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| author | Thomas Voss <mail@thomasvoss.com> | 2024-11-27 20:54:24 +0100 |
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| committer | Thomas Voss <mail@thomasvoss.com> | 2024-11-27 20:54:24 +0100 |
| commit | 4bfd864f10b68b71482b35c818559068ef8d5797 (patch) | |
| tree | e3989f47a7994642eb325063d46e8f08ffa681dc /doc/rfc/rfc8095.txt | |
| parent | ea76e11061bda059ae9f9ad130a9895cc85607db (diff) | |
doc: Add RFC documents
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diff --git a/doc/rfc/rfc8095.txt b/doc/rfc/rfc8095.txt new file mode 100644 index 0000000..09c4569 --- /dev/null +++ b/doc/rfc/rfc8095.txt @@ -0,0 +1,3027 @@ + + + + + + +Internet Engineering Task Force (IETF) G. Fairhurst, Ed. +Request for Comments: 8095 University of Aberdeen +Category: Informational B. Trammell, Ed. +ISSN: 2070-1721 M. Kuehlewind, Ed. + ETH Zurich + March 2017 + + + Services Provided by + IETF Transport Protocols and Congestion Control Mechanisms + +Abstract + + This document describes, surveys, and classifies the protocol + mechanisms provided by existing IETF protocols, as background for + determining a common set of transport services. It examines the + Transmission Control Protocol (TCP), Multipath TCP, the Stream + Control Transmission Protocol (SCTP), the User Datagram Protocol + (UDP), UDP-Lite, the Datagram Congestion Control Protocol (DCCP), the + Internet Control Message Protocol (ICMP), the Real-Time Transport + Protocol (RTP), File Delivery over Unidirectional Transport / + Asynchronous Layered Coding (FLUTE/ALC) for Reliable Multicast, NACK- + Oriented Reliable Multicast (NORM), Transport Layer Security (TLS), + Datagram TLS (DTLS), and the Hypertext Transport Protocol (HTTP), + when HTTP is used as a pseudotransport. This survey provides + background for the definition of transport services within the TAPS + working group. + +Status of This Memo + + This document is not an Internet Standards Track specification; it is + published for informational purposes. + + This document is a product of the Internet Engineering Task Force + (IETF). It represents the consensus of the IETF community. It has + received public review and has been approved for publication by the + Internet Engineering Steering Group (IESG). Not all documents + approved by the IESG are a candidate for any level of Internet + Standard; see Section 2 of RFC 7841. + + Information about the current status of this document, any errata, + and how to provide feedback on it may be obtained at + http://www.rfc-editor.org/info/rfc8095. + + + + + + + + +Fairhurst, et al. Informational [Page 1] + +RFC 8095 Transport Services March 2017 + + +Copyright Notice + + Copyright (c) 2017 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 ....................................................4 + 1.1. Overview of Transport Features .............................4 + 2. Terminology .....................................................5 + 3. Existing Transport Protocols ....................................6 + 3.1. Transport Control Protocol (TCP) ...........................6 + 3.1.1. Protocol Description ................................6 + 3.1.2. Interface Description ...............................8 + 3.1.3. Transport Features ..................................9 + 3.2. Multipath TCP (MPTCP) .....................................10 + 3.2.1. Protocol Description ...............................10 + 3.2.2. Interface Description ..............................10 + 3.2.3. Transport Features .................................11 + 3.3. User Datagram Protocol (UDP) ..............................11 + 3.3.1. Protocol Description ...............................11 + 3.3.2. Interface Description ..............................12 + 3.3.3. Transport Features .................................13 + 3.4. Lightweight User Datagram Protocol (UDP-Lite) .............13 + 3.4.1. Protocol Description ...............................13 + 3.4.2. Interface Description ..............................14 + 3.4.3. Transport Features .................................14 + 3.5. Stream Control Transmission Protocol (SCTP) ...............14 + 3.5.1. Protocol Description ...............................15 + 3.5.2. Interface Description ..............................17 + 3.5.3. Transport Features .................................19 + 3.6. Datagram Congestion Control Protocol (DCCP) ...............20 + 3.6.1. Protocol Description ...............................21 + 3.6.2. Interface Description ..............................22 + 3.6.3. Transport Features .................................22 + + + + + + +Fairhurst, et al. Informational [Page 2] + +RFC 8095 Transport Services March 2017 + + + 3.7. Transport Layer Security (TLS) and Datagram TLS + (DTLS) as a Pseudotransport ...............................23 + 3.7.1. Protocol Description ...............................23 + 3.7.2. Interface Description ..............................24 + 3.7.3. Transport Features .................................25 + 3.8. Real-Time Transport Protocol (RTP) ........................26 + 3.8.1. Protocol Description ...............................26 + 3.8.2. Interface Description ..............................27 + 3.8.3. Transport Features .................................27 + 3.9. Hypertext Transport Protocol (HTTP) over TCP as a + Pseudotransport ...........................................28 + 3.9.1. Protocol Description ...............................28 + 3.9.2. Interface Description ..............................29 + 3.9.3. Transport Features .................................30 + 3.10. File Delivery over Unidirectional Transport / + Asynchronous Layered Coding (FLUTE/ALC) for + Reliable Multicast .......................................31 + 3.10.1. Protocol Description ..............................31 + 3.10.2. Interface Description .............................33 + 3.10.3. Transport Features ................................33 + 3.11. NACK-Oriented Reliable Multicast (NORM) ..................34 + 3.11.1. Protocol Description ..............................34 + 3.11.2. Interface Description .............................35 + 3.11.3. Transport Features ................................36 + 3.12. Internet Control Message Protocol (ICMP) .................36 + 3.12.1. Protocol Description ..............................37 + 3.12.2. Interface Description .............................37 + 3.12.3. Transport Features ................................38 + 4. Congestion Control .............................................38 + 5. Transport Features .............................................39 + 6. IANA Considerations ............................................42 + 7. Security Considerations ........................................42 + 8. Informative References .........................................42 + Acknowledgments ...................................................53 + Contributors ......................................................53 + Authors' Addresses ................................................54 + + + + + + + + + + + + + + + +Fairhurst, et al. Informational [Page 3] + +RFC 8095 Transport Services March 2017 + + +1. Introduction + + Internet applications make use of the services provided by a + transport protocol, such as 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, and integrity protection. + + The IETF has defined a wide variety of transport protocols beyond TCP + and UDP, including SCTP, DCCP, MPTCP, 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), 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 NewReno [RFC6582], + TCP-Friendly Rate Control (TFRC) [RFC5348], or Low Extra Delay + Background Transport (LEDBAT) [RFC6817]). This extends the set of + available transport services beyond those provided to applications by + TCP and UDP. + + The transport protocols described in this document provide a basis + for the definition of transport services provided by common + protocols, as background for the TAPS working group. The protocols + listed here were chosen to help expose as many potential transport + services as possible and are not meant to be a comprehensive survey + or classification of all transport protocols. + +1.1. Overview of Transport Features + + Transport protocols can be differentiated by the features of the + services they provide. + + Some of these provided features are closely related to basic control + function that a protocol needs to work over a network path, such as + addressing. The number of participants in a given association also + determines its applicability: a connection can be between endpoints + (unicast), to one of multiple endpoints (anycast), or simultaneously + to multiple endpoints (multicast). Unicast protocols usually support + bidirectional communication, while multicast is generally + + + + +Fairhurst, et al. Informational [Page 4] + +RFC 8095 Transport Services March 2017 + + + unidirectional. Another feature is whether a transport requires a + control exchange across the network at setup (e.g., TCP) or whether + it is connectionless (e.g., UDP). + + For packet delivery itself, reliability and integrity protection, + ordering, and framing are basic features. However, these features + are implemented with different levels of assurance in different + protocols. As an example, a transport service may provide full + reliability, with detection of loss and retransmission (e.g., TCP). + SCTP offers a message-based service that can provide full or partial + reliability and allows the protocol to minimize the head-of-line + blocking due to the support of ordered and unordered message delivery + within multiple streams. UDP-Lite and DCCP can provide partial + integrity protection to enable corruption tolerance. + + Usually, a protocol has been designed to support one specific type of + delivery/framing: either data needs to be divided into transmission + units based on network packets (datagram service) or a data stream is + segmented and re-combined across multiple packets (stream service). + Whole objects such as files are handled accordingly. This decision + strongly influences the interface that is provided to the upper + layer. + + In addition, transport protocols offer a certain support for + transmission control. For example, a transport service can provide + flow control to allow a receiver to regulate the transmission rate of + a sender. Further, a transport service can provide congestion + control (see Section 4). As an example, TCP and SCTP provide + congestion control for use in the Internet, whereas UDP leaves this + function to the upper-layer protocol that uses UDP. + + Security features are often provided independently of the transport + protocol, via Transport Layer Security (TLS) (see Section 3.7) or by + the application-layer protocol itself. The security properties TLS + provides to the application (such as confidentiality, integrity, and + authenticity) are also features of the transport layer, even though + they are often presently implemented in a separate protocol. + +2. Terminology + + The following terms are used throughout this document and in + subsequent documents produced by the TAPS working group that describe + the composition and decomposition of transport services. + + Transport Feature: a specific end-to-end feature that the transport + layer provides to an application. Examples include + confidentiality, reliable delivery, ordered delivery, message- + versus-stream orientation, etc. + + + +Fairhurst, et al. Informational [Page 5] + +RFC 8095 Transport Services March 2017 + + + Transport Service: a set of transport features, without an + association to any given framing protocol, that provides a + complete service to an application. + + Transport Protocol: an implementation that provides one or more + different transport services using a specific framing and header + format on the wire. + + Application: an entity that uses the transport layer for end-to-end + delivery data across the network (this may also be an upper-layer + protocol or tunnel encapsulation). + +3. Existing Transport Protocols + + This section provides a list of known IETF transport protocols and + transport protocol frameworks. It does not make an assessment about + whether specific implementations of protocols are fully compliant to + current IETF specifications. + +3.1. Transport Control Protocol (TCP) + + TCP is an IETF Standards Track transport protocol. [RFC793] + introduces TCP as follows: + + The Transmission Control Protocol (TCP) is intended for use as a + highly reliable host-to-host protocol between hosts in packet- + switched computer communication networks, and in interconnected + systems of such networks. + + Since its introduction, TCP has become the default connection- + oriented, stream-based transport protocol in the Internet. It is + widely implemented by endpoints and widely used by common + applications. + +3.1.1. Protocol Description + + TCP is a connection-oriented protocol that provides a three-way + handshake to allow a client and server to set up a connection and + negotiate features and provides mechanisms for orderly completion and + immediate teardown of a connection [RFC793] [TCP-SPEC]. TCP is + defined by a family of RFCs (see [RFC7414]). + + TCP provides multiplexing to multiple sockets on each host using port + numbers. A similar approach is adopted by other IETF-defined + transports. An active TCP session is identified by its four-tuple of + local and remote IP addresses and local and remote port numbers. The + destination port during connection setup is often used to indicate + the requested service. + + + +Fairhurst, et al. Informational [Page 6] + +RFC 8095 Transport Services March 2017 + + + TCP partitions a continuous stream of bytes into segments, sized to + fit in IP packets based on a negotiated maximum segment size and + further constrained by the effective Maximum Transmission Unit (MTU) + from Path MTU Discovery (PMTUD). ICMP-based