Network Working Group G. Fairhurst, Ed. Internet-Draft University of Aberdeen Intended status: Informational B. Trammell, Ed. Expires: December 11, 2015 M. Kuehlewind, Ed. ETH Zurich June 09, 2015 Services provided by IETF transport protocols and congestion control mechanisms draft-ietf-taps-transports-05 Abstract This document describes services provided by existing IETF protocols and congestion control mechanisms. It is designed to help application and network stack programmers and to inform the work of the IETF TAPS Working Group. Status of This Memo This Internet-Draft is submitted in full conformance with the provisions of BCP 78 and BCP 79. Internet-Drafts are working documents of the Internet Engineering Task Force (IETF). Note that other groups may also distribute working documents as Internet-Drafts. The list of current Internet- Drafts is at http://datatracker.ietf.org/drafts/current/. Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as "work in progress." This Internet-Draft will expire on December 11, 2015. Copyright Notice Copyright (c) 2015 IETF Trust and the persons identified as the document authors. All rights reserved. This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents (http://trustee.ietf.org/license-info) in effect on the date of publication of this document. Please review these documents carefully, as they describe your rights and restrictions with respect to this document. Code Components extracted from this document must include Simplified BSD License text as described in Section 4.e of Fairhurst, et al. Expires December 11, 2015 [Page 1] Internet-Draft TAPS Transports June 2015 the Trust Legal Provisions and are provided without warranty as described in the Simplified BSD License. Table of Contents 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4 3. Existing Transport Protocols . . . . . . . . . . . . . . . . 4 3.1. Transport Control Protocol (TCP) . . . . . . . . . . . . 4 3.1.1. Protocol Description . . . . . . . . . . . . . . . . 5 3.1.2. Interface description . . . . . . . . . . . . . . . . 6 3.1.3. Transport Protocol Components . . . . . . . . . . . . 6 3.2. Multipath TCP (MPTCP) . . . . . . . . . . . . . . . . . . 7 3.2.1. Protocol Description . . . . . . . . . . . . . . . . 7 3.2.2. Interface Description . . . . . . . . . . . . . . . . 7 3.2.3. Transport Protocol Components . . . . . . . . . . . . 8 3.3. Stream Control Transmission Protocol (SCTP) . . . . . . . 9 3.3.1. Protocol Description . . . . . . . . . . . . . . . . 9 3.3.2. Interface Description . . . . . . . . . . . . . . . . 11 3.3.3. Transport Protocol Components . . . . . . . . . . . . 13 3.4. User Datagram Protocol (UDP) . . . . . . . . . . . . . . 13 3.4.1. Protocol Description . . . . . . . . . . . . . . . . 14 3.4.2. Interface Description . . . . . . . . . . . . . . . . 14 3.4.3. Transport Protocol Components . . . . . . . . . . . . 15 3.5. Lightweight User Datagram Protocol (UDP-Lite) . . . . . . 15 3.5.1. Protocol Description . . . . . . . . . . . . . . . . 15 3.5.2. Interface Description . . . . . . . . . . . . . . . . 16 3.5.3. Transport Protocol Components . . . . . . . . . . . . 16 3.6. Datagram Congestion Control Protocol (DCCP) . . . . . . . 17 3.6.1. Protocol Description . . . . . . . . . . . . . . . . 17 3.6.2. Interface Description . . . . . . . . . . . . . . . . 19 3.6.3. Transport Protocol Components . . . . . . . . . . . . 19 3.7. Realtime Transport Protocol (RTP) . . . . . . . . . . . . 19 3.8. NACK-Oriented Reliable Multicast (NORM) . . . . . . . . . 20 3.8.1. Protocol Description . . . . . . . . . . . . . . . . 20 3.8.2. Interface Description . . . . . . . . . . . . . . . . 21 3.8.3. Transport Protocol Components . . . . . . . . . . . . 21 3.9. Transport Layer Security (TLS) and Datagram TLS (DTLS) as a pseudotransport . . . . . . . . . . . . . . . . . . . . 22 3.9.1. Protocol Description . . . . . . . . . . . . . . . . 23 3.9.2. Interface Description . . . . . . . . . . . . . . . . 23 3.10. Hypertext Transport Protocol (HTTP) over TCP as a pseudotransport . . . . . . . . . . . . . . . . . . . . . 24 3.10.1. Protocol Description . . . . . . . . . . . . . . . . 25 3.10.2. Interface Description . . . . . . . . . . . . . . . 26 3.10.3. Transport Protocol Components . . . . . . . . . . . 26 3.11. WebSockets . . . . . . . . . . . . . . . . . . . . . . . 27 3.11.1. Protocol Description . . . . . . . . . . . . . . . . 27 Fairhurst, et al. Expires December 11, 2015 [Page 2] Internet-Draft TAPS Transports June 2015 3.11.2. Interface Description . . . . . . . . . . . . . . . 27 3.11.3. Transport Protocol Components . . . . . . . . . . . 27 4. Transport Service Features . . . . . . . . . . . . . . . . . 27 4.1. Complete Protocol Feature Matrix . . . . . . . . . . . . 29 5. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 31 6. Security Considerations . . . . . . . . . . . . . . . . . . . 31 7. Contributors . . . . . . . . . . . . . . . . . . . . . . . . 31 8. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 31 9. References . . . . . . . . . . . . . . . . . . . . . . . . . 32 9.1. Normative References . . . . . . . . . . . . . . . . . . 32 9.2. Informative References . . . . . . . . . . . . . . . . . 32 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 38 1. Introduction Most Internet applications make use of the Transport Services provided by TCP (a reliable, in-order stream protocol) or UDP (an unreliable datagram protocol). We use the term "Transport Service" to mean the end-to-end service provided to an application by the transport layer. That service can only be provided correctly if information about the intended usage is supplied from the application. The application may determine this information at design time, compile time, or run time, and may include guidance on whether a feature is required, a preference by the application, or something in between. Examples of features of Transport Services are reliable delivery, ordered delivery, content privacy to in-path devices, integrity protection, and minimal latency. The IETF has defined a wide variety of transport protocols beyond TCP and UDP, including SCTP, DCCP, MP-TCP, and UDP-Lite. Transport services may be provided directly by these transport protocols, or layered on top of them using protocols such as WebSockets (which runs over TCP), RTP (over TCP or UDP) or WebRTC data channels (which run over SCTP over DTLS over UDP or TCP). Services built on top of UDP or UDP-Lite typically also need to specify additional mechanisms, including a congestion control mechanism (such as a windowed congestion control, TFRC or LEDBAT congestion control mechanism). This extends the set of available Transport Services beyond those provided to applications by TCP and UDP. Transport protocols can also be differentiated by the features of the services they provide: for instance, SCTP offers a message-based service providing full or partial reliability and allowing to minimize the head of line blocking due to the support of unordered and unordered message delivery within multiple streams, UDP-Lite provides partial integrity protection, and LEDBAT can provide low- priority "scavenger" communication. Fairhurst, et al. Expires December 11, 2015 [Page 3] Internet-Draft TAPS Transports June 2015 2. Terminology The following terms are defined throughout this document, and in subsequent documents produced by TAPS describing the composition and decomposition of transport services. [EDITOR'S NOTE: we may want to add definitions for the different kinds of interfaces that are important here.] Transport Service Feature: a specific end-to-end feature that a transport service provides to its clients. Examples include confidentiality, reliable delivery, ordered delivery, message- versus-stream orientation, etc. Transport Service: a set of transport service features, without an association to any given framing protocol, which provides a complete service to an application. Transport Protocol: an implementation that provides one or more different transport services using a specific framing and header format on the wire. Transport Protocol Component: an implementation of a transport service feature within a protocol. Transport Service Instance: an arrangement of transport protocols with a selected set of features and configuration parameters that implements a single transport service, e.g. a protocol stack (RTP over UDP). 