--- 1/draft-ietf-taps-transports-03.txt 2015-05-26 06:15:19.200445288 -0700 +++ 2/draft-ietf-taps-transports-04.txt 2015-05-26 06:15:19.264446856 -0700 @@ -1,21 +1,21 @@ Network Working Group G. Fairhurst, Ed. Internet-Draft University of Aberdeen Intended status: Informational B. Trammell, Ed. -Expires: August 31, 2015 M. Kuehlewind, Ed. +Expires: November 28, 2015 M. Kuehlewind, Ed. ETH Zurich - February 27, 2015 + May 27, 2015 Services provided by IETF transport protocols and congestion control mechanisms - draft-ietf-taps-transports-03 + draft-ietf-taps-transports-04 Abstract This document describes services provided by existing IETF protocols and congestion control mechanisms. It is designed to help application and network stack programmers and to inform the work of the IETF TAPS Working Group. Status of This Memo @@ -25,84 +25,135 @@ Internet-Drafts are working documents of the Internet Engineering Task Force (IETF). Note that other groups may also distribute working documents as Internet-Drafts. The list of current Internet- Drafts is at http://datatracker.ietf.org/drafts/current/. Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as "work in progress." - This Internet-Draft will expire on August 31, 2015. + This Internet-Draft will expire on November 28, 2015. Copyright Notice Copyright (c) 2015 IETF Trust and the persons identified as the document authors. All rights reserved. This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents (http://trustee.ietf.org/license-info) in effect on the date of publication of this document. Please review these documents carefully, as they describe your rights and restrictions with respect to this document. Code Components extracted from this document must include Simplified BSD License text as described in Section 4.e of the Trust Legal Provisions and are provided without warranty as described in the Simplified BSD License. +Table of Contents + + 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3 + 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4 + 3. Existing Transport Protocols . . . . . . . . . . . . . . . . 4 + 3.1. Transport Control Protocol (TCP) . . . . . . . . . . . . 4 + 3.1.1. Protocol Description . . . . . . . . . . . . . . . . 5 + 3.1.2. Interface description . . . . . . . . . . . . . . . . 6 + 3.1.3. Transport Protocol Components . . . . . . . . . . . . 6 + 3.2. Multipath TCP (MP-TCP) . . . . . . . . . . . . . . . . . 7 + 3.3. Stream Control Transmission Protocol (SCTP) . . . . . . . 7 + 3.3.1. Protocol Description . . . . . . . . . . . . . . . . 8 + 3.3.2. Interface Description . . . . . . . . . . . . . . . . 10 + 3.3.3. Transport Protocol Components . . . . . . . . . . . . 11 + 3.4. User Datagram Protocol (UDP) . . . . . . . . . . . . . . 12 + 3.4.1. Protocol Description . . . . . . . . . . . . . . . . 12 + 3.4.2. Interface Description . . . . . . . . . . . . . . . . 13 + 3.4.3. Transport Protocol Components . . . . . . . . . . . . 13 + 3.5. Lightweight User Datagram Protocol (UDP-Lite) . . . . . . 14 + 3.5.1. Protocol Description . . . . . . . . . . . . . . . . 14 + 3.5.2. Interface Description . . . . . . . . . . . . . . . . 15 + 3.5.3. Transport Protocol Components . . . . . . . . . . . . 15 + 3.6. Datagram Congestion Control Protocol (DCCP) . . . . . . . 15 + 3.6.1. Protocol Description . . . . . . . . . . . . . . . . 16 + 3.6.2. Interface Description . . . . . . . . . . . . . . . . 17 + 3.6.3. Transport Protocol Components . . . . . . . . . . . . 17 + 3.7. Realtime Transport Protocol (RTP) . . . . . . . . . . . . 18 + 3.8. NACK-Oriented Reliable Multicast (NORM) . . . . . . . . . 18 + 3.8.1. Protocol Description . . . . . . . . . . . . . . . . 18 + 3.8.2. Interface Description . . . . . . . . . . . . . . . . 19 + 3.8.3. Transport Protocol Components . . . . . . . . . . . . 20 + 3.9. Transport Layer Security (TLS) and Datagram TLS (DTLS) as + a pseudotransport . . . . . . . . . . . . . . . . . . . . 20 + 3.9.1. Protocol Description . . . . . . . . . . . . . . . . 21 + 3.9.2. Interface Description . . . . . . . . . . . . . . . . 21 + 3.9.3. Transport Protocol Components . . . . . . . . . . . . 21 + 3.10. Hypertext Transport Protocol (HTTP) over TCP as a + pseudotransport . . . . . . . . . . . . . . . . . . . . . 21 + 3.10.1. Protocol Description . . . . . . . . . . . . . . . . 21 + 3.10.2. Interface Description . . . . . . . . . . . . . . . 22 + 3.10.3. Transport Protocol Components . . . . . . . . . . . 23 + 3.11. WebSockets . . . . . . . . . . . . . . . . . . . . . . . 23 + 3.11.1. Protocol Description . . . . . . . . . . . . . . . . 23 + 3.11.2. Interface Description . . . . . . . . . . . . . . . 24 + 3.11.3. Transport Protocol Components . . . . . . . . . . . 24 + + 4. Transport Service Features . . . . . . . . . . . . . . . . . 24 + 4.1. Complete Protocol Feature Matrix . . . . . . . . . . . . 26 + 5. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 28 + 6. Security Considerations . . . . . . . . . . . . . . . . . . . 28 + 7. Contributors . . . . . . . . . . . . . . . . . . . . . . . . 28 + 8. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 28 + 9. References . . . . . . . . . . . . . . . . . . . . . . . . . 28 + 9.1. Normative References . . . . . . . . . . . . . . . . . . 28 + 9.2. Informative References . . . . . . . . . . . . . . . . . 