Question: but also will offer "a=rtcp-mux", thus allowing for compatibility with either multiplexing or non-multiplexing endpoints. <del> The JSEP implementation will only gather RTP candidates.  This halves the number of candidates that the offerer needs to gather. Applying a description with an m= section that does not contain an "a=rtcp-mux" attribute will cause an error to be returned. </del> <ins> The JSEP implementation will only gather RTP candidates and will insert an "a=rtcp-mux-only" indication into any offers it generates.  This halves the number of candidates that the offerer needs to gather.  Applying a description with an m= section that does not contain an "a=rtcp-mux" attribute will cause an error to be returned. </ins> The default multiplexing policy MUST be set to "require". Implementations MAY choose to reject attempts by the application to

I applied the following principles, which unfortunately are somewhat inconsistent with the current mux-exclusive draft. You use a=rtcp-mux to negotiate mux a=rtcp-mux-only is used by the offerer to say "I really mean it" To that end, offers contain a=rtcp-mux-only if the policy is require. Answers don't contain it because you don't need it to negotiate mux.
Those principles are consistent with bundle-only, although I wrote the mux-only examples as if mux-only should always be included whenever a=rtcp-mux is present.
NAME back at you
NAME ping
LG


Question: identifies the packet as a Version Negotiation packet.  QUIC version 1 uses version 0x00000001.  Operators should expect to observe packets with other version numbers as a result of various <del> Internet experiments, future standards, and greasing.  All deployed versions are maintained in an IANA registry (see </del> <ins> Internet experiments, future standards, and greasing (RFC7801). All deployed versions are maintained in an IANA registry (see </ins> Section 22.2 of QUIC-TRANSPORT). source and destination connection ID: short and long packet

This goes to the end of "Server Name Indication" Some changes are attempts at clarifications, but I could have incorrectly clarified something, so feel free to reject changes.


Question: "Connection Priority" on one Connection does not change it on the other Connections in the same Connection Group. <del> Message Properties are also not entangled.  For example, changing "Lifetime" (see msg-lifetime) of a Message will only affect a single Message on a single Connection. </del> <ins> Message Properties set on a Connection also apply only to that Connection. </ins> A new Connection created by Clone can have a Message Framer assigned via the optional "framer" parameter of the Clone Action.  If this

Applied requested changes
In 3.2 it is clear that you can set Message Properties on Connections and Preconnections, but the rest of the document is not very clear about this. In 4.2 for instance the text only talks about setting Selection Properties and Connection Properties on PreConnections. In 7.1.3 the text says "Connection Properties describe the default behavior for all Messages on a Connection". But I think that Message Properties set on a Connection or PreConnection may also describe the default behavior so this relationship should be clarified. Also, the Message Property defaults given for Properties that do not depend on other Properties sets the defaults for all Messages in a Connection I assume? I also assume that a Message Property set on a Message Context overrides a Message Property set on a PreConnection or Connection, but this is also not specified. In 5.4. Connection Groups, I assume that setting Message Properties on a Connection applies to all Connections in the Connection Group? This is not clear.
In general, I agree. If you set a message property dynamically on a connection, though, I don't think it would affect the group—I could have one stream in a multiplexed protocol have separate behavior, such as priority.
+1 to tfpauly. But this is already written in 5.4, maybe you missed it (Anna)? "Message Properties are also not entangled. For example, changing Lifetime (see Section 7.1.3.1) of a Message will only affect a single Message on a single Connection."
I am fine with an update of a Message Property to a connection applying only to that connection. Priority is not a good example as it is already a special case, but my main issue was that we need to be more clear about the behavior in the specification. NAME the issue is about setting a Message Property on a Connection. The text you refer to talk about setting Message Properties on a message. You could interpret the first part as a general statement, but the way the text is structured it is easy to think it refers to setting Message Properties on a message so I think we should be explicit also about setting Message Properties on a Connection. Could be a second example.
So after looking at the text again, I think the example talking about setting Message Properties on a message should be removed, see comment in the pull request.
Thanks!!


Question: application. Framer: A Framer is a data translation layer that can be added to <del> a Connection to define how application-layer Messages are transmitted over a Protocol Stack.  This is particularly relevant when using a protocol that otherwise presents unstructured streams, such as TCP. </del> <ins> a Connection.  Framers allow extending a Connection's protocol stack to define how to encapsulate or encode outbound Messages, and how to decapsulate or decode inbound data into Messages.  In this way, message boundaries can be preserved when using a Connection object, even with a protocol that otherwise presents unstructured streams, such as TCP.  This is designed based on the fact that many of the current application protocols evolved over TCP, which does not provide message boundary preservation, and since many of these protocols require message boundaries to function, each application layer protocol has defined its own framing.  For example, when an HTTP application sends and receives HTTP messages over a byte-stream transport, it must parse the boundaries of HTTP messages from the stream of bytes. </ins> 4.1.6.

(sorry that it took me so long) This uses some text that is also in the interface draft.
This I would like us to discuss. Well I know we have discussed this before. As I am reading the Interface document after a long while and having read it along with the architecture document, I kind of strongly feel that the architecture document lack of description on framer and the states. These are mentioned and pictured in the architecture document but haven't been clear there. Hence, I kind of feel that we need to move this two sections to architecture document. Please discuss so that I can understand the reasoning of not doing so.
Regarding section 9.1.2, the framer description: I think it's good to be clear and complete in the API document, but I agree that framers should be appropriately described in the architecture document. I have the impression that the text in the -arch document on this matter may indeed be too short, while the text in the -interface document may be too long (although a little bit of repetition surely won't hurt here, it's good for this document to be understandable and clear on its own). Maybe some of it could be moved to -arch, or repeated there? Regarding section 11, Connection states: to me, this is fine as it is. Section 4.1 / figure 4 in the architecture document make state transitions pretty clear - and section 11 / figure 2 are just a more fine-grain look at these states as a way to explain exactly which Events will occur when. This seems needed in the API document, where all other Events and Actions are defined, and I would think that this is at a too detailed / fine-grain level for the architecture document.
NAME after giving it some more thoughts, I think your suggested approach should be fine , meaning we put more description of framer in the architecture document and leave the connection states to the Interface document.
NAME did you want to take a stab at this?
Looks great!


Question: While QUIC is designed as a general-purpose transport protocol, and can carry different application-layer protocols, the current <del> standardized mapping is for HTTP/3 I-D.ietf-quic-http, which describes how QUIC transport features are used for HTTP.  The convention is for HTTP/3 to run over UDP port 443 Port443 but this is not a strict requirement. </del> <ins> standardized mapping is for HTTP/3 RFC9114, which describes how QUIC transport services are used for HTTP.  The convention is for HTTP/3 to run over UDP port 443 Port443 but this is not a strict requirement. </ins> When HTTP/3 is encapsulated in QUIC, which is then encapsulated in UDP, streaming operators (and network operators) might see UDP <del> traffic patterns that are similar to HTTP(S) over TCP.  Since earlier versions of HTTP(S) rely on TCP, UDP ports may be blocked for any port numbers that are not commonly used, such as UDP 53 for DNS. Even when UDP ports are not blocked and HTTP/3 can flow, streaming operators (and network operators) may severely rate-limit this traffic because they do not expect to see legitimate high-bandwidth traffic such as streaming media over the UDP ports that HTTP/3 is using. </del> <ins> traffic patterns that are similar to HTTP(S) over TCP.  UDP ports may be blocked for any port numbers that are not commonly used, such as UDP 53 for DNS.  Even when UDP ports are not blocked and QUIC packets can flow, streaming operators (and network operators) may severely rate-limit this traffic because they do not expect to see legitimate high-bandwidth traffic such as streaming media over the UDP ports that HTTP/3 is using. </ins> As noted in hol-blocking, because TCP provides a reliable, in-order delivery service for applications, any packet loss for a TCP

…transport protocols Also tighten up description of the assumed protocol stack in (what is now) Section 6.1. Note that the UDP and TCP sections have been flipped in order, to improve section flow and clarity.
NAME - If you have a moment before our call on Friday to take a look at this PR, I'd appreciate it. I think it's important to to get that right, in order to address both the INTDIR and TSVART reviews.
From NAME As a bit of transport commentary, I don't find the setup of Section 6 compelling. While UDP is a transport in a technical sense, for the purpose of media, it is almost always a layer upon which the protocol doing the congestion control work runs. To that end, QUIC is just another more standardized case of this, just like SCTP over UDP. Rather than talking about "UDP's behavior" vs "TCP's behavior", I suggest talking about how applications over UDP for media behave vs applications over TCP for media behave.
NAME - I need to look at this more closely. I understand your point about UDP, but (because UDP is not a real transport layer, while TCP and QUIC are), I suspect (without looking closely yet) I'm going to be making sure that I explain this more clearly, without trying to explain the behavior(s) of applications running over TCP and QUIC. As an aside, I'm trying to make this clearer in drafts I'm targeting forwards Media Over QUIC/MOQ, so I actually DO believe you ... Please await results.
A +1 from NAME 's review (URL): 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.
After looking at this issue. I agree with both NAME and NAME that the emphasis should be on the interaction between (see new section describing this) media transport protocols and the transport services provided by underlying transport protocols. I'm moving the formerly-UDP section below the formerly-TCP section, because it's easier to describe what UDP is NOT doing after describing what TCP IS doing.


