Asterisk, FreeSWITCH, Kamailio, OpenSIPS and Kazoo compared

Start with layers, not products

Most comparisons of "Asterisk vs FreeSWITCH vs Kamailio" are framed as if a buyer picks one. That framing produces bad architectures. The projects occupy different positions in the path of a call, and the useful question is which layer you need, then which project is strongest at that layer.

The vocabulary comes from SIP itself. A SIP proxy forwards requests on behalf of others and stays out of the media. A back-to-back user agent (B2BUA) is defined in RFC 3261, as quoted by RFC 7092, as "the logical combination of a User Agent Server (UAS) and User Agent Client (UAC)":40 it ends the call on one side and starts a new one on the other. RFC 7092 then sorts B2BUAs by how deep into the media they go, from a "signaling-only" B2BUA that "operates at the SIP layer but in ways beyond those of [RFC3261] proxies", through a media relay that "terminates the media plane at the IP and transport (UDP/TCP) layers", to a media-termination B2BUA that "operates at the media payload layer".40 That last category is where you can play a prompt, record a call, detect a keypress or transcode between codecs.

Mapped onto that vocabulary, the stack looks like this:

• SIP proxy, registrar and router: Kamailio and OpenSIPS. They route signalling at high volume and hand media to something else.
• Media server and application engine: Asterisk and FreeSWITCH. They terminate calls, handle audio, and run the menu, voicemail, queue or conference logic.
• Media relay: rtpengine. It forwards RTP packets under the control of a proxy, without running any call logic of its own.
• Platform: Kazoo. An API-driven, multi-tenant system that orchestrates proxies and media servers as one cluster.
• GUI distributions: FreePBX (on Asterisk) and FusionPBX (on FreeSWITCH). A web administration layer, plus installers, that turns an engine into a PBX an administrator can run.

Once the layers are clear, most of the "which one" argument dissolves. A carrier edge does not need voicemail; a ten-person office does not need 5,000 call setups a second. The rest of this page goes layer by layer, then puts everything into one table.

Asterisk: the application framework most PBXs are built on

What it is. The project describes Asterisk as "an open source framework for building communications applications" that "powers IP PBX systems, VoIP gateways, conference servers and other custom solutions", adding that "Asterisk is sponsored by Sangoma".2 It is written in C and is built from loadable modules: channel drivers for each signalling protocol, applications such as voicemail and queues, and the dialplan that ties them together. In RFC 7092 terms it is a media-termination B2BUA: every call through Asterisk is two legs with Asterisk in the middle.

Licence. The repository's LICENSE file says: "Asterisk is distributed under the GNU General Public License version 2 and is also available under alternative licenses negotiated directly with Sangoma US Inc."1 Two further clauses in that file matter commercially. Sangoma states that it "holds copyright and/or sufficient licenses to all components of the Asterisk package", which is what lets it offer the dual licence. And on its control protocols (AMI, AGI and ARI), the file says "It is our belief that applications using these protocols to manage or control an Asterisk instance do not have to be licensed under the GPL", backed by an explicit grant if a court ever disagrees.1 That is the clause that makes it safe to build proprietary software that drives Asterisk from outside.

Steward and governance. Sangoma: it holds the copyright, owns the trademark ("The 'Asterisk' name and logos are trademarks owned by Sangoma US Inc"1) and sponsors the project.2 Development happens in the open on GitHub, and outside contributors appear in the commit history (see the activity figures in the table below); Sangoma's copyright position is what lets it sell the alternative licence.

Release cadence. Asterisk has the most formal release policy of anything on this page. "New major releases come roughly once a year, alternating between Standard and LTS", an LTS release is fully supported for four years plus one year of security-only fixes, and a Standard release gets at least one year of full support plus one of security fixes.3 The calendar is fixed: branch on the second Thursday of August, first release candidate on the second Thursday of September, release on the third Thursday of October.3 Its version table, as of today:3

Scroll the table sideways to see every column.

