Question: is running out of resources.  In these cases, MPTCP should send the MP_FASTCLOSE.  This option is illustrated in tcpm_fastclose. <del> If Host A wants to force the closure of an MPTCP connection, the MPTCP Fast Close procedure is as follows: </del> <ins> If Host A wants to force the closure of an MPTCP connection, it has two different options: </ins> <del> Host A sends an ACK containing the MP_FASTCLOSE option on one subflow, containing the key of Host B as declared in the initial connection handshake.  On all the other subflows, Host A sends a regular TCP RST to close these subflows, and tears them down. Host A now enters FASTCLOSE_WAIT state. </del> <ins> Option A (ACK) : Host A sends an ACK containing the MP_FASTCLOSE option on one subflow, containing the key of Host B as declared in the initial connection handshake.  On all the other subflows, Host A sends a regular TCP RST to close these subflows, and tears them down.  Host A now enters FASTCLOSE_WAIT state. </ins> <del> Upon receipt of an MP_FASTCLOSE, containing the valid key, Host B answers on the same subflow with a TCP RST and tears down all subflows.  Host B can now close the whole MPTCP connection (it transitions directly to CLOSED state). </del> <ins> Option R (RST) : Host A sends a RST containing the MP_FASTCLOSE option on all subflows, containing the key of Host B as declared in the initial connection handshake.  Host A can tear the subflows and the connection down immediately. If a host receives a packet with a valid MP_FASTCLOSE option, it shall process it as follows : Upon receipt of an ACK with MP_FASTCLOSE, containing the valid key, Host B answers on the same subflow with a TCP RST and tears down all subflows.  Host B can now close the whole MPTCP connection (it transitions directly to CLOSED state). </ins> As soon as Host A has received the TCP RST on the remaining subflow, it can close this subflow and tear down the whole

The number of retransmissions SHOULD be limited to avoid this connection from being retained for a long time, but this limit is implementation specific. A RECOMMENDED number is 3. If no TCP RST is received in response, Host A SHOULD send a TCP RST with the MPFASTCLOSE option itself when it releases state in order to clear any remaining state at middleboxes. o Upon receipt of a RST with MPFASTCLOSE, containing the valid key, Host B tears down all subflows. Host B can now close the whole MPTCP connection (it transitions directly to CLOSED state). The burden of the MP_FASTCLOSE is on the host that decides to close the connection (typically the server, but not necessarily). The text above allows it to opt for either ACK or RST without adding complexity to the other side. Another point that might be discussed in RFC6824bis is how a host should react when there are all the subflows associates to a Multipath TCP connection are in the CLOSED state, but the connection is still alive. At this point, it is impossible to send or receive data over the connection. It should attempt to reestablish one subflow to keep the connection alive. Olivier


Question: which type of header is present.  The purpose of this bit is invariant across QUIC versions. <del> The long header exposes more information.  In version 1 of QUIC, it is used during connection establishment, including version negotiation, retry, and 0-RTT data.  It contains a version number, as well as source and destination connection IDs for grouping packets belonging to the same flow.  The definition and location of these fields in the QUIC long header are invariant for future versions of QUIC, although future versions of QUIC may provide additional fields in the long header QUIC-INVARIANTS. </del> <ins> The long header exposes more information.  It contains a version number, as well as source and destination connection IDs for associating packets with a QUIC connection.  The definition and location of these fields in the QUIC long header are invariant for future versions of QUIC, although future versions of QUIC may provide additional fields in the long header QUIC-INVARIANTS. In version 1 of QUIC, the long header is used during connection establishment to transmit crypto handshake data, perform version negotiation, retry, and send 0-RTT data. </ins> Short headers contain only an optional destination connection ID and the spin bit for RTT measurement.  In version 1 of QUIC, they are

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: 12.3.11. <del> Control Property [TODO: Discuss] Boolean Preconnection, Connection This property specifies whether an application considers it useful to be informed in case sent data was retransmitted more often than a certain threshold.  When set to true, the effect is twofold: The application may receive events in case excessive retransmissions.  In addition, the transport system considers this as a preference to use transports stacks that can provide this notification.  This is not a strict requirement.  If set to false, no notification of excessive retransmissions will be sent and this transport feature is ignored for protocol selection. The default is to have this option. </del> <ins> Boolean Connection Property - see conn-retrans-notify. </ins> 12.3.12. <del> Control Property [TODO: Discuss] Integer Preconnection, Connection This property specifies after how many retransmissions to inform the application about "Excessive Retransmissions". </del> <ins> Integer Connection Property - see </ins> 12.3.13. <del> Control Property [TODO: Discuss] Boolean Preconnection, Connection This property specifies whether an application considers it useful to be informed when an ICMP error message arrives that does not force termination of a connection.  When set to true, received ICMP errors will be available as SoftErrors.  Note that even if a protocol supporting this property is selected, not all ICMP errors will necessarily be delivered, so applications cannot rely on receiving them.  Setting this option also implies a preference to prefer transports stacks that can provide this notification.  If not set, no events will be sent for ICMP soft error message and this transport feature is ignored for protocol selection. This property applies to Connections and Connection Groups.  The default is not to have this option. </del> <ins> Boolean Connection Property - see </ins> 12.3.14.

This is the second step towards applying our new Properties Classification. It moves up the Connection Properties and annotates most places with stuff to do.
As discussed in
Thanks for moving this forward! I only found a few nits. Also, I have a question about whether two of these Properties should maybe be changed to be Selection Properties, but that's more of a discussion and not something that needs to be done in this URL modulo NAME commentsI carefully checked it, but I find it very hard to be sure that this is 100% ok (as is usually the case IMO with diff's about text blocks being moved around). My suggestion is to land it and see.


Question: SHA-224, SHA-256, SHA-384, and SHA-512 SHS, respectively.  The "none" value is provided for future extensibility, in case of a signature algorithm which does not require hashing before signing. <ins> The usage of MD5 and SHA-224 are deprecated.  The md5_RESERVED and sha224_RESERVED values MUST NOT be offered or negotiated by any implementation. </ins> This field indicates the signature algorithm that may be used.

Not sure if the deprecation cutoff will be at 224 bits (~112 bit security level) or 255 bits (~128 bit security level) yet, so I've just done the former for now. This drops the old curves under 224 bits, updates the extension example bytes, and deprecates MD5.
As noted on-list, SHA1 can't be as easily dealt with here, so that'll be a topic for a separate changeset.
Here's a PR for this: Question: Are we dropping all curves <224 bits or <255 bits? (was consensus ever called on this specific number?)
There was not a specific WG consensus call on this point. I hope we don't need one and that we should keep the curves in that are used and trim the ones that aren't.
Good. In that case, I'll slash everything other than secp256r1 (prime256v1 / NIST P-256), secp384r1 (NIST P-384), secp521r1 (NIST P-521), and sect571r1 (NIST B-571). (plus, of course, new CFRG curves yet to be added) PR updated. URL URL
Based on the fact that apparently precisely zero servers actually use SHA-224 for signatures, can we drop that too while we're at it? (not an exaggeration; the scan cited in the last comment shows none)
I would drop it. spt On Jun 17, 2015, at 20:09, Dave Garrett EMAIL wrote:
OK. PR updated.
NAME This can be closed now that PR has been merged.


Question: REQUIRED.  A unique access token for continuing the request, in the format specified in response-token-single.  This access token MUST be bound to the client instance's key used in the request and <del> MUST NOT be a "bearer" token.  As a consequence, the "bound" field of this access token is always the boolean value "true" and the </del> <ins> MUST NOT be a "bearer" token.  As a consequence, the "flags" array of this access token MUST NOT contain the string "bearer" and the </ins> "key" field MUST be omitted.  This access token MUST NOT be usable at resources outside of the AS.  The client instance MUST present the access token in all requests to the continuation URI as

URL match the flags array in the token request. While this is technically a breaking change, I think we should consider this editorial as this should have been changed when we made the original change to use a flags array in the request in .
:heavycheckmark: Deploy Preview for gnap-core-protocol-editors-draft ready! Built :hammer: Explore the source changes: 8c793bb446d82f053536358106485568fc457842 :mag: Inspect the deploy log: :sunglasses: Browse the preview:
You know, I had that thought, and I checked the request section and it doesn't have quotes. I'm going to add the quotes to both parts. This ended up with some other language changing, not just the examples, so let me know what you think.


Question: Service Registration Protocol for DNS-Based Service Discovery <del> draft-ietf-dnssd-srp-13 </del> <ins> draft-ietf-dnssd-srp-14 </ins> Abstract

Update the draft number, date, and change draft-sekar-ul to draft-ietf-dnssd-update-lease.


Question: If endpoint receives an unexpected value for the transport parameter "enable_multipath", it MUST treat this as a connection error of type <del> MP_CONNECTION_ERROR and close the connection. </del> <ins> TRANSPORT_PARAMETER_ERROR (specified in Section 20.1 of QUIC- TRANSPORT) and close the connection. </ins> This extension does not change the definition of any transport parameter defined in Section 18.2. of QUIC-TRANSPORT.

Fix issue
In section 3, we have However, the error code does not exist (anymore), and I'm not sure we need such a specific error code anyway. Why not raising a in such case?
It's reasonable to use TRANSPORTPARAMETERERROR. Submitted PR for this issue.


