Open Source C++ Telephony Software for Linux
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C++ Telephony Software for Linux
View 49 business solutionsBrowse free open source C++ Telephony Software for Linux and projects below. Use the toggles on the left to filter open source C++ Telephony Software for Linux by OS, license, language, programming language, and project status.
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VoIP monitor
VoIP SIP and SKINNY quality analyzer and packet / audio recording tool
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. -
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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
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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
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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.
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TelcoChisel OS
Advanced Telecom Security Live Linux OS for 5G/4G Cellular Auditing, S
TelcoChisel is an advanced, bootable live Linux operating system purpose-built for cellular security researchers, Software Defined Radio (SDR) engineers, and telecommunications penetration testers. Eliminating dependency conflicts and driver compilation issues, TelcoChisel delivers an out-of-the-box, carrier-grade laboratory equipped with 88 pre-configured telecommunications security tools across 10 specialized domains. Built on Ubuntu 24.04 LTS (Noble Numbat) with the Linux low-latency kernel (`linux-image-lowlatency`), lightweight XFCE4 desktop, and keyboard-driven i3 tiling window manager. -
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Bayonne is supported from Savanna.gnu.org. Please visit there. Sourceforge is generally depreciated except for mailing lists, but updated releases will be posted here from time to time.
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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.
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oreka
Enterprise telephony recording and retrieval system
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. -
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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.
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Stuntman - STUN server and client
High performance, production quality STUN server and client library
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 -
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A project for a highly scalable protocol and framework and a collection of software tools for m2m (machine to machine communication) and IoT Internet of Things. The programs here works on Windows and Linux and are based around VSCP, The Very Simple Control Protocol. Repository is here: https://github.com/grodansparadis/vscp_software and here https://github.com/grodansparadis/vscp_firmware
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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.
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mod_MumbleLink
Positional Audio Communication for Minecraft with Mumble
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 -
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The ASN.1 Compiler
Go to github.com/vlm/asn1c for the latest version.
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 -
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KAlcatel
Application for Alcatel 50x and 70x phones
KAlcatel allows you to manage messages, calls, contacts, todos and the calendar in your Alcatel (tm) One Touch 50x and 70x mobile. This project is not maintained anymore, use Wammu instead (http://wammu.eu/) -
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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.
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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.
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GNU Gatekeeper (GnuGk)
H.323 Gatekeeper for VoIP and videconferencing
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/ -
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The CSTA Switch Simulator is a software telephone environment that can be used to test Computer Telephony Integration (CTI) applications. The simulator complies with the CSTA phase II specification.
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Vodia PBX
A Better PBX For Your Business
Vodia PBX has included a complete suite of business telephony features for on-premise and cloud-based telephony deployments. Sign up at https://vodia.com/en/register and receive your free Vodia license. -
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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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CSTA Inside
Open source CSTA library
Aim of this project is to provide a library for C/C++ developers to use ECMA-269/ECMA-323 CSTAIII XML protocol. It will also include: CSTA interface for FreeSWITCH and example client application. (extending the PBX and protocol support is in progress). Check wiki for details. -
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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.