Question: While QUIC is designed as a general-purpose transport protocol, and can carry different application-layer protocols, the current <del> standardized mapping is for HTTP/3 I-D.ietf-quic-http, which describes how QUIC transport features are used for HTTP.  The convention is for HTTP/3 to run over UDP port 443 Port443 but this is not a strict requirement. </del> <ins> standardized mapping is for HTTP/3 RFC9114, which describes how QUIC transport services are used for HTTP.  The convention is for HTTP/3 to run over UDP port 443 Port443 but this is not a strict requirement. </ins> When HTTP/3 is encapsulated in QUIC, which is then encapsulated in UDP, streaming operators (and network operators) might see UDP <del> traffic patterns that are similar to HTTP(S) over TCP.  Since earlier versions of HTTP(S) rely on TCP, UDP ports may be blocked for any port numbers that are not commonly used, such as UDP 53 for DNS. Even when UDP ports are not blocked and HTTP/3 can flow, streaming operators (and network operators) may severely rate-limit this traffic because they do not expect to see legitimate high-bandwidth traffic such as streaming media over the UDP ports that HTTP/3 is using. </del> <ins> traffic patterns that are similar to HTTP(S) over TCP.  UDP ports may be blocked for any port numbers that are not commonly used, such as UDP 53 for DNS.  Even when UDP ports are not blocked and QUIC packets can flow, streaming operators (and network operators) may severely rate-limit this traffic because they do not expect to see legitimate high-bandwidth traffic such as streaming media over the UDP ports that HTTP/3 is using. </ins> As noted in hol-blocking, because TCP provides a reliable, in-order delivery service for applications, any packet loss for a TCP

I'm also moving one paragraph into the formerly-TCP section from Section 6, because it is specific to ABR strategies that assume a reliable transport. More news to come, when I create a PR for this issue ...
We think is now mergrable. 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.


Question: While QUIC is designed as a general-purpose transport protocol, and can carry different application-layer protocols, the current <del> standardized mapping is for HTTP/3 I-D.ietf-quic-http, which describes how QUIC transport features are used for HTTP.  The convention is for HTTP/3 to run over UDP port 443 Port443 but this is not a strict requirement. </del> <ins> standardized mapping is for HTTP/3 RFC9114, which describes how QUIC transport services are used for HTTP.  The convention is for HTTP/3 to run over UDP port 443 Port443 but this is not a strict requirement. </ins> When HTTP/3 is encapsulated in QUIC, which is then encapsulated in UDP, streaming operators (and network operators) might see UDP <del> traffic patterns that are similar to HTTP(S) over TCP.  Since earlier versions of HTTP(S) rely on TCP, UDP ports may be blocked for any port numbers that are not commonly used, such as UDP 53 for DNS. Even when UDP ports are not blocked and HTTP/3 can flow, streaming operators (and network operators) may severely rate-limit this traffic because they do not expect to see legitimate high-bandwidth traffic such as streaming media over the UDP ports that HTTP/3 is using. </del> <ins> traffic patterns that are similar to HTTP(S) over TCP.  UDP ports may be blocked for any port numbers that are not commonly used, such as UDP 53 for DNS.  Even when UDP ports are not blocked and QUIC packets can flow, streaming operators (and network operators) may severely rate-limit this traffic because they do not expect to see legitimate high-bandwidth traffic such as streaming media over the UDP ports that HTTP/3 is using. </ins> As noted in hol-blocking, because TCP provides a reliable, in-order delivery service for applications, any packet loss for a TCP

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.


Question: While QUIC is designed as a general-purpose transport protocol, and can carry different application-layer protocols, the current <del> standardized mapping is for HTTP/3 I-D.ietf-quic-http, which describes how QUIC transport features are used for HTTP.  The convention is for HTTP/3 to run over UDP port 443 Port443 but this is not a strict requirement. </del> <ins> standardized mapping is for HTTP/3 RFC9114, which describes how QUIC transport services are used for HTTP.  The convention is for HTTP/3 to run over UDP port 443 Port443 but this is not a strict requirement. </ins> When HTTP/3 is encapsulated in QUIC, which is then encapsulated in UDP, streaming operators (and network operators) might see UDP <del> traffic patterns that are similar to HTTP(S) over TCP.  Since earlier versions of HTTP(S) rely on TCP, UDP ports may be blocked for any port numbers that are not commonly used, such as UDP 53 for DNS. Even when UDP ports are not blocked and HTTP/3 can flow, streaming operators (and network operators) may severely rate-limit this traffic because they do not expect to see legitimate high-bandwidth traffic such as streaming media over the UDP ports that HTTP/3 is using. </del> <ins> traffic patterns that are similar to HTTP(S) over TCP.  UDP ports may be blocked for any port numbers that are not commonly used, such as UDP 53 for DNS.  Even when UDP ports are not blocked and QUIC packets can flow, streaming operators (and network operators) may severely rate-limit this traffic because they do not expect to see legitimate high-bandwidth traffic such as streaming media over the UDP ports that HTTP/3 is using. </ins> As noted in hol-blocking, because TCP provides a reliable, in-order delivery service for applications, any packet loss for a TCP

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?


Question: While QUIC is designed as a general-purpose transport protocol, and can carry different application-layer protocols, the current <del> standardized mapping is for HTTP/3 I-D.ietf-quic-http, which describes how QUIC transport features are used for HTTP.  The convention is for HTTP/3 to run over UDP port 443 Port443 but this is not a strict requirement. </del> <ins> standardized mapping is for HTTP/3 RFC9114, which describes how QUIC transport services are used for HTTP.  The convention is for HTTP/3 to run over UDP port 443 Port443 but this is not a strict requirement. </ins> When HTTP/3 is encapsulated in QUIC, which is then encapsulated in UDP, streaming operators (and network operators) might see UDP <del> traffic patterns that are similar to HTTP(S) over TCP.  Since earlier versions of HTTP(S) rely on TCP, UDP ports may be blocked for any port numbers that are not commonly used, such as UDP 53 for DNS. Even when UDP ports are not blocked and HTTP/3 can flow, streaming operators (and network operators) may severely rate-limit this traffic because they do not expect to see legitimate high-bandwidth traffic such as streaming media over the UDP ports that HTTP/3 is using. </del> <ins> traffic patterns that are similar to HTTP(S) over TCP.  UDP ports may be blocked for any port numbers that are not commonly used, such as UDP 53 for DNS.  Even when UDP ports are not blocked and QUIC packets can flow, streaming operators (and network operators) may severely rate-limit this traffic because they do not expect to see legitimate high-bandwidth traffic such as streaming media over the UDP ports that HTTP/3 is using. </ins> As noted in hol-blocking, because TCP provides a reliable, in-order delivery service for applications, any packet loss for a TCP

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


Question: Each report share has a corresponding task ID, report metadata (report ID and, timestamp), the public share sent to each Aggregator, and the recipient's encrypted input share.  Let "task_id", <del> "metadata", "public_share", and "encrypted_input_share" denote these values, respectively.  Given these values, an aggregator decrypts the input share as follows.  First, it constructs an "InputShareAad" message from "task_id", "metadata", and "public_share".  Let this be denoted by "input_share_aad".  Then, the aggregator looks up the HPKE config and corresponding secret key indicated by "encrypted_input_share.config_id" and attempts decryption of the payload with the following procedure: </del> <ins> "report_metadata", "public_share", and "encrypted_input_share" denote these values, respectively.  Given these values, an aggregator decrypts the input share as follows.  First, it constructs an "InputShareAad" message from "task_id", "report_metadata", and "public_share".  Let this be denoted by "input_share_aad".  Then, the aggregator looks up the HPKE config and corresponding secret key indicated by "encrypted_input_share.config_id" and attempts decryption of the payload with the following procedure: </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).

All references to "metadata" are now to "report_metadata". References to "CollectReq" are now "CollectionReq". References to "CollectResp" are now "Collection".
The current version of the draft renames field "metadata" to "report_metadata" in the Report structure, but does not do the same renaming in the ReportShare and InputShareAad structures. I think this naming should be consistent, though I don't have a preference between the alternatives.
Thanks for pointing this out! Can you send prepare a PR?
Fixed in
Thanks a bunch NAME


Question: 7. <del> Since client can receive DNS SVCB answers over unencrypted DNS, on- </del> <ins> Since clients can receive DNS SVCB answers over unencrypted DNS, on- </ins> path attackers can prevent successful discovery by dropping SVCB packets.  Clients should be aware that it might not be possible to distinguish between resolvers that do not have any Designated

Three changes Re-ordered statements in the abstract to place the restriction on unencrypted bootstrapping next to the unencrypted statement (instead of after the encrypted bootstrapping statement, which makes the restriction seem like a conflict where it excludes bootstrapping from a name instead of an IP address) Re-ordered security considerations (placed the "clients can opt to do other things" statement at the end after the MUSTs/SHOULDs for this doc) s/client/clients for grammatical correctness


Question: QUIC is an end-to-end transport protocol.  No information in the protocol header, even that which can be inspected, is mutable by the network.  This is achieved through integrity protection of the wire <del> image WIRE-IMAGE.  Encryption of most control signaling means that less information is visible to the network than is the case with TCP. </del> <ins> image WIRE-IMAGE.  Encryption of most transport-layer control signaling means that less information is visible to the network than is the case with TCP. </ins> Integrity protection can also simplify troubleshooting, because none of the nodes on the network path can modify transport layer

URL , can we be more specific here about the control signal we are talking about is QUIC protocol control signal not really other kind of control signal that the network might have? also here it is compared with TCP, should it also mention TCP with TLS?
I added the term "transport-layer" in PR . Does that help. I don't think TLS needs to be mentioned here.


