jiebi/CodeConvo huggingface dataset:
Endpoints which receive a Capsule with an unknown Capsule Type MUST silently drop that Capsule. <del> Receipt of a CAPSULE HTTP/3 Frame on a stream that is not a client- initiated bidirectional stream MUST be treated as a connection error of type H3_FRAME_UNEXPECTED. CAPSULE frames MUST NOT be sent on a stream before at least one HEADERS frame has been sent on that stream.  This removes the need to buffer capsules when the endpoint needs information from headers to determine how to react to the capsule.  If a CAPSULE frame is received on a stream before a HEADERS frame, the receiver MUST treat this as a connection error of type H3_FRAME_UNEXPECTED. </del> <ins> 4.4. </ins> <del> 4.1. </del> <ins> 4.4.1. </ins> The REGISTER_DATAGRAM_CONTEXT capsule (see iana-types for the value of the capsule type) allows an endpoint to inform its peer of the
8. <del> We can provide DATAGRAM support in HTTP/2 by defining the CAPSULE frame in HTTP/2. We can provide DATAGRAM support in HTTP/1.x by defining its data stream format to a sequence of length-value capsules. TODO: Refactor this document and add definitions for HTTP/1.x and HTTP/2. 9. </del> Since this feature requires sending an HTTP/3 Settings parameter, it "sticks out".  In other words, probing clients can learn whether a server supports this feature.  Implementations that support this
the fact that there are applications using HTTP/3 datagrams enabled on this endpoint. <del> 10. 10.1. This document will request IANA to register the following entry in the "HTTP/3 Frames" registry: </del> <ins> 9. </ins> <del> 10.2. </del> <ins> 9.1. </ins> This document will request IANA to register the following entry in the "HTTP/3 Settings" registry: <del> 10.3. </del> <ins> 9.2. </ins> This document establishes a registry for HTTP capsule type codes. The "HTTP Capsule Types" registry governs a 62-bit space.
5.6. <del> In addition to measurements media players use to guide their segment- by-segment adaptive streaming requests, streaming media providers may also rely on measurements collected from media players to provide analytics that can be used for decisions such as whether the adaptive encoding bitrates in use are the best ones to provide to media players, or whether current media content caching is providing the best experience for viewers.  To that effect, the Consumer Technology Association (CTA) who owns the Web Application Video Ecosystem (WAVE) project has published two important specifications. 5.6.1. </del> <ins> Media players use measurements to guide their segment-by-segment adaptive streaming requests, but may also provide measurements to streaming media providers. In turn, providers may base analytics on these measurements, to guide decisions such as whether adaptive encoding bitrates in use are the best ones to provide to media players, or whether current media content caching is providing the best experience for viewers. To that effect, the Consumer Technology Association (CTA) who owns the Web Application Video Ecosystem (WAVE) project has published two important specifications. CTA-2066: Streaming Quality of Experience Events, Properties and Metrics </ins> CTA-2066 specifies a set of media player events, properties, quality of experience (QoE) metrics and associated terminology for
different algorithms such as loss-based CUBIC RFC8312, delay-based COPA or BBR, or even something completely different. <del> We do have experience with deploying new congestion controllers without melting the Internet (CUBIC is one example), but the point mentioned in tcp-behavior about TCP being implemented in operating system kernels is also different with QUIC.  Although QUIC can be implemented in operating system kernels, one of the design goals when this work was chartered was "QUIC is expected to support rapid, distributed development and testing of features", and to meet this expectation, many implementers have chosen to implement QUIC in user space, outside the operating system kernel, and to even distribute QUIC libraries with their own applications. The decision to deploy a new version of QUIC is relatively uncontrolled, compared to other widely used transport protocols, and this can include new transport behaviors that appear without much notice except to the QUIC endpoints.  At IETF 105, Christian Huitema and Brian Trammell presented a talk on "Congestion Defense in Depth" CDiD, that explored potential concerns about new QUIC congestion controllers being broadly deployed without the testing and instrumentation that current major content providers routinely include.  The sense of the room at IETF 105 was that the current major content providers understood what is at stake when they deploy new congestion controllers, but this presentation, and the related discussion in TSVAREA minutes from IETF 105 (tsvarea-105, are still worth a look for new and rapidly growing content providers. </del> <ins> The Internet community does have experience with deploying new congestion controllers without melting the Internet.  