Simple Project List 軟體列表

166 projects in result set
最後更新: 2007-04-27 08:34

Sipp

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.

(Machine Translation)
最後更新: 2004-12-21 10:19

asterisk-oh323

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.

(Machine Translation)
最後更新: 2007-03-03 14:11

1VideoConference

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.

最後更新: 2009-05-13 14:22

Gnu Gatekeeper ACD

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.

(Machine Translation)
最後更新: 2002-03-03 09:46

jAugment

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.

(Machine Translation)
最後更新: 2007-08-28 14:26

freePBX

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
interface.

(Machine Translation)
最後更新: 2005-04-12 09:58

Cutlass

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.

最後更新: 2009-05-22 11:12

IPP Codecs

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

最後更新: 2011-02-15 13:56

CELT audio codec

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).

(Machine Translation)
最後更新: 2009-01-19 18:28

PJSIP and PJMEDIA

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.

最後更新: 2019-06-14 19:04

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: 2009-05-24 07:00

sip-redirect

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.

最後更新: 2011-12-26 22:34

mediastreamer

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.

最後更新: 2007-07-03 11:49

Asterisk Soap API

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.

(Machine Translation)
最後更新: 2012-12-30 03:36

CQiNet

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。

(Machine Translation)