title: how to determine quality of connectivity from webrtc stats?
body: <p>I want to automatically change the video resolution, if needed( firefox is yet to support bandwidth manipulation), change to audio only mode, based on the quality of the connection. </p>
  
  <p>I get the connections stats through <code>PeerConnection.getStats</code>, but I am not sure what attributes to use to determine the quality. For starters, firefox and chrome give different attributes in the callback.</p>
  
  <p>I do not have a formal education on media manipulation, so only have a rudimentary idea about most of the attributes.</p>
  
  <p>After a little experimenting, for good quality of connection, I am setting the standard as:</p>
  
  <p>in Chrome:</p>
  
  <pre><code>googJitterBufferMs &lt; 200  
  packetsLost/packetsSent &lt; .1   // loss must be less than 10%
  </code></pre>
  
  <p>and Firefox:</p>
  
  <pre><code>droppedFrames/packetsSent &lt; .1   // loss must be less than 10%
  </code></pre>
  
  <p>Am I choosing the right attributes? are my assumptions correct?</p>