Kamailio vs Asterisk: When to Use Each

Introduction¶

Both are open-source SIP platforms, but they solve different problems. Choosing wrong means either fighting Asterisk to scale, or trying to make Kamailio play audio prompts.

What each is¶

Asterisk — B2BUA and media server¶

• A Back-to-Back User Agent: it terminates the incoming call leg and originates a new outgoing leg. It's in every call.
• Handles media: plays prompts, records, does IVR, conferencing, voicemail, DTMF detection, transcoding, music on hold.
• A full PBX: dialplan, queues, features (transfer, park, pickup).
• Stateful and resource-heavy per call — RTP, codecs, channels. One box handles hundreds to low thousands of concurrent calls.

Kamailio — SIP proxy/router/registrar¶

• A SIP proxy: it forwards SIP messages, it is not a call party. It stays in the signalling path (Record-Route) but never touches RTP.
• Registrar / location service, load balancer, SBC-like functions (NAT traversal help, topology hiding, rate limiting, auth).
• Programmable routing in its own config language (kamailio.cfg) or via KEMI (Lua/Python/JS).
• Extremely scalable — tens of thousands of registrations, thousands of calls per second, because it only shuffles small text messages.

The canonical architecture¶

internet
   |
[ Kamailio ]  <- SIP edge: registration, auth, NAT handling, load balancing,
   |             rate limiting, dispatcher to the media tier
   +--> [ Asterisk #1 ]  <- media: IVR, queues, voicemail, conferencing
   +--> [ Asterisk #2 ]
   +--> [ Asterisk #N ]
   |
[ RTPengine ]  <- media relay for NAT / recording (driven by Kamailio)

Kamailio is the front door: it authenticates users, keeps the registration table, protects against floods, and dispatches calls across a pool of Asterisk boxes (dispatcher module + health checks). Asterisk does the call-processing work. RTPengine (Kamailio-controlled) relays media when NAT requires it.

This gives you Kamailio's scale and resilience at the edge with Asterisk's features behind it — and you can add Asterisk capacity horizontally.

Choosing for a project¶

FreeSWITCH is the other media server — more built for scale and concurrency than Asterisk, steeper learning curve; often paired with Kamailio the same way.

Verification and troubleshooting¶

# Kamailio
kamctl ul show                       # registered contacts
kamcmd dispatcher.list               # media-tier targets + health
kamcmd core.shmmem                    # shared memory (tune -m / -M)
sngrep                               # ladder diagrams across both

# Asterisk
asterisk -rx "pjsip show endpoints"
asterisk -rx "core show channels"

• Calls register on Kamailio but fail to connect — the dispatcher has no healthy Asterisk targets, or the Record-Route/loose-routing isn't set so in-dialog requests bypass Kamailio and hit NAT.
• One-way / no audio — RTPengine not engaged (rtpengine_manage() not called in the right routes), or Asterisk's external_media_address wrong. Media is the usual culprit, not signalling.
• Kamailio out of memory under load — shared memory (-m) or package memory (-M) too small; transactions/dialogs accumulating (TM/dialog timers).
• Asterisk box maxed at a few hundred calls — that's expected; add boxes and let Kamailio's dispatcher spread the load. If one call = 100% of a core, it's transcoding — align codecs end to end.
• Trying to do an IVR in Kamailio — wrong tool. Route the call to Asterisk (or an rtpengine + media server) for anything that plays or records audio.
• Registration storms after a restart — Kamailio pike/htable rate limiting, and randomised re-REGISTER timers on the clients.

Related tools and reading¶

• On-site: Port Checker (5060), Ping test.
• Related posts: RTPengine for NAT and transcoding, Your first Asterisk dialplan.