Question: maximum value for a Push ID that the server can use in a PUSH_PROMISE frame.  Consequently, this also limits the number of push streams that the server can initiate in addition to the limit set by the QUIC <del> MAX_STREAM_ID frame. </del> <ins> MAX_STREAMS frame. </ins> The MAX_PUSH_ID frame is always sent on the control stream.  Receipt of a MAX_PUSH_ID frame on any other stream MUST be treated as a

Makes two minor fixes in the GOAWAY text, both editorial: Correctly references the MAXSTREAMS frame instead of the MAXSTREAM_ID frame Notes that servers need not (and cannot) send a GOAWAY if the client has used up all the possible stream IDs; the client simply won't be making any more requests.
This is a whatever...


Question: 2.6. <del> The following diagram shows the communication between each component. </del> <ins> The following diagram shows the communication between each component in the case where the Captive Network has a User Portal, and the User Equipment chooses to visit the User Portal in response to discovering and interacting with the API Server. </ins> In the diagram:

Whether or not the user visits the User Portal is optional: the user could choose not to, or the Captive Portal may not provide one. Clarify that the case presented in the component diagram is the one where the User Portal is present, and where the user chooses to visit it.


Question: layer headers.  Note that in some cases a zero-length HashValue (indicated by "") is passed to HKDF-Expand-Label. <ins> Note: with common hash functions, any label longer than 12 characters requires an additional iteration of the hash function to compute. The labels in this specification have all been chosen to fit within this limit. </ins> Given a set of n InputSecrets, the final "master secret" is computed by iteratively invoking HKDF-Extract with InputSecret_1, InputSecret_2, etc.  The initial secret is simply a string of

Per mailing list discussion, this allows us to have every HKDF-Expand just have one hash block of info. NAME would love your feedback.
Shouldn't "TLS 1.3, " be shortened to "tls13 "? Without that, several of the labels would seem to blow blocks. Also, "derived secret" seems unchanged, and is too long to fit into a block, even with "tls13 " prefix.
Oops. Good catch. Script fail.
NAME fixed?
Yeah, I'll add a note. I still have an issue for the ChangeLog so I will add this as well
Also, minimum label size looks to be still 10, presumably should be 7 with 6-byte prefix (as it 10 with 9-byte prefix).


Question: to the PA protocol: The helper handles each sub-request "PAAggregateSubReq" as follows. <del> It first checks that checks that "PAAggregateSubReq.proto == PAParam.proto".  If not, it aborts and alerts the leader with "incorrect protocol for sub-request".  Otherwise, it computes the HPKE context as where "sk" is its HPKE key and computes the body of the "PAAggregateSubResp".  It then updates its state according to the PA protocol.  After processing all of the sub-requests, the helper encrypts its updated state and constructs its response to the aggregate request. </del> <ins> It first looks up the HPKE config and corresponding secret key associated with "helper_share.config_id".  If not found, then the sub-response consists of an "unrecognized config" alert.  [TODO: We'll want to be more precise about what this means.  See issue#57.] Next, it attempts to decrypt the payload with the following procedure: where "sk" is the HPKE secret key.  If decryption fails, then the sub-response consists of a "decryption error" alert.  [See issue#57.] Otherwise, the helper handles the request for its plaintext input share "input_share" and updates its state as specified by the PA protocol. After processing all of the sub-requests, the helper encrypts its updated state and constructs its response to the aggregate request. </ins> 3.4.2.1.1.

Update the aggregate request text to reflect recent protocol changes. Relax the requirement that all sub-requests use the same key config. Do URL, which was previously merged into the wrong branch.
NAME NAME please have a look when you can.
Per , the first step in the upload protocol is a POST to the leader server which returns information about helpers. In some yet to be specified cases, the response could include a challenge to be incorporated into reports. However I suspect that in most cases, the response will be static and so an idempotent, cacheable GET would be a better fit, especially since it would save clients a roundtrip to the helper on the hot path of submitting reports, and would make it possible for helpers to implement uploadstart by putting some static content in a CDN. HTTP GET requests don't have bodies, so we would have to figure out how to encode all the data in the current into the URI, which could be done as either path fragments in the URI (i.e., ) or as query parameters (i.e., ). We would of course need to make sure that don't preclude proof systems like the one alleged to exist in section 5.2 of "BBCp19".
Also, because the current protocol uses a POST for uploadstart, the implication is that for every report they submit, clients need to hit and then . If we make idempotent and cacheable, then that request can be skipped in the majority of cases. I suppose the significance of this depends on how we expect clients to use PDA: are they going to be continuously streaming hundreds or thousands of reports into the leader as events of interest occur (in which case I think eliminating requests is interesting) or are they expected to accumulate several reports locally before submitting them all in one or a few requests?
I agree this would be a good change, if we can swing it. It's notable that Prio v2 and HH don't require a random challenge from the leader. But as discussed, this is useful for other proof systems one might want to implement for Prio. For example, see the data type implemented here: URL See below for an explanation. There might be other ways to implement the challenge. One possibility is to derive the challenge from the TLS key schedule (or from whatever state the client and leader share for the secure channel).


Question: to the PA protocol: The helper handles each sub-request "PAAggregateSubReq" as follows. <del> It first checks that checks that "PAAggregateSubReq.proto == PAParam.proto".  If not, it aborts and alerts the leader with "incorrect protocol for sub-request".  Otherwise, it computes the HPKE context as where "sk" is its HPKE key and computes the body of the "PAAggregateSubResp".  It then updates its state according to the PA protocol.  After processing all of the sub-requests, the helper encrypts its updated state and constructs its response to the aggregate request. </del> <ins> It first looks up the HPKE config and corresponding secret key associated with "helper_share.config_id".  If not found, then the sub-response consists of an "unrecognized config" alert.  [TODO: We'll want to be more precise about what this means.  See issue#57.] Next, it attempts to decrypt the payload with the following procedure: where "sk" is the HPKE secret key.  If decryption fails, then the sub-response consists of a "decryption error" alert.  [See issue#57.] Otherwise, the helper handles the request for its plaintext input share "input_share" and updates its state as specified by the PA protocol. After processing all of the sub-requests, the helper encrypts its updated state and constructs its response to the aggregate request. </ins> 3.4.2.1.1.

NAME and NAME have both pointed out that adding this dependency on the underlying transport is problematic. Another possibility might be to use some external source of randomness that the leader trusts. comes to mind, though it is somewhat problematic because clients would effectively use the same randomness for some period of time. The security model of BBCp19 would need to be extended to account for this. (I believe this could be done.) Some other solution I'm missing? _ This data type is used to compute the mean and variance of each integer in a sequence of integers. To do so, the client encodes each integer of its input vector as a pair , where is the representation of as a bit vector and is equal to . To prove validity of its input, the proof needs to do two things: verify that (a) is indeed a bit vector, and (b) the integer encoded by is the square root of . In order to check these things efficiently, the proof system uses a random challenge generated by the leader. At a very high level, the client wants to prove that the following boolean conjunction holds: is 1 or 0 AND is 1 or 0 AND ... AND encodes the square root of . In Prio, we need to represent this conjunction as an arithmetic circuit, which the random challenge helps us do. In a bit more detail now: Let's rewrite this expression as w[0] AND w[1] AND ... AND w[s]. As an arithmetic circuit, this expression is actually , where if and only if w[i] is true. If the input is valid, then the expression should evaluate to . But because we're talking about field elements, a malicious client may try to play games like setting and so that the sum is still , but the statement is false. To detect this, we actually compute the following expression: , where is a randomly chosen field element. If the statement is true, then the expression evaluates to . If the statement is false, then with high probability, the expression will evaluate to something other than . (This idea comes from Theorem 5.3)
Thank you for the detailed explanalation NAME Having thought about it more, I suspect an HTTP GET wouldn't be the right way to support proof systems that need a challenge.


