1VideoConverence is Web2.0 audio-video conference call software for Asterisk with support for Web, phone, MSN, Skype, Yahoo, and Jabber clients. This VoIP and VVoIP conferencing app for business, government, education, and health care is based on C#, WinFX, XAML, and .NET 3.0.
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Mediastreamer is a portable C library that allows you to create and run audio and video streams. It is designed for any kind of voice over IP applications. It features RTP connectivity, audio codecs (Speex, iLBC, G711, GSM, and AMR), video codecs (MPEG4, H263, H264, and Theora), sound card I/O, wav file streaming, webcam video capture, echo-cancellation, conferencing, parametric equalization, and various other utilities. It has a modular design that makes it extensible through plugins. This is the media-streaming component of linphone, a GPL SIP video phone.
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Real time communication software built to provide face-to-face advantages to remote gamers.
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QuteCom (formerly WengoPhone) is a multi-platform
VoIP client. The GUI is Qt-based, and the
Video-over-IP engine is based on the eXosip, oSIP,
oRTP, ffmpeg, and libgaim projects. The eXosip
module is extended by a phApi module, which
implements a high-level, easy-to-use call control
API. It supports PC-to-PC voice, video, and chat.
One can use a standard SIP service provider such
as Wengo to be assigned an incoming number, make
calls to PSTN/cell phones, get voice mail, and
more. In addition to SIP/SIMPLE QuteCom provides
IM functionality using libgaim, so it is
compatible with all protocols supported by libgaim.
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isdngw is an H.323 to ISDN gateway using the OpenH323 and PWlib libraries, and the ISDN4Linux subsystem. It is based on isdn2h323.
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The freePBX (formerly Asterisk Management Portal)
project brings together best-of-breed applications
to produce a standardized implementation of
Asterisk, complete with Web-based administrative
interface.
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restund is a modular STUN/TURN server that is designed around the principle of a lightweight core and server modules that extend its functionality. Both UDP and TCP are supported, along with IPv6 and IPv4. Supported modules include STUN, TURN, MySQL database, syslog, and status monitoring.
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YaPS (Yet anothewr Paging Software) is primarily designed to send message to pager devices (including cellular phones) which are able to receive SMS. It is a simple command-line driven program with a global and a personal configuration file, but the protocol implementation is kept as a seperate library. YAPS can transfer Pager Messages with UCP, TAP, and Script Protocol.
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GYach Enhanced is a feature-rich, improved version
of the original Gyach. It is the first Yahoo!
client for Linux with voice chat capabilities. It
offers almost all of the features you would expect
to find in the official Windows Yahoo! client. The
program offers support for chat, conferences,
buddy lists, and My Yahoo content. In addition,
Gyach Enhanced offers many features not available
in the official Yahoo! clients for Windows, Mac,
and Linux. Webcam support is under development and
planned for the future. Unlike the original Gyach,
GYach Enhanced is designed for Linux only.
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python-gammu is a set of Python bindings for
Gammu, the GSM mobile manager for various cellular
phones. It can work with any phone that Gammu
supports, including many models by Nokia, Siemens,
and Alcatel. python-gammu is now part of the Gammu project and it does not have separate releases anymore.
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chan-sccp-b is an extension of the original chan-sccp implementation for the Asterisk soft-PBX. It lets you hook up a Cisco/SCCP Phone to your Asterisk server using the SCCP protocol, which works a lot better than the SIP firmware on the same phone. It provides full phone functionality instead of just a simple SIP channel provider. It offers functionality like shared lines, hotline functionality, guest login, dynamic speeddials, private line automatic ring-down (PLAR), personal softkey configurations, Dundi support, SCCP extended dialplan functions, manager support, and custom device state buttons.
Sipp is a performance testing tool for the SIP protocol. It includes a few basic SipStone user agent scenarios (UAC & UAS) and establishes and releases multiple calls with the INVITE and BYE methods. It also reads XML scenario files describing any performance testing configuration. It features the dynamic display of statistics about running tests, periodic CSV statistics dumps, TCP, UDP, or TLS over IPv4 or IPv6 over multiple sockets or multiplexed with retransmission management, regular expressions and variables in scenario files, conditional branching, and dynamically-adjustable call rates. RTP play (voice, video, and RFC2833 DTMFs) is also supported.
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Julius is a high-performance large vocabulary
continuous speech recognition (LVCSR) engine for
speech-related research and development. You can
construct your own speech recognition system, but
you need a separate English acoustic model and
language model or grammar file.
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partysip is a modular SIP proxy server. It can be used as a registrar and a stateful SIP server. New capabilities can be added with plugins.
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The GnuGk ACD does automatic call distribution for the Gnu Gatekeeper. It is the first step to a VOIP call-center solution based on GnuGk. Incoming calls are routed to agents based on different distribution algorithms.
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