Question: Therefore it needs to respond to the request without waiting for a packet from the target.  However, if the target_host is a DNS name, the UDP proxy MUST perform DNS resolution before replying to the HTTP <del> request.  If errors occur during this process (for example, a DNS resolution failure), the UDP proxy MUST fail the request and SHOULD send details using an appropriate "Proxy-Status" header field PROXY- STATUS. </del> <ins> request.  If errors occur during this process, the UDP proxy MUST fail the request and SHOULD send details using an appropriate "Proxy- Status" header field PROXY-STATUS (for example, if DNS resolution returns an error, the proxy can use the dns_error Proxy Error Type from PROXY-STATUS). </ins> UDP proxies can use connected UDP sockets if their operating system supports them, as that allows the UDP proxy to rely on the kernel to

I also want to point an issue due to this stance. The fact that we are not explicit and discuss the expectations and realities of different IP versions we have interesting failure cases due to this text from Section 3.1: “However, if the target_host is a DNS name, the UDP proxy MUST perform DNS resolution before replying to the HTTP request.” No where in the follow paragraphs in this section do it discuss the slight issue that a DNS resolution can return only an AAAA record and the MASQUE server may only support IPv4. What is the error response? Should this be a proxy-status response. I would not there are currently no response for “IP version not supported”.
These concerns apply equally to TCP proxying via HTTP. We should strive to provide parity to how CONNECT treats these matters. HTTP semantics don't say anything
+1 to Lucas' meta-comment about striving for parity with CONNECT There is no such thing as a "unsupported IP version" DNS failure. Let's take an example where the client wants to CONNECT-UDP via the UDP proxy to the target . This particular host is configured such that querying AAAA will get a valid address, and querying A will yield a "no answer no error" DNS result. Let's assume that in this example the UDP proxy is IPv4-only. In that scenario, the UDP proxy will only query A (because it doesn't know what AAAA is) and it will get "no result no answer" so it will send
NAME in light of the previous comments, can you clarify what you think is missing from the document please?
First of all, I think that just answering lets do like Connect is a poor answer. There are benefits to alignment, but also one should ask if one are preventing useful functionality. In this case I think there are a relevant difference between what could be answered depending on how one implement stuff. Your suggest road says that for a IPv4 only proxy with a DNS name that has AAAA record but no A record the answer is: a) There is no answer For a node that would query both the answer could actually be: b) There is host, but I can't reach it as I don't support the right IP version, try another proxy. Sure, limiting ourselves to a) is easy, but is it the most useful for the protocol?
Disagree it's a poor answer.


Question: Therefore it needs to respond to the request without waiting for a packet from the target.  However, if the target_host is a DNS name, the UDP proxy MUST perform DNS resolution before replying to the HTTP <del> request.  If errors occur during this process (for example, a DNS resolution failure), the UDP proxy MUST fail the request and SHOULD send details using an appropriate "Proxy-Status" header field PROXY- STATUS. </del> <ins> request.  If errors occur during this process, the UDP proxy MUST fail the request and SHOULD send details using an appropriate "Proxy- Status" header field PROXY-STATUS (for example, if DNS resolution returns an error, the proxy can use the dns_error Proxy Error Type from PROXY-STATUS). </ins> UDP proxies can use connected UDP sockets if their operating system supports them, as that allows the UDP proxy to rely on the kernel to

An HTTP proxy that inplements CONNECT for UDP is likely going to implement CONNECT for TCP. How to treat name resolution for both of these has to be consistent. The Proxy-Status header field is the appropriate way to signal such failures. Any text more complicated than what we have now, should be pursued in a separate document IMO. Not added at this late stage. It would be straightforward to form up a document like "DNS considerations for HTTP proxying" that is additive to the specifications we have.
I agree: Proxy-Status is the right place for this discussion, and CONNECT-UDP already recommends Proxy-Status. Also, as a technical matter, this is not how dual-stack connectivity works. As noted above, an IPv4-only UDP client will never issue a AAAA query for a destination hostname.
I agree with what Lucas and Ben said above. Let's move this discussion to an extension draft.
I agree that this discussion probably deserves to be in an extension draft. As is stands today, a DNS failure as described in this thread can just as easily result in a destination-unreachable / connection-closed error for the CONNECT-UDP stream, and an extension to provide detailed information as to why seems like the natural path forward.
Proxy status is absolutely the right thing to do here. The proxy-status is what we are already doing in production for CONNECT and CONNECT-UDP. The current text seems entirely appropriate for this: The only thing I could see saying further is to mention the error parameter, like this: I do not think we need any extension draft for this. This is very straightforward, and doesn't need to say much.
Thanks Tommy, I think that's a useful clarification. I wrote up PR based on your proposal to make this clearer
Ok, I am fine with this resolution.
Thank you for confirming Magnus!


Question: [4] https://about.gitlab.com/ <del> [5] https://trustee.ietf.org/license-for-open-source- repositories.html [6] https://github.com/dontcallmedom/github-notify-ml </del> <ins> [5] https://github.com/dontcallmedom/github-notify-ml </ins>

I was tempted to remove the entire section on administrative policies, though left it as is. NAME what do you think?
Yeah, I figured it best to start small. Let me know if you need anything else before merging!
NAME Interesting that the circle CI build ran on the merge, but not the PR. I didn't see any build failures reported before it merged, but I just got the email reporting a missed reference. Maybe there are some tweaks you need to apply over there as owner of the org.
Yeah, I'll fix it. Thanks for letting me know!
I was thinking of being a little more aggressive about removing some of the text, with matching PRs for the other work, but this works.


Question: prioritization. Note that this property is not a per-message override of the <del> connection Priority - see conn-priority.  The Priority properties may </del> <ins> Connection Priority - see conn-priority.  The Priority properties may </ins> interact, but can be used independently and be realized by different mechanisms; see priority-in-taps.

This changes the use of the term "entanglement" as discussed in , and moves some text around in this section for better clarity and less repetition. Also, addressing , it introduces a framer as an optional parameter to the Clone call.
NAME I incorporated all of your suggestions, thanks!
This is not fully clear from section 5.4., however, if connections in a group are each QUIC streams, I think it should be possible to use different framers on each stream. So it might be okay that a framers is copied when clone is called but changing a framer on one connection should not change the framer on the other connection in a group...?
Also an related editorial comment (that could be addressed in the same PR eventually): this section on groups only at the very end has a paragraph starting with "Note that calling Clone() can result in on-the-wire signaling..." which kind of explains that groups are supposed to be used for multistream protocol but can also be mapped to separate connection if the protocol doesn't not support multiplexing of stream on one connection. I think this information should be added at the very beginning of this section.
Actually, now looking at the framer section, I guess you would rather like to already set the new framer before you clone a connection...?
I agree with everything you say here; about your last suggestion, maybe a framer should be an optional argument of the clone call?
I don’t actually understand the difference between a connection in a Connection Group and an “entangled connection” ? Both terms are used in API, but in different ways (possibly by different authors) - do we really really need the word “entangled”?? if we do we need, can we define and explain this.
I think that having the word is good as it indicates a deeper relationship than mere "grouping". Just based on terms, I'd (correctly) imagine that I can define priorities between members of a group, and that they somehow belong together... as in: Close with a common call, etc. ... but I don't think that I would expect, just from this term, that changing a property on one of them affects all the others. Hence the phrasing: "The connections within a group are 'entangled' with each other". However, I agree that in most other places, we should talk about "groups" instead.


Question: prioritization. Note that this property is not a per-message override of the <del> connection Priority - see conn-priority.  The Priority properties may </del> <ins> Connection Priority - see conn-priority.  The Priority properties may </ins> interact, but can be used independently and be realized by different mechanisms; see priority-in-taps.

I can try to fix this.
Sounds good - we should add that explanation upfront in the definitions then also?
[Yes, I think so.] EDIT: no, now I don't think so anymore. (see )
I have addressed this in PR . Note that the text interchangeably talked about both Connections and Connection Properties as being "entangled". I have now written this such that the term "entanglement" applies to Connection Properties only, whereas Connections in a group are, well, members of the same Connection Group.
Looks good to me.


Question: an HTTP Datagram.  This prevents infinite loops in the presence of routing loops, and matches the choices in IPsec IPSEC. <ins> IPv6 requires that every link have an MTU of at least 1280 bytes IPv6.  Since CONNECT-IP conveys IP packets in HTTP Datagrams and those can in turn be sent in QUIC DATAGRAM frames which cannot be fragmented DGRAM, the MTU of a CONNECT-IP link can be limited by the MTU of the QUIC connection that CONNECT-IP is operating over.  This can lead to situations where the IPv6 minimum link MTU is violated. CONNECT-IP endpoints that support IPv6 MUST ensure that the CONNECT- IP tunnel link MTU is at least 1280 (i.e., that they can send HTTP Datagrams with payloads of at least 1280 bytes).  This can be accomplished using various techniques: if HTTP intermediaries are not in use, both CONNECT-IP endpoints can pad the QUIC INITIAL packets of the underlying QUIC connection that CONNECT-IP is running over. if HTTP intermediaries are in use, CONNECT-IP endpoints can enter in an out of band agreement with the intermediaries to ensure that endpoints and intermediaries pad QUIC INITIAL packets. CONNECT-IP endpoints can also send ICMPv6 echo requests with 1232 bytes of data to ascertain the link MTU and tear down the tunnel if they do not receive a response.  Unless endpoints have an out of band means of guaranteeing that one of the two previous techniques is sufficient, they MUST use this method. </ins> Endpoints MAY implement additional filtering policies on the IP packets they forward.