PMTUD [RFC1191] + [RFC1981] as well as Packetization Layer PMTUD (PLPMTUD) [RFC4821] + have been defined by the IETF. + + Each byte in the stream is identified by a sequence number. The + sequence number is used to order segments on receipt, to identify + segments in acknowledgments, and to detect unacknowledged segments + for retransmission. This is the basis of the reliable, ordered + delivery of data in a TCP stream. TCP Selective Acknowledgment + (SACK) [RFC2018] extends this mechanism by making it possible to + provide earlier identification of which segments are missing, + allowing faster retransmission. SACK-based methods (e.g., Duplicate + Selective ACK) can also result in less spurious retransmission. + + Receiver flow control is provided by a sliding window, which limits + the amount of unacknowledged data that can be outstanding at a given + time. The window scale option [RFC7323] allows a receiver to use + windows greater than 64 KB. + + All TCP senders provide congestion control, such as that described in + [RFC5681]. TCP uses a sequence number with a sliding receiver window + for flow control. The TCP congestion control mechanism also utilizes + this TCP sequence number to manage a separate congestion window + [RFC5681]. The sending window at a given point in time is the + minimum of the receiver window and the congestion window. The + congestion window is increased in the absence of congestion and + decreased if congestion is detected. Often, loss is implicitly + handled as a congestion indication, which is detected in TCP (also as + input for retransmission handling) based on two mechanisms: a + retransmission timer with exponential back-off or the reception of + three acknowledgments for the same segment, so called "duplicated + ACKs" (fast retransmit). In addition, Explicit Congestion + Notification (ECN) [RFC3168] can be used in TCP and, if supported by + both endpoints, allows a network node to signal congestion without + inducing loss. Alternatively, a delay-based congestion control + scheme that reacts to changes in delay as an early indication of + congestion can be used in TCP. This is further described in + Section 4. Examples of different kinds of congestion control schemes + are provided in Section 4. + + TCP protocol instances can be extended (see [RFC7414]). Some + protocol features may also be tuned to optimize for a specific + deployment scenario. Some features are sender-side only, requiring + no negotiation with the receiver; some are receiver-side only; and + some are explicitly negotiated during connection setup. + + + +Fairhurst, et al. Informational [Page 7] + +RFC 8095 Transport Services March 2017 + + + TCP may buffer data, e.g., to optimize processing or capacity usage. + TCP therefore provides mechanisms to control this, including an + optional "PUSH" function [RFC793] that explicitly requests the + transport service not to delay data. By default, TCP segment + partitioning uses Nagle's algorithm [TCP-SPEC] to buffer data at the + sender into large segments, potentially incurring sender-side + buffering delay; this algorithm can be disabled by the sender to + transmit more immediately, e.g., to reduce latency for interactive + sessions. + + TCP provides an "urgent data" function for limited out-of-order + delivery of the data. This function is deprecated [RFC6093]. + + A TCP Reset (RST) control message may be used to force a TCP endpoint + to close a session [RFC793], aborting the connection. + + A mandatory checksum provides a basic integrity check against + misdelivery and data corruption over the entire packet. Applications + that require end-to-end integrity of data are recommended to include + a stronger integrity check of their payload data. The TCP checksum + [RFC1071] [RFC2460] does not support partial payload protection (as + in DCCP/UDP-Lite). + + TCP supports only unicast connections. + +3.1.2. Interface Description + + The User/TCP Interface defined in [RFC793] provides six user + commands: Open, Send, Receive, Close, Status, and Abort. This + interface does not describe configuration of TCP options or + parameters aside from the use of the PUSH and URGENT flags. + + [RFC1122] describes extensions of the TCP/application-layer interface + for: + + o reporting soft errors such as reception of ICMP error messages, + extensive retransmission, or urgent pointer advance, + + o providing a possibility to specify the Differentiated Services + Code Point (DSCP) [RFC3260] (formerly, the Type-of-Service (TOS)) + for segments, + + o providing a flush call to empty the TCP send queue, and + + o multihoming support. + + + + + + +Fairhurst, et al. Informational [Page 8] + +RFC 8095 Transport Services March 2017 + + + In API implementations derived from the BSD Sockets API, TCP sockets + are created using the "SOCK_STREAM" socket type as described in the + IEEE Portable Operating System Interface (POSIX) Base Specifications + [POSIX]. The features used by a protocol instance may be set and + tuned via this API. There are currently no documents in the RFC + Series that describe this interface. + +3.1.3. Transport Features + + The transport features provided by TCP are: + + o connection-oriented transport with feature negotiation and + application-to-port mapping (implemented using SYN segments and + the TCP Option field to negotiate features), + + o unicast transport (though anycast TCP is implemented, at risk of + instability due to rerouting), + + o port multiplexing, + + o unidirectional or bidirectional communication, + + o stream-oriented delivery in a single stream, + + o fully reliable delivery (implemented using ACKs sent from the + receiver to confirm delivery), + + o error detection (implemented using a segment checksum to verify + delivery to the correct endpoint and integrity of the data and + options), + + o segmentation, + + o data bundling (optional; uses Nagle's algorithm to coalesce data + sent within the same RTT into full-sized segments), + + o flow control (implemented using a window-based mechanism where the + receiver advertises the window that it is willing to buffer), and + + o congestion control (usually implemented using a window-based + mechanism and four algorithms for different phases of the + transmission: slow start, congestion avoidance, fast retransmit, + and fast recovery [RFC5681]). + + + + + + + + +Fairhurst, et al. Informational [Page 9] + +RFC 8095 Transport Services March 2017 + + +3.2. Multipath TCP (MPTCP) + + Multipath TCP [RFC6824] is an extension for TCP to support + multihoming for resilience, mobility, and load balancing. It is + designed to be as indistinguishable to middleboxes from non-multipath + TCP as possible. It does so by establishing regular TCP flows + between a pair of source/destination endpoints and multiplexing the + application's stream over these flows. Sub-flows can be started over + IPv4 or IPv6 for the same session. + +3.2.1. Protocol Description + + MPTCP uses TCP options for its control plane. They are used to + signal multipath capabilities, as well as to negotiate data sequence + numbers, advertise other available IP addresses, and establish new + sessions between pairs of endpoints. + + By multiplexing one byte stream over separate paths, MPTCP can + achieve a higher throughput than TCP in certain situations. However, + if coupled congestion control [RFC6356] is used, it might limit this + benefit to maintain fairness to other flows at the bottleneck. When + aggregating capacity over multiple paths, and depending on the way + packets are scheduled on each TCP subflow, additional delay and + higher jitter might be observed before in-order delivery of data to + the applications. + +3.2.2. Interface Description + + By default, MPTCP exposes the same interface as TCP to the + application. [RFC6897], however, describes a richer API for MPTCP- + aware applications. + + This Basic API describes how an application can: + + o enable or disable MPTCP. + + o bind a socket to one or more selected local endpoints. + + o query local and remote endpoint addresses. + + o get a unique connection identifier (similar to an address-port + pair for TCP). + + The document also recommends the use of extensions defined for SCTP + [RFC6458] (see Section 3.5) to support multihoming for resilience and + mobility. + + + + + +Fairhurst, et al. Informational [Page 10] + +RFC 8095 Transport Services March 2017 + + +3.2.3. Transport Features + + As an extension to TCP, MPTCP provides mostly the same features. By + establishing multiple sessions between available endpoints, it can + additionally provide soft failover solutions in the case that one of + the paths becomes unusable. + + Therefore, the transport features provided by MPTCP in addition to + TCP are: + + o multihoming for load balancing, with endpoint multiplexing of a + single byte stream, using either coupled congestion control or + throughput maximization, + + o address family multiplexing (using IPv4 and IPv6 for the same + session), and + + o resilience to network failure and/or handover. + +3.3. User Datagram Protocol (UDP) + + The User Datagram Protocol (UDP) [RFC768] [RFC2460] is an IETF + Standards Track transport protocol. It provides a unidirectional + datagram protocol that preserves message boundaries. It provides no + 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 [RFC8085]. UDP is widely + implemented and widely used by common applications, including DNS. + +3.3.1. Protocol Description + + UDP is a connectionless protocol that maintains message boundaries, + with no connection setup or feature negotiation. The protocol uses + independent messages, ordinarily called "datagrams". It provides + detection of payload errors and misdelivery of packets to an + unintended endpoint, both of which result in discard of received + datagrams, with no indication to the user of the service. + + It is possible to create IPv4 UDP datagrams with no checksum, and + while this is generally discouraged [RFC1122] [RFC8085], certain + special cases permit this use. These datagrams rely on the IPv4 + header checksum to protect from misdelivery to an unintended + endpoint. IPv6 does not permit UDP datagrams with no checksum, + although in certain cases [RFC6936], this rule may be relaxed + [RFC6935]. + + + + + +Fairhurst, et al. Informational [Page 11] + +RFC 8095 Transport Services March 2017 + + + UDP does not provide reliability and does not provide retransmission. + Messages may be reordered, lost, or duplicated in transit. Note that + due to the relatively weak form of checksum used by UDP, applications + that require end-to-end integrity of data are recommended to include + a stronger integrity check of their payload data. + + Because UDP provides no flow control, a receiving application that is + unable to run sufficiently fast, or frequently, may miss messages. + The lack of congestion handling implies UDP traffic may experience + loss when using an overloaded path and may cause the loss of messages + from other protocols (e.g., TCP) when sharing the same network path. + + On transmission, UDP encapsulates each datagram into a single IP + packet or several IP packet fragments. This allows a datagram to be + larger than the effective path MTU. Fragments are reassembled before + delivery to the UDP receiver, making this transparent to the user of + the transport service. When jumbograms are supported, larger + messages may be sent without performing fragmentation. + + UDP on its own does not provide support for segmentation, receiver + flow control, congestion control, PMTUD/PLPMTUD, or ECN. + Applications that require these features need to provide them on + their own or use a protocol over UDP that provides them [RFC8085]. + +3.3.2. Interface Description + + [RFC768] describes basic requirements for an API for UDP. Guidance + on the use of common APIs is provided in [RFC8085]. + + A UDP endpoint consists of a tuple of (IP address, port number). + De-multiplexing using multiple abstract endpoints (sockets) on the + same IP address is supported. The same socket may be used by a + single server to interact with multiple clients. (Note: This + behavior differs from TCP, which uses a pair of tuples to identify a + connection). Multiple server instances (processes) that bind to the + same socket can cooperate to service multiple clients. The socket + implementation arranges to not duplicate the same received unicast + message to multiple server processes. + + Many operating systems also allow a UDP socket to be "connected", + i.e., to bind a UDP socket to a specific (remote) UDP endpoint. + Unlike TCP's connect primitive, for UDP, this is only a local + operation that serves to simplify the local send/receive functions + and to filter the traffic for the specified addresses and ports + [RFC8085]. + + + + + + +Fairhurst, et al. Informational [Page 12] + +RFC 8095 Transport Services March 2017 + + +3.3.3. Transport Features + + The transport features provided by UDP are: + + o unicast, multicast, anycast, or IPv4 broadcast transport, + + o port multiplexing (where a receiving port can be configured to + receive datagrams from multiple senders), + + o message-oriented delivery, + + o unidirectional or bidirectional communication where the + transmissions in each direction are independent, + + o non-reliable delivery, + + o unordered delivery, and + + o error detection (implemented using a segment checksum to verify + delivery to the correct endpoint and integrity of the data; + optional for IPv4 and optional under specific conditions for IPv6 + where all or none of the payload data is protected). + +3.4. Lightweight User Datagram Protocol (UDP-Lite) + + The Lightweight User Datagram Protocol (UDP-Lite) [RFC3828] is an + IETF Standards Track transport protocol. It provides a + unidirectional, datagram protocol that preserves message boundaries. + IETF guidance on the use of UDP-Lite is provided in [RFC8085]. A + UDP-Lite