Two details in that table are worth noticing. Asterisk 21 got two years of full support because of what the project itself calls "a mistake in scheduling".3 And Asterisk 24, due on 15 October, had not shipped by today: the GitHub releases page shows 24.0.0-rc3 published on 22 October 2026, alongside 23.6.0-rc3, 22.12.0-rc3 and 20.22.0-rc3.4 The latest final releases are 23.5.0, 22.11.0 and 20.21.0 (27 August 2026), and the certified branches certified-22.8-cert5 and certified-20.7-cert12 (19 October 2026).4 A slip of a week or two on an annual release is normal; we mention it only because "Asterisk 24 is out" is the kind of claim that gets written from a calendar rather than a releases page.

Programmability. ARI is the modern interface: "an asynchronous API that allows developers to build communications applications", working "by exposing the raw primitive objects in Asterisk - channels, bridges, endpoints, media, etc." over REST and a WebSocket.5 The documentation is candid that the older AMI and AGI interfaces were not "designed to let a developer build their own custom communications application"; ARI exists to fill that gap.5

FreeSWITCH: the media engine, and what changed under SignalWire

What it is. The repository describes FreeSWITCH as "a Software Defined Telecom Stack enabling the digital transformation from proprietary telecom switches to a versatile software implementation that runs on any commodity hardware."7 Like Asterisk it is a C media-termination B2BUA, but its design centre is different: a modular core built to be driven from outside over the Event Socket, "a TCP-based interface that exposes FreeSWITCH call control" in two modes, "inbound, where an external client connects to FreeSWITCH" and "outbound, where FreeSWITCH connects to an external process for a specific call leg."11 A connected client can "Send DTMF, play audio, and control media on active sessions".11 That control model is why FreeSWITCH is the media layer underneath both FusionPBX and Kazoo (below).

Licence. The LICENSE file in the repository is a Debian-format copyright file whose default stanza (Files: *) reads License: MPL-1.1, the Mozilla Public License version 1.1.6 MPL is a file-level copyleft: changes to FreeSWITCH's own files must be shared if you distribute them, but your separate modules and applications need not be. That is materially more permissive for a product company than Asterisk's GPLv2. SignalWire's FreeSWITCH page says the same in plain words: "FreeSWITCH is open source under the MPL license."9

Steward. SignalWire. The README says "SignalWire is the primary sponsor of the FreeSWITCH project and was founded by the original developers of FreeSWITCH",7 and the documentation describes FreeSWITCH as "Maintained by SignalWire".12

What changed, from SignalWire's own statements. There are a lot of second-hand accounts of FreeSWITCH "going closed". Here is only what SignalWire and the repository say:

• The source code is still MPL 1.1 and public on GitHub.6 SignalWire's page says: "The open-source engine stays free and community-maintained."9
• Binary packages need a SignalWire credential. The repository's packaging instructions state: "A SignalWire Personal Access Token (PAT) or an API Token (for enterprise customers) is required to access FreeSWITCH install packages."8 SignalWire's page puts it as "A free SignalWire account is required to access FreeSWITCH downloads and packages."9 The source itself is on GitHub, and the README documents building it from source.7
• There are now two editions. The documentation says "FreeSWITCH ships in two editions": the open-source edition and "FreeSWITCH Enterprise", "the commercially supported edition from SignalWire, with professional support and services".12 SignalWire's product page calls the paid self-hosted edition "Advantage", "The enterprise branch", with "Commercial modules (G.729A, AMD)".9 The packaging script installs "FreeSWITCH Community or FreeSWITCH Enterprise version ... based on the token provided."8

SignalWire's own pages use three names for the commercial offer (Enterprise, Advantage, and "FreeSWITCH Pro" in the product page FAQ)912 and do not say whether security fixes reach the two editions at different times. We could not verify any statement about patch timing from a first-party source, so we make none.

