jiebi/CodeConvo huggingface dataset:
Code change:
affects the direction property of the associated "m=" section on future calls to createOffer and createAnswer.  The permitted values for direction are "recvonly", "sendrecv", "sendonly", and "inactive", <del> mirroring the identically named directional attributes defined in </del> <ins> mirroring the identically named direction attributes defined in </ins> RFC4566. When creating offers, the transceiver direction is directly reflected

Related discussion:
text: Several small updates: Issue 15: local media state, local resources correct, removed comment. - Issue 16: updated wording to use 'calling this method', removed comments, - Issue 17: changed stopTransceiver to stop, removed comments, - Issue 18: changed 'directional attribute' to 'direction attribute', removed comments,
LGTM once lands
18) [rfced] Section 4.2.5: We see "direction attribute" but not "directional attribute" in RFC 3264. Will this text be clear to readers? (We ask because we also see "A direction attribute, determined by applying the rules regarding the offered direction specified in [RFC3264], Section 6.1" in Section 5.3.1 of this document.) Original: More specifically, it indicates the [RFC3264] directional attribute of the associated m= section in the last applied answer (including provisional answers), with "send" and "recv" directions reversed if it was a remote answer.
This should indeed be , as that's the terminology used by 3264.
NAME please close this issue once you consider it resolved.
17) [rfced] Section 4.2.2: Will "stopTransceiver" be clear to readers? We could not find this string elsewhere in this document, anywhere else in this cluster of documents, in any published RFC, or in google searches. Original: The stopped property indicates whether the transceiver has been stopped, either by a call to stopTransceiver or by applying an answer that rejects the associated m= section.
This should be , rather than .
NAME please close this issue once you consider it resolved.
16) [rfced] Section 4.1.16: Should "The effects of this method call" be "The effects of calling this method," and should "This call" be "Calling this method"? Original: The effects of this method call depend on when it is invoked, and differ depending on which specific parameters are changed: ... This call may result in a change to the state of the ICE Agent.
SGTM
NAME please close this issue once you consider it resolved.
15) [rfced] Section 4.1.10: Please confirm that "local media state" and "local resources" (as opposed to remote) are correct in the context of setRemoteDescription.
Code change:
affects the direction property of the associated "m=" section on future calls to createOffer and createAnswer.  The permitted values for direction are "recvonly", "sendrecv", "sendonly", and "inactive", <del> mirroring the identically named directional attributes defined in </del> <ins> mirroring the identically named direction attributes defined in </ins> RFC4566. When creating offers, the transceiver direction is directly reflected

Related discussion:
text: (We ask because we see identical wording in Section 4.1.9 ("setLocalDescription").) Original: This API changes the local media state; among other things, it sets up local resources for sending and encoding media.
yes, this is correct
NAME please close this issue once you consider it resolved.
12) [rfced] Section 4.1.8.1: We had trouble with this sentence. If neither suggestion below is correct, please clarify. Original: While this could also be done with a "sendonly" pranswer, followed by a "sendrecv" answer, the initial pranswer leaves the offer-answer exchange open, which means that the caller cannot send an updated offer during this time. Suggestion : While this could also be done with a "sendonly" pranswer, if followed by a "sendrecv" answer the initial pranswer leaves the offer/answer exchange open, which means that the caller cannot send an updated offer during this time. Suggestion : While this could also be done with a "sendonly" pranswer followed by a "sendrecv" answer, the initial pranswer leaves the offer/answer exchange open, which means that the caller cannot send an updated offer during this time.
suggestion seems right
Code change:
Most applications operating over IP networks and requiring latency this low use the Real-time Transport Protocol (RTP) RFC3550 or WebRTC <del> RFC8825, which uses RTP for the media transport as well as several other protocols necessary for safe operation in browsers. </del> <ins> RFC8825, which uses RTP as its Media Transport Protocol, along with several other protocols necessary for safe operation in browsers. </ins> Worth noting is that many applications for ultra low-latency delivery do not need to scale to more than a few users at a time, which

