Simple Project List Software Download Map

166 projects in result set
LastUpdate: 2018-03-20 23:04

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。

LastUpdate: 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.

LastUpdate: 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.

LastUpdate: 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.

LastUpdate: 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.

LastUpdate: 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。

LastUpdate: 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.

LastUpdate: 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).

LastUpdate: 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.

LastUpdate: 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.

LastUpdate: 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)
LastUpdate: 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.

LastUpdate: 2010-07-19 22:06

oRTP

oRTP is a library implementing the Real-time Transport Protocol (RFC3550), written in C. It is easy to use and provides a packet scheduler for sending and receiving packets on time, adaptive jitter compensation, automatic sending of RTCP compound packets, and the RTCP parser API. It works with IPv6.

LastUpdate: 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.

LastUpdate: 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。