Release cadence. Irregular, and the gap is the single most important health fact about the project. The GitHub releases page shows v1.10.12 on 3 August 2024, then nothing until v1.11.0 on 7 May 2026, followed by v1.11.1 (26 May), v1.11.2 (9 August) and v1.11.3 (28 August 2026).10 The v1.11.0 notes call it "an important release containing critical security fixes and stability improvements, alongside the migration to PCRE2, the addition of Debian 13 Trixie support, the removal of ~30 legacy modules".10 The PCRE2 change carries the project's own warning: "Be aware of breaking changes."10 If you run FreeSWITCH on 1.10.x, read the 1.11 notes before upgrading; module removals and a regular-expression library change can break existing dialplans.

Kamailio: the SIP server for volume

What it is. The project calls itself "Robust and Performant SIP (RFC3261) Server" with registrar, location, proxy, SIP application and redirect server roles, and gives one concrete figure: "used as load balancer in stateless mode, Kamailio can handle over 5000 call setups per second."14 It is written in C and configured with its own routing script language. It does not run voicemail or menus; when media has to be anchored, it delegates. Its rtpengine module "enables media streams to be proxied via an RTP proxy. The only RTP proxy currently known to work with this module is Sipwise rtpengine."19

The module set is the reason people choose it. The project's 25-year retrospective counts "over 500 contributors" and about 250 modules.15 The dispatcher module alone "offers SIP load balancer functionality and it can be used as SIP traffic dispatcher", with algorithms including "round-robin, weight based load balancing, call load distribution, and hashing over SIP message attributes."18 That one module is how most people put Kamailio in front of a pool of Asterisk or FreeSWITCH servers.

Licence. GPL version 2 or later. The COPYING file opens "The GPL applies to this copy of Kamailio software (sip-router)", warns that "GPL mandates all plug-ins developed for the Kamailio software are to be released under GPL license" or a compatible one, and then reproduces GPL version 2; source files carry SPDX-License-Identifier: GPL-2.0-or-later.13 The plug-in clause matters only if you distribute modified binaries; running Kamailio as a service does not trigger it.

Governance and steward. Kamailio is a community project with no single owning company. Its own history: SIP Express Router (SER) was "started in 2001 by Fraunhofer Fokus Research Institute and released under GPLv2 in 2002"; "OpenSER project spawned from SER" in 2005; it "had to change the name to Kamailio in summer of 2008 due to trademark claims"; and in autumn 2008 the Kamailio and SER teams "decided to join back the projects and use further the Kamailio name."15 The copyright footer on its site reads "The Kamailio SIP Server Project".16 The nearest thing to a commercial steward is asipto, which advertises "direct interaction with co-founders and core developers of Kamailio SIP Server project" and sells consultancy, support, training and development.20 Nothing we found on the project's own site gives any company control of it.

Release cadence. Roughly one major release a year, with point releases on the two newest branches. Version 6.0.0 was tagged in January 202517 and 6.1.0 was announced on 18 February 2026.16 The project's news page lists 6.1.4 (20 August 2026), 6.0.8 (16 September 2026) and 6.1.5 (15 October 2026) in the last three months.16 Its written release policy (major releases every six to eight months, "the last two major releases are maintained") is a 2012 wiki page,17 and the actual pace has since slowed to about a year between majors, so read it as intent rather than a contract.

OpenSIPS: the SIP server that also does B2BUA

What it is. "A mature Open Source implementation of a SIP server", which the project insists "is more than a SIP proxy/router as it includes application-level functionalities"; its feature list covers registrar, router/proxy, redirect server, back-to-back user agent, load balancer/dispatcher, NAT traversal and application server.22 The B2BUA support is real code, not marketing: the source tree ships b2b_entities, b2b_logic and b2b_sdp_demux modules next to dispatcher, load_balancer, topology_hiding and rtpengine.21 Like Kamailio it is C with a routing script, and like Kamailio it relays media through an external engine rather than handling audio itself.

