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.
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.
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
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.
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
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
Cutlass is a cross-platform system for secure peer to peer communication, oriented towards small groups. It provides VoIP, IM and file transfer. A major design goal is to be easy to use, even by non-security conscious users.
相關的專案Media Player Classic - Home Cinema, Dumpper v.60.3, Tween, PukiWiki, SmillaEnlarger
G.728, G.729, G.723.1, G.722.2 GSM-FR, GSM-AMR, H.261, H.263, H.264そしてMPEG4 part 2を含むOPAL/OpenH323ライブラリのためのIntel Integrated Performace Primitivesオーディオ／ビデオコーデックプラグイン
相關的專案ip_phone, Media Player Classic - Home Cinema, Skype4Java (a.k.a Skype API for Java), MPC-BE, ffdshow
CELT (Constrained Energy Lapped Transform) is an
ultra-low delay audio codec designed for realtime
transmission of high quality speech and audio.
This is meant to close the gap between traditional
speech codecs (such as Speex) and traditional
audio codecs (such as Vorbis).
pjsip is a multimedia communication library based on the SIP protocol. It is integrated with a rich media and a NAT traversal library supporting the ICE protocol. It is very portable and has a small footprint for embedded use.
相關的專案PukiWiki, FreeStyleWiki, iReport-Designer for JasperReports, Skype4Java (a.k.a Skype API for Java), OpenTween
相關的專案LetterFix for Mac OS X Mail.app, Dumpper v.60.3, PortableApps.com: Portable Software/USB, SmillaEnlarger, postLDAPadmin
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.
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
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
アマチュア無線のオペレーター向け IP （VoIP） ソフトウェアを介して音声します。主な目的は、インターネット リンクを使用するユーザーのリモート アクセスを提供する二次目的でハム リピータです。IRLP および !EchoLink 互換性の目標です。
相關的專案Media Player Classic - Home Cinema, Skype4Java (a.k.a Skype API for Java), Win32 Disk Imager, ip_phone, MPC-BE