Question:
This change allows clients to disable contexts for a given stream. This uses a new REGISTERDATAGRAMNO_CONTEXT capsule for negotiation. The important property here is that it requires capsule support even when contexts are not required. This is critical to the health of the ecosystem as it will ensure that intermediaries that support datagrams but not capsules cannot be deployed, as that would prevent future extensibility.
Editorial comments aside, I support the protocol feature of an optional DATAGRAM Context ID. Using a registration capsule, as proposed in this PR, is a very logical way of achieving that. I plan to implement this in my WIP code.
Looks like there's support for merging this. Some additional editorial work is needed, but we'll do that in followups.
This issue assumes that decide to go with the two-layer design described in . Given that design, some applications might not need the multiplexing provided by context IDs. We have multiple options here: Make context ID mandatory Applications that don't need it waste a byte per datagram Negotiate the presence of context IDs using an HTTP header Context IDs would still be mandatory to implement on servers because the client might send that header Have the method determine whether context IDs are present or not This would prevent extensibility on those methods Create a REGISTERNOCONTEXT message This assumes we use the Message design from We add a register message that means that this stream does not use context IDs This allows avoiding the byte overhead without sacrificing extensibility
Making the context ID mandatory is simple, although slightly wasteful. For the variations where it is optional, I prefer the one that uses a header. It could work with a variant of where we do Header + Message, and the Header determines whether or not you have contexts, and whether or not they can be dynamic.
What are the benefits of message over header to make it optional? To me it seems like header is sufficient, because the context IDs are end-to-end, and should not affect intermediaries, making the header a natural place to put it. I think having the context ID be the default behavior and the header is to opt-out is the right decision if we were to allow removing it.
This is a bit of a bike-shed, but why would it default to having context-IDs instead of not having them?
I think the second option "Negotiate the presence of context IDs using an HTTP header" is the best personally. In terms of: "Context IDs would still be mandatory to implement on servers because the client might send that header", I assume server in this context is the final application server, not a proxy? Even so, I'd argue it's applications which decide if they need this functionality, so if a server only implemented one application and it didn't use Context-IDs, then I don't see why it would need to implement it.
Ian, in your comment, are you talking about an intermediary when you say "proxy"? If so, then an intermediary need not implement anything based on the payload of than H3 DATAGRAM, it only has to pass the data along. That may be worth a clarification.
Having context ID optional has a nice future-proof property - other uses of datagrams in HTTP/3 can decide if they want the second layer or not. I wouldn't even state it as application needing multiplexing or not. Other applications might find it very useful to include a second ((third, fourth, fifth) layer of frame header. I think its fine for us to state as much in this in the spec, while also defining an explicit second layer for multiplexing.
Yes, by proxy I meant intermediary. And I agree about what intermediaries need to do, I just wanted to ensure I understood what server meant when David used it above. Yup, whenever all this gets written up, we can make it clear that intermediaries can ignore much of this.
The method defines the semantics of datagrams for a particular request, thus I think the third option makes most sense. I don't see a point in using explicit negotiation, since in the two cases I can think of the top of the head, the answer is always clear from the method (CONNECT-IP seems to always want context IDs, WebTransport does not ever want context IDs). If some method starts out without context, but wants to add context later, it can add its own negotiation.
One set of use cases for which the method approach(3) is best are those where there are multiple context IDs but they're predefined by the method and don't need negotiation. My strawman is WebRTC.
Looks good overall.

Its Capsule Data field consists of: A variable-length integer indicating the context ID of the <del> datagram (see datagram-contexts). </del> <ins> datagram (see datagram-contexts).  Whether or not this field is present depends on which registration capsules were exchanged on the associated stream: if a REGISTER_DATAGRAM_CONTEXT capsule (see register-capsule) has been sent or received on this stream, then the field is present; if a REGISTER_DATAGRAM_NO_CONTEXT capsule (see register-no-context-capsule) has been sent or received, then this field is absent; if neither has been sent or received, then it is not yet possible to parse this datagram and the receiver MUST either drop that datagram silently or buffer it temporarily while awaiting the registration capsule. </ins> The payload of the datagram, whose semantics are defined by individual applications.  Note that this field can be empty.


Question:
This tries to address both of:
Discussion from editor's conference call: Can we provide a reference for useful properties? Network analytics, application analytics, or both? Currently application analytics, which is OK. Is that clear enough? Who is supposed to notice changes from an expected baseline? Is mentioning QLOG helpful? Have we said clearly enough that baselines themselves can fluctuate, before the kinds of path characteristic changes we're talking about kick in?
I think just "baseline" is clear enough to me, in context with this section listing some relevant stats. I guess "baseline pattern of some relevant statistics" is maybe more precise, but I'm not sure "baseline pattern" really gets there.
from NAME 's review (URL) 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.
This comment helpfully points out the changing network architecture (a LOT of wireless), and we should be clearer about what we're talking about in Section 3.2 (our mental was mostly wired) and in 5.5.3 (where we focused on wireless). This material should go in Section 3.2, and should be referenced in 5.5.3..
Related to
From NAME (URL) 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.
Related to
This works for me. I'll merge it. 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

3.2.1. <del> There are many reasons why path characteristics might change suddenly, but we can divide these reasons into two categories: </del> <ins> There are many reasons why path characteristics might change in normal operation, for example: </ins> If the path topology changes.  For example, routing changes, which can happen in normal operation, may result in traffic being


Question:
This tries to address both of:
Discussion from editor's conference call: Can we provide a reference for useful properties? Network analytics, application analytics, or both? Currently application analytics, which is OK. Is that clear enough? Who is supposed to notice changes from an expected baseline? Is mentioning QLOG helpful? Have we said clearly enough that baselines themselves can fluctuate, before the kinds of path characteristic changes we're talking about kick in?
I think just "baseline" is clear enough to me, in context with this section listing some relevant stats. I guess "baseline pattern of some relevant statistics" is maybe more precise, but I'm not sure "baseline pattern" really gets there.
from NAME 's review (URL) 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.
This comment helpfully points out the changing network architecture (a LOT of wireless), and we should be clearer about what we're talking about in Section 3.2 (our mental was mostly wired) and in 5.5.3 (where we focused on wireless). This material should go in Section 3.2, and should be referenced in 5.5.3..
Related to
From NAME (URL) 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.
Related to
This works for me. I'll merge it. 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

If cross traffic that also traverses part or all of the same path topology increases or decreases, especially if this new cross <del> traffic is "inelastic," and does not, itself, respond to indications of path congestion. To recognize that a path carrying streaming media is not behaving the way it normally does, having an expected baseline that describes the way it normally does is fundamental.  Analytics that aid in that recognition can be more or less sophisticated and can be as simple as noticing that the apparent round trip times for media traffic carried over TCP transport on some paths are suddenly and significantly longer than usual.  Passive monitors can detect changes in the elapsed time between the acknowledgements for specific TCP segments from a TCP receiver since TCP octet sequence numbers and acknowledgements for those sequence numbers are carried in the clear, even if the TCP payload itself is encrypted.  See reliable-behavior for more information. As transport protocols evolve to encrypt their transport header fields, one side effect of increasing encryption is the kind of passive monitoring, or even "performance enhancement" (RFC3135) that was possible with the older transport protocols (UDP, described in unreliable-behavior and TCP, described in reliable-behavior) is no longer possible with newer transport protocols such as QUIC (described in quic-behavior).  The IETF has specified a "latency spin bit" mechanism in Section 17.4 of RFC9000 to allow passive latency monitoring from observation points on the network path throughout the duration of a connection, but currently chartered work in the IETF is focusing on endpoint monitoring and reporting, rather than on passive monitoring. One example is the "qlog" mechanism I-D.ietf-quic-qlog-main-schema, a protocol-agnostic mechanism used to provide better visibility for encrypted protocols such as QUIC (I-D.ietf-quic-qlog-quic-events) and for HTTP/3 (I-D.ietf-quic-qlog-h3-events). </del> <ins> traffic is "inelastic," and does not respond to indications of path congestion. Wireless links (Wi-Fi, 5G, LTE, etc.) often see rapid changes to capacity from changes in radio interference and signal strength as endpoints move. To recognize that a path carrying streaming media has experienced a change, maintaining a baseline that captures its prior properties is fundamental.  Analytics that aid in that recognition can be more or less sophisticated and can usefully operate on several different time scales, from milliseconds to hours or days. Useful properties to monitor for changes can include: round-trip times loss rate (and explicit congestion notification (ECN) (RFC3168 when in use) out of order packet rate packet and byte receive rate application level goodput properties of other connections carrying competing traffic, in addition to the connections carrying the streaming media externally provided measurements, for example from network cards or metrics collected by the operating system </ins> 3.3.


