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| 1 | +--- |
| 2 | +title: Azure Communication Services Media Stats (Native) |
| 3 | +titleSuffix: An Azure Communication Services concept document |
| 4 | +description: Provides usage samples of the Media Stats feature for Native. |
| 5 | +author: jsaurezle-msft |
| 6 | +ms.author: jsaurezlee |
| 7 | + |
| 8 | +services: azure-communication-services |
| 9 | +ms.date: 08/09/2023 |
| 10 | +ms.topic: include |
| 11 | +ms.service: azure-communication-services |
| 12 | +ms.subservice: calling |
| 13 | +--- |
| 14 | + |
| 15 | +## Best practices |
| 16 | +If you want to collect this data for off-line inspection (after a call ends), it is recommended to collect this data and send it to your pipeline ingest after your call has ended. If you transmit this data during a call, it could use internet bandwidth that is needed to continue an Azure Communication Services call (especially when available bandwidth is low). |
| 17 | + |
| 18 | +### Outgoing Audio metrics |
| 19 | +| Metric Name | Description | Comments | |
| 20 | +| ----------- | ----------- | -------- | |
| 21 | +| CodecName | codec name | | |
| 22 | +| Bitrate | audio send bitrate (bps) | General values are in the 24 kbps range (36-128 kbps typical) | |
| 23 | +| JitterInMs | packet jitter (milliseconds) | Lower is better. | |
| 24 | +| PacketsPerSecond | packet rate (packets/sec) | | |
| 25 | +| PacketsLostPerSecond | packet loss rate (packets/sec) | Lower is better. | |
| 26 | +| RoundTripTimeInMs | round-trip time (milliseconds) | Lower is better. It's calculated from RTCP Receiver Report. A round trip time of 200 ms or less is recommended. | |
| 27 | +| PairRoundTripTimeInMs | round-trip time (milliseconds) | Lower is better. It's similar to rttInMS but is calculated from STUN connectivity check. A round trip time of 200 ms or less is recommended. | |
| 28 | +| AvailableBitrate | bandwidth estimation (bps) | | |
| 29 | +| AudioInputLevel | audio volume level from microphone | The value ranges from 0-65536. 0 represents silence | |
| 30 | + |
| 31 | +### Incoming Audio metrics |
| 32 | +| Metric Name | Description | Comments | |
| 33 | +| ----------- | ----------- | -------- | |
| 34 | +| CodecName | codec name | | |
| 35 | +| Bitrate | audio receive bitrate (bps) | General values are in the 24 kbps range (36-128 kbps typical) | |
| 36 | +| JitterInMs | packet jitter (milliseconds) | Lower is better. | |
| 37 | +| PacketsPerSecond | packet rate (packets/sec) | | |
| 38 | +| PacketsLostPerSecond | packet loss rate (packets/sec) | Lower is better. | |
| 39 | +| RoundTripTimeInMs | round-trip time (milliseconds) | Lower is better. It's calculated from RTCP Receiver Report. A round trip time of 200 ms or less is recommended. | |
| 40 | +| PairRoundTripTimeInMs | round-trip time (milliseconds) | Lower is better. It is calculated from STUN connectivity check. A round trip time of 200 ms or less is recommended. | |
| 41 | +| AvailableBitrate | bandwidth estimation (bps) | | |
| 42 | +| JitterBufferInMs | 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. | |
| 43 | +| AudioOutputLevel | audio volume level from receiving stream | The value ranges from 0-65536. 0 represents silence. | |
| 44 | +| HealedRatio | ratio of concealedSamples(except silentConcealedSamples) to total received samples | Information only. | |
| 45 | + |
| 46 | +### Outgoing Video metrics |
| 47 | +| Metric Name | Description | Comments | |
| 48 | +| ----------- | ----------- | -------- | |
| 49 | +| CodecName | codec name | | |
| 50 | +| Bitrate | video send bitrate (bps) | | |
| 51 | +| JitterInMs | packet jitter (milliseconds) | Lower is better. | |
| 52 | +| PacketsPerSecond | packet rate (packets/sec) | | |
| 53 | +| PacketsLostPerSecond | packet loss rate (packets/sec) | Lower is better. | |
