Flash RTMP SIP Gateway

Flash VoIP gateway which based on top of Wowza Media Server

Flash VoIP gateway is based on Wowza Media Server released by Wowza Media Systems. Within the given product, Wowza is implemented to provide audio/video stream transport over the RTMP protocol, while Flashphoner core works with SIP/RTP protocols and provides integration with VoIP gateways.
Features | Get developer license

Download
Get License
FREE
Call me

JS Click2Call

The most popular use case.
Makes site-calls a commonplace.


The product is well-suited for the existing Wowza Media Server based projects that require adding integration options with the SIP network and support for incoming and outgoing audio/video calls form/to PSTN GSM and SIP devices to the existing Wowza application.

Log in
Log out
Chat
Call
1
2
3
4
5
6
7
8
9
*
0
#
  Login
Log in
Cancel
  Incoming call
Caller
calls you
Answer
Hangup
Connecting...
  Security panel
Please allow access to devices
  Transfer
Transfer this call to
Ok
Cansel
  Chat

JavaScript Phone

Made by using Flashphoner JS API
Extremely handy for web-calls.

The product is well-suited for the existing Wowza Media Server based projects that require adding integration options with the SIP network and support for incoming and outgoing audio/video calls form/to PSTN GSM and SIP devices to the existing Wowza application. Try this.

Open source

Flashphoner client is open source.
Easy to customize and integrate.

Find it on GitHub

Click2call
JS Phone
Open source

Flash RTMFP Media Server

Flash streaming media server that supports the RTMFP protocol

- Works over UDP protocol
- Protocol level AES encryption
UDP ensures low-latency streaming, which enables developers to build advanced Flash VoIP applications based on RTMFP Media Server.
AES encryption ensures protocol security.
Features | Get developer license

Download
Docs

Low Latency

The product is designed for developing Flash to Flash applications, where low-latency is a critical design consideration. Java-based extendable server side.

Web Call Server 2.1 - Stable

Web Call Server 3.0 - WebRTC preview

more about WebRTC

Web Call Server is a platform designed for low-latency Flash VoIP/SIP audio/video communications

The server side of the product works using SIP 2.0, RTP, RTMFP protocols. As compared to RTMP-based technologies and products, RTMFP has a number of advantages, namely:
- Works over the UDP protocol
- Protocol-level AES encryption
Features | Get developer license

Download
Docs

Low Latency

UDP ensures low-latency, which enables developers to build advanced Flash VoIP applications based on RTMFP SIP Gateway.
AES encryption ensures protocol security.
The product designed for communications between web-applications on top Adobe Flash Player or Adobe Air platform and VoIP gateways, SIP, PSTN, GSM devices from other side.

Websocket SIP Proxy

For SIP messaging
directly from the browser

Stateful SIP proxy follows the SIP RFC3261 specification and supports SIP 2.0 and HTML5 Websocket protocols. It may be used to integrate an HTML5 web application with standard VoIP servers that do not support the Websocket protocol.
Features

Download
Docs

Signaling on client side

WSSP is an experimental product, with SIP signaling implemented using Javascript and executed by the browser. Media traffic goes through the Flash swf application and Wowza server to be subsequently converted into the RTP protocol to provide compatibility with VoIP gateways and SIP/RTP devices, PSTN network and GSM gateways. The client-side WebRTC is disabled and is not currently used.



Flashphoner Web Call Server 2.1

Web Call Server is a platform for
low-latency voip communications

Flash RTMP SIP Gateway

RTMP-SIP gateway
for web-voip communications, based on Wowza server

Websocket SIP proxy

For SIP messaging
directly from the browser via HTML5 Websocket SIP proxy

or mail to: [email protected]
[email protected]
[email protected]
You can get the license one for all products for free
Don't hesitate to contact us via ticket-system and forums
The product has client-side and server-side. You can get the source code of client-side on the Github website.
Here you can find the document base


Comparison of RTMP-SIP Products September, 8

Web-phones are very popular now.
Voip market is growing fast, but much more faster
growing market of browser web-telephony.

