Open Source C++ 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.

Sipp is a performance testing tool for the SIP protocol. Its main features are basic SIPStone scenarios, TCP/UDP transport, customizable (XML-based) scenarios, dynamic adjustment of call-rate and a comprehensive set of real-time statistics.

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

Vem ai o Novo Disc-OS Aguardem.. Uma Empresa Orkestral Estamos Voltando! Modificações: Asterisk 21 Nova Interface UI Centos 8 Nodejs Disc-OS é uma distribuição de um PABX IP baseado em software livre. Desenvolvido para o mercado brasileiro com interfaces em português, de fácil instalação e configuração, contendo Linux customizado, software Asterisk 1.4 e configurador Disc.

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.

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 .

Google's common Java, C++ and JavaScript library for parsing, formatting, and validating international phone numbers. The Java version is optimized for running on smartphones, and is used by the Android framework since 4.0 (Ice Cream Sandwich). libphonenumber gets the type of the number based on the number itself; able to distinguish Fixed-line, Mobile, Toll-free, Premium Rate, Shared Cost, VoIP, Personal Numbers, UAN, Pager, and Voicemail (whenever feasible). Provides valid example numbers for all countries/regions, with the option of specifying which type of example phone number is needed. Quickly guesses whether a number is a possible phone number by using only the length information, much faster than a full validation. To include the Java code in your application, either integrate with Maven (see wiki) or download the latest jars from the Maven repository.

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.

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...

Intel Integrated Performace Primitives audio/video codecs plug-in for the OPAL/OpenH323 library including G.728, G.729, G.723.1, G.722.2 GSM-FR, GSM-AMR, H.261, H.263, H.264 and MPEG4 part 2.

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.

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.

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/

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.

KPhone is a SIP UA for Linux, supporting a multitude of features. Originally developed by Billy Biggs, it was developed at Wirlab until 2005. It is now developed by a team of volunteers in this project.

This project develops video conferencing and VoIP applications for Mac OS X. XMeeting is based on the work of the OpenH323 project which provides the libraries for H.323 and SIP support.

Audio/Video Conference software for Instant Messengers. It aims to provide Audio/Video conferencing for as many Instant Messengers as possible through a modular design.

Social Networking Features include multiplayer 3D Billiards, Personal Web Server, Web Cam Sever with motion detection, Web Cam Client, VOIP Voice Phone, P2P File Sharing, Instant Message, Chat Rooms and more. 3rd Party Plugins Welcome :)

Speech profile of Person is created that contains elementary sounds uttered. Profile is 1 time download for listeners. The actual audio sample is encoded based on profile. Decode using the profile stored earlier by User, and the audio can be regenerated.

An overlay Networking Protocol, aims in creating a big secured virtual network, optimizing both speed and cost.

Anca is a H.323 client for Qt 3.x. It means that it allows communicating (voice/video) with other similar clients over internet, eg. Gnomemeeting. Its design is minimalist and clean; has a support for embedded devices. It is very modular.

PNX system develops the middle-ware source code to glue Asterisk with a number of powerful telephony products such as: 1) OpenH323 H.323 stack 2) Vovida SIP stack 3) Bayonne Voice Automation Platform The advantages? An advanced IP PBX supporting the wide

A C++ framework utilizing Design Patterns for creating Linux and Windows communications applications that contain Dialogic® products. Includes media and network classes (analog, digital, SIP, H323), multithreaded event handling, distributed app support.

Asterisk Manage API and FAGI C++ Framework. Implemented with high efficient OS independent Sockets wrapper (ACE). Provide C++ API wrappers for Asterisk Manager API and FAGI. A Manage API relay(proxy) server and a FAGI socket server will be implemented.