Open Source VoIP Software
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VoIP Software
View 378 business solutionsBrowse free open source VoIP software and projects below. Use the toggles on the left to filter open source VoIP software by OS, license, language, programming language, and project status.
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IssabelPBX
Issabel PBX - Unified Communications
Open Source and Unified Communications partners created a new platform based on an Elastix® fork (currently purchased by 3CX) to provide the community with continuity, peace of mind and support needed to continue with their PBX and operation developments. Contribute to the funding of Issabel on https://www.patreon.com/issabel -
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NoiseGator (Noise Gate)
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. -
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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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Elastix
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. -
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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)
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Synapse
Matrix reference homeserver
Matrix is an ambitious new ecosystem for open federated Instant Messaging and VoIP. Everything in Matrix happens in a room. Rooms are distributed and do not exist on any single server. Rooms can be located using convenience aliases like #matrix:matrix.org or #test:localhost:8448. Synapse is currently in rapid development, but as of version 0.5 we believe it is sufficiently stable to be run as an internet-facing service for real usage! Create and manage fully distributed chat rooms with no single points of control or failure. Eventually-consistent cryptographically secure synchronization of room state across a global open network of federated servers and services. Send and receive extensible messages in a room with (optional) end-to-end encryption. Use 3rd Party IDs (3PIDs) such as email addresses, phone numbers, Facebook accounts to authenticate, identify and discover users on Matrix. -
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JDA
Java wrapper for the popular chat & VOIP service
JDA strives to provide a clean and full wrapping of the Discord REST api and its Websocket-Events for Java. This library is a helpful tool that provides the functionality to create a discord bot in java. Discord is currently prohibiting the creation and usage of automated client accounts (AccountType.CLIENT). We have officially dropped support for client login as of version 4.2.0! Note that JDA is not a good tool to build a custom discord client as it loads all servers/guilds on startup, unlike a client which does this via lazy loading instead. If you need a bot, use a bot account from the Application Dashboard. Creating the JDA Object is done via the JDABuilder class. After setting the token and other options via setters, the JDA Object is then created by calling the build() method. When build() returns, JDA might not have finished starting up. However, you can use await ready() on the JDA object to ensure that the entire cache is loaded before proceeding. -
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BlackBelt Privacy Tor/i2p+WASTE+VidVoIP
Browse on Tor/i2p, Anon p2p Chat / FileTx, Conf / Video VoIP
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) -
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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.
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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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TurnServer is a implementation of Traversal Using Relay around NAT (TURN) protocol. This protocol allows a client to obtain IP addresses and ports from such a relay.
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OpenSIPS (former OpenSER) is an GPL implementation of a multi-functionality SIP Server that targets to deliver a high-level technical solution (performance, security and quality) to be used in professional SIP server platforms. IMPORTANT: this is no longer the main hosting for the project. This was moved on GITHUB - https://github.com/OpenSIPS/opensips
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Vaak (Elastix-Persian)
http://www.vaakpbx.com
سيستم تلفني واك مبتني بر استريسك و محيط گرافيكي الستيكس، تنها سيستم تلفني ويپ فارسي دنيا . -
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BlackBelt WASTE - ipv4/Tor/i2p +AI+Voice
Modern, AI-Smart, WASTE p2p for ipv4, Tor and i2p + Voice Conference.
Open Source - GPLv3 inc images. A WASTE client. Download and create your own WASTE networks. Move 1000's of GB's at 100MB+ per sec. (800 Mbits per sec) FULL pause and resume capable. Voice Conference, Chat, Transfer files and Participate in Forums in a secure environment. For Windows XP 32/64, Vista 32/64, Win7 32/64, Win8 32/64, Win 10, Win 11, Linux (WINE). *** User Based Access Control - for voice, chats, file transfers and uploads. (useful in NULLNETS) *** Distributed Autonomic-Performance-Tuning - A Goal-Seeking Swarming-Semiotic AI *** AI Connect - AI Managed Connections. *** Self-Organising Anti-Spoofing Technology *** Geared Multi-threading, providing the smoothest performance possible *** Advanced Threat Detect and Manage Technology *** Voice Conferencing Over WASTE *** RNN - Recurring Neural Net - AI Noise Reduction *** Differential Files Transfer - Seriously fast data backups -
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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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CafeSip - Look what Java and SIP can do
A suite of open-source tools and frameworks for creating SIP apps
Session Initialtion Protocol (SIP) is widely used for telephone services over the Internet. CafeSip provides a suite of open-source tools and applications for creating customized SIP services and applications using the Java. -
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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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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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PolycomVVXControl
A command line utility for remote controlling Polycom VVX IP phones
Application to remote control Polycom VVX IP phones via their web interface (using HTTPS). This application is initially intended to perform certain actions on phones running in Microsoft Lync mode. These actions include: * get device information * get status * sign in using PIN authentication * sign out * reboot * factory reset It also supports performing actions in batch reading data from a CSV file. -
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OSDial
OpenSource Call Center and Dialer System
An Open Source (predictive) Dialer. OSDial is a full featured GPL dialer which does Manual, Power and Predictive dialing out of the box. Its easily installed through one of our .iso images. Install manual available at http://store.osdial.com/ Full details available at web site (http://osdial.com). -
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Asterisk Monitor is a HTML interface that acts a operator pannel for asterisk to display user/peer status and calls. This uses a reverse AJAX, PHP and Python to originate, transfer and hangup calls, manage queues and meetme rooms.
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Asterisk Channels Live
This is a PBX dashboard showing the peer,queue,agent at real time
http://www.astchannelslive.com/sourceforge.php This is a dashboard work with Asterisk(FreePBX) any other installation: 1. Showing the peer status 2. queue,agent at real time 3. Hangup, Spy,.... -
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ICTFax
Open source Fax server software for inbound / outbound internet Faxing
Note: for binaries and installation instruction please visit official website. http://ictfax.org/ ICTFax is Open Source Fax Server software to send and receive Fax over Internet & Emails developed over Freeswitch, ICTCore and Angular Framework , It works with T.38 and G.711 pass through and feature email to fax, fax to email, web to fax, fax to web and also support fax machines to send and recieve Fax using ATA. Installation Guide https://www.ictfax.org/ictfax-installation-guide/ Administration Guide https://www.ictfax.org/ictfax-admin-guide/ User Guide https://www.ictfax.org/ictfax-user-guide/ Commercial Support If you do not posses Linux / VoIP expertise to install and setup ICTFax or you are busy, We can help you in installation, configuration and testing to make sure ICTFax setup is functional and tested. Below are Installation package link for custom projects and development work. Please Installation support services. -
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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. The project moved to https://github.com/chan-sccp/chan-sccp
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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/