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 protocol and transport protocol frameworks. [EDITOR'S NOTE: Contributions to the subsections below are welcome] 3.1. Transport Control Protocol (TCP) TCP is an IETF standards track transport protocol. [RFC0793] 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, Fairhurst, et al. Expires December 11, 2015 [Page 4] Internet-Draft TAPS Transports June 2015 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, providing a three way handshake to allow a client and server to set up a connection, and mechanisms for orderly completion and immediate teardown of a connection. TCP is defined by a family of RFCs [RFC4614]. TCP provides multiplexing to multiple sockets on each host using port numbers. An active TCP session is identified by its four-tuple of local and remote IP addresses and local port and remote port numbers. The destination port during connection setup has a different role as it is often used to indicate the requested service. TCP partitions a continuous stream of bytes into segments, sized to fit in IP packets. ICMP-based PathMTU discovery [RFC1191][RFC1981] as well as Packetization Layer Path MTU Discovery (PMTUD) [RFC4821] are supported. 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 TCP's reliable, ordered delivery of data in a stream. TCP Selective Acknowledgment [RFC2018] extends this mechanism by making it possible to identify missing segments more precisely, reducing spurious retransmission. Receiver flow control is provided by a sliding window: limiting 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 64KB. All TCP senders provide Congestion Control: This uses a separate window, where each time congestion is detected, this congestion window is reduced. A receiver detects congestion using one of three mechanisms: A retransmission timer, detection of loss (interpreted as a congestion signal), or Explicit Congestion Notification (ECN) [RFC3168] to provide early signaling (see [I-D.ietf-aqm-ecn-benefits]) A TCP protocol instance can be extended [RFC4614] and tuned. Some features are sender-side only, requiring no negotiation with the receiver; some are receiver-side only, some are explicitly negotiated during connection setup. Fairhurst, et al. Expires December 11, 2015 [Page 5] Internet-Draft TAPS Transports June 2015 By default, TCP segment partitioning uses Nagle's algorithm [RFC0896] 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 enable smoother interactive sessions. [EDITOR'S NOTE: add URGENT and PUSH flag (note [RFC6093] says SHOULD NOT use due to the range of TCP implementations that process TCP urgent indications differently.) ] A checksum provides an Integrity Check and is mandatory across the entire packet. The TCP checksum does not support partial corruption protection as in DCCP/UDP-Lite). This check protects from misdelivery of data corrupted data, but is relatively weak, and applications that require end to end integrity of data are recommended to include a stronger integrity check of their payload data. A TCP service is unicast. 3.1.2. Interface description A User/TCP Interface is defined in [RFC0793] providing six user commands: Open, Send, Receive, Close, Status. This interface does not describe configuration of TCP options or parameters beside use of the PUSH and URGENT flags. In API implementations derived from the BSD Sockets API, TCP sockets are created using the "SOCK_STREAM" socket type. The features used by a protocol instance may be set and tuned via this API. (more on the API goes here) 3.1.3. Transport Protocol Components The transport protocol components provided by TCP are: o unicast o connection setup with feature negotiation and application-to-port mapping o port multiplexing o reliable delivery Fairhurst, et al. Expires December 11, 2015 [Page 6] Internet-Draft TAPS Transports June 2015 o error detection (checksum) o segmentation o stream-oriented delivery in a single stream o data bundling (Nagle's algorithm) o flow control o congestion control [EDITOR'S NOTE: discussion of how to map this to features and TAPS: what does the higher layer need to decide? what can the transport layer decide based on global settings? what must the transport layer decide based on network characteristics?] 3.2. Multipath TCP (MPTCP) Multipath TCP [RFC6824] is an extension for TCP to support multi- homing. It is designed to be as transparent as possible to middle- boxes. It does so by establishing regular TCP flows between a pair of source/destination endpoints, and multiplexing the application's stream over these flows. 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, and advertise other available IP addresses and establish new sessions between pairs of endpoints. 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 - enable or disable MPTCP; - bind a socket to one or more selected local endpoints; - query local and remote endpoint addresses; - get a unique connection identifier (similar to an address-port pair for TCP). The document also recommend the use of extensions defined for SCTP [RFC6458] (see next section) to deal with multihoming. [AUTHOR'S NOTE: research work, and some implementation, also suggest that the scheduling algorithm, as well as the path manager, are Fairhurst, et al. Expires December 11, 2015 [Page 7] Internet-Draft TAPS Transports June 2015 configurable options that should be exposed to higher layer. Should this be discussed here?] 3.2.3. Transport Protocol Components [AUTHOR'S NOTE: shouldn't it be "service feature"?] As an extension to TCP, MPTCP provides mostly the same components. By establishing multiple sessions between available endpoints, it can additionally provide soft failover solutions should one of the paths become unusable. In addition, by multiplexing one byte stream over separate paths, it can achieve a higher throughput than TCP in certain situations (note however that coupled congestion control [RFC6356] 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, an additional delay and higher jitter might be observed observed before in-order delivery of data to the applications. The transport protocol components provided by MPTCP therefore are: o unicast o connection setup with feature negotiation and application-to-port mapping o port multiplexing o reliable delivery o error detection (checksum) o segmentation o stream-oriented delivery in a single stream o flow control o congestion control o endpoint multiplexing of a single byte stream (higher throughput) o resilience to network failure and/or handovers [AUTHOR'S NOTE: it is unclear whether MPTCP has to provide data bundling.] [AUTHOR'S NOTE: AF muliplexing? sub-flows can be started over IPv4 or IPv6 for the same session] Fairhurst, et al. Expires December 11, 2015 [Page 8] Internet-Draft TAPS Transports June 2015 3.3. Stream Control Transmission Protocol (SCTP) SCTP is a message oriented standards track transport protocol and the base protocol is specified in [RFC4960]. It supports multi-homing to handle path failures. An SCTP association has multiple unidirectional streams in each direction and provides in-sequence delivery of user messages only within each stream. This allows to minimize head of line blocking. SCTP is extensible and the currently defined extensions include mechanisms for dynamic re-configurations of streams [RFC6525] and IP-addresses [RFC5061]. Furthermore, the extension specified in [RFC3758] introduces the concept of partial reliability for user messages. SCTP was originally developed for transporting telephony signalling messages and is deployed in telephony signalling networks, especially in mobile telephony networks. Additionally, it is used in the WebRTC framework for data channels and is therefore deployed in all WEB- browsers supporting WebRTC. 3.3.1. 