Media quality statistics
To help you understand media quality in VoIP and video calls that use Azure Communication Services, there's a feature called media quality statistics. Use it to examine the low-level audio, video, and screen-sharing quality metrics for incoming and outgoing call metrics.
Media quality statistics for an ongoing call
Important
There's an interface update on media quality statistics in the SDK, beginning with the SDK version 1.20.1
Media quality statistics is an extended feature of the core Call API. You first need to obtain the mediaStatsFeature API object:
const mediaStatsFeature = call.feature(Features.MediaStats);
To receive the media statistics data, you can subscribe sampleReported event or summaryReported event.
sampleReported event triggers every second. It's suitable as a data source for UI display or your own data pipeline.
summaryReported event contains the aggregated values of the data over intervals, which is useful when you just need a summary.
If you want control over the interval of the summaryReported event, you need to define mediaStatsCollectorOptions of type MediaStatsCollectorOptions.
Otherwise, the SDK uses default values.
const mediaStatsCollectorOptions: SDK.MediaStatsCollectorOptions = {
aggregationInterval: 10,
dataPointsPerAggregation: 6
};
const mediaStatsCollector = mediaStatsFeature.createCollector(mediaStatsSubscriptionOptions);
mediaStatsCollector.on('sampleReported', (sample) => {
console.log('media stats sample', sample);
});
mediaStatsCollector.on('summaryReported', (summary) => {
console.log('media stats summary', summary);
});
In case you don't need to use the media statistics collector, you can call dispose method of mediaStatsCollector.
mediaStatsCollector.dispose();
It's not necessary to call dispose method of mediaStatsCollector every time when the call ends, as the collectorsare reclaimed internally when the call ends.
MediaStatsCollectorOptions
The MediaStatsCollectorOptions is optional, and there are two optional fields in MediaStatsCollectorOptions.
aggregationIntervalis the interval, in seconds, that the statistics are aggregated. The default value is 10.dataPointsPerAggregationdefines how many data points each aggregation event has. The default value is 6.
These two values determine the frequency at which the SDK emits summaryReported event and the number of aggregated data points included in the report.
The summaryReported event raised every aggregationInterval * dataPointsPerAggregation seconds.
For example, if you set the following values:
aggregationInterval= 1dataPointsPerAggregation= 60
The summaryReported event is raised every 60 seconds and contains 60 unique units for each statistic recorded.
If you set the following values:
aggregatinInterval= 60dataPointsPerAggregation= 1
The summaryReported event is raised every 60 seconds and contains 1 unique unit for each statistic recorded.
Best practices
If you want to collect the data for offline inspection, we recommend that you collect the data and send it to your pipeline ingestion after your call ends. If you transmit the data during a call, it could use internet bandwidth needed to continue an Azure Communication Services call (especially when available bandwidth is low).
In either sampleReported event or summaryReported event, the media statistics data are not just a simple key-value mapping.
Here is the type declaration of the event data reported by sampleReported event.
export interface MediaStatsReportSample {
audio: {
send: OutgoingAudioMediaStats<number, string>[];
receive: IncomingAudioMediaStats<number, string>[];
};
video: {
send: OutgoingVideoMediaStats<number, string>[];
receive: IncomingVideoMediaStats<number, string>[];
};
screenShare: {
send: OutgoingScreenShareMediaStats<number, string>[];
receive: IncomingScreenShareMediaStats<number, string>[];
};
transports: TransportMediaStats<number>[];
}
The event data provide the statistics data for each media stream in the call, including both send and receive directions.
It's recommended that you print the event using the console.log to observe its layout and value changes, so you can find a proper way to display or process the data according to your usage scenario.
