jitter(ms)
remote-inbound-rtpoutboundaudiovideo
Packet Jitter measured in milliseconds for this SSRC.
Description
Real number; in milliseconds
Packet Jitter measured in seconds for this SSRC. Calculated as defined in RFC3550 section 6.4.1.
It is an estimate of the statistical variance of the RTP data packet interarrival time, measured in timestamp units.
For rtcStats, we convert the value from seconds to milliseconds to make it easier to read and understand.
This is the metric reported for the outgoing media stream from the remote peer. For the incoming metric see jitter.
See also
Notes
- The higher the jitter value is, the worse the media quality is expected to be and the bigger the playout delay will need to be to account for it
- WebRTC getStats() returns jitter in seconds. For the purpose of visualization, we convert it to milliseconds in rtcStats