Simple Project List 軟體列表

166 projects in result set
最後更新: 2005-05-15 13:56

USB Phone Connector

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.

(Machine Translation)
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: 2011-12-26 22:34


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: 2012-06-25 09:52

CafeSip - Look what Java and SIP can do


最後更新: 2009-07-08 15:32


SuperShaper-SOHO is a traffic shaping setup for DSL connections which prioritizes VoIP and interactive traffic and makes sure P2P traffic doesn't saturate your uplink.

(Machine Translation)
最後更新: 2013-09-08 20:22

ZRTP Protocol Library

ZRTP Protocol Library is an implementation of Phil Zimmermann's ZRTP protocol, created based on and interoperable with Zfone beta 2. Combined with the GNU RTP Stack (ccrtp), this offers the ability to create communication services that natively support the ZRTP protocol.

最後更新: 2005-04-12 09:58


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: 2013-10-08 21:41


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。

最後更新: 2016-10-05 17:22


!VoiceOne インストールして完全に、使いやすい web サーバーを構築して通信様々 なプラグインを追加するためのフレームワークとなる GUI でアスタリスク 1.8 に基づく pbx プラットフォームを構成することができます。

(Machine Translation)
最後更新: 2007-05-01 23:43

AstBill Billing, Routing Management Asterisk VOIP

AstBill is Web-based billing, routing, and
management software for Asterisk and MySQL based
on Drupal. It provides pre- and post-paid billing
services. It completely automates Asterisk billing
from start to finish. Key benefits are the central
Web-based installation, credit control on outgoing
calls, and the call routing module.

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


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: 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: 2007-08-28 14:26


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

(Machine Translation)
最後更新: 2011-05-22 14:01


Sofia-SIP is a SIP user agent library, compliant
with the IETF RFC3261 specification. It can be
used as a building block for SIP client software
for uses such as VoIP, IM, and many other
real-time and person-to-person communication
services. The primary target platform is
GNU/Linux. Sofia-SIP is based on a SIP stack
developed at the Nokia Research Center.

(Machine Translation)
最後更新: 2005-10-17 08:36

Jori's Voice over IP Library

JVOIPLIB is an object-oriented Voice-over-IP (VoIP) library, written in C++. Its purpose is to make it easy to set up and control VoIP sessions. The session's characteristics can be changed during a VoIP call.

(Machine Translation)