Related discussion:
text: NAME - several things here. we probably don't need all of the descriptive text - the idea is that we'd say "assume roughly this protocol stack, and here's what's not included" if distinguishing between "transport protocol" and "media transport protocol" makes sense, that probably needs to be reflected elsewhere in the document (like, a lot of elsewheres) If this approach works, it could either be in the transport protocol section, or in the one you're thinking of for
I don't think there will be a lot of places. We need to pay attention to only the places where we are explicitly referring to the media transport protocol. The rest can stay the same IMO.
NAME - I couldn't reply to this inline, but I looked through the document (searching for "transport protocol", and you're right. I found one place in Section 4.1, and changed the description of RTP as a transport protocol to RTP as a media transport protocol. NAME - Section 4.1 is before where my explanation is now - if this PR makes sense to you, the explanation probably DOES belong much earlier in the doc, perhaps in .
I'm merging this, based on Ali's review.
We want a paragraph to say something about scope along the lines of: This document is for endpoints delivering streaming media traffic. This can include intermediaries like CDNs as well as content owners and end users, plus their device and player providers. We think this can help address a few issues from last call/iesg review, including: (more to be linked as we notice them)
Consider adding a subsection in the introduction about things that are out of scope, and (for extra credit), why they are out of scope.
from NAME 's review (URL) Section 6.1. UDP and Its Behavior UDP is also used for encrypted tunnels (OpenVPN, Wireguard, etc.). Does encrypted tunneling really have no operational impact on streaming apps other than circuit breakers? (I am thinking about 4K streaming over OpenVPN and the like.)
Not sure if this is really about scope clarification? I think this section is intending to talk about usage as part of a transport protocol, but tunnels are sort of more like a path overlay issue or something (might affect mtu, extra hops with extra things that can go wrong, etc.).
Code change:
Most applications operating over IP networks and requiring latency this low use the Real-time Transport Protocol (RTP) RFC3550 or WebRTC <del> RFC8825, which uses RTP for the media transport as well as several other protocols necessary for safe operation in browsers. </del> <ins> RFC8825, which uses RTP as its Media Transport Protocol, along with several other protocols necessary for safe operation in browsers. </ins> Worth noting is that many applications for ultra low-latency delivery do not need to scale to more than a few users at a time, which

Related discussion:
text: NAME - I'm looking more closely at this issue, and I think text that I put in another (individual) draft may be useful here as well. I'll add it in this section (making the obvious substitutions), and I'll see what NAME and NAME think (after the holiday weekend), and then check back with you. If it works for people, the result will likely be clearer to many readers. Reviewer: Michael Scharf Review result: Ready with Issues This document has been reviewed as part of the transport area review team's ongoing effort to review key IETF documents. These comments were written primarily for the transport area directors, but are copied to the document's authors and WG to allow them to address any issues raised and also to the IETF discussion list for information. When done at the time of IETF Last Call, the authors should consider this review as part of the last-call comments they receive. Please always CC EMAIL if you reply to or forward this review. This informational document surveys network and transport protocol issues that affect quality of experience for streaming applications, in particular video. The overview covers many topics across different technologies. Readers may only be familiar with a subset of these topics and could therefore learn quite a bit from this kind of tutorial. Nonetheless, it remains somewhat unclear what the actual objective of some of the text is. For some topics, quite a bit of technical background is provided, but the discussion is not really comprehensive. In many sections, the document neither derives technical challenges, nor system/protocol requirements, nor practical usage guidance. It is just a kind of tutorial. Quite a bit of text also presents ongoing IETF work, and an RFC with this scope may thus get outdated soon. Section 6.2 deals with topics owned by TCPM. In similar past documents, I have asked for a TCPM presentation prior to an IETF last call in order to ensure that owners of running code are in the loop. I believe this strategy has worked well in the past. Having said this, as TSV-ART reviewer I don't strongly disagree with a publication. All these issues may just be OK for an informational RFC. Some more specific comments: Section 3.2.1. Recognizing Changes from an Expected Baseline This section apparently assumes relatively static path properties, e.g., fixed network connectivity.
Code change:
Most applications operating over IP networks and requiring latency this low use the Real-time Transport Protocol (RTP) RFC3550 or WebRTC <del> RFC8825, which uses RTP for the media transport as well as several other protocols necessary for safe operation in browsers. </del> <ins> RFC8825, which uses RTP as its Media Transport Protocol, along with several other protocols necessary for safe operation in browsers. </ins> Worth noting is that many applications for ultra low-latency delivery do not need to scale to more than a few users at a time, which