service may support IPv4 broadcast, multicast, anycast, and + unicast, as well as IPv6 multicast, anycast, and unicast. + + Examples of use include a class of applications that can derive + benefit from having partially damaged payloads delivered rather than + discarded. One use is to provide header integrity checks but allow + delivery of corrupted payloads to error-tolerant applications or to + applications that use some other mechanism to provide payload + integrity (see [RFC6936]). + +3.4.1. Protocol Description + + Like UDP, UDP-Lite is a connectionless datagram protocol, with no + connection setup or feature negotiation. It changes the semantics of + the UDP Payload Length field to that of a Checksum Coverage Length + field and is identified by a different IP protocol/next-header value. + The Checksum Coverage Length field specifies the intended checksum + coverage, with the remaining unprotected part of the payload called + + + + +Fairhurst, et al. Informational [Page 13] + +RFC 8095 Transport Services March 2017 + + + the "error-insensitive part". Therefore, applications using UDP-Lite + cannot make assumptions regarding the correctness of the data + received in the insensitive part of the UDP-Lite payload. + + Otherwise, UDP-Lite is semantically identical to UDP. In the same + way as for UDP, mechanisms for receiver flow control, congestion + control, PMTU or PLPMTU discovery, support for ECN, etc., need to be + provided by upper-layer protocols [RFC8085]. + +3.4.2. Interface Description + + There is no API currently specified in the RFC Series, but guidance + on use of common APIs is provided in [RFC8085]. + + The interface of UDP-Lite differs from that of UDP by the addition of + a single (socket) option that communicates a checksum coverage length + value. The checksum coverage may also be made visible to the + application via the UDP-Lite MIB module [RFC5097]. + +3.4.3. Transport Features + + The transport features provided by UDP-Lite are: + + o unicast, multicast, anycast, or IPv4 broadcast transport (same as + for UDP), + + o port multiplexing (same as for UDP), + + o message-oriented delivery (same as for UDP), + + o unidirectional or bidirectional communication where the + transmissions in each direction are independent (same as for UDP), + + o non-reliable delivery (same as for UDP), + + o non-ordered delivery (same as for UDP), and + + o partial or full payload error detection (where the Checksum + Coverage field indicates the size of the payload data covered by + the checksum). + +3.5. Stream Control Transmission Protocol (SCTP) + + SCTP is a message-oriented IETF Standards Track transport protocol. + The base protocol is specified in [RFC4960]. It supports multihoming + and path failover to provide resilience to path failures. An SCTP + association has multiple streams in each direction, providing + in-sequence delivery of user messages within each stream. This + + + +Fairhurst, et al. Informational [Page 14] + +RFC 8095 Transport Services March 2017 + + + allows it to minimize head-of-line blocking. SCTP supports multiple + stream- scheduling schemes controlling stream multiplexing, including + priority and fair weighting schemes. + + SCTP was originally developed for transporting telephony signaling + messages and is deployed in telephony signaling networks, especially + in mobile telephony networks. It can also be used for other + services, for example, in the WebRTC framework for data channels. + +3.5.1. Protocol Description + + SCTP is a connection-oriented protocol using a four-way handshake to + establish an SCTP association and a three-way message exchange to + gracefully shut it down. It uses the same port number concept as + DCCP, TCP, UDP, and UDP-Lite. SCTP only supports unicast. + + SCTP uses the 32-bit CRC32c for protecting SCTP packets against bit + errors and misdelivery of packets to an unintended endpoint. This is + stronger than the 16-bit checksums used by TCP or UDP. However, + partial payload checksum coverage as provided by DCCP or UDP-Lite is + not supported. + + SCTP has been designed with extensibility in mind. A common header + is followed by a sequence of chunks. [RFC4960] defines how a + receiver processes chunks with an unknown chunk type. The support of + extensions can be negotiated during the SCTP handshake. Currently + defined extensions include mechanisms for dynamic reconfiguration of + streams [RFC6525] and IP addresses [RFC5061]. Furthermore, the + extension specified in [RFC3758] introduces the concept of partial + reliability for user messages. + + SCTP provides a message-oriented service. Multiple small user + messages can be bundled into a single SCTP packet to improve + efficiency. For example, this bundling may be done by delaying user + messages at the sender, similar to Nagle's algorithm used by TCP. + User messages that would result in IP packets larger than the MTU + will be fragmented at the sender and reassembled at the receiver. + There is no protocol limit on the user message size. For MTU + discovery, the same mechanism as for TCP can be used [RFC1981] + [RFC4821], as well as utilization of probe packets with padding + chunks, as defined in [RFC4820]. + + [RFC4960] specifies TCP-friendly congestion control to protect the + network against overload. SCTP also uses sliding window flow control + to protect receivers against overflow. Similar to TCP, SCTP also + supports delaying acknowledgments. [RFC7053] provides a way for the + sender of user messages to request immediate sending of the + corresponding acknowledgments. + + + +Fairhurst, et al. Informational [Page 15] + +RFC 8095 Transport Services March 2017 + + + Each SCTP association has between 1 and 65536 unidirectional 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 unordered or ordered upon request by the upper + layer. Unordered messages can be delivered as soon as they are + completely received. For user messages not requiring fragmentation, + this minimizes head-of-line blocking. On the other hand, ordered + messages sent on the same stream are delivered at the receiver in the + same order as sent by the sender. + + The base protocol defined in [RFC4960] does not allow interleaving of + user messages. Large messages on one stream can therefore block the + sending of user messages on other streams. [SCTP-NDATA] describes a + method to overcome this limitation. This document also specifies + multiple algorithms for the sender-side selection of which streams to + send data from, supporting a variety of scheduling algorithms + including priority-based methods. The stream reconfiguration + extension defined in [RFC6525] allows streams to be reset during the + lifetime of an association and to increase the number of streams, if + the number of streams negotiated in the SCTP handshake becomes + insufficient. + + Each user message sent is delivered to the receiver or, in case of + excessive retransmissions, the association is terminated in a + non-graceful way [RFC4960], similar to TCP behavior. In addition to + this reliable transfer, the partial reliability extension [RFC3758] + allows a sender to abandon user messages. The application can + specify the policy for abandoning user messages. + + SCTP supports multihoming. Each SCTP endpoint 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 reconfiguration extension + specified in [RFC5061] in combination with [RFC4895] can be used to + change these addresses in an authenticated way during the lifetime 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. + + 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, such as unordered delivery or partial + reliability. 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. + + + + + +Fairhurst, et al. Informational [Page 16] + +RFC 8095 Transport Services March 2017 + + + [NAT-SUPP] defines methods for endpoints and middleboxes to provide + NAT traversal 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 [SCTP-DTLS-ENCAPS]. The latter encapsulation is used + within the WebRTC [WEBRTC-TRANS] context. + + An SCTP ABORT chunk may be used to force a SCTP endpoint to close a + session [RFC4960], aborting the connection. + + SCTP has a well-defined API, described in the next subsection. + +3.5.2. Interface Description + + [RFC4960] defines an abstract API for the base protocol. This API + describes the following functions callable by the upper layer of + SCTP: Initialize, Associate, Send, Receive, Receive Unsent Message, + Receive Unacknowledged Message, Shutdown, Abort, SetPrimary, Status, + Change Heartbeat, Request Heartbeat, Get SRTT Report, Set Failure + Threshold, Set Protocol Parameters, and Destroy. The following + notifications are provided by the SCTP stack to the upper layer: + COMMUNICATION UP, DATA ARRIVE, SHUTDOWN COMPLETE, COMMUNICATION LOST, + COMMUNICATION ERROR, RESTART, SEND FAILURE, and NETWORK STATUS + CHANGE. + + An extension to the BSD Sockets API is defined in [RFC6458] and + covers: + + o the base protocol defined in [RFC4960]. The API allows control + over local addresses and port numbers and the primary path. + Furthermore, the application has fine control of parameters like + retransmission thresholds, the path supervision, the delayed + acknowledgment timeout, and the fragmentation point. The API + provides a mechanism to allow the SCTP stack to notify the + application about events if the application has requested them. + These notifications provide information about status changes of + the association and each of the peer addresses. In case of send + failures, including drop of messages sent unreliably, the + application can also be notified, and user messages can be + returned to the application. When sending user messages, the + application can indicate a stream id, a payload protocol + identifier, and an indication of whether ordered delivery is + requested. These parameters can also be provided on message + reception. Additionally, a context can be provided when sending, + which can be used in case of send failures. The sending of + arbitrarily large user messages is supported. + + + + + +Fairhurst, et al. Informational [Page 17] + +RFC 8095 Transport Services March 2017 + + + o the SCTP Partial Reliability extension defined in [RFC3758] to + specify for a user message the Partially Reliable SCTP (PR-SCTP) + policy and the policy-specific parameter. Examples of these + policies defined in [RFC3758] and [RFC7496] are: + + * limiting the time a user message is dealt with by the sender. + + * limiting the number of retransmissions for each fragment of a + user message. If the number of retransmissions is limited to + 0, one gets a service similar to UDP. + + * abandoning messages of lower priority in case of a send buffer + shortage. + + o the SCTP Authentication extension defined in [RFC4895] allowing + management of the shared keys and allowing the HMAC to use and set + the chunk types (which are only accepted in an authenticated way) + and get the list of chunks that are accepted by the local and + remote endpoints in an authenticated way. + + o the SCTP Dynamic Address Reconfiguration extension defined in + [RFC5061]. It allows the manual addition and deletion of local + addresses for SCTP associations, as well as the enabling of + automatic address addition and deletion. Furthermore, the peer + can be given a hint for choosing its primary path. + + A BSD Sockets API extension has been defined in the documents that + specify the following SCTP extensions: + + o the SCTP Stream Reconfiguration extension defined in [RFC6525]. + The API allows triggering of the reset operation for incoming and + outgoing streams and the whole association. It also provides a + way to notify the association about the corresponding events. + Furthermore, the application can increase the number of streams. + + o the UDP Encapsulation of SCTP packets extension defined in + [RFC6951]. The API allows the management of the remote UDP + encapsulation port. + + o the SCTP SACK-IMMEDIATELY extension defined in [RFC7053]. The API + allows the sender of a user message to request the receiver to + send the corresponding acknowledgment immediately. + + o the additional PR-SCTP policies defined in [RFC7496]. The API + allows enabling/disabling the PR-SCTP extension, choosing the + PR-SCTP policies defined in the document, and providing + statistical information about abandoned messages. + + + + +Fairhurst, et al. Informational [Page 18] + +RFC 8095 Transport Services March 2017 + + + Future documents describing SCTP extensions are expected to describe + the corresponding BSD Sockets API extension in a "Socket API + Considerations" section. + + The SCTP Socket API supports two kinds of sockets: + + o one-to-one style sockets (by using the socket type "SOCK_STREAM"). + + o one-to-many style socket (by using the socket type + "SOCK_SEQPACKET"). + + One-to-one style sockets are similar to TCP sockets; there is a 1:1 + relationship between the sockets and the SCTP associations (except + for listening sockets). One-to-many style SCTP sockets are similar + to unconnected UDP sockets, where there is a 1:n relationship between + the sockets and the SCTP associations. + + The SCTP stack can provide information to the applications about + state changes of the individual paths and the association whenever + they occur. These events are delivered similarly to user messages + but are specifically marked as notifications. + + New functions have been introduced to support the use of multiple + local and remote addresses. Additional SCTP-specific send and + receive calls have been defined to permit SCTP-specific information + to be sent without using ancillary data in the form of additional + Control Message (cmsg) calls. These functions provide support for + detecting partial delivery of user messages and notifications. + + The SCTP Socket API allows a fine-grained control of the protocol + behavior through an extensive set of socket options. + + The