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 do and only supports unicast. SCTP uses the 32-bit CRC32c for protecting SCTP packets against bit errors. This is stronger than the 16-bit checksums used by TCP or UDP. However, a partial checksum coverage as provided by DCCP or UDP-Lite is not supported. SCTP has been designed with extensibility in mind. Each SCTP packet starts with a single common header containing the port numbers, a verification tag and the CRC32c checksum. This common header is followed by a sequence of chunks. Each chunk consists of a type field, flags, a length field and a value. [RFC4960] defines how a receiver processes chunks with an unknown chunk type. The support of extensions can be negotiated during the SCTP handshake. SCTP provides a message-oriented service. Multiple small user messages can be bundled into a single SCTP packet to improve the efficiency. For example, this bundling may be done by delaying user messages at the sender side similar to the Nagle algorithm used by TCP. User messages which would result in IP packets larger than the MTU will be fragmented at the sender side and reassembled at the receiver side. There is no protocol limit on the user message size. ICMP-based path MTU discovery as specified for IPv4 in [RFC1191] and for IPv6 in [RFC1981] as well as packetization layer path MTU Fairhurst, et al. Expires December 11, 2015 [Page 9] Internet-Draft TAPS Transports June 2015 discovery as specified in [RFC4821] with probe packets using the padding chunks defined the [RFC4820] are supported. [RFC4960] specifies a TCP friendly congestion control to protect the network against overload. SCTP also uses a sliding window flow control to protect receivers against overflow. Each SCTP association has between 1 and 65536 uni-directional streams in each direction. The number of streams can be different in each direction. Every user-message is sent on a particular stream. User messages can be sent un-ordered or ordered upon request by the upper layer. Un-ordered messages can be delivered as soon as they are completely received. Only all ordered messages sent on the same stream are delivered at the receiver in the same order as sent by the sender. For user messages not requiring fragmentation, this minimises head of line blocking. The base protocol defined in [RFC4960] doesn't allow interleaving of user-messages, which results in sending a large message on one stream can block the sending of user messages on other streams. [I-D.ietf-tsvwg-sctp-ndata] overcomes this limitation. Furthermore, [I-D.ietf-tsvwg-sctp-ndata] specifies multiple algorithms for the sender side selection of which streams to send data from supporting a variety of scheduling algorithms including priority based ones. The stream re- configuration extension defined in [RFC6525] allows to reset streams during the lifetime of an association and to increase the number of streams, if the number of streams negotiated in the SCTP handshake is not sufficient. According to [RFC4960], each user message sent is either delivered to the receiver or, in case of excessive retransmissions, the association is terminated in a non-graceful way, similar to the TCP behaviour. In addition to this reliable transfer, the partial reliability extension defined in [RFC3758] allows the sender to abandon user messages. The application can specify the policy for abandoning user messages. Examples for these policies include: o Limiting the time a user message is dealt with by the sender. o Limiting the number of retransmissions for each fragment of a user message. If the number of retransmissions is limited to 0, one gets a service similar to UDP. o Abandoning messages of lower priority in case of a send buffer shortage. SCTP supports multi-homing. Each SCTP end-point uses a list of IP- addresses and a single port number. These addresses can be any mixture of IPv4 and IPv6 addresses. These addresses are negotiated Fairhurst, et al. Expires December 11, 2015 [Page 10] Internet-Draft TAPS Transports June 2015 during the handshake and the address re-configuration extension specified in [RFC5061] in combination with [RFC4895] can be used to change these addresses in an authenticated way during the livetime of an SCTP association. This allows for transport layer mobility. Multiple addresses are used for improved resilience. If a remote address becomes unreachable, the traffic is switched over to a reachable one, if one exists. Each SCTP end-point supervises continuously the reachability of all peer addresses using a heartbeat mechanism. For securing user messages, the use of TLS over SCTP has been specified in [RFC3436]. However, this solution does not support all services provided by SCTP (for example un-ordered delivery or partial reliability), and therefore the use of DTLS over SCTP has been specified in [RFC6083] to overcome these limitations. When using DTLS over SCTP, the application can use almost all services provided by SCTP. [I-D.ietf-tsvwg-natsupp] defines a methods for end-hosts and middleboxes to provide for NAT support for SCTP over IPv4. For legacy NAT traversal, [RFC6951] defines the UDP encapsulation of SCTP-packets. Alternatively, SCTP packets can be encapsulated in DTLS packets as specified in [I-D.ietf-tsvwg-sctp-dtls-encaps]. The latter encapsulation is used with in the WebRTC context. Having a well defined API is also a feature provided by SCTP as described in the next subsection. 3.3.2. Interface Description [RFC4960] defines an abstract API for the base protocol. An extension to the BSD Sockets API is defined in [RFC6458] and covers: o the base protocol defined in [RFC4960]. o the SCTP Partial Reliability extension defined in [RFC3758]. o the SCTP Authentication extension defined in [RFC4895]. o the SCTP Dynamic Address Reconfiguration extension defined in [RFC5061]. For the following SCTP protocol extensions the BSD Sockets API extension is defined in the document specifying the protocol extensions: o the SCTP SACK-IMMEDIATELY extension defined in [RFC7053]. Fairhurst, et al. Expires December 11, 2015 [Page 11] Internet-Draft TAPS Transports June 2015 o the SCTP Stream Reconfiguration extension defined in [RFC6525]. o the UDP Encapsulation of SCTP packets extension defined in [RFC6951]. o the additional PR-SCTP policies defined in [I-D.ietf-tsvwg-sctp-prpolicies]. Future documents describing SCTP protocol 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 as 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 similar to user messages but are specifically marked as notifications. A couple of 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 allow dealing with the SCTP specific information without using ancillary data in the form of additional cmsgs, which are also defined. 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 behaviour 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. Fairhurst, et al. Expires December 11, 2015 [Page 12] Internet-Draft TAPS Transports June 2015 3.3.3. Transport Protocol Components The transport protocol components provided by SCTP are: o unicast o connection setup with feature negotiation and application-to-port mapping o port multiplexing o reliable or partially reliable delivery o ordered and unordered delivery within a stream o support for multiple concurrent streams o support for stream scheduling prioritization o flow control o message-oriented delivery o congestion control o user message bundling o user message fragmentation and reassembly o strong error detection (CRC32C) o transport layer multihoming for resilience o transport layer mobility [EDITOR'S NOTE: update this list.] 3.4. User Datagram Protocol (UDP) The User Datagram Protocol (UDP) [RFC0768] [RFC2460] is an IETF standards track transport protocol. It provides a uni-directional, datagram protocol which preserves message boundaries. It provides none of the following transport features: error correction, congestion control, or flow control. It can be used to send broadcast datagrams (IPv4) or multicast datagrams (IPv4 and IPv6), in addition to unicast (and anycast) datagrams. IETF guidance on the use of UDP is provided in[RFC5405]. UDP is widely implemented and widely used by common applications, especially DNS. Fairhurst, et al. Expires December 11, 2015 [Page 13] Internet-Draft TAPS Transports June 2015 3.4.1. Protocol Description UDP is a connection-less protocol which maintains message boundaries, with no connection setup or feature negotiation. The protocol uses independent messages, ordinarily called datagrams. The lack of error control and flow control implies messages may be damaged, re-ordered, lost, or duplicated in transit. A receiving application unable to run sufficiently fast or frequently may miss messages. The lack of congestion handling implies UDP traffic may cause the loss of messages from other protocols (e.g., TCP) when sharing the same network paths. UDP traffic can also cause the loss of other UDP traffic in the same or other flows for the same reasons. Messages with bit errors are ordinarily detected by an invalid end- to-end checksum and are discarded before being delivered to an application. There are some exceptions to this general rule, however. UDP-Lite (see [RFC3828], and below) provides the ability for portions of the message contents to be exempt from checksum coverage. It is also possible to create UDP datagrams with no checksum, and while this is generally discouraged [RFC1122] [RFC5405], certain special cases permit its use [RFC6935]. The checksum support considerations for omitting the checksum are defined in [RFC6936]. 