This is a working directory of open source software that carries real voice traffic. A version of it first went up here in 2011, it has been cited since as a reference on open source in the ICT industry, and by this year most of it was out of date. So we rebuilt it. Every project below was rechecked on 6 September 2026 against its own public repository, and anything that had stopped moving was either marked dormant or moved to the retired section at the bottom.
We kept the honest version rather than the flattering one. A directory that lists dead projects as if they were current sends people down weeks of wasted work, and we have been on the receiving end of that ourselves.
How to read this
Each entry gives the date of the most recent public commit at the time of checking. That’s a blunt measure and it isn’t the same as project health, but it does separate software somebody is working on from software nobody is, which is the first thing you need to know before you build on anything.
Licences vary a lot across this list and several have changed over the years. Check the licence yourself against what you actually intend to do, especially if you plan to redistribute commercially. Nothing here is a recommendation for your particular situation.
Switching and call control
These are the engines. Almost everything else in a voice stack ends up sitting on one of them.
| Project | What it does | Last commit |
|---|---|---|
| Asterisk | The open source PBX and telephony toolkit a large part of this industry was built on. Channel drivers, a dialplan, and the ARI and AGI interfaces for driving calls from your own code. | 2 Sep 2026 |
| FreeSWITCH | Softswitch built for scale and for media work. Stronger than Asterisk at transcoding and conferencing, and happier being embedded in a bigger system than driven by a dialplan. | 4 Sep 2026 |
| Yate | Telephony engine covering SIP, H.323 and several mobile core protocols. Used well outside normal PBX territory, including GSM and satellite deployments. | 18 Jun 2026 |
| SEMS | SIP Express Media Server. Application server and B2BUA that pairs naturally with Kamailio for announcements, conferencing and prepaid logic. | 21 Aug 2026 |
PBX distributions and admin interfaces
If you want a PBX with a web interface rather than a dialplan to write, you want one of these rather than raw Asterisk.
| Project | What it does | Last commit |
|---|---|---|
| FreePBX | The long-running web administration layer over Asterisk, and the usual first stop for anyone who wants a working PBX this afternoon. | 26 Aug 2026 |
| Issabel | The community distribution that carried on after Elastix went commercial. Asterisk based, with the same all-in-one unified communications shape. | 11 Aug 2026 |
| Wazo Platform | Asterisk based unified communications platform built API-first, aimed at people constructing a service rather than administering a box. | 4 Aug 2026 |
| ICTPBX Community Edition (ours) | Multi tenant PBX built on ICTCore and FreeSWITCH, with a separate Angular interface. Released under MPL-2.0. | 4 Sep 2026 |
SIP proxies, routing and session border control
Handling signalling at carrier volume is a different job from running a PBX, and mixing the two up is one of the more expensive architecture mistakes in this field.
| Project | What it does | Last commit |
|---|---|---|
| Kamailio | SIP proxy, registrar and router built for very high signalling throughput. Modular and scriptable, and the standard answer when a PBX is the wrong shape. | 4 Sep 2026 |
| OpenSIPS | The other branch of the same family tree, with its own module set and a strong focus on session border control, load balancing and provider-side routing. | 4 Sep 2026 |
| Routr | Cloud native SIP proxy and registrar designed to be configured declaratively and run in Kubernetes, which is a genuinely different approach from the two above. | 5 Sep 2026 |
| Flexisip | SIP proxy from the Linphone team, with presence, push notification handling and conference features aimed at mobile deployments. | 4 Sep 2026 |
| Yeti | Open source class 4 softswitch for wholesale voice, covering routing, rating and the operational side of interconnect traffic. | 5 Sep 2026 |
| GNU Gatekeeper | H.323 gatekeeper doing address translation, admission control, routing and accounting. H.323 is a smaller world now, but this one kept going where its contemporaries did not. | 25 Aug 2026 |
Media servers, RTP and WebRTC
The part that carries actual audio. This category barely existed when the first version of this page was written, and it’s now where most of the new work happens.