Licence. GPL version 2 or later. The COPYING file says "The GPL applies to this copy of OpenSIPS (Open SIP Server) software", carries the same plug-ins-must-be-GPL note as Kamailio, records that "OpenSIPS software is derived from SER (SIP Express Router) project developed by FhG FOKUS", and adds an explicit exception permitting linking with OpenSSL.21

Steward. OpenSIPS Solutions, which describes its business as "Designing and deploying professional VoIP platforms and solutions based on the OpenSIPS project" and sells products built on it, such as an MS Teams SBC.25 The project site itself does not describe a formal governance structure2224, so we describe OpenSIPS Solutions as the commercial steward on the strength of its own description, not a formal role.

Release cadence. The most predictable of the SIP servers: "the major release cycle is 1 year", with the beta usually in the second half of March and the stable release around the beginning of May. Standard releases are supported for 1.5 years and LTS releases for three years.24 The project's version page lists 4.0.x (released 23 June 2026, supported until December 2027) and 3.6.x LTS (released 23 July 2025, supported until 21 May 2028) as the supported stable releases, with 4.1 in development and a beta planned for May 2027.23 Note that 4.0 is not labelled LTS, so anyone wanting three years of fixes should be on 3.6 today. The latest tags, 4.0.2 and 3.6.9, were both cut on 8 September 2026.21

Kamailio and OpenSIPS share an ancestor and much of their configuration style. We think choosing between them is mostly a matter of which module set, documentation and community fit your team; both are built and marketed for carrier-scale use.1422

Kazoo (2600Hz): a platform whose open core has gone quiet

What it is. The README calls Kazoo "an API-based platform that lets you use your existing phones, programming languages and IT skills".26 It is written in Erlang29 and it is a platform rather than a single engine: 2600Hz publishes configuration repositories for each component it orchestrates, named kazoo-configs-freeswitch, kazoo-configs-kamailio, kazoo-configs-couchdb, kazoo-configs-rabbitmq and kazoo-configs-haproxy,30 which tells you the architecture: Kamailio at the edge, FreeSWITCH for media, CouchDB for data, RabbitMQ between the Erlang applications. The Monster UI web front end is a separate repository.30

Licence. The repository is inconsistent, so we report both. The LICENSE file is the Mozilla Public License version 2.0; the README says Kazoo is released under the "OSI-approved MPL 1.1 open source software license".26 The file in the repository is the one that governs the code as distributed, and it says MPL 2.0.

Steward. 2600Hz, which today sells "White Label Communications" (UCaaS, CPaaS and contact centre) that "can be deployed in 2600Hz data centers or in your own".31 Its homepage does not mention open source.31

Release cadence and health. This is where Kazoo differs sharply from everything else on this page:

• The newest release on GitHub is 4.3.142, published 19 January 2022.28
• The last commit to the default master branch was on 12 March 2024, and the last commit to the 4.3 branch on 11 February 2025.26 There were no commits in the six months to today.
• The project's own announcements file says: "Development of KAZOO 5.x is currently closed to public commits as we prepare the infrastructure necessary for the app exchange."27
• Meanwhile the surrounding repositories are busy: the configuration repositories carry branches 5.0 to 5.5 and were updated today, and Monster UI received commits this month.30

Our reading of those facts: a 5.x line exists and 2600Hz is actively packaging it, but its core source is not in the public repository. Kazoo 4.3 remains open, and its installation guide says it "requires Erlang version 19 specifically 19.3.x".29 Our view: we would not start a new self-hosted deployment on the public Kazoo code in 2026 unless you have a relationship with 2600Hz or the in-house Erlang skill to maintain a fork.

FreePBX and FusionPBX: the GUI distributions

Neither of these is an engine. Each is a web administration layer that writes the engine's configuration for you, plus an installer. They are what most small-business deployments actually run.

