Kamailio (SIP server / softswitch)
Kamailio (formerly OpenSER) is a high-performance SIP server. It is not a PBX — it has no media, no IVR, no voicemail. It is a SIP signaling proxy/router/registrar that handles tens of thousands of calls per second on commodity hardware, and it is the de-facto standard Class 4 softswitch in the open-source world.
What Kamailio does
- SIP proxy / router: takes SIP messages, applies routing logic (in a C-like config language), forwards them.
- Registrar: handles SIP REGISTER from millions of endpoints.
- Load balancer / dispatcher: spreads INVITEs across upstream gateways with weight + probing.
- Authentication: digest auth backed by DB or radius.
- Anti-fraud / rate-limiting: pike, htable, ratelimit modules.
- Topology hiding (B2BUA mode): with mod_b2b_logic.
What it does NOT do: media. It only sees SIP signaling. RTP flows directly between endpoints (unless you put rtpengine in line for media anchoring).
Where Kamailio fits
- In front of Asterisk/FreeSWITCH cluster — Kamailio handles registration + routing, Asterisk handles features. The standard scaling pattern.
- As a wholesale carrier softswitch — routing INVITEs between carriers based on rate decks.
- As a SIP load balancer in front of a cluster.
- As a SIP-WebSocket gateway for browser-based SIP.
Configuration model
Configuration is a single big file (kamailio.cfg) in a C-like syntax with route{} blocks. Logic is procedural: when an INVITE arrives, it goes through the main route block, can branch to sub-routes, can hit modules that mutate the message. Steeper learning curve than Asterisk's dialplan but vastly more flexible.
DIDHub + Kamailio
DIDHub as a Kamailio dispatcher upstream — weighted load balancing, active OPTIONS probing, automatic failover: Kamailio dispatcher tutorial.
Related terms
OpenSIPS (SIP server / fork of OpenSER)
Asterisk (open-source PBX framework)
FreeSWITCH
rtpengine (media proxy / relay)
Class 4 vs Class 5 Switches (Trunking vs PBX)
SBC (Session Border Controller)
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