Web-2-Phone - Flash-to-SIP gateway


Nowadays customers are looking for effective and convenient ways to use IP Telephony services. It is very important for them to have different options to place call – either using regular PSTN lines, or using PC Softphone applications, or using Dialers for PDA and communicators.

Today customers can use Web-2-Phone - additional way of communication where customers can use calls using just PCs with internet connection.

Web-2-Phone as Flash-to-SIP gateway that allows to make audio calls directly from webpage. User don't need to install anything. Calls can be place from webbrowser and with a single click.
Web-2-Phone supports SIP protocol making smooth integration with major IP telephony solutions.

Web-2-Phone consists of:
- client side, which is written in Flash, runs in 98% browser at different OS (Windows, Linux, Mac);
- server part, which runs in Windows OS (2000/XP/2003 Server) or Linux/FreeBSD, supports up-to 30-500 concurrent connections.

Web-2-Phone has following features and parameters:
  • Protocol: SIP
  • Codecs: G.711/G.729/Speex/iLBC
  • Operating system: Windows, Linux, FreeBSD, Solaris
  • Browser support: all that has Flash installed
  • Number of concurrent calls: up to 300-500
  • Compatibility: Asterisk, SIP Expreee Router (SER/OpenSER), Kamaillo, Cisco, Alcatel-Lucent and others
Available documents:
Visit our Web-2-Phone Live Demo


Enterprise products

 MESSiX
End-user products

 PSGw