Audio send metrics
| Metric name | Description | Comments |
|---|---|---|
id |
Statistics ID | Used to identify statistics across the events, especially when there are multiple statistics with the same media type and direction in an event. |
codecName |
Codec name | OPUS, G722. |
bitrate |
Audio send bit rate (bits per second) | General values are in the 24-Kbps range (36-128 Kbps is typical). |
jitterInMs |
Packet jitter (milliseconds) | Lower is better. |
packets |
The total number of packets sent. | |
packetsPerSecond |
Packet rate (packets per second) | |
packetsLost |
The total number of packets lost reported from the remote end. | |
packetsLostPerSecond |
Packet loss rate (packets per second) | Lower is better. |
rttInMs |
Round-trip time (milliseconds) | Lower is better. Calculated from the RTCP receiver report. We recommend a round-trip time of 200 ms or less. |
audioInputLevel |
Audio volume level from the microphone | The value ranges from 0 to 65536. A value of 0 represents silence. |
transportId |
Transport ID | Used to associate the stats in transports. |
Audio receive metrics
In the SDK versions ealier than 1.20.1, jitterBufferDelayInMs existed as jitterBufferInMs.
| Metric name | Description | Comments |
|---|---|---|
id |
Statistics ID | Used to identify statistics across the events, especially when there are multiple statistics with the same media type and direction in an event. |
codecName |
Codec name | OPUS, G722. |
bitrate |
Audio receive bitrate (bits per second) | General values are in the 24-Kbps range (36-128 Kbps is typical). |
jitterInMs |
Packet jitter (milliseconds) | Lower is better. |
packets |
The total number of packets received. | |
packetsPerSecond |
Packet rate (packets per second) | |
packetsLost |
The total number of packets lost. | |
packetsLostPerSecond |
Packet loss rate (packets per second) | Lower is better. |
jitterBufferDelayInMs |
Jitter buffer (milliseconds) | Lower is better. The jitter buffer is used for smooth playout. This value is how long the packets of the samples stay in the jitter buffer. |
audioOutputLevel |
Audio volume level from the receiving stream | The value ranges from 0 to 65536. A value of 0 represents silence. |
healedRatio |
Ratio of concealed samples (except silentConcealedSamples) to total received samples |
Information only. |
transportId |
Transport ID | Used to associate the stats in transports. |
Video send metrics
Starting from SDK version 1.20.1, the video send metrics included the altLayouts metric field, which allows for a better representation of simulcast stream statistics.
| Metric name | Description | Comments |
|---|---|---|
id |
Statistics ID | Used to identify statistics across the events, especially when there are multiple statistics with the same media type and direction in an event. |
codecName |
Codec name | H264, VP8, VP9. |
bitrate |
Video send bitrate (bits per second) | |
jitterInMs |
Packet jitter (milliseconds) | Lower is better. |
packets |
The total number of packets sent. | |
packetsPerSecond |
Packet rate (packets per second) | |
packetsLost |
The total number of packets lost reported from the remote end. | |
packetsLostPerSecond |
Packet loss rate (packets per second) | Lower is better. |
rttInMs |
Round-trip time (milliseconds) | Lower is better. Calculated from the RTCP receiver report. We recommend a round-trip time of 200 ms or less. |
frameRateInput |
Frame rate that originates from the video source (frames per second) | |
frameWidthInput |
Frame width of the last frame that originates from the video source (pixels) | |
frameHeightInput |
Frame height of the last frame that originates from the video source (pixels) | |
framesEncoded |
The number of frames successfully encoded for the RTP stream. | |
frameRateEncoded |
Frame rate successfully encoded for the RTP stream (frames per second) | |
framesSent |
The number of frames sent on the RTP stream | |
frameRateSent |
Frame rate sent on the RTP stream (frames per second) | |
frameWidthSent |
Frame width of the encoded frame (pixel) | |
frameHeightSent |
Frame height of the encoded frame (pixel) | |
keyFramesEncoded |
Key frames successfully encoded for the RTP stream | |
transportId |
Transport ID | Used to associate the stats in transports. |
altLayouts |
Simulcast streams | altLayouts contains the same metrics to the video send |
Video receive metrics
In the SDK versions earlier than 1.20.1, jitterBufferDelayInMs existed as jitterBufferInMs.
