Berlin, Germany

Solutions

Built around the Kamailio 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.

SolutionsASIPTO-UCP

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

Internet Telephony Platform

SolutionsASIPTO-LBD

SIP Load Balancer

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

SolutionsASIPTO-SWG

SIP WebRTC Gateway

  • can be integrated in existing Kamailio platforms
  • can be deployed as standalone node
  • connect WebRTC endpoints with classic SIP phones
  • add contextual communications to your servce
  • combined use of WebRTC endpoints and classic SIP phones
  • audio, video, instant messaging and presence

SolutionsASIPTO-LCR

Least Cost Routing

  • select the best available route
  • priority can be based on cost, quality or custom rules
  • high capacity of routing records
  • multiple least cost routing profiles
  • processing of thousands of requests per second
  • failover to next available routes

SolutionsASIPTO-EP-SBC

SIP Edge Proxy – Session Border Controller

Customizable SIP system running at the edge of VoIP platforms – a selection of its features is listed next.

  • protect internal VoIP platform from DoS and DDoS attacks
  • detect password scanning attacks and block them
  • IP whitelisting and blacklisting, traffic ratelimiting
  • normalize SIP headers, add/remove SIP headers
  • gatewaying between SRTP and RTP; 
  • SIP gateway to Microsoft Teams or proprietary PBXes

SolutionsASIPTO-SPE

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