29 + Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 34 + 1. Introduction Most Internet applications make use of the Transport Services provided by TCP (a reliable, in-order stream protocol) or UDP (an unreliable datagram protocol). We use the term "Transport Service" to mean the end-to-end service provided to an application by the transport layer. That service can only be provided correctly if information about the intended usage is supplied from the application. The application may determine this information at design time, compile time, or run time, and may include guidance on whether a feature is required, a preference by the application, or something in between. Examples of features of Transport Services are reliable delivery, ordered delivery, content privacy to in-path devices, integrity protection, and minimal latency. The IETF has defined a wide variety of transport protocols beyond TCP - and UDP, including TCP, SCTP, DCCP, MP-TCP, and UDP-Lite. Transport + and UDP, including SCTP, DCCP, MP-TCP, and UDP-Lite. Transport services may be provided directly by these transport protocols, or layered on top of them using protocols such as WebSockets (which runs - over TCP) or RTP (over TCP or UDP). Services built on top of UDP or - UDP-Lite typically also need to specify additional mechanisms, + over TCP), RTP (over TCP or UDP) or WebRTC data channels (which run + over SCTP over DTLS over UDP or TCP). Services built on top of UDP + or UDP-Lite typically also need to specify additional mechanisms, including a congestion control mechanism (such as a windowed congestion control, TFRC or LEDBAT congestion control mechanism). This extends the set of available Transport Services beyond those provided to applications by TCP and UDP. Transport protocols can also be differentiated by the features of the services they provide: for instance, SCTP offers a message-based - service that does not suffer head-of-line blocking when used with - multiple stream, because it can accept blocks of data out of order, - UDP-Lite provides partial integrity protection, and LEDBAT can - provide low-priority "scavenger" communication. + service providing full or partial reliability and allowing to + minimize the head of line blocking due to the support of unordered + and unordered message delivery within multiple streams, UDP-Lite + provides partial integrity protection, and LEDBAT can provide low- + priority "scavenger" communication. 2. Terminology The following terms are defined throughout this document, and in subsequent documents produced by TAPS describing the composition and decomposition of transport services. - [NOTE: The terminology below was presented at the TAPS WG meeting in - Honolulu. While the factoring of the terminology seems - uncontroversial, there may be some entities which still require names - (e.g. information about the interface between the transport and lower - layers which could lead to the availability or unavailability of - certain transport protocol features). Comments are welcome via the - TAPS mailing list.] - Transport Service Feature: a specific end-to-end feature that a transport service provides to its clients. Examples include confidentiality, reliable delivery, ordered delivery, message- versus-stream orientation, etc. Transport Service: a set of transport service features, without an association to any given framing protocol, which provides a complete service to an application. Transport Protocol: an implementation that provides one or more @@ -246,21 +297,22 @@ [EDITOR'S NOTE: discussion of how to map this to features and TAPS: what does the higher layer need to decide? what can the transport layer decide based on global settings? what must the transport layer decide based on network characteristics?] 3.2. Multipath TCP (MP-TCP) [EDITOR'S NOTE: a few sentences describing Multipath TCP [RFC6824] go here. Note that this adds transport-layer multihoming to the - components TCP provides] + components TCP provides. Simone Ferlin-Oliveira will contribute text + for this section.] 3.3. Stream Control Transmission Protocol (SCTP) SCTP is a message oriented standards track transport protocol and the base protocol is specified in [RFC4960]. It supports multi-homing to handle path failures. An SCTP association has multiple unidirectional streams in each direction and provides in-sequence delivery of user messages only within each stream. This allows to minimize head of line blocking. SCTP is extensible and the currently defined extensions include mechanisms for dynamic re-configurations @@ -292,85 +344,96 @@ SCTP has been designed with extensibility in mind. Each SCTP packet starts with a single common header containing the port numbers, a verification tag and the CRC32c checksum. This common header is followed by a sequence of chunks. Each chunk consists of a type field, flags, a length field and a value. [RFC4960] defines how a receiver processes chunks with an unknown chunk type. The support of extensions can be negotiated during the SCTP handshake. SCTP provides a message-oriented service. Multiple small user messages can be bundled into a single SCTP packet to improve the - efficiency. User messages which would result in IP packets larger - than the MTU will be fragmented at the sender side and reassembled at - the receiver side. There is no protocol limit on the user message - size. [RFC4821] defines a method to perform packetization layer path - MTU discovery with probe packets using the padding chunks defined the - [RFC4820]. + efficiency. For example, this bundling may be done by delaying user + messages at the sender side similar to the Nagle algorithm used by + TCP. User