Video RTC

Video Real Time Communications.

Video Gateway

Our Video Gateway connects peers or endpoints between WebRTC, RTMP and SIP technology to create advanced Video services.

WebRTC (Web Real-Time Communication) is an API definition drafted by the World Wide Web Consortium (W3C) that supports browser-to-browser applications for voice calling, video chat, and messaging without the need of either internal or external plugins. RTMP is an open protocol for Adobe Flash Player compliant browsers.

Video Gateway works in common hardware or Cloud VM servers configurations, providing a highly scalable base system to meet all customers’ business and technical requirements.


  • Software-Based, 100% binary packaged.
  • WebRTC / Flash compliant Browsers.
  • SIP Connect for Contact Centers.
  • Linux Debian 64bit kernels.
  • Special Use Cases available.
  • Mobile SDK for Android & iOS.
  • Web Panel management.
  • Debugging & Logs.


Cloud Ready

Our Video Gateway solutions are software based and ready to run over Cloud environments and Virtual Servers. You can setup our Video RTC platforms over Amazon EC2 servers or any Private/Public Cloud.

Contact Centers

Our Video Gateway is specifically designed to connect efficiently existing Call Centers with our advanced SPLIT module running over SIP and adding live Video RTC to your business calls interactions.

Web Browsers

Our Video Gateway runs over all existing Web Browsers with native WebRTC inside functions and javascript. To connect others Web Browsers, we can add a special plugin and/or a Flash Player for RTMP.

Android & iOS

Our Video Gateway connects endpoints based in a native/hybrid Android or iOS app that you can create and customize according to your own requirements. Our SDK works with standard mobile development tools.

Special Extensions

This powerful RTC solution is fully compliant with

  • SPLIT Forward & Reverse

    Connector for Contact Centers

  • Web & Mobile SDK

    Build hybrid applications

  • Video Recording

    Record Video Calls

  •  CoBrowsing

    Real-Time between two Peers

Video Protocols

Web Real Time Communication


WebRTC (Web Real Time Communication) is an API definition drafted by the World Wide Web Consortium (W3C) that supports browser-to-browser applications for voice calling, video chat, file sharing without plugins.

Real Time Messaging Protocol


The Real Time Messaging Protocol (RTMP) is an open protocol developed by Macromedia / Adobe for streaming audio, video and data over the Internet, and ready to run with any Web Browser with a Flash player installed.

Special Functions

Live Chat

Video Gateway supports Live Chat feature through a DataChannel. You can customize all the User or Agent front-end in Flash or HTML5 according the gateway mode you have deployed for your service. Live Chat is designed to allow quick written messages between peers.

File Sharing

Video Gateway supports sending / receiving files between peers during a live session. You can customize both the User and Agent front-end in Flash or HTML5. File sharing can be linked to any kind of file directory and customized thanks to the DataChannel.

Screen Sharing

Video Gateway supports to share a screen or window from the Agent peers to manage an advanced Call Center interaction with Users. A special extension can be added on demand for your Video RTC projects and it works with most Web Browsers.

Ready to purchase?

Please complete the following form to talk to Sales. Let us know how about your VideoRTC inquiries, and we’ll get back to you within business hours.

For support requests, file a support ticket.

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