====== onStats() ======
wrtc.onStats() is called when Chrome browser has collected statistics about received media stream during a call. (Chrome only, never occurs on other browsers.)

You can use this event to display the statistics (mostly used for debug).

===== Occurs =====
You must set [[ixnx:wrtc:js:api:call:adv.stats_ms|call.adv.stats_ms]] to enable wrtc.onStats()

If you have set [[ixnx:wrtc:js:api:call:adv.stats_ms|call.adv.stats_ms]]=1000 wrtc.onStats() will be called once per second during the duration of the call. (Chrome only)

===== Arguments =====
wrtc.onStat(call, arg)
  * call - the [[ixnx:wrtc:js:api:call:start|call]] object raising the event
  * arg: JSON structure containing:
    * timestamp : date and time data was collected
    * remoteAddr : IP address data stream is coming from
    * remotePort : IP port data stream is coming from
    * remoteType : type of connection:
      * local : connected directly to remote peer
      * serverreflexive : address derived using STUN
      * relayed : connected through TURN server
    * videoBytes : number of video stream bytes received since the beginning of the call
    * videoPackets : number of video stream packets received since the beginning of the call
    * videoLost : number of video stream packets lost (should have arrived but never did) since the beginning of the call
    * videoBitRate : current bitrate video stream is received at, in kbps
    * videoWidth : width of the video frame currently received, in pixels
    * videoHeight : height of the video frame currently received, in pixels
    * videoFrameRate : number of video frames per second received (slow hardwares may drop video frames during render – this is the number of frames received)
    * videoJitterBufMs : size of the video jitter buffer, in ms of video stored
    * audioBytes : number of audio stream bytes received since the beginning of the call
    * audioPackets : number of audio stream packets received since the beginning of the call
    * audioLost : number of audio stream packets lost (should have arrived but never did) since the beginning of the call
    * audioBitRate : current bitrate audio stream is received at, in kbps
    * audioLevel : current audio volume (sound level)
    * audioJitter : current audio jitter
All measurements are of the //incoming// media stream.

<WRAP center round tip 100%>
You can use the value audioLevel to detect if the person you are in call with is speaking right now or not. Usually audioLevel>1000 means you are receiving sound.
</WRAP>

===== See Also =====
[[ixnx:wrtc:js:api:call:adv.stats_ms|call.adv.stats_ms]]