| Metric name | Description | Comments |
|---|---|---|
id |
Statistics ID | Used to identify statistics across the events, especially when there are multiple statistics with the same media type and direction in an event. |
codecName |
Codec name | H264, VP8, VP9. |
bitrate |
Video receive bitrate (bits per second) | |
jitterInMs |
Packet jitter (milliseconds) | Lower is better. |
packets |
The total number of packets received. | |
packetsPerSecond |
Packet rate (packets per second) | |
packetsLost |
The total number of packets lost. | |
packetsLostPerSecond |
Packet loss rate (packets per second) | Lower is better. |
rttInMs |
Round-trip time (milliseconds) | Lower is better. Calculated from the RTCP sender report. We recommend a round-trip time of 200 ms or less. |
streamId |
Stream ID | The streamId value corresponds to id in VideoStreamCommon. It can be used to match the sender. |
jitterBufferDelayInMs |
Jitter buffer (milliseconds) | Lower is better. The jitter buffer is used for smooth playout. This value is how long the packets of the frames stay in the jitter buffer. |
frameRateDecoded |
Frame rate correctly decoded for the RTP stream (frames per second) | |
frameRateReceived |
Frame rate received on the RTP stream (frames per second) | |
frameWidthReceived |
Frame width of the decoded frame (pixel) | |
frameHeightReceived |
Frame height of the decoded frame (pixel) | |
longestFreezeDurationInMs |
Longest freeze duration (milliseconds) | |
totalFreezeDurationInMs |
Total freeze duration (milliseconds) | |
framesReceived |
Total number of frames received on the RTP stream | |
framesDecoded |
Total number of frames correctly decoded for the RTP stream | |
framesDropped |
Total number of frames dropped | |
keyFramesDecoded |
Total number of key frames correctly decoded for the RTP stream | |
transportId |
Transport ID | Used to associate the stats in transports. |
Screen-share send metrics
Currently, statistics fields are the same as video send metrics.
Screen-share receive metrics
Currently, statistics fields are the same as video receive metrics.
Transport metrics
The transport related metrics were separated out after ACS Web SDK 1.20.1.
In earlier versions, rttInMs existed as pairRttInMs in the stats for audio, video, and screenShare.
availableIncomingBitrate was availableBitrate in the receive stats for audio, video, and screenShare.
availableOutgoingBitrate was availableBitrate in the send stats for audio, video, and screenShare.
| Metric name | Description | Comments |
|---|---|---|
id |
Transport ID | Used to associate with the transportId in other stats |
rttInMs |
Round-trip time (milliseconds) | The value is calculated from the STUN connectivity check. We recommend a round-trip time of 200 ms or less. |
availableIncomingBitrate |
Bandwidth estimation (bits per second) | The value may not be available depending on the bandwidth estimation algorithm used in the WebRTC session |
availableOutgoingBitrate |
Bandwidth estimation (bits per second) | The value may not be available depending on the bandwidth estimation algorithm used in the WebRTC session |
What are changed in SDK version 1.20.1 (GA)
We now support MediaStats feature API in 1.20.1 (GA). Compared to the previous beta versions, we also made some minor changes to the API interface in this GA version.
In the previous beta versions, pairRttInMs, availableBitrate were included in audio, video, and screenShare statistics.
Now, these metrics have been separated into transport metrics.
We introduced packets, packetsLost metric fields in audio, video, screenShare statistics. These metrics are useful for calculating the total number of packets sent or recieved between two different time points.
The frameRateOutput in video and screenShare statistics is removed. You can use frameRateDecoded instead.
The metric field jitterBufferInMs has been renamed to jitterBufferDelayInMs to provide a clearer description, as this metric indicates the duration of a packet stay in the jitter buffer.
Media quality statistics for an ongoing call
Media quality statistics is an extended feature of the core Call API. You first need to obtain the MediaStatisticsCallFeature API object:
MediaStatisticsCallFeature mediaStatisticsCallFeature = call.feature(Features.MEDIA_STATISTICS);
The MediaStatisticsCallFeature object has the following API structure:
- The
OnReportReceivedListenerevent listens for periodic reports of the media statistics. getReportIntervalInSecondsgets the interval, in seconds, of the media statistics report generation. The SDK uses10second as default.updateReportIntervalInSeconds()updates the interval, in seconds, of the media statistics report generation. The SDK uses10second as default.- A
MediaStatisticsReportcontains the definition of the outgoing and incoming media statistics, categorized by audio, video, and screen share.getOutgoingStatistics(): The list of media statistics for outgoing media.getAudioStatistics(): The list of media statistics for outgoing audio.getVideoStatistics(): The list of media statistics for outgoing video.getScreenShareStatistics(): The list of media statistics for outgoing screen share.getDataChannelStatistics(): The list of media statistics for data channel.
getIncomingStatistics(): The list of media statistics for incoming media.getAudioStatistics(): The list of media statistics for incoming audio.getVideoStatistics(): The list of media statistics for the incoming video.getScreenShareStatistics(): The list of media statistics for incoming screen share.getDataChannelStatistics(): The list of media statistics for data channel.
getLastUpdatedAt(): The date when the report was generated.