SCTP kernel implementations of FreeBSD, Linux, and Solaris follow + mostly the specified extension to the BSD Sockets API for the base + protocol and the corresponding supported protocol extensions. + +3.5.3. Transport Features + + The transport features provided by SCTP are: + + o connection-oriented transport with feature negotiation and + application-to-port mapping, + + o unicast transport, + + o port multiplexing, + + o unidirectional or bidirectional communication, + + + +Fairhurst, et al. Informational [Page 19] + +RFC 8095 Transport Services March 2017 + + + o message-oriented delivery with durable message framing supporting + multiple concurrent streams, + + o fully reliable, partially reliable, or unreliable delivery (based + on user-specified policy to handle abandoned user messages) with + drop notification, + + o ordered and unordered delivery within a stream, + + o support for stream scheduling prioritization, + + o segmentation, + + o user message bundling, + + o flow control using a window-based mechanism, + + o congestion control using methods similar to TCP, + + o strong error detection (CRC32c), and + + o transport-layer multihoming for resilience and mobility. + +3.6. Datagram Congestion Control Protocol (DCCP) + + The Datagram Congestion Control Protocol (DCCP) [RFC4340] is an IETF + Standards Track bidirectional transport protocol that provides + unicast connections of congestion-controlled messages without + providing reliability. + + The DCCP Problem Statement [RFC4336] describes the goals that DCCP + sought to address. It is suitable for applications that transfer + fairly large amounts of data and that can benefit from control over + the trade-off between timeliness and reliability [RFC4336]. + + DCCP offers low overhead, and many characteristics common to UDP, but + can avoid "re-inventing the wheel" each time a new multimedia + application emerges. Specifically, it includes core transport + functions (feature negotiation, path state management, RTT + calculation, PMTUD, etc.): DCCP applications select how they send + packets and, where suitable, choose common algorithms to manage their + functions. Examples of applications that can benefit from such + transport services include interactive applications, streaming media, + or on-line games [RFC4336]. + + + + + + + +Fairhurst, et al. Informational [Page 20] + +RFC 8095 Transport Services March 2017 + + +3.6.1. Protocol Description + + DCCP is a connection-oriented datagram protocol that provides a + three-way handshake to allow a client and server to set up a + connection and provides mechanisms for orderly completion and + immediate teardown of a connection. + + A DCCP protocol instance can be extended [RFC4340] and tuned using + additional features. Some features are sender-side only, requiring + no negotiation with the receiver; some are receiver-side only; and + some are explicitly negotiated during connection setup. + + DCCP uses a Connect packet to initiate a session and permits each + endpoint to choose the features it wishes to support. Simultaneous + open [RFC5596], as in TCP, can enable interoperability in the + presence of middleboxes. The Connect packet includes a Service Code + [RFC5595] that identifies the application or protocol using DCCP, + providing middleboxes with information about the intended use of a + connection. + + The DCCP service is unicast-only. + + It provides multiplexing to multiple sockets at each endpoint using + port numbers. An active DCCP session is identified by its four-tuple + of local and remote IP addresses and local and remote port numbers. + + The protocol segments data into messages that are typically sized to + fit in IP packets but may be fragmented if they are smaller than the + maximum packet size. A DCCP interface allows applications to request + fragmentation for packets larger than PMTU, but not larger than the + maximum packet size allowed by the current congestion control + mechanism (Congestion Control Maximum Packet Size (CCMPS)) [RFC4340]. + + Each message is identified by a sequence number. The sequence number + is used to identify segments in acknowledgments, to detect + unacknowledged segments, to measure RTT, etc. The protocol may + support unordered delivery of data and does not itself provide + retransmission. DCCP supports reduced checksum coverage, a partial + payload protection mechanism similar to UDP-Lite. There is also a + Data Checksum option, which when enabled, contains a strong Cyclic + Redundancy Check (CRC), to enable endpoints to detect application + data corruption. + + Receiver flow control is supported, which limits the amount of + unacknowledged data that can be outstanding at a given time. + + A DCCP Reset packet may be used to force a DCCP endpoint to close a + session [RFC4340], aborting the connection. + + + +Fairhurst, et al. Informational [Page 21] + +RFC 8095 Transport Services March 2017 + + + DCCP supports negotiation of the congestion control profile between + endpoints, to provide plug-and-play congestion control mechanisms. + Examples of specified profiles include "TCP-like" [RFC4341], "TCP- + friendly" [RFC4342], and "TCP-friendly for small packets" [RFC5622]. + Additional mechanisms are recorded in an IANA registry (see + <http://www.iana.org/assignments/dccp-parameters>). + + A lightweight UDP-based encapsulation (DCCP-UDP) has been defined + [RFC6773] that permits DCCP to be used over paths where DCCP is not + natively supported. Support for DCCP in NAPT/NATs is defined in + [RFC4340] and [RFC5595]. Upper-layer protocols specified on top of + DCCP include DTLS [RFC5238], RTP [RFC5762], and Interactive + Connectivity Establishment / Session Description Protocol (ICE/SDP) + [RFC6773]. + +3.6.2. Interface Description + + Functions expected for a DCCP API include: Open, Close, and + Management of the progress a DCCP connection. The Open function + provides feature negotiation, selection of an appropriate Congestion + Control Identifier (CCID) for congestion control, and other + parameters associated with the DCCP connection. A function allows an + application to send DCCP datagrams, including setting the required + checksum coverage and any required options. (DCCP permits sending + datagrams with a zero-length payload.) A function allows reception + of data, including indicating if the data was used or dropped. + Functions can also make the status of a connection visible to an + application, including detection of the maximum packet size and the + ability to perform flow control by detecting a slow receiver at the + sender. + + There is no API currently specified in the RFC Series. + +3.6.3. Transport Features + + The transport features provided by DCCP are: + + o unicast transport, + + o connection-oriented communication with feature negotiation and + application-to-port mapping, + + o signaling of application class for middlebox support (implemented + using Service Codes), + + o port multiplexing, + + o unidirectional or bidirectional communication, + + + +Fairhurst, et al. Informational [Page 22] + +RFC 8095 Transport Services March 2017 + + + o message-oriented delivery, + + o unreliable delivery with drop notification, + + o unordered delivery, + + o flow control (implemented using the slow receiver function), and + + o partial and full payload error detection (with optional strong + integrity check). + +3.7. Transport Layer Security (TLS) and Datagram TLS (DTLS) as a + Pseudotransport + + Transport Layer Security (TLS) [RFC5246] and Datagram TLS (DTLS) + [RFC6347] are IETF protocols that provide several security-related + features to applications. TLS is designed to run on top of a + reliable streaming transport protocol (usually TCP), while DTLS is + designed to run on top of a best-effort datagram protocol (UDP or + DCCP [RFC5238]). At the time of writing, the current version of TLS + is 1.2, defined in [RFC5246]; work on TLS version is 1.3 [TLS-1.3] + nearing completion. DTLS provides nearly identical functionality to + applications; it is defined in [RFC6347] and its current version is + also 1.2. The TLS protocol evolved from the Secure Sockets Layer + (SSL) [RFC6101] protocols developed in the mid-1990s to support + protection of HTTP traffic. + + While older versions of TLS and DTLS are still in use, they provide + weaker security guarantees. [RFC7457] outlines important attacks on + TLS and DTLS. [RFC7525] is a Best Current Practices (BCP) document + that describes secure configurations for TLS and DTLS to counter + these attacks. The recommendations are applicable for the vast + majority of use cases. + +3.7.1. Protocol Description + + Both TLS and DTLS provide the same security features and can thus be + discussed together. The features they provide are: + + o Confidentiality + + o Data integrity + + o Peer authentication (optional) + + o Perfect forward secrecy (optional) + + + + + +Fairhurst, et al. Informational [Page 23] + +RFC 8095 Transport Services March 2017 + + + The authentication of the peer entity can be omitted; a common web + use case is where the server is authenticated and the client is not. + TLS also provides a completely anonymous operation mode in which + neither peer's identity is authenticated. It is important to note + that TLS itself does not specify how a peering entity's identity + should be interpreted. For example, in the common use case of + authentication by means of an X.509 certificate, it is the + application's decision whether the certificate of the peering entity + is acceptable for authorization decisions. + + Perfect forward secrecy, if enabled and supported by the selected + algorithms, ensures that traffic encrypted and captured during a + session at time t0 cannot be later decrypted at time t1 (t1 > t0), + even if the long-term secrets of the communicating peers are later + compromised. + + As DTLS is generally used over an unreliable datagram transport such + as UDP, applications will need to tolerate lost, reordered, or + duplicated datagrams. Like TLS, DTLS conveys application data in a + sequence of independent records. However, because records are mapped + to unreliable datagrams, there are several features unique to DTLS + that are not applicable to TLS: + + o Record replay detection (optional). + + o Record size negotiation (estimates of PMTU and record size + expansion factor). + + o Conveyance of IP don't fragment (DF) bit settings by application. + + o An anti-DoS stateless cookie mechanism (optional). + + Generally, DTLS follows the TLS design as closely as possible. To + operate over datagrams, DTLS includes a sequence number and limited + forms of retransmission and fragmentation for its internal + operations. The sequence number may be used for detecting replayed + information, according to the windowing procedure described in + Section 4.1.2.6 of [RFC6347]. DTLS forbids the use of stream + ciphers, which are essentially incompatible when operating on + independent encrypted records. + +3.7.2. Interface Description + + TLS is commonly invoked using an API provided by packages such as + OpenSSL, wolfSSL, or GnuTLS. Using such APIs entails the + manipulation of several important abstractions, which fall into the + following categories: long-term keys and algorithms, session state, + and communications/connections. + + + +Fairhurst, et al. Informational [Page 24] + +RFC 8095 Transport Services March 2017 + + + Considerable care is required in the use of TLS APIs to ensure + creation of a secure application. The programmer should have at + least a basic understanding of encryption and digital signature + algorithms and their strengths, public key infrastructure (including + X.509 certificates and certificate revocation), and the Sockets API. + See [RFC7525] and [RFC7457], as mentioned above. + + As an example, in the case of OpenSSL, the primary abstractions are + the library itself, method (protocol), session, context, cipher, and + connection. After initializing the library and setting the method, a + cipher suite is chosen and used to configure a context object. + Session objects may then be minted according to the parameters + present in a context object and associated with individual + connections. Depending on how precisely the programmer wishes to + select different algorithmic or protocol options, various levels of + details may be required. + +3.7.3. Transport Features + + Both TLS and DTLS employ a layered architecture. The lower layer is + commonly called the "record protocol". It is responsible for: + + o message fragmentation, + + o authentication and integrity via message authentication codes + (MACs), + + o data encryption, and + + o scheduling transmission using the underlying transport protocol. + + DTLS augments the TLS record protocol with: + + o ordering and replay protection, implemented using sequence + numbers. + + Several protocols are layered on top of the record protocol. These + include the handshake, alert, and change cipher spec protocols. + There is also the data protocol, used to carry application traffic. + The handshake protocol is used to establish cryptographic and + compression parameters when a connection is first set up. In DTLS, + this protocol also has a basic fragmentation and retransmission + capability and a cookie-like mechanism to resist DoS attacks. (TLS + compression is not recommended at present). The alert protocol is + used to inform the peer of various conditions, most of which are + terminal for the connection. The change cipher spec protocol is used + to synchronize changes in cryptographic parameters for each peer. + + + + +Fairhurst, et al. Informational [Page 25] + +RFC 8095 Transport Services March 2017 + + + The data protocol, when used with an appropriate cipher, provides: + + o authentication of one end or both ends of a connection, + + o confidentiality, and + + o cryptographic integrity protection. + + Both TLS and DTLS are unicast-only. + +3.8. Real-Time 