Simple Project List 軟體列表

82 projects in result set
最後更新: 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.

最後更新: 2014-03-19 01:35

Zentyal

Zentyal Server aims at offering small and medium businesses (SMBs) a native drop-in replacement for Windows Small Business Server and Microsoft Exchange Server which can be set up in less than 30 minutes and is both easy to use and affordable.

最後更新: 2012-09-01 23:21

Hanasu

JXTA を使用して P2P のメッセンジャー。安全な RSA/AES のエンドツー エンドの暗号化と VoIP 機能を提供しています。

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Development Status: 3 - Alpha
目標受眾: End Users/Desktop
自然語言: English, German
程式語言: Java
使用者介面: Java Swing
最後更新: 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。

最後更新: 2014-01-23 16:33

baresip

baresip is a bare-bones SIP user agent. It supports SIP, SDP, RTP/RTCP, and STUN/TURN/ICE, and IPv4 and IPv6, and is RFC-compliant and has portable C89 and C99 source code. A modular plugin architecture provides stdio, cons, and evdev user interfaces, celt, g711, g722, gsm, ilbc, l16, and speex audio codecs, alsa, coreaudio, gst, portaudio, oss, winwav, and mda audio drivers, speex_pp, speex_aec, speex_resamp, and sndfile audio filters, the avcodec video codec, avformat, quicktime, qtcapture, v4l, and v4l2 video sources, sdl, opengl, and x11 video display drivers, and srtp media encoding.

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最後更新: 2014-04-12 12:27

libre

libre is a generic library for real-time communications with asynchronous I/O support. It is written in portable POSIX source code that conforms to the ANSI C89 and ISO C99 standards. It is robust and fast, with a low memory footprint. It also features RFC compliance and support for IPv4 and IPv6. Protocol implementations include SIP, SDP, RTP/RTCP, BFCP, DNS, STUN/TURN/ICE, HTTP, and WebSockets.

最後更新: 2019-03-08 00:16

OpenSIPS/OpenSER-a versatile SIP Server

OpenSIPS(旧OpenSER)は、GPLライセンスの元に実装している多機能なSIPサーバで、プロフェッショナルなSIPサーバプラットフォームで使用されるハイレベルな技術的ソリューション(パフォーマンス、セキュリティ及び品質)を提供することを目的としています。

Database Environment: Other API
Development Status: 6 - Mature
自然語言: English
程式語言: C
使用者介面: No Input/Output (Daemon)
最後更新: 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.

最後更新: 2013-09-10 19:08

Discretio

Discretio is an application (and service) for secure VoIP on smartphones. It allows users to establish calls over the Internet while using state of the art encryption technologies to avoid eavesdropping.

最後更新: 2010-12-13 14:34

SEMS

SEMS is a media and application server for SIP based VoIP services. It shows good performance doing basic services like announcements and conference for combination with external application servers. Thanks to its easy-to-use and flexible application development framework and back-to-back user agent support, application logic and media serving can be combined in the same process. Basic applications like announcement, pre-call announcement, RBT, conference, voicemail, mailbox, and lots of example applications are available. Scripting can be done in Python and a simple state machine description language. Support All commonly used free codecs (including g711, gsm, iLBC, speex, adpcm, and l16) are supported. Other features include wideband, ZRTP encryption, a SIP registrar client, an XMLRPC server/client, and a DIAMETER client.

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最後更新: 2012-03-18 01:52

Speech synthesis for asterisk

Speech synthesis for asterisk is an Asterisk AGI script that uses Google Translate to convert text to speech and play it back to the user. It supports a variety of different languages, local caching of voice data, and a choice between 8 kHz or 16 kHz sample rates to provide the best possible sound quality along with the use of wideband codecs.

最後更新: 2011-09-20 08:02

AsterClick

AsterClick is a system for developing with Asterisk AMI and HTML5 WebSockets. It is composed of two parts: a server-side middleware and a client-side JavaScript class. The server-side middleware mediates between Asterisk AMI and multiple HTML5 browsers connected via WebSockets. The JavaScript class manages the WebSockets connection and provides methods like addEventListener() and removeEventListener() that take AMI events as parameters. AsterClick does away with browsers polling servers by exploiting the persistent nature of HTML5 WebSocket connections. The communications protocol between client and server is based on XML. Commands can be sent via the JavaScript class using XML strings, XML objects, or JSON objects. A client can connect to multiple Asterisk servers at the same time. The server-side component of AsterClick has hooks for both custom AsterClick commands and server side plugins and related events that all share the same XML stream.

最後更新: 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。

最後更新: 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.

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最後更新: 2012-02-01 22:22

Asterisk speech recognition

Asterisk speech recognition is an AGI script that makes use of the Google voice recognition engine in order to render speech to text and return it back to the dialplan as an asterisk channel variable.

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