As noted in RFC8312, both the CUBIC congestion controller and its predecessor BIC have significantly different behavior from Reno-style congestion controllers such as TCP NewReno RFC6582, but both CUBIC and BIC were added to the Linux kernel in order to allow experimentation and analysis, and both were then selected as the default TCP congestion controllers in Linux, and both were deployed globally. The point mentioned in tcp-behavior about TCP congestion controllers being implemented in operating system kernels is different with QUIC. Although QUIC can be implemented in operating system kernels, one of the design goals when this work was chartered was "QUIC is expected to support rapid, distributed development and testing of features", and to meet this expectation, many implementers have chosen to implement QUIC in user space, outside the operating system kernel, and to even distribute QUIC libraries with their own applications. It is worth noting that streaming operators using HTTP/3, carried over QUIC, can expect more frequent deployment of new congestion controller behavior than has been the case with HTTP/1 and HTTP/2, carried over TCP. </ins> It is worth considering that if TCP-based HTTP traffic and UDP-based HTTP/3 traffic are allowed to enter operator networks on roughly
to predict usage and provision bandwidth, caching, and other mechanisms to meet the needs of users.  In some cases (sec-predict), this is relatively routine, but in other cases, it is more difficult <del> (sec-unpredict, sec-extreme). </del> <ins> (sec-unpredict). </ins> And as with other parts of the ecosystem, new technology brings new challenges.  For example, with the emergence of ultra-low-latency
still being transmitted to the cache, and while the cache does not yet know the size of the object.  Some of the popular caching systems were designed around cache footprint and had deeply ingrained <del> assumptions about knowing the size of objects being stored, so the change in design requirements in long-established systems caused some errors in production.  Incidents occurred where a transmission error in the connection from the upstream source to the cache could result in the cache holding a truncated segment and transmitting it to the end user's device.  In this case, players rendering the stream often had the video freeze until the player was reset.  In some cases the truncated object was even cached that way and served later to other players, causing continued stalls at the same spot in the video for all players playing the segment delivered from that cache node. </del> <ins> assumptions about knowing the size of objects that are being stored, so the change in design requirements in long-established systems caused some errors in production.  Incidents occurred where a transmission error in the connection from the upstream source to the cache could result in the cache holding a truncated segment and transmitting it to the end user's device.  In this case, players rendering the stream often had the video freeze until the player was reset.  In some cases the truncated object was even cached that way and served later to other players as well, causing continued stalls at the same spot in the video for all players playing the segment delivered from that cache node. </ins> 3.5.
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
6.1. <ins> Within sec-trans, the term "Media Transport Protocol" is used to describe to describe the protocol of interest.  This is easier to understand if the reader assumes a protocol stack that looks something like this: where "Media Format" would be something like an RTP payload format RFC2736 or ISOBMFF ISOBMFF, "Media Transport Protocol" would be something like RTP or HTTP, and "Transport Protocol" would be something like TCP or UDP. Not all possible streaming media applications follow this model, but for the ones that do, it seems useful to have names for the protocol layers between Media Format and Transport Layer It is worth noting explicitly that the Media Transport Protocol and Transport Protocol layers might each include more than one protocol. For example, a Media Transport Protocol might run over HTTP, or over WebTransport, which in turn runs over HTTP. It is worth noting explicitly that more complex network protocol stacks are certainly possible - for instance, packets with this protocol stack may be carried in a tunnel, or in a VPN.  If these environments are present, streaming media operators may need to analyze their effects on applications as well. 6.2. </ins> For most of the history of the Internet, we have trusted UDP-based applications to limit their impact on other users.  One of the strategies used was to use UDP for simple query-response application
Various HTTP versions have been used for media delivery.  HTTP/1.0, HTTP/1.1 and HTTP/2 are carried over TCP, and TCP's transport <del> behavior is described in tcp-behavior.  HTTP/3 is carried over QUIC, and QUIC's transport behavior is described in quic-behavior. </del> <ins> behavior is described in reliable-behavior.  HTTP/3 is carried over QUIC, and QUIC's transport behavior is described in quic-behavior. </ins> Unreliable media delivery using RTP and other UDP-based protocols is <del> also discussed in ultralow, udp-behavior, and hop-by-hop-encrypt, but it is difficult to give general guidance for these applications.  For instance, when packet loss occurs, the most appropriate response may depend on the type of codec being used. </del> <ins> also discussed in ultralow, unreliable-behavior, and hop-by-hop- encrypt, but it is difficult to give general guidance for these applications.  For instance, when packet loss occurs, the most appropriate response may depend on the type of codec being used. </ins> 3.
