This article will describe key metrics for voice workloads in more detail.

Round Trip Latency

  • Measures the time taken to send a data packet from the beacon server to the remote endpoint (Remote IP). Round trip time is affected by the physical distance between 2 endpoints and the transmission speed and any overhead on the routes in-between.  
  • The latency impacts the call quality between 2 people.  
  • Too much latency causes breaks and gaps in voice conversations.  
  • When you are on the phone and people end up talking at the same time, the likely cause is too much latency.  

Packet Reorder Ratio

  • Packet reordering is important because excessive reordering of packets will also affect call quality.  
  • When packets arrive in a different order than they were send it can be seen as packet loss or network congestion.  
  • This metric is also used to calculate the Mean Opinion Score. 
  • Packet reordering can impact packet send rate which will increase round trip time.  
  • Calls can be distorted and cut out at times.  

Packet Loss Rate

  • Microsoft recommends to keep packet loss less than 1% during a 15 second call.  
  • If packet loss is less than 3%, acceptable call quality can be maintained. Thus the default threshold for a healthy scan is packet loss < 3%.  
  • Packet loss rate is used in the Mean Opinion Score calculation. 
  • Excessive packet loss during a call will result in degraded voice quality and call attendees may sound like a robot.  

Mean Opinion Score

MOS is a prediction of end-user audio quality experience. Multiple factors are considered in calculating the MOS (Mean Opinion Score). The score ranges from 1-5. The highest score is usually around 4.4 because of the audio codec in use.  
A MOS < 3 will result in poor call quality 
A MOS < 2 will result in critical reduction in call quality.   

Degradation Average

  • This metric shows the impact of jitter and packet loss. This value should always be less than 1 for acceptable user experience.  
  • If you see degradation here, you will encounter audio distortion. This is the result of network congestion or insufficient bandwidth, which impacts packet loss and jitter.  

Average Jitter

  • Audio packets are sent at regular intervals. Sometimes they are not received with the same intervals (usually because of network latency).  
  • The buffer waits for all packets before reconstructing them in the correct order.  
  • Jitter is the size of the buffer that is needed to store packets before reconstructing them.  
  • Jitter value is calculated over a 15s period.  
  • Low jitter means that the connection to the call is healthy.  
  • Medium / High jitter is a sign of network congestion.  
  • Jitter is also used in the Mean Opinion Score calculation.  
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