While we haven't used sourceforge, I think in a few years, git makes it particularly easy to push and manage additional remotes. Also, as we are preparing to update all our code around ucommon API release 6 (ucommon 6.0.0), and with it release and distribute updated sets of our core applications and services (including sipwitch 1.4), this seemed like a particularly good time to re-introduce sourceforge. As we produce and update these releases we will also make them available here.
SIP (Session Initiation Protocol) is an IETF standard protocol for interconnecting telephone devices over TCP/IP networks. GNU SIP Witch is a call server which implements the SIP protocol standard while supporting generic phone system features like call forwarding, hunt groups and call distribution, call coverage and ring groups, holding and call transfer, as well as offering SIP specific capabilities such as presence and messaging. GNU SIP Witch will support use of secure telephone extensions and enables communication privacy through the use of peer-to-peer audio and video sessions directly between connected endpoints. GNU SIP Witch also supports placing and receiving calls directly with remote users over the public Internet without requiring the use of mediating VOIP "service providers" in what is commonly called SIP "Business-to-Business" (b2b) calling.... read more
SIP Witch is an official package of the GNU Project as of August 10th 2007. GNU SIP Witch is also part of GNU Telephony & the GNU Telecom subsystem.
GNU SIP Witch is a call and registration server for the SIP protocol. As a call server it services call registration for SIP devices and destination routing through SIP gateways. GNU SIP Witch does not perform codec operations or media proxying and thereby enables SIP endpoints to directly peer negotiate call setting and process peer to peer media streaming even when when multiple SIP Witch call nodes at multiple locations are involved. This means GNU SIP Witch operates without introducing additional media latency or offering a central point for media capture.... read more
GNU Telephony is happy to announce that with the latest release of the GNU RTP Stack, GNU ccrtp 1.5, we are introducing a free software framework for developing both the secure RTP profile for VOIP (as defined by RFC 3711), and also a GNU GPL licensed implementation of Phil Zimmermann's ZRTP protocol for voice encryption as used in "Zfone". By offering a native secure RTP framework that can be directly embedded in newly developed VOIP applications, GNU Telephony intends to promote the development and widespread use of secure and intercept free voice and video communication services worldwide.... read more
Bayonne2 is the telephone server of GNU Telephony. Very recent and rapidly introduced releases of GNU Bayonne2 have focused on expanding the potential use of the GNU Bayonne2 server in several key areas; support for XML application serving, introduction of Bayonne web services for integration and network management, and core features for building Bayonne based office telephone systems.
XML application services have been introduced through BayonneXML, which is a CallXML-like XML dialect. BayonneXML allows one to use a Bayonne server to query a web site, and retrieve a voice navigable XML document. Additional queries can be made, and new documents can be retrieved, based on results of user input on forms and fields. This approach places most of the logic for control of a Bayonne server at the backend of the web site rather than the scripting engine local to Bayonne.... read more
Tycho Softworks and GNU Telephony is pleased to announce the immediate availability of GNU Bayonne2 1.2. With this release, we are delivering new functionality for using Bayonne as a free software based IMS application platform. Included in this release are new abilities to deliver audio from arbitrary url's and audio media formats directly to telephone callers connected via the SIP protocol, as well as the ability to switch and route SIP voice calls. New application integration support has been added to better enable use of Bayonne as a middleware telephony platform and to deliver new B-to-B and B-to-C voice services by sitting between traditional enterprise applications like SAP, SQL Ledger, groupware servers, etc, and your enterprise telephone network.
I will be speaking on November 24th at 16:30 at the International World Forum on Free Knowledge, at the Museum of Contemporary Art of Zulia, in Maracaibo, Venezuela. The international forum on free knowledge offers an oppertunity to present the nessisity of freedom in learning and community development. My presentation will be on the application of freedom in telecommunications, about GNU Bayonne 2, and how free software and VOIP can liberate users from the bounds of traditional telephone carriers and proprietary solutions. Further information about the International World Forum on Free Knowledge can be found by visiting http://conexionsocial.org.ve/foromundial
GNU Telephony is happy to announce the availability of the first 1.0 release candidate for GNU Bayonne 2. GNU Bayonne 2, is a telephony application server which allows small and large businesses, and commercial telephone carriers, to create, deploy, and manage their own interactive voice response applications, both on wired and VOIP telephone networks, using free software.
GNU Bayonne 2 1.0 is composed of a subset of those services and features found in the recently introduced, and very rapidly advancing GNU Bayonne 2 development effort. Features were chosen for introduction in this release candidate that were already stable and effective for production use and supportable under GNU/Linux and other platforms.... read more
Today I released Bayonne 2 0.9.1 with GNU Troll service bindings. This release was meant to introduce GNU Troll voip-pstn gateway services binding operating under GNU Bayonne 2. It was also meant to enable others to become familiar with how troll services operate, to offer proof of concept for the services binding plugin model, and to initiate further development of Troll services. Basic incoming call handling should work under troll at this point although much work remains to be completed in Troll. Troll services coexist with Bayonne 2 scripting, and either Troll or scripting server mode may be selected at runtime.... read more
I will be speaking at ClueCon August 4th in Chicago, Illinios. I will be demonstrating GNU Bayonne 2 live using SIP and perhaps IAX if work on that driver is completed on time. I will also be talking about how GNU Bayonne 2 was developed, the fundimentals of Bayonne architecture, how one develops applications for GNU Bayonne 2, and what my immediate plans are for its further development this year. There should be many other projects attending ClueCon as well, so I think it should be a very interesting event. I will post a link to the slides for my presentation early in July.
I have made available this morning a second generation Bayonne telephony server for immediate testing and further development. This new server offers support both for wired and protocol stack based telephony drivers, including initial support for SIP and H323. Bayonne 2 uses a simplified driver model and exposes core functionality both through an interface library and a model script driven voice application server. This is an initial release to allow full public testing and development of Bayonne 2 to officially begin, and as such has a limited functional range which will be expanded upon over the summer months.... read more
I have consolidated GNU Bayonne, GNU Common C++, and many related packages under GNU Telephony (http://www.gnutelephony.org). The purpose of this action was to consolidate management of Bayonne and other free software packages I separately maintain. This action was also taken to better support existing Bayonne users, to initiate development of a new second generation Bayonne telephony application server, and to introduce a number of new packages, starting with a pstn/sip gateway server.