Code change:
4.3. <ins> Finally, the "On-Demand" use cases described in this section do not have a tight linkage between ingest and streaming, allowing significant transcoding, processing, insertion of video clips in a news article, etc.  The latency constraints for the use cases in this section may be dominated by the time required for whatever actions are required before media are available for streaming. </ins> 4.3.1. *Senders/Receivers*: One to Many *Bi-directional*: No *Latency*: On Demand Where media is ingested and processed for a system to later serve it <del> to clients as on-demand media.  This may be media captured from live output or provided as a pre-recorded source, and may optionally be transcoded upon ingest. </del> <ins> to clients as on-demand media.  This media provided from a pre- recorded source, or captured from live output, but in either case, this media is not immediately passed to viewers, but is stored for "on-demand" retrieval, and may be transcoded upon ingest. </ins> 4.3.2.

Related discussion:
text: I think the first point illustrates nicely why protocols like RTP have low-level control over packetization, while the existing MoQ proposals defer it entirely to the QUIC stack.
NAME - that's super helpful. It helps us explain a few things - why we need to know this, in order to explain why we're shooting for One Protocol, and why we think the first three use cases in the draft are different (they're all RTPish), and why we're not looking at RTP in the BOF - right now, the draft just says AVTCORE is looking at RTP over QUIC, which really isn't much of a justification, engineering-wise.
Victor's points are well made, though I think picking the boundary of the number of round trips necessary to rely on standard QUIC loss detection is a challenge. 5 RTTs provides almost 5 opportunities to repair a loss, which in my experience is almost always enough. Similarly, I'd expect it's enough for BBR to complete various probing cycles. It's also important to understand what RTT we have in mind here. I think the RTT used to understand what domain you're in is approximately smoothed RTT (not min RTT), because the number of rounds for loss recovery or BBR gain cycling or Cubic packet conservation matter the most.
I agree that the numeric ranges are not useful when designing a protocol but they're useful enough as product requirements. However focusing on the RTT completely misses the point. Network latency involves fixed delays (min RTT) but most of it is caused by queuing (primarily congestion control). This includes any queues within the application, the network layer, intermediate routers, and even the receiver. You can think of latency as the "frame RTT" which is decoupled from the packet RTT. Of course the packet RTT is important, but it just doesn't factor in how much data has been queued. For example, it's just useless to talk about RTMP and HLS/DASH latency in terms of network RTTs. In my opinion, the problem with those protocols is that they impose unnecessary dependencies, making it slow to drop the bitrate to reduce queue sizes. Sending media over a single stream means that every frame depends on every previous frame regardless of the GoP structure (no frame dropping), and it also means that new GoPs are dependent on the previous GoPs (no skipping ahead).