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Asterisk vs FreeSWITCH vs Kamailio vs OpenSIPS
These four are the pillars of open-source VoIP, but they don't do the same job. The single biggest distinction: Asterisk and FreeSWITCH are media servers (they touch the audio), while Kamailio and OpenSIPS are SIP proxies (they route signalling and usually don't touch the audio). Get that right and everything else falls into place.
Media servers vs SIP proxies
A media server / B2BUA sits in the call. It answers, plays audio, records, runs IVR, mixes conferences and transcodes between codecs. That power costs CPU per call, so a single box handles hundreds to low-thousands of concurrent calls.
A SIP proxy only routes the SIP messages that set up calls — it decides where a call goes, then (typically) lets the audio flow directly between endpoints. It barely touches each call, so one instance can route tens of thousands of calls per second. It does not play audio, record or run IVR on its own.
Asterisk
The most widely used open-source PBX. Batteries-included: IVR, voicemail, queues, conferencing, call recording and a huge module ecosystem, plus friendly interfaces like FreePBX. Easiest place to build features fast. Best for PBXs, call centers and IVR up to a few thousand concurrent calls per server. Most commercial contact-center stacks (including the engine behind Qallixo) build on Asterisk.
FreeSWITCH
Also a media server, engineered from the start for high concurrency, clean multi-tenancy and heavy media work (transcoding, conferencing, video). Steeper learning curve than Asterisk but scales further per box and is a favourite for large carriers and CPaaS platforms. FusionPBX is the popular web GUI on top of it.
Kamailio
A battle-tested SIP proxy/router built for massive scale — registrars, load balancers, SBC front-ends and carrier signalling handling tens of thousands of calls per second. It routes; it doesn't record or play audio. To add media (recording, transcoding, NAT traversal) you pair it with a media relay such as RTPengine.
OpenSIPS
The other major open-source SIP proxy — a close sibling of Kamailio (they share a common ancestor). Same core role: high-throughput SIP routing, load balancing and SBC duties, with a rich scripting language and its own module set. Choosing between Kamailio and OpenSIPS usually comes down to team familiarity and specific module needs rather than raw capability.
Where RTPengine fits
Because Kamailio and OpenSIPS don't handle media, they're commonly deployed with RTPengine — a fast media proxy that relays RTP audio, handles NAT and can transcode or record. A typical carrier design is: Kamailio/OpenSIPS for signalling at the edge + RTPengine for media + Asterisk/FreeSWITCH behind them for IVR and application logic.
Quick chooser
- Need IVR, recording, queues, a PBX → Asterisk (or FreeSWITCH for heavier media/scale).
- Need to route/balance huge SIP volume or build an SBC → Kamailio or OpenSIPS.
- Need a media relay in front of proxies → RTPengine.
- Building a serious carrier/CPaaS → often all of them together, each doing its job.
Or skip the plumbing entirely
All four are powerful — and all four mean you provision servers, secure SIP, tune media and stay on top of updates. Qallixo gives you the finished contact-center product built on this stack, so you get dialers, IVR, recording and WhatsApp without running any of it. If that trade-off interests you, see the Vicidial alternative comparison.
Want to install them yourself?
We've written step-by-step Ubuntu guides for each: