Original text:
5.5.2. <del> In the event of a lost packet on a TCP connection with SACK support (a common case for segmented delivery in practice), loss of a packet can provide a confusing bandwidth signal to the receiving </del>  application.  Because of the sliding window in TCP, many packets may be accepted by the receiver without being available to the application until the missing packet arrives.  Upon the arrival of get access to a lot of data at the same time. To a receiver measuring bytes received per unit time at the <del> application layer, and interpreting it as an estimate of the available network bandwidth, this appears as a high jitter in the goodput measurement, presenting as a stall, followed by a sudden leap that can far exceed the actual capacity of the transport path from the server when the hole in the received data is filled by a later </del>  retransmission. <del> It is worth noting that more modern transport protocols such as QUIC have mitigation of head-of-line blocking as a protocol design goal. See quic-behavior for more details. </del> 5.5.3. As many end devices have moved to wireless connections for the final high-bandwidth traffic such as streaming media over the UDP ports that HTTP/3 is using. <del> As noted in hol-blocking, because TCP provides a reliable, in-order delivery service for applications, any packet loss for a TCP </del>  connection causes head-of-line blocking, so that no TCP segments arriving after a packet is lost will be delivered to the receiving application until retransmission of the lost packet has been

Reviewer comments:
attempted fix for --the section is saying noisy measurements are a concern, not that head of line blocking per se is a concern.
From NAME 's review (URL) Section 5.5.2 Head-of-Line Blocking If Head-of-Line Blocking is indeed a relevant operational problem, it would be useful to add a corresponding reference (e.g., with measurements).
Also mentioned in section 6.3 about QUIC.
NAME 'said benefit from backing by a reference.
Fig. 1 from URL
Change the section title for 5.5.2: it's not really about head-of-line blocking specifically, it's about a noisy goodput signal when measuring data received at the application layer, which requires appropriate smoothing in the player's bitrate detection algorithm.
The text in 6.3 isn't about head-of-line blocking, either.
Discussion on editor conference call: we really like the diagram Ali provided ... Reviewer: Michael Scharf Review result: Ready with Issues This document has been reviewed as part of the transport area review team's ongoing effort to review key IETF documents. These comments were written primarily for the transport area directors, but are copied to the document's authors and WG to allow them to address any issues raised and also to the IETF discussion list for information. When done at the time of IETF Last Call, the authors should consider this review as part of the last-call comments they receive. Please always CC EMAIL if you reply to or forward this review. This informational document surveys network and transport protocol issues that affect quality of experience for streaming applications, in particular video. The overview covers many topics across different technologies. Readers may only be familiar with a subset of these topics and could therefore learn quite a bit from this kind of tutorial. Nonetheless, it remains somewhat unclear what the actual objective of some of the text is. For some topics, quite a bit of technical background is provided, but the discussion is not really comprehensive. In many sections, the document neither derives technical challenges, nor system/protocol requirements, nor practical usage guidance. It is just a kind of tutorial. Quite a bit of text also presents ongoing IETF work, and an RFC with this scope may thus get outdated soon. Section 6.2 deals with topics owned by TCPM. In similar past documents, I have asked for a TCPM presentation prior to an IETF last call in order to ensure that owners of running code are in the loop. I believe this strategy has worked well in the past. Having said this, as TSV-ART reviewer I don't strongly disagree with a publication. All these issues may just be OK for an informational RFC. Some more specific comments: Section 3.2.1. Recognizing Changes from an Expected Baseline This section apparently assumes relatively static path properties, e.g., fixed network connectivity. It would be more useful to analyze the impact of Wifi and 4G/5G networks in one place. Some of this follows later in sections 3.2.1 and Yet, the split between these three sections is unclear. Having a discussion about today's Internet path characteristics and the dynamics at one place could be more useful. The text could also better distinguish between what matters to endpoints and considerations relevant only for middleboxes (such as passive monitors). Section 3.3 Path Requirements The statement "to find the bandwidth requirements for a router on the delivery path" may assume that the bottleneck on the path will be routers. Yet, on a path the bottlenecks could also be a link layer device (e.g., an Ethernet Switch, a Wifi access point, etc.). RFC 6077 (specifically Section 3.1.3) also explains that issue. A better wording may be "to find the bandwidth requirements for a delivery path". Section 3.6. Unpredictable Usage Profiles / Section 3.7. Extremely Unpredictable Usage Profiles I