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.
- secure peer-to-peer communication
- person to person or group voice calls
- person to person video calls
- screen sharing
- instant messaging
- presence status
- call detail records