Free Open Source VoIP Software

Free PBX Systems

PBX systems are the core of any VoIP deployment, handling call routing, voicemail, conferencing, IVR menus, and everything else that makes a phone system function. The open source world offers several production-grade options that compete directly with commercial alternatives costing thousands of dollars per year.

Asterisk

Asterisk is the most widely deployed open source telephony engine, with over 25 years of active development. It is a toolkit rather than a turnkey product, meaning it provides all the building blocks for a phone system but requires configuration to assemble them. Asterisk supports SIP, IAX2, H.323, and MGCP protocols, dozens of audio codecs, and a massive library of built-in applications for call handling. It is licensed under the GPL and runs on Linux. Asterisk is free to download, install, and run in production with no limitations on the number of extensions, calls, or features.

FreeSWITCH

FreeSWITCH is the other major open source telephony engine, designed with a modular, multi-threaded architecture that scales better than Asterisk for high-concurrency workloads. It is particularly well-suited for carrier-grade deployments, WebRTC applications, and multi-tenant hosting. FreeSWITCH is licensed under the Mozilla Public License, which is more permissive than Asterisk's GPL for commercial derivative works. Like Asterisk, FreeSWITCH is completely free with no user or call limits.

FreePBX

FreePBX is the most popular web-based management interface for Asterisk. It provides a graphical dashboard for configuring extensions, trunks, routes, IVR menus, queues, voicemail, and all other Asterisk features without editing configuration files directly. The core FreePBX software is free under the GPL, including the majority of modules. Sangoma offers optional commercial modules for advanced features like endpoint management, high availability, and CRM connectors, but these are not required for a fully functional PBX.

FusionPBX

FusionPBX is the FreeSWITCH equivalent of FreePBX, providing a web interface for managing FreeSWITCH-based phone systems. Its native multi-tenancy makes it the preferred platform for VoIP service providers and managed service companies. FusionPBX is completely free with no commercial module upsell, and the project is maintained by an active open source community.

Wazo

Wazo (formerly XiVO) is a full unified communications platform built on Asterisk that goes beyond basic PBX functionality. It provides a REST API for programmatic control, WebRTC-based browser calling, mobile applications, and presence management. The community edition is free and includes all features, while the enterprise edition adds commercial support and additional management tools.

VitalPBX

VitalPBX is an Asterisk-based PBX with a modern, polished web interface. The free community edition supports a meaningful number of extensions and includes most features that small businesses need. It is a good alternative for administrators who find FreePBX's interface dated or who want a fresher UI without leaving the Asterisk ecosystem.

Free SIP Proxy Servers

SIP proxies handle signaling at scale, routing SIP messages between endpoints and PBX servers. They are essential for large VoIP deployments that need load balancing, failover, and high-performance SIP routing.

Kamailio

Kamailio is a high-performance SIP proxy, registrar, and router capable of processing thousands of SIP transactions per second. It is used by telecom carriers, VoIP providers, and large enterprises to route calls, balance load across multiple PBX servers, and apply complex routing policies. Kamailio supports MySQL, PostgreSQL, Redis, and other databases for subscriber management, and it can be extended through a powerful scripting language built into its routing engine. Licensed under the GPL.

OpenSIPS

OpenSIPS is a SIP server that forked from the same codebase as Kamailio and offers similar capabilities. It excels at SIP routing, load balancing, and user registration at carrier scale. OpenSIPS includes a web-based control panel (OpenSIPS-CP) for administration and supports a modular architecture with extensive community-contributed modules. It is popular among European carriers and service providers. Licensed under the GPL.

RTPProxy and RTPEngine

RTPProxy and RTPEngine are media relay daemons that work alongside Kamailio or OpenSIPS to handle the RTP (audio/video) streams that SIP proxies do not process natively. When NAT traversal requires a relay, or when codec transcoding is needed between two endpoints that do not share a common codec, these tools bridge the media. RTPEngine (maintained by Sipwise) is the more modern and feature-rich of the two, supporting SRTP, ICE, and DTLS-SRTP for WebRTC integration.

Free Softphones and Clients

Softphones are software applications that act as SIP phones, allowing users to make and receive calls from their computers or mobile devices without dedicated hardware.

