OUR PRODUCTS

“Bridging the gap between Web and VoIP”

Flash-VideoIO

Flash-VideoIO is a reusable and generic Flash application to enable seamless web video conferencing and messaging applications, and can be used in a variety of use cases related to audio and video communication, e.g., live camera view, recording of multimedia messages, playing video files from web server or via streaming, live video call and conferencing using client-server as well as peer-to-peer technology. We have built many applications using this component, including web-based video conferencing and presentation, Facebook video chat, online video office, and so on.

Generic application for video conferencing, recording and playback
Extensive Javascript API for control and indication
Built-in default video control user interface; allows customizing
Supports RTMP (client-server) and RTMFP (peer-to-peer) streaming, and HTTP playback
Supports RTMFP groups for application level multicast communication
Usable in HTML/Javascript as well as another Flash application
Interworks with SIP-RTMP gateway for web to phone calls
Portable across different platforms and browsers
Enables acoustic echo cancellation and group communication when available

Please visit the project web site and tutorials for details on the open source examples that use this component. Please contact us for professional consulting related to this component and how you can use this in your web site.

View the demo video of voice-and-video-on-web (VVoW) project built using Flash-VideoIO below. Use the full screen mode to enjoy the demo.

RTMPlite (and SIP-RTMP gateway)

RTMPlite is an open source light weight RTMP server written in Python and is ideal for protyping and starting new Flash media streaming related projects. It also includes an open source SIP-RTMP gateway that enables web-to-phone (and reverse) calls by interoperating between Flash Player application and standard VoIP user agents based on SIP and related protocols.

Lightweight RTMP server for live and offline streaming
Built-in SIP-RTMP gateway to enable interworking between web and VoIP
Built-in RTMP client library and application
Translates between RTMP and SIP registration, call signaling and RTP media transport
Supports both wideband (default) and narrowband Speex voice codec
Supports external media transcoding such as Speex to/from G.711.
Comes with example video phone and test client applications
Supports external white-labelled Flash user interface using Flash VideoIO
Also runs on Amazon EC2 machines with different external and internal IP addresses

Please visit the project web site for details on this open source project. Please contact us for professional consulting or an alternative commercial license for this product.

View the demo video of SIP-RTMP gateway in action below. Use the full screen mode to enjoy the demo.

Audio Call	Video Call

AIRphone: desktop video phone

Download Intencity AIRphone now.

This application requires Adobe® AIR™ to be installed for Mac OS or Windows.

Intencity AIRphone is a free desktop video phone application built on Adobe Integrated Runtime (AIR) platform. It uses the Flash-VideoIO component to handle end-to-end media path, and RESTful web service of IIT web conference to handle the signaling. It presents a device user interface on desktop and allows two party video call. The details on how to use follows:

It automatically launches after install, and has built-in auto-update feature.
On first launch, it prompts you to (1) edit your name, (2) take a picture snapshot, and (3) complete registration, so that others can see your name and picture in their AIRphone instance.
After registration on first launch, or on subsequent launches, it automaticaly logs you in using your registered identity. The light on OPEN button blinks slowly in green to indicate that it has logged in. The light is red during login or on error.
To place an outbound call, click on the START button to open the right panel with list of currently online users. Use mouse or Prev/Next buttons to select the user you want to connect to, and then click the START button again. A call invitation is placed, which automatically times out in 30 seconds. If the other user accepts the call, his video appears on the right panel.
When you receive an incoming call, the caller's video appears on the right panel, with a note about incoming call. To accept the call, click on the START button. To reject the call, click on the END button. When the call is accepted, your video starts in the left panel.
To terminate an active call, cancel an outbound call, or to reject an incoming call, click on the END button.
In an active call you can click on the display or sound button to toggle your camera or microphone. You can also control the speaker volume using the volume buttons. The three buttons on the right are general purpose buttons useful based on the context, if displayed on the right panel. For example, during online user list display, the first and third button signal the Prev and Next command displayed on the right panel.
There is a very small RESET button at the end of AIRPHONE text. Clicking the RESET button will remove your registration information from local storage. The registration information is not deleted on the server though, hence you will not be able to re-register using the same name.

Please see the demo video of AIRphone below. Use full screen mode to enjoy the demo.

Flash-Network

Flash Network is a set of library and application to allow building any type of network applications running in the browser. It defines extensive Javascript API for various network socket feature such as client and server TCP socket, generic UDP socket and real-time RTP/RTCP socket. It uses Flash Player and Adobe Integrated Runtime (AIR) to facilitate seamless integration with your web application. The idea is to allow any type of network application including multimedia communication application from your web page.

Please visit the project web site for getting started with this project as a software developer. We also have a demonstration of Web-based SIP phone built using this project. Please contact us for professional consulting of this product.

Please see the demo video of Web-based SIP phone below. Use full screen mode to enjoy the demo.

39 Peers (SIP-related protocol stack)

The 39 Peers project implements protocol stack for SIP and related standards in Python for use in real-time communication applications in client-server as well as peer-to-peer mode. In particular it contains implementations of IETF standards for SIP (RFCs 3261, 3263, 3264), SDP (RFC 4566), RTP/RTCP (RFC 3550, 3551), authentication (RFC 2617), DTMF transport (RFC 2833, 2198), URI (RFC 2396), NAT traversal (RFC 3489bis), XMPP (RFC 3920, 3921) and sample applications for SIP user agent, SIP proxy and registration server, peer-to-peer SIP adaptor, Google chat connector, distributed hash table emulator, etc.

Please visit the project web site for details on this open source project. Please contact us for professional consulting or an alternative commercial license for parts of this product.

Licensing

While most of our software pieces are available as open source license, typically GPL or LGPL, we also sell low cost alternative commercial license if the viral nature of GPL is not suitable for your deployment. In particular, the alternative commercial license allows you to combine pieces of our software with your other proprietary elements. Please visit the services page for details on this alternative commercial license.