consistec is the leading German supplier for VoIP monitoring & fraud prevention

Detect and solve VoIP quality and infrastructure problems in a timely manner.

Our caplon© systems are used by leading German and international telecommunications providers. We also offer customized solutions for small and medium-sized enterprises. With our passive approach, in which VoIP monitoring and fraud prevention can be seamlessly combined, we are the leading supplier in Germany.

With caplon©, VoIP infrastructures can also be monitored across multiple sites and signalling / registration and VoIP quality problems can be quickly detected and rectified. The signalling data can also be analyzed to identify fraud scenarios as they arise and to prevent them at an early stage. In this way, considerable financial damage can be effectively avoided.

We have more than 150+ man years of experience in the telecommunications sector. Benefit from our industry-specific know-how and our excellent support.

 

Monitoring of quality, availability and performance

Increase the satisfaction of your customers and save valuable time of your employees, who can concentrate on their core tasks instead of costly error analyses.

 

Overview of the complete VoIP communication

  • Clear presentation of important metrics in individual and user-definable dashboards
  • Extensive filter options both globally per dashboard and locally per widget
  • Configuration of nodes and trunks for which specific metrics are calculated online
  • Efficient troubleshooting by drill-down analyses from anomalies in graphs to the associated calls and network data

Call statistics and call analyses

  • Call statistics: Call rate, status distribution, number of concurrent calls, number of errors, call duration, Post Dial Delay, ASR, NER, …
  • Extensive filter options for individual calls and associated network data
  • Visualization of call flows in message sequence charts
  • Reliable correlation of all call legs belonging to a call (even if the call ID changes)
  • Analysis and display of DTMF tones (in-band, as telephone event (RFC-4733 or RFC-2833) and as SIP info event)

Call quality analysis

  • Determination of call quality by analysis of RTP streams: MOS, Jitter, Packet Loss, Out-Of-Order Packets, Burst Ratio, Sequence Errors, …
  • Fast detection of the call legs for a call that mainly contribute to poor quality by presenting the quality values.
  • Detection of quality degradations by displaying the MOS values over time according to ITU-T G.107
  • Detect burst-like quality drops in the call by determining the quality values per call every 10 seconds
  • Analysis of the RTCP data and comparison to the values calculated from the RTP streams
  • one-way audio detection

Analysis of SIP registrations

  • Detection of registration interruptions by displaying the registration status per AOR and Contact over time
  • Easy analysis of registration problems by filterable lists with all SIP registrations / SIP registration attempts with the option of PCAP downloads
  • Overview of the general registration status by continuous calculation of relevant statistics: Number of registered users, registration rate, status distribution of SIP registrations, Registration Request Delay (RRD), Ineffective Registration Attempts (IRA), …

VoIP Alarming and Reporting

  • Intuitive creation of rules via a rule wizard, for which an alarm is generated if they are violated
  • Check of absolute threshold values or comparison with historical values (e.g. 20% more failing calls than "normal") ---> System learns normal behaviour of VoIP infrastructure
  • Individual monitoring of nodes or trunks by specific rule creation
  • Immediate information about problems by alerting via UI, e-mail or SNMP
  • Detailed information about alarms incl. statistics on related historical values
  • Individually definable reports that are sent to a dedicated mailing list in a fixed time interval (e.g. once a week or once a month).

Fast handling of support cases through system support

  • Option to permanently store RTP traffic for dedicated calls for a more detailed analysis and to download the corresponding network packets ("RTP trigger")
  • User-tracking page on which both registration and signaling data is shown for a dedicated customer in order to get a quick idea about the root-cause of potential problems
  • Saving calls as favorites (with descriptive names) to be able to quickly find them again or to simply reference them in tickets (also via deep links).
  • Incoming calls and registrations can be shown in a filterable live view, including raw message view.
  • Convenient option to analyse traffic online via Wireshark plugin ("use as full-fledged network tracer") even for non VoIP-specific protocols

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