I think there are two option here, 1) close/deny the connect request, and 2) use streams instead. I think we should consider both option, especially given stream are just there to use.
Isn't this a dup of ? That's fixed with . I guess the text could be generalized to talk about more than just IPv6 MTU issues.
No, this aspect on what you do when the MTU required is not present is not resolved by that PR. I would note that closing gives the endpoints one experience that is likely easier to handle in implementation and do avoid the issue with reordering and the varying packet treatment that moving to stream would result.
I would suggest adding a sentence saying that if an endpoint realizes that the IPv6 tunnel MTU goes lower than 1280, then it MUST close the stream.
IETF 114: Note that if something happens and your MTU gets smaller, you may need to destroy that link. Potentially make this option a SHOULD do PMTUD, otherwise you could end up in a situation where you're not a compliant IPv6 tunnel. Proposal:
Even if PR was an improvement to the description of the IPv6 minimal link MTU issue the text still leaves an open issue in that it doesn't make clear which of the endpoints that are responsible for HTTP intermediaries or if is truly both which has another implication. So a Connect-IP endpoint will be aware of the HTTP datagram supported MTU and if its HTTP/3 connection can support sending at least 1280 bytes of IPv6 packets without fragmentation. However, if there is two or more HTTP intermediaries in between the client and the server then there are HTTP intermediaries that might have an HTTP/3 connection which has less than 1280 bytes MTU for HTTP datagram payloads in-between the client and server. Thus, arises the issue of how to detect this. PR states: if HTTP intermediaries are in use, CONNECT-IP endpoints can enter in an out of band agreement with the intermediaries to ensure that endpoints and intermediaries pad QUIC INITIAL packets. From my perspective the big question is how a client or a server knows that there are intermediaries, or the most important aspect, that there are no unknown HTTP intermediaries. To my understanding both the client and the server can potentially bring in HTTP intermediaries unknown to the other part.


Question: an HTTP Datagram.  This prevents infinite loops in the presence of routing loops, and matches the choices in IPsec IPSEC. <ins> IPv6 requires that every link have an MTU of at least 1280 bytes IPv6.  Since CONNECT-IP conveys IP packets in HTTP Datagrams and those can in turn be sent in QUIC DATAGRAM frames which cannot be fragmented DGRAM, the MTU of a CONNECT-IP link can be limited by the MTU of the QUIC connection that CONNECT-IP is operating over.  This can lead to situations where the IPv6 minimum link MTU is violated. CONNECT-IP endpoints that support IPv6 MUST ensure that the CONNECT- IP tunnel link MTU is at least 1280 (i.e., that they can send HTTP Datagrams with payloads of at least 1280 bytes).  This can be accomplished using various techniques: if HTTP intermediaries are not in use, both CONNECT-IP endpoints can pad the QUIC INITIAL packets of the underlying QUIC connection that CONNECT-IP is running over. if HTTP intermediaries are in use, CONNECT-IP endpoints can enter in an out of band agreement with the intermediaries to ensure that endpoints and intermediaries pad QUIC INITIAL packets. CONNECT-IP endpoints can also send ICMPv6 echo requests with 1232 bytes of data to ascertain the link MTU and tear down the tunnel if they do not receive a response.  Unless endpoints have an out of band means of guaranteeing that one of the two previous techniques is sufficient, they MUST use this method. </ins> Endpoints MAY implement additional filtering policies on the IP packets they forward.

Thus resulting in a HTTP intermediary to intermediary connection where neither endpoint knows if the requirement is fulfilled. Thus my interpretation is that the only way to be sure that it works it to actually execute on the third bullet in the PR: CONNECT-IP endpoints can also send ICMPv6 echo requests with 1232 bytes of data to ascertain the link MTU and tear down the tunnel if they do not receive a response. Unless endpoints have an out of band means of guaranteeing that one of the two previous techniques is sufficient, they MUST use this method. I would also note that this only verifies from the acting endpoint towards its peer, not the reverse path. For that to be verified the peer need to send ICMP probes the other direction. Thus, my conclusion is that Connect-IP still don't ensure that the minimal MTU is fulfilled.
Isn't the quick-fix to require that the ICMP reply also be 1232 byte sized?
Normally I would say now, as you want to ensure that the reply makes it back that shouldn't be padded. Also if the response is anything else than echo reply you also should not pad. However, I think the implication is the same either way. Both masque client and server need to send ICMP or modify its ICMP processing. Given that I would think having each endpoint doing independent processing of its forward path might actually be simpler and enables using a standard off the shelf ICMP implementation and not have to modify it.
Our implementation has userspace ICMP request/response handling for echo request/response, it wasn't too hard to write. I don't propose modifying anything, but the structures were ~50 lines of code to set up. I agree we shouldn't be modifying responses we did not generate.
I think the existing text already is generic to "endpoint", not client or server: So this implies that both sides need to send the PING if they haven't guaranteed the MTU. This seems like this is already handled?
If both endpoints are required to send ping it would be resolved. However, that is not what is written in the draft currently. What is written is the there are various techniques, and if an endpoint choses something else than verifying its forward path using ping, it becomes unclear who is actually responsible for any HTTP intermediaries. Which results in an unclear requirement and very likely failure of the MTU verification resulting in an MTU black hole.


Question: an HTTP Datagram.  This prevents infinite loops in the presence of routing loops, and matches the choices in IPsec IPSEC. <ins> IPv6 requires that every link have an MTU of at least 1280 bytes IPv6.  Since CONNECT-IP conveys IP packets in HTTP Datagrams and those can in turn be sent in QUIC DATAGRAM frames which cannot be fragmented DGRAM, the MTU of a CONNECT-IP link can be limited by the MTU of the QUIC connection that CONNECT-IP is operating over.  This can lead to situations where the IPv6 minimum link MTU is violated. CONNECT-IP endpoints that support IPv6 MUST ensure that the CONNECT- IP tunnel link MTU is at least 1280 (i.e., that they can send HTTP Datagrams with payloads of at least 1280 bytes).  This can be accomplished using various techniques: if HTTP intermediaries are not in use, both CONNECT-IP endpoints can pad the QUIC INITIAL packets of the underlying QUIC connection that CONNECT-IP is running over. if HTTP intermediaries are in use, CONNECT-IP endpoints can enter in an out of band agreement with the intermediaries to ensure that endpoints and intermediaries pad QUIC INITIAL packets. CONNECT-IP endpoints can also send ICMPv6 echo requests with 1232 bytes of data to ascertain the link MTU and tear down the tunnel if they do not receive a response.  Unless endpoints have an out of band means of guaranteeing that one of the two previous techniques is sufficient, they MUST use this method. </ins> Endpoints MAY implement additional filtering policies on the IP packets they forward.

Well, it does say that endpoints MUST use this method unless the already have other coordination. I could see some clarification text to say that the guarantee for the other methods must include knowing there isn't an intermediary — which is actually pretty easy to guarantee in many cases.
That is what I intended when writing , but perhaps I wasn't clear. Can one of you propose a PR that would clarify this?
As per , the IPv6 minimum link MTU is 1280 bytes. However, states that it is possible to have a functional QUIC connection on a path that only allows 1200 bytes of UDP payloads. Such a path would only allow HTTP Datagrams with payloads up to 1180 bytes. If we only send IPv6 packets over QUIC DATAGRAM frames, such a setup would result in violating the IPv6 minimum link MTU. We should describe this problem and how to react to it. Possible solutions include failing the CONNECT-IP request, using the DATAGRAM capsule, or different solutions altogether.
This is only a problem for the end-2-end QUIC handshake packets, right? I was assuming that you either have a QUIC tunnel that has a larger MTU in order to use datagrams or you have to use stream-based transmission for the e2e handshake and then need a way to switch to datagram later. In any case we need to say more in the draft!
From the most recent session, I think we may have consensus to address this by noting that CONNECT-IP over H3 requires a minimum link MTU of X bytes on every HTTP hop for IPv6 compliance, leaving support for smaller links to an extension.
I'm not sure the consensus was clear, but I think it's upon us to make a proposal and see if it flies =)
That sounds like the best path forward.
I propose a simple solution that should be enough for most use-cases: if any endpoint detects that the path MTU is too low, it MUST close the tunnel if the tunnel had IPv6 addresses or routes on it. Endpoints are not required to measure the path MTU. endpoints that wish to use IPv6 SHOULD ensure that their path MTU is sufficient. For example, that can be guaranteed by padding QUIC initial packets to 1320 bytes.
That seems reasonable to me as a proposal!


Question: an HTTP Datagram.  This prevents infinite loops in the presence of routing loops, and matches the choices in IPsec IPSEC. <ins> IPv6 requires that every link have an MTU of at least 1280 bytes IPv6.  Since CONNECT-IP conveys IP packets in HTTP Datagrams and those can in turn be sent in QUIC DATAGRAM frames which cannot be fragmented DGRAM, the MTU of a CONNECT-IP link can be limited by the MTU of the QUIC connection that CONNECT-IP is operating over.  This can lead to situations where the IPv6 minimum link MTU is violated. CONNECT-IP endpoints that support IPv6 MUST ensure that the CONNECT- IP tunnel link MTU is at least 1280 (i.e., that they can send HTTP Datagrams with payloads of at least 1280 bytes).  This can be accomplished using various techniques: if HTTP intermediaries are not in use, both CONNECT-IP endpoints can pad the QUIC INITIAL packets of the underlying QUIC connection that CONNECT-IP is running over. if HTTP intermediaries are in use, CONNECT-IP endpoints can enter in an out of band agreement with the intermediaries to ensure that endpoints and intermediaries pad QUIC INITIAL packets. CONNECT-IP endpoints can also send ICMPv6 echo requests with 1232 bytes of data to ascertain the link MTU and tear down the tunnel if they do not receive a response.  Unless endpoints have an out of band means of guaranteeing that one of the two previous techniques is sufficient, they MUST use this method. </ins> Endpoints MAY implement additional filtering policies on the IP packets they forward.

