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The Origination and Destination Call Quality Metrics tables provide the following information, depending on the environment you are monitoring. Video metrics are reported in the following circumstances:
(Avaya only) The QoS priority setting for voice packets at Layer 2.
(Cisco PSTN calls only) The total echo return loss on the network. ACOM measures how significantly the voice gateway reduced the echo. ACOM includes echo reduction that occurs with or without the activity of an echo cancellation device.
The percentage of packets within burst periods that are lost or discarded.
The average duration of all high-loss periods in a data stream.
The percentage of frames in a data stream that are concealment frames, which the endpoints generate to conceal packet loss. Includes both early and late packets. Either the cumulative value (the total since the beginning of the stream) or the maximum value (the highest since the beginning of the stream).
The Mean Opinion Score (MOS) based on metric factors from both directions of data flow.
The Differentiated Services Code Point setting of the incoming RTP packets.
The “length” of echo cancellation processing. Based on the distance between a voice gateway and the endpoint. Typical values range from 8 milliseconds to 32 milliseconds.
The average length of frozen video instances.
The frequency of long and noticeable frozen video periods for an entire session. Expressed as a percentage of session time.
The percentage of lost or discarded packets in the gaps between bursts in a data stream.
The average duration of periods of good performance (low loss) between periods of data loss in a data stream.
Packet delay that distorts the quality of a voice conversation.Either the average or the maximum value.
The definition of Max Jitter varies by monitored environment:
Delay that the jitter buffer introduces while it holds one or more packets to reduce variations in packet arrival times.
The packets that are lost when jitter hinders the caching capacity of the jitter buffer.
The number of times that jitter exceeded the maximum size setting of the jitter buffer. Packets arrive too quickly to be contained by the jitter buffer. Over runs usually result in packet loss.
The number of times that the jitter buffer became empty. Packets arrive too slowly to be contained by the jitter buffer. Under runs usually indicate that delays are too lengthy for the buffer setting.
One-way delay. Calculated from the origination party to the destination party. Includes propagation delay, network delay, and packetization delay.
The Mean Opinion Score, which is based on call legs traveling toward the endpoint to reflect listener perception of quality. Either the average or minimum value.
A counter embedded in data to prevent a data packet from circulating through the network indefinitely. The counter decrements each time that the packet passes through a router or a switch. Either the minimum or maximum value in a data stream.
The Mean Opinion Score (MOS) is an industry standard method for gauging call quality. MOS is an estimation of how impairments to a voice signal affect listener perception of call quality. Either the average or the minimum value.
MOS listening quality value that is based only on network factors, such as codec, packet loss, packet reordering, packet errors, and jitter. Either the average or the minimum value.
(Microsoft only) A breakdown of the metrics used in calculating the Network MOS. Degradation refers to an impairment factor that decreased the MOS.
(Microsoft only) The average portion of an audio signal that is noise and not actual voice data. Measured in decibels.
The percentage of data packets that were lost in transit. These packets were sent but never received at the destination. Either the average or the maximum value.
The number of data packets that were lost in transit. These packets were sent but never received at the destination.
The number of data packets in a stream.
The width and height of the video image, in pixels. Higher resolutions require more bandwidth and more CPU resources. Support for resolution settings varies by endpoint.
The number of times that at least one packet arrived out of order. Either the actual or the maximum number of sequence falls in a stream.
The number of times that at least one consecutive packet was lost. Either the actual or the maximum number of sequence jumps in a stream.
The number of call seconds that had more than 5 percent concealment events from the start of the data stream.
The average audio signal level in decibels (dBm0).
The number of bits sent per second for an entire video stream. Bit rates provide a gauge of codec performance. Either the average or the maximum value.
The percentage of consecutive video packets that were lost in transit.
The average amount of time for decoding frames in a stream. A slower decoding rate can be the result of conditions on the endpoint, such as lack of CPU resources, and can affect call quality.
The average number of unique consecutive images, or video frames, lost due to corruption and error concealment for the entire system. Video frames can span multiple packets.
The average number of frames that were sent or received per second for an entire stream.
The variation in delay among video packets in the same stream. Either the average or the maximum value.
The maximum time for a video packet to travel between the calling parties. Measured from end-to-end in one direction. Calculated by taking the average round-trip time for a call leg in a given video call and dividing it in half.
The percentage of video packets that were lost in transit. These packets were sent but never received at the destination. Either the average or the maximum value.
The number of video packets in a stream.
The number of video packets that were lost in transit. These packets were sent but never received at the destination.
(Avaya only) The VLAN that carried the RTP packets.
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