Network Working Group                                  G. Fairhurst, Ed.
Internet-Draft                                    University of Aberdeen
Intended status: Informational                          B. Trammell, Ed.
Expires: November 28, December 11, 2015                            M. Kuehlewind, Ed.
                                                              ETH Zurich
                                                            May 27,
                                                           June 09, 2015

  Services provided by IETF transport protocols and congestion control
                               mechanisms
                     draft-ietf-taps-transports-04
                     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 November 28, 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
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   3
   2.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . .   4
   3.  Existing Transport Protocols  . . . . . . . . . . . . . . . .   4
     3.1.  Transport Control Protocol (TCP)  . . . . . . . . . . . .   4
       3.1.1.  Protocol Description  . . . . . . . . . . . . . . . .   5
       3.1.2.  Interface description . . . . . . . . . . . . . . . .   6
       3.1.3.  Transport Protocol Components . . . . . . . . . . . .   6
     3.2.  Multipath TCP (MP-TCP) (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) . . . . . . .   7   9
       3.3.1.  Protocol Description  . . . . . . . . . . . . . . . .   8   9
       3.3.2.  Interface Description . . . . . . . . . . . . . . . .  10  11
       3.3.3.  Transport Protocol Components . . . . . . . . . . . .  11  13
     3.4.  User Datagram Protocol (UDP)  . . . . . . . . . . . . . .  12  13
       3.4.1.  Protocol Description  . . . . . . . . . . . . . . . .  12  14
       3.4.2.  Interface Description . . . . . . . . . . . . . . . .  13  14
       3.4.3.  Transport Protocol Components . . . . . . . . . . . .  13  15
     3.5.  Lightweight User Datagram Protocol (UDP-Lite) . . . . . .  14  15
       3.5.1.  Protocol Description  . . . . . . . . . . . . . . . .  14  15
       3.5.2.  Interface Description . . . . . . . . . . . . . . . .  15  16
       3.5.3.  Transport Protocol Components . . . . . . . . . . . .  15  16
     3.6.  Datagram Congestion Control Protocol (DCCP) . . . . . . .  15  17
       3.6.1.  Protocol Description  . . . . . . . . . . . . . . . .  16  17
       3.6.2.  Interface Description . . . . . . . . . . . . . . . .  17  19
       3.6.3.  Transport Protocol Components . . . . . . . . . . . .  17  19
     3.7.  Realtime Transport Protocol (RTP) . . . . . . . . . . . .  18  19
     3.8.  NACK-Oriented Reliable Multicast (NORM) . . . . . . . . .  18  20
       3.8.1.  Protocol Description  . . . . . . . . . . . . . . . .  18  20
       3.8.2.  Interface Description . . . . . . . . . . . . . . . .  19  21
       3.8.3.  Transport Protocol Components . . . . . . . . . . . .  20  21
     3.9.  Transport Layer Security (TLS) and Datagram TLS (DTLS) as
           a pseudotransport . . . . . . . . . . . . . . . . . . . .  20  22
       3.9.1.  Protocol Description  . . . . . . . . . . . . . . . .  21  23
       3.9.2.  Interface Description . . . . . . . . . . . . . . . .  21
       3.9.3.  Transport Protocol Components . . . . . . . . . . . .  21  23
     3.10. Hypertext Transport Protocol (HTTP) over TCP as a
           pseudotransport . . . . . . . . . . . . . . . . . . . . .  21  24
       3.10.1.  Protocol Description . . . . . . . . . . . . . . . .  21  25
       3.10.2.  Interface Description  . . . . . . . . . . . . . . .  22  26
       3.10.3.  Transport Protocol Components  . . . . . . . . . . .  23  26
     3.11. WebSockets  . . . . . . . . . . . . . . . . . . . . . . .  23  27
       3.11.1.  Protocol Description . . . . . . . . . . . . . . . .  23  27
       3.11.2.  Interface Description  . . . . . . . . . . . . . . .  24  27
       3.11.3.  Transport Protocol Components  . . . . . . . . . . .  24  27
   4.  Transport Service Features  . . . . . . . . . . . . . . . . .  24  27
     4.1.  Complete Protocol Feature Matrix  . . . . . . . . . . . .  26  29
   5.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  28  31
   6.  Security Considerations . . . . . . . . . . . . . . . . . . .  28  31
   7.  Contributors  . . . . . . . . . . . . . . . . . . . . . . . .  28  31
   8.  Acknowledgments . . . . . . . . . . . . . . . . . . . . . . .  28  31
   9.  References  . . . . . . . . . . . . . . . . . . . . . . . . .  28  32
     9.1.  Normative References  . . . . . . . . . . . . . . . . . .  28  32
     9.2.  Informative References  . . . . . . . . . . . . . . . . .  29  32
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  34  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.