elapsed time between the acknowledgements for specific TCP segments from a TCP receiver since TCP octet sequence numbers and acknowledgements for those sequence numbers are carried in the clear, <del> even if the TCP payload itself is encrypted.  See tcp-behavior for more information. </del> <ins> even if the TCP payload itself is encrypted.  See reliable-behavior for more information. </ins> As transport protocols evolve to encrypt their transport header fields, one side effect of increasing encryption is the kind of passive monitoring, or even "performance enhancement" (RFC3135) that was possible with the older transport protocols (UDP, described in <del> udp-behavior and TCP, described in tcp-behavior) is no longer possible with newer transport protocols such as QUIC (described in quic-behavior).  The IETF has specified a "latency spin bit" mechanism in Section 17.4 of RFC9000 to allow passive latency </del> <ins> unreliable-behavior and TCP, described in reliable-behavior) is no longer possible with newer transport protocols such as QUIC (described in quic-behavior).  The IETF has specified a "latency spin bit" mechanism in Section 17.4 of RFC9000 to allow passive latency </ins> monitoring from observation points on the network path throughout the duration of a connection, but currently chartered work in the IETF is focusing on endpoint monitoring and reporting, rather than on passive
feedback mechanisms, UDP-based applications that send and receive substantial amounts of information are expected to provide their own feedback mechanisms, and to respond to the feedback the application <del> receives.  This expectation is most recently codified in Best Current Practice RFC8085. </del> <ins> receives.  This expectation is most recently codified as a Best Current Practice RFC8085. </ins> In contrast to adaptive segmented delivery over a reliable transport as described in adapt-deliver, some applications deliver streaming
6.3. <del> The past success of the largely TCP-based Internet is evidence that the mechanisms TCP used to achieve equilibrium quickly, at a point where TCP senders do not interfere with other TCP senders for sustained periods of time (RFC5681), have been largely successful. The Internet has continued to work even when the specific TCP mechanisms used to reach equilibrium changed over time (RFC7414). Because TCP provides a common tool to avoid contention, even when significant TCP-based applications like FTP were largely replaced by other significant TCP-based applications like HTTP, the transport behavior remained safe for the Internet. Modern TCP implementations (I-D.ietf-tcpm-rfc793bis) continue to probe for available bandwidth, and "back off" when a network path is saturated, but may also work to avoid growing queues along network paths, which prevent TCP senders from detecting quickly when a network path becoming saturated.  Congestion control mechanisms such as COPA COPA18 and BBR I-D.cardwell-iccrg-bbr-congestion-control make these decisions based on measured path delays, assuming that if the measured path delay is increasing, the sender is injecting packets onto the network path faster than the network can forward them (or the receiver can accept them) so the sender should adjust its sending rate accordingly. Although common TCP behavior has changed significantly since the days of Jacobson-Karels and RFC2001, even adding new congestion controllers such as CUBIC RFC8312, the common practice of implementing TCP as part of an operating system kernel has acted to limit how quickly TCP behavior can change.  Even with the widespread use of automated operating system update installation on many end- user systems, streaming media providers could have a reasonable expectation that they could understand TCP transport protocol behaviors, and that those behaviors would remain relatively stable in the short term. 6.4. </del> The QUIC protocol, developed from a proprietary protocol into an IETF standards-track protocol RFC9000, turns many of the statements made <del> in udp-behavior and tcp-behavior on their heads. </del> <ins> in reliable-behavior and unreliable-behavior on their heads. </ins> Although QUIC provides an alternative to the TCP and UDP transport protocols, QUIC is itself encapsulated in UDP.  As noted elsewhere in
