What is WebRTC?

The foundation

WebRTC is an abbreviation of Web-Real-Time-Communication. In the first place, the Google Hangouts developer team started working on a solution to integrate real-time audio and video in the webbrowser.
After years of development, it is now accepted and supported by almost all populair webbrowsers. Basic functionallity is sending and receiving audio and video. These days, features like screensharing and realtime collaborative drawing are almost standards in these kind of applications.

Integration with VoIP

As telephony over IP (VoIP) became more mature, developers started to see opportunity’s to integrate these worlds. Now it’s pretty easy (for technical guys) to set up a VoIP connection using your webbrowser. For users, there is a need to make things easier and you don’t want to bother them with technicall difficulties. So there is a need for an abstraction layer, so users can login using an e-mailaddress and a password. When they are logged in, they can place and receive calls using the webbbrowser. Offcource, for the first time they have to give the permission to the browser for using the microphone and speaker devices etc.

Expanding with more features

So, calling using a webbrowser is not very special anymore these days. But as the users started using it, important features which are available on VoIP phones are requested like, can I see the presence of my collegues?

Then we saw the need of these ‘basic’ features and decided to start developing a ‘universal’ platform so most opensource telephony platforms can easily connect to that platform with all features in place. Besides more features are requested and we develop if demand is high and keep users and companies with us.