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. On transmission, UDP encapsulates each datagram into an IP packet, which may in turn be fragmented by IP. Applications concerned with fragmentation or that have other requirements such as receiver flow control, congestion control, PathMTU discovery/PLPMTUD, support for ECN, etc need to be provided by protocols other than UDP [RFC5405]. 3.4.2. Interface Description [RFC0768] describes basic requirements for an API for UDP. Guidance on use of common APIs is provided in [RFC5405]. A UDP endpoint consists of a tuple of (IP address, port number). Demultiplexing using multiple abstract endpoints (sockets) on the same IP address are 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) binding 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. Fairhurst, et al. Expires December 11, 2015 [Page 14] Internet-Draft TAPS Transports June 2015 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 [RFC5405]. 3.4.3. Transport Protocol Components The transport protocol components provided by UDP are: o unidirectional o port multiplexing o 2-tuple endpoints o IPv4 broadcast, multicast and anycast o IPv6 multicast and anycast o IPv6 jumbograms o message-oriented delivery o error detection (checksum) o checksum optional 3.5. Lightweight User Datagram Protocol (UDP-Lite) The Lightweight User Datagram Protocol (UDP-Lite) [RFC3828] is an IETF standards track transport protocol. UDP-Lite provides a bidirectional set of logical unicast or multicast message streams over a datagram protocol. IETF guidance on the use of UDP-Lite is provided in [RFC5405]. 3.5.1. Protocol Description UDP-Lite is a connection-less datagram protocol, with no connection setup or feature negotiation. The protocol use messages, rather than a byte-stream. Each stream of messages is independently managed, therefore retransmission does not hold back data sent using other logical streams. It provides multiplexing to multiple sockets on each host using port numbers. An active UDP-Lite session is identified by its four-tuple Fairhurst, et al. Expires December 11, 2015 [Page 15] Internet-Draft TAPS Transports June 2015 of local and remote IP addresses and local port and remote port numbers. UDP-Lite fragments packets into IP packets, constrained by the maximum size of IP packet. UDP-Lite changes the semantics of the UDP "payload length" field to that of a "checksum coverage length" field. Otherwise, UDP-Lite is semantically identical to UDP. Applications using UDP-Lite therefore can not make assumptions regarding the correctness of the data received in the insensitive part of the UDP-Lite payload. 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 [RFC5405]. 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 support error tolerate payload corruption when used over paths that include error-prone links, another application is when header integrity checks are required, but payload integrity is provided by some other mechanism (e.g. [RFC6936]. A UDP-Lite service may support IPv4 broadcast, multicast, anycast and unicast. 3.5.2. Interface Description There is no current API specified in the RFC Series, but guidance on use of common APIs is provided in [RFC5405]. 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: at the sender, this specifies the intended checksum coverage, with the remaining unprotected part of the payload called the "error- insensitive part". The checksum coverage may also be made visible to the application via the UDP-Lite MIB module [RFC5097]. 3.5.3. Transport Protocol Components The transport protocol components provided by UDP-Lite are: o unicast o IPv4 broadcast, multicast and anycast o port multiplexing Fairhurst, et al. Expires December 11, 2015 [Page 16] Internet-Draft TAPS Transports June 2015 o non-reliable, non-ordered delivery o message-oriented delivery o partial integrity protection 3.6. Datagram Congestion Control Protocol (DCCP) Datagram Congestion Control Protocol (DCCP) [RFC4340] is an IETF standards track bidirectional transport protocol that provides unicast connections of congestion-controlled unreliable messages. [EDITOR'S NOTE: Gorry Fairhurst signed up as a contributor for this section.] The DCCP Problem Statement describes the goals that DCCP sought to address [RFC4336]. 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]. It 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 functions (feature negotiation, path state management, RTT calculation, PMTUD, etc): This allows applications to use a compatible method defining how they send packets and where suitable to choose common algorithms to manage their functions. Examples of suitable applications include interactive applications, streaming media or on-line games [RFC4336]. 3.6.1. Protocol Description DCCP is a connection-oriented datagram protocol, providing a three way handshake to allow a client and server to set up a connection, and mechanisms for orderly completion and immediate teardown of a connection. The protocol is defined by a family of RFCs. It provides multiplexing to multiple sockets on each host using port numbers. An active DCCP session is identified by its four-tuple of local and remote IP addresses and local port and remote port numbers. At connection setup, DCCP also exchanges the the service code [RFC5595] mechanism to allow transport instantiations to indicate the service treatment that is expected from the network. The protocol segments data into messages, typically sized to fit in IP packets, but which may be fragmented providing they are less than the A DCCP interface MAY allow applications to request fragmentation for packets larger than PMTU, but not larger than the maximum packet Fairhurst, et al. Expires December 11, 2015 [Page 17] Internet-Draft TAPS Transports June 2015 size allowed by the current congestion control mechanism (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 ordered or unordered delivery of data, and does not itself provide retransmission. There is a Data Checksum option, which contains a strong CRC, lets endpoints detect application data corruption. It also supports reduced checksum coverage, a partial integrity mechanisms similar to UDP-lIte. Receiver flow control is supported: limiting the amount of unacknowledged data that can be outstanding at a given time. A DCCP protocol instance can be extended [RFC4340] and tuned. Some features are sender-side only, requiring no negotiation with the receiver; some are receiver-side only, some are explicitly negotiated during connection setup. DCCP supports negotiation of the congestion control profile, to provide Plug and Play congestion control mechanisms. examples of specified profiles include [RFC4341] [RFC4342] [RFC5662]. All IETF- defined methods provide Congestion Control. DCCP use a Connect packet to start a session, and permits half- connections that allow each client to choose 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 field [RFC5595] designed to allow middle boxes and endpoints to identify the characteristics required by a session. A lightweight UDP-based encapsulation (DCCP-UDP) has been defined [RFC6773] that permits DCCP to be used over paths where it is not natively supported. Support in NAPT/NATs is defined in [RFC4340] and [RFC5595]. Upper layer protocols specified on top of DCCP include: DTLS [RFC5595], RTP [RFC5672], ICE/SDP [RFC6773]. A DCCP service is unicast. A common packet format has allowed tools to evolve that can read and interpret DCCP packets (e.g. Wireshark). Fairhurst, et al. Expires December 11, 2015 [Page 18] Internet-Draft TAPS Transports June 2015 3.6.2. Interface Description API characteristics include: - Datagram transmission. - Notification of the current maximum packet size. - Send and reception of zero- length payloads. - Set the Slow Receiver flow control at a receiver. - Detect a Slow receiver at the sender. There is no current API specified in the RFC Series. 