Interconnecting world.

Today IT proposes many solutions to create secure data channels for audio, video and other media data. There are many standards and components behind of them.

We want to share VoIP related source code and ideas implemented by us - it is VoIP/SIP/RTP toolkit to allow IP calls and softphone based on lib. It is work in progress yet.


The softphone and toolkit are written in C++ and can be compiled for Windows/OSX currently. Linux is in progress.

Please contact us to get more information.

Source code is available at Bitbucket's repository .

Open standards

Toolkit is based on already known standards:

Supported audio codecs are G711 / iLBC / iSAC / Opus (stereo at 48KHz).

The toolkit is compatible with most modern SIP phones.


  • supports SRTP
  • traverses NATs via ICE
  • supports OPUS codec at 48KHz in stereo