Why the Consumer statistics does not include jitter? The provided sample stats in the documentation also confirms that.
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Because calculating jitter from sender side is not easy at all.
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Understand, but Jitter is an important WebRTC metric for diagnosing connectivity issues specially for participants (Consumers). Can we hope to have it in the near future? Or do you have any other suggestion/workaround?
You can get it from local stats in consuming WebRTC devices.
Yeah I tried that. Unfortunately the local stats provides jitter only for audio. For video I’m getting jitterBufferDelay and jitterBufferEmittedCount but not jitter (tested on Chrome). 
We cannot calculate jitter on a sending RTP stream, just that.
Said that, jitter is not important for video since many RTP packets may belong to the same instant of time and there are retransmissions on another RTX stream. That’s why Chrome does not even show “jitter” for received video streams.
Anyway, we cannot do magic.
Thanks for the clarification. 