Merge pull request #278695 from kuipan-msft/kuipan/update_ACS_logs

update ACS CallSummary and CallDiagnostics logs

Author: prmerger-automator[bot]

articles/communication-services/concepts/analytics/logs/voice-and-video-logs.md

@@ -85,6 +85,8 @@ The call summary log contains data to help you identify key properties of all ca
 |     `participantTenantId`               |    The ID of the Microsoft tenant associated with the identity of the participant. The tenant can either be the Azure tenant that owns the Azure Communication Services resource or the Microsoft tenant of an M365 identity. This field is used to guide cross-tenant redaction.
 |`participantType` | 	Description of the participant as a combination of its client (Azure Communication Services or Teams), and its identity, (Azure Communication Services or Microsoft 365). Possible values include: Azure Communication Services (Azure Communication Services identity and Azure Communication Services SDK), Teams (Teams identity and Teams client), Azure Communication Services as Teams external user (Azure Communication Services identity and Azure Communication Services SDK in Teams call or meeting), Azure Communication Services as Microsoft 365 user (M365 identity and Azure Communication Services client), and Teams Voice Apps.
 | `pstnPartcipantCallType `|It represents the type and direction of PSTN participants including Emergency calling, direct routing, transfer, forwarding, etc.| 
+| `ParticipantEndSubCode `| it represents the Calling SDK error subcode that the SDK emits (when relevant) for each `participantId` value.| 
+| `ResultCategory `|It represents the category of the participant ending the call. It can be one of these 4 values: Success, ExpectedError, UnexpectedClientError, UnexpectedServerError. | 
 
 ### Call diagnostic log schema
 
@@ -121,6 +123,11 @@ For each endpoint within a call, a distinct call diagnostic log is created for o
 |     `VideoFrameRateAvg`     |    The average number of video frames that are transmitted per second during a video/screensharing call. The video frame rate can impact the quality and smoothness of the video stream, with higher frame rates generally resulting in smoother and more fluid motion. The standard frame rate for WebRTC video is typically 30 frames per second (fps), although this can vary depending on the specific implementation and network conditions.  <br><br> The stream quality is considered poor when this value is less than 7 for video stream, or less than 1 for screensharing stream. 
 |     `RecvResolutionHeight`     |    The average of vertical size of the incoming video stream that is transmitted during a video/screensharing call. It's measured in pixels and is one of the factors that determines the overall resolution and quality of the video stream. The specific resolution used may depend on the capabilities of the devices and network conditions involved in the call.  <br><br> The stream quality is considered poor when this value is less than 240 for video stream, or less than 768 for screensharing stream. 
 |     `RecvFreezeDurationPerMinuteInMs`     |    The average freeze duration in milliseconds per minute for incoming video/screensharing stream. Freezes are typically due to bad network condition and can degrade the stream quality.  <br><br> The stream quality is considered poor when this value is greater than 6,000 ms for video stream, or greater than 25,000 ms for screensharing stream. 
+|     `PacketUtilization`     |    The packets sent or received for a given media stream.  <br><br> Usually the longer the call, the higher the value is. If this value is zero, it could indicate that media is not flowing. 
+|     `VideoBitRateAvg`     |    The average bitrate (bits pers second) for a video or screenshare stream.  <br><br> A low bitrate value could indicate poor network issue. The minimum bitrate (bandwidth) required can be found here: [Network bandwidth](../../voice-video-calling/network-requirements.md#network-bandwidth).
+|     `VideoBitRateMax`     |    The maximum bitrate (bits pers second) for a video or screenshare stream.  <br><br> A low bitrate value could indicate poor network issue. The minimum bitrate (bandwidth) required can be found here: [Network bandwidth](../../voice-video-calling/network-requirements.md#network-bandwidth).
+|     `StreamDirection`     |    The direction of the media stream. It is either Inbound or Outbound.
+|     `CodecName`     |    The name of the codec used for processing media streams. It can be OPUS, G722, H264S, SATIN etc.
 
 ### Call client operations log schema