3.6.3. Transport Protocol Components The transport protocol components provided by DCCP are: o unicast o connection setup with feature negotiation and application-to-port mapping o Service Codes o port multiplexing o non-reliable, ordered delivery o flow control (slow receiver function) o drop notification o timestamps o message-oriented delivery o partial integrity protection 3.7. Realtime Transport Protocol (RTP) RTP provides an end-to-end network transport service, suitable for applications transmitting real-time data, such as audio, video or data, over multicast or unicast network services, including TCP, UDP, UDP-Lite, DCCP. [EDITOR'S NOTE: Varun Singh signed up as contributor for this section. Given the complexity of RTP, suggest to have an abbreviated section here contrasting RTP with other transports, and focusing on those features that are RTP-unique.] Fairhurst, et al. Expires December 11, 2015 [Page 19] Internet-Draft TAPS Transports June 2015 3.8. NACK-Oriented Reliable Multicast (NORM) NORM is an IETF standards track protocol specified in [RFC5740]. The protocol was designed to support reliable bulk data dissemination to receiver groups using IP Multicast but also provides for point-to- point unicast operation. Its support for bulk data dissemination includes discrete file or computer memory-based "objects" as well as byte- and message-streaming. NORM is designed to incorporate packet erasure coding as an inherent part of its selective ARQ in response to receiver negative acknowledgements. The packet erasure coding can also be proactively applied for forward protection from packet loss. NORM transmissions are governed by TCP-friendly congestion control. NORM's reliability, congestion control, and flow control mechanism are distinct components and can be separately controlled to meet different application needs. 3.8.1. Protocol Description [EDITOR'S NOTE: needs to be more clear about the application of FEC and packet erasure coding; expand ARQ.] The NORM protocol is encapsulated in UDP datagrams and thus provides multiplexing for multiple sockets on hosts using port numbers. For purposes of loosely coordinated IP Multicast, NORM is not strictly connection-oriented although per-sender state is maintained by receivers for protocol operation. [RFC5740] does not specify a handshake protocol for connection establishment and separate session initiation can be used to coordinate port numbers. However, in-band "client-server" style connection establishment can be accomplished with the NORM congestion control signaling messages using port binding techniques like those for TCP client-server connections. NORM supports bulk "objects" such as file or in-memory content but also can treat a stream of data as a logical bulk object for purposes of packet erasure coding. In the case of stream transport, NORM can support either byte streams or message streams where application- defined message boundary information is carried in the NORM protocol messages. This allows the receiver(s) to join/re-join and recover message boundaries mid-stream as needed. Application content is carried and identified by the NORM protocol with encoding symbol identifiers depending upon the Forward Error Correction (FEC) Scheme [RFC3452] configured. NORM uses NACK-based selective ARQ to reliably deliver the application content to the receiver(s). NORM proactively measures round-trip timing information to scale ARQ timers appropriately and to support congestion control. For multicast operation, timer-based feedback suppression is uses to achieve group size scaling with low feedback traffic levels. The feedback suppression is not applied for unicast operation. Fairhurst, et al. Expires December 11, 2015 [Page 20] Internet-Draft TAPS Transports June 2015 NORM uses rate-based congestion control based upon the TCP-Friendly Rate Control (TFRC) [RFC4324] principles that are also used in DCCP [RFC4340]. NORM uses control messages to measure RTT and collect congestion event (e..g, loss event, ECN event, etc) information from the receiver(s) to support dynamic rate control adjustment. The TCP- Friendly Multicast Congestion Control (TFMCC) [RFC4654] used provides some extra features to support multicast but is functionally equivalent to TFRC in the unicast case. NORM's reliability mechanism is decoupled from congestion control. This allows alternative arrangements of transport services to be invoked. For example, fixed-rate reliable delivery can be supported or unreliable (but optionally "better than best effort" via packet erasure coding) delivery with rate-control per TFRC can be achieved. Additionally, alternative congestion control techniques may be applied. For example, TFRC rate control with congestion event detection based on ECN for links with high packet loss (e.g., wireless) has been implemented and demonstrated with NORM. While NORM is NACK-based for reliability transfer, it also supports a positive acknowledgment (ACK) mechanism that can be used for receiver flow control. Again, since this mechanism is decoupled from the reliability and congestion control, applications that have different needs in this aspect can use the protocol differently. One example is the use of NORM for quasi-reliable delivery where timely delivery of newer content may be favored over completely reliable delivery of older content within buffering and RTT constraints. 3.8.2. Interface Description The NORM specification does not describe a specific application programming interface (API) to control protocol operation. A freely- available, open source reference implementation of NORM is available at https://www.nrl.navy.mil/itd/ncs/products/norm, and a documented API is provided for this implementation. While a sockets-like API is not currently documented, the existing API supports the necessary functions for that to be implemented. 3.8.3. Transport Protocol Components The transport protocol components provided by NORM are: o unicast o multicast o port multiplexing (UDP ports) Fairhurst, et al. Expires December 11, 2015 [Page 21] Internet-Draft TAPS Transports June 2015 o reliable delivery o unordered delivery of in-memory data or file bulk content objects o error detection (UDP checksum) o segmentation o stream-oriented delivery in a single stream o object-oriented delivery of discrete data or file items o data bundling (Nagle's algorithm) o flow control (timer-based and/or ack-based) o congestion control o packet erasure coding (both proactively and as part of ARQ) 3.9. Transport Layer Security (TLS) and Datagram TLS (DTLS) as a pseudotransport Transport Layer Security (TLS) and Datagram TLS are IETF protocols that provide several security-related features to applications. TLS is designed to run on top of TCP, DTLS is designed to run on top of UDP. At the time of writing, the current version of TLS is 1.2; it is defined in [RFC5246]. DTLS provides nearly identical functionality; it is defined in {RFC6347}} and also at version 1.2. 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. [NOTE: The Logjam authors (weakdh.org) give (inconclusive) evidence that one of the recommendations of [RFC7525], namely use to DHE-1024 as a fallback, may not be sufficient in all cases to counter an attacker with the resources of a nation-state. It is unclear at this time if the RFC is going to be updated as a result or whether there will be an RFC7525bis.] Fairhurst, et al. Expires December 11, 2015 [Page 22] Internet-Draft TAPS Transports June 2015 3.9.1. 