update

Author: amagginetti

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

@@ -114,7 +114,14 @@ For each endpoint within a call, a distinct call diagnostic log is created for o
 |     `jitterMax`             |     The maximum jitter value measured between packets for each media stream. Bursts in network conditions can cause problems in the audio/video traffic flow.  |
 |     `packetLossRateAvg`     |     The average percentage of packets that are lost. Packet loss directly affects audio quality. Small, individual lost packets have almost no impact, whereas back-to-back burst losses cause audio to cut out completely. The packets being dropped and not arriving at their intended destination cause gaps in the media. This situation results in missed syllables and words, along with choppy video and sharing. <br><br>A packet loss rate of greater than 10% (0.1) is likely having a negative quality impact. This metric is measured for each media stream over the `participantDuration` period in a group call or over the `callDuration` period in a P2P call.    |
 |     `packetLossRateMax`     |     This value represents the maximum packet loss rate (percentage) for each media stream over the `participantDuration` period in a group call or over the `callDuration` period in a P2P call. Bursts in network conditions can cause problems in the audio/video traffic flow.
-|     `Inputnewvalue`     |     This.
+|     `JitterBufferSizeAvg`     |     The average size of jitter buffer over the duration of each media stream. A jitter buffer is a shared data area where voice packets can be collected, stored, and sent to the voice processor in evenly spaced intervals. Jitter buffer is used to counter the effects of jitter. <br><br> Jitter buffers can be either static or dynamic. Static jitter buffers are set to a fixed size, while dynamic jitter buffers can adjust their size based on network conditions. The goal of the jitter buffer is to provide a smooth and uninterrupted stream of audio and video data to the user. <br><br> In the web SDK, this 'JitterBufferSizeAvg' is the average value of the 'jitterBufferDelay' during the call, the 'jitterBufferDelay' is the duration of an audio sample or a video frame that stays in the jitter buffer. <br><br> Normally when 'JitterBufferSizeAvg' value is greater than 200 ms, it will cause a negative quality impact.   
+|     `JitterBufferSizeMax`     |    The maximum jitter buffer size measured during the duration of each media stream.  <br><br> Normally when this value is greater than 200 ms, it will cause a negative quality impact.  
+|     `HealedDataRatioAvg`     |    The average percentage of lost or damaged data packets that are successfully reconstructed or recovered by the healer over the duration of audio stream. Healed data ratio is a measure of the effectiveness of error correction techniques used in VoIP systems.  <br><br> When this value is greater than 0.1 (10%),  we consider the stream as bad quality. 
+|     `HealedDataRatioMax`     |    The maximum healed data ratio measured during the duration of each media stream. <br><br> When this value is greater than 0.1 (10%),  we consider the stream as bad quality. 
+|     `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. 
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 ### P2P vs. group calls