So how does the endpoint determine that the Path MTU is too low? In the scenario that looks like this C - I1 -I2 - S and it is the I1-I2 path that can't support 1280 byte datagrams. The I nodes are HTTP intermediaries. The client will not know that the datagram MTU size is too low on these nodes, to my understanding the datagram payloads will simply be dropped by the I nodes in either direction.
In that scenario, if the endpoint wants to check it can send an ICMPv6 echo request with 1232 bytes of data - if it receives an ICMPv6 echo reply that means that the path supports the minimum IPv6 MTU.
NAME then we are making it mandatory to send ICMPv6 echo to the external address assigned for the client, and for the server sending one to the client, this to determine that minimal 1280 bytes are working over MASQUE. Because to me there are no way the MASQUE endpoint can determine in all cases if there are HTTP intermediaries or not. Else we have to put requirements on the HTTP intermediaries to understand MASQUE, and to my understanding what was not desired. And maybe using ICMPv6 is the easiest method with the lowest implementation burden. The issue I see with this aspect is that many applications when establishing a Connect-IP path might not want to wait for the ICMPv6 response before sending other traffic so we have a situation where there is a small, but non-zero risk that these packets goes into a black hole, and then the ICMP response comes back and the MASQUE Connect-IP stream will be closed due to the response to indicate the failure. I think this is all manageable and not different from failures that can occur for other reasons.
I don't see why this should be mandatory. When I plug in an Ethernet cable into my computer, I don't know whether the IP stack on the next router spuriously drops packets above 1000 bytes long, and my OS has no requirement to measure the path MTU.
But for the said router dropping IPv6 packets below 1280 would be a protocol violation. The issue here is the interaction with the HTTP intermediaries and what is under the control of Connect-IP function. So Connect-IP basically provides a IPv6 capable link.


Question: an HTTP Datagram.  This prevents infinite loops in the presence of routing loops, and matches the choices in IPsec IPSEC. <ins> IPv6 requires that every link have an MTU of at least 1280 bytes IPv6.  Since CONNECT-IP conveys IP packets in HTTP Datagrams and those can in turn be sent in QUIC DATAGRAM frames which cannot be fragmented DGRAM, the MTU of a CONNECT-IP link can be limited by the MTU of the QUIC connection that CONNECT-IP is operating over.  This can lead to situations where the IPv6 minimum link MTU is violated. CONNECT-IP endpoints that support IPv6 MUST ensure that the CONNECT- IP tunnel link MTU is at least 1280 (i.e., that they can send HTTP Datagrams with payloads of at least 1280 bytes).  This can be accomplished using various techniques: if HTTP intermediaries are not in use, both CONNECT-IP endpoints can pad the QUIC INITIAL packets of the underlying QUIC connection that CONNECT-IP is running over. if HTTP intermediaries are in use, CONNECT-IP endpoints can enter in an out of band agreement with the intermediaries to ensure that endpoints and intermediaries pad QUIC INITIAL packets. CONNECT-IP endpoints can also send ICMPv6 echo requests with 1232 bytes of data to ascertain the link MTU and tear down the tunnel if they do not receive a response.  Unless endpoints have an out of band means of guaranteeing that one of the two previous techniques is sufficient, they MUST use this method. </ins> Endpoints MAY implement additional filtering policies on the IP packets they forward.

Per IPv6 RFC that requires us to support 1280 at least. What have been the rough consensus so far is that we are not defining an fragmentation and reassembly mechanism, or rather how to ensure that one uses QUIC streams when the transport is HTTP/3 for below 1280 IPv6 packets and instead reject the service. Thus we are in the position of having to answer how does one know when to reject the service. The alternatives I see are: 1) Force the Connect-IP endpoints to ICMP ping each other to verify that the path supports 1280 IPv6 packets 2) Have the HTTP intermediaries be Connect-IP aware to the level that they can reject the HTTP request if the path does not support 1280 packets. Where 2) have the downside that we still likely have some black holes in cases where the HTTP intermediaries are not connect-IP aware and ignores the requirements. Where 1) have the downside that the endpoints request can't be rejected during the request response sequence and instead have to be closed with an error message indicating the paths failure to support the packet.
I'm opposed to requiring ICMP pings because the majority of use-cases will not have this problem. How about we add text like this: CONNECT-IP endpoints that enable IPv6 connectivity in the tunnel MUST ensure that the tunnel MTU is at least 1280 bytes to meet the IPv6 minimum required MTU. This can be accomplished multiple different ways, including: Endpoints padding QUIC initial packets when it is known that there are no intermediaries present Endpoints sending ICMPv6 echo requests with 1232 bytes of data an out of band agreement between one of the endpoints and any involved intermediaries to have them pad QUIC initial packets
So I understand your desire to not add complexity anywhere for a likely seldom arising case. However, I do fear that not addressing this with a clear assignment on whose responsibility it is to avoid the issue will result in that none does anything of it. Because an endpoint that uses HTTP/3 can verify its first hop. However, the complexities arise when you have HTTP intermediaries in the mix.


Question: an HTTP Datagram.  This prevents infinite loops in the presence of routing loops, and matches the choices in IPsec IPSEC. <ins> IPv6 requires that every link have an MTU of at least 1280 bytes IPv6.  Since CONNECT-IP conveys IP packets in HTTP Datagrams and those can in turn be sent in QUIC DATAGRAM frames which cannot be fragmented DGRAM, the MTU of a CONNECT-IP link can be limited by the MTU of the QUIC connection that CONNECT-IP is operating over.  This can lead to situations where the IPv6 minimum link MTU is violated. CONNECT-IP endpoints that support IPv6 MUST ensure that the CONNECT- IP tunnel link MTU is at least 1280 (i.e., that they can send HTTP Datagrams with payloads of at least 1280 bytes).  This can be accomplished using various techniques: if HTTP intermediaries are not in use, both CONNECT-IP endpoints can pad the QUIC INITIAL packets of the underlying QUIC connection that CONNECT-IP is running over. if HTTP intermediaries are in use, CONNECT-IP endpoints can enter in an out of band agreement with the intermediaries to ensure that endpoints and intermediaries pad QUIC INITIAL packets. CONNECT-IP endpoints can also send ICMPv6 echo requests with 1232 bytes of data to ascertain the link MTU and tear down the tunnel if they do not receive a response.  Unless endpoints have an out of band means of guaranteeing that one of the two previous techniques is sufficient, they MUST use this method. </ins> Endpoints MAY implement additional filtering policies on the IP packets they forward.

I get that, but since the complexity is added by intermediaries we just need to make it clear that the intermediary needs to not be silly
One part is to detect if the link is capable of 1232 bytes, however, if not, we also not to provide more guidance what to do. I think there are two options: either close the connection or use quic streams instead of data datagrams.


Question: The ACK frame uses the least significant bit of the type value (that is, type 0x03) to indicate ECN feedback and report receipt of QUIC packets with associated ECN codepoints of ECT(0), ECT(1), or ECN-CE <del> in the packet's IP header.  ECN Counts are only present when the ACK </del> <ins> in the packet's IP header.  ECN counts are only present when the ACK </ins> frame type is 0x03. When present, there are three ECN counts, as shown in ecn-count- format. <del> The three ECN Counts are: </del> <ins> The ECN count fields are: </ins> A variable-length integer representing the total number of packets received with the ECT(0) codepoint in the packet number space of

This one slipped the net.
Either way is fine by me. Just ship it already :PThanks, Martin!


Question: Verifier Owner may need to trust the Verifier before giving the Endorsement, Reference Values, or Appraisal Policy to it.  This can be done similarly to how a Relying Party might establish trust in a <del> Verifier as discussed above, and in such a case, mutual attestation might even be needed as discussed in rpowner-trust. </del> <ins> Verifier as discussed above, and in such a case, mutual authentication or attestation might even be needed as discussed in rpowner-trust. </ins> 8.

TLS authentication is sufficient for establishing trust in the various entities before sending them privacy sensitive data. TLS is also widely supported today and attestation between servers is not. Most likely TLS will be the mechanism. This PR adds mention to in addition to attestation.
Addresses
says that the way for an Endorser to trust and Verifier is to use attestation. While I won't categorically say this should never be done, it seems like this is overly complex and a somewhat misuse of attestation. I think of the Verifier as a service like a web site, not an implementation like a TEE, so TLS or similar, not attestations, seems like the more efficient and conventional way to do this. I think attestation is associated with actual HW and SW, for example the Dell server, Intel CPU and Web stack used to run the service. The operator of a Verifier gets to change those around without having to tell the people using the service. On the other hand, if a Verification Service changes owners, say from Chinese to American, then Endorser concerned about confidential stuff in the Endorsements really wants to know. I also think that Verifier Owner should not be lumped in with Endorser. The Verifier Owner is very likely to be tightly coupled organizationally with the Verifier, where the Endorser is not likely to be.
PR addresses this well enough.
Reviewing the trust relationships described in the architecture document got me thinking about sharpening a distinction between device attestation (e.g., EAT for phone) and service authenticity (e.g. HTTPS for a web site): Device Attestation Focus is on the implementation, the HW and SW The legal entity of interest is the device manufacturer The claims are about the HW and the SW, not the device owner Example: a mobile phone (not its owner/user) Example: integrity of a remote device Service Authenticity Focus is on the provider of the service, not the HW or SW The legal entity of interest is the service provider There is no equivalent of claims, but if there was they would be about the business or person operating the service Example: a web site Example: an email provider (IMAP service) Service providers (e.g., web sites) use whatever HW and SW they want to use. As a user of the service we don’t care what HW and SW they select.


Question: Verifier Owner may need to trust the Verifier before giving the Endorsement, Reference Values, or Appraisal Policy to it.  This can be done similarly to how a Relying Party might establish trust in a <del> Verifier as discussed above, and in such a case, mutual attestation might even be needed as discussed in rpowner-trust. </del> <ins> Verifier as discussed above, and in such a case, mutual authentication or attestation might even be needed as discussed in rpowner-trust. </ins> 8.