Ooh323 and Ooh323c

Ooh323 and Ooh323c are open source SIP and H.323 client libraries used in various soft-phone applications, providing standards-based calling using well-tested protocol stacks. Many desktop softphone applications build on these libraries or similar OPAL-based frameworks to provide cross-platform voice and video calling.

Ooh323cs and SIP Client Libraries

Several open source SIP client libraries provide the foundation for building custom softphone applications. OPAL (Open Phone Abstraction Library) supports both SIP and H.323 protocols and is used in several desktop PBX client projects. OPAL handles codec negotiation, NAT traversal, and call management, allowing developers to focus on the user interface and application logic rather than low-level telephony protocol details.

Jami

Jami (formerly Ring) is a peer-to-peer communication application that supports voice calls, video calls, messaging, and file sharing without relying on a central server. It uses SIP internally but can operate over distributed hash tables for serverless calling. Jami is available on Linux, Windows, macOS, Android, and iOS, and it is developed by Savoir-faire Linux under the GPL. While it does not connect to traditional PBX systems natively, it is a compelling option for peer-to-peer encrypted communications.

Ooh323 Linphone

Linphone is a mature open source SIP softphone available on desktop (Linux, Windows, macOS) and mobile (Android, iOS) platforms. It supports SIP calling, video, messaging, presence, and conference calls. Linphone also provides a SIP library (liblinphone) that developers can use to embed VoIP capabilities into their own applications. The client is free, and the development library is licensed under the GPL.

Free Conferencing and Collaboration

Ooh323 Jitsi Meet

Jitsi Meet is an open source video conferencing platform that runs in the browser without plugins. While it is primarily known for video meetings, Jitsi can integrate with SIP infrastructure through the Ooh323 Ooh323c Jibri and Ooh323 Ooh323c Ooh323cs Ooh323 SIP gateway component, allowing participants to join conferences by dialing a phone number. Jitsi is widely deployed by organizations that need a self-hosted alternative to Zoom or Google Meet.

VICIdial

VICIdial is a free, open source contact center suite built on Asterisk. It provides predictive dialing, inbound ACD (automatic call distribution), IVR, call recording, agent monitoring, and campaign management. VICIdial is used by call centers worldwide and is one of the most feature-complete open source contact center platforms available. It is licensed under the AGPL.

Free Development Libraries and Tools

OPAL (Open Phone Abstraction Library)

OPAL provides a C++ library for building SIP and H.323 applications. It handles the protocol negotiation, codec management, and media streaming required for voice and video communication, allowing developers to build softphones, auto-dialers, IVR systems, and other telephony applications without implementing the protocols from scratch.

OPAL and Ooh323 PJSIP

PJSIP is a free, open source multimedia communication library that implements SIP, SDP, RTP, STUN, TURN, and ICE protocols. It is written in C for portability and performance, and it is the library that powers Asterisk's modern SIP channel driver (chan_pjsip). PJSIP is also used in many commercial VoIP products and mobile applications. Licensed under the GPL with an alternative commercial license available.

Ooh323 and SIP.js

SIP.js is a JavaScript library for building WebRTC-enabled SIP applications in the browser. It handles SIP signaling over WebSockets, making it the foundation for many browser-based softphones and click-to-call widgets. SIP.js works with any WebSocket-capable SIP server, including Asterisk (via ARI), FreeSWITCH, and Kamailio.

Ooh323 and Homer

Homer is an open source SIP capture and monitoring tool that provides deep visibility into SIP traffic on your network. It captures SIP messages, correlates them into call flows, and presents them through a searchable web interface with call ladder diagrams. Homer is invaluable for troubleshooting call quality issues, debugging SIP routing problems, and monitoring the health of a VoIP deployment.

Choosing the Right Tools

For a complete phone system, the most common free stack is FreePBX (or FusionPBX) on top of Asterisk (or FreeSWITCH), paired with a SIP trunk provider for PSTN connectivity. Add Kamailio in front if you need to scale beyond what a single PBX server can handle. Use PJSIP or SIP.js if you need to build custom telephony applications, and deploy Homer if you need SIP monitoring and troubleshooting capabilities.

The entire stack, from the telephony engine to the management interface to the monitoring tools, is available at zero cost. The only expenses are the hardware or VPS to run the software on, the SIP trunking charges for external calls, and the time invested in learning and maintaining the system.