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 transport services, including TCP, + UDP, UDP-Lite, DCCP, TLS, and DTLS. + +3.8.1. Protocol Description + + The RTP standard [RFC3550] defines a pair of protocols: RTP and the + RTP Control Protocol (RTCP). The transport does not provide + connection setup, instead relying on out-of-band techniques or + associated control protocols to setup, negotiate parameters, or tear + down a session. + + An RTP sender encapsulates audio/video data into RTP packets to + transport media streams. The RFC Series specifies RTP payload + formats that allow packets to carry a wide range of media and + specifies a wide range of multiplexing, error control, and other + support mechanisms. + + If a frame of media data is large, it will be fragmented into several + RTP packets. Likewise, several small frames may be bundled into a + single RTP packet. + + An RTP receiver collects RTP packets from the network, validates them + for correctness, and sends them to the media decoder input queue. + Missing packet detection is performed by the channel decoder. The + playout buffer is ordered by time stamp and is used to reorder + packets. Damaged frames may be repaired before the media payloads + are decompressed to display or store the data. Some uses of RTP are + able to exploit the partial payload protection features offered by + DCCP and UDP-Lite. + + RTCP is a control protocol that works alongside an RTP flow. Both + the RTP sender and receiver will send RTCP report packets. This is + used to periodically send control information and report performance. + + + + +Fairhurst, et al. Informational [Page 26] + +RFC 8095 Transport Services March 2017 + + + Based on received RTCP feedback, an RTP sender can adjust the + transmission, e.g., perform rate adaptation at the application layer + in the case of congestion. + + An RTCP receiver report (RTCP RR) is returned to the sender + periodically to report key parameters (e.g., the fraction of packets + lost in the last reporting interval, the cumulative number of packets + lost, the highest sequence number received, and the inter-arrival + jitter). The RTCP RR packets also contain timing information that + allows the sender to estimate the network round-trip time (RTT) to + the receivers. + + The interval between reports sent from each receiver tends to be on + the order of a few seconds on average, although this varies with the + session rate, and sub-second reporting intervals are possible for + high rate sessions. The interval is randomized to avoid + synchronization of reports from multiple receivers. + +3.8.2. Interface Description + + There is no standard API defined for RTP or RTCP. Implementations + are typically tightly integrated with a particular application and + closely follow the principles of application-level framing and + integrated layer processing [ClarkArch] in media processing + [RFC2736], error recovery and concealment, rate adaptation, and + security [RFC7202]. Accordingly, RTP implementations tend to be + targeted at particular application domains (e.g., voice-over-IP, + IPTV, or video conferencing), with a feature set optimized for that + domain, rather than being general purpose implementations of the + protocol. + +3.8.3. Transport Features + + The transport features provided by RTP are: + + o unicast, multicast, or IPv4 broadcast (provided by lower-layer + protocol), + + o port multiplexing (provided by lower-layer protocol), + + o unidirectional or bidirectional communication (provided by lower- + layer protocol), + + o message-oriented delivery with support for media types and other + extensions, + + o reliable delivery when using erasure coding or unreliable delivery + with drop notification (if supported by lower-layer protocol), + + + +Fairhurst, et al. Informational [Page 27] + +RFC 8095 Transport Services March 2017 + + + o connection setup with feature negotiation (using associated + protocols) and application-to-port mapping (provided by lower- + layer protocol), + + o segmentation, and + + o performance metric reporting (using associated protocols). + +3.9. Hypertext Transport Protocol (HTTP) over TCP as a Pseudotransport + + The Hypertext Transfer Protocol (HTTP) is an application-level + protocol widely used on the Internet. It provides object-oriented + delivery of discrete data or files. Version 1.1 of the protocol is + specified in [RFC7230] [RFC7231] [RFC7232] [RFC7233] [RFC7234] + [RFC7235], and version 2 is specified in [RFC7540]. HTTP is usually + transported over TCP using ports 80 and 443, although it can be used + with other transports. When used over TCP, it inherits TCP's + properties. + + Application-layer protocols may use HTTP as a substrate with an + existing method and data formats, or specify new methods and data + formats. 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 [RFC7235] and TLS [RFC5246], and the ability of HTTP + to traverse firewalls, which allows it to work over many types of + infrastructure and in cases where an application server often needs + to support HTTP anyway. + + Depending on application need, 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] addresses this issue, provides some + guidelines, and identifies concerns about the use of HTTP standard + ports 80 and 443, the use of the HTTP URL scheme, and interaction + with existing firewalls, proxies, and NATs. + + Representational State Transfer (REST) [REST] is another example of + how applications can use HTTP as a transport protocol. REST is an + architecture style that may be used to build applications using HTTP + as a communication protocol. + +3.9.1. Protocol Description + + The Hypertext Transfer Protocol (HTTP) is a request/response + protocol. A client sends a request containing a request method, URI, + and protocol version followed by message whose design is inspired by + + + +Fairhurst, et al. Informational [Page 28] + +RFC 8095 Transport Services March 2017 + + + MIME (see [RFC7231] for the differences between an HTTP object and a + MIME message), containing information about the client and request + modifiers. The message can also contain a message body carrying + application data. The server responds with a status or error code + followed by a message containing information about the server and + information about the data. This may include a message body. It is + possible to specify a data format for the message body using MIME + media types [RFC2045]. The protocol has additional features; some + relevant to pseudotransport are described below. + + Content negotiation, specified in [RFC7231], is a mechanism provided + by HTTP to allow selection of a representation for a requested + resource. The client and server negotiate acceptable data formats, + character sets, and data encoding (e.g., data can be transferred + compressed using gzip). 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 an object + range request [RFC7233]. The protocol provides powerful cache + control signaling defined in [RFC7234]. + + The persistent connections of HTTP 1.1 and HTTP 2.0 allow multiple + request/response transactions (streams) during the lifetime of a + single HTTP connection. This reduces overhead during connection + establishment and mitigates transport-layer slow-start that would + have otherwise been incurred for each transaction. HTTP 2.0 + connections can multiplex many request/response pairs in parallel on + a single transport connection. Both are important to reduce latency + for HTTP's primary use case. + + HTTP can be combined with security mechanisms, such as TLS (denoted + by HTTPS). This adds protocol properties provided by such a + mechanism (e.g., authentication and encryption). The TLS + Application-Layer Protocol Negotiation (ALPN) extension [RFC7301] can + be used to negotiate the HTTP version within the TLS handshake, + eliminating the latency incurred by additional round-trip exchanges. + Arbitrary cookie strings, included as part of the request headers, + are often used as bearer tokens in HTTP. + +3.9.2. 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 HTTPS is used, the API exposes a + registration of callbacks when a server requests client + authentication and when certificate verification is needed. + + + + +Fairhurst, et al. Informational [Page 29] + +RFC 8095 Transport Services March 2017 + + + The World Wide Web Consortium (W3C) has standardized the + XMLHttpRequest API [XHR]. This API can be used for sending HTTP/ + HTTPS requests and receiving server responses. Besides the XML data + format, the request and response data format can also be JSON, HTML, + and plain text. JavaScript and XMLHttpRequest are ubiquitous + programming models for websites and more general applications where + native code is less attractive. + +3.9.3. Transport Features + + The transport features provided by HTTP, when used as a + pseudotransport, are: + + o unicast transport (provided by the lower-layer protocol, usually + TCP), + + o unidirectional or bidirectional communication, + + o transfer of objects in multiple streams with object content type + negotiation, supporting partial transmission of object ranges, + + o ordered delivery (provided by the lower-layer protocol, usually + TCP), + + o fully reliable delivery (provided by the lower-layer protocol, + usually TCP), + + o flow control (provided by the lower-layer protocol, usually TCP), + and + + o congestion control (provided by the lower-layer protocol, usually + TCP). + + HTTPS (HTTP over TLS) additionally provides the following features + (as provided by TLS): + + o authentication (of one or both ends of a connection), + + o confidentiality, and + + o integrity protection. + + + + + + + + + + +Fairhurst, et al. Informational [Page 30] + +RFC 8095 Transport Services March 2017 + + +3.10. File Delivery over Unidirectional Transport / Asynchronous + Layered Coding (FLUTE/ALC) for Reliable Multicast + + FLUTE/ALC is an IETF Standards Track protocol specified in [RFC6726] + and [RFC5775]. It provides object-oriented delivery of discrete data + or files. Asynchronous Layer Coding (ALC) provides an underlying + reliable transport service and FLUTE a file-oriented specialization + of the ALC service (e.g., to carry associated metadata). [RFC6726] + and [RFC5775] are non-backward-compatible updates of [RFC3926] and + [RFC3450], which are Experimental protocols; these Experimental + protocols are currently largely deployed in the 3GPP Multimedia + Broadcast / Multicast Service (MBMS) (see [MBMS], Section 7) and + similar contexts (e.g., the Japanese ISDB-Tmm standard). + + The FLUTE/ALC protocol has been designed to support massively + scalable reliable bulk data dissemination to receiver groups of + arbitrary size using IP Multicast over any type of delivery network, + including unidirectional networks (e.g., broadcast wireless + channels). However, the FLUTE/ALC protocol also supports point-to- + point unicast transmissions. + + FLUTE/ALC bulk data dissemination has been designed for discrete file + or memory-based "objects". Although FLUTE/ALC is not well adapted to + byte and message streaming, there is an exception: FLUTE/ALC is used + to carry 3GPP Dynamic Adaptive Streaming over HTTP (DASH) when + scalability is a requirement (see [MBMS], Section 5.6). + + FLUTE/ALC's reliability, delivery mode, congestion control, and flow/ + rate control mechanisms can be separately controlled to meet + different application needs. Section 4.1 of [RFC8085] describes + multicast congestion control requirements for UDP. + +3.10.1. Protocol Description + + The FLUTE/ALC protocol works on top of UDP (though it could work on + top of any datagram delivery transport protocol), without requiring + any connectivity from receivers to the sender. Purely unidirectional + networks are therefore supported by FLUTE/ALC. This guarantees + scalability to an unlimited number of receivers in a session, since + the sender behaves exactly the same regardless of the number of + receivers. + + FLUTE/ALC supports the transfer of bulk objects such as file or + in-memory content, using either a push or an on-demand mode. In push + mode, content is sent once to the receivers, while in on-demand mode, + content is sent continuously during periods of time that can greatly + exceed the average time required to download the session objects (see + [RFC5651], Section 4.2). + + + +Fairhurst, et al. Informational [Page 31] + +RFC 8095 Transport Services March 2017 + + + This enables receivers to join a session asynchronously, at their own + discretion, receive the content, and leave the session. In this + case, data content is typically sent continuously, in loops (also + known as "carousels"). FLUTE/ALC also supports the transfer of an + object stream, with loose real-time constraints. This is + particularly useful to carry 3GPP DASH when scalability is a + requirement and unicast transmissions over HTTP cannot be used + ([MBMS], Section 5.6). In this case, packets are sent in sequence + using push mode. FLUTE/ALC is not well adapted to byte and message + streaming, and other solutions could be preferred (e.g., FECFRAME + [RFC6363] with real-time flows). + + The FLUTE file delivery instantiation of ALC provides a metadata + delivery service. Each object of the FLUTE/ALC session is described + in a dedicated entry of a File Delivery Table (FDT), using an XML + format (see [RFC6726], Section 3.2). This metadata can include, but + is not restricted to, a URI attribute (to identify and locate the + object), a media type attribute, a size attribute, an encoding + attribute, or a message digest attribute. Since the set of objects + sent within a session can be dynamic, with new objects being added + and old ones removed, several instances of the FDT can be sent, and a + mechanism is provided to identify a new FDT instance. + + Error detection and verification of the protocol control information + relies on the underlying transport (e.g., UDP