FreePBX runs on Asterisk. Its get-started page walks you through bringing up "a FreePBX 17 system": install Debian 12 (at least 2 GB of RAM and 20 GB of disk), run the GitHub-hosted install script, then finish setup in a browser.35 The site describes FreePBX as "the world's most popular open source IP PBX" and is open about the commercial layer: "commercial modules from Sangoma and other add-ons".36 It states that "FreePBX is tested and maintained by Sangoma's Quality Assurance and development teams" and "has zero restrictions on integrated features, such as the number of extensions, users, queues and IVRs".35 The core framework module is licensed GPL version 3 (its LICENSE file); FreePBX is a collection of modules, each with its own licence, so check the module you depend on rather than assuming.37 Version state: the get-started page installs FreePBX 17, and the framework repository's newest tags are release/17.0.33.4 (26 October 2026) and release/18.0.2 (22 September 2026).37 We could not find a first-party announcement of FreePBX 18 as the recommended release, so treat 17 as current.

FusionPBX runs on FreeSWITCH. Its documentation describes it as providing "a customizable and flexible web interface to the very powerful and highly scalable multi-platform voice switch" that "provides a multi-tenant interface", optimised for Debian and Ubuntu.33 The repository describes it as "A full-featured domain based multi-tenant PBX and voice switch for FreeSwitch".32 That multi-tenancy is the main difference from FreePBX: one FusionPBX server can host many separate companies, each on its own domain. There is no LICENSE file at the repository root (GitHub reports no licence), but the source files carry the header "FusionPBX Version: MPL 1.1".32 The most recent release is 5.6.5 (21 September 2026), following 5.5.7 in January 2026 and 5.4.4 in July 2025,34 so roughly one minor line a year with active commits between.

rtpengine: the media relay nobody talks about

Every proxy-centred design eventually has to deal with media that cannot flow directly between endpoints: NAT, IPv4 to IPv6, WebRTC on one side and plain RTP on the other. rtpengine is the usual answer. Its README: "The Sipwise NGCP rtpengine is a proxy for RTP traffic and other UDP based media traffic. It's meant to be used with the Kamailio SIP proxy and forms a drop-in replacement for any of the other available RTP and media proxies."38 The features that make it the default are "In-kernel packet forwarding for low-latency and low-CPU performance" with "Automatic fallback to normal userspace operation if kernel module is unavailable", bridging "between IPv4 and IPv6 user agents", and, through the control protocol, "Full SDP parsing and rewriting" and ICE support.38 "Currently the only supported platform is GNU/Linux."38 Both Kamailio and OpenSIPS ship an rtpengine control module.1921

Licence and steward. GPL version 3 (the repository's LICENSE file); developed by Sipwise as part of its NGCP platform.38 Cadence: very frequent; the releases page shows maintenance releases on several branches within a single week, for example mr26.3.1.1 on 19 October 2026 and mr26.2.1.5 and mr26.0.1.26 on 16 October 2026.39

The master comparison table

Languages are as GitHub reports them for each repository. Every licence below was read from the repository's own licence file or source headers today; every version and date from the project's releases page or version page. Activity is our own measurement through the GitHub API: commits on the default branch in the six months from 26 April to 26 October 2026, and the number of distinct authors of those commits. For FreePBX we measured only the framework module, which understates the project as a whole.

Scroll the table sideways to see every column.

Commit counts are a crude signal. A large count can mean many small commits; a small one can mean careful, batched work. What they reliably show is the difference between a project that is moving and one that is not: every row is moving except Kazoo, whose public core has stopped. The author count is the more telling column. Kamailio's commits came from 88 distinct authors and Asterisk's from 54; FreeSWITCH's 137 commits came from 10. That does not make FreeSWITCH unhealthy, but it does mean its direction depends on a small team, and in practice on SignalWire.

How they are deployed together

Three patterns cover most real systems.

