Total number of bytes received within the audio stream.
Audio codec name for the audio stream, e.g. "opus". Only in Chrome
Average time packets stay in a jitter buffer (temporary storage buffer used to capture incoming data packets). It is used to ensure the continuity of streams by smoothing out packet arrival times during periods of network congestion. It's calculated as jitterBufferDelay for the last Config.rtcStatsCollectionInterval(sum of the time each frame takes from the time it is received and to the time it exits the jitter buffer) divided by jitterBufferEmittedCount for the same interval (total number of frames that have come out of the jitter buffer). Measured in milliseconds. Only in Chrome
Packet loss in the audio stream. Values are in the range 0..1, where 0 means no loss and 1 means full loss.
Total number of audio packets lost for the audio stream.
Total number of packets received within the audio stream.
The time at which the call statistics are collected (in UNIX timestamp format).