Press about us
"We have the chance to have a look and test your software which downloaded from your web site
and we find it perfect solution to combine with our existing VOIP product."
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RSDevs.com - PSGw details
PSGw 1.6 for Linux (SIP version) details
PSGw can handle only 1 (ONE) simultaneous call due to the fact that Skype can handle only one call.
||PSGw uses preinstalled Skype client and Skype API. Thus it is necessary to have Skype running and logged in.
||PSGw is compatible with most of SIP applications and devices. PGw supports SIP proxies and SIP registrars with authorization.
Images with typical PSGw configuration for single computer is available here, for enterprise configuration here.
Also avaliable PDF White Papers.
- Linux (kernel from 2.6)
- Skype for Linux (since version 1.2) istalled
- sound card with OSS software support
- GUI (Kde, Gnome, other)>
- 16 MB hard disk space
- 256 MB memory
SIP module features
- registrar support
- proxy support
- authorization for proxy and registrar
- external IP address mapping support
- adjustable UDP port range
- "arbitrary" mapping of SIP names to Skype names
- STUN server support
- adjustable jitter buffer
PSGw supports codecs:
PSGw can run as standalone SIP endpoint. It also can register at SIP Registrar.
PSGw can use SIP Proxy server to accept and place calls. Very often SIP Registrar works also as SIP Proxy.
Registrar and proxy authorization
PSGw can use basic SIP authorization to make secure access to SIP Registrar and SIP Proxy server.
Map SIP destination to Skype users
PSGw has powerful feature "Map SIP destination to Skype username" (also called "Arbitrary mapping"). This feature
allows to "map" SIP destination username to Skype username.
How it works: assumed that SIP endpoint calls PSGw (running at IP address 192.168.0.1)
with SIP address "firstname.lastname@example.org". PSGw uses "myskypeintheoffice" as Skype destination
and connects call to Skype user "myskypeintheoffice". Anything entered before "@" in SIP address will be regarded as Skype username.
Jitter buffer is used to mimizime the influence of network conditions. It also helps to maintain the voice quality.
STUN server helps to bypass firewall or NAT (Network Address Translation).
This feature allows to change SIP User-Agent field.
PSGw for Linux uses SIP protocol version 2.0 and is compatible with most of SIP applications and devices.
Note for Asterisk users: PSGw for Linux is compatible with Asterisk.
February 22, 2010.
New version of H.323/SIP SoftPhone released
February 21, 2010.
New version of H.323/SIP MCU (Conference Server) released
October 2, 2009.
Press about us
May 18, 2009.
RSDevs.com is one of the Top 10 Finalists of Cisco Developer Contest !!!
March 20, 2009.
RSDevs.com introduces H.323/SIP Video Softphone for PDA -
Windows Mobile 2003/5.0/6.0 Video SoftPhone for SIP and H.323 networks
July 14, 2008.
RSDevs.com introduces "Web-2-Phone" - Flash-based phone with SIP/H.323/Skype support
July 19, 2007.
PSGw 3.5 (SIP version) demo released
February 1, 2007.
PSGw for Linux 1.6 (SIP version) released
January 1, 2007.
Merry Christmas and Happy New Year!
July 24, 2006.
PSGw for Linux 1.5 (SIP version) released
July 18, 2006.
PSGw 3.5 Standard and Pro now has unique feature - arbitrary SIP destination!
July 1, 2006.
PSGw 3.3 Pro (SIP and H.323 version) allows to set SIP and H.323 DTMF type
June 29, 2006.
PSGw 3.3 Standard (SIP version) allows to set SIP DTMF type
June 20, 2006.
PSGw for Linux 1.2 (SIP version) is released for beta-testing!
May 18, 2006.
Updated PSGw has codec management feature and prefix for Skype username