We have added load balancing to Flashphoner Web Call Server product.
With the load balancing feature you can allocate your users between several server nodes.
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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.
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We plan to release WebRTC support in April – May 2013.
So, according our plans we are going to present new product with RTMFP and WebRTC support simultaneously.
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You may know that the Adobe Company has recently published the RTMFP specifications.
It allowed us to improve protocol implementation, as well as to introduce several major improvements related to audio and video quality, latency and stability of the server.
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Flashphoner team proudly presents to Flash and VoIP community our two new products that may be interesting for VoIP experts as well as Action Script developers. Both of these products take advantage of newer RTMFP protocol where letter “F” means transition from “Message” to “Media Flow”.

Principal difference between these two protocols is how they communicate over the network. RTMFP is based on UDP, whereas RTMP utilizes TCP. This change gives a number of specific advantages for audio/video delivery in real-time. While most of the products with similar functionality are still using RTMP, we’ve made the transition and have got some impressive results.
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Finally we released beta of our new product – Flashphoner RTMFP!

This is new generation of Flashphoner product.
Our team did a great job and made two huge improvements:
1. Support of RTMFP protocol, which allow you make calls from browser with really miniature latency.
2. Run away from Wowza Media Server and came to our own RTMFP Media Server, which makes whole product more compact

Please try our beta version and send your feedbacks to [email protected]
You can download it here.

Today we release beta version of the new Flashphoner product – RTMFP Media Server.
This is simple mediaserver with the RTMFP protocol support,
which allow descrease latency to the minimum in the web-conference and videochats.

Please check it here.

Flashphoner, provider of Flash-to-SIP gateways, and SoliCall, provider of audio quality solutions, announced today on a combined solution that improves audio quality in Flash-based systems.
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Some of our customers asks:
- So what about HTML5 phone? Is it already done?

In this article, we want to explain situation regarding HTML5, SIP, WebRTC phone solution for Windows, iOS, Androind, iPad, iPhone.

Some peoples reads tech news and thinks that future is already here, and HTML5 is a standard, which gives us possibility to create real time audio/video communications in desktop or mobile browsers like iPhone iPad devices.

Unfortunatelly it`s not true.
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Why RTMFP

RTMFP is protocol based on UDP, developed by Adobe for low-latency streaming.

Flash platform uses RTMFP for peer to peer connections and media relay between  Adobe Flash Player instances. This feature supported since Flash Player 10.

RTMP protocol is implemented in many streaming solutions like Wowza Media Server, Flash Media Server and other servers designed for streaming and broadcast targets.
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