In Lync 2013 we changed how we classify poor calls in QoE. The flag is called ClassifiedPoorCall and it is now available in the MediaLine table. It is moved here, because we now support classifying poor calls for audio, video and application sharing.
The conditions we use to classify poor calls are shown in the 3 tables below. The poor call flag is set if one or more the conditions are met. Please note that a record in the MediaLine table can cover multiple media streams. The flagging occurs on the MediaLine level, so if you want to understand specifically which stream was the reason for the classification you need to look at the individual streams and use the columns in the tables below.
Column in AudioStream Table
Network MOS Degradation for the whole call. This metric shows the amount the Network MOS was reduced because of jitter and packet loss
Round trip time
The packet loss rate
Average network jitter
Average ratio of concealed samples generated by audio healing to typical samples
Column in VideoStream Table
The packet loss rate after forward error correction has been applied
The percentage of total video frames that are lost
Average video frame rate used by the receiver
Percentage of the call below the low frame rate threshold
The average video frame rate received during the call
The average video frame rate sent during the call
Percentage of the call where the client experienced high CPU load when processing video
Column in AppSharingStream Table
This value is the percentage of the content from the sharer that did not reach the viewer. Content may be discarded (or spoiled) when the sharer discards tiles from the graphics source or the ASMCU tiles discards tiles from Sharer respectively.
Acceptable value of the average RDP tile processing latency in the AS Conferencing Server over the duration of the viewing session
Optimal value for the relative one-way delay between the two media endpoints involved in the application sharing. This is a single-hop latency measure
Thanks for the article, quite informative.
With the generally adopted value of 150ms max RTT for an optimal VoIP call, isn't the >500ms roundtrip value a bit generous?
Jens>Hi Alessio. You might be right, but for now we are sticking with 500 ms.
Thank you. Great to know!
The 500ms rating is for RTP, while the commonly accepted 150ms is for UDP packets, seen as testing with PING.