Related discussion:
text: It would be more useful to analyze the impact of Wifi and 4G/5G networks in one place. Some of this follows later in sections 3.2.1 and Yet, the split between these three sections is unclear. Having a discussion about today's Internet path characteristics and the dynamics at one place could be more useful. The text could also better distinguish between what matters to endpoints and considerations relevant only for middleboxes (such as passive monitors). Section 3.3 Path Requirements The statement "to find the bandwidth requirements for a router on the delivery path" may assume that the bottleneck on the path will be routers. Yet, on a path the bottlenecks could also be a link layer device (e.g., an Ethernet Switch, a Wifi access point, etc.). RFC 6077 (specifically Section 3.1.3) also explains that issue. A better wording may be "to find the bandwidth requirements for a delivery path". Section 3.6. Unpredictable Usage Profiles / Section 3.7. Extremely Unpredictable Usage Profiles I am not fully convinced by the distinction between "unpredictable" and "extremely unpredictable". To me, these two sections could be merged and maybe also be shortened. More specifically, Section 3.7 lists a lot of statistics from the past. What is IMHO a bit missing in Section 3.7 are actual operational considerations. For instance, are there any lessons learnt for the future? Note that Section 3.6 has a reasonable conclusion at the end. Section 4. Latency Considerations and Section 4.1. Ultra Low-Latency I am surprised by the definition "ultra low-latency (less than 1 second)", as well as some of the other numbers. For other real-time communication use cases, "ultra low-latency" would probably imply a latency requirement of the order of one millisecond. For instance, isochronous traffic for motion control in industrial networks may require a latency of 1 ms, and such "ultra-low latency" requirements are discussed elsewhere, e.g., in the DetNet WG. The terminology in this section should be better explained to readers dealing with networked systems with much harder latency requirements.
Code change:
Most applications operating over IP networks and requiring latency this low use the Real-time Transport Protocol (RTP) RFC3550 or WebRTC <del> RFC8825, which uses RTP for the media transport as well as several other protocols necessary for safe operation in browsers. </del> <ins> RFC8825, which uses RTP as its Media Transport Protocol, along with several other protocols necessary for safe operation in browsers. </ins> Worth noting is that many applications for ultra low-latency delivery do not need to scale to more than a few users at a time, which

Related discussion:
text: And a reference should be added for these definitions in the context of video streaming. Section 5.5.2 Head-of-Line Blocking If Head-of-Line Blocking is indeed a relevant operational problem, it would be useful to add a corresponding reference (e.g., with measurements). Section 5.5.3. Wide and Rapid Variation in Path Capacity The statement "As many end devices have moved to wireless connectivity for the final hop (Wi-Fi, 5G, or LTE), new problems in bandwidth detction have emerged from radio interference and signal strength effects." could be moved to earlier parts of the document. Quite a bit of new computers apparently only have Wifi connectivity and no Ethernet port, i.e., Wifi may just be their default. Also, wireless may not only be used on the last hop, for instance in meshed setups. This may make the problem even harder. Section 6. Evolution of Transport Protocols and Transport Protocol Behaviors I really wonder whether UDP vs. TCP vs. QUIC is actually the relevant distinction. What may actually matter are the transport services provided by the protocol stack (e.g., RFC 8095). I fully agree to a related comment in the INTDIR review. Section 6.1. UDP and Its Behavior UDP is also used for encrypted tunnels (OpenVPN, Wireguard, etc.). Does encrypted tunneling really have no operational impact on streaming apps other than circuit breakers? (I am thinking about 4K streaming over OpenVPN and the like.) Section 6.2. TCP and Its Behavior This text should IMHO be presented and be discussed in TCPM. Personally, I am not convinced that this document is a good place for discussing the long history of TCP congestion control. If the objective of the document is to provide a tutorial, IMHO it would be more useful to briefly explain the state-of-the-art in year 2022. Some further specific comments on TCP congestion control: It is surprising that RFC 5681 is not referenced.793bis has normative statements regarding congestion control, and most stacks are compatible to what is written in 793bis. Several operating systems use CUBIC by default and support standard Reno. Another general good reference for TCP standards is RFC 7414.Does ECN, DCTCP, etc.
Code change:
Most applications operating over IP networks and requiring latency this low use the Real-time Transport Protocol (RTP) RFC3550 or WebRTC <del> RFC8825, which uses RTP for the media transport as well as several other protocols necessary for safe operation in browsers. </del> <ins> RFC8825, which uses RTP as its Media Transport Protocol, along with several other protocols necessary for safe operation in browsers. </ins> Worth noting is that many applications for ultra low-latency delivery do not need to scale to more than a few users at a time, which

