04-10-2012, 01:21 PM
GNU Telephony
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GNU SIP Witch Initially Released
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.
GNU SIP Witch can be used together with and to interconnect common free software and IETF standard compliant voice and video capable SIP based desktop softphone applications such as Ekiga, Twinkle, Linphone, and OpenWango, as well as most standard compliant SIP telephone devices. When used together with a standard compliant SIP media application server such as GNU Bayonne, GNU SIP Witch will be able to offer users and remote callers access voice messaging and media application services. In the future GNU SIP Witch will offer STUN services and optional packet forwarding to facilitate interconnection when behind NAT's and firewalls in IPV4 networks. GNU SIP Witch also supports IPV6.
GNU SIP Witch is not a SIP proxy, a multi-protocol telephone server, or a IP-PBX, and does not try to address the same things like asterisk, yate, or GNU bayonne2, all of which make use of direct media processing where media connections and streaming are at least initially established between each endpoint and the IP-PBX server itself. Direct media processing adds additional latency, and by introducing a central point where all media streams can be processed, compromises both the privacy of calls and security of encrypted telephone sessions.
Instead, GNU SIP Witch focuses on doing just one thing as a pure SIP call server, and will try to do that one thing very well. My goal is to focus on achieving a network scalable telephone architecture built around the SIP protocol that can be deeply embedded, which can support secure calling nodes, that can integrate well with other SIP based/standards compliant components, and, by not engaging in media processing, that is not license encumbered by the use of "mandated" patent encumbered and proprietary telephony media codecs.
GNU SIP Witch Announced
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.
GNU SIP Witch is designed to support network scaling of telephony services, rather than the heavily compute-bound solutions we find in use today. This means a call node has a local authentication/registration database, and this will be mirrored, so that any active call node in a cluster will be able to accept and service a call. This allows for the possibility of live failover support in the future as well.
GNU SIP Witch is not a SIP "router", and does not try to address the same things as a project like iptel "Ser". GNU SIP Witch is being designed to create on-premise SIP telephone systems, telecenter servers, and Internet hosted SIP telephone systems. One important feature will include use of URI routing to support direct peer to peer calls between service domains over the public internet without needing mediation of an intermediary "service provider" so that people can publish and call sip: uri's unconstrained. GNU SIP Witch is about freedom to communicate and the removal of artifical barriers and constraints whether imposed by monopoly service providers or by governments.
GNU Telephony Secure Calling Announced
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.
The Twinkle softphone package, starting with release 0.9, is the first complete free software package to make use of Secure call features offered in the GNU RTP Stack and may well be the first stand-alone zrtp compatible VOIP client available anywhere.
The GNU RTP stack can be used to develop secure communications for GNU/Linux hosted applications. The stack may also be used to develop application on various BSD systems including Mac OS/X, on Microsoft Windows, and even for embedded systems. We have tested and built the GNU RTP Stack with Handhelds Open Embedded build environment, and we look forward to implementing a Twinkle based secure calling solutions on Linux kernel powered cell phones in the future.
Bayonne; Telephony, XML, and webservices
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.
Bayonne2 introduces BayonneXML as a service binding, which is a Bayonne2 feature that allows services to be derived as plugins from the core engine library and Bayonne server framework with a minimum of additional coding effort, while using the same configuration files, management services, telephony drivers, etc. Bayonne2 service bindings will be created in the future for other XML dialects such as for Daisy to enable electronic talking books, and for other purposes, including dedicated telephony switch integrations and to offer other service profiles such as for building TMS application services platforms and ACD switches.