GNU Telephony

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'''Welcome to GNU Telephony'''
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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!
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__NOTOC__
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== How you can participate ==
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{| style="width: 50%; float: right; margin-left: 2em; background-color: orange;"
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|-
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| style="border-width: 0;" |
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'''Latest News'''
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* Intitial integration between [http://friendika.gnutelephony.org our Friendica instance] and GNU SIP Witch has started. [http://www.friendica.com/ Friendica] is similar to Facebook, but is decentralized, free (as in freedom) software, secured, private, modular, extensible, unincorporated, federated and available at no cost.
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** [http://friendika.gnutelephony.org/display/dyfet/59370 Discussion at GNU Telephony Friendica Network]
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** [https://github.com/dyfet/friendica-addons/tree/master/sipwitch Initial GNU SIP Witch plugin for Friendica at Github]
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* GNU SIP Witch 1.2.1 is now available. New in 1.2.*
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** Support for usercache functionality
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** Revised manpages
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** Fix for cpu loading with subscriber plugin
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Packages for Debian 6.x Squeeze are available by adding
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GNU Telephony is a meta 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 crytographic 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!
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<code><nowiki>deb http://www.gnutelephony.org/archive/ squeeze/</nowiki></code>
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'''Latest News'''
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to your /etc/apt/sources.list.
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You might like to review [[GNU SIP Witch configuration]] after installation.
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We presented "[[Secure Call|Secure Calling]] and Communication Privacy" at this year's Scale/8x conference in Los Angeles, California and I am speaking about GNU SIP Witch and Secure Calling at Harvard later this week.  [[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.
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'''Older News'''
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* Free Software Foundation now supports the work of GNU Telephony through its Working Together for Free Software fund. Further info https://my.fsf.org/donate/working-together/telephony
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'''How you can participate'''
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|}
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 [http://meta.wikipedia.org/wiki/MediaWiki_User%27s_Guide User's Guide].  You can also help by creating domain calling networks bottom-up, by testing and use various deployment models, and by
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 [http://meta.wikipedia.org/wiki/MediaWiki_User%27s_Guide User's Guide].  You can also help by creating domain calling networks bottom-up, by testing and use various deployment models, and by
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helping us document basic sipwitch use cases better.  You can help by contributing code to the community and communicating freely using free software.
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helping us document basic sipwitch use cases better.  You can help by contributing code to the community and communicating freely using free software. You can also help by donating to the Free Software Foundation, which now supports the work of GNU Telephony through its Working Together for Free Software fund - see further info at https://my.fsf.org/donate/working-together/telephony .
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'''Project Status'''
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GNU SIP Witch is now in an early beta release cycle and is being made available initially in several GNU/Linux distributions, including Fedora, and Ubuntu starting with 10.04.  GNU SIP Witch development is focused both on it's role as a desktop VoIP mediation service and as a low latency/low overhead embeddable VoIP telephone server.  In both these roles, we are developing GNU SIP Witch peer-to-peer capabilities to offer direct call interconnection without requiring intermediary service or directory providers.  In particular, we are focused on a "secure domain calling" model at the moment, where GNU SIP Witch offers a primary SIP service for an entire domain and then redirects calls to individual users registered through it.
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GNU Bayonne is the telephony server of GNU Telephony and the [http://www.gnu.org 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, ccscript-4, and ccaudio-3 and will further explore it's role as a Telephony integration server.
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The stable release is [[GNU Bayonne]] '''2''', the current release series is 1.5.x, and currently supports SIP, H.323, and Voicetronix drivers. [[GNU Bayonne]] 2 can be used on 32 and 64 bit GNU/Linux systems, various BSD systems, Mac OS/X, and Microsoft Windows. Work also exists on support for Dialogic, Aculab, and Synway hardware. Other drivers will be added as time and community support allows to be developed.  Future releases of GNU Bayonne 2 will be derived from the Arnhem codebase and focus on becoming a general purpose IP-PBX softswitch.
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== Project Status ==
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The stable release currently performs script driven IVR applications written in [[GNU Bayonne]]'s native [[Bayonne Scripting|scripting language]], as well as access, conversion, and playing of audio from remote URL's. The stable release also performs basic switching interconnect functions, including tone detection and dtmf regeneration, that are needed for basic gateway operations. The latest release can also operate as a SIP proxy and register for external SIP devices, which can be used to build phone systems and gatewaysThe stable release supports integration of external perl, python, php, C#, and Java applications; the ability to perform XML query operations and voice rendering of [[BayonneXML]] documents with a web site; and a build-in webserver offering html pages to browsers and standard compliant [http://www.xmlrpc.com/spec XMLRPC] services for programatic control and integration[http://www.xmlrpc.com/spec XMLRPC] is also offered as a local Unix domain socket if one does not wish to expose the server to remote access, and may be offered over SIP transport as wellBoth future releases of Bayonne and Arnhem will support
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[[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 providerGNU 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 extensionsGNU 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 protocolsAs 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.
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these core capabilities.
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The [[GNU Telephony Open Embedded]] project has recently had it's first success, in building installable packages of [[GNU Common C++]] and [[GNU ccRTP]] for GNU/Linux on ArmThese packages are built for use on ipaq's either using [http://gpe.handhelds.org/ GPE] or [http://opie.handhelds.org/cgi-bin/moin.cgi/ OPIE].  We would like to port a oftphone client like sflphoned and/or Twinkle to Ipaq.
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GNU Bayonne is the telephony server of GNU Telephony and the [http://www.gnu.org 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 yearsFuture releases of [[GNU Bayonne]] will be based on [[ucommon]] and will further explore it's role as a Telephony integration server.
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'''How we license our code'''
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== How we license our code ==
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In GNU Telephony we generally license under the [http://www.fsf.org/licensing/gpl.html GNU GPL] version 2 or [http://gplv3.fsf.org/ 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 include the same [http://gcc.gnu.org/onlinedocs/libstdc++/17_intro/license.html Runtime Library Exception] used for libstdc++ in the [http://gcc.gnu.org GNU Compiler Collection].
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In GNU Telephony we generally license under the [http://www.fsf.org/licensing/gpl.html GNU GPL] version 2 or [http://gplv3.fsf.org/ 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 [http://www.gnu.org/licenses/gcc-exception.html Runtime Library Exception] used for libstdc++ in the [http://gcc.gnu.org GNU Compiler Collection].

Revision as of 20:22, 22 January 2012

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

Latest News

Packages for Debian 6.x Squeeze are available by adding

deb http://www.gnutelephony.org/archive/ squeeze/

to your /etc/apt/sources.list. You might like to review GNU SIP Witch configuration after installation.

Older News

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. You can also help by donating to the Free Software Foundation, which now supports the work of GNU Telephony through its Working Together for Free Software fund - see further info at https://my.fsf.org/donate/working-together/telephony .

Project Status

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.

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.

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