Merge pull request #248526 from jsaurezlee-msft/main

Update available Media Stats table for native platforms (#9)

Author: American-Dipper

articles/communication-services/concepts/voice-video-calling/includes/media-stats/media-stats-native-metrics.md

@@ -13,7 +13,7 @@ ms.subservice: calling
 ---
 
 ## Best practices
-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).
+If you want to collect this data for off-line inspection (after a call ends), it's 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).
 
 ### Outgoing Audio metrics
 | Metric Name | Description | Comments |
@@ -22,48 +22,32 @@ If you want to collect this data for off-line inspection (after a call ends), it
 | Bitrate | audio send bitrate (bps) | General values are in the 24 kbps range (36-128 kbps typical) |
 | JitterInMs | packet jitter (milliseconds) | Lower is better. |
 | PacketsPerSecond | packet rate (packets/sec) | |
-| PacketsLostPerSecond | packet loss rate (packets/sec) | Lower is better. |
 | 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. |
-| 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. |
 | AvailableBitrate | bandwidth estimation (bps) | |
-| AudioInputLevel | audio volume level from microphone | The value ranges from 0-65536. 0 represents silence |
 
 ### Incoming Audio metrics
 | Metric Name | Description | Comments |
 | ----------- | ----------- | -------- |
 | CodecName | codec name | |
-| Bitrate | audio receive bitrate (bps) | General values are in the 24 kbps range (36-128 kbps typical) |
 | JitterInMs | packet jitter (milliseconds) | Lower is better. |
 | PacketsPerSecond | packet rate (packets/sec) | |
 | PacketsLostPerSecond | packet loss rate (packets/sec) | Lower is better. |
-| 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. |
-| 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. |
-| AvailableBitrate | bandwidth estimation (bps) | |
-| 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. |
-| AudioOutputLevel | audio volume level from receiving stream | The value ranges from 0-65536. 0 represents silence. |
-| HealedRatio | ratio of concealedSamples(except silentConcealedSamples) to total received samples | Information only. |
 
 ### Outgoing Video metrics
 | Metric Name | Description | Comments |
 | ----------- | ----------- | -------- |
 | CodecName | codec name | |
 | Bitrate | video send bitrate (bps) | |
-| JitterInMs | packet jitter (milliseconds) | Lower is better. |
 | PacketsPerSecond | packet rate (packets/sec) | |
 | PacketsLostPerSecond | packet loss rate (packets/sec) | Lower is better. |
-| 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. |
-| 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. |
+| 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. |
 | AvailableBitrate | bandwidth estimation (bps) | 1.5 Mbps or higher is recommended for high-quality video for upload/download. |
 | FrameRateInput | frame rate originating from the video source (frames/sec) | |
 | FrameWidthInput | frame width of the last frame originating from video source (pixel) | |
 | FrameHeightInput | frame height of the last frame originating from video source (pixel) | |
-| FrameRateEncoded | frame rate successfully encoded for the RTP stream (frames/sec) | |
 | FrameRateSent | frame rate sent on the RTP stream (frames/sec) | |
 | FrameWidthSent | frame width of the encoded frame (pixel) | |
 | FrameHeightSent | frame height of the encoded frame (pixel) | |
-| FramesSent | frames sent on the RTP stream | |
-| FramesEncoded | frames successfully encoded for the RTP stream | |
-| KeyFramesEncoded | key frames successfully encoded for the RTP stream  | |
 
 ### Incoming Video metrics
 | Metric Name | Description | Comments |
@@ -73,22 +57,11 @@ If you want to collect this data for off-line inspection (after a call ends), it
 | JitterInMs | packet jitter (milliseconds) | Lower is better. |
 | PacketsPerSecond | packet rate (packets/sec) | |
 | PacketsLostPerSecond | packet loss rate (packets/sec) | Lower is better. |
-| 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. |
-| PairRoundTripTimeInMs | round-trip time (milliseconds) | Lower is better. A round trip time of 200 ms or less is recommended. |
-| AvailableBitrate | bandwidth estimation (bps) | 1.5 Mbps or higher is recommended for high-quality video for upload/download. |
-| 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. |
-| StreamId | stream id | The streamId value corresponds to id in VideoStreamCommon. It can be used to match the sender. |
-| FrameRateOutput | frame rate output (frames/sec) | |
-| FrameRateDecoded | frame rate correctly decoded for the RTP stream (frames/sec) | |
+| StreamId | stream ID | The streamId value corresponds to the ID of the video of the RemoteParticipant. It can be used to match the sender. |
 | FrameRateReceived | frame rate received on the RTP stream (frames/sec) | |
 | 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 | |
 
 ### ScreenShare Send metrics
 Currently stats fields are the same as *Video Send metrics*