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Kamailio

Free#30 of 53 in Voice APIs

Kamailio: A Linux-based SIP routing foundation for teams building self-hosted voice infrastructure. Ranked #30 of 53 in Voice APIs by our editors (6.8/10); pricing: Free plan; best for open-source SIP routing infrastructure.

6.8/10Editor score
Visit Kamailio
Kamailio6.8 Visit Kamailio

At a glance

  • Editor score
    6.8 / 10
  • Pricing
    Free plan
  • Best for
    Open-source SIP routing infrastructure
  • Free plan
    Yes
  • Paid from
    Free
  • Inbound calling
    Yes
  • Founded
    2001
  • Facts checked
    24 Sep 2026
Kamailio screenshot

  • 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

Our verdict on Kamailio

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.

Your product? Our editors can test Kamailio in depth and publish a review, labelled as paid · from $149/mo →

Kamailio pricing

Plans Free planFree Free to use — no paid tier required for the core job.
See plans on kamailio.org

Kamailio fact sheet

Free planYes
Paid fromFree
Inbound callingYes
Outbound callingYes
Phone number supportNo
SIP supportYes
Call recordingNot verified
Concurrent call limitNot verified
SDK languagesNot verified
DeploymentSelf-hosted
PlatformsLinux
SupportCommunity, Docs
Built forSmall business, Mid-market, Enterprise (editorial estimate)
Integrations9 integrations: MySQL, PostgreSQL, Oracle, Radius, LDAP, Redis …
PricingFree plan
Websitekamailio.org
Facts checked24 Sep 2026

Kamailio integrations

Kamailio lists 9 integrations on its own site.

  • MySQL
  • PostgreSQL
  • Oracle
  • Radius
  • LDAP
  • Redis
  • Cassandra
  • MongoDB
  • Memcached

Alternatives to Kamailio

See all Kamailio alternatives →

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Featured on iTechGuides

Featured on iTechGuides — Kamailio 6.8/10

Kamailio is listed in our Voice APIs directory. Add the badge to your site — it links back to this page.

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Reviewed by iTechGuides Editors · Editorial team · Updated Oct 2026

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