Open Source C++ VoIP Software

VoIPmonitor is open source network packet sniffer with commercial frontend for SIP SKINNY MGCP RTP and RTCP VoIP protocols running on linux. VoIPmonitor is designed to analyze quality of VoIP call based on network parameters - delay variation and packet loss according to ITU-T G.107 E-model which predicts quality on MOS scale. Calls with all relevant statistics are saved to MySQL or ODBC database. Optionally each call can be saved to pcap file with either only SIP / SKINNY protocol or SIP/RTP/RTCP/T.38/udptl protocols. VoIPmonitor can also decode audio.

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

Open Source - GPLv3 inc images. *** PLEASE NOTE: There are now 2 seperate versions here. *** One is Pre Firefox 57. The other is Post Firefox 57. *** For those providing mirrors, please enable your users to realize this. Vidalia Based, Tor as a Service Solution. MicroSip: enables FREE PC to PC video calling with no account sign-up and no middleman server. WASTE: enables FREE Conference VoIP, chat, file transfer and support. *** AI Powered *** Tor/i2p: enables safer browsing. Tor/i2p Profile: Browse over Tor/i2p on Firefox (Vanilla, ESR), Waterfox (Classic, New), Palemoon , LibreWolf and legacy CyberFox. As with all versions of Tor - do not rely on this for strong anonymity. A usability enhanced Privacy Pack. Use alongside Tor Browser Bundle. An installer, for : Vista 32/64, Win7 32/64, Win8 32/64, Win10 32/64, Win11, Linux (Wine)

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.

What is t38modem? From your application view point it's a fax/voice modem pool. From IP network view point it's a H.323/SIP endpoint with T.38 fax support. From your view point it's a gateway between an application and IP network. Works with HylaFAX.

New version 1.2. This is the code to STUNTMAN - an open source STUN server and client code by john selbie. Compliant with the latest RFCs including 5389, 5769, and 5780. Also includes backwards compatibility for RFC 3489. ICE and WebRTC ready. Version 1.2 compiles on Linux, MacOS, BSD, and Solaris. Supports the STUN protocol on both UDP and TCP for both IPv4 and IPv6. Windows binaries are also provided. Additional features are in development. This is a mirror of the code on https://github.com/jselbie/stunserver More details on the project's website: http://www.stunprotocol.org

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/

A Mod so that Minecraft now natively supports Mumble's positional audio feature. This means: Directional and positionally attenuated VOIP in relation to the game world. Please visit the Forum for information about the newest Version! Main Forum-Thread: http://www.minecraftforum.net/forums/mapping-and-modding/minecraft-mods/1272675 TheSkorm's Fork on GitHub: https://github.com/TheSkorm/mod_mumblelink Mumble: http://mumble.sourceforge.net Donations are greatly appreciated: https://sourceforge.net/donate/?user_id=2370023

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.

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.

Comprehensive business management system The full-cycle automation ERP system - [ Full Freeware ] IP PBX - Complete synchronization with CRM system - Call accounting for PBX (PBX billing) - Phone call control and monitoring - Phone recording systems Accounting, Financial management - Trade management, warehouse - Management and tax accounting Corporate information system - Client relations management system (CRM) - Project, tasks, orders management - On-line conferences (forums) - Instant messenger, chat, sending SMS Staff management system [HRM] - Personnel records, documents circulation - Time and Attendance tracking system - Computer usage monitoring (time tracking) - Work time accounting (five levels) - Accrual and deduction system - Fines system (two levels) - Salary calculation Access control systems Management of access rights to the premises Video surveillance system (IP,Analog) Photofixation, Videofixation Technological process automation

A command line SIP/H323 softphone capable of sending and receiving audio files as well as sending out of band DTMF digits. Supports a BNF format configuration language for scripting call scenarios. Useful for example system testing.

FreeSentral is a full IP PBX consisting of a Linux Distribution, an IP PBX and a Web Graphical User Interface for easy configuration.

Jabbin is an Open Source social application that combines VoIP, Instant Messaging and Social Networking, allowing you to focus on what you really care about: your friends.

Open source software components of the AastraLink Pro 160 Supplied from CDROM "71-001049-02 REV01" by Aastra Telecom

The goal of this project is to provide a liveCD allowing the user to analyze networks for VoIP installations. This project gives you a global network state.

Apreta is about sharing information in real-time meetings. It provides desktop sharing, presentation sharing, a shared whiteboard, and audio conferencing.

Video Applications for Asterisk (libh43m, app_h324m, app_mp4, app_rtsp, app_transcode, app_swf) H324m stack, mp4 record and playblack, rtsp support, video transcoding and swf file playback.

A C++ lib for the Asterisk PBX AGI interface.

COMONet is a novel concept in which a voice call is routed via an ad hoc mobile network avoiding the conventional Mobile Service Provider. COMONet is consisting of community of mobile users who stay between two mobile users who are making the call.

This Gui Element is used by Voicent Call Center Manager to track and update statistics.

A client-side library for secure voice conference.

Fideliphone: High Quality Internet Telephony

This is an Interactive Voice Response System using TAPI (MS TAPI 2.x).You can use this as an answering machine.But most of all this is meant as an illustration of how you can use TAPI API for developing IVR applications in VC++

REPOSITORY MOVED TO GITHUB!! https://github.com/medooze/media-server Video Multiconference Media Server with WebRTC support. Provide Multiconference and video broadcasting services to any SIP service. Supports VP8, H264, MP4V-ES, H263 and H263P, continuous presence, RTMP flash broadcasting, adhoc conferences, load balancing and administrative WEB interface. JSR309 driver implementation under development. .