Simple Project List Software Map

83 projects in result set
最后更新: 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.

最后更新: 2015-01-24 05:08

NoiseGator (Noise Gate)

軽量ノイズ ゲート アプリケーションにオーディオ入力をオーディオ出力を介してオーディオのルートです。リアルタイム オーディオ レベルは、分析し、オーディオ バイパスとして通常平均レベルがしきい値を上回る場合。しかし平均レベルがしきい値を下回る場合、ゲートは閉じてし、オーディオをカットします。仮想オーディオ ケーブルを使用するとサウンド input(microphone) をいずれかのノイズ ゲートとして機能したり output(speakers) に聞こえます。もともと誰もが話していたときにバック グラウンド ノイズをカットする Skype 用に設計された、それはあなた自身のマイクからの騒音のゲートまたはあなたのスピーカーを通してあなたのマイクを再生する使用できます。要件: - これを実行する Java 6 またはそれ以降が必要です。-仮想オーディオ ケーブル (または多くのポートを持つ 2 番目のサウンド カードまたはサウンド カードと共に実質の 1) VOIPs で使用するために必要です。Mac ユーザーは !SoundFlower を使用することができます、Windows ユーザーが VAC(paid) または声のチェンジャー ソフトウェアに付属している無料のものを使用できます。

(Machine Translation)
最后更新: 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。

最后更新: 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.

最后更新: 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.

(Machine Translation)
最后更新: 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.

最后更新: 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.

最后更新: 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.

最后更新: 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.

最后更新: 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.

最后更新: 2020-09-02 20:15

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。

最后更新: 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.

(Machine Translation)
最后更新: 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.

最后更新: 2009-10-16 22:23

Imptalk

Real time communication software built to provide face-to-face advantages to remote gamers.

(Machine Translation)
最后更新: 2012-04-22 12:29

restund

restund is a modular STUN/TURN server that is designed around the principle of a lightweight core and server modules that extend its functionality. Both UDP and TCP are supported, along with IPv6 and IPv4. Supported modules include STUN, TURN, MySQL database, syslog, and status monitoring.