While QUIC is designed as a general-purpose transport protocol, and can carry different application-layer protocols, the current <del> standardized mapping is for HTTP/3 I-D.ietf-quic-http, which describes how QUIC transport features are used for HTTP.  The convention is for HTTP/3 to run over UDP port 443 Port443 but this is not a strict requirement. </del> <ins> standardized mapping is for HTTP/3 RFC9114, which describes how QUIC transport services are used for HTTP.  The convention is for HTTP/3 to run over UDP port 443 Port443 but this is not a strict requirement. </ins> When HTTP/3 is encapsulated in QUIC, which is then encapsulated in UDP, streaming operators (and network operators) might see UDP <del> traffic patterns that are similar to HTTP(S) over TCP.  Since earlier versions of HTTP(S) rely on TCP, UDP ports may be blocked for any port numbers that are not commonly used, such as UDP 53 for DNS. Even when UDP ports are not blocked and HTTP/3 can flow, streaming operators (and network operators) may severely rate-limit this traffic because they do not expect to see legitimate high-bandwidth traffic such as streaming media over the UDP ports that HTTP/3 is using. </del> <ins> traffic patterns that are similar to HTTP(S) over TCP.  UDP ports may be blocked for any port numbers that are not commonly used, such as UDP 53 for DNS.  Even when UDP ports are not blocked and QUIC packets can flow, streaming operators (and network operators) may severely rate-limit this traffic because they do not expect to see legitimate high-bandwidth traffic such as streaming media over the UDP ports that HTTP/3 is using. </ins> As noted in hol-blocking, because TCP provides a reliable, in-order delivery service for applications, any packet loss for a TCP
both were then selected as the default TCP congestion controllers in Linux, and both were deployed globally. <del> The point mentioned in tcp-behavior about TCP congestion controllers being implemented in operating system kernels is different with QUIC. Although QUIC can be implemented in operating system kernels, one of the design goals when this work was chartered was "QUIC is expected to support rapid, distributed development and testing of features," and to meet this expectation, many implementers have chosen to implement QUIC in user space, outside the operating system kernel, and to even distribute QUIC libraries with their own applications. It is worth noting that streaming operators using HTTP/3, carried over QUIC, can expect more frequent deployment of new congestion controller behavior than has been the case with HTTP/1 and HTTP/2, carried over TCP. </del> <ins> The point mentioned in reliable-behavior about TCP congestion controllers being implemented in operating system kernels is different with QUIC.  Although QUIC can be implemented in operating system kernels, one of the design goals when this work was chartered was "QUIC is expected to support rapid, distributed development and testing of features," and to meet this expectation, many implementers have chosen to implement QUIC in user space, outside the operating system kernel, and to even distribute QUIC libraries with their own applications.  It is worth noting that streaming operators using HTTP/3, carried over QUIC, can expect more frequent deployment of new congestion controller behavior than has been the case with HTTP/1 and HTTP/2, carried over TCP. </ins> It is worth considering that if TCP-based HTTP traffic and UDP-based HTTP/3 traffic are allowed to enter operator networks on roughly
certainly as examining decrypted traffic. Because HTTPS has historically layered HTTP on top of TLS, which is <del> in turn layered on top of TCP, intermediaries do have access to unencrypted TCP-level transport information, such as retransmissions, and some carriers exploited this information in attempts to improve transport-layer performance RFC3135.  The most recent standardized version of HTTPS, HTTP/3 I-D.ietf-quic-http, uses the QUIC protocol RFC9000 as its transport layer.  QUIC relies on the TLS 1.3 initial handshake RFC8446 only for key exchange RFC9001, and encrypts almost all transport parameters itself except for a few invariant header fields.  