Question: to the PA protocol: The helper handles each sub-request "PAAggregateSubReq" as follows. <del> It first checks that checks that "PAAggregateSubReq.proto == PAParam.proto".  If not, it aborts and alerts the leader with "incorrect protocol for sub-request".  Otherwise, it computes the HPKE context as where "sk" is its HPKE key and computes the body of the "PAAggregateSubResp".  It then updates its state according to the PA protocol.  After processing all of the sub-requests, the helper encrypts its updated state and constructs its response to the aggregate request. </del> <ins> It first looks up the HPKE config and corresponding secret key associated with "helper_share.config_id".  If not found, then the sub-response consists of an "unrecognized config" alert.  [TODO: We'll want to be more precise about what this means.  See issue#57.] Next, it attempts to decrypt the payload with the following procedure: where "sk" is the HPKE secret key.  If decryption fails, then the sub-response consists of a "decryption error" alert.  [See issue#57.] Otherwise, the helper handles the request for its plaintext input share "input_share" and updates its state as specified by the PA protocol. After processing all of the sub-requests, the helper encrypts its updated state and constructs its response to the aggregate request. </ins> 3.4.2.1.1.

Assuming that a unique challenge is needed for every report[^1], then intuitively the leader would have to maintain a mapping of reports to issued challenges so that the leader and helper(s) can later use the challenge value when evaluating the client's proof. But now that the leader has to maintain state per-report, I think the upload protocol is missing a notion of a report identifier. The contains a , but IIUC the will be the same across many reports[^2] (e.g., if a browser is sending daily reports on how often users click a button, they use the same every time they submit). So needs to include a report ID, which must be echoed back to the leader in so that the leader can look up the challenge for the report. [^1] Is this true? Or would it suffice for each to use the same challenge across multiple reports? [^2] I notice that the design doc lacks a strong definition of a report and how it relates to tasks, inputs, measurements: because GET should be "safe" in the sense of not changing any server state, and it seems like the server would have to maintain a mapping of reports to issued challenges so that the leader and helper can later use the challenge value when evaluating the client's proof.
Continuing on the assumption that causes a state change on the leader, POST is then the appropriate HTTP method to use. , though it's a little unusual. There may be some big drawback to cacheable POST responses, but I think that might let us support proof systems that require live challenges while also eliminating the majority of requests when the proof system is such that the will change very rarely.
The requirement for the leader to keep state across upload start and upload finish may be avoidable. Suppose the leader has a long-lived PRF key K. It would could choose a random nonce N, compute R = PRF(K, N) as the challenge, and hand (N, R) to the client in its upload start response. The client would then use R to generate the proof and send N to the leader in its upload finish request. The leader could then re-derive R. NAME and I kicked around the idea of having the client generate a report ID. I think this may end up being useful, but we weren't sure if it was better to have the leader choose it or the client.


Question: to the PA protocol: The helper handles each sub-request "PAAggregateSubReq" as follows. <del> It first checks that checks that "PAAggregateSubReq.proto == PAParam.proto".  If not, it aborts and alerts the leader with "incorrect protocol for sub-request".  Otherwise, it computes the HPKE context as where "sk" is its HPKE key and computes the body of the "PAAggregateSubResp".  It then updates its state according to the PA protocol.  After processing all of the sub-requests, the helper encrypts its updated state and constructs its response to the aggregate request. </del> <ins> It first looks up the HPKE config and corresponding secret key associated with "helper_share.config_id".  If not found, then the sub-response consists of an "unrecognized config" alert.  [TODO: We'll want to be more precise about what this means.  See issue#57.] Next, it attempts to decrypt the payload with the following procedure: where "sk" is the HPKE secret key.  If decryption fails, then the sub-response consists of a "decryption error" alert.  [See issue#57.] Otherwise, the helper handles the request for its plaintext input share "input_share" and updates its state as specified by the PA protocol. After processing all of the sub-requests, the helper encrypts its updated state and constructs its response to the aggregate request. </ins> 3.4.2.1.1.

To conform to the security model of BBCp19, each report would need its own challenge. It may be possible to relax this, but this would be risky.
I like this a lot! I'm a big fan of deterministic derivation schemes like this -- besides reintroducing the possibility of an idempotent GET, eliminating storage requirements for challenges makes it much easier to scale up leaders. If the report IDs are big enough (UUIDs?) we could even use them as . I would say the leader, because if you let the client do it, then the leader would have to check for things like collisions with existing report IDs or other malicious ID choices. If the leader chooses it, then all it has to do is generate a few bytes of randomness. I agree.
NAME we'd appreciate your feedback on this issue because it involves security considerations for ZKP systems on secret-shared data [1]. I'll quickly sum up the problem so that you don't need to worry about the conversation above. The problem at hand regards systems like [1, Theorem 5.3] that require interaction between the prover and verifier. Specifically, we're considering a special case of Theorem 5.3 in which the verifier sends the prover a random "challenge" that is used to generate and verify a (non-interactive) FLPCP (call it ): We refer to the prover's first and second messages as "upload start" and "upload finish" respectively. Each message corresponds to an HTTP request made by a web client to a web server. Ideally, the verifier would be able to handle these requests statelessly, i.e., without having to store . Consider the following change. Suppose the verifier has a long-term PRF key . In response to an upload start request, it chooses a unique "upload id" --- maybe this is a 16-byte integer chosen at random --- and computes and sends to the prover. In the upload finish request, the prover sends so that the verifier can re-derive instead of needing to remember it: The (potential) problem with this variant is that a malicious client is able to replay a challenge as many times as it wants.


Question: to the PA protocol: The helper handles each sub-request "PAAggregateSubReq" as follows. <del> It first checks that checks that "PAAggregateSubReq.proto == PAParam.proto".  If not, it aborts and alerts the leader with "incorrect protocol for sub-request".  Otherwise, it computes the HPKE context as where "sk" is its HPKE key and computes the body of the "PAAggregateSubResp".  It then updates its state according to the PA protocol.  After processing all of the sub-requests, the helper encrypts its updated state and constructs its response to the aggregate request. </del> <ins> It first looks up the HPKE config and corresponding secret key associated with "helper_share.config_id".  If not found, then the sub-response consists of an "unrecognized config" alert.  [TODO: We'll want to be more precise about what this means.  See issue#57.] Next, it attempts to decrypt the payload with the following procedure: where "sk" is the HPKE secret key.  If decryption fails, then the sub-response consists of a "decryption error" alert.  [See issue#57.] Otherwise, the helper handles the request for its plaintext input share "input_share" and updates its state as specified by the PA protocol. After processing all of the sub-requests, the helper encrypts its updated state and constructs its response to the aggregate request. </ins> 3.4.2.1.1.

This behavior isn't accounted for in Definition 3.9, so when you try to "compile" this system into a ZKP on secret-shared data as described in Section 6.2 and consider its security with respect to Definition 6.4 (Setting I), you can no longer appeal to Theorem 6.6 to reason about its security. Our question is therefore: What impact do you expect this change to have on the soundness of the system? Can you think of an attack that exploits the fact that challenges can be replayed? My expectation is that the problem is merely theoretical, i.e., we ought to be able to close the gap with an appropriate refinement of Definition of 3.9 and reproving Theorem 5.3. [1] URL
Thanks for looping me in. In the discussion below, I am assuming that the client sends its data vector (called "x" in the paper [1]) at the same time as it sends its proof to the verifier. Even if the client sends its data vector in the first message, it seems that the client can reuse an (R,N) pair from upload request 1 to later submit upload request 2, in which case the client sees (R, N) before it chooses its data vector and proof. So I think we can assume that the client's data vector and proof can depend on the verifier's random value R. (I'll call this value "r" to match the paper.) For soundness: the client/prover must commit to its data vector before it learns the verifier's random value r. If the prover can choose its data value and proof in a way that depends on the verifier's randomness r, the soundness guarantee no longer holds. There is also an efficient attack. So I suspect that the stateless-verifier optimization is unsound. The attack works as follows: given r, the verifier finds a non-satisfying vector x such that sumi ri C(Ai(x)) = 0, again using the notation from Theorem 5.3 in the paper [1]. For the circuits C used in most applications, it will be easy to find such a vector x. Then, the prover constructs a proof that x that sumi ri C(Ai(x)) = 0.


Question: to the PA protocol: The helper handles each sub-request "PAAggregateSubReq" as follows. <del> It first checks that checks that "PAAggregateSubReq.proto == PAParam.proto".  If not, it aborts and alerts the leader with "incorrect protocol for sub-request".  Otherwise, it computes the HPKE context as where "sk" is its HPKE key and computes the body of the "PAAggregateSubResp".  It then updates its state according to the PA protocol.  After processing all of the sub-requests, the helper encrypts its updated state and constructs its response to the aggregate request. </del> <ins> It first looks up the HPKE config and corresponding secret key associated with "helper_share.config_id".  If not found, then the sub-response consists of an "unrecognized config" alert.  [TODO: We'll want to be more precise about what this means.  See issue#57.] Next, it attempts to decrypt the payload with the following procedure: where "sk" is the HPKE secret key.  If decryption fails, then the sub-response consists of a "decryption error" alert.  [See issue#57.] Otherwise, the helper handles the request for its plaintext input share "input_share" and updates its state as specified by the PA protocol. After processing all of the sub-requests, the helper encrypts its updated state and constructs its response to the aggregate request. </ins> 3.4.2.1.1.

