Einfache Projektliste Software-Karte

394 Projekte im Ergebnis
Letztes Update: 2007-07-03 11:49

Asterisk Soap API

Asterisk-soap is an extensible SOAP (Web services)
API with the aim to support all administration
features available in the Asterisk VoIP server.
Its primary goal is to provide a framework between
Asterisk and multi-platform frontends.

(Machine Translation)
Letztes Update: 2006-06-08 09:18

Star Asterisk API

Star Asterisk API is a high performance API that
connects to the manager interface of Asterisk or
to AstManProxy. It is easy to use,
object-oriented, and easy to extend to suit your
particular requirements.

(Machine Translation)
Letztes Update: 2011-12-26 22:34

mediastreamer

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.

Letztes Update: 2010-10-19 14:50

QuteCom

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.

(Machine Translation)
Letztes Update: 2009-05-23 00:18

isdngw

isdngw is an H.323 to ISDN gateway using the OpenH323 and PWlib libraries, and the ISDN4Linux subsystem. It is based on isdn2h323.

(Machine Translation)
Letztes Update: 2007-08-28 14:26

freePBX

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.

(Machine Translation)
Letztes Update: 2012-04-22 12:29

restund

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.

Letztes Update: 2001-08-15 15:34

Yet another Pager Software

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.

(Machine Translation)
Letztes Update: 2021-11-04 17:38

Elastix

Elastixは、AsteriskベースのPBXに使いやすいインターフェースを付加するためにベストなツール群を統合化したアプライアンスソフトウェアです。また、オープンソースのテレフォニーのためのベストなソフトウェアパッケージとするために、独自のユーティリティの設定も追加されています。

Letztes Update: 2012-07-18 20:26

EMIPLIB

EMIPLIB is a library to facilitate the development of programs that need to stream several kinds of media over IP. The library consists of several kinds of components that can be linked together in various ways, thereby providing a flexible framework. It also provides some ready-to-use classes for the transmission of audio and video over IP. Streams originating from the same participant can be synchronized.

Letztes Update: 2010-12-22 22:59

ScopServ Communicator

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.

(Machine Translation)
Letztes Update: 2005-04-04 10:11

GYach Enhanced

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.

(Machine Translation)
Letztes Update: 2009-05-24 07:00

sip-redirect

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.

Letztes Update: 2004-02-09 12:36

ISDN Voice Box Answering Machine

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.

(Machine Translation)
Letztes Update: 2007-04-27 08:34

Sipp

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.

(Machine Translation)