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,
   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.

   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
   o  ordered delivery for each byte stream

   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 (MP-TCP)

   [EDITOR'S NOTE: a few sentences describing (MPTCP)

   Multipath TCP [RFC6824] go
   here.  Note that this adds transport-layer multihoming to the
   components TCP provides.  Simone Ferlin-Oliveira will contribute text
   for this section.]

3.3.  Stream Control Transmission Protocol (SCTP)

   SCTP is a message oriented standards track transport protocol and the
   base protocol is specified in [RFC4960]. an extension for TCP to support multi-
   homing.  It supports multi-homing is designed 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 be as transparent as possible to
   minimize head middle-
   boxes.  It does so by establishing regular TCP flows between a pair
   of line blocking.  SCTP is extensible source/destination endpoints, and multiplexing 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 application's
   stream over these flows.

3.2.1.  Protocol Description

   MPTCP uses TCP options for transporting telephony signalling
   messages and is deployed in telephony signalling networks, especially
   in mobile telephony networks.  Additionally, it is its control plane.  They are used in the WebRTC
   framework for to
   signal multipath capabilities, as well as to negotiate data channels sequence
   numbers, and is therefore deployed in all WEB-
   browsers supporting WebRTC.

   [EDITOR'S NOTE: Michael Tuexen advertise other available IP addresses and Karen Nielsen signed up as
   contributors for these sections.]

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 new
   sessions between pairs of endpoints.

3.2.2.  Interface Description

   By default, MPTCP exposes the same port number concept interface as
   DCCP, TCP, UDP, and UDP-Lite do and only supports unicast.

   SCTP uses TCP to the 32-bit CRC32c
   application.  [RFC6897] however describes a richer API for protecting SCTP packets against bit
   errors. MPTCP-
   aware applications.

   This is stronger than the 16-bit checksums used by TCP Basic API describes how an application can - enable or
   UDP.  However, disable
   MPTCP; - bind a partial checksum coverage as provided by DCCP socket to one or
   UDP-Lite is not supported.

   SCTP has been designed with extensibility in mind.  Each 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 packet
   starts
   [RFC6458] (see next section) to deal with a single common header containing multihoming.

   [AUTHOR'S NOTE: research work, and some implementation, also suggest
   that 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 scheduling algorithm, as well as the SCTP handshake.

   SCTP provides a message-oriented service.  Multiple small user
   messages can path manager, are
   configurable options that should be bundled into a single SCTP packet exposed to improve the
   efficiency.  For example, higher layer.  Should
   this bundling may be done by delaying user
   messages at the sender side similar 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 Nagle algorithm used 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
   TCP.  User messages which would result in IP packets larger multiplexing one byte stream over
   separate paths, it can achieve a higher throughput than the
   MTU will be fragmented TCP in
   certain situations (note however that coupled congestion control
   [RFC6356] might limit this benefit to maintain fairness to other
   flows at the sender side bottleneck).  When aggregating capacity over multiple
   paths, and reassembled at the
   receiver side.  There is no protocol limit depending on the user message size.
   ICMP-based path MTU discovery as specified for IPv4 in [RFC1191] and
   for IPv6 in [RFC1981] as well as packetization layer path MTU
   discovery as specified in [RFC4821] with probe way packets using the
   padding chunks defined the [RFC4820] are supported.

   [RFC4960] specifies a scheduled on each TCP friendly congestion control
   subflow, an additional delay and higher jitter might be observed
   observed before in-order delivery of data 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 applications.

   The transport protocol components provided by MPTCP therefore are:

   o  unicast

   o  connection setup with feature negotiation and 65536 uni-directional streams application-to-port
      mapping

   o  port multiplexing

   o  reliable delivery

   o  error detection (checksum)

   o  segmentation

   o  stream-oriented delivery in each direction.  The number a single stream

   o  flow control

   o  congestion control

   o  endpoint multiplexing of streams can be different in each
   direction.  Every user-message is sent on a particular stream.  User
   messages 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 sent un-ordered started
   over IPv4 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 IPv6 for the same
   stream are delivered at session]

3.3.  Stream Control Transmission Protocol (SCTP)

   SCTP is a message oriented standards track transport protocol and the receiver
   base protocol is specified in the same order as sent by the
   sender.  For [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 not requiring fragmentation, this
   minimises only within each stream.  This allows to
   minimize head of line blocking.  The base protocol  SCTP is extensible and the currently
   defined in
   [RFC4960] doesn't allow interleaving extensions include mechanisms for dynamic re-configurations
   of user-messages, which results streams [RFC6525] and IP-addresses [RFC5061].  Furthermore, the
   extension specified in sending a large message on one stream can block [RFC3758] introduces the sending concept of partial
   reliability for user messages.