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 Data authenticity o Optionally authentication of the peer entity [Note: Both TLS and DTLS provide replay protection, although it is optional in DTLS. The TLS RFC discusses this only in the security considerations and thus views it as a feature that is implicit in the ones listed above. DTLS mentions it as an explicit feature.] The authentication of the peer entity can be omitted, although this is a rare use case. In many use cases (e.g. the Web), authentication is not mutual, however (e.g. only the Web server is authenticated, but not the client). It is important to note that TLS itself does not specify how a peering entity is to be authenticated. This is part of the application logic; i.e. the authentication decision rests with the application. As an 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 the purposes of the application or whether the handshake should be aborted. As DTLS is used over the unreliable UDP transport, it needs to add three features to provide the same security guarantees as TLS: * Message fragmentation * Message reordering * Message loss As a result, DTLS provides features that UDP lacks. [EDITOR'S NOTE: Need to describe how this is achieved?] 3.9.2. Interface Description TLS is commonly used with a socket-like interface, although details can vary between implementations. This is particularly true for the choice which cryptographic algorithms to use, see below. [TODO: DTLS interface] Fairhurst, et al. Expires December 11, 2015 [Page 23] Internet-Draft TAPS Transports June 2015 Both TLS and DTLS allow to employ a multitude of cipher suites for encryption, hashing and applying message integrity. It is no easy task to choose safe settings here. [RFC7525] provides guidance. [TODO: list the RFCs?] [TODO: more detail?] ### Transport Protocol Components Both TLS and DTLS employ a layered architecture. The lower layer is commonly called the record protocol. It is responsible for fragmenting messages, applying message authentication codes (MACs), encrypting data, and sending it via the underlying transport protocol. Several essential protocols run on top of the record protocol in order to carry out the handshake and establish a secure session. [EDITOR'S NOTE: TLS can also compress, but this has been found to be a security weakness. It is not described here.] 3.10. Hypertext Transport Protocol (HTTP) over TCP as a pseudotransport Hypertext Transfer Protocol (HTTP) is an application-level protocol widely used on the Internet. Version 1.1 of the protocol is specified in [RFC7230] [RFC7231] [RFC7232] [RFC7233] [RFC7234] [RFC7235], and version 2 in [RFC7540]. Furthermore, HTTP is used as a substrate for other application-layer protocols. There are various reasons for this practice listed in [RFC3205]; these include being a well-known and well-understood protocol, reusability of existing servers and client libraries, easy use of existing security mechanisms such as HTTP digest authentication [RFC2617] and TLS [RFC5246], the ability of HTTP to traverse firewalls which makes it work with a lot of infrastructure, and cases where a application server often needs to support HTTP anyway. Depending on application's needs, the use of HTTP as a substrate protocol may add complexity and overhead in comparison to a special- purpose protocol (e.g. HTTP headers, suitability of the HTTP security model etc.). [RFC3205] address this issues and provides some guidelines and concerns about the use of HTTP standard port 80 and 443, the use of HTTP URL scheme and interaction with existing firewalls, proxies and NATs. Though not strictly bound to TCP, HTTP is almost exclusively run over TCP, and therefore inherits its properties when used in this way. Fairhurst, et al. Expires December 11, 2015 [Page 24] Internet-Draft TAPS Transports June 2015 3.10.1. Protocol Description Hypertext Transfer Protocol (HTTP) is a request/response protocol. A client sends a request containing a request method, URI and protocol version followed by a MIME-like message (see [RFC7231] for the differences between an HTTP object and a MIME message), containing information about the client and request modifiers. The message can contain a message body carrying application data as well. The server responds with a status or error code followed by a MIME-like message containing information about the server and information about carried data and it can include a message body. It is possible to specify a data format for the message body using MIME media types [RFC2045]. Furthermore, the protocol has numerous additional features; features relevant to pseudotransport are described below. Content negotiation, specified in [RFC7231], is a mechanism provided by HTTP for selecting a representation on a requested resource. The client and server negotiate acceptable data formats, charsets, data encoding (e.g. data can be transferred compressed, gzip), etc. HTTP can accommodate exchange of messages as well as data streaming (using chunked transfer encoding [RFC7230]). It is also possible to request a part of a resource using range requests specified in [RFC7233]. The protocol provides powerful cache control signalling defined in [RFC7234]. HTTP 1.1's and HTTP 2.0's persistent connections can be use to perform multiple request-response transactions during the life-time of a single HTTP connection. Moreover, HTTP 2.0 connections can multiplex many request/response pairs in parallel on a single connection. This reduces connection establishment overhead and the effect of TCP slow-start on each transaction, important for HTTP's primary use case. It is possible to combine HTTP with security mechanisms, like TLS (denoted by HTTPS), which adds protocol properties provided by such a mechanism (e.g. authentication, encryption, etc.). TLS's Application-Layer Protocol Negotiation (ALPN) extension [RFC7301] can be used for HTTP version negotiation within TLS handshake which eliminates addition round-trip. Arbitrary cookie strings, included as part of the MIME headers, are often used as bearer tokens in HTTP. Application layer protocols using HTTP as substrate may use existing method and data formats, or specify new methods and data formats. Furthermore some protocols may not fit a request/response paradigm and instead rely on HTTP to send messages (e.g. [RFC6546]). Because HTTP is working in many restricted infrastructures, it is also used to tunnel other application-layer protocols. Fairhurst, et al. Expires December 11, 2015 [Page 25] Internet-Draft TAPS Transports June 2015 3.10.2. Interface Description There are many HTTP libraries available exposing different APIs. The APIs provide a way to specify a request by providing a URI, a method, request modifiers and optionally a request body. For the response, callbacks can be registered that will be invoked when the response is received. If TLS is used, API expose a registration of callbacks in case a server requests client authentication and when certificate verification is needed. World Wide Web Consortium (W3C) standardized the XMLHttpRequest API [XHR], an API that can be use for sending HTTP/HTTPS requests and receiving server responses. Besides XML data format, request and response data format can also be JSON, HTML and plain text. Specifically JavaScript and XMLHttpRequest are a ubiquitous programming model for websites, and more general applications, where native code is less attractive. Representational State Transfer (REST) [REST] is another example how applications can use HTTP as transport protocol. REST is an architecture style for building application on the Internet. It uses HTTP as a communication protocol. 3.10.3. Transport Protocol Components The transport protocol components provided by HTTP, when used as a pseudotransport, are: o unicast o reliable delivery o ordered delivery o message and stream-oriented o object range request o message content type negotiation o congestion control HTTPS (HTTP over TLS) additionally provides the following components: o authentication (of one or both ends of a connection) o confidentiality Fairhurst, et al. Expires December 11, 2015 [Page 26] Internet-Draft TAPS Transports June 2015 o integrity protection 3.11. WebSockets [RFC6455] [EDITOR'S NOTE: Salvatore Loreto will contribute text for this section.] 