We trust the service provider to pick good HW and SW and to configure and operate it correctly. They might change the HW and the SW and as a user we won’t know. For example we trust Bank of Xxx to operate their banking web site correctly with little concern for their HW. This is why attestation of the HW and SW is not so useful to the user of a service. Users of mobile phones and IoT devices are not trustworthy to pick secure HW and SW. They are not professionals like the people running web sites. There is little accountability against them if they pick bad SW and HW. Therefor we want attestation for these usage scenarios. This is why EAT, FIDO and Android Attestation exist. I don’t think the line between device attestation and service authenticity is completely black and white. In some use cases attestation may be bent to provide some service authenticity. But I don’t think it makes sense to add attestation to the millions of web sites out there, add support for it to browsers and make attestation an extra layer of security for web browsing. I think it is fine for a service provider to use attestation inside their service to monitor integrity and such of the HW and SW providing the service. But usually that won’t be manifest outside the service. It could be awkward for the user to the service to have to re evaluate the service every time the service provider changes HW. Another way to look at this is from a certification / regulatory view. If the thing you're looking at is a candidate for implementation certification such as Common Criteria, FIPS, FIDO certification, then it probably lines up with device attestation. On the other hand, if the thing your are looking at lines up with regulation (e.g., banking regulation, GDPR,...) and to some degree OWASP, then it probably lines up with service authenticity. I think the RATS architecture document should probably have text that describes when attestation is applicable and when it is not, for example saying something like “attestation is not a replacement or augmentation for HTTPS/TLS"
In the last editors session, the editor's rough consensus was that https does not provide the authenticity but it provides a secure channel. Instead, authenticity is checked via successful authentication rooted in identity documents (I hope I captured appropriately, please bash).


Question: Verifier Owner may need to trust the Verifier before giving the Endorsement, Reference Values, or Appraisal Policy to it.  This can be done similarly to how a Relying Party might establish trust in a <del> Verifier as discussed above, and in such a case, mutual attestation might even be needed as discussed in rpowner-trust. </del> <ins> Verifier as discussed above, and in such a case, mutual authentication or attestation might even be needed as discussed in rpowner-trust. </ins> 8.

Taking that into account, how does this effect this issue? Looking at the last paragraph, "Remote attestation procedures are not a replacement for secure authentication, they are based on secure authentication." might be missing vital parts. Does anything need to be added or modified here?
Note that is INCORRECTLY referenced here. Is the resolution of a different issue about HTTPS vs Attestation.
I think comments from NAME do need to be taken into account when the text to address this issue is written. My text in opening the issue is not intended as the text to resolve the issue. Probably the text that resolves the issue needs to refer to "HTTPS plus the web certificate infrastructure that includes public CA's and browsers and/or private CA's used in B-to-B..." or such.
I think the RATS architecture document should probably have text that describes when attestation is applicable and when it is not, for example saying something like “attestation is not a replacement or augmentation for HTTPS/TLS" The design team disagrees with this. Every authentication (including HTTPS) can be said in terms of attestation. It is the verifier that allows the decoupling of the RP to know the expected state of the Attester.


Question: Local Endpoint is specified, the implementation should either use an ephemeral port or generate an error. <del> If the Path Selection Properties do not require a single network interface or path, but allow the use of multiple paths, the Listener object should register for incoming traffic on all of the network interfaces or paths that conform to the Path Selection Properties. The set of available paths can change over time, so the implementation should monitor network path changes and register and de-register the Listener across all usable paths.  When using multiple paths, the Listener is generally expected to use the same port for listening on each. If the Protocol Selection Properties allow multiple protocols to be used for listening, and the implementation supports it, the Listener object should register across the eligble protocols for each path. This means that inbound Connections delivered by the implementation may have heterogeneous protocol stacks. </del> <ins> If the Selection Properties do not require a single network interface or path, but allow the use of multiple paths, the Listener object should register for incoming traffic on all of the network interfaces or paths that conform to the Properties.  The set of available paths can change over time, so the implementation should monitor network path changes and register and de-register the Listener across all usable paths.  When using multiple paths, the Listener is generally expected to use the same port for listening on each. If the Selection Properties allow multiple protocols to be used for listening, and the implementation supports it, the Listener object should register across the eligble protocols for each path.  This means that inbound Connections delivered by the implementation may have heterogeneous protocol stacks. </ins> 4.8.1.

Several small updates, mostly to clean things up a bit, and align property names and such.
NAME - I think you're wrong: my understanding (and also the way the text is written - otherwise we'd have to change it in the API draft) was that Connection Properties can (and should) also be set on Preconnections, to give as much input as possible for Connection establishment. In the API draft, the Selection Properties in Section 5.2 can only be used on Preconnections: you can't ask for congestion control after establishment. You can't say that you'd rather have multistreaming support after establishment. But that doesn't exclude Connection properties from being usable earlier. Required minimum checksum coverage could influence choosing UDP-Lite vs. UDP. Bounds on Send or Receive Rate could influence the interface choice. The capacity profile could influence various things (and no, there isn't a matching Selection Property now - there is a per-Message override). So... I think that either you're wrong, or I'm missing something and the API draft heavily lags behind a decision that was made. I hope not!
NAME The API drafts says: I interpret this as only the Selection Properties influence selection and the Connection properties are used to further configure the connections according to user preferences. Otherwise what is the difference between them?
If you interpret it as you do, I think the text isn't clear enough (indeed it isn't super clear - in particular, "eventually established" is misleading IMO). Though, note that it says that Connection properties can also be used later. Regarding the difference between them: Selection Properties only make sense for Preconnections.
NAME - sorry lost this thread. But I saw that you have started cleaning up some Connection properties that had mixed Connection and Selection properties up in the API draft. Does it mean that we are now in synch and agree that only Selection properties are used in the selection and Connection properties are used for configuration (both during establisment and also after the connection setup).
NAME the other PR means nothing about what I said here: it only tried to address issue which is about a wording inconsistency in a specific property. You say that Connection properties can also be used during establishment - if you agree with allowing this, then why not allow to configure them as early as possible and allow them to influence selection?


Question: Local Endpoint is specified, the implementation should either use an ephemeral port or generate an error. <del> If the Path Selection Properties do not require a single network interface or path, but allow the use of multiple paths, the Listener object should register for incoming traffic on all of the network interfaces or paths that conform to the Path Selection Properties. The set of available paths can change over time, so the implementation should monitor network path changes and register and de-register the Listener across all usable paths.  When using multiple paths, the Listener is generally expected to use the same port for listening on each. If the Protocol Selection Properties allow multiple protocols to be used for listening, and the implementation supports it, the Listener object should register across the eligble protocols for each path. This means that inbound Connections delivered by the implementation may have heterogeneous protocol stacks. </del> <ins> If the Selection Properties do not require a single network interface or path, but allow the use of multiple paths, the Listener object should register for incoming traffic on all of the network interfaces or paths that conform to the Properties.  The set of available paths can change over time, so the implementation should monitor network path changes and register and de-register the Listener across all usable paths.  When using multiple paths, the Listener is generally expected to use the same port for listening on each. If the Selection Properties allow multiple protocols to be used for listening, and the implementation supports it, the Listener object should register across the eligble protocols for each path.  This means that inbound Connections delivered by the implementation may have heterogeneous protocol stacks. </ins> 4.8.1.

What's bad about having more information available that can guide the selection process?
NAME the problem is that the Connection properties are not expressed in a way suitable for selection. If they should also be used for Selection they would have to be reworked to be of type Preference. Otherwise I do not know what they mean for selection. What does it mean if I pick say a particular connection group transmission scheduler? Is it required or preferred? Most connection properties also do not look very useful to me for selection so in case there are some we need I think it would be simpler to just define a matching selection property.
NAME we're only discussing whether it's allowed to use them as additional input or not. I agree, some (like the scheduler choice) don't seem useful, but some do... generally we may not have good guidance on how they should be used by an implementation, but giving the system freedom to use as many inputs as possible is probably a good thing. Regarding the type, require/prefer//ignore/avoid/prohibit obviously works fine for a "wishlist" of boolean character: having something, or not having it. The selection properties are generally like that - except "Interface Instance or Type" and "Provisioning Domain Instance or Type" which are, well... wishlist-collections. Consider the capacity profile. Surely configuring this early can be useful input - e.g. if someone picks low latency / interactive, you may even want to put this traffic on a specific network interface! But if you'd want to make a copy of the capacity profile as a "proper" selection property, how would you do it? Apply require/prefer/ignore/avoid/prohibit for each of its values? Then I could say "prefer" low latency/non-interactive but "avoid" low latency/non-interactive, which would surely be odd. Apply the qualifier to the whole thing? Then it's about, e.g. "avoiding" to use a capacity profile at all. That's odd as well. I think it's in the nature of the current selection properties that they merely say "I want to be able to do this, and this is how important it is to me", which is different from configuring something. That's also why, as written, they simply wouldn't work for later connection configuration.


Question: Local Endpoint is specified, the implementation should either use an ephemeral port or generate an error. <del> If the Path Selection Properties do not require a single network interface or path, but allow the use of multiple paths, the Listener object should register for incoming traffic on all of the network interfaces or paths that conform to the Path Selection Properties. The set of available paths can change over time, so the implementation should monitor network path changes and register and de-register the Listener across all usable paths.  When using multiple paths, the Listener is generally expected to use the same port for listening on each. If the Protocol Selection Properties allow multiple protocols to be used for listening, and the implementation supports it, the Listener object should register across the eligble protocols for each path. This means that inbound Connections delivered by the implementation may have heterogeneous protocol stacks. </del> <ins> If the Selection Properties do not require a single network interface or path, but allow the use of multiple paths, the Listener object should register for incoming traffic on all of the network interfaces or paths that conform to the Properties.  The set of available paths can change over time, so the implementation should monitor network path changes and register and de-register the Listener across all usable paths.  When using multiple paths, the Listener is generally expected to use the same port for listening on each. If the Selection Properties allow multiple protocols to be used for listening, and the implementation supports it, the Listener object should register across the eligble protocols for each path.  This means that inbound Connections delivered by the implementation may have heterogeneous protocol stacks. </ins> 4.8.1.

