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SIP Solutions

Built around the Kamailio (OpenSER) SIP server, integrating other popular Open Source applications and technologies (Asterisk, FreeSWITCH, SEMS), Asipto’s solutions offer the shortest time to roll out your SIP or WebRTC service, leaving open the way to extend to new functionalities as you go.

Internet Telephony Platform

SIP Unified Communication Platform

  • IP communication sessions (voice, video, gaming, a.s.o.) with NAT traversal support
  • presence and instant messaging
  • administrative web front-end
  • dns based addressing, multiple domains enabled
  • rich telephony features and media services
  • accounting, least cost routing and enum peering engine
SIP Load Balancer

  • hot plug of new components in the SIP network
  • customizable load balancing rules
  • redirect traffic based on DID or service type
  • secure the SIP internal network
  • high availability and failover
SIP Prepaid Engine

  • real-time charging and SIP session disconnect
  • per-second precision billing
  • multiple charging plans
  • cost and time unit per destination
  • voucher system for loading credit
  • listen the credit value
SIP Number Portability Server

  • solve number portability demands in SIP networks with focus on performance and scalability
  • seamless integration with any SIP server
  • high capacity of records for ported numbers
  • dual mode: redirect server or proxy server
  • processing of thousands of requests per second
  • interrogation from one or many SIP servers at the same time

For more details, please contact us.