Related discussion:
text: really not matter at all, for instance, if a data center hosts video playout servers? Section 6.3. QUIC and Its Behavior As already noted, the discussion about head-of-line-blocking would really benefit from backing by a reference. Section 7. Streaming Encrypted Media As far as I know, streaming users sometimes use encrypted tunnels such as OpenVPN or WireGuard (or IPsec) to access video content. I may miss something, but it is unclear to me how that fits into the categories presented in Section
Code change:
If the EAP-TLS peer authenticates successfully, the EAP-TLS server MUST send an EAP-Request packet with EAP-Type=EAP-TLS containing TLS records conforming to the version of TLS used.  The message <del> flow ends with the EAP-TLS server sending an EAP-Success message. </del> <ins> flow ends with a protected success indication from the EAP-TLS server, follwed by an EAP-Response packet of EAP-Type=EAP-TLS and no data from the EAP-TLS peer, follwed by EAP-Success from the server. </ins> If the EAP-TLS server authenticates successfully, the EAP-TLS peer MUST send an EAP-Response message with EAP-Type=EAP-TLS containing

Related discussion:
text: Hi! I performed a second AD review on draft-ietf-emu-eap-tls13. This time on -18. Thank you to the authors, the responsible co-chair, and the WG for all of the work to redesign the -13 protocol in response to the initial IESG review in December 2020. My review feedback consists of my own minimal additional comments and highlights of where previously stated IESG ballot items do not appear to be responded to or addressed. All items appear to be minor. ** Section 2.4. Previously, this text referenced TLS 1.3 or higher necessitating more ambiguous reference to version numbers. However, now the text only notes TLS 1.3. OLD When EAP-TLS is used with TLS version 1.3, the EAP-TLS peers and EAP- TLS servers MUST comply with the compliance requirements (mandatory- to-implement cipher suites, signature algorithms, key exchange algorithms, extensions, etc.) for the TLS version used. For TLS 1.3 the compliance requirements are defined in Section 9 of [RFC8446]. NEW When EAP-TLS is used with TLS version 1.3, the EAP-TLS peers and EAP-TLS servers MUST comply with the compliance requirements (mandatory- to-implement cipher suites, signature algorithms, key exchange algorithms, extensions, etc.) defined in Section 9 of [RFC8446]. ==[ COMMENT (Ben Kaduk) Section 2.1.4 The two paragraphs below replaces the corresponding paragraphs in Section 2.1.3 of [RFC5216] when EAP-TLS is used with TLS 1.3 or higher. The other paragraphs in Section 2.1.3 of [RFC5216] still apply with the exception that SessionID is deprecated. If the EAP-TLS peer authenticates successfully, the EAP-TLS server MUST send an EAP-Request packet with EAP-Type=EAP-TLS containing TLS records conforming to the version of TLS used. The message flow ends with the EAP-TLS server sending an EAP-Success message.
Code change:
If the EAP-TLS peer authenticates successfully, the EAP-TLS server MUST send an EAP-Request packet with EAP-Type=EAP-TLS containing TLS records conforming to the version of TLS used.  The message <del> flow ends with the EAP-TLS server sending an EAP-Success message. </del> <ins> flow ends with a protected success indication from the EAP-TLS server, follwed by an EAP-Response packet of EAP-Type=EAP-TLS and no data from the EAP-TLS peer, follwed by EAP-Success from the server. </ins> If the EAP-TLS server authenticates successfully, the EAP-TLS peer MUST send an EAP-Response message with EAP-Type=EAP-TLS containing

