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
}