   SCTP was originally developed for transporting telephony signalling
   messages on other streams.  [I-D.ietf-tsvwg-sctp-ndata]
   overcomes this limitation.  Furthermore, [I-D.ietf-tsvwg-sctp-ndata]
   specifies multiple algorithms for and is deployed in telephony signalling networks, especially
   in mobile telephony networks.  Additionally, it is used in the sender side selection of which
   streams to send WebRTC
   framework for data from channels and is therefore deployed in all WEB-
   browsers supporting WebRTC.

3.3.1.  Protocol Description

   SCTP is a variety of scheduling
   algorithms including priority based ones.  The stream re-
   configuration extension defined in [RFC6525] allows connection oriented protocol using a four way handshake to reset streams
   during the lifetime of
   establish an SCTP association and a three way message exchange to increase the number of
   streams, if
   gracefully shut it down.  It uses the same port number of streams negotiated in the concept as
   DCCP, TCP, UDP, and UDP-Lite do and only supports unicast.

   SCTP handshake is
   not sufficient.

   According to [RFC4960], each user message sent is either delivered to
   the receiver or, in case of excessive retransmissions, uses the
   association 32-bit CRC32c for protecting SCTP packets against bit
   errors.  This is terminated in a non-graceful way, similar to stronger than the 16-bit checksums used by 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 or
   UDP.  However, a user message partial checksum coverage as provided by DCCP or
   UDP-Lite is dealt not supported.

   SCTP has been designed with by extensibility in mind.  Each SCTP packet
   starts with a single common header containing the sender.

   o  Limiting port numbers, a
   verification tag and the number of retransmissions for each fragment of a user
      message.  If the number of retransmissions CRC32c checksum.  This common header is limited to 0, one
      gets
   followed by a service similar to UDP.

   o  Abandoning messages sequence of lower priority in case chunks.  Each chunk consists of a send buffer
      shortage.

   SCTP supports multi-homing.  Each SCTP end-point uses type
   field, flags, a list of IP-
   addresses length field and a single port number.  These addresses can be any
   mixture of IPv4 and IPv6 addresses.  These addresses are negotiated
   during the handshake and the address re-configuration extension
   specified in [RFC5061] in combination value.  [RFC4960] defines how a
   receiver processes chunks with [RFC4895] an unknown chunk type.  The support of
   extensions can be used to
   change these addresses in an authenticated way negotiated during the livetime of
   an SCTP association.  This allows for transport layer mobility.
   Multiple addresses are used for improved resilience.  If handshake.

   SCTP provides a remote
   address becomes unreachable, the traffic is switched over to message-oriented service.  Multiple small user
   messages can be bundled into a
   reachable one, if one exists.  Each single SCTP end-point supervises
   continuously packet to improve the reachability of all peer addresses using a heartbeat
   mechanism.
   efficiency.  For securing user messages, the use of TLS over SCTP has been
   specified in [RFC3436].  However, example, this solution does not support all
   services provided bundling may be done by SCTP (for example un-ordered delivery or partial
   reliability), and therefore delaying user
   messages at the use of DTLS over SCTP has been
   specified in [RFC6083] sender side similar to overcome these limitations.  When using
   DTLS over SCTP, the application can use almost all services provided Nagle algorithm used by SCTP.

   [I-D.ietf-tsvwg-natsupp] defines a methods for end-hosts
   TCP.  User messages which would result in IP packets larger than the
   MTU will be fragmented at the sender side and
   middleboxes to provide reassembled at the
   receiver side.  There is no protocol limit on the user message size.
   ICMP-based path MTU discovery as specified for NAT support IPv4 in [RFC1191] and
   for SCTP over IPv4.  For
   legacy NAT traversal, [RFC6951] defines the UDP encapsulation of
   SCTP-packets.  Alternatively, SCTP packets can be encapsulated IPv6 in
   DTLS packets [RFC1981] as well as packetization layer path MTU
   discovery as specified in [I-D.ietf-tsvwg-sctp-dtls-encaps].  The
   latter encapsulation is used [RFC4821] with in probe packets using the WebRTC context.

   Having a well
   padding chunks defined API is the [RFC4820] are supported.

   [RFC4960] specifies a TCP friendly congestion control to protect the
   network against overload.  SCTP also uses a feature provided by sliding window flow
   control to protect receivers against overflow.

   Each SCTP as
   described 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 next subsection.