checksum). + + To provide robustness against packet loss and improve the efficiency + of the on-demand mode, FLUTE/ALC relies on packet erasure coding + (Application-Layer Forward Error Correction (AL-FEC)). AL-FEC + encoding is proactive (since there is no feedback and therefore no + (N)ACK-based retransmission), and ALC packets containing repair data + are sent along with ALC packets containing source data. Several FEC + Schemes have been standardized; FLUTE/ALC does not mandate the use of + any particular one. Several strategies concerning the transmission + order of ALC source and repair packets are possible, in particular, + in on-demand mode where it can deeply impact the service provided + (e.g., to favor the recovery of objects in sequence or, at the other + extreme, to favor the recovery of all objects in parallel), and + FLUTE/ALC does not mandate nor recommend the use of any particular + one. + + A FLUTE/ALC session is composed of one or more channels, associated + to different destination unicast and/or multicast IP addresses. ALC + packets are sent in those channels at a certain transmission rate, + with a rate that often differs depending on the channel. FLUTE/ALC + does not mandate nor recommend any strategy to select which ALC + packet to send on which channel. FLUTE/ALC can use a multiple rate + congestion control building block (e.g., Wave and Equation Based Rate + + + +Fairhurst, et al. Informational [Page 32] + +RFC 8095 Transport Services March 2017 + + + Control (WEBRC)) to provide congestion control that is feedback free, + where receivers adjust their reception rates individually by joining + and leaving channels associated with the session. To that purpose, + the ALC header provides a specific field to carry congestion-control- + specific information. However, FLUTE/ALC does not mandate the use of + a particular congestion control mechanism although WEBRC is mandatory + to support for the Internet ([RFC6726], Section 1.1.4). FLUTE/ALC is + often used over a network path with pre-provisioned capacity + [RFC8085] where there are no flows competing for capacity. In this + case, a sender-based rate control mechanism and a single channel are + sufficient. + + [RFC6584] provides per-packet authentication, integrity, and anti- + replay protection in the context of the ALC and NORM protocols. + Several mechanisms are proposed that seamlessly integrate into these + protocols using the ALC and NORM header extension mechanisms. + +3.10.2. Interface Description + + The FLUTE/ALC specification does not describe a specific API to + control protocol operation. Although open source and commercial + implementations have specified APIs, there is no IETF-specified API + for FLUTE/ALC. + +3.10.3. Transport Features + + The transport features provided by FLUTE/ALC are: + + o unicast, multicast, anycast, or IPv4 broadcast transmission, + + o object-oriented delivery of discrete data or files and associated + metadata, + + o fully reliable or partially reliable delivery (of file or in- + memory objects), using proactive packet erasure coding (AL-FEC) to + recover from packet erasures, + + o ordered or unordered delivery (of file or in-memory objects), + + o error detection (based on the UDP checksum), + + o per-packet authentication, + + o per-packet integrity, + + o per-packet replay protection, and + + o congestion control for layered flows (e.g., with WEBRC). + + + +Fairhurst, et al. Informational [Page 33] + +RFC 8095 Transport Services March 2017 + + +3.11. NACK-Oriented Reliable Multicast (NORM) + + NORM is an IETF Standards Track protocol specified in [RFC5740]. It + provides object-oriented delivery of discrete data or files. + + 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. Support for bulk data dissemination + includes discrete file or computer memory-based "objects" as well as + byte and message streaming. + + NORM can incorporate packet erasure coding as a part of its selective + Automatic Repeat reQuest (ARQ) in response to negative + acknowledgments from the receiver. The packet erasure coding can + also be proactively applied for forward protection from packet loss. + NORM transmissions are governed by TCP-Friendly Multicast Congestion + Control (TFMCC) [RFC4654]. The reliability, congestion control, and + flow control mechanisms can be separately controlled to meet + different application needs. + +3.11.1. Protocol Description + + The NORM protocol is encapsulated in UDP datagrams and thus provides + multiplexing for multiple sockets on hosts using port numbers. For + 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. 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/rejoin 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 + [RFC5052] 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 + + + + + +Fairhurst, et al. Informational [Page 34] + +RFC 8095 Transport Services March 2017 + + + operation, timer-based feedback suppression is used 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) [RFC5348] principles that are also used in DCCP + [RFC4340]. NORM uses control messages to measure RTT and collect + congestion event information (e.g., reflecting a loss event or ECN + event) from the receiver(s) to support dynamic adjustment or the + rate. TCP-Friendly Multicast Congestion Control (TFMCC) [RFC4654] + provides extra features to support multicast but is functionally + equivalent to TFRC for unicast. + + Error detection and verification of the protocol control information + relies on the on the underlying transport (e.g., UDP checksum). + + The reliability mechanism is decoupled from congestion control. This + allows invocation of alternative arrangements of transport services, + for example, to support, fixed-rate reliable delivery or unreliable + delivery (that may optionally be "better than best effort" via packet + erasure coding) using TFRC. Alternative congestion control + techniques may be applied, for example, TFRC with congestion event + detection based on ECN. + + While NORM provides NACK-based reliability, it also supports a + positive acknowledgment (ACK) mechanism that can be used for receiver + flow control. This mechanism is decoupled from the reliability and + congestion control, supporting applications with different needs. + One example is 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. + +3.11.2. Interface Description + + The NORM specification does not describe a specific 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. + + + + + + + + + + +Fairhurst, et al. Informational [Page 35] + +RFC 8095 Transport Services March 2017 + + +3.11.3. Transport Features + + The transport features provided by NORM are: + + o unicast or multicast transport, + + o unidirectional communication, + + o stream-oriented delivery in a single stream or object-oriented + delivery of in-memory data or file bulk content objects, + + o fully reliable (NACK-based) or partially reliable (using erasure + coding both proactively and as part of ARQ) delivery, + + o unordered delivery, + + o error detection (relies on UDP checksum), + + o segmentation, + + o data bundling (using Nagle's algorithm), + + o flow control (timer-based and/or ACK-based), and + + o congestion control (also supporting fixed-rate reliable or + unreliable delivery). + +3.12. Internet Control Message Protocol (ICMP) + + The Internet Control Message Protocol (ICMP) [RFC792] for IPv4 and + ICMP for IPv6 [RFC4443] are IETF Standards Track protocols. It is a + connectionless unidirectional protocol that delivers individual + messages, without error correction, congestion control, or flow + control. Messages may be sent as unicast, IPv4 broadcast, or + multicast datagrams (IPv4 and IPv6), in addition to anycast + datagrams. + + While ICMP is not typically described as a transport protocol, it + does position itself over the network layer, and the operation of + other transport protocols can be closely linked to the functions + provided by ICMP. + + Transport protocols and upper-layer protocols can use received ICMP + messages to help them make appropriate decisions when network or + endpoint errors are reported, for example, to implement ICMP-based + Path MTU Discovery (PMTUD) [RFC1191] [RFC1981] or assist in + Packetization Layer PMTUD (PLPMTUD) [RFC4821]. Such reactions to + received messages need to protect from off-path data injection + + + +Fairhurst, et al. Informational [Page 36] + +RFC 8095 Transport Services March 2017 + + + [RFC8085] to avoid an application receiving packets created by an + unauthorized third party. An application therefore needs to ensure + that all messages are appropriately validated by checking the payload + of the messages to ensure they are received in response to actually + transmitted traffic (e.g., a reported error condition that + corresponds to a UDP datagram or TCP segment was actually sent by the + application). This requires context [RFC6056], such as local state + about communication instances to each destination (e.g., in TCP, + DCCP, or SCTP). This state is not always maintained by UDP-based + applications [RFC8085]. + +3.12.1. Protocol Description + + ICMP is a connectionless unidirectional protocol. It delivers + independent messages, called "datagrams". Each message is required + to carry a checksum as an integrity check and to protect from + misdelivery to an unintended endpoint. + + ICMP messages typically relay diagnostic information from an endpoint + [RFC1122] or network device [RFC1812] addressed to the sender of a + flow. This usually contains the network protocol header of a packet + that encountered a reported issue. Some formats of messages can also + carry other payload data. Each message carries an integrity check + calculated in the same way as for UDP; this checksum is not optional. + + The RFC Series defines additional IPv6 message formats to support a + range of uses. In the case of IPv6, the protocol incorporates + neighbor discovery [RFC4861] [RFC3971] (provided by ARP for IPv4) and + Multicast Listener Discovery (MLD) [RFC2710] group management + functions (provided by IGMP for IPv4). + + Reliable transmission cannot be assumed. A receiving application + that is unable to run sufficiently fast, or frequently, may miss + messages since there is no flow or congestion control. In addition, + some network devices rate-limit ICMP messages. + +3.12.2. Interface Description + + ICMP processing is integrated in many connection-oriented transports + but, like other functions, needs to be provided by an upper-layer + protocol when using UDP and UDP-Lite. + + On some stacks, a bound socket also allows a UDP application to be + notified when ICMP error messages are received for its transmissions + [RFC8085]. + + + + + + +Fairhurst, et al. Informational [Page 37] + +RFC 8095 Transport Services March 2017 + + + Any response to ICMP error messages ought to be robust to temporary + routing failures (sometimes called "soft errors"), e.g., transient + ICMP "unreachable" messages ought to not normally cause a + communication abort [RFC5461] [RFC8085]. + +3.12.3. Transport Features + + ICMP does not provide any transport service directly to applications. + Used together with other transport protocols, it provides + transmission of control, error, and measurement data between + endpoints or from devices along the path to one endpoint. + +4. Congestion Control + + Congestion control is critical to the stable operation of the + Internet. A variety of mechanisms are used to provide the congestion + control needed by many Internet transport protocols. Congestion is + detected based on sensing of network conditions, whether through + explicit or implicit feedback. The congestion control mechanisms + that can be applied by different transport protocols are largely + orthogonal to the choice of transport protocol. This section + provides an overview of the congestion control mechanisms available + to the protocols described in Section 3. + + Many protocols use a separate window to determine the maximum sending + rate that is allowed by the congestion control. The used congestion + control mechanism will increase the congestion window if feedback is + received that indicates that the currently used network path is not + congested and will reduce the window otherwise. Window-based + mechanisms often increase their window slowing over multiple RTTs, + while decreasing strongly when the first indication of congestion is + received. One example is an Additive Increase Multiplicative + Decrease (AIMD) scheme, where the window is increased by a certain + number of packets/bytes for each data segment that has been + successfully transmitted, while the window decreases multiplicatively + on the occurrence of a congestion event. This can lead to a rather + unstable, oscillating sending rate but will resolve a congestion + situation quickly. Examples of window-based AIMD schemes include TCP + NewReno [RFC5681], TCP Cubic [CUBIC] (the default mechanism for TCP + in Linux), and CCID 2 specified for DCCP [RFC4341]. + + Some classes of applications prefer to use a transport service that + allows sending at a more stable rate that is slowly varied in + response to congestion. Rate-based methods offer this type of + congestion control and have been defined based on the loss ratio and + observed round-trip time, such as TFRC [RFC5348] and TFRC-SP + + + + + +Fairhurst, et al. Informational [Page 38] + +RFC 8095 Transport Services March 2017 + + + [RFC4828]. These methods utilize a throughput equation to determine + the maximum acceptable rate. Such methods are used with DCCP CCID 3 + [RFC4342], CCID 4 [RFC5622], WEBRC [RFC3738], and other applications. + + Another class of applications prefers a transport service that yields + to other (higher-priority) traffic, such as interactive + transmissions. While most traffic in the Internet uses loss-based + congestion control and therefore tends to fill the network buffers + (to a certain level