Optional
bitrateCurrent bitrate in bytes per second.
A unique id that is associated with the object that was inspected to produce this StatsBase object.
Optional
jitterPacket Jitter measured in seconds for this SSRC.
Optional
jitterThe purpose of the jitter buffer is to recombine RTP packets into frames (in the case of video) and have smooth playout. The model described here assumes that the samples or frames are still compressed and have not yet been decoded. It is the sum of the time, in seconds, each audio sample or a video frame takes from the time the first packet is received by the jitter buffer (ingest timestamp) to the time it exits the jitter buffer (emit timestamp).
Optional
jitterThe total number of audio samples or video frames that have come out of the jitter buffer (increasing jitterBufferDelay).
Optional
midMedia stream "identification-tag" negotiated and present in the local and remote descriptions.
Optional
mimeThe codec MIME media type/subtype. e.g., video/vp8 or equivalent.
Optional
packetNumber of packets lost since last collection.
Optional
packetThe ratio of packet loss.
Optional
packetCurrent packet rate in packets per second.
The timestamp, associated with this object. The time is relative to the UNIX epoch (Jan 1, 1970, UTC).
Optional
totalTotal number of bytes received for this SSRC.
Optional
totalTotal number of RTP packets lost for this SSRC. Note that because of how this is estimated, it can be negative if more packets are received than sent.
Optional
totalTotal number of packets received for this SSRC.
The value of the MediaStreamTrack's id attribute.
Represents the statistics object for an input audio stream.