| Project | What it does | Last commit |
|---|---|---|
| rtpengine | Kernel assisted RTP proxy and media relay, normally paired with Kamailio or OpenSIPS. Handles NAT traversal, recording and transcoding at volume. | 4 Sep 2026 |
| Janus | General purpose WebRTC server built as a set of plugins, so it becomes a gateway, an SFU, a recorder or a bridge into SIP depending on what you load. | 4 Sep 2026 |
| mediasoup | WebRTC selective forwarding unit shipped as a library rather than an application, which suits teams building their own product instead of deploying someone else’s. | 4 Sep 2026 |
| LiveKit | WebRTC infrastructure with SDKs, an SFU and an agents framework. It has become the default choice for new real-time voice and video applications, including AI ones. | 5 Sep 2026 |
| Jitsi Meet | Complete open source video conferencing that you can self host, with the largest community of anything in this category. | 5 Sep 2026 |
| coturn | The STUN and TURN server nearly every WebRTC deployment ends up running. Unglamorous, and the thing that makes calls work behind real networks. | 6 Sep 2026 |
| SpanDSP | DSP library covering fax modulation, tone handling and echo cancellation. Most open source T.38 faxing traces back to this. | 31 Jul 2026 |
SIP stacks and libraries
If you’re building rather than deploying, this is the layer you link against.
| Project | What it does | Last commit |
|---|---|---|
| PJSIP | Portable C stack covering SIP, SDP and media. Small enough for embedded work, complete enough for desktop and mobile clients, and the SIP channel driver behind modern Asterisk. | 4 Sep 2026 |
| Sofia-SIP | The C SIP stack FreeSWITCH uses, maintained under that project since the original Nokia release. | 3 Aug 2026 |
| reSIProcate | Standards focused C++ SIP stack with a long record in products that need protocol correctness ahead of convenience. | 2 Sep 2026 |
| baresip and libre | Modular SIP user agent and the portable protocol library underneath it. Useful both as a scriptable endpoint and as a base for custom clients. | 4 Sep 2026 |
| drachtio | SIP server controlled from Node.js, for developers who would rather write signalling logic in application code than in a routing script. | 20 Aug 2026 |
| JsSIP | SIP over WebSocket in the browser. If you need a softphone inside a web page, this or the one below is where you start. | 6 May 2026 |
| SIP.js | The other established JavaScript SIP library, with a different API shape and its own following. | 15 Jun 2026 |
| asterisk-audiosocket (ours) | Dependency free implementation of the Asterisk AudioSocket protocol for Node.js and TypeScript, for streaming call audio into your own code. | 10 Aug 2026 |
| asterisk-ami-node (ours) | Dependency free Asterisk Manager Interface client for Node.js, typed, for reading events and issuing actions. | 9 Aug 2026 |
| freeswitch-esl-node (ours) | The same idea for the FreeSWITCH Event Socket, so both switches can be driven from the same kind of Node.js service. | 9 Aug 2026 |
Testing, capture and monitoring
Underrated, and the difference between a platform you can operate and one you can only hope about.
| Project | What it does | Last commit |
|---|---|---|
| SIPp | Traffic generator and test tool for SIP. Scenario driven, and still the standard way to load test a signalling path before your customers do it for you. | 24 Aug 2026 |
| HOMER | SIP capture and correlation across a whole estate, so you can reconstruct one call across every hop it took instead of guessing. | 4 Sep 2026 |
| sngrep | Terminal SIP capture with call flow diagrams. The fastest way to see what is happening on a box you are logged into right now. | 17 Aug 2026 |
Softphones and endpoints
The client side thinned out badly over the past decade. Several well known open source softphones simply stopped, and this is what is left with real development behind it.
| Project | What it does | Last commit |
|---|---|---|
| Linphone | SIP client across desktop and mobile built on the Belledonne stack, with encrypted voice and video and a library you can embed in your own app. | 4 Sep 2026 |
| Jami | Distributed communication platform with no central server. This is what SFLphone became, so if you followed an old link to SFLphone, this is where it went. | 5 Sep 2026 |
| Twinkle | Long-running Linux SIP softphone with encrypted media. Development is slow rather than stopped. | 6 Apr 2026 |
Contact centre, dialling and fax
The application layer above the switch. Our own projects are in here, and we have labelled them rather than slipping them in as neutral picks.
| Project | What it does | Last commit |
|---|---|---|
| VICIdial | Web based inbound and outbound contact centre suite over Asterisk, with predictive dialling. Very widely deployed and still commercially supported. | actively released |
| GOautodial | Contact centre distribution in the same territory, packaged with its own interface and API. | 28 Aug 2026 |
| ICTDialer (ours) | Open source auto dialler for voice, fax and messaging campaigns, with a drag and drop IVR designer. The open source edition runs on FreeSWITCH. | 12 Aug 2026 |
| ICTFax (ours) | Multi tenant fax server covering T.38 origination and termination, built on FreeSWITCH and ICTCore. | 31 Aug 2026 |
| ICTCore (ours) | The unified communications framework and REST API underneath the two above, so voice and fax workflows can be driven from application code. | 31 Aug 2026 |
| HylaFAX | The original Unix fax server. Old, and still in service in a lot of places that need PSTN faxing to keep working. | legacy, still deployed |
Billing, rating and mediation
Where a lot of open source telecom projects quietly go to die, so check the activity dates in this section particularly carefully.
