mediaStream resource type

Namespace: microsoft.graph.callRecords

Represents information about a media stream between two endpoints in a call.

Properties

Property	Type	Description
audioCodec	microsoft.graph.callRecords.audioCodec	Codec name used to encode audio for transmission on the network. Possible values are: unknown, invalid, cn, pcma, pcmu, amrWide, g722, g7221, g7221c, g729, multiChannelAudio, muchv2, opus, satin, satinFullband, rtAudio8, rtAudio16, silk, silkNarrow, silkWide, siren, xmsRta, unknownFutureValue.
averageAudioDegradation	Double	Average Network Mean Opinion Score degradation for stream. Represents how much the network loss and jitter has impacted the quality of received audio.
averageAudioNetworkJitter	Duration	Average jitter for the stream computed as specified in RFC 3550, denoted in ISO 8601 format. For example, 1 second is denoted as 'PT1S', where 'P' is the duration designator, 'T' is the time designator, and 'S' is the second designator.
averageBandwidthEstimate	Int64	Average estimated bandwidth available between two endpoints in bits per second.
averageFreezeDuration	Duration	Average duration of the received freezing time in the video stream.
averageJitter	Duration	Average jitter for the stream computed as specified in RFC 3550, denoted in ISO 8601 format. For example, 1 second is denoted as 'PT1S', where 'P' is the duration designator, 'T' is the time designator, and 'S' is the second designator.
averagePacketLossRate	Double	Average packet loss rate for stream.
averageRatioOfConcealedSamples	Double	Ratio of the number of audio frames with samples generated by packet loss concealment to the total number of audio frames.
averageReceivedFrameRate	Double	Average frames per second received for all video streams computed over the duration of the session.
averageRoundTripTime	Duration	Average network propagation round-trip time computed as specified in RFC 3550, denoted in ISO 8601 format. For example, 1 second is denoted as 'PT1S', where 'P' is the duration designator, 'T' is the time designator, and 'S' is the second designator.
averageVideoFrameLossPercentage	Double	Average percentage of video frames lost as displayed to the user.
averageVideoFrameRate	Double	Average frames per second received for a video stream, computed over the duration of the session.
averageVideoPacketLossRate	Double	Average fraction of packets lost, as specified in RFC 3550, computed over the duration of the session.
endDateTime	DateTimeOffset	UTC time when the stream ended. The DateTimeOffset type represents date and time information using ISO 8601 format and is always in UTC time. For example, midnight UTC on Jan 1, 2014 is 2014-01-01T00:00:00Z. This field is only available for streams that use the SIP protocol.
isAudioForwardErrorCorrectionUsed	Boolean	Indicates whether the forward error correction (FEC) was used at some point during the session. The default value is null.
lowFrameRateRatio	Double	Fraction of the call where frame rate is less than 7.5 frames per second.
lowVideoProcessingCapabilityRatio	Double	Fraction of the call that the client is running less than 70% expected video processing capability.
maxAudioNetworkJitter	Duration	Maximum of audio network jitter computed over each of the 20 second windows during the session, denoted in ISO 8601 format. For example, 1 second is denoted as 'PT1S', where 'P' is the duration designator, 'T' is the time designator, and 'S' is the second designator.
maxJitter	Duration	Maximum jitter for the stream computed as specified in RFC 3550, denoted in ISO 8601 format. For example, 1 second is denoted as 'PT1S', where 'P' is the duration designator, 'T' is the time designator, and 'S' is the second designator.
maxPacketLossRate	Double	Maximum packet loss rate for the stream.
maxRatioOfConcealedSamples	Double	Maximum ratio of packets concealed by the healer.
maxRoundTripTime	Duration	Maximum network propagation round-trip time computed as specified in RFC 3550, denoted in ISO 8601 format. For example, 1 second is denoted as 'PT1S', where 'P' is the duration designator, 'T' is the time designator, and 'S' is the second designator.
packetUtilization	Int64	Packet count for the stream.
postForwardErrorCorrectionPacketLossRate	Double	Packet loss rate after FEC has been applied aggregated across all video streams and codecs.
rmsFreezeDuration	Duration	Average duration of the received freezing time in the video stream represented in root mean square.
startDateTime	DateTimeOffset	UTC time when the stream started. The DateTimeOffset type represents date and time information using ISO 8601 format and is always in UTC time. For example, midnight UTC on Jan 1, 2014 is 2014-01-01T00:00:00Z. This field is only available for streams that use the SIP protocol.
streamDirection	microsoft.graph.callRecords.mediaStreamDirection	Indicates the direction of the media stream. Possible values are: callerToCallee, calleeToCaller.
streamId	String	Unique identifier for the stream.
videoCodec	microsoft.graph.callRecords.videoCodec	Codec name used to encode video for transmission on the network. Possible values are: unknown, invalid, av1, h263, h264, h264s, h264uc, h265, rtvc1, rtVideo, xrtvc1, unknownFutureValue.
wasMediaBypassed	Boolean	True if the media stream bypassed the Mediation Server and went straight between client and PSTN Gateway/PBX, false otherwise.

JSON representation

The following JSON representation shows the resource type.

{
  "audioCodec": "String",
  "averageAudioDegradation": "Double",
  "averageAudioNetworkJitter": "String (duration)",
  "averageBandwidthEstimate": 1024,
  "averageFreezeDuration": "String (duration)",
  "averageJitter": "String (duration)",
  "averagePacketLossRate": "Double",
  "averageRatioOfConcealedSamples": "Double",
  "averageReceivedFrameRate": "Double",
  "averageRoundTripTime": "String (duration)",
  "averageVideoFrameLossPercentage": "Double",
  "averageVideoFrameRate": "Double",
  "averageVideoPacketLossRate": "Double",
  "endDateTime": "String (timestamp)",
  "isAudioForwardErrorCorrectionUsed": "Boolean",
  "lowFrameRateRatio": "Double",
  "lowVideoProcessingCapabilityRatio": "Double",
  "maxAudioNetworkJitter": "String (duration)",
  "maxJitter": "String (duration)",
  "maxPacketLossRate": "Double",
  "maxRatioOfConcealedSamples": "Double",
  "maxRoundTripTime": "String (duration)",
  "packetUtilization": 1024,
  "postForwardErrorCorrectionPacketLossRate": "Double",
  "rmsFreezeDuration": "String (duration)",
  "startDateTime": "String (timestamp)",
  "streamDirection": "String",
  "streamId": "String",
  "videoCodec": "String",
  "wasMediaBypassed": true
}