Below is a listing of open source software having to do with Telephony. For a list of VoIP and business telephone providers please visit Voip-Info.org.
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.
Unified Communications Server
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.
Nokia NBU, NBF, NFB, NFC and ARC backup file parser, extractor and viewer. It can help you to check content of backup or extract files from it. Requires MS .Net Framework 2
A simple noise gate app intended for use with VOIPs like Skype.
Ever wanted to cut out background noise when talking with others on Skype? Now it's possible! NoiseGator is a light-weight noise gate application that routes audio through an audio input to an audio output. In real-time the audio level is analysed and if the average level is higher than the threshold the audio bypasses as normal. However, if the average level goes below the threshold, the gate closes and the audio is cut. When used with a virtual audio cable it can act as a noise gate for a either a sound input(microphone) or sound output(speakers). Can also be used to gate noise from your own mic or play your microphone through your speakers. REQUIREMENTS: - Java 7 or higher for Windows. - Java 6 or higher for Mac. Java 7 recommended. - A virtual audio cable is required for use with VOIPs: For Windows users I recommend the VB-Cable driver (http://vb-audio.pagesperso-orange.fr/Cable/index.htm). Mac users can use SoundFlower.
trixbox CE is an easy to install, VOIP phone system based on the Asterisk PBX. trixbox is designed for home or office use. trixbox CE includes CentOS linux, mysql, and all the tools needed to run a business quality phone system. (formerly asterisk@home)
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.
The libSRTP project has been moved to https://github.com/cisco/libsrtp
Simple to install and use Tor client / server with WASTE and VideoVoIP
*** 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. MicroSip: enables FREE PC to PC video calling with no account sign-up and no middleman server. WASTE: enables chat, file transfer and support. Tor: enables safer browsing. Tor Firefox Profile: Browse over Tor. As with all versions of Tor - do not rely on this for strong anonymity. A usability enhanced Privacy Pack. An installer, for : Windows XP 32/64, Vista 32/64, Win7 32/64, Win8 32/64, Win10 32/64.
Vicidial Contact Center Suite
This software suite is designed to extend the functionality of the Asterisk PBX through platform-independant web-client applications. Includes the VICIDIAL inbound/outbound contact center application. The suite is scalable across multiple Asterisk servers.
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.
سيستم تلفني واك مبتني بر استريسك و محيط گرافيكي الستيكس، تنها سيستم تلفني ويپ فارسي دنيا .
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
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 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.
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...
Cellular manager for mobile phones/modems
Gammu is a cellular manager for mobile phones/modems. It contains libraries and functions for ringtones,logos,phonebook,SMS,etc. (used by external software), a command line version (with backup/restore) and SMS gateway (with MySQL and PostgreSQL supp
Siproxd is a proxy/masquerading daemon for the SIP protocol. It allows SIP clients (softphones & hardphones) to work behind an IP masquerading firewall or router.
JFritz is a application written in Java, which makes it possible for AVM FRITZ!Box users to download the caller history from their box and manage it on their desktop pc.
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.
Kalkun is a simple web-based SMS (Short Message Service) management, it use gammu-smsd (part of gammu family) as SMS gateway engine to deliver and retrieve messages from your phone/modem.
An open source 3GPP LTE implementation.
OpenLTE is an open source implementation of the 3GPP LTE specifications. Currently, octave code is available for test and simulation of downlink transmit and receive functionality and uplink PRACH transmit and receive functionality. In addition, GNU Radio applications are available for downlink transmit and receive to and from a file, downlink receive using rtl-sdr, HackRF, or USRP B2X0, LTE I/Q file recording using rtl-sdr, HackRF, or USRP B2X0, and a simple eNodeB using USRP B2X0. The current focus is on extending the capabilities of the GNU Radio applications and adding capabilities to the simple base station application (LTE_fdd_enodeb).
Replacement for the SCCP channel driver in Asterisk. Extended features include Shared Lines, Presence / BLF, customizable Feature Buttons, and Custom Device State. Visit our discussion mailing list for help and join us as a developer if you like.
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
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.