There no many players at this market
They are few, but they are all different and interesting.
more…


Java Script Phone – How It Works? August, 5

Obviously, you can not make Java Script Phone.
JS technology simply does not allow it.
However, it is possible with Flashphoner JavaScript API.

How does it work?
The scheme is very simple.
Flashphoner server provides an API for flash applications allows you create your own flash-phones.
We made it as a library flashphoner_api.swc.
(API description here)
more…


JavaScript Phone With Flashphoner July, 27

JavaScript better then Flash for any user interfaces in the web.

Why?
There are specific reasons
1. Load faster
2. Work faster
3. Language is easier
4. Made faster – no compile needed
5. GUI elements can be placed in any place on th page
more…


How to try May, 18

Flashphoner – the only solution on the market with such good price/quality ratio. You couldn`t find software with this list of characteristics and for this price. Also Flashphoner has Free Developer Version for 10-connects (unlimited in time).


Easy integration May, 18

In this age of rapid changes, any software is valuable not only by their features, but mostly by simplicity of the integration into existing complex. Flashphoner made great strides in this direction. Integration takes 1-2 weeks in average, including testing and debugging. Reasons – detailed documentation and jet support team, which helps implement Flashphoner in a few days.


Load Balancing May, 18

When you provide users with a new feature – be ready for impact. When the calls flow will increased several times – you can easily balance the Flashphoner load, by placing it on multiple servers. Flashphoner load monitored by a special http interface, which issue this statistics of each server on request from your web server. So your load balancer can easily track Flashphoner load statistics and distribute connects for the optimal load blancing.


Flashphoner default client May, 18

Yes, except the server side, Flashphoner also has a client applications in complete. There are two kinds of this – Web phone with many buttons, and Click-to-Call button with one “Call button” and predetermined call destination. Both are open source and you can easily change their interfaces and functional. Both have Java Script API.


Support for DTMF (tone dial) May, 18

DTMF (Dual-Tone Multi Frequency), or tone dialing – the basis of voice menu navigation – is absolutely essential if you are processing a lot of calls. For many years, organizations around the world to save on phone consultants wages using all the advantages of voice menu (IVR, Interactive Voice Response).
Internet telephony does not lag a step, and Flashphoner – Flash-VoIP server (gateway) also has DTMF support.
This means that users, whom call you from website, will be able to use your convenient voice menu.


Java Script API May, 18

Flashphone, built into the website, often require automatic control. You want the users to manage their click2call buttons or web-phones from the control panel of for your service. Flashphoner client application has muli-functional Java Script API.
Your site will be able to fully manage your flashphone by the special scripts functions.


Support SIP 2.0 May, 18

SIP protocol – the heart of Flashphoner. SIP support, this universal protocol for establish call sessions, makes it possible to call from Flash to anywhere. Flashphoner working on the RFC3261 specification.
At this point, SIP – is the best protocol for establishing call sessions. SIP also supports such features as HOLD, TRANSFER, CONFERENCE and much more.
All of this enhances Flashphoner opportunities.


Built in load balancer May, 24

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.
more…


Flashphoner Web Call Server 3.0 WebRTC HTML5/Websockets Preview Has Been Released April, 29

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.
more…


Flashphoner RTMFP SIP Gateway Has Been Renamed to Flashphoner Web Call Server April, 16

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.
more…


The New Major Version of Flashphoner RTMFP SIP Gateway – 2.1 Has Been Released March, 14

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.
more…


Flash – SIP Communications over RTMFP October, 17

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.
more…


Flashphoner RTMFP SIP Gateway October, 12

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.


Flashphoner RTMFP Media Server October, 10

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 and SoliCall Improve Quality in Flash-Based Solutions September, 13

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.
more…


Do You Think HTML5-WebRTC Phone Ready? Not Yet July, 24

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.
more…


More About RTMFP July, 18

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.
more…