3.11.1. Protocol Description 3.11.2. Interface Description 3.11.3. Transport Protocol Components 4. Transport Service Features The transport protocol components analyzed in this document which can be used as a basis for defining common transport service features, normalized and separated into categories, are as follows: o Destination selection * unicast * broadcast (IPv4 only) * multicast * anycast * transport layer multihoming for resilience * transport layer mobility * port multiplexing * service codes o Connection setup * connection setup with feature negotiation and application-to- port mapping o Delivery * reliable delivery Fairhurst, et al. Expires December 11, 2015 [Page 27] Internet-Draft TAPS Transports June 2015 * partially reliable delivery * unreliable delivery * packet erasure coding * ordered delivery * unordered delivery * stream-oriented delivery * message-oriented delivery * message fragmentation * object-oriented delivery of discrete data or file items * unordered delivery of in-memory data or file bulk content objects * object range request * object content type negotiation * single streaming * multiple streaming * stream scheduling prioritization * segmentation * data bundling (Nagle's algorithm) * message bundling o Transmission control * timer-based rate control * ack-based flow control * drop notification * packet erasure coding * congestion control Fairhurst, et al. Expires December 11, 2015 [Page 28] Internet-Draft TAPS Transports June 2015 o Integrity protection * checksum for error detection * partial checksum protection * checksum optional * cryptographic integrity protection o Security * authentication of one end of a connection * authentication of both ends of a connection * confidentiality The next revision of this document will define transport service features based upon this list. [EDITOR'S NOTE: this section will drawn from the candidate features provided by protocol components in the previous section - please discuss on taps@ietf.org list] 4.1. Complete Protocol Feature Matrix [EDITOR'S NOTE: Dave Thaler has signed up as a contributor for this section. Michael Welzl also has a beginning of a matrix which could be useful here.] [EDITOR'S NOTE: The below is a strawman proposal below by Gorry Fairhurst for initial discussion] The table below summarises protocol mechanisms that have been standardised. It does not make an assessment on whether specific implementations are fully compliant to these specifications. Fairhurst, et al. Expires December 11, 2015 [Page 29] Internet-Draft TAPS Transports June 2015 +-----------------+---------+---------+---------+---------+---------+ | Mechanism | UDP | UDP-L | DCCP | SCTP | TCP | +-----------------+---------+---------+---------+---------+---------+ | Unicast | Yes | Yes | Yes | Yes | Yes | | | | | | | | | Mcast/IPv4Bcast | Yes(2) | Yes | No | No | No | | | | | | | | | Port Mux | Yes | Yes | Yes | Yes | Yes | | | | | | | | | Mode | Dgram | Dgram | Dgram | Dgram | Stream | | | | | | | | | Connected | No | No | Yes | Yes | Yes | | | | | | | | | Data bundling | No | No | No | Yes | Yes | | | | | | | | | Feature Nego | No | No | Yes | Yes | Yes | | | | | | | | | Options | No | No | Support | Support | Support | | | | | | | | | Data priority | * | * | * | Yes | No | | | | | | | | | Data bundling | No | No | No | Yes | Yes | | | | | | | | | Reliability | None | None | None | Select | Full | | | | | | | | | Ordered deliv | No | No | No | Stream | Yes | | | | | | | | | Corruption Tol. | No | Support | Support | No | No | | | | | | | | | Flow Control | No | No | Support | Yes | Yes | | | | | | | | | PMTU/PLPMTU | (1) | (1) | Yes | Yes | Yes | | | | | | | | | Cong Control | (1) | (1) | Yes | Yes | Yes | | | | | | | | | ECN Support | (1) | (1) | Yes | TBD | Yes | | | | | | | | | NAT support | Limited | Limited | Support | TBD | Support | | | | | | | | | Security | DTLS | DTLS | DTLS | DTLS | TLS, AO | | | | | | | | | UDP encaps | N/A | None | Yes | Yes | None | | | | | | | | | RTP support | Support | Support | Support | ? | Support | +-----------------+---------+---------+---------+---------+---------+ Note (1): this feature requires support in an upper layer protocol. Fairhurst, et al. Expires December 11, 2015 [Page 30] Internet-Draft TAPS Transports June 2015 Note (2): this feature requires support in an upper layer protocol when used with IPv6. 5. IANA Considerations This document has no considerations for IANA. 6. 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 components or mechanisms for providing these features. Each RFC listed in this document discusses the security considerations of the specification it contains. 7. Contributors [Editor's Note: turn this into a real contributors section with addresses once we figure out how to trick the toolchain into doing so] o Section 3.2 on MPTCP was contributed by Simone Ferlin-Oliviera (ferlin@simula.no) and Olivier Mehani (olivier.mehani@nicta.com.au) o Section 3.4 on UDP was contributed by Kevin Fall (kfall@kfall.com) o Section 3.3 on SCTP was contributed by Michael Tuexen (tuexen@fh- muenster.de) o Section 3.8 on NORM was contributed by Brian Adamson (brian.adamson@nrl.navy.mil) o Section 3.9 on MPTCP was contributed by Ralph Holz (ralph.holz@nicta.com.au) and Olivier Mehani (olivier.mehani@nicta.com.au) o Section 3.10 on HTTP was contributed by Dragana Damjanovic (ddamjanovic@mozilla.com) 8. Acknowledgments Thanks to Karen Nielsen, Joe Touch, and Michael Welzl for the comments, feedback, and discussion. This work is partially supported by the European Commission under grant agreement FP7-ICT-318627 mPlane; support does not imply endorsement. Fairhurst, et al. Expires December 11, 2015 [Page 31] Internet-Draft TAPS Transports June 2015 [EDITOR'S NOTE: add H2020-NEAT ack]. 9. References 9.1. Normative References [RFC0791] Postel, J., "Internet Protocol", STD 5, RFC 791, September 1981. 9.2. Informative References [RFC0768] Postel, J., "User Datagram Protocol", STD 6, RFC 768, August 1980. [RFC0793] Postel, J., "Transmission Control Protocol", STD 7, RFC 793, September 1981. [RFC0896] Nagle, J., "Congestion control in IP/TCP internetworks", RFC 896, January 1984. [RFC1122] Braden, R., "Requirements for Internet Hosts - Communication Layers", STD 3, RFC 1122, October 1989. [RFC1191] Mogul, J. and S. Deering, "Path MTU discovery", RFC 1191, November 1990. [RFC1981] McCann, J., Deering, S., and J. Mogul, "Path MTU Discovery for IP version 6", RFC 1981, August 1996. [RFC2018] Mathis, M., Mahdavi, J., Floyd, S., and A. Romanow, "TCP Selective Acknowledgment Options", RFC 2018, October 1996. [RFC2045] Freed, N. and N. Borenstein, "Multipurpose Internet Mail Extensions (MIME) Part One: Format of Internet Message Bodies", RFC 2045, November 1996. [RFC2460] Deering, S. and R. Hinden, "Internet Protocol, Version 6 (IPv6) Specification", RFC 2460, December 1998. [RFC2617] Franks, J., Hallam-Baker, P., Hostetler, J., Lawrence, S., Leach, P., Luotonen, A., and L. Stewart, "HTTP Authentication: Basic and Digest Access Authentication", RFC 2617, June 1999. [RFC3168] Ramakrishnan, K., Floyd, S., and D. Black, "The Addition of Explicit Congestion Notification (ECN) to IP", RFC 3168, September 2001. Fairhurst, et al. Expires December 11, 2015 [Page 32] Internet-Draft TAPS Transports June 2015 [RFC3205] Moore, K., "On the use of HTTP as a Substrate", BCP 56, RFC 3205, February 2002. [RFC3390] Allman, M., Floyd, S., and C. Partridge, "Increasing TCP's Initial Window", RFC 3390, October 2002. [RFC3436] Jungmaier, A., Rescorla, E., and M. Tuexen, "Transport Layer Security over Stream Control Transmission Protocol", RFC 3436, December 2002. [RFC3452] Luby, M., Vicisano, L., Gemmell, J., Rizzo, L., Handley, M., and J. Crowcroft, "Forward Error Correction (FEC) Building Block", RFC 3452, December 2002. [RFC3758] Stewart, R., Ramalho, M., Xie, Q., Tuexen, M., and P. Conrad, "Stream Control Transmission Protocol (SCTP) Partial Reliability Extension", RFC 3758, May 2004. [RFC3828] Larzon, L-A., Degermark, M., Pink, S., Jonsson, L-E., and G. Fairhurst, "The Lightweight User Datagram Protocol (UDP-Lite)", RFC 3828, July 2004. [RFC4324] Royer, D., Babics, G., and S. Mansour, "Calendar Access Protocol (CAP)", RFC 4324, December 2005. [RFC4336] Floyd, S., Handley, M., and E. Kohler, "Problem Statement for the Datagram Congestion Control Protocol (DCCP)", RFC 4336, March 2006. [RFC4340] Kohler, E., Handley, M., and S. Floyd, "Datagram Congestion Control Protocol (DCCP)", RFC 4340, March 2006. [RFC4341] Floyd, S. and E. Kohler, "Profile for Datagram Congestion Control Protocol (DCCP) Congestion Control ID 2: TCP-like Congestion Control", RFC 4341, March 2006. [RFC4342] Floyd, S., Kohler, E., and J. Padhye, "Profile for Datagram Congestion Control Protocol (DCCP) Congestion Control ID 3: TCP-Friendly Rate Control (TFRC)", RFC 4342, March 2006. [RFC4614] Duke, M., Braden, R., Eddy, W., and E. Blanton, "A