am not fully convinced by the distinction between "unpredictable" and "extremely unpredictable". To me, these two sections could be merged and maybe also be shortened. More specifically, Section 3.7 lists a lot of statistics from the past. What is IMHO a bit missing in Section 3.7 are actual operational considerations. For instance, are there any lessons learnt for the future? Note that Section 3.6 has a reasonable conclusion at the end. Section 4. Latency Considerations and Section 4.1. Ultra Low-Latency I am surprised by the definition "ultra low-latency (less than 1 second)", as well as some of the other numbers. For other real-time communication use cases, "ultra low-latency" would probably imply a latency requirement of the order of one millisecond. For instance, isochronous traffic for motion control in industrial networks may require a latency of 1 ms, and such "ultra-low latency" requirements are discussed elsewhere, e.g., in the DetNet WG. The terminology in this section should be better explained to readers dealing with networked systems with much harder latency requirements. And a reference should be added for these definitions in the context of video streaming. Section 5.5.2 Head-of-Line Blocking If Head-of-Line Blocking is indeed a relevant operational problem, it would be useful to add a corresponding reference (e.g., with measurements). Section 5.5.3. Wide and Rapid Variation in Path Capacity The statement "As many end devices have moved to wireless connectivity for the final hop (Wi-Fi, 5G, or LTE), new problems in bandwidth detction have emerged from radio interference and signal strength effects." could be moved to earlier parts of the document. Quite a bit of new computers apparently only have Wifi connectivity and no Ethernet port, i.e., Wifi may just be their default. Also, wireless may not only be used on the last hop, for instance in meshed setups. This may make the problem even harder. Section 6. Evolution of Transport Protocols and Transport Protocol Behaviors I really wonder whether UDP vs. TCP vs. QUIC is actually the relevant distinction. What may actually matter are the transport services provided by the protocol stack (e.g., RFC 8095). I fully agree to a related comment in the INTDIR review. Section 6.1. UDP and Its Behavior UDP is also used for encrypted tunnels (OpenVPN, Wireguard, etc.). Does encrypted tunneling really have no operational impact on streaming apps other than circuit breakers? (I am thinking about 4K streaming over OpenVPN and the like.) Section 6.2. TCP and Its Behavior This text should IMHO be presented and be discussed in TCPM. Personally, I am not convinced that this document is a good place for discussing the long history of TCP congestion control. If the objective of the document is to provide a tutorial, IMHO it would be more useful to briefly explain the state-of-the-art in year 2022. Some further specific comments on TCP congestion control: It is surprising that RFC 5681 is not referenced.793bis has normative statements regarding congestion control, and most stacks are compatible to what is written in 793bis. Several operating systems use CUBIC by default and support standard Reno. Another general good reference for TCP standards is RFC 7414.Does ECN, DCTCP, etc. really not matter at all, for instance, if a data center hosts video playout servers? Section 6.3. QUIC and Its Behavior As already noted, the discussion about head-of-line-blocking would really benefit from backing by a reference. Section 7. Streaming Encrypted Media As far as I know, streaming users sometimes use encrypted tunnels such as OpenVPN or WireGuard (or IPsec) to access video content. I may miss something, but it is unclear to me how that fits into the categories presented in Section

Revised text:
5.5.2.  <ins> In addition to smoothing over an appropriate time scale to handle network jitter (see RFC5481), ABR systems relying on measurements at the application layer also have to account for noise from the in- order data transmission at the transport layer. For instance, in the event of a lost packet on a TCP connection with SACK support (a common case for segmented delivery in practice), loss of a packet can provide a confusing bandwidth signal to the receiving </ins> application.  Because of the sliding window in TCP, many packets may be accepted by the receiver without being available to the application until the missing packet arrives.  Upon the arrival of get access to a lot of data at the same time. To a receiver measuring bytes received per unit time at the  <ins> application layer and interpreting it as an estimate of the available network bandwidth, this appears as a high jitter in the goodput measurement, presenting as a stall followed by a sudden leap that can far exceed the actual capacity of the transport path from the server when the hole in the received data is filled by a later </ins> retransmission.  5.5.3. As many end devices have moved to wireless connections for the final high-bandwidth traffic such as streaming media over the UDP ports that HTTP/3 is using.  <ins> As noted in noisy-measurements, because TCP provides a reliable, in- order delivery service for applications, any packet loss for a TCP </ins> connection causes head-of-line blocking, so that no TCP segments arriving after a packet is lost will be delivered to the receiving application until retransmission of the lost packet has been