1. The single box. One Asterisk server with FreePBX, or one FreeSWITCH server with FusionPBX, registering desk phones and softphones directly and connecting to one or two SIP trunk providers. Nothing else. This is the right shape for a single office and it is where most small-business deployments should stop.

2. Proxy in front of media servers. Kamailio or OpenSIPS takes every registration and every call, authenticates, applies rate limits and topology hiding, then uses dispatcher or load_balancer to hand calls that need media work to a pool of Asterisk or FreeSWITCH servers.1821 The proxy layer holds the state that has to be fast and highly available (registrations, routing); the media servers become stateless, interchangeable workers you can add or drain. rtpengine sits alongside the proxy for calls whose media must be anchored but need no application logic.19 Kazoo is this pattern pre-assembled, with Kamailio at the edge and FreeSWITCH behind it.30

3. The SBC at the carrier edge. Kamailio or OpenSIPS plus rtpengine as a session border controller between your network and carriers or customers: topology hiding so internal addresses never leak, SIP normalisation between vendors' dialects, rate limiting and fraud screening, and media anchoring or IPv4/IPv6 bridging through rtpengine.38 OpenSIPS Solutions sells a packaged SBC on exactly this basis.25 No media server is needed here at all.

How to choose, by use case

Everything in this section is our view. It rests on the layer analysis, the licence terms and the health signals above, and on the general principle that the fewest moving parts that do the job is the right design.

Small-business PBX (one site, under about 50 phones). Our view: FreePBX on Asterisk, installed the way the FreePBX project documents it.35 The reasons are the predictable release calendar,3 four years of full support on an LTS line, the largest pool of administrators and integrators who already know it, and the fact that the GUI covers the day-to-day changes (extensions, ring groups, menus) without touching configuration files. FusionPBX is an equally capable choice if you, or the integrator you hire, prefer FreeSWITCH. Do not put Kamailio in front of a single office; it adds a second system to secure and operate and buys nothing at that size.

Carrier edge, SBC, or high-volume routing. Our view: Kamailio or OpenSIPS with rtpengine, no media server. Pick Kamailio if you value the broadest module set and a community with no single vendor behind it15; pick OpenSIPS if you want B2BUA behaviour inside the SIP server itself21 or a vendor with a packaged SBC behind the project.25 Both are GPL; neither licence affects you unless you distribute modified binaries.

Multi-tenant hosted PBX (you sell phone service to many small companies). Our view: FusionPBX on FreeSWITCH for a first deployment, because multi-tenancy is built into its data model33 rather than added on; then put Kamailio in front as you outgrow one server. Kazoo was built for exactly this and is the most complete design on paper, but with no public core release since January 202228 and 5.x "closed to public commits"27, we would only choose it as a commercial relationship with 2600Hz, not as an open-source project. A hosted business also carries regulatory obligations (in the United States, 911 and dispatchable location; see our emergency calling page) that no software choice discharges for you. Canadian operators have their own CRTC obligations; we have not summarised them here.

Programmable media (IVR, voice bots, recording, transcription). This is the closest call on the page. FreeSWITCH's Event Socket11 and MPL licence6 make it the natural engine for a product company that will write proprietary modules or drive calls from its own service. Asterisk's ARI5 is the cleaner API for an application developer, and Sangoma's written position that ARI and AMI clients are not derivative works1 removes the GPL worry for code that talks to Asterisk over those interfaces. Our view: if you will extend the engine itself, prefer FreeSWITCH, accepting that official binary packages now require a SignalWire token8 and that its release cadence has been uneven10; if you will only control calls from outside, prefer Asterisk and ARI for the calendar and the support windows.

Where a hosted auto attendant fits

If what you need is a menu that answers your business number and sends callers to a few mobiles, all of the above is more system than the job requires: a server to patch, a SIP trunk to buy, and a dialplan to maintain. That narrower job is what our service does; nothing on this page describes it. If you want to own your telephony, or you need queues, desk phones and recording, the open-source stack is the better choice, and FreePBX or FusionPBX is where we would start.