Open Source Windows Internet Phone Software

Mumble is an open source, low-latency, high quality voice chat software primarily intended for use while gaming. It includes game linking, so voice from other players comes from the direction of their characters, and has echo cancellation so the sound from your loudspeakers won't be audible to other players.

Elastix is a software-based PBX powered by 3CX and based on Debian. An open-standards solution, Elastix is an easy to install and manage UC system compatible with popular IP phones, gateways and SIP trunks. Elastix is complete with unified communications features such as integrated WebRTC video conferencing, chat, presence and softphones and smartphone clients for Windows, Mac, iOS and Android.

Kiax is a softphone (soft phone, VoIP client) with a simple and comfortable user interface for making VoIP calls to Asterisk PBX. It depends on the iaxclient library to use Asterisk's IAX2 protocol for easy call configuration and audio setup.

A TAPI driver for SIP. SIPTAPI gives you a click2dial feature with any TAPI enabled application (e.g. MS Outlook) and SIP PBX/proxy. This is the "free" version of SIPTAPI. There is also an enhanced commercial version available at www.ipcom.at.

This project has been superseded by OpalVoip (https://sourceforge.net/projects/opalvoip/) and H323Plus (https://sourceforge.net/projects/h323plus/) The OpenH323 project provides full featured, interoperable, Open Source implementation of the ITU H.323 teleconferencing protocol that can be used by personal developers and commercial users without charge.

OfficeSIP Softphone and Messenger are two enterprise VoIP SIP clients written in C# in .NET Framework. The SIP clients make use of Microsoft UCC API SDK, ensuring the highest quality of audio and video. Compatible with Office Communications Server. See also open source, cross-platform: 1) simple messenger Brief Msg at http://briefmsg.org 2) MUVConf is a multi-user video conferencing, see demo video http://youtu.be/YrBU-Aqtvrk, download https://code.google.com/p/muvconf/downloads/list

Peers is a very simple softphone. It's a SIP User-Agent, written in java, it works on windows, linux and mac. It can be used with SIP servers like opensips or asterisk IPBX. It supports G711 codec (PCMU and PCMA) and telephone-events (DTMF).

A lightweight cross platform IP telephony client using the IAX protocol, designed for use with the asterisk open source PBX.

Skype Audio Player allows playing audio files (mp3, wav, wma) during Skype calls to both parties. It is a simple tool without luxurious functions, useful for those who study foreign languages using Skype and need to play sample dialogs or audio tests.

The project has moved! Please find current versions at https://www.gnugk.org/ The GNU Gatekeeper (GnuGk) is a full featured H.323 gatekeeper under GPL license. It supports VoIP and videoconferencing and includes Radius and database support and many call routing functions. The project has moved! Please find current versions at https://www.gnugk.org/

VMukti open source is leading Asterisk/ Yate enabled p2p Video IP Communications Suite for Web / PSTN. These serverless broadband ready platform enable OS community to save 90% on capital & operating costs over proprietary software for conferencing &

Software to use the Creative Labs USB VOIP Blaster on NetBSD, Linux, and Windows.

Open79XX XMLDirectory is an open PHP-based XML menuing system for providing on-screen services to the Cisco 79XX Series IP Phones. It contains dynamically generated phone directories, on-screen memos, links to other services, and many more options.

Simple RFC3261 Compliant SIP Proxy.

GoVictor is a channel changer for the popular windows voip client Roger Wilco. It operates in the background, watching for hotkeys associated with predefined RW channels.

This is a Java wrapper for Ffmpeg. The JMF (Java Media Framework) plugins system lets one use JMStudio or other application to play H.263/rfc2190 streams. Support for H.263+ and MPEG-4 is currently under test, please contact us for further information.

With SIP Proxy you will have the opportunity to eavesdrop and manipulate SIP traffic. Furthermore, predefined security test cases can be executed to find weak spots in VoIP devices. Security analysts can add and execute custom test cases.

CPhone is a cross platform voip client which uses the H323 protocol. It compiles and runs on Linux, BSD, Windows and Mac (OS X) boxes. It is based on the libraries from www.openh323.org and Trolltech.

The Cisco IP Telephony Services project contains many IP phone services and utilities for the Cisco 79XX phones and CallManager. This project houses several scripts, services, and administration tools. Recruiting! Contact me and become a developer.

An example registrar portal for provisioning ENUM domains with ENUM trial registries. Currently supported is the Austrian ENUM Trial registry, other registries may be added.

ST-PBX Live is a GNU/Debian Linux distribution integrating Asterisk, freePBX and other tools to try from CD or install in HD a complete VoIP system.

VoiceXML Voice Mail Appliation written in Java. Inlcudes a distribution for the Voxeo Prophecy platform.

Next generation web-based, multi-tenant, distributed, mulit-lingual, inbound, outbound Video enabled VoIP & VVoIP call/ contact center solution for Asterisk with Built-in phone, IVR, CRM, Predictive dialer, ACD, Chat, Mail, Fax, Video and other features.

Teamspeak enables people to speak with one another over the Internet. The Administration of a Server can be done by Telnet, Mysql, Web-Frontend or with the Teamspeak-Client. This Project is a Interface to Perl.

Cross-platform Java Teamspeak client