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