Connection properties really are about configuring something, but I still say that being able to use them as early input is good. I agree that the implementation isn't super clear as they don't come with a require/prefer/... etc. qualifier, but then we just offer some freedom to implementers.
NAME I just pushed a commit that updates the selection and connection properties text, removing "connection" almost everywhere, but adding a disclaimer in the beginning that says that connection properties should be configured as early as possible because they can influence decisions made by the implementation, and "In the remainder of this document, we only refer to Selection Properties because they are the more typical case." Frankly I'm not sure this is better... but it does remove most of the "selection and connection" double mentions that you pointed out.
Fixed up branch building and resolved merge conflicts. I'll merge this in later today before the posting deadline.
Cool thanks! Sorry if I messed something up
As discussed in
Thank you for updating this – looks good! just needs s/send-params/msg-properties/ and merging


Question: Server Initial datagram as shown in fig-server-initial typically containing three packets: an Initial packet with the beginning of the server's side of the TLS handshake, a Handshake packet with the rest <del> of the server's side of the TLS handshake, and initial 1-RTT data, if </del> <ins> of the server's side of the TLS handshake, and 1-RTT data, if </ins> present. The Client Completion datagram contains at least one Handshake 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: 6.1. <del> When applicable for non-interactive tokens, all Origins SHOULD implement a robust storage-query mechanism for checking that tokens sent by clients have not been spent before.  Such tokens only need to be checked for each Origin individually.  But all Origins must perform global double-spend checks to avoid clients from exploiting the possibility of spending tokens more than once against distributed token checking systems.  For the same reason, the global data storage must have quick update times.  While an update is occurring it may be possible for a malicious client to spend a token more than once. 6.2. </del> When a Client holds tokens for an Issuer, it is possible for any verifier to invoke that client to redeem tokens for that Issuer. This can lead to an attack where a malicious verifier can force a

The section references seem to be breaking the build, so I'll fix that later before we land this.
Origins currently choose whether their challenges are per-origin or cross-origin. For cross-origin, double-spend prevention is only as good as the coordination between origins and the Issuer. Per-origin allows double-spend prevention to be isolated to a single origin; also prevents the cache of tokens being take up by some other origin.
Write this as "any origins that share a redemption context must also share double spend state for that context."


Question: IANA are requested to create a new entry in the "HTTP Problem Type" registry established by PROBLEM. <ins> 9.5. IANA are requested to create a new entry in the "HTTP Problem Type" registry established by PROBLEM. </ins> 10.  References 10.1.  URIs <del> [1] https://www.iana.org/assignments/media-types </del> <ins> [1] https://iana.org/assignments/media-types [2] https://iana.org/assignments/http-problem-types </ins> Index

We debated this at some length. This defines an optional-to-use signaling method for the gateway so that it can inform the client when the key configuration is not acceptable. Clients can use this as a prompt to refresh their configuration.
(Originally posted by NAME in URL): Any lifetime information that comes with a key is helpful, but in practice is insufficient for clients to drive key updates. Key rotation may not occur on a fixed schedule and it may not occur synchronously (some gateway instances might not get the key at the right time. This means that clients cannot distinguish between failures due to configuration mismatch or request tampering (or whatever else might cause the gateway to reply without an encapsulated response). If the client can't distinguish then the client's recovery implementation is a lot more complicated. Does it always try to forcibly update its configuration? Does it implement some sort of backoff? Can a relay abuse the client's recovery implementation to somehow put the client into a state where it is stuck with an old configuration? It seems vastly simpler to just be clear that the configuration is wrong as a signal to the client that it's time to update. This was done in ODoH with a 401 response code. proposed a new problem statement, which I think is the right way to deal with this. Yes, there are potential privacy problems based on automated updates, but those depend on the client's consistency story and is therefore not something new a signal would introduce.
This looks great -- thanks!


Question: The entries in iana-setting-table are registered by this document. <del> Additionally, each code of the format "0x1f * N + 0x21" for non- negative integer values of N (that is, 0x21, 0x40, ..., through 0x3ffffffffffffffe) MUST NOT be assigned by IANA. </del> <ins> Each code of the format "0x1f * N + 0x21" for non-negative integer values of N (that is, 0x21, 0x40, ..., through 0x3ffffffffffffffe) MUST NOT be assigned by IANA and MUST NOT appear in the listing of assigned values. </ins> 11.2.3.

IANA's review wanted to know whether the 148 quadrillion GREASE values should be listed on the registry page as Reserved, and what note should be present if not. They should not, lest it be challenging to actually download the registry page within the lifetime of the human race. This rewords the restriction and clarifies that the restriction should be the note.
NAME Can you add the same text to 22.2 (Transport parameter registry) in the transport doc as well?
NAME said
We believe the note, updated based on IANA feedback during Last Call, is sufficient. Reserving 148 quadrillion values in the registry would cause the registry to be impossible to download in the lifetime of humanity; see . The purpose is described in , to exercise the requirement to ignore unknown frame types (a.k.a. GREASE). Proposal is to make no change; this was already covered during IANA review.
If IANA has agreed, then I am fine with it. BTW, I was not asking to make those zillions of reservations but leaving a note in the registry ;-)
It's a fair question but the WG and the responsible AD are satisfied with our IANA considerations at this stage, closing with no action.
Please see the for full context. This issue is tracking the following IANA question:
Copying my e-mail reply here: unknown values; having an explicit list encourages an explicitly defined reaction. This also applies to and , where the latter has better wording.
NAME replied to IANA . Closing this issue.
Actually, NAME said in response to the e-mail that this guidance should be in the document: Let's re-open to track. I don't think all instances of this question need separate issues, though.
So the related questions were: Should the GREASE values be listed in the registry as Reserved? No, because some rough estimation suggests it would take a few millennia to download the registry page if this were the case. Should there be a note indicating which values cannot be assigned? I'm of two minds, because while I really don't want people embedding the formula into their parsers, I do want people embedding the formula into their generators. says no Reserved listing, yes note.
I think this is proposal ready. Forwarding Mike's response to the list, to there is an archive to link to. Lars >


Question: RFC6455. All requests for which the CoAP client has not received a response <del> yet are cancelled when the connection is closed.  If the client observes one or more resources over the WebSocket connection, then the CoAP server (or intermediary in the role of the CoAP server) MUST remove all entries associated with the client from the lists of observers when the connection is closed. </del> <ins> yet are cancelled when the connection is closed. </ins> 4.

Added tag for updates: Removed Observe references scattered throughout the draft Added appendix to collect clarifications to RFC7641 in one place
Captured from : There needs to be a section on how resources can be observed over TCP/TLS. Questions include: Do servers need to include a sequence number in notifications, given that TCP/TLS provides reliable, in-order delivery? How do servers throttle the stream of notifications when the resource changes its state more frequently than the server can send notifications? Does a client have to re-register its interest in a resource if it hasn't received a notification for a while but the connection is still up?
URL In a conference call with Klaus, Simon, Bill and myself we discussed the issue and the raised questions. QUESTION 1: Do servers need to include a sequence number in notifications, given that TCP/TLS provides reliable, in-order delivery? The text from the CoAP over Websockets could be re-used. Here is what it could say: Since the TCP provides ordered delivery of messages, the mechanism for reordering detection when observing resources [RFC7641] is not needed. The value of the Observe Option in notifications therefore MAY be empty on transmission and MUST be ignored on reception. The Observe Option still has to be sent since it contains other information as well. QUESTION 2: Does a client have to re-register its interest in a resource if it hasn't received a notification for a while but the connection is still up? We believe it is enough to send keep-alive message for the CoAP over TCP draft (as opposed to send a re-registration message when used over UDP). QUESTION 3: How do servers throttle the stream of notifications when the resource changes its state more frequently than the server can send notifications? TCP offers a flow control mechanism as a means for the receiver to govern the amount of data sent by the sender. This is functionality that can be re-used.
URL (1) -- I agree with the proposed solution. (2) -- we need text that empty messages are no-ops and can be used as a keep-alive mechanism. (3) -- this is a problem that most applications that start using reliable streams run into. Early implementations sometimes ignore the need for proper flow control.


Question: RFC6455. All requests for which the CoAP client has not received a response <del> yet are cancelled when the connection is closed.  If the client observes one or more resources over the WebSocket connection, then the CoAP server (or intermediary in the role of the CoAP server) MUST remove all entries associated with the client from the lists of observers when the connection is closed. </del> <ins> yet are cancelled when the connection is closed. </ins> 4.

That is not a reason to add more mechanism (that would also not be in those early implementations), but the solution should be to use the flow control mechanisms we already have. Maybe an LWIG document could have information about flow control support in implementations, e.g., URL
URL Has someone thought about the following setup, that is, will it work with keep-alives only? There is a cluster behind a load balancer. I client sets up a connection, is assigned to one node and starts observing. For some reason the node fails and through a fail-over strategy the client is handed over to another node. All further requests are answered transparently be the new node. However, the Observe relationship got lost...
URL Subissue 1 is covered by the Websockets draft; close this when merged. Subissue 2 is covered by text in the Websockets draft; the failover problem described can be addressed by proper implementation of failover. This may however be costly, so we should have some more discussion about this. Subissue 3 is an implementation quality issue. Just don't implement stream-based without some flow control.
On the 7/6 call with the authors, NAME agreed to review and determine whether all issues were addressed.
This issue does not include a question about the use of confirmable messages for notifications in to determine client aliveness and interest: A notification can be confirmable or non-confirmable ... An acknowledgement message signals to the server that the client is alive and interested in receiving further notifications; if the server does not receive an acknowledgement in reply to a confirmable notification, it will assume that the client is no longer interested and will eventually remove the associated entry from the list of observers This is inappropriate for CoAP over reliable transports.
The phrasing is a bit unfortunate. What is meant is "if the confirmable notification times out". The same would be true in TCP -- if the connection times out, you remove the interests. We could clarify this in a section of CoAP-TCP about the impact of no longer sending back and forth acknowledgements.
Discussing the scope of this issue with the authors. This probably needs a more complete treatment rather than small references like: Since the TCP protocol provides ordered delivery of messages, the mechanism for reordering detection when observing resources [RFC7641] is not needed.