Then, subscribe to the addOnReportReceivedListener event to get regular updates about the current media quality statistics:
mediaStatisticsCallFeature.addOnReportReceivedListener(handleReportReceivedListener);
// Optionally, set the interval for media statistics report generation
mediaStatisticsCallFeature.updateReportIntervalInSeconds(15);
private void handleReportReceivedListener(MediaStatisticssReportEvent args) {
// Obtain the media statistics report instance
MediaStatisticsReport report = args.getReport();
// Obtain the outgoing media statistics for audio
List<OutgoingAudioStatistics> outgoingAudioStatistics = report.getOutgoingStatistics().getAudioStatistics();
// Obtain the outgoing media statistics for video
List<OutgoingVideoStatistics> outgoingVideoStatistics = report.getOutgoingStatistics().getVideoStatistics();
// Obtain the outgoing media statistics for screen share
List<OutgoingScreenShareStatistics> outgoingScreenShareStatistics = report.getOutgoingStatistics().getScreenShareStatistics();
// Obtain the outgoing media statistics for data channel
List<OutgoingDataChannelStatistics> outgoingDataChannelStatistics = report.getOutgoingStatistics().getDataChannelStatistics();
// Obtain the incoming media statistics for audio
List<IncomingAudioStatistics> incomingAudioStatistics = report.getIncomingStatistics().getAudioStatistics();
// Obtain the incoming media statistics for video
List<IncomingVideoStatistics> incomingVideoStatistics = report.getIncomingStatistics().getVideoStatistics();
// Obtain the incoming media statistics for screen share
List<IncomingScreenShareStatistics> incomingScreenShareStatistics = report.getIncomingStatistics().getScreenShareStatistics();
// Obtain the incoming media statistics for data channel
List<IncomingDataChannelStatistics> incomingDataChannelStatistics = report.getIncomingStatistics().getDataChannelStatistics();
}
Best practices
If you want to collect the data for offline inspection, we recommend that you collect the data and send it to your pipeline ingestion after your call ends. If you transmit the data during a call, it could use internet bandwidth needed to continue an Azure Communication Services call (especially when available bandwidth is low).
Outgoing audio metrics
| Metric name | Description | Comments |
|---|---|---|
CodecName |
Codec name | |
BitrateInBps |
Audio send bitrate (bits per second) | General values are in the 24-Kbps range (36-128 Kbps is typical). |
JitterInMs |
Packet jitter (milliseconds) | Lower is better. |
PacketCount |
The total number of packets sent. |
Incoming audio metrics
| Metric name | Description | Comments |
|---|---|---|
CodecName |
Codec name | |
JitterInMs |
Packet jitter (milliseconds) | Lower is better. |
PacketCount |
The total number of packets sent. | |
PacketsLostPerSecond |
Packet loss rate (packets per second) | Lower is better. |
Outgoing video metrics
| Metric name | Description | Comments |
|---|---|---|
CodecName |
Codec name | |
BitrateInBps |
Video send bitrate (bits per second) | |
PacketCount |
The total number of packets sent. | |
FrameRate |
Frame rate sent on the RTP stream (frames per second) | |
FrameWidth |
Frame width of the encoded frame (pixels) | |
FrameHeight |
Frame height of the encoded frame (pixels) |
Incoming video metrics
| Metric name | Description | Comments |
|---|---|---|
CodecName |
Codec name | |
BitrateInBps |
Video receive bitrate (bits per second) | |
JitterInMs |
Packet jitter (milliseconds) | Lower is better. |
PacketCount |
The total number of packets sent. | |
PacketsLostPerSecond |
Packet loss rate (packets per second) | Lower is better. |
StreamId |
Stream ID | The streamId value corresponds to the ID of the video of the remote participant. It can be used to match the sender. |
FrameRate |
Frame rate received on the RTP stream (frames per second) | |
FrameWidth |
Frame width of the decoded frame (pixels) | |
FrameHeight |
Frame height of the decoded frame (pixels) | |
TotalFreezeDurationInMs |
Total freeze duration (milliseconds) |
Outgoing screen share metrics
Currently, statistics fields are the same as Outgoing video metrics.