Question:
This tries to address both of:
Discussion from editor's conference call: Can we provide a reference for useful properties? Network analytics, application analytics, or both? Currently application analytics, which is OK. Is that clear enough? Who is supposed to notice changes from an expected baseline? Is mentioning QLOG helpful? Have we said clearly enough that baselines themselves can fluctuate, before the kinds of path characteristic changes we're talking about kick in?
I think just "baseline" is clear enough to me, in context with this section listing some relevant stats. I guess "baseline pattern of some relevant statistics" is maybe more precise, but I'm not sure "baseline pattern" really gets there.
from NAME 's review (URL) 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.
This comment helpfully points out the changing network architecture (a LOT of wireless), and we should be clearer about what we're talking about in Section 3.2 (our mental was mostly wired) and in 5.5.3 (where we focused on wireless). This material should go in Section 3.2, and should be referenced in 5.5.3..
Related to
From NAME (URL) 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.
Related to
This works for me. I'll merge it. 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

5.5.3. <del> As many end devices have moved to wireless connectivity for the final hop (Wi-Fi, 5G, or LTE), new problems in bandwidth detection have emerged from radio interference and signal strength effects. </del> <ins> As many end devices have moved to wireless connections for the final hop (such as Wi-Fi, 5G, LTE, etc.), new problems in bandwidth detection have emerged. </ins> <del> Each of these technologies can experience sudden changes in capacity as the end user device moves from place to place and encounters new sources of interference.  Microwave ovens, for example, can cause a throughput degradation of more than a factor of 2 while active Micro. 5G and LTE likewise can easily see rate variation by a factor of 2 or more over a span of seconds as users move around. </del> <ins> In most real-world operating environments, wireless links can often experience sudden changes in capacity as the end user device moves from place to place or encounters new sources of interference. Microwave ovens, for example, can cause a throughput degradation in Wi-Fi of more than a factor of 2 while active Micro.  5G and LTE likewise can easily see rate variation by a factor of 2 or more over a span of seconds as users move around. </ins> These swings in actual transport capacity can result in user <del> experience issues that can be exacerbated by insufficiently responsive ABR algorithms. </del> <ins> experience issues when interacting with ABR algorithms that aren't tuned to handle the capacity variation gracefully. </ins> 5.6.


Question:
This is based on the post-IETF 107 MOPS meeting where we talked about this. I chased some more references.
We should try to improve this reference - it's a long page, additions are at the top, so we have to either scroll for a long time or search for the date to get the relevant part of the page. Maybe reach out to ATT to get fragment tags that work?

Operational Considerations for Streaming Media <del> draft-ietf-mops-streaming-opcons-00 </del> <ins> draft-ietf-mops-streaming-opcons-01 </ins> Abstract


Question:
This is based on the post-IETF 107 MOPS meeting where we talked about this. I chased some more references.
We should try to improve this reference - it's a long page, additions are at the top, so we have to either scroll for a long time or search for the date to get the relevant part of the page. Maybe reach out to ATT to get fragment tags that work?

watch for increasing numbers of end users uploading significant amounts of content. <ins> 2.6. The causes of unpredictable usage described in sec-unpredictable were more or less the result of human choices, but we were reminded during a post-IETF 107 meeting that humans are not always in control, and forces of nature can cause enormous fluctuations in traffic patterns. In his talk, Sanjay Mishra Mishra reported that after the CoViD-19 pandemic broke out in early 2020, Comcast's streaming and web video consumption rose by 38%, with their reported peak traffic up 32% overall between March 1 to March 30 Comcast, AT&T reported a 28% jump in core network traffic (single day in April, as compared to pre stay-at-home daily average traffic), with video accounting for nearly half of all mobile network traffic, while social networking and web browsing remained the highest percentage (almost a quarter each) of overall mobility traffic ATT, and Verizon reported similar trends with video traffic up 36% over an average day (pre COVID-19) Verizon. We note that other operators saw similar spikes during this time period.  Craig Labowitz Labovitz reported Weekday peak traffic increases over 45%-50% from pre-lockdown levels, A 30% increase in upstream traffic over their pre-pandemic levels, and A steady increase in the overall volume of DDoS traffic, with amounts exceeding the pre-pandemic levels by 40%. (He attributed this increase to the significant rise in gaming-related DDoS attacks (LabovitzDDoS), as gaming usage also increased.) </ins> 3. 3.1.


Question:
Ali, this PR looks nice. Two meta-issues (which we can talk about whenever). I changed my references in the draft to a more complete format (using ins: can result in odd formatting in the References section). We don't have to do that for all the references in -04, but that would be a good thing to do in an editorial pass. I see that you're making some terminology changes (from "delivery" to "streaming", stuff like that). I agree with the changes you made in this PR, but we may want to make sure that they are made globally, for consistency.
I was not sure about the reference format, normally I just use the casual format: Authors, "Title," Journal/Conference, Date. I saw the missing author names and realized "ins" would do the trick. By no means, I am tied to that format - don't know even know what "ins" stands for here :)
From URL : "as in the author list, ins is an abbreviation for "initials/surname";" I tend to think an editorial pass on references would be better as a separate commit, where they need cleaning. Anything that somehow made it in and breaks the "make" needs fixing, but others maybe later?
I suggest a definition along these lines: Streaming is transmission of a continuous content from a server to a client and its simultaneous consumption by the client. ("simultaneous" is the key here) This has two implications: 1) Server transmission rate (loosely or tightly) matches to client consumption rate. That is, no buffer overrun or underrun is desirable/acceptable. 2) Client consumption rate is also limited by real-time constraints as opposed to just bandwidth availability. That is, client cannot fetch the content not available yet.
+1, this sgtm. But this probably implies a refactor of the intro section and the abstract, to better focus on the relevant use case.
I can take a crack at this.
Ali has text for this, will create a PR for the issue.

Abstract This document provides an overview of operational networking issues <del> that pertain to quality of experience in delivery of video and other </del> <ins> that pertain to quality of experience in streaming of video and other </ins> high-bitrate media over the internet. 1.


