Kamailio – build large platforms for VoIP and realtime communications

Kamailio is an open source implementation of a SIP Signaling Server. SIP is an open standard protocol specified by the IETF. This is an industrial-strength, free server for realtime communication, based on the Session Initiation Protocol (SIP RFC3261).

The software is designed to permit Realtime Communications such as IP telephony and presence infrastructures up to large scale. With embedded support for WebSockets/WebRTC, HTTP, XSRP and XMLrpc as well as Json-rpc it’s a modern server, up to date with current standards on IPv4/IPv6 and TLS security.

Kamailio’s performance and robustness allows it to serve millions of users and accommodate needs of very large operators. With a low-cost dual-CPU, the Kamailio server is able to power IP telephony services in an area as large as the Bay Area during peak hours.

Features include:

  • Robust and Performant SIP (RFC3261) Server flavour:
    • Registrar server.
    • Location server.
    • Proxy server.
    • SIP Application server.
    • Redirect server.
  • Flexibility:
    • Small footprint – suitable for embedded devices – the binary file is small size, functionality can be stripped/added via modules.
    • Plug&play module interface – ability to add new extensions, without touching the core, therefore assuring a great stability of core components.
    • Modular architecture – core, internal libraries and module interface to extend the server’s functionality.
    • Extension repository – overall more than 150 modules are included in the Kamailio source tree.
  • SIP Routing Capabilities:
    • Stateless and transactional stateful SIP Proxy processing.
    • Serial and parallel forking.
    • NAT traversal support for SIP and RTP traffic.
    • Load balancing with many distribution algorithms and failover support.
    • Flexible least cost routing.
    • Routing failover.
    • Replication for High Availability (HA).
  • Transport Layers:
    • Support for communication via UDP, TCP, TLS and SCTP.
    • IPv4 and IPv6.
    • Transport layer gatewaying (IPv4 to IPv6, UDP to TLS, a.s.o.).
    • SCTP multi-homing and multi-streaming.
    • WebSocket for WebRTC.
  • Asynchronous Processing:
    • Asynchronous TCP handling.
    • Asynchronous SIP message processing.
    • Asynchronous inter-process message queues communication system.
    • Distributed message queue.
  • Secure Communication:
    • Digest SIP User authentication.
    • Authorization via ACL or group membership.
    • IP and Network authentication.
    • TLS support for SIP signaling.
    • Transparent handling of SRTP for secure audio.
    • TLS domain name extension support.
    • Authentication and authorization against database (MySQL, PostgreSQL, UnixODBC, BerkeleyDB, Oracle, text files), RADIUS and DIAMETER.
  • IP and DNS:
    • Support for SRV and NAPTR DNS lookups.
    • SRV DNS failover.
    • DNSsec support.
    • ENUM support.
    • internal DNS caching system – avoid DNS blocking.
    • IP level Blacklists.
    • Multi-homed and multi-domain support.
    • Topology hiding – hide IP addresses in SIP headers to protect your network architecture.
  • Accounting:
    • Event based accounting.
    • Configurable accounting data details.
    • Multi-leg call accounting.
    • Storage to database, Radius or Diameter.
    • Prepaid engine.
  • External Interaction via:
    • RPC control interface – via XMLRPC, JSONRPC, UDP or TCP.
    • RabbitMQ and NSQ connectors.
  • Rich Communication Services:
    • SIP SIMPLE Presence Server (rich presence).
    • Presence User Agent.
    • XCAP client capabilities.
    • Embedded XCAP Server.
    • Presence DialogInfo support – SLA/BLA.
    • Instant Messaging.
    • Embedded MSRP relay.
  • Monitoring and Troubleshooting:
    • SNMP – interface to Simple Network Management Protocol.
    • Config file step-by-step debugger.
    • Remote control via XMLRPC.
    • Internal statistics exported via RPC and SNMP.
    • Flexible debug and error message logging system – log custom messages including any header or pseudo-variable and parts of SIP message structure.
  • Extensibility APIs:
    • Perl Programming Interface – embed your extensions written in Perl.
    • Java SIP Servlet Application Interface – write Java SIP Servlets to extent your VoIP services and integrate with web services.
    • Lua Programming Interface.
    • JavaScript Programming Interface.
    • Managed Code (C#) Programming Interface.
    • Python Programming Interface.
    • Java Programming Interface.
    • Generic Event API via TCP connections.
  • Supports Multiple Database Backends:
    • (MySQL, PostgreSQL, SQLite, UnixODBC, BerkeleyDB, Oracle, text files) and other database types which have unixodbc drivers.
    • Connections pool.
    • Different backends can be used simultaneously (e.g., accounting to Oracle and authorization against MySQL).
    • Connectors for Memcached, Redis, MongoDB and Cassandra no-SQL backends.
  • Interconnectivity:
    • Straightforward interconnection with PSTN gateways.
    • Gateway to sms or xmpp and other IM services.
    • Interoperability with SIP enabled devices and applications such as SIP phones (Snom, Cisco, etc.), Media Servers (Asterisk, FreeSwitch, etc).
  • IMS:
    • Diameter support and authentication.
    • Charging, QOS, ISC.

Website: www.kamailio.org
Support: Doucmentation, GitHub, Twitter
Developer: Kamailio developers
License: GNU GPL

Kamailio is written in C. Learn C with our recommended free books and free tutorials.

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