IHU is a Voice over IP (VoIP) application that creates an audio stream between two computers easily and with the minimal network traffic. It works without an intervening server, can use either UDP or TCP for transport, can encrypt the data stream, and can work without a GUI. It uses the Speex codec to help reduce bandwidth consumption. It supports ALSA and JACK for audio service.
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GYach Enhanced is a feature-rich, improved version
of the original Gyach. It is the first Yahoo!
client for Linux with voice chat capabilities. It
offers almost all of the features you would expect
to find in the official Windows Yahoo! client. The
program offers support for chat, conferences,
buddy lists, and My Yahoo content. In addition,
Gyach Enhanced offers many features not available
in the official Yahoo! clients for Windows, Mac,
and Linux. Webcam support is under development and
planned for the future. Unlike the original Gyach,
GYach Enhanced is designed for Linux only.
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Sipp is a performance testing tool for the SIP protocol. It includes a few basic SipStone user agent scenarios (UAC & UAS) and establishes and releases multiple calls with the INVITE and BYE methods. It also reads XML scenario files describing any performance testing configuration. It features the dynamic display of statistics about running tests, periodic CSV statistics dumps, TCP, UDP, or TLS over IPv4 or IPv6 over multiple sockets or multiplexed with retransmission management, regular expressions and variables in scenario files, conditional branching, and dynamically-adjustable call rates. RTP play (voice, video, and RFC2833 DTMFs) is also supported.
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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.
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Star Asterisk API is a high performance API that
connects to the manager interface of Asterisk or
to AstManProxy. It is easy to use,
object-oriented, and easy to extend to suit your
particular requirements.
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StreamBOX-LiveCD is a KNOPPIX-based boot CD which is specially designed to stream MP3s. It also includes some programs to stream in the Ogg Vorbis format, to cut audio, to view video and video streams, and to perform other similar tasks. It is intended to be used at smaller radio stations.
Related ProjectsDarik's Boot and Nuke, iReport-Designer for JasperReports, Media Player Classic - Home Cinema, CaptureStream, FreeStyleWiki |
minisip is a SIP VoIP soft phone that implements
additional security features such as mutual
authentication, encryption and integrity of on-going
calls, and encryption of the signaling (SIP over TLS).
These security features use work-in-progress IETF
standards (SRTP and MIKEY).
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GSUtil is a utility for saving, restoring, and
rebooting GrandStream BudgeTone 100 and GX2000
VOIP phones, amongst others. It's written in Perl
and was developed using standard Perl modules, so
no extra modules need to be installed.
Related ProjectsPeaZip, Clonezilla, OpenTween, PukiWiki, MeCab |
VoiceBuntu (formerly Ubunterisk) is an Ubuntu-based live CD that uses Asterisk and VoiceOne to provide VoIP service without any system installation process. VoiceOne
is a Web-based GUI for the Asterisk PBX. Ubunterisk can be used as a phone client as well as a PBX server. Ubunterisk can be administered either remotely or by accessing its local GUI. A capser-rw filesystem is used to store the system's
data persistently.
Related ProjectsRealTerm: Serial/TCP Terminal, FreeStyleWiki, PukiWiki, iReport-Designer for JasperReports, SmillaEnlarger |
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)
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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。
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セッション開始プロトコル(SIP)は、インターネット上での電話サービスのために幅広く用いられています。CafeSipは、Javaを用いた独自のSIPサービス及びアプリケーションを作成するためのオープンソース・ツールとアプリケーションのパッケージを提供します。
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OSP ツールキットは ETSI OSP VoIP ピアリング プロトコル (ETSI TS 101 321) のクライアント側の実装です。OSP ツールキット プロジェクトが 1998 年に開始され、コードは多くの商用およびオープン ソースの VoIP 製品に組み込まれています。
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VMuktiは、Asterisk/ Yateが有効にされたp2pの動画用IP Communications Suiteです。 これらのサーバレスなブロードバンド対応のオープンソースプラットフォームは、遠隔会議やコールセンター用の独占的ソフトウェアに比べて90%の運用コストを節約します。
Related ProjectsMPC-BE, Dumpper v.60.3, Skype4Java (a.k.a Skype API for Java), PeerCast-Users@jp, MeCab |
アスタリスク + a2billing ラズベリー pi シングル ボード コンピューターのための画像します。
Related ProjectsMPC-BE, Skype4Java (a.k.a Skype API for Java), OneSQLiteAdmin, IAX2 API For Java, MeCab |
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