Question:
Ali, this PR looks nice. Two meta-issues (which we can talk about whenever). I changed my references in the draft to a more complete format (using ins: can result in odd formatting in the References section). We don't have to do that for all the references in -04, but that would be a good thing to do in an editorial pass. I see that you're making some terminology changes (from "delivery" to "streaming", stuff like that). I agree with the changes you made in this PR, but we may want to make sure that they are made globally, for consistency.
I was not sure about the reference format, normally I just use the casual format: Authors, "Title," Journal/Conference, Date. I saw the missing author names and realized "ins" would do the trick. By no means, I am tied to that format - don't know even know what "ins" stands for here :)
From URL : "as in the author list, ins is an abbreviation for "initials/surname";" I tend to think an editorial pass on references would be better as a separate commit, where they need cleaning. Anything that somehow made it in and breaks the "make" needs fixing, but others maybe later?
I suggest a definition along these lines: Streaming is transmission of a continuous content from a server to a client and its simultaneous consumption by the client. ("simultaneous" is the key here) This has two implications: 1) Server transmission rate (loosely or tightly) matches to client consumption rate. That is, no buffer overrun or underrun is desirable/acceptable. 2) Client consumption rate is also limited by real-time constraints as opposed to just bandwidth availability. That is, client cannot fetch the content not available yet.
+1, this sgtm. But this probably implies a refactor of the intro section and the abstract, to better focus on the relevant use case.
I can take a crack at this.
Ali has text for this, will create a PR for the issue.

compound annual growth rate continuing at 34% (from Appendix D of CVNI). <ins> A substantial part of this growth is due to increased use of streaming video, although the amount of video traffic in real-time communications (for example, online videoconferencing) has also grown significantly.  While both streaming video and videoconferencing have real-time delivery and latency requirements, these requirements vary from one application to another.  For example, videoconferencing demands an end-to-end (one-way) latency of a few hundreds of milliseconds whereas live streaming can tolerate latencies of several seconds. This document specifically focuses on the streaming applications and defines streaming as follows: Streaming is transmission of a continuous media from a server to a client and its simultaneous consumption by the client.  Here, continous media refers to media and associated streams such as video, audio, metadata, etc.  In this definition, the critical term is "simultaneous", as it is not considered streaming if one downloads a video file and plays it after the download is completed, which would be called download-and-play. This has two implications.  First, server's transmission rate must (loosely or tightly) match to client's consumption rate for an uninterrupted playback.  That is, the client must not run out of data (buffer underrun) or take more than it can keep (buffer overrun) as any excess media is simply discarded.  Second, client's consumption rate is limited by not only bandwidth availability but also the real- time constraints.  That is, the client cannot fetch media that is not available yet. </ins> In many contexts, video traffic can be handled transparently as generic application-level traffic.  However, as the volume of video traffic continues to grow, it's becoming increasingly important to


Question:
Ali, this PR looks nice. Two meta-issues (which we can talk about whenever). I changed my references in the draft to a more complete format (using ins: can result in odd formatting in the References section). We don't have to do that for all the references in -04, but that would be a good thing to do in an editorial pass. I see that you're making some terminology changes (from "delivery" to "streaming", stuff like that). I agree with the changes you made in this PR, but we may want to make sure that they are made globally, for consistency.
I was not sure about the reference format, normally I just use the casual format: Authors, "Title," Journal/Conference, Date. I saw the missing author names and realized "ins" would do the trick. By no means, I am tied to that format - don't know even know what "ins" stands for here :)
From URL : "as in the author list, ins is an abbreviation for "initials/surname";" I tend to think an editorial pass on references would be better as a separate commit, where they need cleaning. Anything that somehow made it in and breaks the "make" needs fixing, but others maybe later?
I suggest a definition along these lines: Streaming is transmission of a continuous content from a server to a client and its simultaneous consumption by the client. ("simultaneous" is the key here) This has two implications: 1) Server transmission rate (loosely or tightly) matches to client consumption rate. That is, no buffer overrun or underrun is desirable/acceptable. 2) Client consumption rate is also limited by real-time constraints as opposed to just bandwidth availability. That is, client cannot fetch the content not available yet.
+1, this sgtm. But this probably implies a refactor of the intro section and the abstract, to better focus on the relevant use case.
I can take a crack at this.
Ali has text for this, will create a PR for the issue.

further.  In this case, the immersive content is typically referred to as volumetric media.  One way to represent the volumetric media is to use point clouds, where streaming a single object may easily <del> require a bitrate of 30 Mbps or higher.  Refer to PCC for more details. </del> <ins> require a bitrate of 30 Mbps or higher.  Refer to MPEGI and PCC for more details. </ins> 2.2.


Question:
Ali, this PR looks nice. Two meta-issues (which we can talk about whenever). I changed my references in the draft to a more complete format (using ins: can result in odd formatting in the References section). We don't have to do that for all the references in -04, but that would be a good thing to do in an editorial pass. I see that you're making some terminology changes (from "delivery" to "streaming", stuff like that). I agree with the changes you made in this PR, but we may want to make sure that they are made globally, for consistency.
I was not sure about the reference format, normally I just use the casual format: Authors, "Title," Journal/Conference, Date. I saw the missing author names and realized "ins" would do the trick. By no means, I am tied to that format - don't know even know what "ins" stands for here :)
From URL : "as in the author list, ins is an abbreviation for "initials/surname";" I tend to think an editorial pass on references would be better as a separate commit, where they need cleaning. Anything that somehow made it in and breaks the "make" needs fixing, but others maybe later?
I suggest a definition along these lines: Streaming is transmission of a continuous content from a server to a client and its simultaneous consumption by the client. ("simultaneous" is the key here) This has two implications: 1) Server transmission rate (loosely or tightly) matches to client consumption rate. That is, no buffer overrun or underrun is desirable/acceptable. 2) Client consumption rate is also limited by real-time constraints as opposed to just bandwidth availability. That is, client cannot fetch the content not available yet.
+1, this sgtm. But this probably implies a refactor of the intro section and the abstract, to better focus on the relevant use case.
I can take a crack at this.
Ali has text for this, will create a PR for the issue.

3.1. Adaptive BitRate (ABR) is a sort of application-level response <del> strategy in which the receiving media player attempts to detect the available bandwidth of the network path by experiment or by observing the successful application-layer download speed, then chooses a video bitrate (among the limited number of available options) that fits within that bandwidth, typically adjusting as changes in available bandwidth occur in the network or changes in capabilities occur in the player (such as available memory, CPU, display size, etc.). </del> <ins> strategy in which the streaming client attempts to detect the available bandwidth of the network path by observing the successful application-layer download speed, then chooses a bitrate for each of the video, audio, subtitles and metadata (among the limited number of available options) that fits within that bandwidth, typically adjusting as changes in available bandwidth occur in the network or changes in capabilities occur during the playback (such as available memory, CPU, display size, etc.). </ins> The choice of bitrate occurs within the context of optimizing for <del> some metric monitored by the video player, such as highest achievable video quality, or lowest rate of expected rebuffering events. </del> <ins> some metric monitored by the client, such as highest achievable video quality or lowest chances for a rebuffering (playback stall). </ins> 3.2. <del> ABR playback is commonly implemented by video players using HLS </del> <ins> ABR playback is commonly implemented by streaming clients using HLS </ins> RFC8216 or DASH DASH to perform a reliable segmented delivery of <del> video data over HTTP.  Different player implementations and receiving devices use different strategies, often proprietary algorithms (called rate adaptation or bitrate selection algorithms), to perform available bandwidth estimation/prediction and the bitrate selection. Most players only use passive observations, i.e., they do not generate probe traffic to measure the available bandwidth. </del> <ins> media over HTTP.  Different implementations use different strategies ABRSurvey, often proprietary algorithms (called rate adaptation or bitrate selection algorithms) to perform available bandwidth estimation/prediction and the bitrate selection.  Most clients only use passive observations, i.e., they do not generate probe traffic to measure the available bandwidth. </ins> This kind of bandwidth-measurement systems can experience trouble in several ways that can be affected by networking design choices.


