Open-source VoIP: Kazoo, OpenSIPS and Kamailio

The guides

Kazoo from its docs and source: Erlang architecture, account tree, Crossbar API, an auto attendant in JSON, and where the open-source project stands in 2026.

Open-source VoIP · 24 min read

OpenSIPS in depth: the SIP server, its modules and a working load balancer.

OpenSIPS from its own docs and source: the SER lineage, the script and key modules, 3.6 LTS against 4.0, and a dispatcher plus rtpengine load balancer.

Open-source VoIP · 30 min read

Kamailio in depth: the SIP server that routes, and does nothing else.

How Kamailio works: SER lineage, processes and memory, kamailio.cfg routes, KEMI, key modules, current releases, and a registrar plus dispatcher config.

The layers, in one paragraph

A SIP proxy or registrar (Kamailio, OpenSIPS) routes signalling and scales to very large numbers of endpoints without touching audio. A media server or PBX (Asterisk, FreeSWITCH) answers calls, plays prompts, records and bridges. A platform (Kazoo by 2600Hz) combines these into a multi-tenant service with an API, accounts and billing hooks. Knowing which layer a project occupies settles most "which should I use" questions.