Open Source Internet Phone Software

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.

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.

This project implements a simple STUN server and client on Windows, Linux, and Solaris. The STUN protocol (Simple Traversal of UDP through NATs) is described in the IETF RFC 3489, available at http://www.ietf.org/rfc/rfc3489.txt

Open Phone Abstraction Library (OPAL) is a C++ multi-platform, multi-protocol library for Fax, Video & Voice over IP and other networks. Also included is the Portable Tool Library (PTLib) which is a C++ multi-platform abstraction library and collection o

This project has been superseded by OpalVoip (https://sourceforge.net/projects/opalvoip/) and H323Plus (https://sourceforge.net/projects/h323plus/) The OpenH323 project provides full featured,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.

Kablink open team collaboration software uses social networking to unify team workspaces w/ real-time web conferencing. Collaboration for knowledge networking, program management, communities-of-practice, telework, other business process/functional areas

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).

Activa brings the Asterisk IP PBX to the call center. Built on top of Asterisk, Activa components enable successful call center implementations adding value in areas such as computer telephony, screenpop&click2dial, agent control, automatic dialing...

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

jfPhone is a VoIP/SIP SoftPhone for Desktops.

Voxy - the simplest way to integrate VoiceXML applications in Asterisk. Configure your dial plan with the URL of your VoiceXML application and it's done.

The XTapi project is aimed at someday growing up to be a reference implementation for JTAPI. At this time it is not the intention to cover every JTAPI package, just the Core and Media packages are covered.

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/

wxCommunicator is a cross platform SIP softphone written in C++ utilizing customized sipXtapi user agent library and wxWidgets 2.8.9 GUI library. For a list of supported features see http://wxcommunicator.sourceforge.net/features.html .

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.

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.

An implementation of the iax protocol using java. Only alaw, ulaw codec Supported.

OpenGatekeeper is an Open Source H.323 Gatekeeper based on the work done by the OpenH323 project.

A software SIP (Voice over IP) Framework, including a phone, written in Python

Maintained version of old asterisk applications ported to 1.4 that for copyright reasons cannot be included into official Digium's releases and that for some reason the author is not keeping up-to-date. The project also require Zaptel instead of DAHDI.

Open Source Voip Billing / Calling Cards / Callshops / LCR For Asterisk

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

Real-Time Collaboration (RTC) and conferencing project. This product has been deprecated. Please see the replacement product, CoolCollaborator, at http://coolcollaboratr.sourceforge.net/