Question:
We've been getting help from people, but the ACKs section hasn't changed much since -03 was submitted in November 2020.
I found like two names that we had missed, and added them to the ACKs section. I think we can merge this any time.

9. <del> Thanks to Mark Nottingham, Glenn Deen, Dave Oran, Aaron Falk, Kyle Rose, Leslie Daigle, Lucas Pardue, Matt Stock, Alexandre Gouaillard, and Mike English for their very helpful reviews and comments. </del> <ins> Thanks to Alexandre Gouaillard, Aaron Falk, Dave Oran, Glenn Deen, Kyle Rose, Leslie Daigle, Lucas Pardue, Mark Nottingham, Matt Stock, Mike English, Roni Even, and Will Law for very helpful suggestions, reviews and comments. (If we missed your name, please let us know!) </ins> 10.  References


Question: Section 3.1 says to include the PPM task ID in a field on the problem details JSON object, but the task ID may be any 32 byte value, and may not be representable as a Unicode string. I propose that we base64-encode the task ID here to resolve this.

errors.  The "instance" value MUST be the endpoint to which the request was targeted.  The problem document MUST also include a "taskid" member which contains the associated PPM task ID, encoded <del> with base64 RFC4648 (this value is always known, see task- configuration). </del> <ins> with base64 using the standard alphabet RFC4648 (this value is always known, see task-configuration). </ins> In the remainder of this document, we use the tokens in the table above to refer to error types, rather than the full URNs.  For


Question:
Would be helpful to readers if this reference was fixed in the Editor's Copy: [I-D.draft-cfrg-patton-vdaf] " BROKEN REFERENCE ".
Maybe change each reference to ? That way it'll render like in the text, which reads a bit nicer.

This protocol is extensible and allows for the addition of new cryptographic schemes that implement the VDAF interface specified in <del> I-D.draft-cfrg-patton-vdaf.  Candidates include: </del> <ins> VDAF.  Candidates include: </ins> "prio3", which allows for aggregate statistics such as sum, mean, histograms, etc.  This class of VDAFs is based on Prio CGB17 and


Question:
Would be helpful to readers if this reference was fixed in the Editor's Copy: [I-D.draft-cfrg-patton-vdaf] " BROKEN REFERENCE ".
Maybe change each reference to ? That way it'll render like in the text, which reads a bit nicer.

client ever sees the values for individual clients. In order to address this problem, the aggregators engage in a secure, <del> multi-party computation specified by the chosen VDAF I-D.draft-cfrg- patton-vdaf in order to prepare a report for aggregation.  At the beginning of this computation, each aggregator is in possession of an input share uploaded by the client.  At the end of the computation, each aggregator is in posession of either an "output share" that is ready to be aggregated or an indication that a valid output share could not be computed. </del> <ins> multi-party computation specified by the chosen VDAF VDAF in order to prepare a report for aggregation.  At the beginning of this computation, each aggregator is in possession of an input share uploaded by the client.  At the end of the computation, each aggregator is in posession of either an "output share" that is ready to be aggregated or an indication that a valid output share could not be computed. </ins> To facilitiate this computation, the input shares generated by the client include information used by the aggregators during aggregation


Question:
Would be helpful to readers if this reference was fixed in the Editor's Copy: [I-D.draft-cfrg-patton-vdaf] " BROKEN REFERENCE ".
Maybe change each reference to ? That way it'll render like in the text, which reads a bit nicer.

"vdaf_verify_param": The aggregator's VDAF verification parameter output by the setup algorithm computed jointly by the aggregators <del> before the start of the PPM protocol I-D.draft-cfrg-patton-vdaf). [OPEN ISSUE: This is yet to be specified.  See issue#161.] </del> <ins> before the start of the PPM protocol VDAF).  [OPEN ISSUE: This is yet to be specified.  See issue#161.] </ins> Finally, the collector is configured with the HPKE secret key corresponding to "collector_hpke_config".


Question:
The only language I can find in the document that prescribes serialization is in Section 1.2 (please point out if I missed anything): We intend this statement to be prescriptive, i.e., encoding/decoding of messages is as defined in RFC8446. Another valid interpretation might be "we define the fields of each message following the conventions of RFC8446, but leaving encoding/decoding unspecified". I think it would help to refine this statement a bit. WDYTA:
SGTM FWIW (this was our interpretation, and I think it makes sense to clarify)
Yeah, this seems like a fine clarification. I don't think it was under specified before, but more information for those unfamiliar with TLS-style encoding doesn't hurt.

The following terms are used: <del> This document uses the presentation language of RFC8446. </del> <ins> This document uses the presentation language of RFC8446 to define messages in the PPM protocol.  Encoding and decoding of these messages as byte strings also follows RFC8446. </ins> 2.


Question:
This change is primarily editorial and includes the following changes: Replace references to "Prio" with references to a generic VDAF. (This section was written long ago when we had Prio in mind.) Elaborate on known issues for collect requests. Discuss Sybil attacks, including enumerating the different types ().
From dkg in Vienna:
It seems like there are a bunch of assumptions about Sybil attacks generally that are not well articulated yet. This would certainly help.
The list of attacks is really a list of things we ought to fix in the protocol, so I would not even bother including them in the security considerations. If we want to use this as an opportunity to identify open issues that need to be addressed before we consider the protocol secure, I would simply list the open issues and keep discussion in GitHub. (That is, the text here seems redundant with the issue )

6. <del> Prio assumes a powerful adversary with the ability to compromise an unbounded number of clients.  In doing so, the adversary can provide malicious (yet truthful) inputs to the aggregation function.  Prio also assumes that all but one server operates honestly, where a dishonest server does not execute the protocol faithfully as specified.  The system also assumes that servers communicate over secure and mutually authenticated channels.  In practice, this can be done by TLS or some other form of application-layer authentication. In the presence of this adversary, Prio provides two important properties for computing an aggregation function F: Privacy.  The aggregators and collector learn only the output of F computed over all client inputs, and nothing else. Robustness.  As long as the aggregators execute the input- validation protocol correctly, a malicious client can skew the output of F only by reporting false (untruthful) input.  The output cannot be influenced in any other way. There are several additional constraints that a Prio deployment must satisfy in order to achieve these goals: Minimum batch size.  The aggregation batch size has an obvious impact on privacy.  (A batch size of one hides nothing of the input.) Aggregation function choice.  Some aggregation functions leak slightly more than the function output itself. [TODO: discuss these in more detail.] </del> <ins> PPM assumes an active attacker that controls the network and has the ability to statically corrupt any number of clients, aggregators, and collectors.  That is, the attacker can learn the secret state of any party prior to the start of its attack.  For example, it may coerce a client into providing malicious input shares for aggregation or coerce an aggregator into diverting from the protocol specified (e.g., by divulging its input shares to the attacker). In the presence of this adversary, PPM aims to achieve the following high-level secure aggregation goals: Privacy.  Clients trust that some aggregator is honest.  That is, as long as at least one aggregator executes the protocol faithfully, the parties learn nothing beyond the aggregate result (i.e., the output of the aggregation function computed over the honest measurements). Correctness.  The collector trusts that the aggregators execute the protocol correctly.  That is, as long as the aggregators execute the protocol faithfully, a malicious client can skew the aggregate result only by reporting a false (untruthful) measurement.  The result cannot be influenced in any other way. Currently, the specification does not achieve these goals.  In particular, there are several open issues that need to be addressed before these goals are met.  Details for each issue are below. When crafted maliciously, collect requests may leak more information about the measurements than the system intends.  For example, the spec currently allows sequences of collect requests to reveal an aggregate result for a batch smaller than the minimum batch size.  [OPEN ISSUE: See issue#195.  This also has implications for how we solve issue#183.] Even benign collect requests may leak information beyond what one might expect intuitively.  For example, the Poplar1 VDAF VDAF can be used to compute the set of heavy hitters among a set of arbitrary bit strings uploaded by clients.  This requires multiple evaluations of the VDAF, the results of which reveal information to the aggregators and collector beyond what follows from the heavy hitters themselves.  Note that this leakage can be mitigated using differential privacy.  [OPEN ISSUE: We have yet not specified how to add DP.] The core PPM spec does not defend against Sybil attacks.  In this type of attack, the adversary adds to a batch a number of reports that skew the aggregate result in its favor.  For example: The result may reveal additional information about the honest measurements, leading to a privacy violation; or the result may have some property that is desirable to the adversary ("stats poisoning").  The upload sub-protocol includes an extensions mechanism that can be used to prevent -- or at least mitigate -- these types of attacks.  See upload-extensions.  [OPEN ISSUE: No such extension has been implemented, so we're not yet sure if the current mechanism is sufficient.] Attacks may also come from the network.  Thus, it is required that the aggregators and collector communicate with one another over mutually authenticated and confidential channels.  The core PPM spec does not specify such a mechanism beyond requiring server authentication for HTTPS sessions.  Note that clients are not required to authenticate themselves.  [OPEN ISSUE: It might be better to be prescriptive about leader authentication in leader- helper channels and collector authenticaiton in collector-leader channels.  For the latter we have issue#155.] </ins> 6.1. <ins> [OPEN ISSUE: This subsection is a bit out-of-date.] </ins> In this section, we enumerate the actors participating in the Prio system and enumerate their assets (secrets that are either inherently valuable or which confer some capability that enables further attack