if Active Queue Management (AQM) is used), low- + priority congestion control methods often react to changes in delay + as an earlier indication of congestion. This approach tends to + induce less loss than a loss-based method but does generally not + compete well with loss-based traffic across shared bottleneck links. + Therefore, methods such as LEDBAT [RFC6817] are deployed in the + Internet for scavenger traffic that aims to only utilize otherwise + unused capacity. + +5. Transport Features + + The transport protocol features described in this document can be + used as a basis for defining common transport features. These are + listed below with the protocols supporting them: + + o Control Functions + + * Addressing + + + unicast (TCP, MPTCP, UDP, UDP-Lite, SCTP, DCCP, TLS, RTP, + HTTP, ICMP) + + + multicast (UDP, UDP-Lite, RTP, ICMP, FLUTE/ALC, NORM). Note + that, as TLS and DTLS are unicast-only, there is no widely + deployed mechanism for supporting the features listed under + the Security bullet (below) when using multicast addressing. + + + IPv4 broadcast (UDP, UDP-Lite, ICMP) + + + anycast (UDP, UDP-Lite). Connection-oriented protocols such + as TCP and DCCP have also been deployed using anycast + addressing, with the risk that routing changes may cause + connection failure. + + * Association type + + + connection-oriented (TCP, MPTCP, DCCP, SCTP, TLS, RTP, HTTP, + NORM) + + + connectionless (UDP, UDP-Lite, FLUTE/ALC) + + + +Fairhurst, et al. Informational [Page 39] + +RFC 8095 Transport Services March 2017 + + + * Multihoming support + + + resilience and mobility (MPTCP, SCTP) + + + load balancing (MPTCP) + + + address family multiplexing (MPTCP, SCTP) + + * Middlebox cooperation + + + application-class signaling to middleboxes (DCCP) + + + error condition signaling from middleboxes and routers to + endpoints (ICMP) + + * Signaling + + + control information and error signaling (ICMP) + + + application performance reporting (RTP) + + o Delivery + + * Reliability + + + fully reliable delivery (TCP, MPTCP, SCTP, TLS, HTTP, FLUTE/ + ALC, NORM) + + + partially reliable delivery (SCTP, NORM) + + - using packet erasure coding (RTP, FLUTE/ALC, NORM) + + - with specified policy for dropped messages (SCTP) + + + unreliable delivery (SCTP, UDP, UDP-Lite, DCCP, RTP) + + - with drop notification to sender (SCTP, DCCP, RTP) + + + error detection + + - checksum for error detection (TCP, MPTCP, UDP, UDP-Lite, + SCTP, DCCP, TLS, DTLS, FLUTE/ALC, NORM, ICMP) + + - partial payload checksum protection (UDP-Lite, DCCP). + Some uses of RTP can exploit partial payload checksum + protection feature to provide a corruption-tolerant + transport service. + + + + +Fairhurst, et al. Informational [Page 40] + +RFC 8095 Transport Services March 2017 + + + - checksum optional (UDP). Possible with IPv4 and, in + certain cases, with IPv6. + + * Ordering + + + ordered delivery (TCP, MPTCP, SCTP, TLS, RTP, HTTP, FLUTE) + + + unordered delivery permitted (UDP, UDP-Lite, SCTP, DCCP, + RTP, NORM) + + * Type/framing + + + stream-oriented delivery (TCP, MPTCP, SCTP, TLS, HTTP) + + - with multiple streams per association (SCTP, HTTP2) + + + message-oriented delivery (UDP, UDP-Lite, SCTP, DCCP, DTLS, + RTP) + + + object-oriented delivery of discrete data or files and + associated metadata (HTTP, FLUTE/ALC, NORM) + + - with partial delivery of object ranges (HTTP) + + * Directionality + + + unidirectional (UDP, UDP-Lite, DCCP, RTP, FLUTE/ALC, NORM) + + + bidirectional (TCP, MPTCP, SCTP, TLS, HTTP) + + o Transmission control + + * flow control (TCP, MPTCP, SCTP, DCCP, TLS, RTP, HTTP) + + * congestion control (TCP, MPTCP, SCTP, DCCP, RTP, FLUTE/ALC, + NORM). Congestion control can also provided by the transport + supporting an upper-layer transport (e.g., TLS, RTP, HTTP). + + * segmentation (TCP, MPTCP, SCTP, TLS, RTP, HTTP, FLUTE/ALC, + NORM) + + * data/message bundling (TCP, MPTCP, SCTP, TLS, HTTP) + + * stream scheduling prioritization (SCTP, HTTP2) + + * endpoint multiplexing (MPTCP) + + + + + +Fairhurst, et al. Informational [Page 41] + +RFC 8095 Transport Services March 2017 + + + o Security + + * authentication of one end of a connection (TLS, DTLS, FLUTE/ + ALC) + + * authentication of both ends of a connection (TLS, DTLS) + + * confidentiality (TLS, DTLS) + + * cryptographic integrity protection (TLS, DTLS) + + * replay protection (TLS, DTLS, FLUTE/ALC) + +6. IANA Considerations + + This document does not require any IANA actions. + +7. Security Considerations + + This document surveys existing transport protocols and protocols + providing transport-like services. Confidentiality, integrity, and + authenticity are among the features provided by those services. This + document does not specify any new features or mechanisms for + providing these features. Each RFC referenced by this document + discusses the security considerations of the specification it + contains. + +8. Informative References + + [ClarkArch] + Clark, D. and D. Tennenhouse, "Architectural + Considerations for a New Generation of Protocols", + Proceedings of ACM SIGCOMM, DOI 10.1145/99517.99553, 1990. + + [CUBIC] Rhee, I., Xu, L., Ha, S., Zimmermann, A., Eggert, L., and + R. Scheffenegger, "CUBIC for Fast Long-Distance Networks", + Work in Progress, draft-ietf-tcpm-cubic-04, February 2017. + + [MBMS] 3GPP, "Multimedia Broadcast/Multicast Service (MBMS); + Protocols and codecs", 3GPP TS 26.346, 2015, + <http://www.3gpp.org/DynaReport/26346.htm>. + + [NAT-SUPP] Stewart, R., Tuexen, M., and I. Ruengeler, "Stream Control + Transmission Protocol (SCTP) Network Address Translation + Support", Work in Progress, draft-ietf-tsvwg-natsupp-09, + May 2016. + + + + + +Fairhurst, et al. Informational [Page 42] + +RFC 8095 Transport Services March 2017 + + + [POSIX] IEEE, "Standard for Information Technology -- Portable + Operating System Interface (POSIX(R)) Base Specifications, + Issue 7", IEEE 1003.1, DOI 10.1109/ieeestd.2016.7582338, + <http://ieeexplore.ieee.org/document/7582338/>. + + [REST] Fielding, R., "Architectural Styles and the Design of + Network-based Software Architectures, Chapter 5: + Representational State Transfer", Ph.D. + Dissertation, University of California, Irvine, 2000. + + [RFC768] Postel, J., "User Datagram Protocol", STD 6, RFC 768, + DOI 10.17487/RFC0768, August 1980, + <http://www.rfc-editor.org/info/rfc768>. + + [RFC792] Postel, J., "Internet Control Message Protocol", STD 5, + RFC 792, DOI 10.17487/RFC0792, September 1981, + <http://www.rfc-editor.org/info/rfc792>. + + [RFC793] Postel, J., "Transmission Control Protocol", STD 7, + RFC 793, DOI 10.17487/RFC0793, September 1981, + <http://www.rfc-editor.org/info/rfc793>. + + [RFC1071] Braden, R., Borman, D., and C. Partridge, "Computing the + Internet checksum", RFC 1071, DOI 10.17487/RFC1071, + September 1988, <http://www.rfc-editor.org/info/rfc1071>. + + [RFC1122] Braden, R., Ed., "Requirements for Internet Hosts - + Communication Layers", STD 3, RFC 1122, + DOI 10.17487/RFC1122, October 1989, + <http://www.rfc-editor.org/info/rfc1122>. + + [RFC1191] Mogul, J. and S. Deering, "Path MTU discovery", RFC 1191, + DOI 10.17487/RFC1191, November 1990, + <http://www.rfc-editor.org/info/rfc1191>. + + [RFC1812] Baker, F., Ed., "Requirements for IP Version 4 Routers", + RFC 1812, DOI 10.17487/RFC1812, June 1995, + <http://www.rfc-editor.org/info/rfc1812>. + + [RFC1981] McCann, J., Deering, S., and J. Mogul, "Path MTU Discovery + for IP version 6", RFC 1981, DOI 10.17487/RFC1981, August + 1996, <http://www.rfc-editor.org/info/rfc1981>. + + [RFC2018] Mathis, M., Mahdavi, J., Floyd, S., and A. Romanow, "TCP + Selective Acknowledgment Options", RFC 2018, + DOI 10.17487/RFC2018, October 1996, + <http://www.rfc-editor.org/info/rfc2018>. + + + + +Fairhurst, et al. Informational [Page 43] + +RFC 8095 Transport Services March 2017 + + + [RFC2045] Freed, N. and N. Borenstein, "Multipurpose Internet Mail + Extensions (MIME) Part One: Format of Internet Message + Bodies", RFC 2045, DOI 10.17487/RFC2045, November 1996, + <http://www.rfc-editor.org/info/rfc2045>. + + [RFC2460] Deering, S. and R. Hinden, "Internet Protocol, Version 6 + (IPv6) Specification", RFC 2460, DOI 10.17487/RFC2460, + December 1998, <http://www.rfc-editor.org/info/rfc2460>. + + [RFC2710] Deering, S., Fenner, W., and B. Haberman, "Multicast + Listener Discovery (MLD) for IPv6", RFC 2710, + DOI 10.17487/RFC2710, October 1999, + <http://www.rfc-editor.org/info/rfc2710>. + + [RFC2736] Handley, M. and C. Perkins, "Guidelines for Writers of RTP + Payload Format Specifications", BCP 36, RFC 2736, + DOI 10.17487/RFC2736, December 1999, + <http://www.rfc-editor.org/info/rfc2736>. + + [RFC3168] Ramakrishnan, K., Floyd, S., and D. Black, "The Addition + of Explicit Congestion Notification (ECN) to IP", + RFC 3168, DOI 10.17487/RFC3168, September 2001, + <http://www.rfc-editor.org/info/rfc3168>. + + [RFC3205] Moore, K., "On the use of HTTP as a Substrate", BCP 56, + RFC 3205, DOI 10.17487/RFC3205, February 2002, + <http://www.rfc-editor.org/info/rfc3205>. + + [RFC3260] Grossman, D., "New Terminology and Clarifications for + Diffserv", RFC 3260, DOI 10.17487/RFC3260, April 2002, + <http://www.rfc-editor.org/info/rfc3260>. + + [RFC3436] Jungmaier, A., Rescorla, E., and M. Tuexen, "Transport + Layer Security over Stream Control Transmission Protocol", + RFC 3436, DOI 10.17487/RFC3436, December 2002, + <http://www.rfc-editor.org/info/rfc3436>. + + [RFC3450] Luby, M., Gemmell, J., Vicisano, L., Rizzo, L., and J. + Crowcroft, "Asynchronous Layered Coding (ALC) Protocol + Instantiation", RFC 3450, DOI 10.17487/RFC3450, December + 2002, <http://www.rfc-editor.org/info/rfc3450>. + + [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. + Jacobson, "RTP: A Transport Protocol for Real-Time + Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550, + July 2003, <http://www.rfc-editor.org/info/rfc3550>. + + + + + +Fairhurst, et al. Informational [Page 44] + +RFC 8095 Transport Services March 2017 + + + [RFC3738] Luby, M. and V. Goyal, "Wave and Equation Based Rate + Control (WEBRC) Building Block", RFC 3738, + DOI 10.17487/RFC3738, April 2004, + <http://www.rfc-editor.org/info/rfc3738>. + + [RFC3758] Stewart, R., Ramalho, M., Xie, Q., Tuexen, M., and P. + Conrad, "Stream Control Transmission Protocol (SCTP) + Partial Reliability Extension", RFC 3758, + DOI 10.17487/RFC3758, May 2004, + <http://www.rfc-editor.org/info/rfc3758>. + + [RFC3828] Larzon, L-A., Degermark, M., Pink, S., Jonsson, L-E., Ed., + and G. Fairhurst, Ed., "The Lightweight User Datagram + Protocol (UDP-Lite)", RFC 3828, DOI 10.17487/RFC3828, July + 2004, <http://www.rfc-editor.org/info/rfc3828>. + + [RFC3926] Paila, T., Luby, M., Lehtonen, R., Roca, V., and R. Walsh, + "FLUTE - File Delivery over Unidirectional Transport", + RFC 3926, DOI 10.17487/RFC3926, October 2004, + <http://www.rfc-editor.org/info/rfc3926>. + + [RFC3971] Arkko, J., Ed., Kempf, J., Zill, B., and P. Nikander, + "SEcure Neighbor Discovery (SEND)", RFC 3971, + DOI 10.17487/RFC3971, March 2005, + <http://www.rfc-editor.org/info/rfc3971>. + + [RFC4336] Floyd, S., Handley, M., and E. Kohler, "Problem Statement + for the Datagram Congestion Control Protocol (DCCP)", + RFC 4336, DOI 10.17487/RFC4336, March 2006, + <http://www.rfc-editor.org/info/rfc4336>. + + [RFC4340] Kohler, E., Handley, M., and S. Floyd, "Datagram + Congestion Control Protocol (DCCP)", RFC 4340, + DOI 10.17487/RFC4340, March 2006, + <http://www.rfc-editor.org/info/rfc4340>. + + [RFC4341] Floyd, S. and E. Kohler, "Profile for Datagram Congestion + Control Protocol (DCCP) Congestion Control ID 2: TCP-like + Congestion Control", RFC 4341, DOI 10.17487/RFC4341, March + 2006, <http://www.rfc-editor.org/info/rfc4341>. + + [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, + DOI 10.17487/RFC4342, March 2006, + <http://www.rfc-editor.org/info/rfc4342>. + + + + + +Fairhurst, et al. Informational [Page 45] + +RFC 8095 Transport Services March 2017 + + + [RFC4443] Conta, A., Deering, S., and M. Gupta, Ed., "Internet + Control Message Protocol (ICMPv6) for the Internet + Protocol Version 6 (IPv6) Specification", RFC 4443, + DOI 10.17487/RFC4443, March 2006, + <http://www.rfc-editor.org/info/rfc4443>. + + [RFC4654] Widmer, J. and M. Handley, "TCP-Friendly Multicast + Congestion Control (TFMCC): Protocol Specification", + RFC 4654, DOI 10.17487/RFC4654, August 2006, + <http://www.rfc-editor.org/info/rfc4654>. + + [RFC4820] Tuexen, M., Stewart, R., and P. Lei, "Padding Chunk and + Parameter for the Stream Control Transmission Protocol + (SCTP)", RFC 4820, DOI 10.17487/RFC4820, March 2007, + <http://www.rfc-editor.org/info/rfc4820>. + + [RFC4821] Mathis, M. and J. Heffner, "Packetization Layer Path MTU + Discovery", RFC 4821, DOI 10.17487/RFC4821, March 2007, + <http://www.rfc-editor.org/info/rfc4821>. + + [RFC4828] Floyd, S. and E. Kohler, "TCP Friendly Rate Control + (TFRC): The Small-Packet (SP) Variant", RFC 4828, + DOI 10.17487/RFC4828, April 2007, + <http://www.rfc-editor.org/info/rfc4828>. + + [RFC4861] Narten, T., Nordmark, E., Simpson, W., and H. Soliman, + "Neighbor Discovery for IP version 6 (IPv6)", RFC 4861, + DOI 10.17487/RFC4861, September 2007, + <http://www.rfc-editor.org/info/rfc4861>. + + [RFC4895] Tuexen, M., Stewart, R., Lei, P., and E. Rescorla, + "Authenticated Chunks for the Stream Control Transmission + Protocol (SCTP)", RFC 4895, DOI 10.17487/RFC4895, August + 2007, <http://www.rfc-editor.org/info/rfc4895>. + + [RFC4960] Stewart, R., Ed., "Stream Control Transmission Protocol", + RFC 4960, DOI 10.17487/RFC4960, September 2007, + <http://www.rfc-editor.org/info/rfc4960>. + + [RFC5052] Watson, M., Luby, M., and L. Vicisano, "Forward Error + Correction (FEC) Building Block", RFC 5052, + DOI 10.17487/RFC5052, August 2007, + <http://www.rfc-editor.org/info/rfc5052>. + + + + + + + + +Fairhurst, et al. Informational [Page 46] + +RFC 8095 Transport Services March 2017 + + + [RFC5061] Stewart, R., Xie, Q., Tuexen, M., Maruyama, S., and