, among open-standard web browsers. Unlike many other new technologies that present themselves as a money-saving way to streamline communications, WebRTC will enable developers to cater to the context of business communications enclosed in existing applications without the need for extra software. Chrome and Mozilla Firefox currently support the WebRTC standard; Safari and Internet Explorer do not, but Microsoft is currently developing a similar standard, and Apple may support WebRTC in the future. (You can still use WebRTC with some Apple devices even without full Apple support, however.)

Using WebRTC for Voice Calls
WebRTC is an application programming interface (API) that communication vendors must supply to make it useful for users. Several basic services exist that can show how to connect two browsers to make a call. WhatsApp is one vendor rumored to use WebRTC to provide web calling capability between users. The premise behind WebRTC is that it provides a new way for web users to communicate through rich video.

3CX has been using this new technology since it was first developed, investing in it to continue its development. Using WebRTC, 3CX can place and take calls to anyone using the Internet without special software. Unlike Voice over IP (VoIP) calls, which require the use of an application or software such as Microsoft Skype, WebRTC can deliver calls instantly online just like using a web page. Development continues on web and video conferencing technology using WebRTC, but when the functionality becomes available, it could eliminate the need for proprietary standards, additional hardware, and software requirements typically associated with such conferencing. 3CX’s vision for this technology is to start a revolution in web communications.

What Does This Mean for VoIP
Although WebRTC is still in its infancy, it’s unknown what the technology’s long-term effects will be for VoIP. Latency, jitter, quality of service, and security may or may not be an issue. All these factors may ride on the surface of a user’s web connection. How secure is the Internet connection? How fast is it? Answers to these questions may determine WebRTC call connection quality.

WebRTC supports a number of voice codecs that are different from those many Session Initiation Protocol systems use. WebRTC uses VP8 for video rather than the commonly used H264 codec. For voice, it uses G.711, iSAC, and iLBC. Based on its network architecture, WebRTC takes web apps that the W3C edits into an API for web browser developers (Google, Mozilla, etc.) on an overall WebRTC platform. Within the platform, a voice engine, video engine, and transport session use the codecs mentioned. In the voice engine, iSAC and iLBC codecs, NetEQ for voice, echo canceler, and noise reduction are employed.

Although the signaling protocol for WebRTC is still undefined, it shares many of the same protocols, such as the Skinny Real-Time Transport Protocol (RTP) and RTP Control Protocol for media transport; Secure RTP for security; Session Traversal Utilities for NAT, Traversal Using Relay NAT, and Interactive Connectivity Establishment for network address translation (NAT) transversal; and the G.711 voice codec.

WebRTC may never fully replace VoIP, but it will battle VoIP for big business in the upcoming years. VoIP software and hardware providers will most likely advocate against the use of the WebRTC technology available through web browsers, but WebRTC may provide a valuable solution to international VoIP blocking.