From GNU Telephony
Visit our IRC-channel #gnutelephony on freenode.
GNU Telephony is a project to enable anyone to use free as in freedom software for telephony, and with the freedom to do so on any platform they choose to use. We also wish to make it easy to use the Internet for real-time voice and video communication, and in fact for all forms of real-time collaboration. Finally we wish to make it possible to communicate securely and in complete privacy by applying distributed cryptographic solutions. Our goal is to enable secure and private real-time communication worldwide over the Internet that is free as in freedom, and is also free as in no cost too!
How you can participate
Packages for Debian 6.x Squeeze are available by adding
By participating in Debian Popularity Contest, you can automatically give us feedback about the number of installations - thank you!
We are running a generally open wiki for this project. Once you login you can edit any page on this site to correct and improve it. For information on editing, see the MediaWiki User's Guide. You can also help by creating domain calling networks bottom-up, by testing and use various deployment models, and by helping us document basic sipwitch use cases better. You can help by contributing code to the community and communicating freely using free software.
GNU SIP Witch is a secure peer-to-peer VoIP server that uses the SIP protocol. Calls can be made peer-to-peer behind NAT firewalls*, and without needing a service provider. GNU SIP Witch supports using secure telephone extensions, for placing and receiving calls directly over the Internet, and intercept-free peer-to-peer audio and video extensions. GNU SIP Witch also is being introduced as a desktop VoIP mediation service to enable the construction of participatory bottom-up secure calling networks and to enable replacement of Skype with free software and published protocols. As a desktop mediation service, GNU SIP Witch can solve issues like NAT* in one place for all user agents, and offer new ways to route and redirect VoIP much like gstreamer does for desktop media. We have chosen to adopt Friendica in GNU Telephony, to work on extending it to offer realtime secure voice and video with GNU SIP WITCH and ZRTP enabled clients, and to make it formally part of the GNU Free Call Roadmap. (* Work in progress.)
GNU Bayonne is the telephony server of GNU Telephony and the GNU Project. The production release of GNU Bayonne 1 is 1.2.15 and has a long history in production telecommunication environments. GNU Bayonne supports IVR scripting using hardware from Voicetronix, Dialogic, Aculab, CAPI drivers, and Quicklink drivers under GNU/Linux. GNU Bayonne 1 can integrate perl and python applications, and has been commercially deployed in production use for several years. Future releases of GNU Bayonne will be based on ucommon and will further explore it's role as a Telephony integration server.
How we license our code
In GNU Telephony we generally license under the GNU GPL version 2 or later. We are licensing new code under the GNU GPL Version 3 or later and we will re-license future releases of many existing packages this way as well. Some of our C++ frameworks and libraries may also include the same Runtime Library Exception used for libstdc++ in the GNU Compiler Collection.