Introduction

VOIP – SIP (Voice Over IP – Session Initiation Protocol) web service monitor verifies the availability or performance issues of your VOIP communications. The monitor replicates calls to your SIP device and analyzes the call response to monitor the quality.

Creating SIP web service monitor

After selecting the web service monitor type, to configure the SIP web service monitor:

  1. From the Add Web Service window, provide details for the following parameters and click Save:
  • Name: Refers to the name of the web service monitor.
    Note: The value entered in the Name field must be unique. If the name is not unique, the screen displays the error message: Name Already Exists.
  • Call Plan (One to Many): Refers to monitoring many Gateways configured at the destination locations from a Gateway configured at the source location.
  • Call Plan (Any to Any): Refers to every Gateway configured at Locations, monitor each other.
  • Call Duration: Refers to the duration that verifies the call quality. Call duration cannot exceed the time configured for Call Frequency.
  • Codec Format: Refers to the Codec format that determines the format of the voice packets received during the call.
  • Frequency: The time interval to monitor the chosen Gateways.
    The screen displays a confirmation message about the successful addition of the web service monitor.
  1. Click OK.
    Synthetics page displays the configured web service monitor.

To manage a configured web service monitor:

  • Edit – To modify the existing details of the configured web service monitors.
  • Delete – To remove any configured web service monitor.
  • Scheduled Maintenance – To move any configured web service monitor to Scheduled Maintenance.

After configuring, view Metrics and graphical representations for additional information.

Metrics for SIP monitor types

Metrics for SIP Monitor Types
Monitor TypesUnitsRecommended MetricsDescription
SIP-availability.down.location.countLocation Count: The number of locations where the configured host is down
SIPMillisecondssip.call.latencyLatency: Time between the moment a voice packet is transmitted and the moment it reaches its destination
SIPPercentagesip.call.plPacket loss: The percentage of packets dropped by the network on the way to the destination address
SIP-sip.call.mosMean Opinion Score (MOS): Measure of voice quality
SIPMillisecondssip.call.jitterJitter: The time taken to measure the voice instability
SIPMillisecondssip.call.rttRound trip time: Duration for data to travel to the target destination for the speed test and back

Graphical representation for metrics

Graphical Representation

Graphical Representation

What to do next