Incoming screen share metrics
Currently, statistics fields are the same as Incoming video metrics.
Outgoing data channel metrics
| Metric name | Description | Comments |
|---|---|---|
PacketCount |
The total number of packets sent. |
Incoming data channel metrics
| Metric name | Description | Comments |
|---|---|---|
JitterInMs |
Packet jitter (milliseconds) | Lower is better. |
PacketCount |
The total number of packets sent. |
Media quality statistics for an ongoing call
Media quality statistics is an extended feature of the core Call API. You first need to obtain the mediaStatisticsCallFeature API object:
var mediaStatisticsCallFeature = self.call.feature(Features.mediaStatistics)
The mediaStatisticsCallFeature object has the following API structure:
- The
didReceiveReportdelegate method listens for periodic reports of the media statistics. reportIntervalInSecondsgets the interval, in seconds, of the media statistics report generation. The SDK uses10second as default.updateReportInterval(inSeconds)updates the interval, in seconds, of the media statistics report generation. The SDK uses10second as default.- A
MediaStatisticsReportobject contains the definition of the outgoing and incoming media statistics, categorized by audio, video, and screen share.outgoingMediaStatistics: The list of media statistics for outgoing media.audio: The list of media statistics for the outgoing audio.video: The list of media statistics for the outgoing video.screenShare: The list of media statistics for the outgoing screen share.dataChannel: The list of media statistics for the outgoing data channel.
incomingMediaStatistics: The list of media statistics for incoming media.audio: The list of media statistics for the incoming audio.video: The list of media statistics for the incoming video.screenShare: The list of media statistics for the incoming screen share.dataChannel: The list of media statistics for the incoming data channel.
lastUpdated: The date when the report was generated.
Then, implement the didReceiveReport delegate to get regular updates about the current media quality statistics:
// Optionally, set the interval for media statistics report generation
mediaStatisticsCallFeature.updateReportInterval(inSeconds: 15)
mediaStatisticsCallFeature.delegate = MediaStatisticsDelegate()
public class MediaStatisticsDelegate : MediaStatisticsCallFeatureDelegate
{
public func mediaStatisticsCallFeature(_ mediaStatisticsCallFeature: MediaStatisticsCallFeature,
didReceiveReport args: MediaStatisticsReportReceivedEventArgs) {
let report = args.report
// Obtain the outgoing media statistics for audio
let outgoingAudioStatistics = report.outgoingStatistics.audio
// Obtain the outgoing media statistics for video
let outgoingVideoStatistics = report.outgoingStatistics.video
// Obtain the outgoing media statistics for screen share
let outgoingScreenShareStatistics = report.outgoingStatistics.screenShare
// Obtain the outgoing media statistics for data channel
let outgoingDataChannelStatistics = report.outgoingStatistics.dataChannel
// Obtain the incoming media statistics for audio
let incomingAudioStatistics = report.incomingStatistics.audio
// Obtain the incoming media statistics for video
let incomingVideoStatistics = report.incomingStatistics.video
// Obtain the incoming media statistics for screen share
let incomingScreenShareStatistics = report.incomingStatistics.screenShare
// Obtain the incoming media statistics for data channel
let incomingDataChannelStatistics = report.incomingStatistics.dataChannel
}
}
Best practices
If you want to collect the data for offline inspection, we recommend that you collect the data and send it to your pipeline ingestion after your call ends. If you transmit the data during a call, it could use internet bandwidth needed to continue an Azure Communication Services call (especially when available bandwidth is low).
