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)
相關的專案Media Player Classic - Home Cinema, Skype4Java (a.k.a Skype API for Java), Dumpper v.60.3, GLOBALBASE PROJECT, SmillaEnlarger
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.
相關的專案SmillaEnlarger, SharpDevelop-jp, Properties Editor, Darik's Boot and Nuke, DeSmuME
Jiplet Container (Java SIP Servlet) is a servlet-like development and runtime environment for SIP applications. The SIP protocol is widely used for voice services over IP networks. This product enables developers to create server-side SIP applications using a component-based model similar to that envisioned by the J2EE architecture. The Jiplet container runs as a standalone server as well as a JBOSS service.
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.
SipExchange is a softswitch that provides standard SIP services like location, proxy, and presence. Using the SipExchange application, service providers can offer VoIP telephone services to their subscribers as well as other services based on voice, video, and instant messaging. SipExchange supports many of the standard subscriber features offered by the traditional telephone exchanges and PBXs. In addition, SipExchange supports external call control capabilities which service providers and software developers can use to create new features and services rapidly and plug them into the SipExchange application. SipExchange works with standard SIP phones that adhere to the SIP protocol standards. Its software architecture is flexible, scalable, and easily extensible. It runs as an enterprise application inside the JBoss server and takes advantage of many services that a J2EE server provides. SipExchange provides a portal-based user interface with which system administrators can manage subscribers and features as well as perform other routine operations. From the portal, subscribers can manage their profiles, view the call detail records, and customize the features to which they have subscribed. Service providers can easily add additional content to the portal and customize the look and feel.
asterisk-oh323 adds H.323 support to the ASTERISK soft PBX. It does this
by interfacing the OpenH323 library to ASTERISK through a loadable
module. The package provides the channel driver as well as a wrapper in
a shared library form. It is able to initiate and receive calls to and
from H.323 endpoints, and has been successfully tested with the H.323
terminals on the OpenH323 site (ohphone, openphone), GnomeMeeting,
Microsoft NetMeeting, Cisco routers, H.323 Snom phones, and other hardware and software H.323 IP phones.
The GnuGk ACD does automatic call distribution for the Gnu Gatekeeper. It is the first step to a VOIP call-center solution based on GnuGk. Incoming calls are routed to agents based on different distribution algorithms.
The freePBX (formerly Asterisk Management Portal)
project brings together best-of-breed applications
to produce a standardized implementation of
Asterisk, complete with Web-based administrative
相關的專案Hinemos, SmillaEnlarger, MPC-BE, Media Player Classic - Home Cinema, 再生龍
sip-redirect is a tiny SIP redirect server. It supports IPv4 and IPv6, but the IPv6 support is optional. The RFC 3261 was the base for this simple and very configurable implementation. There is neither TCP nor multicast support programmed in.
jAugment is a JINI-based, network-aware framework and set of initial applications for wearable computers and other computers with uncommon input/output-devices. It supports user interfaces from text-only to 3D.
相關的專案Open Jane, Media Player Classic - Home Cinema, Properties Editor, ギコナビ, Dumpper v.60.3
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.
相關的專案Darik's Boot and Nuke, OpenOffice.org 独自ビルドプロジェクト, iReport-Designer for JasperReports, OpenTween, FreeStyleWiki
1VideoConverence is Web2.0 audio-video conference call software for Asterisk with support for Web, phone, MSN, Skype, Yahoo, and Jabber clients. This VoIP and VVoIP conferencing app for business, government, education, and health care is based on C#, WinFX, XAML, and .NET 3.0.
相關的專案OpenOffice.org 独自ビルドプロジェクト, Dumpper v.60.3, Tween, PukiWiki, SmillaEnlarger
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
相關的專案Nucleus日本語版, RealTerm: Serial/TCP Terminal, PukiWiki, DeSmuME, Darik's Boot and Nuke
USB Phone Connector is a framework for connecting USB "phones" (typically a device incorporating a USB-addressable speaker, microphone, LCD display, and like peripherals in a telephone-like form factor) to a softphone application. Newer versions support Skype as well as the linphone SIP phone.
Mediastreamer is a portable C library that allows you to create and run audio and video streams. It is designed for any kind of voice over IP applications. It features RTP connectivity, audio codecs (Speex, iLBC, G711, GSM, and AMR), video codecs (MPEG4, H263, H264, and Theora), sound card I/O, wav file streaming, webcam video capture, echo-cancellation, conferencing, parametric equalization, and various other utilities. It has a modular design that makes it extensible through plugins. This is the media-streaming component of linphone, a GPL SIP video phone.
相關的專案Amateras, PukiWiki, Dumpper v.60.3, NNDD - ニコ動専用ブラウザ, Properties Editor