messages which would result in IP packets larger than the + MTU will be fragmented at the sender side and reassembled at the + receiver side. There is no protocol limit on the user message size. + ICMP-based path MTU discovery as specified for IPv4 in [RFC1191] and + for IPv6 in [RFC1981] as well as packetization layer path MTU + discovery as specified in [RFC4821] with probe packets using the + padding chunks defined the [RFC4820] are supported. [RFC4960] specifies a TCP friendly congestion control to protect the network against overload. SCTP also uses a sliding window flow control to protect receivers against overflow. Each SCTP association has between 1 and 65536 uni-directional streams in each direction. The number of streams can be different in each direction. Every user-message is sent on a particular stream. User - messages can be sent ordered or un-ordered upon request by the upper - layer. Only all ordered messages sent on the same stream are - delivered at the receiver in the same order as sent by the sender. - For user messages not requiring fragmentation, this minimises head of - line blocking. The base protocol defined in [RFC4960] doesn't allow - interleaving of user-messages, which results in sending a large - message on one stream can block the sending of user messages on other - streams. [I-D.ietf-tsvwg-sctp-ndata] overcomes this limitation and - also allows to specify a scheduler for the sender side streams - selection. The stream re-configuration extension defined in - [RFC6525] allows to reset streams during the lifetime of an - association and to increase the number of streams, if the number of - streams negotiated in the SCTP handshake is not sufficient. + messages can be sent un-ordered or ordered upon request by the upper + layer. Un-ordered messages can be delivered as soon as they are + completely received. Only all ordered messages sent on the same + stream are delivered at the receiver in the same order as sent by the + sender. For user messages not requiring fragmentation, this + minimises head of line blocking. The base protocol defined in + [RFC4960] doesn't allow interleaving of user-messages, which results + in sending a large message on one stream can block the sending of + user messages on other streams. [I-D.ietf-tsvwg-sctp-ndata] + overcomes this limitation. Furthermore, [I-D.ietf-tsvwg-sctp-ndata] + specifies multiple algorithms for the sender side selection of which + streams to send data from supporting a variety of scheduling + algorithms including priority based ones. The stream re- + configuration extension defined in [RFC6525] allows to reset streams + during the lifetime of an association and to increase the number of + streams, if the number of streams negotiated in the SCTP handshake is + not sufficient. According to [RFC4960], each user message sent is either delivered to the receiver or, in case of excessive retransmissions, the association is terminated in a non-graceful way, similar to the TCP behaviour. In addition to this reliable transfer, the partial reliability extension defined in [RFC3758] allows the sender to abandon user messages. The application can specify the policy for abandoning user messages. Examples for these policies include: o Limiting the time a user message is dealt with by the sender. o Limiting the number of retransmissions for each fragment of a user - message. + message. If the number of retransmissions is limited to 0, one + gets a service similar to UDP. o Abandoning messages of lower priority in case of a send buffer shortage. SCTP supports multi-homing. Each SCTP end-point uses a list of IP- addresses and a single port number. These addresses can be any mixture of IPv4 and IPv6 addresses. These addresses are negotiated during the handshake and the address re-configuration extension - specified in [RFC5061] can be used to change these addresses during - the livetime of an SCTP association. This allows for transport layer - mobility. Multiple addresses are used for improved resilience. If a - remote address becomes unreachable, the traffic is switched over to a + specified in [RFC5061] in combination with [RFC4895] can be used to + change these addresses in an authenticated way during the livetime of + an SCTP association. This allows for transport layer mobility. + Multiple addresses are used for improved resilience. If a remote + address becomes unreachable, the traffic is switched over to a reachable one, if one exists. Each SCTP end-point supervises continuously the reachability of all peer addresses using a heartbeat mechanism. For securing user messages, the use of TLS over SCTP has been specified in [RFC3436]. However, this solution does not support all services provided by SCTP (for example un-ordered delivery or partial reliability), and therefore the use of DTLS over SCTP has been specified in [RFC6083] to overcome these limitations. When using DTLS over SCTP, the application can use almost all services provided by SCTP. - For legacy NAT traversal, [RFC6951] defines the UDP encapsulation of + [I-D.ietf-tsvwg-natsupp] defines a methods for end-hosts and + middleboxes to provide for NAT support for SCTP over IPv4. For + legacy NAT traversal, [RFC6951] defines the UDP encapsulation of SCTP-packets. Alternatively, SCTP packets can be encapsulated in DTLS packets as specified in [I-D.ietf-tsvwg-sctp-dtls-encaps]. The latter encapsulation is used with in the WebRTC context. Having a well defined API is also a feature provided by SCTP as described in the next subsection. 