Outgoing audio metrics
| Metric name | Description | Comments |
|---|---|---|
CodecName |
Codec name | |
BitrateInBps |
Audio send bitrate (bits per second) | General values are in the 24-Kbps range (36-128 Kbps is typical). |
JitterInMs |
Packet jitter (milliseconds) | Lower is better. |
PacketCount |
The total number of packets sent. |
Incoming audio metrics
| Metric name | Description | Comments |
|---|---|---|
CodecName |
Codec name | |
JitterInMs |
Packet jitter (milliseconds) | Lower is better. |
PacketCount |
The total number of packets sent. | |
PacketsLostPerSecond |
Packet loss rate (packets per second) | Lower is better. |
Outgoing video metrics
| Metric name | Description | Comments |
|---|---|---|
CodecName |
Codec name | |
BitrateInBps |
Video send bitrate (bits per second) | |
PacketCount |
The total number of packets sent. | |
FrameRate |
Frame rate sent on the RTP stream (frames per second) | |
FrameWidth |
Frame width of the encoded frame (pixels) | |
FrameHeight |
Frame height of the encoded frame (pixels) |
Incoming video metrics
| Metric name | Description | Comments |
|---|---|---|
CodecName |
Codec name | |
BitrateInBps |
Video receive bitrate (bits per second) | |
JitterInMs |
Packet jitter (milliseconds) | Lower is better. |
PacketCount |
The total number of packets sent. | |
PacketsLostPerSecond |
Packet loss rate (packets per second) | Lower is better. |
StreamId |
Stream ID | The streamId value corresponds to the ID of the video of the remote participant. It can be used to match the sender. |
FrameRate |
Frame rate received on the RTP stream (frames per second) | |
FrameWidth |
Frame width of the decoded frame (pixels) | |
FrameHeight |
Frame height of the decoded frame (pixels) | |
TotalFreezeDurationInMs |
Total freeze duration (milliseconds) |
Outgoing screen share metrics
Currently, statistics fields are the same as Outgoing video metrics.
Incoming screen share metrics
Currently, statistics fields are the same as Incoming video metrics.
Outgoing data channel metrics
| Metric name | Description | Comments |
|---|---|---|
PacketCount |
The total number of packets sent. |
Incoming data channel metrics
| Metric name | Description | Comments |
|---|---|---|
JitterInMs |
Packet jitter (milliseconds) | Lower is better. |
PacketCount |
The total number of packets sent. |
Media quality statistics for an ongoing call
Media quality statistics is an extended feature of the core CommunicationCall API. You first need to obtain the MediaStatisticsCallFeature API object:
MediaStatisticsCallFeature mediaStatisticsCallFeature = call.Features.MediaStatistics;
The MediaStatisticsCallFeature feature object has the following API structure:
- The
ReportReceivedevent listens for periodic reports of the media statistics. ReportIntervalInSecondsgets the interval, in seconds, of the media statistics report generation. The SDK uses10second as default.UpdateReportIntervalInSeconds()updates the interval, in seconds, of the media statistics report generation. The SDK uses10second as default.- A
MediaStatisticsReportobject contains the definition of the outgoing and incoming media statistics, categorized by audio, video, and screen share.OutgoingMediaStatistics: The list of media statistics for outgoing media.Audio: The list of media statistics for the outgoing audio.Video: The list of media statistics for the outgoing video.ScreenShare: The list of media statistics for the outgoing screen share.DataChannel: The list of media statistics for the outgoing data channel.
IncomingMediaStatistics: The list of media statistics for incoming media.Audio: The list of media statistics for the incoming audio.Video: The list of media statistics for the incoming video.ScreenShare: The list of media statistics for the incoming screen share.DataChannel: The list of media statistics for the incoming data channel.
LastUpdateAt: The date when the report was generated.