Related discussion:
text: If the EAP-TLS server authenticates successfully, the EAP-TLS peer MUST send an EAP-Response message with EAP-Type=EAP-TLS containing TLS records conforming to the version of TLS used. Just to walk through the various scenarios here; if the last message of the TLS handshake for a given TLS version is sent by the TLS client, that will go in an EAP-Response, so the server is then able to send EAP-Success directly. On the other hand, if the last TLS handshake message is sent by the server, that will have to go in an EAP-TLS EAP-Request, and so the peer will have to send an empty EAP-TLS response in order for the server to be able to wend EAP-Success? Do we need to have any text here to handle that empty EAP-TLS Eap-Request case? ==[ COMMENT(Alvaro Retana) Is the intention of the Appendix to provide additional formal updates to rfc5216? That is what it looks like to me, but there's no reference to it from the main text. Please either reference the Appendix when talking about some of the other updates (if appropriate) or specifically include the text somewhere more prominent. [Roman] As proposed above, please provide some forward reference. ==[ COMMENT (Éric Vyncke) I find "This section updates Section 2.1.1 of [RFC5216]." a little ambiguous as it the 'updated section' is not identified clearly. I.e., as the sections in RFC 5216 are not too long, why not simply providing whole new sections ? [Roman] I'd propose a simple fix by using (approximate phrases such as) either: This section updates Section xxx of [RFC5216] by amending it with the following text. (for the cases where the intent is to keep the existing text in rfc5216 but add this new text) This section updates Section xxx of [RFC5216] by replacing it with the following text. (for the cases where the intent is to replace an entire section in rfc5216) ==[ COMMENT (Éric Vyncke) None of us are native English speaker, but "e.g." as "i.e."
Code change:
If the EAP-TLS peer authenticates successfully, the EAP-TLS server MUST send an EAP-Request packet with EAP-Type=EAP-TLS containing TLS records conforming to the version of TLS used.  The message <del> flow ends with the EAP-TLS server sending an EAP-Success message. </del> <ins> flow ends with a protected success indication from the EAP-TLS server, follwed by an EAP-Response packet of EAP-Type=EAP-TLS and no data from the EAP-TLS peer, follwed by EAP-Success from the server. </ins> If the EAP-TLS server authenticates successfully, the EAP-TLS peer MUST send an EAP-Response message with EAP-Type=EAP-TLS containing

Related discussion:
text: are usually followed by a comma while "but" has usually no comma before ;-) [Roman] A simple s/e.g./e.g.,/ and s/i.e./i.e.,/ would catch this. Regards, Roman
==[ COMMENT (Éric Vyncke) None of us are native English speaker, but "e.g." as "i.e." are usually followed by a comma while "but" has usually no comma before ;-) [Roman] A simple s/e.g./e.g.,/ and s/i.e./i.e.,/ would catch this. Eric is correct that US english i.e. and e.g. is usually followed by a comma. I corrected this in d9f5028 I also run a word spell check on the document. Word does not suggest removing any comma after "but" but rather insert an additional comma after a but. I think this can be left to the RFC editor. The word spell check caught two other grammar errors (s after plural, no s after singular) RFC 8446 does not use capitalization on post-handshake as that is just a name for a set of messages. I corrected these things in
I Made a new PR
Made a commit with NEW When EAP-TLS is used with TLS version 1.3, the EAP-TLS peers and EAP-TLS servers MUST comply with the compliance requirements (mandatory- to-implement cipher suites, signature algorithms, key exchange algorithms, extensions, etc.) defined in Section 9 of [RFC8446].
Section 2.1.4 Changed the text to: The message flow ends with a protected success indication from the EAP-TLS server, follwed by an EAP-Response packet of EAP-Type=EAP-TLS and no data from the EAP-TLS peer, follwed by EAP-Success from the server.
I added a forward reference to the appendix in the introduction
NAME looks like you have got everything covered? are there any comments from Roman which still need to be addressed? should we submit a new version and ask for IETF last call?
I have tried to do everything. Problem with the updates thing.