Question: RFC6455. All requests for which the CoAP client has not received a response <del> yet are cancelled when the connection is closed.  If the client observes one or more resources over the WebSocket connection, then the CoAP server (or intermediary in the role of the CoAP server) MUST remove all entries associated with the client from the lists of observers when the connection is closed. </del> <ins> yet are cancelled when the connection is closed. </ins> 4.

The value of the Observe Option in notifications MAY be empty on transmission and MUST be ignored on reception.
My reading of the discussion thread gives me the impression that we had closed the initial set of questions. The subsequently raised issue regarding the wording in RFC 7641 I am wondering whether there is any change needed. We definitely are not going to change RFC 7641. I personally don't think that the highlighted text from RFC7641 will be wrongly implemented. I would suggest to close issue.
NAME - this would be - any comments on that pull request?


Question: gateway to convert between the signaling on the web side to the SIP signaling may be needed. <del> When a new codec is specified, and the SDP for the new codec is specified in the MMUSIC WG, no other standardization should be required for it to be possible to use that in the web browsers. Adding new codecs which might have new SDP parameters should not change the APIs between the browser and Javascript application. As soon as the browsers support the new codecs, old applications written before the codecs were specified should automatically be able to use the new codecs where appropriate with no changes to the JS applications. </del> <ins> When an SDP for a new codec is specified, no other standardization should be required for it to be possible to use that in the web browsers.  Adding new codecs which might have new SDP parameters should not change the APIs between the browser and Javascript application.  As soon as the browsers support the new codecs, old applications written before the codecs were specified should automatically be able to use the new codecs where appropriate with no changes to the JS applications. </ins> The particular choices made for WebRTC, and their implications for the API offered by a browser implementing WebRTC, are described in I-

Tweaks to future proof document.
Spencer had the following comment: I found the reference to MMUSIC WG in When a new codec is specified, and the SDP for the new codec is specified in the MMUSIC WG, no other standardization should be required for it to be possible to use that in the web browsers. to be odd. MMUSIC may be around forever, but this work might be refactored at some point in the future. Is the point that When SDP for a new codec is specified, no other standardization should be required for it to be used in the web browsers. Or is there another way to say this? I'm also wondering if the statement is true for any WebRTC endpoint, not just browsers.


Question: 9.6. <del> The main MLS key schedule provides a "resumption_secret" that is used as a PSK to inject entropy from one epoch into another.  This </del> <ins> The main MLS key schedule provides a "resumption_psk" that is used as a PSK to inject entropy from one epoch into another.  This </ins> functionality is used in the reinitialization and branching processes described in reinitialization and sub-group-branching, but may be used by applications for other purposes. Some uses of resumption PSKs might call for the use of PSKs from historical epochs.  The application SHOULD specify an upper limit on <del> the number of past epochs for which the "resumption_secret" may be </del> <ins> the number of past epochs for which the "resumption_psk" may be </ins> stored. 9.7. <del> The main MLS key schedule provides a per-epoch "authentication_secret".  If one of the parties is being actively impersonated by an attacker, their "authentication_secret" will differ from that of the other group members.  Thus, members of a group MAY use their "authentication_secrets" within an out-of-band authentication protocol to ensure that they share the same view of the group. </del> <ins> The main MLS key schedule provides a per-epoch "authentication_code". If one of the parties is being actively impersonated by an attacker, their "authentication_code" will differ from that of the other group members.  Thus, members of a group MAY verify out of band that their "authentication_code" values are all the same, in order to ensure that they share the same view of the group. </ins> 10.

Looks good! I would maybe add 1 sentence explaining the rule for what is called a secret and what is called a key. The rule says that a secret will go through some key derivation to obtain a key/keys, is that right?
NAME Yep, that's exactly right. Added a sentence. Thanks for the feedback everyone!
Reopening due to incomplete discussion and inaccuracy. The claims presented are incorrect in response by NAME demonstrate a misunderstanding of the roles of these keys and uses, and constitutes a change to the fundamental uses thereof. The authenticationsecret is NOT a public value nor was it ever introduced as such. If an application is using it as a public value, that onus is on the developers, and I advise against using Cisco's implementation as a standard for MLS. As shown in the name and all original discussion, the authenticationsecret is intended as a secret value (or else 'public' in the same way that the group secret is 'public). It is used for further derivation to support authentication. This should be corrected, and such foundational changes without WG discussion should be avoided.
Clearly we have a disagreement here NAME but maybe let's hold off on attributing misunderstanding. It seems like you are claiming that in order for MLS to be secure, the needs to be protected to the same level as the . What is this based on? If that is true, then we should probably remove the altogether. The "out-of-band" protocol in which it would be confirmed would have to be part of the MLS security analysis, thus no longer out-of-band.
NAME that is not the claim at all - different uses imply different compromise risks. This should not be a new concept to understand. As demonstrated in the terminology, the authenticationsecret is linked to authentication, not confidentiality. MLS, per specification, outsources authentication (the AS is not explicitly defined and all entity authentication within MLS relies on it). Even though it is definitionally out-of-scope and therefore out-of-band (certificates, authentication verification are entirely out-of-band of the protocol spec), we do not say that signature secret keys can be "public" - yet you are asserting that an authenticationsecret which can support epoch-level authentication would be public.


Question: 9.6. <del> The main MLS key schedule provides a "resumption_secret" that is used as a PSK to inject entropy from one epoch into another.  This </del> <ins> The main MLS key schedule provides a "resumption_psk" that is used as a PSK to inject entropy from one epoch into another.  This </ins> functionality is used in the reinitialization and branching processes described in reinitialization and sub-group-branching, but may be used by applications for other purposes. Some uses of resumption PSKs might call for the use of PSKs from historical epochs.  The application SHOULD specify an upper limit on <del> the number of past epochs for which the "resumption_secret" may be </del> <ins> the number of past epochs for which the "resumption_psk" may be </ins> stored. 9.7. <del> The main MLS key schedule provides a per-epoch "authentication_secret".  If one of the parties is being actively impersonated by an attacker, their "authentication_secret" will differ from that of the other group members.  Thus, members of a group MAY use their "authentication_secrets" within an out-of-band authentication protocol to ensure that they share the same view of the group. </del> <ins> The main MLS key schedule provides a per-epoch "authentication_code". If one of the parties is being actively impersonated by an attacker, their "authentication_code" will differ from that of the other group members.  Thus, members of a group MAY verify out of band that their "authentication_code" values are all the same, in order to ensure that they share the same view of the group. </ins> 10.

The authenticationsecret is private in the same way that the exportervalue is: based on key schedule derivation, compromise of these keys should not undermine MLS, but secrecy of them can be leveraged. So, unless you are now proposing removing mention of certificates, etc. from the core protocol, there really isn't a logical basis for removing 'out-of-band' authentication mechanisms.
What I'm trying to get at here is: What is the negative consequence if an is known by someone other than the members of a group? For example, leaking a signature private key would enable the attacker to impersonate the corresponding member. Leaking the would allow the attacker to decrypt all of the group's messages. What is the bad consequence of an becoming public?
NAME I am also assuming that this push to use epoch-level authentication values as public codes is based on Cisco's implementation choice. I would be curious to know why there is such insistence that all implementations also go about using authentication in the same way. If an implementation chooses to use it as a secret value to then derive an authentication code based on yet other information, why is that a problem?
Publishing the authenticationsecret as a public value has a similar effect as publishing the exportervalue as public - it is not an issue for MLS per protocol spec, but it would definitely be an issue for the application if it is then used for something secret thereafter. One reason not to specify these as public values is that they are derived following the conditions placed on keys and therefore could be used as secret values if desired by an application. Forcing or annotating them as immediately public, or as final 'codes', when such a use as a secret key is an out-of-scope potential is artificial and unnecessary.
I'm not pushing anything, I'm just trying to get the protocol to be clear. If this thing needs to be secret, then it needs to be secret, and we need to document that so that people keep it secret. Thanks for elaborating more on your concerns. I see what you're saying -- basically the same point that NAME and I discussed . I think the core of the disagreement here is what we mean when we say that a value is "public" or "secret".


Question: 9.6. <del> The main MLS key schedule provides a "resumption_secret" that is used as a PSK to inject entropy from one epoch into another.  This </del> <ins> The main MLS key schedule provides a "resumption_psk" that is used as a PSK to inject entropy from one epoch into another.  This </ins> functionality is used in the reinitialization and branching processes described in reinitialization and sub-group-branching, but may be used by applications for other purposes. Some uses of resumption PSKs might call for the use of PSKs from historical epochs.  The application SHOULD specify an upper limit on <del> the number of past epochs for which the "resumption_secret" may be </del> <ins> the number of past epochs for which the "resumption_psk" may be </ins> stored. 9.7. <del> The main MLS key schedule provides a per-epoch "authentication_secret".  If one of the parties is being actively impersonated by an attacker, their "authentication_secret" will differ from that of the other group members.  Thus, members of a group MAY use their "authentication_secrets" within an out-of-band authentication protocol to ensure that they share the same view of the group. </del> <ins> The main MLS key schedule provides a per-epoch "authentication_code". If one of the parties is being actively impersonated by an attacker, their "authentication_code" will differ from that of the other group members.  Thus, members of a group MAY verify out of band that their "authentication_code" values are all the same, in order to ensure that they share the same view of the group. </ins> 10.

