Einfache Projektliste Software-Karte

Internet-Telefon
166 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: 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: 2007-03-03 14:11

1VideoConference

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.

Letztes Update: 2009-03-23 10:21

SipExchange

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.

(Machine Translation)
Letztes Update: 2009-05-22 11:12

IPP Codecs

G.728, G.729, G.723.1, G.722.2 GSM-FR, GSM-AMR, H.261, H.263, H.264そしてMPEG4 part 2を含むOPAL/OpenH323ライブラリのためのIntel Integrated Performace Primitivesオーディオ/ビデオコーデックプラグイン

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: 2012-01-08 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: 2011-12-26 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: 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: 2009-05-13 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: 2012-02-14 05:20

The Milkfish Embedded SIP Router

Milkfishは、組込み用のコミュニケーション・ソフトウェア・プロジェクトです。主な目的は、1つのパブリックIPアドレスを使用してWAN接続を共有し、LAN内の複数のSIP電話機の実用的なセットアップを可能にする安価なハードウェアの基本的なNATトラバーサル・ソリューションを提供することです。

Letztes Update: 2017-09-24 18:21

GNU Gatekeeper (GnuGk)

GNU ゲートキーパー (!GnuGk) における注目 H.323 ゲートキーパー GPL ライセンスの下でです。それ VoIP やビデオ会議をサポートする Radius とデータベースのサポートが含まれています、多くルーティング関数を呼び出します。

(Machine Translation)
Letztes Update: 2009-01-19 18:28

PJSIP and PJMEDIA

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.

Letztes Update: 2005-04-12 09:58

Cutlass

Cutlass is a cross-platform system for secure peer to peer communication, oriented towards small groups. It provides VoIP, IM and file transfer. A major design goal is to be easy to use, even by non-security conscious users.

Letztes Update: 2012-01-26 20:05

STUN Client and Server

このプロジェクトは、Windows、Linux及びSolarisで動作するシンプルなSTUNサーバとクライアントを実装します。STUNプロトコル(Simple Traversal of UDP through NATs)は、IETF RFC3489で定義されています(RFC3489はhttp://www.ietf.org/rfc/rfc3489.txtから入手できます)