Kamailio review 2026: open-source SIP server for VoIP and WebRTC platforms. Explore features, architecture, deployment options and top alternatives.
Kamailio is a free, open-source SIP server used to build large-scale VoIP, WebRTC, presence, and instant-messaging platforms. Its origins date to September 2001, when the project began as SIP Express Router (SER) at Fraunhofer FOKUS, a research institute in Berlin, meaning Kamailio's codebase predates the finalized SIP v2.0 specification (RFC 3261) and its early developers helped shape the protocol's real-world extensions.
The project went through several identity changes: a 2005 fork became OpenSER, which was renamed Kamailio in 2008 following a trademark dispute, and later that year the Kamailio and original SER communities merged their codebases into a single unified project under the Kamailio name. Today it is maintained by a distributed community of contributors rather than a single company.
Released under GPLv2, Kamailio is designed for extreme scale, capable of processing thousands of call setups per second, and is used by telecom carriers and VoIP providers worldwide to route what its maintainers describe as billions of minutes of voice traffic every month.
Kamailio's networking layer supports asynchronous TCP, UDP, and SCTP transport with TLS encryption for secure signaling, dual IPv4/IPv6 support, and native WebSocket support for WebRTC-based real-time communication applications.
On the application layer, it provides SIMPLE-based instant messaging and presence with an embedded XCAP server and MSRP relay, IMS extensions for VoLTE deployments, ENUM lookups, DID and least-cost routing, and load balancing with automated failover across SIP endpoints.
Operationally, Kamailio includes built-in accounting, authentication, and authorization modules, integrates with backend systems including MySQL, PostgreSQL, Oracle, Radius, LDAP, Redis, Cassandra, MongoDB, and Memcached, and exposes JSON-RPC/XML-RPC control interfaces plus SNMP monitoring for production operations teams.
Kamailio is entirely free and open source under the GPLv2 license, with no paid tiers, license fees, or vendor lock-in. Anyone can download the source from GitHub, compile it, and deploy it on their own infrastructure without cost.
Because there is no company behind Kamailio, there is no official paid support tier either; organizations that need commercial support typically contract independent consultants or specialist VoIP engineering firms within the Kamailio ecosystem, or rely on community support through the project's mailing lists and IRC/chat channels.
Kamailio is an open-source SIP server used to build large-scale VoIP, WebRTC, presence, and instant-messaging platforms for telecom carriers and service providers.
Yes, Kamailio is completely free and open source under the GPLv2 license with no licensing fees.
Kamailio's history dates to September 2001 as SIP Express Router (SER), later forked into OpenSER in 2005 and renamed Kamailio in 2008.
Kamailio focuses on SIP signaling, proxying, and routing at very high scale, while Asterisk is a broader open-source PBX with stronger built-in media handling and call features.
Kamailio is designed to process thousands of call setups per second, making it suitable for carrier-grade VoIP deployments.
Yes, Kamailio includes native WebSocket support that enables browser-based WebRTC communication applications.