Question:
Specifically: Specify use of the "one-shot" HPKE APIs defined in RFC9180, replacing equivalent use of the "multi-shot" APIs. (This change is intended to be a clarification, not a functional change.) Move the task_id parameter from the "info" parameter to the "aad" parameter. This is intended to protect against key-commitment-related attacks[1]. (This is a functional change.) Clarify that the URL field is an encapsulated key, rather than an encryption context. (This is a clarification, not a functional change.) [1] See discussion in URL
Per URL section 8.1: PPM effectively uses the single-shot APIs; we should consider refactoring PPM's usage of HPKE to use only one of applicationinfo & AAD. Following the advice of the HPKE RFC, we would want to use applicationinfo. This would permit slightly simpler implementations. I am not sure how much it would simplify the HPKE computations.
Should we then formally use the single shot HPKE APIs from ?
That would be my recommendation (though, somewhat confusingly given the advice quoted above, those APIs still allow specifying both application info & AAD). Implementations can & do already use the single-shot APIs, so I strongly suspect specifying the single-shot APIs will be feasible. & .]
+1 to using single-shot APIs, though I'm not sure I agree that using either info or add (but not both) is simpler. What's the reasoning?
The reasoning is: a) it follows the advice of the HPKE spec (quoted above) b) AFAICT, in a single-shot setting, application info & AAD serve exactly the same purpose, so using both is arbitrary/confusing/a potential source of mistakes. (in a multi-shot setting, application info is bound to all encryptions/decryptions done by the same context; AAD is bound to a single encryption/decryption)
TBH I'm not sure I agree with the quoted guidance here. There is a concept in protocol design known as "domain separation", the goal of which is to provide some kind of binding of long-term secret key operations to the context in which they were used. Imagine, for example, that someone used an HPKE secret key for PPM and some other protocol. Suppose further that both are using the empty string for . Then the security of our PPM deployment depends on how the other protocol uses the derived AEAD key, since in both protocols may end up deriving the same AEAD key. One way to avoid creating this attack surface is to pick an string that no other application is likely to choose so that derived keys are guaranteed to not collide (except with some negligible probability). That said, I'd go for changing how we use and . Something like this might be simpler: Let be a fixed string that identifies the protocol. It would be good if this string were relatively long, say the SHA-256 hash of "ppm-00". Let encode the task ID, sender/receiver roles, etc.
I definitely agree that application info should include information allowing domain separation, for the reasons you mention. (Indeed, the current PPM spec includes the strings "ppm input share" or "ppm aggregate share" in its application infos, presumably for this purpose.) But I think that's orthogonal to the question of whether we should also use AAD, since the application info could also include information other than the "domain separation parameter". Picking one example to make things concrete, a client upload request sets its application info to and its AAD to . But it could just as easily set the application info to (i.e. the concatenation of the two previous strings) and not use AAD. (Or alternatively, it could place that string in the AAD and not use application info.) My argument is that it's simpler to use only one of the parameters, since in the single-shot setting that PPM is using HPKE, the two parameters serve exactly the same purpose (i.e. including additionally-authenticated data).
As I see it, our goal is to make sure that decryption only succeeds if the sender and receiver agree on the "context", i.e., the task ID, sender/receiver role, nonce, extensions, and so on. This way a MITM cannot force the aggregator to interpret an input share incorrectly. Does this sound reasonable to you? From the point of view of this threat model, I don't think it's immediately clear that sticking the context in or is equivalent. For example, if we stuff the context in , then it influences the derivation of the AEAD key; but depending on the AEAD, it might be feasible for an attacker to find a find two keys that decrypt the same ciphertext. (See URL) More analysis will be needed to say for sure whether this weakness in the AEAD amounts to an attack against our protocol. In the meantime, sticking the "context" in seems like a more conservative choice to me.
Given your concern, your approach SGTM. It sounds like this may be a concern worth raising with the folks working on the HPKE spec, given the advice to "use the Setup info parameter for specifying auxiliary authenticated information", but I'm not going to chase this down currently.
So I think there's two tasks for this issue: Move task ID from to Use single-shot APIs for HPKE Anything else?
Yep, I think that's it.
(reopening for now as the fix PR has been reverted temporarily)
Perfect, great work!

To generate the report, the client begins by sharding its measurement into a sequence of input shares as specified by the VDAF in use.  To <del> encrypt an input share, the client first generates an HPKE RFC9180 context for the aggregator by running </del> <ins> encrypt an input share, the client generates an HPKE RFC9180 ciphertext and encapsulated key for the aggregator by running </ins> <del> where "pk" is the aggregator's public key and "server_role" is the Role of the intended recipient ("0x02" for the leader and "0x03" for the helper).  In general, the info string for computing the HPKE context is suffixed by two bytes, the first of which identifies the role of the sender and the second of which identifies the role of the intended recipient. "enc" is the HPKE encapsulated key and "context" is the HPKE context used by the client for encryption.  The payload is encrypted as where "input_share" is the aggregator's input share and "nonce" and "extensions" are the corresponding fields of "Report".  Clients MUST NOT use the same "enc" for multiple reports. </del> <ins> where "pk" is the aggregator's public key; "server_role" is the Role of the intended recipient ("0x02" for the leader and "0x03" for the helper); "task_id", "nonce", and "extensions" are the corresponding fields of "Report"; and "input_share" is the aggregator's input share. </ins> The leader responds to well-formed requests to "[leader]/upload" with HTTP status code 200 OK and an empty body.  Malformed requests are