In the QUIC short header, the only transport-level parameter which is sent "in the clear" is the Destination Connection ID RFC8999, and even in the QUIC long header, the only transport-level parameters sent "in the clear" are the Version, Destination </del> <ins> in turn layered on top of TCP, intermediaries have historically had access to unencrypted TCP-level transport information, such as retransmissions, and some carriers exploited this information in attempts to improve transport-layer performance RFC3135.  The most recent standardized version of HTTPS, HTTP/3 RFC9114, uses the QUIC protocol RFC9000 as its transport layer.  QUIC relies on the TLS 1.3 initial handshake RFC8446 only for key exchange RFC9001, and encrypts almost all transport parameters itself except for a few invariant header fields.  In the QUIC short header, the only transport-level parameter which is sent "in the clear" is the Destination Connection ID RFC8999, and even in the QUIC long header, the only transport- level parameters sent "in the clear" are the Version, Destination </ins> Connection ID, and Source Connection ID.  For these reasons, HTTP/3 is significantly more "opaque" than HTTPS with HTTP/1 or HTTP/2.
4.1. <del> (To the editor: check this repository URL after the draft is adopted. The working group may create its own repository) </del> <ins> This document is in the Github repository at: </ins> <del> This document is in the Github repository at https://github.com/GrumpyOldTroll/ietf-mops-drafts.  Readers are welcome to open issues and send pull requests for this document. </del> <ins> https://github.com/ietf-wg-mops/draft-ietf-mops-streaming-opcons Readers are welcome to open issues and send pull requests for this document. </ins> Substantial discussion of this document should take place on the MOPS working group mailing list (mops@ietf.org). <ins> Join: https://www.ietf.org/mailman/listinfo/mops Search: https://mailarchive.ietf.org/arch/browse/mops/ </ins> 4.2. Presentations: IETF 105 BOF: https://www.youtube.com/watch?v=4G3YBVmn9Eo&t=47m21s <del> IETF 106 Mops meeting: https://www.youtube.com/ </del> <ins> IETF 106 meeting: https://www.youtube.com/ </ins> watch?v=4_k340xT2jM&t=7m23s <del> List archive search for doc mentions: https://mailarchive.ietf.org/arch/browse/mops/?q=draft-ietf-mops- streaming-opcons </del> 5. This document requires no actions from IANA.
Internet Transport Protocols" describes the impact of increased encryption of transport headers in general terms. <ins> It is also worth noting that considerations for heavily-encrypted transport protocols also come into play when streaming media is carried over IP-level VPNs and tunnels, with the additional consideration that an intermediary that does not possess credentials allowing decryption will not have visibility to the source and destination IP addresses of the packets being carried inside the tunnel. </ins> 7.2. Although the IETF has put considerable emphasis on end-to-end
On-demand media streaming refers to the playback of pre-recorded media based on a user's action.  In some cases, on-demand media is produced as a by-product of a live media production, using the same <del> segments as the live event, but freezing the manifest after the live event has finished.  In other cases, on-demand media is constructed out of pre-recorded assets with no streaming necessarily involved during the production of the on-demand content. </del> <ins> segments as the live event, but freezing the manifest that describes the media available from the media server after the live event has finished.  In other cases, on-demand media is constructed out of pre- recorded assets with no streaming necessarily involved during the production of the on-demand content. </ins> On-demand media generally is not subject to latency concerns, but other timing-related considerations can still be as important or even
Media servers can provide media streams at various bitrates because the media has been encoded at various bitrates.  This is a so-called "ladder" of bitrates that can be offered to media players as part of <del> the manifest that describes the media being requested by the media player, so that the media player can select among the available </del> <ins> the manifest, so that the media player can select among the available </ins> bitrate choices. The media server may also choose to alter which bitrates are made