3.3.2. Interface Description [RFC4960] defines an abstract API for the base protocol. An @@ -444,55 +507,53 @@ o connection setup with feature negotiation and application-to-port mapping o port multiplexing o reliable or partially reliable delivery o ordered and unordered delivery within a stream - o support for multiple prioritised streams + o support for multiple concurrent streams - o flow control (slow receiver function) + o support for stream scheduling prioritization - o message-oriented delivery + o flow control + o message-oriented delivery o congestion control - o application PDU bundling + o user message bundling - o application PDU fragmentation and reassembly + o user message fragmentation and reassembly - o integrity check + o strong error detection (CRC32C) o transport layer multihoming for resilience o transport layer mobility [EDITOR'S NOTE: update this list.] 3.4. User Datagram Protocol (UDP) The User Datagram Protocol (UDP) [RFC0768] [RFC2460] is an IETF standards track transport protocol. It provides a uni-directional, datagram protocol which preserves message boundaries. It provides none of the following transport features: error correction, congestion control, or flow control. It can be used to send broadcast datagrams (IPv4) or multicast datagrams (IPv4 and IPv6), in addition to unicast (and anycast) datagrams. IETF guidance on the use of UDP is provided in[RFC5405]. UDP is widely implemented and widely used by common applications, especially DNS. - [EDITOR'S NOTE: Kevin Fall signed up as a contributor for this - section.] - 3.4.1. Protocol Description UDP is a connection-less protocol which maintains message boundaries, with no connection setup or feature negotiation. The protocol uses independent messages, ordinarily called datagrams. The lack of error control and flow control implies messages may be damaged, re-ordered, lost, or duplicated in transit. A receiving application unable to run sufficiently fast or frequently may miss messages. The lack of congestion handling implies UDP traffic may cause the loss of messages from other protocols (e.g., TCP) when sharing the same @@ -564,23 +624,20 @@ o checksum optional 3.5. Lightweight User Datagram Protocol (UDP-Lite) The Lightweight User Datagram Protocol (UDP-Lite) [RFC3828] is an IETF standards track transport protocol. UDP-Lite provides a bidirectional set of logical unicast or multicast message streams over a datagram protocol. IETF guidance on the use of UDP-Lite is provided in [RFC5405]. - [EDITOR'S NOTE: Gorry Fairhurst signed up as a contributor for this - section.] - 3.5.1. Protocol Description UDP-Lite is a connection-less datagram protocol, with no connection setup or feature negotiation. The protocol use messages, rather than a byte-stream. Each stream of messages is independently managed, therefore retransmission does not hold back data sent using other logical streams. It provides multiplexing to multiple sockets on each host using port numbers. An active UDP-Lite session is identified by its four-tuple @@ -759,60 +816,391 @@ 3.7. Realtime Transport Protocol (RTP) RTP provides an end-to-end network transport service, suitable for applications transmitting real-time data, such as audio, video or data, over multicast or unicast network services, including TCP, UDP, UDP-Lite, DCCP. [EDITOR'S NOTE: Varun Singh signed up as contributor for this section.] -3.8. Transport Layer Security (TLS) and Datagram TLS (DTLS) as a - - pseudo transport +3.8. NACK-Oriented Reliable Multicast (NORM) - [NOTE: A few words on TLS [RFC5246] and DTLS [RFC6347] here, and how - they get used by other protocols to meet security goals as an add-on - interlayer above transport.] + NORM is an IETF standards track protocol specified in [RFC5740]. The + protocol was designed to support reliable bulk data dissemination to + receiver groups using IP Multicast but also provides for point-to- + point unicast operation. Its support for bulk data dissemination + includes discrete file or computer memory-based "objects" as well as + byte- and message-streaming. NORM is designed to incorporate packet + erasure coding as an inherent part of its selective ARQ in response + to receiver negative acknowledgements. The packet erasure coding can + also be proactively applied for forward protection from packet loss. + NORM transmissions are governed by TCP-friendly congestion control. + NORM's reliability, congestion control, and flow control mechanism + are distinct components and can be separately controlled to meet + different application needs. 3.8.1. Protocol Description + [EDITOR'S NOTE: needs to be more clear about the application of FEC + and packet erasure coding; expand ARQ.] + + The NORM protocol is encapsulated in UDP datagrams and thus provides + multiplexing for multiple sockets on hosts using port numbers. For + purposes of loosely coordinated IP Multicast, NORM is not strictly + connection-oriented although per-sender state is maintained by + receivers for protocol operation. [RFC5740] does not specify a + handshake protocol for connection establishment and separate session + initiation can be used to coordinate port numbers. However, in-band + "client-server" style connection establishment can be accomplished + with the NORM congestion control signaling messages using port + binding techniques like those for TCP client-server connections. + + NORM supports bulk "objects" such as file or in-memory content but + also