Tuesday 24 May 2016

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.
Welcome to Kamailio® – the Open Source SIP Server
Kalamailio Features
Features
  • 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
For Price and Information
For Price and Information
For pricing or have any further questions
Please email us at sales@arnserver.com
Or can live chat on Skype
ID: arnserver

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