You might not know what Web Real Time Communications (WebRTC) is, but you should. At Vonjour we're working diligently on making it more ubiquitous and empower the voice and video communications across the globe.
WebRTC allows real-time voice and video connections between mulitple web browsers, without any plugins or third-party software—you'll only need the latest versions of Chrome or Firefox.
Web browsers have been powerful channels to surf the web and communicate through text. However, the browser has lagged as a medium to power two voice and video streams.
The underlying the challenge behind browser based voice and video calls has been access to voice and video compression-decompression algorithms, more commonly known as codecs. Traditionally, codec technologies were owned by a few companies, and these companies would charge very costly licensing fees to utilize their codecs in a third-party application.
While codecs were limited in scope, there were also limitations in browsers to send two way communications in real-time. Specifically, browsers were capable of requesting or sending data, but not real-time two-way streaming.
For web based real time communications to be more ubiquitious, everyone needed access to high quality codecs. Google pushed the RTC community forward in 2010 when it acquired GIPs and On2. GIPs was the leading provider of VoIP codecs and On2 made a video codec. Google open sourced both projects, which had profound implications for IP communications and business VoIP services.
WebRTC is a big deal for Vonjour. It allows us to make communication tools accessible on any browser and outside of the teleco network—making it easier to communicate directly with your clients.
Imagine a call center without having to purchase any costly hardware device. Where you can take a call on the same device you're inputting you're working on your CRM or logging a ticket into ZenDesk. It makes for a more integrated experience.
Daniel Tawfik is the CEO of Vonjour.com