can treat a stream of data as a logical bulk object for purposes + of packet erasure coding. In the case of stream transport, NORM can + support either byte streams or message streams where application- + defined message boundary information is carried in the NORM protocol + messages. This allows the receiver(s) to join/re-join and recover + message boundaries mid-stream as needed. Application content is + carried and identified by the NORM protocol with encoding symbol + identifiers depending upon the Forward Error Correction (FEC) Scheme + [RFC3452] configured. NORM uses NACK-based selective ARQ to reliably + deliver the application content to the receiver(s). NORM proactively + measures round-trip timing information to scale ARQ timers + appropriately and to support congestion control. For multicast + operation, timer-based feedback suppression is uses to achieve group + size scaling with low feedback traffic levels. The feedback + suppression is not applied for unicast operation. + + NORM uses rate-based congestion control based upon the TCP-Friendly + Rate Control (TFRC) [RFC4324] principles that are also used in DCCP + [RFC4340]. NORM uses control messages to measure RTT and collect + congestion event (e..g, loss event, ECN event, etc) information from + the receiver(s) to support dynamic rate control adjustment. The TCP- + Friendly Multicast Congestion Control (TFMCC) [RFC4654] used provides + some extra features to support multicast but is functionally + equivalent to TFRC in the unicast case. + + NORM's reliability mechanism is decoupled from congestion control. + This allows alternative arrangements of transport services to be + invoked. For example, fixed-rate reliable delivery can be supported + or unreliable (but optionally "better than best effort" via packet + erasure coding) delivery with rate-control per TFRC can be achieved. + Additionally, alternative congestion control techniques may be + applied. For example, TFRC rate control with congestion event + detection based on ECN for links with high packet loss (e.g., + wireless) has been implemented and demonstrated with NORM. + + While NORM is NACK-based for reliability transfer, it also supports a + positive acknowledgment (ACK) mechanism that can be used for receiver + flow control. Again, since this mechanism is decoupled from the + reliability and congestion control, applications that have different + needs in this aspect can use the protocol differently. One example + is the use of NORM for quasi-reliable delivery where timely delivery + of newer content may be favored over completely reliable delivery of + older content within buffering and RTT constraints. + 3.8.2. Interface Description + The NORM specification does not describe a specific application + programming interface (API) to control protocol operation. A freely- + available, open source reference implementation of NORM is available + at https://www.nrl.navy.mil/itd/ncs/products/norm, and a documented + API is provided for this implementation. While a sockets-like API is + not currently documented, the existing API supports the necessary + functions for that to be implemented. + 3.8.3. Transport Protocol Components -3.9. Hypertext Transport Protocol (HTTP) as a pseudotransport + The transport protocol components provided by NORM are: - [RFC3205] + o unicast - [EDITOR'S NOTE: No identified contributor for this section yet.] + o multicast + + o port multiplexing (UDP ports) + + o reliable delivery + + o ordered delivery for each byte or message stream + + o unordered delivery of in-memory data or file bulk content objects + + o error detection (UDP checksum) + + o segmentation + + o stream-oriented delivery in a single stream + + o object-oriented delivery of discrete data or file items + + o data bundling (Nagle's algorithm) + + o flow control (timer-based and/or ack-based) + + o congestion control + + o packet erasure coding (both proactively and as part of ARQ) + +3.9. Transport Layer Security (TLS) and Datagram TLS (DTLS) as a + pseudotransport + + [NOTE: A few words on TLS [RFC5246] and DTLS [RFC6347] here, and how + they get used by other protocols to meet security goals as an add-on + interlayer above transport. Kevin Fall will contribute text to this + section.] 3.9.1. Protocol Description 3.9.2. Interface Description 3.9.3. Transport Protocol Components -3.10. WebSockets +3.10. Hypertext Transport Protocol (HTTP) over TCP as a pseudotransport - [RFC6455] + Hypertext Transfer Protocol (HTTP) is an application-level protocol + widely used on the Internet. Version 1.1 of the protocol is + specified in [RFC7230] [RFC7231] [RFC7232] [RFC7233] [RFC7234] + [RFC7235], and version 2 in [RFC7540]. Furthermore, HTTP is used as + a substrate for other application-layer protocols. There are various + reasons for this practice listed in [RFC3205]; these include being a + well-known and well-understood protocol, reusability of existing + servers and client libraries, easy use of existing security + mechanisms such as HTTP digest authentication [RFC2617] and TLS + [RFC5246], the ability of HTTP to traverse firewalls which makes it + work with a lot of infrastructure, and cases where a application + server often needs to support HTTP anyway. - [EDITOR'S NOTE: No identified contributor for this section yet.] + Depending on application's needs, the use of HTTP as a substrate + protocol may add complexity and overhead in comparison to a special- + purpose protocol (e.g. HTTP headers, suitability of the HTTP + security model etc.). [RFC3205] address this issues and provides + some guidelines and concerns about the use of HTTP standard port 80 + and 443, the use of HTTP URL scheme and interaction with existing + firewalls, proxies and NATs. Also, though not strictly bound to TCP, + HTTP is almost exclusively run over TCP, and therefore inherits its + properties when used in this way. This can have disadvantages, when + the application does not naturally require single-streamed, reliable + transport. 