Then, subscribe to the SampleReported event to get regular updates about the current media quality statistics:
mediaStatisticsCallFeature.ReportReceived += MediaStatisticsCallFeature_ReportReceived;
// Optionally, set the interval for media statistics report generation
mediaStatisticsCallFeature.UpdateReportIntervalInSeconds(15);
private void MediaStatisticsCallFeature_ReportReceived(object sender, MediaStatisticsReportReceivedEventArgs args)
// Obtain the media statistics report instance
MediaStatisticsReport report = args.Report;
// Obtain the outgoing media statistics for audio
IReadOnlyList<OutgoingAudioStatistics> outgoingAudioStatistics = report.OutgoingStatistics.Audio;
// Obtain the outgoing media statistics for video
IReadOnlyList<OutgoingVideoStatistics> outgoingVideoStatistics = report.OutgoingStatistics.Video;
// Obtain the outgoing media statistics for screen share
IReadOnlyList<OutgoingScreenShareStatistics> outgoingScreenShareStatistics = report.OutgoingStatistics.ScreenShare;
// Obtain the outgoing media statistics for data channel
IReadOnlyList<OutgoingDataChannelStatistics> outgoingDataChannelStatistics = report.OutgoingStatistics.DataChannel;
// Obtain the incoming media statistics for audio
IReadOnlyList<IncomingAudioStatistics> incomingAudioStatistics = report.IncomingStatistics.Audio;
// Obtain the incoming media statistics for video
IReadOnlyList<IncomingVideoStatistics> incomingVideoStatistics = report.IncomingStatistics.Video;
// Obtain the incoming media statistics for screen share
IReadOnlyList<IncomingScreenShareStatistics> incomingScreenShareStatistics = report.IncomingStatistics.ScreenShare;
// Obtain the incoming media statistics for data channel
IReadOnlyList<IncomingDataChannelStatistics> incomingDataChannelStatistics = report.IncomingStatistics.DataChannel;
}
Best practices
If you want to collect the data for offline inspection, we recommend that you collect the data and send it to your pipeline ingestion after your call ends. If you transmit the data during a call, it could use internet bandwidth needed to continue an Azure Communication Services call (especially when available bandwidth is low).
Outgoing audio metrics
| Metric name | Description | Comments |
|---|---|---|
CodecName |
Codec name | |
BitrateInBps |
Audio send bitrate (bits per second) | General values are in the 24-Kbps range (36-128 Kbps is typical). |
JitterInMs |
Packet jitter (milliseconds) | Lower is better. |
PacketCount |
The total number of packets sent. |
Incoming audio metrics
| Metric name | Description | Comments |
|---|---|---|
CodecName |
Codec name | |
JitterInMs |
Packet jitter (milliseconds) | Lower is better. |
PacketCount |
The total number of packets sent. | |
PacketsLostPerSecond |
Packet loss rate (packets per second) | Lower is better. |
Outgoing video metrics
| Metric name | Description | Comments |
|---|---|---|
CodecName |
Codec name | |
BitrateInBps |
Video send bitrate (bits per second) | |
PacketCount |
The total number of packets sent. | |
FrameRate |
Frame rate sent on the RTP stream (frames per second) | |
FrameWidth |
Frame width of the encoded frame (pixels) | |
FrameHeight |
Frame height of the encoded frame (pixels) |
Incoming video metrics
| Metric name | Description | Comments |
|---|---|---|
CodecName |
Codec name | |
BitrateInBps |
Video receive bitrate (bits per second) | |
JitterInMs |
Packet jitter (milliseconds) | Lower is better. |
PacketCount |
The total number of packets sent. | |
PacketsLostPerSecond |
Packet loss rate (packets per second) | Lower is better. |
StreamId |
Stream ID | The streamId value corresponds to the ID of the video of the remote participant. It can be used to match the sender. |
FrameRate |
Frame rate received on the RTP stream (frames per second) | |
FrameWidth |
Frame width of the decoded frame (pixels) | |
FrameHeight |
Frame height of the decoded frame (pixels) | |
TotalFreezeDurationInMs |
Total freeze duration (milliseconds) |
Outgoing screen share metrics
Currently, statistics fields are the same as Outgoing video metrics.
Incoming screen share metrics
Currently, statistics fields are the same as Incoming video metrics.
Outgoing data channel metrics
| Metric name | Description | Comments |
|---|---|---|
PacketCount |
The total number of packets sent. |
Incoming data channel metrics
| Metric name | Description | Comments |
|---|---|---|
JitterInMs |
Packet jitter (milliseconds) | Lower is better. |
PacketCount |
The total number of packets sent. |