Einfache Projektliste Software-Karte

Internet-Telefon
166 Projekte im Ergebnis
Letztes Update: 2010-06-12 08:30

trixbox

trixbox CEは、インストールが容易なAsterrisk PBXベースのVOIP電話システムです。trixboxは、自宅やオフィスで使用する目的で設計されています。trixbox CEにはCentOS Linux、Mysql、ビジネス品質の電話システムに必要な各種ツールが含まれています。

Letztes Update: 2013-05-26 01:46

RemoteTracker

これは、それが紛失したり盗まれたりした場合、お使いのデバイスを追跡する SIM カードの変更検出機能付き盗難防止ソフトウェアです。任意の電話から送信されるフォーマット SMS をキャッチを動作し、戻って有用な情報を送る。他の人が目標、あなたの想像力を使用する !

(Machine Translation)
Letztes Update: 2014-03-17 15:36

Yet Another Telephony Engine

Yate is a next-generation telephony engine. While currently focused on Voice over Internet Protocol (VoIP) and PSTN, its power lies in its ability to be easily extended. It supports SIP, H.323, IAX, MGCP, Jingle, Jabber, E1, T1, analogic, robbed bit, ISDN PRI, BRI, and SS7. YateClient is an Instant Messenger and Voice application for Jabber/Jingle IM, SIP, H.323, and IAX VoIP protocols.

Letztes Update: 2019-06-14 19:04

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: 2010-02-02 11:26

Asterisk

Asterisk is a hybrid TDM and packet voice PBX (Private Branch eXchange) and IVR platform with ACD functionality. It acts as middleware between the Internet (IAX, SIP, MGCP, Skinny, H.323), telephony channels (like Zaptel, T1, PRI, E1, FXO, FXS, VoIP, VoFR, ISDN, modems, Internet Phone Jack, etc.), and applications (like voice-mail, conferencing, directories, MP3 players, intercoms, etc.). It has many advanced features such as a codec translation API. The base distribution includes several channel backends, as well as applications. However, the beauty of Asterisk is its ability to be extended using its APIs, dynamic module loader, and AGI scripting interface. End users can even write their own applications that run on the system in C or any scripting language of their choice.

Letztes Update: 2011-12-26 14:04

linphone

Linphone is an audio and video Internet phone with GTK+ and console interfaces. It uses the SIP protocol, and is compatible with most SIP clients and gateways. It can use various audio and video codecs such as Speex, GSM, G711, G722, ilbc, amr, Theora, H263-1998, MPEG4, H264, VP8, and snow.

Letztes Update: 2014-01-14 22:32

SFLphone

SFLphone is an SIP/IAX2 compatible softphone. The goal is to create a robust enterprise-class desktop phone. While it can serve home users very well, it is designed for intensive corporate use.

Letztes Update: 2007-01-08 17:05

bayonne

Bayonne is the telephony server of the GNU project. It offers a script-driven threaded multi-line state event telephony service on GNU/Linux, xBSD, and Microsoft Windows for building voice response systems, and uses telephony plugins for runtime driver configuration. It also features "TGI" for making Perl applications "telephony aware". It may be used to build telephony-based system administration, home automation, automated attendant, v-commerce, and voice messaging systems.

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: 2008-07-24 11:29

Speex

Speex is a patent-free compression format designed especially for speech. It is specialized for voice communications at low bit-rates in the 2-45 kbps range. Possible applications include Voice over IP (VoIP), Internet audio streaming, audio books, and archiving of speech data (e.g. voice mail).

Letztes Update: 2009-03-25 07:41

FAAC

The FAAC project includes the AAC encoder FAAC and decoder FAAD2. It supports several MPEG-4 object types (LC, Main, LTP, HE AAC, PS) and file formats (ADTS AAC, raw AAC, MP4), multichannel and gapless en/decoding as well as MP4 metadata tags. The codecs are compatible with standard-compliant audio applications using one or more of these profiles.

Letztes Update: 2019-09-09 06:18

Mumble

Mumbleはゲーマー向けのボイスチャットソフトウェアです。遅延が少なく、高品質な通話が可能なのが特徴です。多人数で同時に会話が可能で、ゲームと連携しての使用や、ゲーム画面上に操作ウィンドウをオーバーレイ表示する、といった設定が可能です。

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)

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.