3.10.1. Protocol Description + Hypertext Transfer Protocol (HTTP) is a request/response protocol. A + client sends a request containing a request method, URI and protocol + version followed by a MIME-like message (see [RFC7231] for the + differences between an HTTP object and a MIME message), containing + information about the client and request modifiers. The message can + contain a message body carrying application data as well. The server + responds with a status or error code followed by a MIME-like message + containing information about the server and information about carried + data and it can include a message body. It is possible to specify a + data format for the message body using MIME media types [RFC2045]. + Furthermore, the protocol has numerous additional features; features + relevant to pseudotransport are described below. + + Content negotiation, specified in [RFC7231], is a mechanism provided + by HTTP for selecting a representation on a requested resource. The + client and server negotiate acceptable data formats, charsets, data + encoding (e.g. data can be transferred compressed, gzip), etc. HTTP + can accommodate exchange of messages as well as data streaming (using + chunked transfer encoding [RFC7230]). It is also possible to request + a part of a resource using range requests specified in [RFC7233]. + The protocol provides powerful cache control signalling defined in + [RFC7234]. + + HTTP 1.1's and HTTP 2.0's persistent connections can be use to + perform multiple request-response transactions during the life-time + of a single HTTP connection. Moreover, HTTP 2.0 connections can + multiplex many request/response pairs in parallel on a single + connection. This reduces connection establishment overhead and the + effect of TCP slow-start on each transaction, important for HTTP's + primary use case. + + It is possible to combine HTTP with security mechanisms, like TLS + (denoted by HTTPS), which adds protocol properties provided by such a + mechanism (e.g. authentication, encryption, etc.). TLS's + Application-Layer Protocol Negotiation (ALPN) extension [RFC7301] can + be used for HTTP version negotiation within TLS handshake which + eliminates addition round-trip. Arbitrary cookie strings, included + as part of the MIME headers, are often used as bearer tokens in HTTP. + + Application layer protocols using HTTP as substrate may use existing + method and data formats, or specify new methods and data formats. + Furthermore some protocols may not fit a request/response paradigm + and instead rely on HTTP to send messages (e.g. [RFC6546]). Because + HTTP is working in many restricted infrastructures, it is also used + to tunnel other application-layer protocols. + 3.10.2. Interface Description + There are many HTTP libraries available exposing different APIs. The + APIs provide a way to specify a request by providing a URI, a method, + request modifiers and optionally a request body. For the response, + callbacks can be registered that will be invoked when the response is + received. If TLS is used, API expose a registration of callbacks in + case a server requests client authentication and when certificate + verification is needed. + + World Wide Web Consortium (W3C) standardized the XMLHttpRequest API + [XHR], an API that can be use for sending HTTP/HTTPS requests and + receiving server responses. Besides XML data format, request and + response data format can also be JSON, HTML and plain text. + Specifically JavaScript and XMLHttpRequest are a ubiquitous + programming model for websites, and more general applications, where + native code is less attractive. + + Representational State Transfer (REST) [REST] is another example how + applications can use HTTP as transport protocol. REST is an + architecture style for building application on the Internet. It uses + HTTP as a communication protocol. + 3.10.3. Transport Protocol Components + The transport protocol components provided by HTTP, when used as a + pseudotransport over TCP, are: + + o unicast + + o reliable delivery + + o ordered delivery + + o message and stream-oriented + + o object range request + + o message content type negotiation + + o congestion control + + HTTPS (HTTP over TLS) additionally provides the following components: + + o authentication (of one or both ends of a connection) + + o confidentiality + + o integrity protection + +3.11. WebSockets + + [RFC6455] + + [EDITOR'S NOTE: Salvatore Loreto will contribute text for this + section.] + +3.11.1. Protocol Description +3.11.2. Interface Description + +3.11.3. Transport Protocol Components + 4. Transport Service Features + The transport protocol components analyzed in this document which can + be used as a basis for defining common transport service features, + normalized and separated into categories, are as follows: + + o Destination selection + + * unicast + + * broadcast (IPv4 only) + + * multicast + + * anycast + + * transport layer multihoming for resilience + + * transport layer mobility + + * port multiplexing + + * service codes + + o Connection setup + + * connection setup with feature negotiation and application-to- + port mapping + + o Delivery + + * reliable delivery + + * partially reliable delivery + + * unreliable delivery + + * packet erasure coding + + * ordered delivery + + * unordered delivery + + * stream-oriented delivery + * message-oriented delivery + + * message fragmentation + + * object-oriented delivery of discrete data or file items + + * unordered delivery of in-memory data or file bulk content + objects + + * object range request + + * object content type negotiation + + * single streaming + + * multiple streaming + + * stream scheduling prioritization + + * segmentation + + * data bundling (Nagle's algorithm) + + * message bundling + + o Transmission control + + * timer-based rate control + + * ack-based flow control + + * drop notification + + * packet erasure coding + + * congestion control + + o Integrity protection + + * checksum for error detection + + * partial checksum protection + + * checksum optional + + * cryptographic integrity protection + + o Security + * authentication of one end of a connection + + * authentication of both ends of a connection + + * confidentiality + + The next revision of this document will define transport service + features based upon this list. + [EDITOR'S NOTE: this section will drawn from the candidate features provided by protocol components in the previous section - please discuss on taps@ietf.org list] 4.1. Complete Protocol Feature Matrix [EDITOR'S NOTE: Dave Thaler has signed up as a contributor for this section. Michael Welzl also has a beginning of a matrix which could be useful here.] @@ -891,20 +1279,26 @@ [Editor's Note: turn this into a real contributors section with addresses once we figure out how to trick the toolchain into doing so] o Section 3.4 on UDP was contributed by Kevin Fall (kfall@kfall.com) o Section 3.3 on SCTP was contributed by Michael Tuexen (tuexen@fh- muenster.de) + o Section 3.8 on NORM was contributed by Brian Adamson + (brian.adamson@nrl.navy.mil) + + o Section 3.10 on HTTP was contributed by Dragana Damjanovic + (ddamjanovic@mozilla.com) + 8. Acknowledgments This work is partially supported by the European Commission under grant agreement FP7-ICT-318627 mPlane; support does not imply endorsement. 9. References 9.1. Normative References @@ -927,65 +1321,85 @@ [RFC1191] Mogul, J. and S. Deering, "Path MTU discovery", RFC 1191, November 1990. [RFC1981] McCann, J., Deering, S., and J. Mogul, "Path MTU Discovery for IP version 6", RFC 1981, August 1996. [RFC2018] Mathis, M., Mahdavi, J., Floyd, S., and A. Romanow, "TCP Selective Acknowledgment Options", RFC 2018, October 1996. + [RFC2045] Freed, N. and N. Borenstein, "Multipurpose Internet Mail + Extensions (MIME) Part One: Format of Internet Message + Bodies", RFC 2045, November 1996. + [RFC2460] Deering, S. and R. Hinden, "Internet Protocol, Version 6 (IPv6) Specification", RFC 2460, December 1998. + [RFC2617] Franks, J., Hallam-Baker, P., Hostetler, J., Lawrence, S., + Leach, P., Luotonen, A., and L. Stewart, "HTTP + Authentication: Basic and Digest Access Authentication", + RFC 2617, June 1999. + [RFC3168] Ramakrishnan, K., Floyd, S., and D. Black, "The Addition of Explicit Congestion Notification (ECN) to IP", RFC 3168, September 2001. [RFC3205] Moore, K., "On the use of HTTP as a Substrate", BCP 56, RFC 3205, February 2002. [RFC3390] Allman, M., Floyd, S., and C. Partridge, "Increasing TCP's Initial Window", RFC 3390, October 2002. [RFC3436] Jungmaier, A., Rescorla, E., and M. Tuexen, "Transport Layer Security over Stream Control Transmission Protocol", RFC 3436, December 2002. + [RFC3452] Luby, M., Vicisano, L., Gemmell, J., Rizzo, L., Handley, + M., and J. Crowcroft, "Forward Error Correction (FEC) + Building Block", RFC 3452, December 2002. + [RFC3758] Stewart, R., Ramalho, M., Xie, Q., Tuexen, M., and P. Conrad, "Stream Control Transmission Protocol (SCTP) Partial Reliability Extension", RFC 3758, May 2004. [RFC3828] Larzon, L-A., Degermark, M., Pink, S., Jonsson, L-E., and G. Fairhurst, "The Lightweight User Datagram Protocol (UDP-Lite)", RFC 3828, July 2004. + [RFC4324] Royer, D., Babics, G., and S. Mansour, "Calendar Access + Protocol (CAP)", RFC 4324, December 2005. + [RFC4336] Floyd, S., Handley, M., and E. Kohler, "Problem Statement for the Datagram Congestion Control Protocol (DCCP)", RFC 4336, March 2006. [RFC4340] Kohler, E., Handley, M., and S. Floyd, "Datagram Congestion Control Protocol (DCCP)", RFC 4340, March 2006. [RFC4341] Floyd, S. and E. Kohler, "Profile for Datagram Congestion Control Protocol (DCCP) Congestion Control ID 2: TCP-like Congestion Control", RFC 4341, March 2006. [RFC4342] Floyd, S., Kohler, E., and J. Padhye, "Profile for Datagram Congestion Control Protocol (DCCP) Congestion Control ID 3: TCP-Friendly Rate Control (TFRC)", RFC 4342, March 2006. [RFC4614] Duke, M., Braden, R., Eddy, W., and E. Blanton, "A Roadmap for Transmission Control Protocol (TCP) Specification Documents", RFC 4614, September 2006. + [RFC4654] Widmer, J. and M. Handley, "TCP-Friendly Multicast + Congestion Control (TFMCC): Protocol Specification", RFC + 4654, August 2006. + [RFC4820] Tuexen, M., Stewart, R., and P. Lei, "Padding Chunk and Parameter for the Stream Control Transmission Protocol (SCTP)", RFC 4820, March 2007. [RFC4821] Mathis, M. and J. Heffner, "Packetization Layer Path MTU Discovery", RFC 