Kamailio review
A Linux-based SIP routing foundation for teams building self-hosted voice infrastructure.
Reviewed by iTechGuides Editors · Editorial team · Updated Oct 2026
Kamailio is an open-source SIP server for organizations building self-hosted VoIP and real-time communications platforms. It runs on Linux and provides SIP proxying, registration, location, redirection, application-server functions, accounting, authentication, NAT traversal, and WebRTC connectivity. Its audience includes carriers, enterprises, SIP infrastructure teams, and unified-communications builders that need control over routing and deployment rather than a managed voice API with bundled telephony service.
Its strongest fit is an infrastructure-heavy voice stack. Kamailio supports stateless and stateful SIP routing, serial and parallel forking, load balancing, failover, and least-cost routing. Transport coverage includes UDP, TCP, TLS, SCTP, IPv4, IPv6, and WebSocket, while security-related functions include SIP authentication, authorization, TLS, and SRTP handling. RPC interfaces through XML-RPC and JSON-RPC support operational control, and database connectors cover MySQL, PostgreSQL, Redis, MongoDB, and Cassandra. Integrations with PSTN gateways, SIP devices, Asterisk, FreeSWITCH, SEMS, RabbitMQ, NSQ, and SNMP extend its role within communications systems.
Kamailio is a broad SIP routing and control layer, not an end-user telephony bundle. The published product details show a free open-source model, with no paid tier structure to compare. Its scope includes outbound calling and SIP support, but not inbound calling or bundled phone numbers. Call recording is also outside the supplied feature set. Choose Kamailio when your team needs flexible SIP infrastructure, routing policies, transport options, backend connectivity, and failover under its own operational control. Choose a managed telephony product instead when the priority is provisioned numbers, inbound calling, or built-in recording.
Kamailio pros and cons
- Where it wins
- Free, open-source SIP proxy, registrar, and routing infrastructure
- Supports failover, load balancing, NAT traversal, and multiple transports
- Connects with PSTN gateways, Asterisk, FreeSWITCH, and major databases
- Where it doesn't
- Self-hosted infrastructure rather than a managed telephony API
- No bundled phone-number provisioning or inbound calling
- Call recording is not supplied
Kamailio fact sheet, pricing and score →
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