| 54 | +| RoundTripTimeInMs | round-trip time (milliseconds) | Lower is better. It is calculated from RTCP Receiver Report. A round trip time of 200 ms or less is recommended. | |
| 55 | +| PairRoundTripTimeInMs | round-trip time (milliseconds) | Lower is better. It is similar to rttInMS but is calculated from STUN connectivity check. A round trip time of 200 ms or less is recommended. | |
| 56 | +| AvailableBitrate | bandwidth estimation (bps) | 1.5 Mbps or higher is recommended for high-quality video for upload/download. | |
| 57 | +| FrameRateInput | frame rate originating from the video source (frames/sec) | | |
| 58 | +| FrameWidthInput | frame width of the last frame originating from video source (pixel) | | |
| 59 | +| FrameHeightInput | frame height of the last frame originating from video source (pixel) | | |
| 60 | +| FrameRateEncoded | frame rate successfully encoded for the RTP stream (frames/sec) | | |
| 61 | +| FrameRateSent | frame rate sent on the RTP stream (frames/sec) | | |
| 62 | +| FrameWidthSent | frame width of the encoded frame (pixel) | | |
| 63 | +| FrameHeightSent | frame height of the encoded frame (pixel) | | |
| 64 | +| FramesSent | frames sent on the RTP stream | | |
| 65 | +| FramesEncoded | frames successfully encoded for the RTP stream | | |
| 66 | +| KeyFramesEncoded | key frames successfully encoded for the RTP stream | | |
| 67 | + |
| 68 | +### Incoming Video metrics |
| 69 | +| Metric Name | Description | Comments | |
| 70 | +| ----------- | ----------- | -------- | |
| 71 | +| CodecName | codec name | | |
| 72 | +| Bitrate | video receive bitrate (bps) | | |
| 73 | +| JitterInMs | packet jitter (milliseconds) | Lower is better. | |
| 74 | +| PacketsPerSecond | packet rate (packets/sec) | | |
| 75 | +| PacketsLostPerSecond | packet loss rate (packets/sec) | Lower is better. | |
| 76 | +| RoundTripTimeInMs | round-trip time (milliseconds) | Lower is better. It is calculated from RTCP Receiver Report. A round trip time of 200 ms or less is recommended. | |
| 77 | +| PairRoundTripTimeInMs | round-trip time (milliseconds) | Lower is better. A round trip time of 200 ms or less is recommended. | |
| 78 | +| AvailableBitrate | bandwidth estimation (bps) | 1.5 Mbps or higher is recommended for high-quality video for upload/download. | |
| 79 | +| JitterBufferInMs | jitter buffer (milliseconds) | Lower is better. The jitter buffer is used for smooth playout. This value is how long the packets of the frame stay in the jitter buffer. | |
| 80 | +| StreamId | stream id | The streamId value corresponds to id in VideoStreamCommon. It can be used to match the sender. | |
| 81 | +| FrameRateOutput | frame rate output (frames/sec) | | |
| 82 | +| FrameRateDecoded | frame rate correctly decoded for the RTP stream (frames/sec) | | |
| 83 | +| FrameRateReceived | frame rate received on the RTP stream (frames/sec) | | |
| 84 | +| FrameWidthReceived | frame width of the decoded frame (pixel) | | |
| 85 | +| FrameHeightReceived | frame height of the decoded frame (pixel) | | |
| 86 | +| LongestFreezeDurationInMs | longest freeze duration (milliseconds) | | |
| 87 | +| TotalFreezeDurationInMs | total freeze duration (milliseconds) | | |
| 88 | +| FramesReceived | total number of frames received on the RTP stream | | |
| 89 | +| FramesDecoded | total number of frames correctly decoded for the RTP stream | | |
| 90 | +| FramesDropped | total number of frames dropped | | |
| 91 | +| KeyFramesDecoded | total number of key frames correctly decoded for the RTP stream | | |
| 92 | + |
| 93 | +### ScreenShare Send metrics |
| 94 | +Currently stats fields are the same as *Video Send metrics* |
| 95 | + |
| 96 | +### ScreenShare Receive metrics |
| 97 | +Currently stats fields are the same as *Video Receive metrics* |
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