Question:
Specifically: Specify use of the "one-shot" HPKE APIs defined in RFC9180, replacing equivalent use of the "multi-shot" APIs. (This change is intended to be a clarification, not a functional change.) Move the task_id parameter from the "info" parameter to the "aad" parameter. This is intended to protect against key-commitment-related attacks[1]. (This is a functional change.) Clarify that the URL field is an encapsulated key, rather than an encryption context. (This is a clarification, not a functional change.) [1] See discussion in URL
Per URL section 8.1: PPM effectively uses the single-shot APIs; we should consider refactoring PPM's usage of HPKE to use only one of applicationinfo & AAD. Following the advice of the HPKE RFC, we would want to use applicationinfo. This would permit slightly simpler implementations. I am not sure how much it would simplify the HPKE computations.
Should we then formally use the single shot HPKE APIs from ?
That would be my recommendation (though, somewhat confusingly given the advice quoted above, those APIs still allow specifying both application info & AAD). Implementations can & do already use the single-shot APIs, so I strongly suspect specifying the single-shot APIs will be feasible. & .]
+1 to using single-shot APIs, though I'm not sure I agree that using either info or add (but not both) is simpler. What's the reasoning?
The reasoning is: a) it follows the advice of the HPKE spec (quoted above) b) AFAICT, in a single-shot setting, application info & AAD serve exactly the same purpose, so using both is arbitrary/confusing/a potential source of mistakes. (in a multi-shot setting, application info is bound to all encryptions/decryptions done by the same context; AAD is bound to a single encryption/decryption)
TBH I'm not sure I agree with the quoted guidance here. There is a concept in protocol design known as "domain separation", the goal of which is to provide some kind of binding of long-term secret key operations to the context in which they were used. Imagine, for example, that someone used an HPKE secret key for PPM and some other protocol. Suppose further that both are using the empty string for . Then the security of our PPM deployment depends on how the other protocol uses the derived AEAD key, since in both protocols may end up deriving the same AEAD key. One way to avoid creating this attack surface is to pick an string that no other application is likely to choose so that derived keys are guaranteed to not collide (except with some negligible probability). That said, I'd go for changing how we use and . Something like this might be simpler: Let be a fixed string that identifies the protocol. It would be good if this string were relatively long, say the SHA-256 hash of "ppm-00". Let encode the task ID, sender/receiver roles, etc.
I definitely agree that application info should include information allowing domain separation, for the reasons you mention. (Indeed, the current PPM spec includes the strings "ppm input share" or "ppm aggregate share" in its application infos, presumably for this purpose.) But I think that's orthogonal to the question of whether we should also use AAD, since the application info could also include information other than the "domain separation parameter". Picking one example to make things concrete, a client upload request sets its application info to and its AAD to . But it could just as easily set the application info to (i.e. the concatenation of the two previous strings) and not use AAD. (Or alternatively, it could place that string in the AAD and not use application info.) My argument is that it's simpler to use only one of the parameters, since in the single-shot setting that PPM is using HPKE, the two parameters serve exactly the same purpose (i.e. including additionally-authenticated data).
As I see it, our goal is to make sure that decryption only succeeds if the sender and receiver agree on the "context", i.e., the task ID, sender/receiver role, nonce, extensions, and so on. This way a MITM cannot force the aggregator to interpret an input share incorrectly. Does this sound reasonable to you? From the point of view of this threat model, I don't think it's immediately clear that sticking the context in or is equivalent. For example, if we stuff the context in , then it influences the derivation of the AEAD key; but depending on the AEAD, it might be feasible for an attacker to find a find two keys that decrypt the same ciphertext. (See URL) More analysis will be needed to say for sure whether this weakness in the AEAD amounts to an attack against our protocol. In the meantime, sticking the "context" in seems like a more conservative choice to me.
Given your concern, your approach SGTM. It sounds like this may be a concern worth raising with the folks working on the HPKE spec, given the advice to "use the Setup info parameter for specifying auxiliary authenticated information", but I'm not going to chase this down currently.
So I think there's two tasks for this issue: Move task ID from to Use single-shot APIs for HPKE Anything else?
Yep, I think that's it.
(reopening for now as the fix PR has been reverted temporarily)
Perfect, great work!

report share as invalid with the error "hpke-unknown-config-id". Otherwise, it decrypts the payload with the following procedure: <del> where "sk" is the HPKE secret key, "task_id" is the task ID, "nonce" and "extensions" are the nonce and extensions of the report share respectively, and "server_role" is 0x02 if the aggregator is the leader and 0x03 otherwise.  If decryption fails, the aggregator marks the report share as invalid with the error "hpke-decrypt-error". Otherwise, it outputs the resulting "input_share". </del> <ins> where "sk" is the HPKE secret key, and "server_role" is the role of the aggregator ("0x02" for the leader and "0x03" for the helper).  If decryption fails, the aggregator marks the report share as invalid with the error "hpke-decrypt-error".  Otherwise, it outputs the resulting "input_share". </ins> 4.3.1.4.


Question:
Specifically: Specify use of the "one-shot" HPKE APIs defined in RFC9180, replacing equivalent use of the "multi-shot" APIs. (This change is intended to be a clarification, not a functional change.) Move the task_id parameter from the "info" parameter to the "aad" parameter. This is intended to protect against key-commitment-related attacks[1]. (This is a functional change.) Clarify that the URL field is an encapsulated key, rather than an encryption context. (This is a clarification, not a functional change.) [1] See discussion in URL
Per URL section 8.1: PPM effectively uses the single-shot APIs; we should consider refactoring PPM's usage of HPKE to use only one of applicationinfo & AAD. Following the advice of the HPKE RFC, we would want to use applicationinfo. This would permit slightly simpler implementations. I am not sure how much it would simplify the HPKE computations.
Should we then formally use the single shot HPKE APIs from ?
That would be my recommendation (though, somewhat confusingly given the advice quoted above, those APIs still allow specifying both application info & AAD). Implementations can & do already use the single-shot APIs, so I strongly suspect specifying the single-shot APIs will be feasible. & .]
+1 to using single-shot APIs, though I'm not sure I agree that using either info or add (but not both) is simpler. What's the reasoning?
The reasoning is: a) it follows the advice of the HPKE spec (quoted above) b) AFAICT, in a single-shot setting, application info & AAD serve exactly the same purpose, so using both is arbitrary/confusing/a potential source of mistakes. (in a multi-shot setting, application info is bound to all encryptions/decryptions done by the same context; AAD is bound to a single encryption/decryption)
TBH I'm not sure I agree with the quoted guidance here. There is a concept in protocol design known as "domain separation", the goal of which is to provide some kind of binding of long-term secret key operations to the context in which they were used. Imagine, for example, that someone used an HPKE secret key for PPM and some other protocol. Suppose further that both are using the empty string for . Then the security of our PPM deployment depends on how the other protocol uses the derived AEAD key, since in both protocols may end up deriving the same AEAD key. One way to avoid creating this attack surface is to pick an string that no other application is likely to choose so that derived keys are guaranteed to not collide (except with some negligible probability). That said, I'd go for changing how we use and . Something like this might be simpler: Let be a fixed string that identifies the protocol. It would be good if this string were relatively long, say the SHA-256 hash of "ppm-00". Let encode the task ID, sender/receiver roles, etc.
I definitely agree that application info should include information allowing domain separation, for the reasons you mention. (Indeed, the current PPM spec includes the strings "ppm input share" or "ppm aggregate share" in its application infos, presumably for this purpose.) But I think that's orthogonal to the question of whether we should also use AAD, since the application info could also include information other than the "domain separation parameter". Picking one example to make things concrete, a client upload request sets its application info to and its AAD to . But it could just as easily set the application info to (i.e. the concatenation of the two previous strings) and not use AAD. (Or alternatively, it could place that string in the AAD and not use application info.) My argument is that it's simpler to use only one of the parameters, since in the single-shot setting that PPM is using HPKE, the two parameters serve exactly the same purpose (i.e. including additionally-authenticated data).
As I see it, our goal is to make sure that decryption only succeeds if the sender and receiver agree on the "context", i.e., the task ID, sender/receiver role, nonce, extensions, and so on. This way a MITM cannot force the aggregator to interpret an input share incorrectly. Does this sound reasonable to you? From the point of view of this threat model, I don't think it's immediately clear that sticking the context in or is equivalent. For example, if we stuff the context in , then it influences the derivation of the AEAD key; but depending on the AEAD, it might be feasible for an attacker to find a find two keys that decrypt the same ciphertext. (See URL) More analysis will be needed to say for sure whether this weakness in the AEAD amounts to an attack against our protocol. In the meantime, sticking the "context" in seems like a more conservative choice to me.
Given your concern, your approach SGTM. It sounds like this may be a concern worth raising with the folks working on the HPKE spec, given the advice to "use the Setup info parameter for specifying auxiliary authenticated information", but I'm not going to chase this down currently.
So I think there's two tasks for this issue: Move task ID from to Use single-shot APIs for HPKE Anything else?
Yep, I think that's it.
(reopening for now as the fix PR has been reverted temporarily)
Perfect, great work!