I mean something very specific: That the security guarantees of MLS will not be harmed if a public value is provided to the attacker, and they will be harmed if a secret value is provided to the attacker. Importantly, "public" does not mean that the value must be published, or that the security guarantees of an application using MLS will not be harmed if you publish the value. It seems to me that in that specific taxonomy, the is "public" and not "secret". Do you disagree?
BTW, the notion of something being "public" at one layer but "secret" at another is not new. For example, values exported from TLS aren't secret in the sense that publishing them would compromise TLS. You could authenticate your TLS connection by exporting a value and shouting it through megaphones across a crowd. But the main way people use them in practice is as secrets in some other protocol. Every WebRTC call uses SRTP keys that are exported from TLS, and many people would be very sad if these "public" values were published.
If it would help, I would be happy to add some text to the Security Considerations clarifying this distinction. Basically, draw a clear boundary around which values have to be kept confidential to the client in order to realize the security properties of the protocol. But I still think is a clearer name. Doesn't mean you have to use it in any particular way in your application; doesn't prevent it being used as a shared secret; but also doesn't imply that the MLS library has to worry about its secrecy.
Within the bounds of that particular taxonomy there is not an issue. However, while you may be following a given taxonomy, that is not a general one, one that would be widely understood by implementors, nor is it specified in the protocol spec. It is not an unreasonable taxonomy, but it certainly not a general one. If you are looking for MLS usage spec to align to your taxonomy then I do think that there should be a clarification section added. BTW as a point case on more standard terminology use, the TLS 1.3 spec uses the exact phrasing for exporters as "exportermastersecret", from which the actual exporter may be derived. This aligns very closely to the authenticationsecret, from which an authentication code later may be derived.


Question: 9.6. <del> The main MLS key schedule provides a "resumption_secret" that is used as a PSK to inject entropy from one epoch into another.  This </del> <ins> The main MLS key schedule provides a "resumption_psk" that is used as a PSK to inject entropy from one epoch into another.  This </ins> functionality is used in the reinitialization and branching processes described in reinitialization and sub-group-branching, but may be used by applications for other purposes. Some uses of resumption PSKs might call for the use of PSKs from historical epochs.  The application SHOULD specify an upper limit on <del> the number of past epochs for which the "resumption_secret" may be </del> <ins> the number of past epochs for which the "resumption_psk" may be </ins> stored. 9.7. <del> The main MLS key schedule provides a per-epoch "authentication_secret".  If one of the parties is being actively impersonated by an attacker, their "authentication_secret" will differ from that of the other group members.  Thus, members of a group MAY use their "authentication_secrets" within an out-of-band authentication protocol to ensure that they share the same view of the group. </del> <ins> The main MLS key schedule provides a per-epoch "authentication_code". If one of the parties is being actively impersonated by an attacker, their "authentication_code" will differ from that of the other group members.  Thus, members of a group MAY verify out of band that their "authentication_code" values are all the same, in order to ensure that they share the same view of the group. </ins> 10.

TLS is generic in 'exporters' after an exportermastersecret is derived. Thus there is alignment of the term 'secret' to things that can be used for keys (have sufficient entropy), not as an alignment to public/secret at different layers. I am strongly opposed to the use of authenticationcode, and see no clear definition for that in your customized taxonomy. As such, it would follow the more standard interpretations, such as message authentication codes, etc. which are indeed explicitly public values. Codes are normally an end-state in themselves, not keys. Perhaps it is a clearer name for the Cisco implementation design, but it is not a clear name for something that can/should be used for further derivations as may be more generally applicable. Following the TLS norm for exporters, the name would be "authenticationmastersecret". The shortened "authenticationsecret" was a good name. If you are going to provide a write-up for the custom taxonomy, then an adjustment to "authenticationexporter" could perhaps be made.
The notion of a having boundary to the protocol and values that may or may not safely be exported across the boundary is not exotic or customized. You see it in FIPS 140-2, PKCS, and WebCrypto, and in tools like ProVerif and Tamarin. The only choice I'm making is in the specific words, in particular reserving "secret" for "something that should not be exposed outside the protocol implementation because it would compromise the protocol". I hope you agree that is different from say in this regard. It would be OK for an MLS implementation to have an API for an application to access the , but it would not be OK to have an API to access the . I'm not wedded to , I just want something other than . If you want to align with the exported values (to which I agree it is similar), we could call it an .
The distinction between those documents and tools, and the MLS spec is that those tools clearly define the boundaries and terminology, vs. just anyone making terminology assumptions without them being specified in the document. Not to be pedantic, but if you want to assume specific meanings for terms relative to a threat model for MLS only, versus a general usage ("exporter secrets" being secrets used in other things), then you really do need to specify those assumptions within the specification itself and not various side discussions.


Question: 9.6. <del> The main MLS key schedule provides a "resumption_secret" that is used as a PSK to inject entropy from one epoch into another.  This </del> <ins> The main MLS key schedule provides a "resumption_psk" that is used as a PSK to inject entropy from one epoch into another.  This </ins> functionality is used in the reinitialization and branching processes described in reinitialization and sub-group-branching, but may be used by applications for other purposes. Some uses of resumption PSKs might call for the use of PSKs from historical epochs.  The application SHOULD specify an upper limit on <del> the number of past epochs for which the "resumption_secret" may be </del> <ins> the number of past epochs for which the "resumption_psk" may be </ins> stored. 9.7. <del> The main MLS key schedule provides a per-epoch "authentication_secret".  If one of the parties is being actively impersonated by an attacker, their "authentication_secret" will differ from that of the other group members.  Thus, members of a group MAY use their "authentication_secrets" within an out-of-band authentication protocol to ensure that they share the same view of the group. </del> <ins> The main MLS key schedule provides a per-epoch "authentication_code". If one of the parties is being actively impersonated by an attacker, their "authentication_code" will differ from that of the other group members.  Thus, members of a group MAY verify out of band that their "authentication_code" values are all the same, in order to ensure that they share the same view of the group. </ins> 10.

Such specification is part of what a standard is meant to do. Within other documents this is clearly defined - i.e., what is known as a threat model - such as one sees in ProVerif and Tamarin. Assuming that you are going to add such a taxonomy, and that it follows that described by you in , then the correct term would be "authenticationkey". To quote you what you said there, which correctly aligns to a reasonable usage of the authenticationkey. pointed out that there is a lack of consistency in the usage of "keys" vs "secrets" throughout the document - something that still has not been addressed and should not be dropped from the issue for edits. Regardless of what taxonomy is being used or even assumed, there is no consistency in it in the spec. You seem to be commenting on adjustments only for the "key" and "secret" uses that appear as variables, but the commentary within the spec interchangeably use these terms for the same thing. Whatever is used should be applied consistently. It is not reasonable to assume that developers will correctly guess reasons for terminology separations particularly when they are inconsistently enforced.
On your latter point about usage of "keys" vs "secrets" throughout the document — Sorry if I misinterpreted your intent here. If you're seeing places where there are inconsistencies, it would be helpful if you could highlight those more specifically, or ideally send a PR to fix them.
NAME had a call about this today and hashed out a couple of things: We should be clear in general about what we mean by secrets/keys, in that we generally use them interchangeably, but sometimes use one or the other to indicate how a specific value is used The need for secrecy of the value labeled depends on the context in which it is used. So we should provide some text to suggest that there are different ways to use it, and use a fairly neutral name. NAME suggested , which I like a lot. I have attempted to implement these in
The spec currently uses a mix of terminologies for keys/secrets.


Question: 9.6. <del> The main MLS key schedule provides a "resumption_secret" that is used as a PSK to inject entropy from one epoch into another.  This </del> <ins> The main MLS key schedule provides a "resumption_psk" that is used as a PSK to inject entropy from one epoch into another.  This </ins> functionality is used in the reinitialization and branching processes described in reinitialization and sub-group-branching, but may be used by applications for other purposes. Some uses of resumption PSKs might call for the use of PSKs from historical epochs.  The application SHOULD specify an upper limit on <del> the number of past epochs for which the "resumption_secret" may be </del> <ins> the number of past epochs for which the "resumption_psk" may be </ins> stored. 9.7. <del> The main MLS key schedule provides a per-epoch "authentication_secret".  If one of the parties is being actively impersonated by an attacker, their "authentication_secret" will differ from that of the other group members.  Thus, members of a group MAY use their "authentication_secrets" within an out-of-band authentication protocol to ensure that they share the same view of the group. </del> <ins> The main MLS key schedule provides a per-epoch "authentication_code". If one of the parties is being actively impersonated by an attacker, their "authentication_code" will differ from that of the other group members.  Thus, members of a group MAY verify out of band that their "authentication_code" values are all the same, in order to ensure that they share the same view of the group. </ins> 10.

In the key schedule we have the following: senderdatasecret | "sender data" encryptionsecret | "encryption" exportersecret | "exporter" authenticationsecret | "authentication" externalsecret | "external" confirmationkey | "confirm" membershipkey | "membership" resumptionsecret | "resumption" It would be best to provide a coherency to how we use these terms and check for consistency throughout, to avoid confusion for readers: A simple solution would be to use "secret" for all cases, i.e., changing membershipkey to membershipsecret.Another alternative, which may take more refining but could be useful to readers is to separate out terms based on items that are for internal protocol use versus external protocol use. For example, encryption, confirmation, membership, etc., would use "key" and exporter, authentication, etc., could use "secret", or vice versa. --Break-- Ref the mailinglist on confirmation vs authentication secrets, perhaps the naming of these two are a little misleading - authentication normally being within bounds of a protocol while confirmation is external to it. This confusion probably sparked some of that discussion to begin with. It could_ be worth considering switching those terms for clarity.
I actually think we have a fairly coherent taxonomy here: "secret" is something that gets fed into HKDF to derive further things "key" is something that gets fed into another algorithm, e.g., HMAC Looking through the list, the ones that deviate from this pattern are: , which is exported to the application but expected to be kept private , which is exported to the application but not expected to be kept private Given that, I would propose we update the table in question to something like the following, using because that's how it's used, and to follow the "security code" terminology used by applications:
Things are easier to understand like this! The "Purpose" column is quite valuable I think, I remember I was quite overwhelmed when I first read it and had no idea what all those secret / keys were used for. This helps to understand the big picture.