4821, March 2007. [RFC4895] Tuexen, M., Stewart, R., Lei, P., and E. Rescorla, "Authenticated Chunks for the Stream Control Transmission Protocol (SCTP)", RFC 4895, August 2007. @@ -1020,20 +1434,24 @@ Middlebox Traversal", RFC 5596, September 2009. [RFC5662] Shepler, S., Eisler, M., and D. Noveck, "Network File System (NFS) Version 4 Minor Version 1 External Data Representation Standard (XDR) Description", RFC 5662, January 2010. [RFC5672] Crocker, D., "RFC 4871 DomainKeys Identified Mail (DKIM) Signatures -- Update", RFC 5672, August 2009. + [RFC5740] Adamson, B., Bormann, C., Handley, M., and J. Macker, + "NACK-Oriented Reliable Multicast (NORM) Transport + Protocol", RFC 5740, November 2009. + [RFC6773] Phelan, T., Fairhurst, G., and C. Perkins, "DCCP-UDP: A Datagram Congestion Control Protocol UDP Encapsulation for NAT Traversal", RFC 6773, November 2012. [RFC5925] Touch, J., Mankin, A., and R. Bonica, "The TCP Authentication Option", RFC 5925, June 2010. [RFC5681] Allman, M., Paxson, V., and E. Blanton, "TCP Congestion Control", RFC 5681, September 2009. @@ -1041,20 +1459,24 @@ Transport Layer Security (DTLS) for Stream Control Transmission Protocol (SCTP)", RFC 6083, January 2011. [RFC6093] Gont, F. and A. Yourtchenko, "On the Implementation of the TCP Urgent Mechanism", RFC 6093, January 2011. [RFC6525] Stewart, R., Tuexen, M., and P. Lei, "Stream Control Transmission Protocol (SCTP) Stream Reconfiguration", RFC 6525, February 2012. + [RFC6546] Trammell, B., "Transport of Real-time Inter-network + Defense (RID) Messages over HTTP/TLS", RFC 6546, April + 2012. + [RFC6298] Paxson, V., Allman, M., Chu, J., and M. Sargent, "Computing TCP's Retransmission Timer", RFC 6298, June 2011. [RFC6935] Eubanks, M., Chimento, P., and M. Westerlund, "IPv6 and UDP Checksums for Tunneled Packets", RFC 6935, April 2013. [RFC6936] Fairhurst, G. and M. Westerlund, "Applicability Statement for the Use of IPv6 UDP Datagrams with Zero Checksums", RFC 6936, April 2013. @@ -1077,56 +1499,97 @@ Addresses", RFC 6824, January 2013. [RFC6951] Tuexen, M. and R. Stewart, "UDP Encapsulation of Stream Control Transmission Protocol (SCTP) Packets for End-Host to End-Host Communication", RFC 6951, May 2013. [RFC7053] Tuexen, M., Ruengeler, I., and R. Stewart, "SACK- IMMEDIATELY Extension for the Stream Control Transmission Protocol", RFC 7053, November 2013. + [RFC7230] Fielding, R. and J. Reschke, "Hypertext Transfer Protocol + (HTTP/1.1): Message Syntax and Routing", RFC 7230, June + 2014. + + [RFC7231] Fielding, R. and J. Reschke, "Hypertext Transfer Protocol + (HTTP/1.1): Semantics and Content", RFC 7231, June 2014. + + [RFC7232] Fielding, R. and J. Reschke, "Hypertext Transfer Protocol + (HTTP/1.1): Conditional Requests", RFC 7232, June 2014. + + [RFC7233] Fielding, R., Lafon, Y., and J. Reschke, "Hypertext + Transfer Protocol (HTTP/1.1): Range Requests", RFC 7233, + June 2014. + + [RFC7234] Fielding, R., Nottingham, M., and J. Reschke, "Hypertext + Transfer Protocol (HTTP/1.1): Caching", RFC 7234, June + 2014. + + [RFC7235] Fielding, R. and J. Reschke, "Hypertext Transfer Protocol + (HTTP/1.1): Authentication", RFC 7235, June 2014. + + [RFC7301] Friedl, S., Popov, A., Langley, A., and E. Stephan, + "Transport Layer Security (TLS) Application-Layer Protocol + Negotiation Extension", RFC 7301, July 2014. + [RFC7323] Borman, D., Braden, B., Jacobson, V., and R. Scheffenegger, "TCP Extensions for High Performance", RFC 7323, September 2014. + [RFC7540] Belshe, M., Peon, R., and M. Thomson, "Hypertext Transfer + Protocol Version 2 (HTTP/2)", RFC 7540, May 2015. + [I-D.ietf-aqm-ecn-benefits] Welzl, M. and G. Fairhurst, "The Benefits and Pitfalls of using Explicit Congestion Notification (ECN)", draft-ietf- aqm-ecn-benefits-00 (work in progress), October 2014. [I-D.ietf-tsvwg-sctp-dtls-encaps] Tuexen, M., Stewart, R., Jesup, R., and S. Loreto, "DTLS Encapsulation of SCTP Packets", draft-ietf-tsvwg-sctp- dtls-encaps-09 (work in progress), January 2015. [I-D.ietf-tsvwg-sctp-prpolicies] Tuexen, M., Seggelmann, R., Stewart, R., and S. Loreto, "Additional Policies for the Partial Reliability Extension of the Stream Control Transmission Protocol", draft-ietf- tsvwg-sctp-prpolicies-07 (work in progress), February 2015. [I-D.ietf-tsvwg-sctp-ndata] Stewart, R., Tuexen, M., Loreto, S., and R. Seggelmann, - "Stream Schedulers and a New Data Chunk for the Stream - Control Transmission Protocol", draft-ietf-tsvwg-sctp- - ndata-02 (work in progress), January 2015. + "Stream Schedulers and User Message Interleaving for the + Stream Control Transmission Protocol", draft-ietf-tsvwg- + sctp-ndata-03 (work in progress), March 2015. + + [I-D.ietf-tsvwg-natsupp] + Stewart, R., Tuexen, M., and I. Ruengeler, "Stream Control + Transmission Protocol (SCTP) Network Address Translation + Support", draft-ietf-tsvwg-natsupp-07 (work in progress), + February 2015. + + [XHR] van Kesteren, A., Aubourg, J., Song, J., and H. Steen, + "XMLHttpRequest working draft + (http://www.w3.org/TR/XMLHttpRequest/)", 2000. + + [REST] Fielding, R., "Architectural Styles and the Design of + Network-based Software Architectures, Ph. D. (UC Irvune), + Chapter 5: Representational State Transfer", 2000. Authors' Addresses Godred Fairhurst (editor) University of Aberdeen School of Engineering, Fraser Noble Building Aberdeen AB24 3UE Email: gorry@erg.abdn.ac.uk - Brian Trammell (editor) ETH Zurich Gloriastrasse 35 8092 Zurich Switzerland Email: ietf@trammell.ch Mirja Kuehlewind (editor) ETH Zurich