"AggregateShareReq" is done as follows: where "pk" is the HPKE public key encoded by the collector's HPKE <del> key, and server_role is is "0x02" for the leader and "0x03" for a helper. </del> <ins> key, "server_role" is "0x02" for the leader and "0x03" for a helper. </ins> The collector decrypts these aggregate shares using the opposite process.  Specifically, given an encrypted input share, denoted "enc_share", for a given batch interval, denoted "batch_interval", decryption works as follows: <del> where "sk" is the HPKE secret key, "task_id" is the task ID for a given collect request, and "server_role" is the role of the server that sent the aggregate share ("0x02" for the leader and "0x03" for the helper). </del> <ins> where "sk" is the HPKE secret key, "task_id" is the task ID for the collect request, and "server_role" is the role of the server that sent the aggregate share ("0x02" for the leader and "0x03" for the helper). </ins> 4.4.5.


Question:
AggregateShareReq should have an aggregation parameter field instead of aggregation job ID.
Thanks NAME for spotting this!

"batch_interval" is the batch interval of the request. <ins> "agg_param", an aggregation parameter for the VDAF being executed. </ins> "report_count" is the number of reports included in the aggregation. "checksum" is the checksum computed over the set of client reports. <del> "job_id" is the ID of the aggregation job. </del> This request MUST be authenticated as described in https-sender-auth. To handle the leader's request, the helper first ensures that the request meets the requirements for batch parameters following the


Question:
Align with the latest VDAF draft. This requires two minor protocol changes: 02 added a value called the "publicshare" to the VDAF syntax. This is to support Poplar1 (and likely other schemes that need a similar extractability property). To accommodate this, the publicshare field has been added to the Report and ReportShare structures and appended to the AAD for input share encryption. (This may end up being useful for privacy.) 02 added the report count to the aggregate result computation. This is already passed to the Collector in the CollectResp, so we just need to update the aggregate result computation in the draft to match.
Rebased and pushed a minor update: The public share in Report and ReportStore now both have the same length prefix.
Rebased.
This will be VDAF-02 or 03, depending on when we need to cut DAP-02.
VDAF-03 is out, I will send a PR by end of week. URL

where "[leader]" is the first entry in the task's aggregator endpoints.  The payload is structured as follows: <del> This message is called the Client's report.  It consists of a "header" and the encrypted input share of each of the Aggregators. The header consists of the task ID and report "metadata".  The metadata consists of the following fields: </del> <ins> This message is called the Client's report.  It consists of the report metaata, the "public share" output by the VDAF's input- distribution algorithm, and the encrypted input share of each of the Aggregators.  The header consists of the task ID and report "metadata".  The metadata consists of the following fields: </ins> A timestamp representing the time at which the report was generated.  Specifically, the "time" field is set to the number of


Question:
Align with the latest VDAF draft. This requires two minor protocol changes: 02 added a value called the "publicshare" to the VDAF syntax. This is to support Poplar1 (and likely other schemes that need a similar extractability property). To accommodate this, the publicshare field has been added to the Report and ReportShare structures and appended to the AAD for input share encryption. (This may end up being useful for privacy.) 02 added the report count to the aggregate result computation. This is already passed to the Collector in the CollectResp, so we just need to update the aggregate result computation in the draft to match.
Rebased and pushed a minor update: The public share in Report and ReportStore now both have the same length prefix.
Rebased.
This will be VDAF-02 or 03, depending on when we need to cut DAP-02.
VDAF-03 is out, I will send a PR by end of week. URL

4.4.1.3. Each report share has a corresponding task ID, report metadata <del> (timestamp, nonce, and extensions), and encrypted input share.  Let "task_id", "metadata", and "encrypted_input_share" denote these values, respectively.  Given these values, an aggregator decrypts the input share as follows.  First, the aggregator looks up the HPKE config and corresponding secret key indicated by </del> <ins> (timestamp, nonce, and extensions), the public share sent to each Aggregator, and the recipient's encrypted input share.  Let "task_id", "metadata", "public_share", and "encrypted_input_share" denote these values, respectively.  Given these values, an aggregator decrypts the input share as follows.  First, the aggregator looks up the HPKE config and corresponding secret key indicated by </ins> "encrypted_input_share.config_id".  If not found, then it marks the report share as invalid with the error "hpke-unknown-config-id". Otherwise, it decrypts the payload with the following procedure:


Question:
Align with the latest VDAF draft. This requires two minor protocol changes: 02 added a value called the "publicshare" to the VDAF syntax. This is to support Poplar1 (and likely other schemes that need a similar extractability property). To accommodate this, the publicshare field has been added to the Report and ReportShare structures and appended to the AAD for input share encryption. (This may end up being useful for privacy.) 02 added the report count to the aggregate result computation. This is already passed to the Collector in the CollectResp, so we just need to update the aggregate result computation in the draft to match.
Rebased and pushed a minor update: The public share in Report and ReportStore now both have the same length prefix.
Rebased.
This will be VDAF-02 or 03, depending on when we need to cut DAP-02.
VDAF-03 is out, I will send a PR by end of week. URL

"vdaf_verify_key" is the VDAF verification key shared by the aggregators; "agg_id" is the aggregator ID ("0x00" for the Leader and <del> "0x01" for the helper); and "agg_param" is the opaque aggregation parameter distributed to the aggregtors by the collector. </del> <ins> "0x01" for the helper); "agg_param" is the opaque aggregation parameter distributed to the aggregtors by the collector; "public_share" is the public share generated by the client and distributed to each aggregator; and "input_share" is the aggregator's input share. </ins> If either step fails, the aggregator marks the report as invalid with error "vdaf-prep-error".  Otherwise, the value "out" is interpreted


Question:
Align with the latest VDAF draft. This requires two minor protocol changes: 02 added a value called the "publicshare" to the VDAF syntax. This is to support Poplar1 (and likely other schemes that need a similar extractability property). To accommodate this, the publicshare field has been added to the Report and ReportShare structures and appended to the AAD for input share encryption. (This may end up being useful for privacy.) 02 added the report count to the aggregate result computation. This is already passed to the Collector in the CollectResp, so we just need to update the aggregate result computation in the draft to match.
Rebased and pushed a minor update: The public share in Report and ReportStore now both have the same length prefix.
Rebased.
This will be VDAF-02 or 03, depending on when we need to cut DAP-02.
VDAF-03 is out, I will send a PR by end of week. URL

aggregate shares into an aggregate result using the VDAF's "agg_shares_to_result" algorithm.  In particular, let "agg_shares" denote the ordered sequence of aggregator shares, ordered by <del> aggregator index, and let "agg_param" be the opaque aggregation parameter.  The final aggregate result is computed as follows: </del> <ins> aggregator index, let "report_count" denote the report count sent by the Leader, and let "agg_param" be the opaque aggregation parameter. The final aggregate result is computed as follows: </ins> 4.5.4.


