- Fully customizable GUI (open source SWF)
- Call transfer to an external/internal SIP extension
- Nat pass thru / works on any WIFI / 3G / 4G
- Dual call to SIP extension and RTMP extension
- Video and Voice calling enabled
- Auto-register / Auto-dial functions
- Fullscreen for video
- CIF/QCIF default screen sizes
- Bandwidth management
- Presence management
- Integrated Chat
Skinable and CustomizableWe provide you complete SWF source code to change all buttons, colors, fields and revamp your web phone by yourself with a Flash Editor. We also provide different skin modes to quickly change the behavior of your web phone.
No plugin installation requiredUsers unlike to install any plugin during web navigation. Our product runs directly and never require to download anything to the user computer, it works immediately.
Voicecall and Videocall enabledOur solutions are always ready both for voice and video calls. You can create any kind amazing face to face communication services as well as a simple web phone to connect your PBX.
Adaptive Bandwidth ManagementWe use a RTMP protocol to manage both audio and video RT streams. It’s much more efficient and easy to run than SIP. You can read more about it here.
Sorry, WebRTC is not yet ready…WebRTC protocol is not yet ready to run in all web browsers and still on a early stage. Our RTMP based solution is interoperable with WebRTC thru Asterisk, so don’t wait anymore dreaming the future…
Running over any Web BrowserOur solution works on any flash enabled web browser like Internet Explorer, Firefox, Safari, Opera, Chrome... and any Desktop Operating System like Microsoft Windows, Mac OSX or Linux.
IP - IVR
- September 12, 2014
- April 30, 2014
- February 26, 2014