Welcome to Kamailio® – the Open Source SIP Server

Kamailio® (successor of former OpenSER and SER) is an Open Source SIP Server released under GPL, able to handle thousands of call setups per second. Kamailio can be used to build large platforms for VoIP and realtime communications – presence, WebRTC, Instant messaging and other applications.  It can also easily be applied to  scaling up SIP-to-PSTN gateways, PBX systems or media servers like Asterisk™, FreeSWITCH™ or SEMS.

Among the powerful features: asynchronous TCP, UDP and SCTP, secure communication via TLS for VoIP (voice, video, text); WebSocket support for WebRTC; IPv4 and IPv6; SIMPLE instant messaging and presence with embedded XCAP server and MSRP relay; asynchronous operations; IMS extensions; ENUM; DID and least cost routing; load balancing; routing fail-over; accounting, authentication and authorization; support for many backend systems such as MySQL, Postgres, Oracle, Radius, LDAP, Redis, Cassandra, MongoDB, Memcached; Json and XMLRPC control interface, SNMP monitoring. 


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~ RELEVANT PAST EVENTS~

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Kamailio Bossies'09


  • November 04, 2008 – Kamailio (OpenSER) and SIP Express Router (SER) teamed up to integrate back their source trees, known as the SIP Router project.

The SIP Router Project

Kamailio can be used on systems with limited resources as well as on carrier grade servers. It is written in pure C for Unix/Linux-like systems with architecture specific optimizations to offer high performances. Kamailio Project aims to be a collaborative environment of its users to develop secure and extensible SIP server to provide modern Unified Communication and VoIP services.