Roadmap for Transmission Control Protocol (TCP) Specification Documents", RFC 4614, September 2006. [RFC4654] Widmer, J. and M. Handley, "TCP-Friendly Multicast Congestion Control (TFMCC): Protocol Specification", RFC 4654, August 2006. Fairhurst, et al. Expires December 11, 2015 [Page 33] Internet-Draft TAPS Transports June 2015 [RFC4820] Tuexen, M., Stewart, R., and P. Lei, "Padding Chunk and Parameter for the Stream Control Transmission Protocol (SCTP)", RFC 4820, March 2007. [RFC4821] Mathis, M. and J. Heffner, "Packetization Layer Path MTU Discovery", RFC 4821, March 2007. [RFC4895] Tuexen, M., Stewart, R., Lei, P., and E. Rescorla, "Authenticated Chunks for the Stream Control Transmission Protocol (SCTP)", RFC 4895, August 2007. [RFC4960] Stewart, R., "Stream Control Transmission Protocol", RFC 4960, September 2007. [RFC5061] Stewart, R., Xie, Q., Tuexen, M., Maruyama, S., and M. Kozuka, "Stream Control Transmission Protocol (SCTP) Dynamic Address Reconfiguration", RFC 5061, September 2007. [RFC5097] Renker, G. and G. Fairhurst, "MIB for the UDP-Lite protocol", RFC 5097, January 2008. [RFC5246] Dierks, T. and E. Rescorla, "The Transport Layer Security (TLS) Protocol Version 1.2", RFC 5246, August 2008. [RFC5348] Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP Friendly Rate Control (TFRC): Protocol Specification", RFC 5348, September 2008. [RFC5405] Eggert, L. and G. Fairhurst, "Unicast UDP Usage Guidelines for Application Designers", BCP 145, RFC 5405, November 2008. [RFC5595] Fairhurst, G., "The Datagram Congestion Control Protocol (DCCP) Service Codes", RFC 5595, September 2009. [RFC5596] Fairhurst, G., "Datagram Congestion Control Protocol (DCCP) Simultaneous-Open Technique to Facilitate NAT/ Middlebox Traversal", RFC 5596, September 2009. [RFC5662] Shepler, S., Eisler, M., and D. Noveck, "Network File System (NFS) Version 4 Minor Version 1 External Data Representation Standard (XDR) Description", RFC 5662, January 2010. [RFC5672] Crocker, D., "RFC 4871 DomainKeys Identified Mail (DKIM) Signatures -- Update", RFC 5672, August 2009. Fairhurst, et al. Expires December 11, 2015 [Page 34] Internet-Draft TAPS Transports June 2015 [RFC5740] Adamson, B., Bormann, C., Handley, M., and J. Macker, "NACK-Oriented Reliable Multicast (NORM) Transport Protocol", RFC 5740, November 2009. [RFC6773] Phelan, T., Fairhurst, G., and C. Perkins, "DCCP-UDP: A Datagram Congestion Control Protocol UDP Encapsulation for NAT Traversal", RFC 6773, November 2012. [RFC5925] Touch, J., Mankin, A., and R. Bonica, "The TCP Authentication Option", RFC 5925, June 2010. [RFC5681] Allman, M., Paxson, V., and E. Blanton, "TCP Congestion Control", RFC 5681, September 2009. [RFC6083] Tuexen, M., Seggelmann, R., and E. Rescorla, "Datagram Transport Layer Security (DTLS) for Stream Control Transmission Protocol (SCTP)", RFC 6083, January 2011. [RFC6093] Gont, F. and A. Yourtchenko, "On the Implementation of the TCP Urgent Mechanism", RFC 6093, January 2011. [RFC6525] Stewart, R., Tuexen, M., and P. Lei, "Stream Control Transmission Protocol (SCTP) Stream Reconfiguration", RFC 6525, February 2012. [RFC6546] Trammell, B., "Transport of Real-time Inter-network Defense (RID) Messages over HTTP/TLS", RFC 6546, April 2012. [RFC6298] Paxson, V., Allman, M., Chu, J., and M. Sargent, "Computing TCP's Retransmission Timer", RFC 6298, June 2011. [RFC6347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer Security Version 1.2", RFC 6347, January 2012. [RFC6356] Raiciu, C., Handley, M., and D. Wischik, "Coupled Congestion Control for Multipath Transport Protocols", RFC 6356, October 2011. [RFC6455] Fette, I. and A. Melnikov, "The WebSocket Protocol", RFC 6455, December 2011. [RFC6458] Stewart, R., Tuexen, M., Poon, K., Lei, P., and V. Yasevich, "Sockets API Extensions for the Stream Control Transmission Protocol (SCTP)", RFC 6458, December 2011. Fairhurst, et al. Expires December 11, 2015 [Page 35] Internet-Draft TAPS Transports June 2015 [RFC6691] Borman, D., "TCP Options and Maximum Segment Size (MSS)", RFC 6691, July 2012. [RFC6824] Ford, A., Raiciu, C., Handley, M., and O. Bonaventure, "TCP Extensions for Multipath Operation with Multiple Addresses", RFC 6824, January 2013. [RFC6897] Scharf, M. and A. Ford, "Multipath TCP (MPTCP) Application Interface Considerations", RFC 6897, March 2013. [RFC6935] Eubanks, M., Chimento, P., and M. Westerlund, "IPv6 and UDP Checksums for Tunneled Packets", RFC 6935, April 2013. [RFC6936] Fairhurst, G. and M. Westerlund, "Applicability Statement for the Use of IPv6 UDP Datagrams with Zero Checksums", RFC 6936, April 2013. [RFC6951] Tuexen, M. and R. Stewart, "UDP Encapsulation of Stream Control Transmission Protocol (SCTP) Packets for End-Host to End-Host Communication", RFC 6951, May 2013. [RFC7053] Tuexen, M., Ruengeler, I., and R. Stewart, "SACK- IMMEDIATELY Extension for the Stream Control Transmission Protocol", RFC 7053, November 2013. [RFC7230] Fielding, R. and J. Reschke, "Hypertext Transfer Protocol (HTTP/1.1): Message Syntax and Routing", RFC 7230, June 2014. [RFC7231] Fielding, R. and J. Reschke, "Hypertext Transfer Protocol (HTTP/1.1): Semantics and Content", RFC 7231, June 2014. [RFC7232] Fielding, R. and J. Reschke, "Hypertext Transfer Protocol (HTTP/1.1): Conditional Requests", RFC 7232, June 2014. [RFC7233] Fielding, R., Lafon, Y., and J. Reschke, "Hypertext Transfer Protocol (HTTP/1.1): Range Requests", RFC 7233, June 2014. [RFC7234] Fielding, R., Nottingham, M., and J. Reschke, "Hypertext Transfer Protocol (HTTP/1.1): Caching", RFC 7234, June 2014. [RFC7235] Fielding, R. and J. Reschke, "Hypertext Transfer Protocol (HTTP/1.1): Authentication", RFC 7235, June 2014. Fairhurst, et al. Expires December 11, 2015 [Page 36] Internet-Draft TAPS Transports June 2015 [RFC7301] Friedl, S., Popov, A., Langley, A., and E. Stephan, "Transport Layer Security (TLS) Application-Layer Protocol Negotiation Extension", RFC 7301, July 2014. [RFC7323] Borman, D., Braden, B., Jacobson, V., and R. Scheffenegger, "TCP Extensions for High Performance", RFC 7323, September 2014. [RFC7457] Sheffer, Y., Holz, R., and P. Saint-Andre, "Summarizing Known Attacks on Transport Layer Security (TLS) and Datagram TLS (DTLS)", RFC 7457, February 2015. [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, May 2015. [RFC7540] Belshe, M., Peon, R., and M. Thomson, "Hypertext Transfer Protocol Version 2 (HTTP/2)", RFC 7540, May 2015. [I-D.ietf-aqm-ecn-benefits] Welzl, M. and G. Fairhurst, "The Benefits and Pitfalls of using Explicit Congestion Notification (ECN)", draft-ietf- aqm-ecn-benefits-00 (work in progress), October 2014. [I-D.ietf-tsvwg-sctp-dtls-encaps] Tuexen, M., Stewart, R., Jesup, R., and S. Loreto, "DTLS Encapsulation of SCTP Packets", draft-ietf-tsvwg-sctp- dtls-encaps-09 (work in progress), January 2015. [I-D.ietf-tsvwg-sctp-prpolicies] Tuexen, M., Seggelmann, R., Stewart, R., and S. Loreto, "Additional Policies for the Partial Reliability Extension of the Stream Control Transmission Protocol", draft-ietf- tsvwg-sctp-prpolicies-07 (work in progress), February 2015. [I-D.ietf-tsvwg-sctp-ndata] Stewart, R., Tuexen, M., Loreto, S., and R. Seggelmann, "Stream Schedulers and User Message Interleaving for the Stream Control Transmission Protocol", draft-ietf-tsvwg- sctp-ndata-03 (work in progress), March 2015. [I-D.ietf-tsvwg-natsupp] Stewart, R., Tuexen, M., and I. Ruengeler, "Stream Control Transmission Protocol (SCTP) Network Address Translation Support", draft-ietf-tsvwg-natsupp-07 (work in progress), February 2015. Fairhurst, et al. Expires December 11, 2015 [Page 37] Internet-Draft TAPS Transports June 2015 [XHR] van Kesteren, A., Aubourg, J., Song, J., and H. Steen, "XMLHttpRequest working draft (http://www.w3.org/TR/XMLHttpRequest/)", 2000. [REST] Fielding, R., "Architectural Styles and the Design of Network-based Software Architectures, Ph. D. (UC Irvune), Chapter 5: Representational State Transfer", 2000. Authors' Addresses Godred Fairhurst (editor) University of Aberdeen School of Engineering, Fraser Noble Building Aberdeen AB24 3UE Email: gorry@erg.abdn.ac.uk Brian Trammell (editor) ETH Zurich Gloriastrasse 35 8092 Zurich Switzerland Email: ietf@trammell.ch Mirja Kuehlewind (editor) ETH Zurich Gloriastrasse 35 8092 Zurich Switzerland Email: mirja.kuehlewind@tik.ee.ethz.ch Fairhurst, et al. Expires December 11, 2015 [Page 38]