The proof is valid, so the verifier will accept always. To remove interaction, the Fiat-Shamir-like technique in Section 6.2.3 is probably the best way to go. We only sketched the idea in the paper (without formal analysis or proof) but if it's going to be important for your application, I'm happy to sanity-check the protocol you come up with. Another idea—the details of which I have not checked carefully—would be to modify your protocol as follows: The prover sends its data vector x to verifiers, split into k shares x1, ..., xk—one for each of the k verifiers. Each verifier holds a PRF key ki computes ri = PRF(ki, xi) and they return r = r1 + ... + rk to the prover. The verifiers store no state. The prover computes the proof pi and sends (x1, pi1), ..., (xk, pik) to the verifiers—again, one for each of the k verifiers. Each verifier computes ri = PRF(ki, xi), they jointly compute r = r1 + ... + r_k, and check the proof using randomness r. The Fiat-Shamir-like approach seems slightly cleaner to me, since it only requires a single prover-to-verifier message. Either way, please let me know if I misunderstood your question or if anything else here is unclear. [1] URL
Thanks for the nice explanation, Henry. It's perfectly clear and answers my question. It also points to something that I, for one, missed the importance of: the need for the prover to "commit" to the input shares before generating the r-dependent proof. I appreciate you pointing this out. The only requirement for making upload start not idempotent is so that the randomness r can be sent to the client. It would be nice to do away with this requirement. I think the way forward is using the Fiat-Shamir technique. NAME I'll get cracking on this and let you know when I have something for you to look at.
NAME I believe the change is something like this. Let be a hash function with range . Suppose there are verifiers. The prover proceeds as follows: Splits its inputs into shares .


Question: to the PA protocol: The helper handles each sub-request "PAAggregateSubReq" as follows. <del> It first checks that checks that "PAAggregateSubReq.proto == PAParam.proto".  If not, it aborts and alerts the leader with "incorrect protocol for sub-request".  Otherwise, it computes the HPKE context as where "sk" is its HPKE key and computes the body of the "PAAggregateSubResp".  It then updates its state according to the PA protocol.  After processing all of the sub-requests, the helper encrypts its updated state and constructs its response to the aggregate request. </del> <ins> It first looks up the HPKE config and corresponding secret key associated with "helper_share.config_id".  If not found, then the sub-response consists of an "unrecognized config" alert.  [TODO: We'll want to be more precise about what this means.  See issue#57.] Next, it attempts to decrypt the payload with the following procedure: where "sk" is the HPKE secret key.  If decryption fails, then the sub-response consists of a "decryption error" alert.  [See issue#57.] Otherwise, the helper handles the request for its plaintext input share "input_share" and updates its state as specified by the PA protocol. After processing all of the sub-requests, the helper encrypts its updated state and constructs its response to the aggregate request. </ins> 3.4.2.1.1.

For each , do: Choose blinding factor from at random. Let . Set . Derive field element from (e.g., seed a PRNG with and map the output to a field element by rejection sampling on chunks of the output). Generate the proof using and . Split into shares . For each send to the -th verifier.
Yes, this looks right to me!
As of URL, the upload start request no longer exists. When we're ready to specify Prio, we'll want to include the Fiat-Shamir approach for protocols that use joint randomness. I've noted this and am closing the issue.


Question: configured not to perform a cookie exchange.  The default SHOULD be that the exchange is performed, however.  In addition, the server MAY choose not to do a cookie exchange when a session is resumed or, more <del> generically, when the DTLS handshake uses a PSK-based key exchange. Servers which process 0-RTT requests and send 0.5-RTT responses without a cookie exchange risk being used in an amplification attack if the size of outgoing messages greatly exceeds the size of those that are received.  A server SHOULD limit the amount of data it sends toward a client address before it verifies that the client is able to receive data at that address.  A client address is valid after a cookie exchange or handshake completion.  A server MAY apply a higher limit based on heuristics, such as if the client uses the same IP address as the connection on which the cookie was created.  Clients MUST be prepared to do a cookie exchange with every handshake. </del> <ins> generically, when the DTLS handshake uses a PSK-based key exchange and the IP address matches one associated with the PSK.  Servers which process 0-RTT requests and send 0.5-RTT responses without a cookie exchange risk being used in an amplification attack if the size of outgoing messages greatly exceeds the size of those that are received.  A server SHOULD limit the amount of data it sends toward a client address to three times the amount of data sent by the client before it verifies that the client is able to receive data at that address.  A client address is valid after a cookie exchange or handshake completion.  Clients MUST be prepared to do a cookie exchange with every handshake.  Note that cookies are only valid for the existing handshake and cannot be stored for future handshakes. </ins> If a server receives a ClientHello with an invalid cookie, it MUST terminate the handshake with an "illegal_parameter" alert.  This

Thanks for this document -- badly needed and easy to read. Thanks also to Bernard Adoba for his recent tsvart review. I support his comments. COMMENTS: Sec 2. It might be useful to introduce the term "epoch" in the glossary, for those who read this front to back. Sec 4.2.3: "The encrypted sequence number is computed by XORing the leading bytes of the Mask with the sequence number. Decryption is accomplished by the same process." The text is unclear if the XOR is applied to the expanded sequence number or merely the 1-2 octets on the wire. I presume it's the latter, but this should be clarified. Sec 4.2.3: It's implied here that the snkey rotates with the epoch. As this is different from QUIC, it's probably worth spelling out. Sec 5.1 is a bit vague about the amplification limit; why not at least RECOMMEND 3, as we've converged on this elsewhere? Sec 5.1. Reading between the lines, it's clear that the cookie can't be used as address verification across connections in the way that a NEWTOKEN token is. It would be good to spell this out for clients -- use the resumption token or whatever instead. Sec 7.2 "As noted above, the receipt of any record responding to a given flight MUST be taken as an implicit acknowledgement for the entire flight." I think this should be s/entire flight/entire previous flight? Sec 7.2 "Upon receipt of an ACK that leaves it with only some messages from a flight having been acknowledged an implementation SHOULD retransmit the unacknowledged messages or fragments." This language appears inconsistent with Figure 12, where Record 1 has not been acknowledged but is also not retransmitted. It appears there is an implied handling of empty ACKs that isn't written down anywhere in Sec 7.2 Sec 9. Should there be any flow control limits on newconnectionid? Or should receivers be free to simply drop CIDs they can't handle? It might be good to specify. Finally, a really weird one. Reading this document and references to connection ID prompted to me to think how QUIC-LB could apply to DTLS.


Question: configured not to perform a cookie exchange.  The default SHOULD be that the exchange is performed, however.  In addition, the server MAY choose not to do a cookie exchange when a session is resumed or, more <del> generically, when the DTLS handshake uses a PSK-based key exchange. Servers which process 0-RTT requests and send 0.5-RTT responses without a cookie exchange risk being used in an amplification attack if the size of outgoing messages greatly exceeds the size of those that are received.  A server SHOULD limit the amount of data it sends toward a client address before it verifies that the client is able to receive data at that address.  A client address is valid after a cookie exchange or handshake completion.  A server MAY apply a higher limit based on heuristics, such as if the client uses the same IP address as the connection on which the cookie was created.  Clients MUST be prepared to do a cookie exchange with every handshake. </del> <ins> generically, when the DTLS handshake uses a PSK-based key exchange and the IP address matches one associated with the PSK.  Servers which process 0-RTT requests and send 0.5-RTT responses without a cookie exchange risk being used in an amplification attack if the size of outgoing messages greatly exceeds the size of those that are received.  A server SHOULD limit the amount of data it sends toward a client address to three times the amount of data sent by the client before it verifies that the client is able to receive data at that address.  A client address is valid after a cookie exchange or handshake completion.  Clients MUST be prepared to do a cookie exchange with every handshake.  Note that cookies are only valid for the existing handshake and cannot be stored for future handshakes. </ins> If a server receives a ClientHello with an invalid cookie, it MUST terminate the handshake with an "illegal_parameter" alert.  This

The result is here: URL Please note the rather unfortunate third-to-last paragraph. I'm happy to take the answer that this use case doesn't matter, since I made it up today. But if it does, it would be very helpful if (1) DTLS 1.3 clients MUST include a connectionid extension in their ClientHello, even if zero length, and/or (2) this draft updated 4.1.4 of 8446 to allow the server to include connectionid in HelloRetryRequest even if the client didn't offer it. Thoughts? NITS: s/select(HandshakeType)/select(msg_type). Though with pseudocode your mileage may vary as to what's clearer. s/consitute/constitute Sec 5.7 In table 1, why include one ACK in the diagram but not the other? It's clear from the note, but the figure is a weird omission.
Addressed in URL


Question: 9. <del> Running DNS over https:// relies on the security of the underlying HTTP connection.  By requiring at least RFC7540 levels of support for TLS this protocol expects to use current best practices for secure </del> <ins> Running DNS over HTTPS relies on the security of the underlying HTTP connection.  By requiring at least RFC7540 levels of support for TLS, this protocol expects to use current best practices for secure </ins> transport. Session level encryption has well known weaknesses with respect to

I think that we should specify behaviour for truncated responses with the TC bit more clearly. Right now the text is a little vague. There seem to be three options: mandate the generation of an HTTP error mandate the forwarding of the truncated response explicitly allow either forbid the use of truncated responses and recommend fallback to TCP if the DNS API server is just making DNS requests to another server I think that the draft current vacilates in the direction of 3, but I tend to like 1 or 4. Choosing 4 wouldn't be consistent with the dnsop DNS-over-HTTP draft using DOH though.
is the right answer, so the text should just be stronger about it. If DOH requires or , that means that the DNS server code will need to be heavily reworked to capture these from going out, which seems pointless. DOH clients can act like normal stubs: either they do or don't know about truncation and TC.
I agree - I don't think doh needs to behave differently here than any other datagram based client - i.e. its not a transport error. NAME can you do a pr to clarify?
Will do.


Question: related to the JSON literals "true" and "false" and have no direct syntactical representation in JSONPath. <del> "NodesType" is an abstraction of a "filter-path" (which appears in a test expression or as a function argument).  Members of </del> <ins> "NodesType" is an abstraction of a "filter-query" (which appears in a test expression or as a function argument).  Members of </ins> "NodesType" have no direct syntactical representation in JSONPath. The abstract instances above can be obtained from the concrete

XPath terminology is a bit weird, because XPath has a somewhat competing standard XQuery... XPath is based on DSSSL, which had a complex expression language for building paths in the XML trees that was based on the Scheme language. (Which is the experience that makes me hesitant just jumping headlong into another expression language...)
LGTM with that change. Thanks!


Question: meanings. Readers are expected to be familiar with the terms and concepts <del> described in RFC7252 and RFC7641.  Readers are also expected to be familiar with RFC7049 and understand I-D.greevenbosch-appsawg-cbor- cddl.  Terminology for constrained environments, such as "constrained device", "constrained-node network", is defined in RFC7228. </del> <ins> described in CoAP RFC7252, Observe RFC7641, Blockwise RFC7959, COSE I-D.ietf-cose-msg, CBOR RFC7049, CDDL I-D.greevenbosch-appsawg-cbor- cddl, and constrained environments RFC7228. </ins> 2.

UTF-8 über alles.
Abstract: Explain that OSCOAP works across intermediaries and over any layer.Intro: Explain that OSCOAP works for Observe and Blockwise.Intro: Move details on replay and freshness to some other section.Terminology: Explain the references and add COSE.


Question: old key.  The "outer" JWS represents the current account holder's assent to this request. <del> On receiving keyChange request, the server MUST perform the following steps in addition to the typical JWS validation: </del> <ins> On receiving a keyChange request, the server MUST perform the following steps in addition to the typical JWS validation: </ins> Validate the POST request belongs to a currently active account, as described in message-transport.

Some grammar fixes POST-as-GETs should be indicated as having a signature in flow Specify that the JWS profile we define in Section 6.2 is for ACME request bodies, to distinuish it from the profile used for External Account Binding. They have incompatible requirements: MUST NOT use MAC vs MUST use MAC. Clarify that Replay-Nonce is an HTTP header field since it's defined near JWS header parameters. Don't include "wildcard": false in examples. Section 7.1.4 requires that it MUST be absent when false. There was lingering language in the Section 7.5.1 about "updating" an authorization body in response to fields in a challenge POST. Since the challenge POST is now described explicitly as "an empty JSON body", this language didn't make sense. Fixed.
Per offline review, I added back in a sentence that contained a MUST:


Question: of the following octets.  Note that the first two octets indicate the extension type ("supported_groups" extension): <ins> As of TLS 1.3, servers are permitted to send the "supported_groups" extension to the client.  If the server has a group it prefers to the ones in the "key_share" extension but is still willing to accept the ClientHello, it SHOULD send "supported_groups" to update the client's view of its preferences.  Clients MUST NOT act upon any information found in "supported_groups" prior to successful completion of the handshake, but MAY use the information learned from a successfully completed handshake to change what groups they offer to a server in subsequent connections. </ins> [[TODO: IANA Considerations.]] 6.3.2.3.

You'll also need to update the last paragraph of the MTI Extensions section (currently section 8.2). Now that we're going to be down to only one client-only extension defined in this document, that whole paragraph can probably be cut in favor of a one-off note in the remaining extension's section (signature_algorithms will be the last).
So that the clients can learn about what groups a server supports. The idea here is to allow a client to offer P256 but then learn that a server supports 25519 for next time without the server having to reject.


Question: The status of the Connection, which can be one of the following: Establishing, Established, Closing, or Closed. <ins> Whether the connection can be used to send data.  A connection can not be used for sending if the connection was created unidirectional receive only or if a message with the final property was sent over this connection. Whether the connection can be used to receive data.  A connection can not be used for reading if the connection was created unidirectional send only or if a message with the final property received was received.  The latter is only supported by certain transport protocols, e.g., by TCP as half-closed connection. </ins> Transport Features of the protocols that conform to the Required and Prohibited Transport Preferences, which might be selected by the transport system during Establishment.  These features

This closed As we have no methods on a connection object that query properties, cansend/canreceive also became connection properties. As with all the other connection properties, these are indeed a little underspecified. I want to keep it that way till we fix it for all connection properties with and
Aehm… that comes surprising – NAME why didn't you bring this up in ? Do you think this pull-request is the right place to discuss this issue? I would prefer we merge it now and make a new issue for the question whether the uni-directional connection should be more reflected in the connection. I am not particularly found on keeping it a selection property but would like to have the concept of unidirectional connections rather sooner than later. Our consensus in was that unidirectional connection should use the same abstraction for QUIC unidirectional streams, multicast sources/sinks, and half-closed connections – despite they are constructed in completely different ways. The Selection property is just for the QUIC style ones…

Aehm… that comes surprising – NAME why didn't you bring this up in ? Do you think this pull-request is the right place to discuss this issue? Happy to move the discussion back over to the issue if you like. Will continue the thread here for now though since you've got points to address below. fine, but keep open then, because adding the selection property is possibly-necessary but definitely-not-sufficient to fully address unidirectional connection semantics. good, I agree with this. then I'm confused why this PR purports to since it only addresses part of the issue...?
NAME agreed. The question (and discussion) was about support for including some support for unidirectional connections (with focus on QUIC and half-closed), but omitted details on multicast, as this has not laned yet. So indeed, this pull-requests does everything requested by , but does not fully capture the topic. I will move the open issues (Multicast support, is the Selection Property enough, find some place for more explanation of Uni/Bidi Streams) to once this PR is merged.
In Section 9 – "Setting and Querying of Connection Properties" – the properties of an established connection and path are heavily underspecified. Should be fixed along with .
I'm working on this issue within our pull request for , using the terminology as simplified there.


Question: The status of the Connection, which can be one of the following: Establishing, Established, Closing, or Closed. <ins> Whether the connection can be used to send data.  A connection can not be used for sending if the connection was created unidirectional receive only or if a message with the final property was sent over this connection. Whether the connection can be used to receive data.  A connection can not be used for reading if the connection was created unidirectional send only or if a message with the final property received was received.  The latter is only supported by certain transport protocols, e.g., by TCP as half-closed connection. </ins> Transport Features of the protocols that conform to the Required and Prohibited Transport Preferences, which might be selected by the transport system during Establishment.  These features

For Section 9, I'm proposing the following: An application can query Connection Properties on a connection. Connection Properties include the status of the connection and whether it is possible to send and to receive data on it. (One of them might not be possible, e.g., in a unidirectional connection.) Additionally, Connection Properties include the Connection's Transport Properties. Transport Properties include Selection Properties, so the application can know which Protocol Selection Properties had been specified, and which ones are actually supported by the protocol that has been chosen. For Establishing connections (i.e., we are not done racing yet, the final protocols have not been selected yet), the application gets the Selection Properties that had been specified on the Preconnection earlier. For Established, Closing, or Closed connections (i.e., the protocols have been selected), the application learns whether the Selection Properties are actually provided by the selected protocols. For example, if the application specified earlier, "Control Checksum Coverage: Prefer", now it will get "Control checksum coverage: True", indicating that on this connection it can indeed control the checksum coverage. Transport Properties also include Protocol Properties (Generic and Specific). These Protocol Properties can still be modified by the application. Basically, Protocol Properties are the only thing that the application can still set, everything else it can only query at this point. Finally, the application can query properties of the path. Not sure how well we have to specify them, probably this part will be implementation-specific just like, e.g., the Interface Instance or Type. Does everyone agree, or is there anything to discuss here? For me this is not a big change, it's just putting what was already there in terms of . The only thing I'm wondering if whether the Protocol Selection Properties of the finally chosen protocols are still "Selection Properties", as the selection is already done. That's why they were called "Transport Features" before, but now I want less terms and they're almost the same thing...
OK - so I commented on these things before - but in these things I think I agree. I'd personally prefer to keep the features as attributes implemented in the transport (etc) and the properties as the things requested. Indeed this begs the question about what to call the resulting transport when you selected properties and now get a transport. I think still calling them (for consistency) properties seems really good to me. :-).


Question: The status of the Connection, which can be one of the following: Establishing, Established, Closing, or Closed. <ins> Whether the connection can be used to send data.  A connection can not be used for sending if the connection was created unidirectional receive only or if a message with the final property was sent over this connection. Whether the connection can be used to receive data.  A connection can not be used for reading if the connection was created unidirectional send only or if a message with the final property received was received.  The latter is only supported by certain transport protocols, e.g., by TCP as half-closed connection. </ins> Transport Features of the protocols that conform to the Required and Prohibited Transport Preferences, which might be selected by the transport system during Establishment.  These features

As QUIC has unidirectional streams, we might want to be able to accommodate them in our transport API. Do we want to handle them fundamentally different from a half-closed connection? This should match the decisions made in .
Side note: we don't support half-closed connections to begin with. As for unidirectional streams, is this the final decision in QUIC's design, or might it change? I'd suggest postponing this until we can be sure.
NAME regarding half-closed connections: we don't have any top-level state in the connection for being half-closed. However, being able to express that a message is final in a given direction when sending allows the same functionality. As I've discussed before, we absolutely need the ability to express sending a TCP FIN, but I whole-heartedly agree it doesn't need to be a top level concept. I think this is somewhat different from half-closing, since a unidirectional stream never allows sending on one end. You could imagine that it is a connection object that returns an error if you ever try to send on it, etc. If you wanted it to look like half-closing, you could open a connection (stream), and then mark it complete in a direction immediately? But that's a bit odd.
All ok, I agree with your suggestion to fin-mark a message
I agreee with NAME regarding half-closed. My suggestion is to explicitly state that connections can be uni- or bi-directional and to add a method to query that (cansend / canrecv). This aligns nicely with final messages after which can_send becomes false. I can do a PR for that.
Thanks! I like this
Still missing unidirectional stream for multicast. – please continue discussion of multicast aspects in The connection properties (can send/receive data) are still a little under-specified - please discuss in details in – we should fix this together with .
We need unidirectional connections. You know, for multicast. And ICN. And unidirectional streaming protocols.
This is a dup of , which has more discussion
I am re-opening this ticket as only dealt with unidirectional streams in unicast context (added with PR ): Current status: The current model for unidirectional streams supports half-closed connections and QUIC style unidirectional streams so far.


Question: The status of the Connection, which can be one of the following: Establishing, Established, Closing, or Closed. <ins> Whether the connection can be used to send data.  A connection can not be used for sending if the connection was created unidirectional receive only or if a message with the final property was sent over this connection. Whether the connection can be used to receive data.  A connection can not be used for reading if the connection was created unidirectional send only or if a message with the final property received was received.  The latter is only supported by certain transport protocols, e.g., by TCP as half-closed connection. </ins> Transport Features of the protocols that conform to the Required and Prohibited Transport Preferences, which might be selected by the transport system during Establishment.  These features

We don't know yet whether it is sufficient to have canread/canwrite property on the connection level to model unidirectional connections We don't have pre-connection methods to create multicast sessions yet
there's a selection property for this, tagged discuss, which we haven't discussed yet. I'm moving this down to the appendix as part of my restructuring in my work on
If help is still wanted here, I'm maybe up for an attempt or 2 at adding multicast support. Has there been some prior discussion I can review about rough ideas on pre-connection methods for multicast sessions, or should I just try making something up?

ok, with the caveat that we need to keep multicast as an open issueLGTM About the multicast concern: I agree that it would be good if this could function for multicast reception too, and I couldn't tell if this is suitable for multicast. I suspect that multicast support is much harder anyway - see RFC 3678... so for now, this is a good step forward IMO, and let's discuss how this relates to multicast when we see a concrete text proposal that addresses it.


Question: implementation level only -- they do not influence the result of the evaluation.) <del> A function argument is a "filter-path" or a "comparable". </del> <ins> A function argument is a "filter-query" or a "comparable". </ins> According to filter-selector, a "function-expr" is valid as a <del> "filter-path" or a "comparable". </del> <ins> "filter-query" or a "comparable". </ins> Any function expressions in a query must be well-formed (by conforming to the above ABNF) and well-typed, otherwise the JSONPath


Question: packets arrive.  If a peer could timeout within an Probe Timeout (PTO, see Section 6.2.2 of QUIC-RECOVERY), it is advisable to test for liveness before sending any data that cannot be retried safely. <ins> Note that it is likely that only applications or application protocols will know what information can be retried. </ins> 10.3.

What data can and cannot be retried safely?
That's not something that QUIC knows. HTTP does though. As advice, I don't think that this needs much more expansion, but I'd be happy to take a PR.
I'm not so sure this is a HTTP-only issue. Unlike TLS 1.3, there is QUIC data being sent under encryption, and that might have side effects.
LGTM, one nit. ("using applications" is fine, it's just a bit awkward.)


Question: simplifications are available.  Instead of being configured with (or discovering) the service registration domain, the (proposed) special- use domain name (see RFC6761) "default.service.arpa" is used.  The <del> details of how SRP server(s) are discovered will be specific to the constrained network, and therefore we do not suggest a specific </del> <ins> details of how SRP registrar(s) are discovered will be specific to the constrained network, and therefore we do not suggest a specific </ins> mechanism here. <del> SRP clients on constrained networks are expected to receive from the network a list of SRP servers with which to register.  It is the responsibility of a Constrained-Node Network supporting SRP to provide one or more SRP server addresses.  It is the responsibility of the SRP server supporting a Constrained-Node Network to handle the updates appropriately.  In some network environments, updates may be accepted directly into a local "default.service.arpa" zone, which has only local visibility.  In other network environments, updates for names ending in "default.service.arpa" may be rewritten internally to names with broader visibility. </del> <ins> SRP requestors on constrained networks are expected to receive from the network a list of SRP registrars with which to register.  It is the responsibility of a Constrained-Node Network supporting SRP to provide one or more SRP registrar addresses.  It is the responsibility of the SRP registrar supporting a Constrained-Node Network to handle the updates appropriately.  In some network environments, updates may be accepted directly into a local "default.service.arpa" zone, which has only local visibility.  In other network environments, updates for names ending in "default.service.arpa" may be rewritten internally to names with broader visibility. </ins> 2.1.3.

This is part of the previous pull request, but I missed it because it was interleaved with some unrelated changes.


Question: acknowledgement is received that establishes loss or delivery of packets. <del> When an ACK frame is received that establishes loss after a threshold number of consecutive PTOs (pto_count is more than kPersistentCongestionThreshold, see cc-consts-of-interest), the network is considered to be experiencing persistent congestion, and the sender's congestion window MUST be reduced to the minimum congestion window (kMinimumWindow).  This response of collapsing the congestion window on persistent congestion is functionally similar to a sender's response on a Retransmission Timeout (RTO) in TCP RFC5681. </del> <ins> When an ACK frame is received that establishes loss of all in-flight packets sent prior to a threshold number of consecutive PTOs (pto_count is more than kPersistentCongestionThreshold, see cc- consts-of-interest), the network is considered to be experiencing persistent congestion, and the sender's congestion window MUST be reduced to the minimum congestion window (kMinimumWindow).  This response of collapsing the congestion window on persistent congestion is functionally similar to a sender's response on a Retransmission Timeout (RTO) in TCP RFC5681. </ins> 7.6.

Follow-up from
Consider the following sequence of packets being sent In our old loss recovery code (before unifying TLP and RTO, ), we used the following logic for determining if an RTO was not spurious, which would then lead to collapse of the congestion window: In the example above would be packet 14. This means that to count as a verified RTO, an ACK must acknowledge 15 or 16, but not any packets smaller or equal than 14. With , we now collapse the congestion window when is larger than 2 and we receive an ACK which leads us to declare any packet lost. In the example above this means any ACK received after sending packet 15 which declares a packet lost would suffice. Even worse, this logic already triggers if we receive the ACK immediately after sending 15, i.e. when the peer didn't even have a chance to receive it. This seems a lot more aggressive than what we had before before. Is this intentional?
This is unintentional. I belIeve Jana is intending to fix this in PR , so please take a look at that.
If I understand correctly, is a fix for and only makes the description of the constant and the pseudo code consistent.
I believe the text is now correct, but the pseudocode is still incorrect in the way you describe. I think we should introduce a largestsentbefore_congestion variable as a replacement, but I'm open to other solutions.
I'm removing design. I consider this a bug fix, but still very important.
See which I believe fixes the text.
I've been using to mark things that will result in changes that affect implementations. This would seem to fit that description. Whether you consider it worth discussing is different. If it isn't worth discussing, maybe it will just be fixed by and everything will be fine without discussion in Tokyo. WIthout the label, this won't make the change log, and that might be a problem.
Fair point, I think fixes the text, but I'll re-label this as design to ensure it's included in the changelogs.
The spurious RTO detection calls an RTO spurious if any unacked packets sent prior to the RTO are acked after the RTO.


Question: acknowledgement is received that establishes loss or delivery of packets. <del> When an ACK frame is received that establishes loss after a threshold number of consecutive PTOs (pto_count is more than kPersistentCongestionThreshold, see cc-consts-of-interest), the network is considered to be experiencing persistent congestion, and the sender's congestion window MUST be reduced to the minimum congestion window (kMinimumWindow).  This response of collapsing the congestion window on persistent congestion is functionally similar to a sender's response on a Retransmission Timeout (RTO) in TCP RFC5681. </del> <ins> When an ACK frame is received that establishes loss of all in-flight packets sent prior to a threshold number of consecutive PTOs (pto_count is more than kPersistentCongestionThreshold, see cc- consts-of-interest), the network is considered to be experiencing persistent congestion, and the sender's congestion window MUST be reduced to the minimum congestion window (kMinimumWindow).  This response of collapsing the congestion window on persistent congestion is functionally similar to a sender's response on a Retransmission Timeout (RTO) in TCP RFC5681. </ins> 7.6.

The implication here is that any packet that was received but not acked can cause the RTO to be labeled spurious, even if there were more packets that were in fact lost. This is not always right. Consider the following example: Packets 5, 6, 7 are sent Packet 5 is received, but 6 and 7 are dropped Ack of 5 is dropped TLPs (packets 8 and 9) are sent and dropped RTO (packet 10) is sent and received ACK acks 10 and 5 Since 5 was sent prior to RTO, RTO is marked spurious. A crisp definition of spurious RTO may be useful to continue. Here's what I think is sensible: A spurious RTO is an unnecessary RTO. If the RTO timer had not gone off, things would have been fine. Based on this definition, clearly the RTO was necessary in the example, and therefore not spurious. We need to change the mechanism for detecting spurious RTOs, and I'll propose the following. An RTO is spurious if an ack is received after the RTO fires that acks only packets sent prior to the RTO. In other words, RTO is spurious if ack is received after the RTO, and (instead of , as the draft currently states).
This was recently changed because Lars ran into an issue() where the RTO'd packet was acknowledged for the first time in the same ACK as the pre-RTO packets. In that case, I think it was due to an overly aggressive MinRTO and some pretty coarse grained timers, but I think we want to declare that RTO spurious, since everything would have been fine if it hadn't been sent?
NAME : does not change the semantics substantively of what was in the draft. The intent was always to declare spurious RTO if (packet prior to RTO was newly acked). I'm proposing that we change that to (no packet sent after RTO was newly acked).
In light of , I think this is OBE?
Yes, this is OBE. Closing.


Question: malformed requests are handled as described in pa-error-common- aborts.)  It first looks for the PA parameters "PAParam" for which "PAAggregateReq.task_id" is equal to the task id derived from <del> "PAParam".  Next, it looks up the HPKE config and corresponding secret key associated with "PAAggregateReq.helper_hpke_config_id". If not found, then it aborts and alerts the leader with "unrecognized key config".  [NOTE: In this situation, the leader has no choice but to abort.  This falls into the class of error scenarios that are addressable by running with multiple helpers.] [TODO: Don't require all sub-requests to pertain to the same HPKE config.] </del> <ins> "PAParam". </ins> The response consists of the helper's updated state and a sequence of _sub-responses_, where the i-th sub-response corresponds to the sub-

Update the aggregate request text to reflect recent protocol changes. Relax the requirement that all sub-requests use the same key config. Do URL, which was previously merged into the wrong branch.
NAME NAME please have a look when you can.
Per , the first step in the upload protocol is a POST to the leader server which returns information about helpers. In some yet to be specified cases, the response could include a challenge to be incorporated into reports. However I suspect that in most cases, the response will be static and so an idempotent, cacheable GET would be a better fit, especially since it would save clients a roundtrip to the helper on the hot path of submitting reports, and would make it possible for helpers to implement uploadstart by putting some static content in a CDN. HTTP GET requests don't have bodies, so we would have to figure out how to encode all the data in the current into the URI, which could be done as either path fragments in the URI (i.e., ) or as query parameters (i.e., ). We would of course need to make sure that don't preclude proof systems like the one alleged to exist in section 5.2 of "BBCp19".
Also, because the current protocol uses a POST for uploadstart, the implication is that for every report they submit, clients need to hit and then . If we make idempotent and cacheable, then that request can be skipped in the majority of cases. I suppose the significance of this depends on how we expect clients to use PDA: are they going to be continuously streaming hundreds or thousands of reports into the leader as events of interest occur (in which case I think eliminating requests is interesting) or are they expected to accumulate several reports locally before submitting them all in one or a few requests?
I agree this would be a good change, if we can swing it. It's notable that Prio v2 and HH don't require a random challenge from the leader. But as discussed, this is useful for other proof systems one might want to implement for Prio. For example, see the data type implemented here: URL See below for an explanation. There might be other ways to implement the challenge. One possibility is to derive the challenge from the TLS key schedule (or from whatever state the client and leader share for the secure channel).


Question: malformed requests are handled as described in pa-error-common- aborts.)  It first looks for the PA parameters "PAParam" for which "PAAggregateReq.task_id" is equal to the task id derived from <del> "PAParam".  Next, it looks up the HPKE config and corresponding secret key associated with "PAAggregateReq.helper_hpke_config_id". If not found, then it aborts and alerts the leader with "unrecognized key config".  [NOTE: In this situation, the leader has no choice but to abort.  This falls into the class of error scenarios that are addressable by running with multiple helpers.] [TODO: Don't require all sub-requests to pertain to the same HPKE config.] </del> <ins> "PAParam". </ins> The response consists of the helper's updated state and a sequence of _sub-responses_, where the i-th sub-response corresponds to the sub-

NAME and NAME have both pointed out that adding this dependency on the underlying transport is problematic. Another possibility might be to use some external source of randomness that the leader trusts. comes to mind, though it is somewhat problematic because clients would effectively use the same randomness for some period of time. The security model of BBCp19 would need to be extended to account for this. (I believe this could be done.) Some other solution I'm missing? _ This data type is used to compute the mean and variance of each integer in a sequence of integers. To do so, the client encodes each integer of its input vector as a pair , where is the representation of as a bit vector and is equal to . To prove validity of its input, the proof needs to do two things: verify that (a) is indeed a bit vector, and (b) the integer encoded by is the square root of . In order to check these things efficiently, the proof system uses a random challenge generated by the leader. At a very high level, the client wants to prove that the following boolean conjunction holds: is 1 or 0 AND is 1 or 0 AND ... AND encodes the square root of . In Prio, we need to represent this conjunction as an arithmetic circuit, which the random challenge helps us do. In a bit more detail now: Let's rewrite this expression as w[0] AND w[1] AND ... AND w[s]. As an arithmetic circuit, this expression is actually , where if and only if w[i] is true. If the input is valid, then the expression should evaluate to . But because we're talking about field elements, a malicious client may try to play games like setting and so that the sum is still , but the statement is false. To detect this, we actually compute the following expression: , where is a randomly chosen field element. If the statement is true, then the expression evaluates to . If the statement is false, then with high probability, the expression will evaluate to something other than . (This idea comes from Theorem 5.3)
Thank you for the detailed explanalation NAME Having thought about it more, I suspect an HTTP GET wouldn't be the right way to support proof systems that need a challenge.


Question: malformed requests are handled as described in pa-error-common- aborts.)  It first looks for the PA parameters "PAParam" for which "PAAggregateReq.task_id" is equal to the task id derived from <del> "PAParam".  Next, it looks up the HPKE config and corresponding secret key associated with "PAAggregateReq.helper_hpke_config_id". If not found, then it aborts and alerts the leader with "unrecognized key config".  [NOTE: In this situation, the leader has no choice but to abort.  This falls into the class of error scenarios that are addressable by running with multiple helpers.] [TODO: Don't require all sub-requests to pertain to the same HPKE config.] </del> <ins> "PAParam". </ins> The response consists of the helper's updated state and a sequence of _sub-responses_, where the i-th sub-response corresponds to the sub-

Assuming that a unique challenge is needed for every report[^1], then intuitively the leader would have to maintain a mapping of reports to issued challenges so that the leader and helper(s) can later use the challenge value when evaluating the client's proof. But now that the leader has to maintain state per-report, I think the upload protocol is missing a notion of a report identifier. The contains a , but IIUC the will be the same across many reports[^2] (e.g., if a browser is sending daily reports on how often users click a button, they use the same every time they submit). So needs to include a report ID, which must be echoed back to the leader in so that the leader can look up the challenge for the report. [^1] Is this true? Or would it suffice for each to use the same challenge across multiple reports? [^2] I notice that the design doc lacks a strong definition of a report and how it relates to tasks, inputs, measurements: because GET should be "safe" in the sense of not changing any server state, and it seems like the server would have to maintain a mapping of reports to issued challenges so that the leader and helper can later use the challenge value when evaluating the client's proof.
Continuing on the assumption that causes a state change on the leader, POST is then the appropriate HTTP method to use. , though it's a little unusual. There may be some big drawback to cacheable POST responses, but I think that might let us support proof systems that require live challenges while also eliminating the majority of requests when the proof system is such that the will change very rarely.
The requirement for the leader to keep state across upload start and upload finish may be avoidable. Suppose the leader has a long-lived PRF key K. It would could choose a random nonce N, compute R = PRF(K, N) as the challenge, and hand (N, R) to the client in its upload start response. The client would then use R to generate the proof and send N to the leader in its upload finish request. The leader could then re-derive R. NAME and I kicked around the idea of having the client generate a report ID. I think this may end up being useful, but we weren't sure if it was better to have the leader choose it or the client.