| Project | What it does | Last commit |
|---|---|---|
| CGRateS | Real time rating, charging and mediation engine. The most actively developed option here by a distance, and built for carrier scale. | 5 Sep 2026 |
| ASTPP | Open source VoIP billing and class 4 and 5 softswitch built on FreeSWITCH, covering wholesale and retail cases. | 26 Jun 2026 |
| MagnusBilling | Billing and call management for Asterisk, popular with smaller providers and resellers. | 2 Sep 2026 |
| A2Billing | Calling card platform and real time billing engine for Asterisk. Still running in places, but the public repository has been quiet since early 2023. Treat it as dormant. | 30 Jan 2023 |
Voice AI building blocks
New since the last version of this page, and the most fast-moving section on it. Everything here is self hostable, which is the reason it matters for telecom: call audio often cannot leave your infrastructure.
| Project | What it does | Last commit |
|---|---|---|
| Whisper | Speech recognition models that changed what is affordable in this space. Accurate across many languages, and heavy to run. | 31 Aug 2026 |
| whisper.cpp | The port that made Whisper practical on ordinary CPUs, which is what most people running it next to a PBX actually use. | 4 Sep 2026 |
| Vosk | Lightweight offline speech recognition with streaming support and small models, which suits telephony audio and constrained hardware. | 9 Aug 2026 |
| Piper | Fast local neural text to speech, good enough for phone calls and light enough to run per-channel. Note the project moved: the old Rhasspy repository is archived and this is the maintained one. | 4 Sep 2026 |
| Kaldi | The speech recognition toolkit that most of this field learned on. Still referenced constantly, though new work rarely starts here now. | 22 Sep 2025 |
| asterisk-ai-voice-agent (ours) | Self hosted AI voice agent for Asterisk. Streaming speech to text into a model and back out as speech over AudioSocket or the newer WebSocket channel, with barge-in handling. | 5 Sep 2026 |
| piper-tts-server (ours) | Piper synthesis served at the pace a call needs rather than as fast as it renders, which is the difference between audio a caller hears and audio a jitter buffer discards. | 5 Sep 2026 |
| tvbench (ours) | Measures what a caller actually hears from a voice agent: frame pacing, holes in speech, and how long the agent keeps talking after being interrupted. | 6 Sep 2026 |
| pbx-mcp (ours) | Model Context Protocol server for Asterisk and FreeSWITCH, so an AI assistant can inspect channels, registrations and call state directly. | 2 Sep 2026 |
The four entries above marked as ours are the pieces we needed for telephony work and could not find. The rest of what we release is on our open source projects page, and the reasoning behind the benchmark is written up here.
Retired, renamed or gone commercial
These were on earlier versions of this page. They’re kept here so anyone arriving from an old link finds out what happened in a few seconds rather than a few days.
- sipXecs stopped. The last public commit was December 2021 and the project site no longer responds.
- Elastix is gone as an open source distribution and the domain no longer resolves. The community carried on as Issabel, which is listed above. If you’re still running an Elastix box, plan the move.
- SFLphone wasn’t abandoned, it was renamed. It became Jami.
- QuteCom, CSipSimple and Sipdroid all stopped. Current mobile clients are generally built on PJSIP or the Linphone stack instead.
- PBX in a Flash and Druid are no longer going concerns.
- Plivo is very much alive but it’s a commercial cloud service now, not software you deploy yourself.
- Coqui TTS is worth knowing about because the models are still good and widely used, but the company behind it closed and the repository has been quiet since August 2024. Fork it rather than depend on it.
- Freeside is dormant rather than dead, last touched in July 2024. Check its state carefully before building billing on it.
- Kazoo from 2600Hz has slowed considerably, with its last public commit in February 2025. Still substantial software, but not moving quickly.
- Rhasspy is archived in its original form. Its voice components were split out, and Piper is the piece most people came for.
Why we keep this page current
ICT Innovations builds on this ecosystem and contributes back to it, so keeping an accurate picture of what’s alive is work we’d be doing anyway. Publishing it costs us nothing and saves other people the same round of checking.
If something here is wrong, out of date, or missing a project that genuinely belongs, tell us through the Contact Us page and we’ll verify it and fix the entry. Corrections that arrive with a repository link get dealt with fastest, because that’s the part we have to check anyway.