M. + Kozuka, "Stream Control Transmission Protocol (SCTP) + Dynamic Address Reconfiguration", RFC 5061, + DOI 10.17487/RFC5061, September 2007, + <http://www.rfc-editor.org/info/rfc5061>. + + [RFC5097] Renker, G. and G. Fairhurst, "MIB for the UDP-Lite + protocol", RFC 5097, DOI 10.17487/RFC5097, January 2008, + <http://www.rfc-editor.org/info/rfc5097>. + + [RFC5238] Phelan, T., "Datagram Transport Layer Security (DTLS) over + the Datagram Congestion Control Protocol (DCCP)", + RFC 5238, DOI 10.17487/RFC5238, May 2008, + <http://www.rfc-editor.org/info/rfc5238>. + + [RFC5246] Dierks, T. and E. Rescorla, "The Transport Layer Security + (TLS) Protocol Version 1.2", RFC 5246, + DOI 10.17487/RFC5246, August 2008, + <http://www.rfc-editor.org/info/rfc5246>. + + [RFC5348] Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP + Friendly Rate Control (TFRC): Protocol Specification", + RFC 5348, DOI 10.17487/RFC5348, September 2008, + <http://www.rfc-editor.org/info/rfc5348>. + + [RFC5461] Gont, F., "TCP's Reaction to Soft Errors", RFC 5461, + DOI 10.17487/RFC5461, February 2009, + <http://www.rfc-editor.org/info/rfc5461>. + + [RFC5595] Fairhurst, G., "The Datagram Congestion Control Protocol + (DCCP) Service Codes", RFC 5595, DOI 10.17487/RFC5595, + September 2009, <http://www.rfc-editor.org/info/rfc5595>. + + [RFC5596] Fairhurst, G., "Datagram Congestion Control Protocol + (DCCP) Simultaneous-Open Technique to Facilitate NAT/ + Middlebox Traversal", RFC 5596, DOI 10.17487/RFC5596, + September 2009, <http://www.rfc-editor.org/info/rfc5596>. + + [RFC5622] Floyd, S. and E. Kohler, "Profile for Datagram Congestion + Control Protocol (DCCP) Congestion ID 4: TCP-Friendly Rate + Control for Small Packets (TFRC-SP)", RFC 5622, + DOI 10.17487/RFC5622, August 2009, + <http://www.rfc-editor.org/info/rfc5622>. + + [RFC5651] Luby, M., Watson, M., and L. Vicisano, "Layered Coding + Transport (LCT) Building Block", RFC 5651, + DOI 10.17487/RFC5651, October 2009, + <http://www.rfc-editor.org/info/rfc5651>. + + + +Fairhurst, et al. Informational [Page 47] + +RFC 8095 Transport Services March 2017 + + + [RFC5681] Allman, M., Paxson, V., and E. Blanton, "TCP Congestion + Control", RFC 5681, DOI 10.17487/RFC5681, September 2009, + <http://www.rfc-editor.org/info/rfc5681>. + + [RFC5740] Adamson, B., Bormann, C., Handley, M., and J. Macker, + "NACK-Oriented Reliable Multicast (NORM) Transport + Protocol", RFC 5740, DOI 10.17487/RFC5740, November 2009, + <http://www.rfc-editor.org/info/rfc5740>. + + [RFC5762] Perkins, C., "RTP and the Datagram Congestion Control + Protocol (DCCP)", RFC 5762, DOI 10.17487/RFC5762, April + 2010, <http://www.rfc-editor.org/info/rfc5762>. + + [RFC5775] Luby, M., Watson, M., and L. Vicisano, "Asynchronous + Layered Coding (ALC) Protocol Instantiation", RFC 5775, + DOI 10.17487/RFC5775, April 2010, + <http://www.rfc-editor.org/info/rfc5775>. + + [RFC6056] Larsen, M. and F. Gont, "Recommendations for Transport- + Protocol Port Randomization", BCP 156, RFC 6056, + DOI 10.17487/RFC6056, January 2011, + <http://www.rfc-editor.org/info/rfc6056>. + + [RFC6083] Tuexen, M., Seggelmann, R., and E. Rescorla, "Datagram + Transport Layer Security (DTLS) for Stream Control + Transmission Protocol (SCTP)", RFC 6083, + DOI 10.17487/RFC6083, January 2011, + <http://www.rfc-editor.org/info/rfc6083>. + + [RFC6093] Gont, F. and A. Yourtchenko, "On the Implementation of the + TCP Urgent Mechanism", RFC 6093, DOI 10.17487/RFC6093, + January 2011, <http://www.rfc-editor.org/info/rfc6093>. + + [RFC6101] Freier, A., Karlton, P., and P. Kocher, "The Secure + Sockets Layer (SSL) Protocol Version 3.0", RFC 6101, + DOI 10.17487/RFC6101, August 2011, + <http://www.rfc-editor.org/info/rfc6101>. + + [RFC6347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer + Security Version 1.2", RFC 6347, DOI 10.17487/RFC6347, + January 2012, <http://www.rfc-editor.org/info/rfc6347>. + + [RFC6356] Raiciu, C., Handley, M., and D. Wischik, "Coupled + Congestion Control for Multipath Transport Protocols", + RFC 6356, DOI 10.17487/RFC6356, October 2011, + <http://www.rfc-editor.org/info/rfc6356>. + + + + + +Fairhurst, et al. Informational [Page 48] + +RFC 8095 Transport Services March 2017 + + + [RFC6363] Watson, M., Begen, A., and V. Roca, "Forward Error + Correction (FEC) Framework", RFC 6363, + DOI 10.17487/RFC6363, October 2011, + <http://www.rfc-editor.org/info/rfc6363>. + + [RFC6458] Stewart, R., Tuexen, M., Poon, K., Lei, P., and V. + Yasevich, "Sockets API Extensions for the Stream Control + Transmission Protocol (SCTP)", RFC 6458, + DOI 10.17487/RFC6458, December 2011, + <http://www.rfc-editor.org/info/rfc6458>. + + [RFC6525] Stewart, R., Tuexen, M., and P. Lei, "Stream Control + Transmission Protocol (SCTP) Stream Reconfiguration", + RFC 6525, DOI 10.17487/RFC6525, February 2012, + <http://www.rfc-editor.org/info/rfc6525>. + + [RFC6582] Henderson, T., Floyd, S., Gurtov, A., and Y. Nishida, "The + NewReno Modification to TCP's Fast Recovery Algorithm", + RFC 6582, DOI 10.17487/RFC6582, April 2012, + <http://www.rfc-editor.org/info/rfc6582>. + + [RFC6584] Roca, V., "Simple Authentication Schemes for the + Asynchronous Layered Coding (ALC) and NACK-Oriented + Reliable Multicast (NORM) Protocols", RFC 6584, + DOI 10.17487/RFC6584, April 2012, + <http://www.rfc-editor.org/info/rfc6584>. + + [RFC6726] Paila, T., Walsh, R., Luby, M., Roca, V., and R. Lehtonen, + "FLUTE - File Delivery over Unidirectional Transport", + RFC 6726, DOI 10.17487/RFC6726, November 2012, + <http://www.rfc-editor.org/info/rfc6726>. + + [RFC6773] Phelan, T., Fairhurst, G., and C. Perkins, "DCCP-UDP: A + Datagram Congestion Control Protocol UDP Encapsulation for + NAT Traversal", RFC 6773, DOI 10.17487/RFC6773, November + 2012, <http://www.rfc-editor.org/info/rfc6773>. + + [RFC6817] Shalunov, S., Hazel, G., Iyengar, J., and M. Kuehlewind, + "Low Extra Delay Background Transport (LEDBAT)", RFC 6817, + DOI 10.17487/RFC6817, December 2012, + <http://www.rfc-editor.org/info/rfc6817>. + + [RFC6824] Ford, A., Raiciu, C., Handley, M., and O. Bonaventure, + "TCP Extensions for Multipath Operation with Multiple + Addresses", RFC 6824, DOI 10.17487/RFC6824, January 2013, + <http://www.rfc-editor.org/info/rfc6824>. + + + + + +Fairhurst, et al. Informational [Page 49] + +RFC 8095 Transport Services March 2017 + + + [RFC6897] Scharf, M. and A. Ford, "Multipath TCP (MPTCP) Application + Interface Considerations", RFC 6897, DOI 10.17487/RFC6897, + March 2013, <http://www.rfc-editor.org/info/rfc6897>. + + [RFC6935] Eubanks, M., Chimento, P., and M. Westerlund, "IPv6 and + UDP Checksums for Tunneled Packets", RFC 6935, + DOI 10.17487/RFC6935, April 2013, + <http://www.rfc-editor.org/info/rfc6935>. + + [RFC6936] Fairhurst, G. and M. Westerlund, "Applicability Statement + for the Use of IPv6 UDP Datagrams with Zero Checksums", + RFC 6936, DOI 10.17487/RFC6936, April 2013, + <http://www.rfc-editor.org/info/rfc6936>. + + [RFC6951] Tuexen, M. and R. Stewart, "UDP Encapsulation of Stream + Control Transmission Protocol (SCTP) Packets for End-Host + to End-Host Communication", RFC 6951, + DOI 10.17487/RFC6951, May 2013, + <http://www.rfc-editor.org/info/rfc6951>. + + [RFC7053] Tuexen, M., Ruengeler, I., and R. Stewart, "SACK- + IMMEDIATELY Extension for the Stream Control Transmission + Protocol", RFC 7053, DOI 10.17487/RFC7053, November 2013, + <http://www.rfc-editor.org/info/rfc7053>. + + [RFC7202] Perkins, C. and M. Westerlund, "Securing the RTP + Framework: Why RTP Does Not Mandate a Single Media + Security Solution", RFC 7202, DOI 10.17487/RFC7202, April + 2014, <http://www.rfc-editor.org/info/rfc7202>. + + [RFC7230] Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer + Protocol (HTTP/1.1): Message Syntax and Routing", + RFC 7230, DOI 10.17487/RFC7230, June 2014, + <http://www.rfc-editor.org/info/rfc7230>. + + [RFC7231] Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer + Protocol (HTTP/1.1): Semantics and Content", RFC 7231, + DOI 10.17487/RFC7231, June 2014, + <http://www.rfc-editor.org/info/rfc7231>. + + [RFC7232] Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer + Protocol (HTTP/1.1): Conditional Requests", RFC 7232, + DOI 10.17487/RFC7232, June 2014, + <http://www.rfc-editor.org/info/rfc7232>. + + + + + + + +Fairhurst, et al. Informational [Page 50] + +RFC 8095 Transport Services March 2017 + + + [RFC7233] Fielding, R., Ed., Lafon, Y., Ed., and J. Reschke, Ed., + "Hypertext Transfer Protocol (HTTP/1.1): Range Requests", + RFC 7233, DOI 10.17487/RFC7233, June 2014, + <http://www.rfc-editor.org/info/rfc7233>. + + [RFC7234] Fielding, R., Ed., Nottingham, M., Ed., and J. Reschke, + Ed., "Hypertext Transfer Protocol (HTTP/1.1): Caching", + RFC 7234, DOI 10.17487/RFC7234, June 2014, + <http://www.rfc-editor.org/info/rfc7234>. + + [RFC7235] Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer + Protocol (HTTP/1.1): Authentication", RFC 7235, + DOI 10.17487/RFC7235, June 2014, + <http://www.rfc-editor.org/info/rfc7235>. + + [RFC7301] Friedl, S., Popov, A., Langley, A., and E. Stephan, + "Transport Layer Security (TLS) Application-Layer Protocol + Negotiation Extension", RFC 7301, DOI 10.17487/RFC7301, + July 2014, <http://www.rfc-editor.org/info/rfc7301>. + + [RFC7323] Borman, D., Braden, B., Jacobson, V., and R. + Scheffenegger, Ed., "TCP Extensions for High Performance", + RFC 7323, DOI 10.17487/RFC7323, September 2014, + <http://www.rfc-editor.org/info/rfc7323>. + + [RFC7414] Duke, M., Braden, R., Eddy, W., Blanton, E., and A. + Zimmermann, "A Roadmap for Transmission Control Protocol + (TCP) Specification Documents", RFC 7414, + DOI 10.17487/RFC7414, February 2015, + <http://www.rfc-editor.org/info/rfc7414>. + + [RFC7457] Sheffer, Y., Holz, R., and P. Saint-Andre, "Summarizing + Known Attacks on Transport Layer Security (TLS) and + Datagram TLS (DTLS)", RFC 7457, DOI 10.17487/RFC7457, + February 2015, <http://www.rfc-editor.org/info/rfc7457>. + + [RFC7496] Tuexen, M., Seggelmann, R., Stewart, R., and S. Loreto, + "Additional Policies for the Partially Reliable Stream + Control Transmission Protocol Extension", RFC 7496, + DOI 10.17487/RFC7496, April 2015, + <http://www.rfc-editor.org/info/rfc7496>. + + [RFC7525] Sheffer, Y., Holz, R., and P. Saint-Andre, + "Recommendations for Secure Use of Transport Layer + Security (TLS) and Datagram Transport Layer Security + (DTLS)", BCP 195, RFC 7525, DOI 10.17487/RFC7525, May + 2015, <http://www.rfc-editor.org/info/rfc7525>. + + + + +Fairhurst, et al. Informational [Page 51] + +RFC 8095 Transport Services March 2017 + + + [RFC7540] Belshe, M., Peon, R., and M. Thomson, Ed., "Hypertext + Transfer Protocol Version 2 (HTTP/2)", RFC 7540, + DOI 10.17487/RFC7540, May 2015, + <http://www.rfc-editor.org/info/rfc7540>. + + [RFC8085] Eggert, L., Fairhurst, G., and G. Shepherd, "UDP Usage + Guidelines", BCP 145, RFC 8085, DOI 10.17487/RFC8085, + March 2017, <http://www.rfc-editor.org/info/rfc8085>. + + [SCTP-DTLS-ENCAPS] + Tuexen, M., Stewart, R., Jesup, R., and S. Loreto, "DTLS + Encapsulation of SCTP Packets", Work in Progress, + draft-ietf-tsvwg-sctp-dtls-encaps-09, January 2015. + + [SCTP-NDATA] + Stewart, R., Tuexen, M., Loreto, S., and R. Seggelmann, + "Stream Schedulers and User Message Interleaving for the + Stream Control Transmission Protocol", Work in Progress, + draft-ietf-tsvwg-sctp-ndata-08, October 2016. + + [TCP-SPEC] Eddy, W., Ed., "Transmission Control Protocol + Specification", Work in Progress, draft-ietf-tcpm- + rfc793bis-04, December 2016. + + [TLS-1.3] Rescorla, E., "The Transport Layer Security (TLS) Protocol + Version 1.3", Work in Progress, draft-ietf-tls-tls13-18, + October 2016. + + [WEBRTC-TRANS] + Alvestrand, H., "Transports for WebRTC", Work in + Progress, draft-ietf-rtcweb-transports-17, October 2016. + + [XHR] van Kesteren, A., Aubourg, J., Song, J., and H. Steen, + "XMLHttpRequest Level 1", World Wide Web Consortium NOTE- + XMLHttpRequest-20161006, October 2016, + <http://www.w3.org/TR/XMLHttpRequest/>. + + + + + + + + + + + + + + + +Fairhurst, et al. Informational [Page 52] + +RFC 8095 Transport Services March 2017 + + +Acknowledgments + + Thanks to Joe Touch, Michael Welzl, Spencer Dawkins, and the TAPS + working group for the comments, feedback, and discussion. This work + is supported by the European Commission under grant agreement No. + 318627 mPlane and from the Horizon 2020 research and innovation + program under grant agreements No. 644334 (NEAT) and No. 688421 + (MAMI). This support does not imply endorsement. + +Contributors + + In addition to the editors, this document is the work of Brian + Adamson, Dragana Damjanovic, Kevin Fall, Simone Ferlin-Oliviera, + Ralph Holz, Olivier Mehani, Karen Nielsen, Colin Perkins, Vincent + Roca, and Michael Tuexen. + + o Section 3.2 on MPTCP was contributed by Simone Ferlin-Oliviera + (ferlin@simula.no) and Olivier Mehani + (olivier.mehani@nicta.com.au). + + o Section 3.3 on UDP was contributed by Kevin Fall + (kfall@kfall.com). + + o Section 3.5 on SCTP was contributed by Michael Tuexen (tuexen@fh- + muenster.de) and Karen Nielsen (karen.nielsen@tieto.com). + + o Section 3.7 on TLS and DTLS was contributed by Ralph Holz + (ralph.holz@nicta.com.au) and Olivier Mehani + (olivier.mehani@nicta.com.au). + + o Section 3.8 on RTP contains contributions from Colin Perkins + (csp@csperkins.org). + + o Section 3.9 on HTTP was contributed by Dragana Damjanovic + (ddamjanovic@mozilla.com). + + o Section 3.10 on FLUTE/ALC was contributed by Vincent Roca + (vincent.roca@inria.fr). + + o Section 3.11 on NORM was contributed by Brian Adamson + (brian.adamson@nrl.navy.mil). + + + + + + + + + + +Fairhurst, et al. Informational [Page 53] + +RFC 8095 Transport Services March 2017 + + +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 + Gloriastrasse 35 + 8092 Zurich + Switzerland + + Email: mirja.kuehlewind@tik.ee.ethz.ch + + + + + + + + + + + + + + + + + + + + + + + + + +Fairhurst, et al. Informational [Page 54] + |