Open Source C++ Telephony 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.

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.

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.

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

SIMCardManager is Windows tool that can read a SIM card information (ICCID, IMSI, Pin states) and SIM card content : phone book (ADN), Fixed dialing (FDN) and SMS messages. It also enables you to authenticate with PIN when activated and export SIM card content to text files. You need to have a PC/SC compatible smart card reader or dongle with SIM slot. You can find other useful tools on my website http://www.idrix.fr

Seagull is a multi-protocol traffic generator. Especially targeted towards IMS, Seagull supports Diameter (RFC3588 and all applications) over TCP/SCTP and IPv4/IPv6 , TCAP (over SS7 or Sigtran), XCAP over HTTP and Radius.

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)

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

An open implementation of the SS7 core protocols, MTP, SCCP, ISUP, and TCAP.

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.

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.

The Open VXI VoiceXML interpreter is a portable open source library that interprets the VoiceXML dialog markup language. It is designed to serve as a reference for parties interested in understanding how VoiceXML markup might be executed.

Enterprise telephony recording and retrieval system with web based user interface. The project currently supports recording voice from VoIP SIP, Cisco Skinny (aka SCCP), raw RTP and audio sound device and runs on multiple operating systems and database systems. It can record audio from most PBX and telephony systems such as BroadWorks, Metaswitch, Asterisk, FreeSwitch, OpenSIPS, Avaya, Nortel, Mitel, Siemens, Cisco Call Manager, Cosmocom, NEC, etc... It is amongst others being used in Call Centers and Contact Centers for Quality monitoring (QM) purposes.

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.

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 ASN.1 compiler turns ASN.1 specifications into C code. The asn1c is shipped together with conformant BER/DER/XER/PER codecs. The X.509, GSM TAP3, MEGACO, RRC and LDAP encoding and decoding examples are part of the source code distribution. NOTE: THE asn1c PROJECT HAS LARGELY MOVED TO GITHUB: http://github.com/vlm/asn1c

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

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...

Easily manage and control everything on your Android & iOS devices from computer. Sync, transfer and back up any data among different devices, including Android phone, Android tablet, iPhone, iPad and computer. Discover and download awesome content for your Android and iOS devices right from your desktop or laptop. Customize wallpaper and DIY ringtones of your own! And when I say everything and any data, I mean apps, games, contacts, messages, photos, music, videos, software, files, documents, etc.

Project moved: Find the latest code at https://www.h323plus.org/ Follows on from now depreciated OpenH323 and designed as a drop in replacement for OpenH323. Development of advanced open source H.323 including application sharing, video conferencing and incorporates new research and development work. Project moved: Find the latest code at https://www.h323plus.org/

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.

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/

This is a Win32, VC++ MFC application to manage Nokia mobile phones melodies. Includes VC++ source code and Rational Rose UML model.

A GSM sim card management tool capable of creating, editing, deleting, backup and restore operations on GSM sim card phonebook and SMS records.