ObexTool is a graphical frontend for ObexFTP, which is able to
communicate with mobiles and other devices using the Obex Protocol. The
Siemens S45, S45i, S25, S35, SL45i, SL45, M50, C55, S55, C65, C65V,
Ericsson R320, T68i, Sony/Ericsson T300, Ki700, and Nokia 6230 and 6670
have been reported to work with obexftp, though it should also work with
other phones.
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pjsip is a multimedia communication library based on the SIP protocol. It is integrated with a rich media and a NAT traversal library supporting the ICE protocol. It is very portable and has a small footprint for embedded use.
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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Real time communication software built to provide face-to-face advantages to remote gamers.
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ivam (ISDN Voice Box Answering Machine) is a
telephony application server system for ISDN and
Linux. It consists of two parts: a C-coded daemon
responsible for call setup, and high-level Python
scripted applications.
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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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Asterisk speech recognition is an AGI script that makes use of the Google voice recognition engine in order to render speech to text and return it back to the dialplan as an asterisk channel variable.
SEMS is a media and application server for SIP based VoIP services. It shows good performance doing basic services like announcements and conference for combination with external application servers. Thanks to its easy-to-use and flexible application development framework and back-to-back user agent support, application logic and media serving can be combined in the same process. Basic applications like announcement, pre-call announcement, RBT, conference, voicemail, mailbox, and lots of example applications are available. Scripting can be done in Python and a simple state machine description language. Support All commonly used free codecs (including g711, gsm, iLBC, speex, adpcm, and l16) are supported. Other features include wideband, ZRTP encryption, a SIP registrar client, an XMLRPC server/client, and a DIAMETER client.
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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Scopserv Communicator is an audio/video Internet phone and instant messenger that supports some of the most popular VoIP and instant messaging protocols such as SIP, Jabber (XMPP), AIM/ICQ, MSN, Yahoo! Messenger, and a whole lot of other useful features. ScopServ Communicator is based on the SIP Communicator softphone.
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SipExchange is a softswitch that provides standard SIP services like location, proxy, and presence. Using the SipExchange application, service providers can offer VoIP telephone services to their subscribers as well as other services based on voice, video, and instant messaging. SipExchange supports many of the standard subscriber features offered by the traditional telephone exchanges and PBXs. In addition, SipExchange supports external call control capabilities which service providers and software developers can use to create new features and services rapidly and plug them into the SipExchange application. SipExchange works with standard SIP phones that adhere to the SIP protocol standards. Its software architecture is flexible, scalable, and easily extensible. It runs as an enterprise application inside the JBoss server and takes advantage of many services that a J2EE server provides. SipExchange provides a portal-based user interface with which system administrators can manage subscribers and features as well as perform other routine operations. From the portal, subscribers can manage their profiles, view the call detail records, and customize the features to which they have subscribed. Service providers can easily add additional content to the portal and customize the look and feel.
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sip-redirect is a tiny SIP redirect server. It supports IPv4 and IPv6, but the IPv6 support is optional. The RFC 3261 was the base for this simple and very configurable implementation. There is neither TCP nor multicast support programmed in.
letterize generates alphabetic mnemonics for a phone number, then
filters them for phonetic plausibility in English.
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Asterisell is a Web application for rating, showing to customers, and billing of Asterisk VoIP calls.
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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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