Code change:
From NAME
NAME and NAME - I agree with Magnus in his email (cut and pasted in the description), where he said "what about RMTP and SRT?", but I'm thinking that since we already have a "Why QUIC?" section, we should probably have a "Why NOT other protocols" section, rather than trying to talk about that in (what was) Section 6.1. That could also conveniently map into saying why we're not looking at "Interactive Media" in this document (and, transitively, in the BOF), and why we're not looking at "On Demand" (Media) in this document (and in the BOF). Please let me know if this is a BAD idea soon (I'm typing PRs).
NAME for media ingest, Magnus has a good point about covering not just HLS and DASH, but also making (even brief) reference to non-segmented protocols like RTMP and SRT (but really, this can be expanded to RIST, ZiXi, etc etc etc). This can just be a sentence or two; I'm happy to write it.
NAME - I had a complicated day yesterday, which became less complicated last night, so I'm not coherent enough to type text. I'll see what I can do in the next couple of hours, and if I can include these NAME suggestions.
The current definition of latency targets in the draft is based on specific numeric ranges. Those do make sense (since they ultimately come from use case requirements), but I believe we should focus more on what consequences those have for the protocol design. Reflecting on our experience with QUIC used for live video vs QUIC used for video conferencing, my proposed definition of the latency target for live video would be "latency an order of magnitude higher than one RTT". Two major examples of why this is important: When the target latency is on the order of one RTT, it makes sense to use FEC and codec-level packet loss concealment. When the target latency is in the range of many, many RTTs, selectively retransmitting only lost packets is almost always better than FEC. When the target latency is on the order of one RTT, it is impossible to use congestion control schemes like BBR, since BBR has probing mechanisms that rely on temporarily inducing delay and amortizing the consequences of that over multiple RTTs. 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). WebRTC avoids these unnecessary dependencies which is how it's able to rapidly respond to congestion to keep latency low. I've been working on a "latency theory" document if any of these ideas are resonating with people. I feel like it's too early though, and we should be focusing on the use-cases and requirements instead.
NAME - please let me know when you think it's not too early to focus on "latency theory". I'd love to know more (in particular, I'd love to know if I'd be able to contribute effectively to that important topic). Just filing this away for now, as you suggest.

Related discussion:
text: 4.1.3. *Senders/Receivers*: Many to Many *Bi-directional*: Yes *Latency*: <del> Ull-200 </del> <ins> Ull-50 to Ull-200 </ins> Where media is both sent and received; This may include audio from both microphone(s) or other inputs, or may include "screen sharing" or inclusion of other content such as slide, document, or video presentation.  This may be done as client/server, or peer to peer <del> with a many to many relationship of both senders and receivers. </del> <ins> with a many to many relationship of both senders and receivers.  The target for latency may be as large as Ull-200 for some media types such as audio, but other media types in this use case have much more stringent latency targets. </ins> 4.2. <ins> The use cases in this section, unlike the use cases described in interact, still have "humans in the loop", but these humans expect media to be "responsive", where the responsiveness is more on the order of 5 to 10 RTTs.  This allows the use of protocol mechanisms that require more than one or two RTTs - as noted in interact, end- to-end recovery from packet loss and congestion avoidance are two such protocol mechanisms that can be used with Live Media. To illustrate the difference, the responsiveness expected with videoconferencing is much greater than watching a video, even if the video is being produced "live" and sent to a platform for syndication and distribution. </ins> 4.2.1. *Senders/Receivers*: One to One *Bi-directional*: No *Latency*: