jiebi/CodeConvo huggingface dataset:
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17.7. <ins> The "EncryptWithLabel" function defined in public-key-encryption avoids the risk of confusion between ciphertexts produced for different purposes in different contexts.  Each context is assigned a distinct label that is incorporated into the signature.  This registry records the labels defined in this document, and allows additional labels to be registered in case extensions add other types of public-key encryption using the same HPKE keys used elsewhere in MLS. Template: Label: The string to be used as the "Label" parameter to "EncryptWithLabel" Recommended: Same as in Reference: The document where this credential is defined Initial contents: 17.8. </ins> The exporter function defined in exporters allows applications to derive key material from the MLS key schedule.  Like the TLS exporter RFC8446, the MLS exporter uses a label to distinguish between

Related discussion:
text: As discussed in the virtual interim 2022-12-16, it is possible that extensions might be tempted to re-use HPKE public keys created for MLS. Even within MLS, we have two different public-key encryption cases, Welcome and Commit (though these currently use different keys). Currently, for Commit encryption, the HPKE parameter is set to the group context. The content of is actually undefined for Welcome encryption (!) This PR makes the following changes: Define general and functions that populte the HPKE parameter with an encoded label and context Apply these to Commit and Welcome encryption: Commit: label="UpdatePathNode", context= Welcome: label="Welcome", context= Add an IANA registry of public-key encryption labels (as for signing) There is also a bit of reorganization of the Ciphersuites section to split it into three subsections.
Note that any previous analysis of MLS had to take this particular reuse into account. This can't be the case for any arbitrary extensions reusing the key. So if MLS uses an extension that reuses the key, none of the existing analysis applies, which is a much bigger issue. This said, this reuse did decrease security -- it weakened FS in that a joiner had to hold on to their HPKE leaf key, which also allowed to decrypt the init secret from the welcome message, long after joining.
I think this captures what we discussed during the interim. FWIW, I'm wondering if we didn't have a domain separation issue in the past when we only had one HPKE public key in KeyPackages and used it for both the Welcome subsequently for UpdatePath as well. Either way, I think this is a good defense-in-depth mechanism.
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QUIC permits the use of a generic code in place of a specific error code; see Section 11 of QUIC-TRANSPORT.  For TLS alerts, this includes replacing any alert with a generic alert, such as <del> handshake_failure (0x128 in QUIC).  Endpoints MAY use a generic error code to avoid possibly exposing confidential information. </del> <ins> handshake_failure (0x0128 in QUIC).  Endpoints MAY use a generic error code to avoid possibly exposing confidential information. </ins> 4.9.

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text: One possible resolution to ; all literals are zero-padded to full bytes.
IANA made the following observation while creating the H3 registries: While the IANA copy of the registry uses two digits throughout, the QUIC draft is similarly inconsistent about whether literal hex values have any leading zeros. Is it worth updating our hex literals to use full bytes (, , etc.) or synchonizing on some other format? We probably shouldn't synchronize on trimming, since IANA has already skewed the other way.
NAME did fix this or do we need to inform IANA also?
We should probably shoot IANA an e-mail to update the registries with leading zeroes where they hadn't already assumed them.
I'll let you do that. The only registries that are affected are in HTTP/3 and QPACK.
Fixed both in the registry and the documents.
Yeah, this is easy to understand. The tables must have been annoying.
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details about record padding can be found in Section 5.4 and Appendix E.3 of RFC 8446. <del> Finally, endpoints can use the CID to attach arbitrary metadata to each record they receive.  This may be used as a mechanism to communicate per-connection information to on-path observers.  There is no straightforward way to address this concern with CIDs that contain arbitrary values.  Implementations concerned about this aspects SHOULD refuse to use CIDs. </del> <ins> Finally, endpoints can use the CID to attach arbitrary per-connection metadata to each record they receive on a given connection.  This may be used as a mechanism to communicate per-connection information to on-path observers.  There is no straightforward way to address this concern with CIDs that contain arbitrary values.  Implementations concerned about this aspect SHOULD refuse to use CIDs. </ins> 9.

Related discussion:
text: This is supposed to all be editorial stuff, and I split a couple changes out into separate commits in case they're not actually desired.
Code change:
supported scheme but an invalid value then the server MUST return an error of type "invalidContact". <ins> If the server wishes to present the client with terms under which the ACME service is to be used, it MUST indicate the URL where such terms can be accessed in the "terms-of-service" subfield of the "meta" field in the directory object, and the server MUST reject new-account requests that do not have the "terms-of-service-agreed" set to "true".  Clients SHOULD NOT automatically agree to terms by default. Rather, they SHOULD require some user interaction for agreement to terms. </ins> The server creates an account and stores the public key used to verify the JWS (i.e., the "jwk" element of the JWS header) to authenticate future requests from the account.  The server returns this account object in a 201 (Created) response, with the account URL in a Location header field. <ins> 7.3.1. </ins> If the server already has an account registered with the provided account key, then it MUST return a response with a 200 (OK) status code and provide the URL of that account in the Location header field.  This allows a client that has an account key but not the <del> corresponding account URI to recover the account URL. </del> <ins> corresponding account URL to recover the account URL. </ins> <del> If the server wishes to present the client with terms under which the ACME service is to be used, it MUST indicate the URL where such terms can be accessed in the "terms-of-service" subfield of the "meta" field in the directory object, and the server MUST reject new-account requests that do not have the "terms-of-service-agreed" set to "true".  Clients SHOULD NOT automatically agree to terms by default. Rather, they SHOULD require some user interaction for agreement to terms. </del> <ins> If a client wishes to recover an existing account and does not want a non existing account to be created, then it SHOULD do so by sending a POST request with an empty update.  That is, it should send a JWS whose payload is an empty object ({}). 7.3.2. </ins> If the client wishes to update this information in the future, it sends a POST request with updated information to the account URL. The server MUST ignore any updates to "order" fields or any other <del> fields it does not recognize. </del> <ins> fields it does not recognize.  If the server accepts the update, it MUST return a response with a 200 (OK) status code and the resulting account object. </ins> For example, to update the contact information in the above account, the client could send the following request: <ins> 7.3.3. </ins> Servers SHOULD NOT respond to GET requests for account resources as these requests are not authenticated.  If a client wishes to query the server for information about its account (e.g., to examine the

Related discussion:
text: This PR is a first go at addressing the following points I raised on the mailing list.
This allows us to address the account URL recovery use case in without having special handling for empty payloads.
Code change:
The "jwk" and "kid" fields are mutually exclusive.  Servers MUST reject requests that contain both. <del> For new-reg requests, and for revoke-cert requests authenticated by certificate key, there MUST be a "jwk" field. </del> <ins> For new-account requests, and for revoke-cert requests authenticated by certificate key, there MUST be a "jwk" field. </ins> For all other requests, there MUST be a "kid" field.  This field must <del> contain the account URI received by POSTing to the new-reg resource. </del> <ins> contain the account URI received by POSTing to the new-account resource. </ins> Note that authentication via signed JWS request bodies implies that GET requests are not authenticated.  Servers MUST NOT respond to GET

Related discussion:
text: In the term "registration" was changed to "account". In and in some of the remaining references to "registration" were removed, but some 'new-reg' and references to "registration" remained. This PR removes, hopefully, the last references to "registration".
Code change:
behaviors, and that those behaviors would remain relatively stable in the short term. <del> 6.3. </del> <ins> 6.4. </ins> The QUIC protocol, developed from a proprietary protocol into an IETF standards-track protocol RFC9000, turns many of the statements made

Related discussion:
text: NAME - several things here. we probably don't need all of the descriptive text - the idea is that we'd say "assume roughly this protocol stack, and here's what's not included" if distinguishing between "transport protocol" and "media transport protocol" makes sense, that probably needs to be reflected elsewhere in the document (like, a lot of elsewheres) If this approach works, it could either be in the transport protocol section, or in the one you're thinking of for
I don't think there will be a lot of places. We need to pay attention to only the places where we are explicitly referring to the media transport protocol. The rest can stay the same IMO.
NAME - I couldn't reply to this inline, but I looked through the document (searching for "transport protocol", and you're right. I found one place in Section 4.1, and changed the description of RTP as a transport protocol to RTP as a media transport protocol. NAME - Section 4.1 is before where my explanation is now - if this PR makes sense to you, the explanation probably DOES belong much earlier in the doc, perhaps in .
I'm merging this, based on Ali's review.
We want a paragraph to say something about scope along the lines of: This document is for endpoints delivering streaming media traffic. This can include intermediaries like CDNs as well as content owners and end users, plus their device and player providers. We think this can help address a few issues from last call/iesg review, including: (more to be linked as we notice them)
Consider adding a subsection in the introduction about things that are out of scope, and (for extra credit), why they are out of scope.
from NAME 's review (URL) 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.)
Not sure if this is really about scope clarification? I think this section is intending to talk about usage as part of a transport protocol, but tunnels are sort of more like a path overlay issue or something (might affect mtu, extra hops with extra things that can go wrong, etc.).
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behaviors, and that those behaviors would remain relatively stable in the short term. <del> 6.3. </del> <ins> 6.4. </ins> The QUIC protocol, developed from a proprietary protocol into an IETF standards-track protocol RFC9000, turns many of the statements made

Related discussion:
text: NAME - I'm looking more closely at this issue, and I think text that I put in another (individual) draft may be useful here as well. I'll add it in this section (making the obvious substitutions), and I'll see what NAME and NAME think (after the holiday weekend), and then check back with you. If it works for people, the result will likely be clearer to many readers. 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.
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behaviors, and that those behaviors would remain relatively stable in the short term. <del> 6.3. </del> <ins> 6.4. </ins> The QUIC protocol, developed from a proprietary protocol into an IETF standards-track protocol RFC9000, turns many of the statements made

Related discussion:
text: It would be more useful to analyze the impact of Wifi and 4G/5G networks in one place. Some of this follows later in sections 3.2.1 and Yet, the split between these three sections is unclear. Having a discussion about today's Internet path characteristics and the dynamics at one place could be more useful. The text could also better distinguish between what matters to endpoints and considerations relevant only for middleboxes (such as passive monitors). Section 3.3 Path Requirements The statement "to find the bandwidth requirements for a router on the delivery path" may assume that the bottleneck on the path will be routers. Yet, on a path the bottlenecks could also be a link layer device (e.g., an Ethernet Switch, a Wifi access point, etc.). RFC 6077 (specifically Section 3.1.3) also explains that issue. A better wording may be "to find the bandwidth requirements for a delivery path". Section 3.6. Unpredictable Usage Profiles / Section 3.7. Extremely Unpredictable Usage Profiles I am not fully convinced by the distinction between "unpredictable" and "extremely unpredictable". To me, these two sections could be merged and maybe also be shortened. More specifically, Section 3.7 lists a lot of statistics from the past. What is IMHO a bit missing in Section 3.7 are actual operational considerations. For instance, are there any lessons learnt for the future? Note that Section 3.6 has a reasonable conclusion at the end. Section 4. Latency Considerations and Section 4.1. Ultra Low-Latency I am surprised by the definition "ultra low-latency (less than 1 second)", as well as some of the other numbers. For other real-time communication use cases, "ultra low-latency" would probably imply a latency requirement of the order of one millisecond. For instance, isochronous traffic for motion control in industrial networks may require a latency of 1 ms, and such "ultra-low latency" requirements are discussed elsewhere, e.g., in the DetNet WG. The terminology in this section should be better explained to readers dealing with networked systems with much harder latency requirements.
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behaviors, and that those behaviors would remain relatively stable in the short term. <del> 6.3. </del> <ins> 6.4. </ins> The QUIC protocol, developed from a proprietary protocol into an IETF standards-track protocol RFC9000, turns many of the statements made

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text: And a reference should be added for these definitions in the context of video streaming. Section 5.5.2 Head-of-Line Blocking If Head-of-Line Blocking is indeed a relevant operational problem, it would be useful to add a corresponding reference (e.g., with measurements). Section 5.5.3. Wide and Rapid Variation in Path Capacity The statement "As many end devices have moved to wireless connectivity for the final hop (Wi-Fi, 5G, or LTE), new problems in bandwidth detction have emerged from radio interference and signal strength effects." could be moved to earlier parts of the document. Quite a bit of new computers apparently only have Wifi connectivity and no Ethernet port, i.e., Wifi may just be their default. Also, wireless may not only be used on the last hop, for instance in meshed setups. This may make the problem even harder. Section 6. Evolution of Transport Protocols and Transport Protocol Behaviors I really wonder whether UDP vs. TCP vs. QUIC is actually the relevant distinction. What may actually matter are the transport services provided by the protocol stack (e.g., RFC 8095). I fully agree to a related comment in the INTDIR review. Section 6.1. UDP and Its Behavior UDP is also used for encrypted tunnels (OpenVPN, Wireguard, etc.). Does encrypted tunneling really have no operational impact on streaming apps other than circuit breakers? (I am thinking about 4K streaming over OpenVPN and the like.) Section 6.2. TCP and Its Behavior This text should IMHO be presented and be discussed in TCPM. Personally, I am not convinced that this document is a good place for discussing the long history of TCP congestion control. If the objective of the document is to provide a tutorial, IMHO it would be more useful to briefly explain the state-of-the-art in year 2022. Some further specific comments on TCP congestion control: It is surprising that RFC 5681 is not referenced.793bis has normative statements regarding congestion control, and most stacks are compatible to what is written in 793bis. Several operating systems use CUBIC by default and support standard Reno. Another general good reference for TCP standards is RFC 7414.Does ECN, DCTCP, etc.
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behaviors, and that those behaviors would remain relatively stable in the short term. <del> 6.3. </del> <ins> 6.4. </ins> The QUIC protocol, developed from a proprietary protocol into an IETF standards-track protocol RFC9000, turns many of the statements made

Related discussion:
text: really not matter at all, for instance, if a data center hosts video playout servers? Section 6.3. QUIC and Its Behavior As already noted, the discussion about head-of-line-blocking would really benefit from backing by a reference. Section 7. Streaming Encrypted Media As far as I know, streaming users sometimes use encrypted tunnels such as OpenVPN or WireGuard (or IPsec) to access video content. I may miss something, but it is unclear to me how that fits into the categories presented in Section
Code change:
TLS_AES_128_GCM_SHA256 and TLS_AES_256_GCM_SHA384 would yield two PSKs, one for HKDF-SHA256 and another for HKDF-SHA384.  In contrast, if TLS_AES_128_GCM_SHA256 and TLS_CHACHA20_POLY1305_SHA256 are <del> supported, only one derived key is necessary. </del> <ins> supported, only one derived key is necessary.  Each ciphersuite uniquely identifies the target KDF.  Future specifications that change the way the KDF is negotiated will need to update this specification to make clear how target KDFs are determined for the import process. </ins> EPSKs MAY be imported before the start of a connection if the target KDFs, protocols, and context string(s) are known a priori.  EPSKs MAY <del> also be imported for early data use if they are bound to protocol settings and configurations that would otherwise be required for early data with normal (ticket-based PSK) resumption.  Minimally, that means Application-Layer Protocol Negotiation RFC7301, QUIC transport parameters (if used for QUIC), and any other relevant parameters that are negotiated for early data MUST be provisioned alongside these EPSKs. </del> <ins> also be imported for early data use if they are bound to the protocol settings and configuration that are required for sending early data. Minimally, that means Application-Layer Protocol Negotiation value RFC7301, QUIC transport parameters (if used for QUIC), and any other relevant parameters that are negotiated for early data MUST be provisioned alongside these EPSKs. </ins> 4.2.

Related discussion:
text: I think it may be useful to call out that the "multiple clients" and "multiple servers" case can occur even when there are only two endpoints that know the key (that is, when they do not have predetermined client/server roles). struct { opaque clientmac; opaque server_mac; } Context; We may not need the whole 16k to hold a MAC address, especially since this is just an example. If an attacker then redirects a ClientHello intended for one node to a different node, the receiver will compute a different context string and the handshake will not complete. Similarly to the above, this could be "a different node (including the node that generated the ClientHello)".
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6.6. During the handshake, some packet protection keys might not be <del> available when a packet arrives.  In particular, Handshake and 0-RTT packets cannot be processed until the Initial packets arrive, and 1-RTT packets cannot be processed until the handshake completes. Endpoints MAY ignore the loss of Handshake, 0-RTT, and 1-RTT packets that might arrive before the peer has packet protection keys to process those packets. </del> <ins> available when a packet arrives and the receiver can choose to drop the packet.  In particular, Handshake and 0-RTT packets cannot be processed until the Initial packets arrive and 1-RTT packets cannot be processed until the handshake completes.  Endpoints MAY ignore the loss of Handshake, 0-RTT, and 1-RTT packets that might have arrived before the peer had packet protection keys to process those packets. Endpoints MUST NOT ignore the loss of packets that were sent after the earliest acknowledged packet in a given packet number space. </ins> 6.7.

Related discussion:
text: Currently, the "Ignoring Loss of Undecryptable Packets" section indicates that some loss can be ignored, but now that we have multiple packet number spaces, I think we can constrain the set of ignorable losses to those below the smallest acknowledged packet in a packet number space. Ignoring the loss of a subsequently sent packet was not the intent.
I might have to put in a lint for HTAB...