Question:
And while we're at it, update an obsolete reference to Oblivious HTTP, which has long since been adopted by its WG. VDAF-05 isn't expected until Monday 3/13, but we can get this change ready to go.
DAP-04 will need some routine changes: [x] Update version tags for HPKE to () [x] Bump VDAF-03 to 05 () [x] Update change log () [x] Run spell check ()

participating clients or permit some attacks on robustness.  This auxiliary information could be removed by having clients submit reports to an anonymizing proxy server which would then use Oblivious <del> HTTP I-D.thomson-http-oblivious to forward inputs to the DAP leader, without requiring any server participating in DAP to be aware of whatever client authentication or attestation scheme is in use. </del> <ins> HTTP I-D.draft-ietf-ohai-ohttp-07 to forward inputs to the DAP leader, without requiring any server participating in DAP to be aware of whatever client authentication or attestation scheme is in use. </ins> 7.4.


Question:
Partially addresses . (We also need to decide if/how to allow a particular PA protocol to use multiple helpers for privacy.) Right now, multiple helpers are used for redundancy in case one of the helpers dropped out. This removes this mechanism from the core protocol.
Squashed and rebased.
That's a good idea,
Hmm... Do we need to decide. Per our discussion on Wednesday, if the clients learn this out of band then we won't need this field at all because clients will get it separately and helpers don't need to know what the other helpers are, right?
My takeaway is that the parameters must contain all input needed to run the protocol, including the helpers. And all of this is configured out of band. You could punt everything except for the cryptographic goop out of the parameters structure, forcing clients to know the leader and helper and get their public keys out of band as well, which would be fine. It's just different. I think both work just fine, but I lean towards having params specify everything. It just seems simpler to me. That said... making public key discovery part of the client upload protocol here is more complex, I think. So, .
Squashed.
LGTM. Multiple helpers "for privacy" seems like it's inherently a per-protocol thing (Prio supports it, Hits does not). If and when we decide to address that, should we move this to the protocol-specific slot in PAParam?

*Collect:* The collector makes one or more requests to the leader in order to obtain the final output of the protocol.  Before the output can be computed, the aggregators (i.e, the leader and <del> helpers) need to have verified and aggregated a sufficient number </del> <ins> helper) need to have verified and aggregated a sufficient number </ins> of inputs.  Depending on the PA protocol, it may be possible for the aggregators to do so immediately when reports are uploaded. (See prio.)  However, in general it is necessary for them to wait until (a) enough reports have been uploaded and (b) the collector has made a request.  (See hits.) <del> [TODO: Say that the protocol involves multiple helpers.  The downside of using secret sharing is that the protocol requires at least two servers to be online during the entire data aggregation process.  To ameliorate this problem, we run the protocol in parallel with multiple helpers.] </del> 3. This section specifies a protocol for executing generic PA tasks.


Question:
Partially addresses . (We also need to decide if/how to allow a particular PA protocol to use multiple helpers for privacy.) Right now, multiple helpers are used for redundancy in case one of the helpers dropped out. This removes this mechanism from the core protocol.
Squashed and rebased.
That's a good idea,
Hmm... Do we need to decide. Per our discussion on Wednesday, if the clients learn this out of band then we won't need this field at all because clients will get it separately and helpers don't need to know what the other helpers are, right?
My takeaway is that the parameters must contain all input needed to run the protocol, including the helpers. And all of this is configured out of band. You could punt everything except for the cryptographic goop out of the parameters structure, forcing clients to know the leader and helper and get their public keys out of band as well, which would be fine. It's just different. I think both work just fine, but I lean towards having params specify everything. It just seems simpler to me. That said... making public key discovery part of the client upload protocol here is more complex, I think. So, .
Squashed.
LGTM. Multiple helpers "for privacy" seems like it's inherently a per-protocol thing (Prio supports it, Hits does not). If and when we decide to address that, should we move this to the protocol-specific slot in PAParam?

"task": The PA task. <del> "helper_urls": The helpers' endpoint URLs. </del> <ins> "helper_url": The helpers endpoint URL. </ins> "collector_config": The HPKE configuration of the collector (described in hpke-config).  [OPEN ISSUE: Maybe the collector's


Question:
Partially addresses . (We also need to decide if/how to allow a particular PA protocol to use multiple helpers for privacy.) Right now, multiple helpers are used for redundancy in case one of the helpers dropped out. This removes this mechanism from the core protocol.
Squashed and rebased.
That's a good idea,
Hmm... Do we need to decide. Per our discussion on Wednesday, if the clients learn this out of band then we won't need this field at all because clients will get it separately and helpers don't need to know what the other helpers are, right?
My takeaway is that the parameters must contain all input needed to run the protocol, including the helpers. And all of this is configured out of band. You could punt everything except for the cryptographic goop out of the parameters structure, forcing clients to know the leader and helper and get their public keys out of band as well, which would be fine. It's just different. I think both work just fine, but I lean towards having params specify everything. It just seems simpler to me. That said... making public key discovery part of the client upload protocol here is more complex, I think. So, .
Squashed.
LGTM. Multiple helpers "for privacy" seems like it's inherently a per-protocol thing (Prio supports it, Hits does not). If and when we decide to address that, should we move this to the protocol-specific slot in PAParam?

3.3.2. <del> For each URL "[helper]" in "PAUploadStartResp.helper_urls", the client sends a GET request to "[helper]/key_config".  The helper responds with status 200 and an "HpkeConfig" message.  Next, the client collects the set of helpers it will upload shares to.  It ignores a helper if: </del> <ins> Let "[helper]" denote the helper URL encoded by "PAUploadStartResp.param.helper_url".  When a client sends a GET request to "[helper]/key_config", the helper responds with status 200 and an "HpkeConfig" message.  The client aborts if any of the following happen: </ins> the client and helper failed to establish a secure, helper- authenticated channel;


Question:
Partially addresses . (We also need to decide if/how to allow a particular PA protocol to use multiple helpers for privacy.) Right now, multiple helpers are used for redundancy in case one of the helpers dropped out. This removes this mechanism from the core protocol.
Squashed and rebased.
That's a good idea,
Hmm... Do we need to decide. Per our discussion on Wednesday, if the clients learn this out of band then we won't need this field at all because clients will get it separately and helpers don't need to know what the other helpers are, right?
My takeaway is that the parameters must contain all input needed to run the protocol, including the helpers. And all of this is configured out of band. You could punt everything except for the cryptographic goop out of the parameters structure, forcing clients to know the leader and helper and get their public keys out of band as well, which would be fine. It's just different. I think both work just fine, but I lean towards having params specify everything. It just seems simpler to me. That said... making public key discovery part of the client upload protocol here is more complex, I think. So, .
Squashed.
LGTM. Multiple helpers "for privacy" seems like it's inherently a per-protocol thing (Prio supports it, Hits does not). If and when we decide to address that, should we move this to the protocol-specific slot in PAParam?

the key config specifies a KEM, KDF, or AEAD algorithm the client doesn't recognize. <del> If the set of supported helpers is empty, then the client aborts and alerts the leader with "no supported helpers".  Otherwise, for each supported helper the client issues a POST request to "[leader]/upload_finish" with a payload constructed as described below. </del> [OPEN ISSUE: Should the request URL encode the PA task?  This would be necessary if we make "upload_start" an idempotent GET per issue#48.]