I'd bet you could do something similar with Meet and Zoom–my understanding is video bitrates for those services are lower than for e.g. Netflix which we showed are much lower than network capacities. But it might be tricky because of the latency-sensitivity angle, and we did not look into it in our paper.
[1] https://www.sandvine.com/hubfs/Sandvine_Redesign_2019/Downlo...
For a given quality, bitrate will generally be higher in RTC apps (though quality may be lower in general depending on the context and network conditions obviously) because of tradeoffs between encoding latency and efficiency. However, RTC apps generally already try to underutilize links because queuing is bad for latency and latency matters a lot for the RTC case.
op used the term presumably to describe "live content" eg. the source material is not available as a whole (because the recording is not finished); which can be considered a subset of "streaming video"
the sensitivity in regard to transport characteristics stems from the fact that "live content" places an upper bound for the time required for processing and transferring the content-bits to the clients (for it to be considered "live").