We are eager to share with you a piece of news that may be of interest for you.
1. We have changed the name of our core product Flashphoner RTMFP SIP Gateway 2.1
to Flashphoner Web Call Server 2.1
2. Preview with WebRTC support –
Flashphoner Web Call Server 3.0 WebRTC HTML5/Websockets preview – has been released.
WebRTC – is a browser technology similar to Adobe Flash Player
featuring VoIP functions. The key difference is that WebRTC doesn’t function as a third-party plug-in
(Adobe Flash Player), but as an integral part of the browser.
To put it in simply, it is Javascript/HTML5 API for VoIP.
WebRTC – is a high-flying, future-oriented technology.
For the moment being, WebRTC stack has been implemented for production usage only in the latest Google Chrome versions.
Please, try out Flashphoner Web Call Server 3.0 WebRTC preview.
This version supports both RTMFP and WebRTC in one browser phone.
We took care to enable simultaneous operation of these options from the same Javascript/CSS interface.
WebRTC protocols are also used as HTML5 Websocket protocol. It means that you have two implementation options: one entirely based on Adobe Flash Player, the other – on Javascript/CSS/HTML5.
Take the following steps to try it out:
1. Download
2. Install
3. Switch on WebRTC and configure the calls on your Asterisk server
4. For further information and limitation of WebRTC use with Flashphoner Web Call Server 3.0, please refer to our WebRTC documentation.
What goes next?
WebRTC performs media streaming via the SAVPF(SRTP+RTCP feedback) with G.711 and VP8 codecs.
Currently, there are few devices and softphones supporting these protocols and codecs.
We are trying on WebRTC connection of a browser with any SIP device no matter whether the third-party device supports SAVPF, VP8 or not.
We hope you find the information useful.
For more information, contact as at the address specified on our website: flashphoner.com
Best regards
Flashphoner team.
