Asterisk and FreeSWITCH are the two dominant open-source soft switches that power the majority of the world's self-hosted VoIP infrastructure. After 12+ years and 250+ projects on both platforms, here is our definitive 2026 comparison — including how each integrates with the new generation of AI voice agents.
Architecture: The Fundamental Difference
Asterisk is a telephony application server — it processes calls through a flexible dialplan (extensions.conf or AEL) and executes applications. FreeSWITCH is an event-driven soft switch with a more modular, XML-based configuration. Asterisk feels like programming; FreeSWITCH feels like configuring a system.
| Characteristic | Asterisk | FreeSWITCH |
|---|---|---|
| Core design | Telephony application server | Carrier-grade soft switch |
| Configuration | INI/dialplan (extensions.conf) | XML (directory, dialplan, Sofia config) |
| Primary API | AMI (Manager), ARI (REST), AGI | ESL (Event Socket Library) |
| Language support | C extensions, AGI (any language) | Lua, JavaScript, Python via ESL |
| Channel model | Legacy + PJSIP (dual stack) | Sofia SIP (single, modern stack) |
| WebRTC | Via http_websocket (PJSIP) | Via Sofia SIP + verto module |
| First release | 1999 | 2006 |
| Licence | GPLv2 | MPL 1.1 |
Call Capacity & Scalability
On equivalent hardware (8-core, 32GB RAM), FreeSWITCH consistently outperforms Asterisk in raw concurrent call capacity. FreeSWITCH's event-driven, multi-threaded architecture scales more linearly with cores. Asterisk has improved significantly with PJSIP, but FreeSWITCH still holds an edge at the high end.
| Metric | Asterisk (PJSIP) | FreeSWITCH |
|---|---|---|
| Concurrent calls (8-core VPS) | ~1,500–2,000 | ~3,000–5,000 |
| Calls per second (CPS) | ~50–80 | ~150–300 |
| Memory per call | ~2–4 MB | ~1–2 MB |
| Horizontal scaling | Via Kamailio/OpenSIPS load balancer | Native ESL clustering |
Predictive Diallers and Contact Centres
For outbound contact centre applications (predictive diallers, progressive diallers), Asterisk with Vicidial or GoAutoDial is the dominant choice. The ecosystem around Asterisk-based dialler solutions is mature, well-documented, and supported by dozens of vendors. FreeSWITCH-based diallers exist (FusionPBX + custom scripts) but are less common and have a smaller support ecosystem.
AI Voice Agent Integration
This is where the 2026 comparison gets interesting. AI voice agents (LiveKit, VAPI, Pipecat) all speak WebRTC or SIP. Both Asterisk and FreeSWITCH can bridge AI agents to PSTN via SIP, but the approach differs.
| Integration Point | Asterisk | FreeSWITCH |
|---|---|---|
| SIP bridge to LiveKit | PJSIP endpoint → LiveKit SIP connector | Sofia SIP → LiveKit SIP connector |
| WebRTC to PSTN | Asterisk WebRTC (http_websocket) → PJSIP | FreeSWITCH Verto module |
| Media handling | RTPEngine for transcoding at scale | Built-in transcoding + RTPEngine |
| Answering machine detection | AMD application (built-in) | mod_com_amd or custom Lua |
| Call recording + transcription | MixMonitor + Deepgram webhook | mod_audio_stream + Deepgram webhook |
For AI voice agent deployments: if you are running at <500 concurrent AI calls, Asterisk + PJSIP + LiveKit SIP is simpler to set up and maintain. For >1,000 concurrent AI calls, FreeSWITCH's superior throughput and lower memory footprint makes it the better choice.
WebRTC Support
Both platforms support WebRTC in 2026, but FreeSWITCH's Verto module offers a more polished WebRTC experience with a cleaner JSON-RPC API. Asterisk's WebRTC via PJSIP works well but requires careful DTLS/SRTP configuration. For production browser-based softphones, we slightly prefer FreeSWITCH Verto — but both work.
When to Choose Asterisk
- You are building a traditional PBX (IVR, ring groups, voicemail, queues) — Asterisk's dialplan is more intuitive for this use case
- You need a Vicidial-based outbound dialler platform
- Your team already knows Asterisk — the productivity advantage is real
- You need the largest support ecosystem (forums, third-party tools, hardware certifications)
- Budget for sub-1,000 concurrent calls
When to Choose FreeSWITCH
- You need to handle >1,000 concurrent calls on a single node
- You are building a carrier-grade switch or SBC (Session Border Controller)
- Your application requires high calls-per-second (CPS) throughput
- You are building a multi-tenant hosted PBX (FusionPBX + FreeSWITCH scales well for this)
- You want better native WebRTC support via Verto
Both Asterisk and FreeSWITCH are exceptional platforms that will power production VoIP infrastructure for years to come. Asterisk wins on ecosystem, tooling, and ease of use for traditional PBX applications. FreeSWITCH wins on raw throughput, carrier-grade reliability, and modern WebRTC when that capacity is needed. For most new projects in 2026, start with Asterisk unless you specifically know you need FreeSWITCH's scale characteristics. Our VoIP team has 12+ years on both platforms and can help you choose and deploy the right solution.