SIP Connection Details

Name of the SIP Connection of the customer service.

Connection status (see Management Statuses)

SIP Connection type:

Unique IP Group ID for the SIP connection of the service created in the Live Platform database.

Name of the Service Provider tenant.

Name of the customer attached to the service.

Name of the service.

Unique ID of the SIP connection created in the Live Platform database.

See Management Statuses.

See Management Statuses.

Shows the Active alarms for all categories including Critical, Major, Minor and Warning alarms. Click each category to drill down to the Active Alarms page.

Shows the total number of call streams.

Max Concurrent Calls chart shows the maximum concurrent calls distributed over time. The chart shows when the maximum concurrent calls peaked compared to when they dipped. You can compare this to other charts to identify correlation. Max Concurrent Calls is the maximum number of calls opened at the same time in the server. Note that if you click a bar to open the Calls List page, the number of calls shown in the Calls List page might be different to the number shown in the graph; only calls that end within the time range are displayed in the Calls List page; if a call exceeds the time range, it won't be displayed in the Calls List page.

MOS - Mean Opinion Score (specified by ITU-T recommendation P.800) - the average grade on quality scales of Good to Failed, given to voice calls made over a VoIP network at the conclusion of the testing.

Jitter can result from uneven delays between received voice packets. To space packets evenly, the jitter buffer adds delay. The higher the measurement, the greater the impact of the jitter buffer’s delay on audio quality. Two Jitter values are shown, one value for the caller side and one value for the callee side.

Delay (or latency) - the time it takes for information to travel from source to destination (round-trip time). Sources of delay include voice encoding / decoding, link bandwidth and jitter buffer depth. Two Delay values are shown, one value for the caller side and one value for the callee side.

Lost packets - RTP packets that aren’t received by the voice endpoint for processing, resulting in distorted voice transmission. Two Packet Loss % values are shown, one value for the caller side and one value for the callee side. Packet Loss can be greater than 100%.

Successful / Failed Calls chart shows the % and # of calls whose performance was evaluated as successful or failed, distributed over time The chart lets you assess calls performance at a glance. The chart shows when successful calls peaked compared to when failed calls peaked. You can compare this to other charts to identify correlations. Click to drill down to the details of the selected data segment.