Einfache Projektliste Software-Karte

Internet-Telefon
166 Projekte im Ergebnis
Letztes Update: 2013-10-08 21:41

sipp

Sippは、SIPプロトコルのためのパフォーマンステストツールです。 その主な特徴は、ベーシックなSIPStoneシナリオ、TCP/UDPトランスポート、カスタマイズ(XMLベース)可能なシナリオ、コールレートの動的アジャストメント、包括的なリアルタイム統計などです。

Letztes Update: 2011-09-10 01:10

VoiceOne

VoiceOne is a Linux distribution that gives you the ability to install a PBX platform with an easy to use Web-based GUI. It also provides a framework for building a communication server adding various plugins. Its main features are Asterisk 1.8 with realtime configuration with MySQL, a Ubuntu 10.04 base, and support for both hard disk and Compact Flash card installation.

Letztes Update: 2009-01-13 15:15

Kannel WAP and SMS Gateway

Kannel is a WAP gateway. It attempts to provide this essential part of the WAP infrastructure freely to everyone so the market potential for WAP services, both from wireless operators and specialized service providers, will be realized as efficiently as possible. It also works as an SMS gateway for GSM networks. Almost all GSM phones can use it to send and receive SMS messages, so this is a way to serve many more clients than just those using a WAP phone. Kannel was among the first WAP gateways to be certified as WAP 1.1 compliant.

Letztes Update: 2006-01-11 11:05

SIP Express Router

SER or SIP Express Router is a very fast and flexible SIP (RFC3261) server. It can handle thousands of calls per second on low-budget hadware. A C shell-like scripting language provides full control over the server's behavior. Its modular architecture allows only required functionality to be loaded. The following modules are available: accounting, digest authentication, CPL scripts, ENUM support, instant messaging, MySQL support, PostgreSQL support, a presence agent, Radius authentication and accounting, Diameter authentication, record routing, an SMS gateway, a Jabber gateway, NAT traversal support transaction module, a registrar, and user location.

Letztes Update: 2006-01-28 20:37

sipsak

sipsak is a command line tool for performing
various tests on Session Initiation Protocol
(SIP) applications and devices. It can make several
different tests, send the contents of a file, and
interpret and react on the responses. It supports (de-) registration with given contact URIs and digest authentication.

(Machine Translation)
Letztes Update: 2002-03-03 09:46

jAugment

jAugment is a JINI-based, network-aware framework and set of initial applications for wearable computers and other computers with uncommon input/output-devices. It supports user interfaces from text-only to 3D.

(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: 2004-12-21 10:19

asterisk-oh323

asterisk-oh323 adds H.323 support to the ASTERISK soft PBX. It does this
by interfacing the OpenH323 library to ASTERISK through a loadable
module. The package provides the channel driver as well as a wrapper in
a shared library form. It is able to initiate and receive calls to and
from H.323 endpoints, and has been successfully tested with the H.323
terminals on the OpenH323 site (ohphone, openphone), GnomeMeeting,
Microsoft NetMeeting, Cisco routers, H.323 Snom phones, and other hardware and software H.323 IP phones.

(Machine Translation)
Letztes Update: 2012-01-08 00:16

SIPFwd

The SIP forwarding daemon (implemented as a stateless SIP proxy) allows you to seamlessly forward SIP requests to other SIP servers. Its main purpose is to enable users to use their own domain name in SIP URIs without the hassle of having to run a full-blown SIP server (by forwarding SIP requests to third-party SIP servers). Configuration information is stored in an SQLite database, and low resource consumption is a main priority for the project.

Letztes Update: 2011-02-15 13:56

CELT audio codec

CELT (Constrained Energy Lapped Transform) is an
ultra-low delay audio codec designed for realtime
transmission of high quality speech and audio.
This is meant to close the gap between traditional
speech codecs (such as Speex) and traditional
audio codecs (such as Vorbis).

(Machine Translation)
Letztes Update: 2010-09-09 09:59

StarPound

音声およびデータ アプリケーションを収束、オープン ソースのソフトウェア プロジェクトを提供します。!StarPound プラットフォームは、ビジネス プロセスの電源を使用して音声とデータのソリューションを迅速に開発することができますモデリングします。さらに詳しい情報: http://www.starpound.org。

(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: 2009-05-13 14:22

Gnu Gatekeeper ACD

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.

(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: 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)