3.3.2.  Interface Description

   [RFC4960] defines an abstract API for upper
   layer.  Un-ordered messages can be delivered as soon as they are
   completely received.  Only all ordered messages sent on the base protocol.  An
   extension to same
   stream are delivered at the BSD Sockets API is defined receiver in [RFC6458] and covers:

   o 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].

   o  the SCTP Partial Reliability extension defined
   [RFC4960] doesn't allow interleaving of user-messages, which results
   in [RFC3758].

   o sending a large message on one stream can block the SCTP Authentication 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 [RFC4895].

   o [RFC6525] allows to reset streams
   during the SCTP Dynamic Address Reconfiguration extension defined lifetime of an association and to increase the number of
   streams, if the number of streams negotiated in
      [RFC5061].

   For the following SCTP protocol extensions the BSD Sockets API
   extension handshake is defined
   not sufficient.

   According to [RFC4960], each user message sent is either delivered to
   the receiver or, in case of excessive retransmissions, the document specifying
   association is terminated in a non-graceful way, similar to the protocol
   extensions:

   o TCP
   behaviour.  In addition to this reliable transfer, the SCTP SACK-IMMEDIATELY partial
   reliability extension defined in [RFC7053]. [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 SCTP Stream Reconfiguration extension defined in [RFC6525]. time a user message is dealt with by the sender.

   o  Limiting the UDP Encapsulation number of SCTP packets extension defined in
      [RFC6951].

   o retransmissions for each fragment of a user
      message.  If the additional PR-SCTP policies defined in
      [I-D.ietf-tsvwg-sctp-prpolicies].

   Future documents describing SCTP protocol extensions are expected number of retransmissions is limited to
   describe the corresponding BSD Sockets API extension in 0, one
      gets 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 service similar to TCP sockets, there is UDP.

   o  Abandoning messages of lower priority in case of a 1:1
   relationship between the sockets and the send buffer
      shortage.

   SCTP associations (except
   for listening sockets).  One-to-many style supports multi-homing.  Each SCTP sockets are similar
   to unconnected UDP sockets as there is end-point uses a 1:n relationship between the
   sockets list of IP-
   addresses and the SCTP associations.

   The SCTP stack a single port number.  These addresses can provide information to the applications about
   state changes be any
   mixture of the individual paths IPv4 and the association whenever
   they occur. IPv6 addresses.  These events are delivered similar to user messages but addresses are specifically marked as notifications.

   A couple of new functions have been introduced to support negotiated
   during the use of
   multiple local and remote addresses.  Additional SCTP-specific send handshake and receive calls have been defined to allow dealing with the SCTP
   specific information without using ancillary data address re-configuration extension
   specified in [RFC5061] in combination with [RFC4895] can be used to
   change these addresses in an authenticated way during the form of
   additional cmsgs, which are also defined.  These functions provide
   support for detecting partial delivery livetime of user messages and
   notifications.

   The
   an SCTP socket API association.  This allows for transport layer mobility.
   Multiple addresses are used for improved resilience.  If a fine-grained control of remote
   address becomes unreachable, the protocol
   behaviour through an extensive set of socket options.

   The SCTP kernel implementations of FreeBSD, Linux and Solaris follow
   mostly 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 extension 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 BSD Sockets API application can use almost all services provided
   by SCTP.

   [I-D.ietf-tsvwg-natsupp] defines a methods for the base
   protocol end-hosts and
   middleboxes to provide for NAT support for SCTP over IPv4.  For
   legacy NAT traversal, [RFC6951] defines the corresponding supported protocol extensions.

3.3.3.  Transport Protocol Components 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 transport protocol components
   latter encapsulation is used with in the WebRTC context.

   Having a well defined API is also a feature 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 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 unordered delivery within a stream covers:

   o  support for multiple concurrent streams  the base protocol defined in [RFC4960].

   o  support for stream scheduling prioritization  the SCTP Partial Reliability extension defined in [RFC3758].

   o  flow control  the SCTP Authentication extension defined in [RFC4895].

   o  message-oriented delivery
   o  congestion control

   o  user message bundling

   o  user message fragmentation and reassembly  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  strong error detection (CRC32C)  the SCTP SACK-IMMEDIATELY extension defined in [RFC7053].

   o  transport layer multihoming for resilience  the SCTP Stream Reconfiguration extension defined in [RFC6525].

   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  the UDP Encapsulation of SCTP packets extension defined in
      [RFC6951].

   o  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), additional PR-SCTP policies defined in
   addition
      [I-D.ietf-tsvwg-sctp-prpolicies].

   Future documents describing SCTP protocol extensions are expected to unicast (and anycast) datagrams.  IETF guidance on
   describe the
   use corresponding BSD Sockets API extension in a "Socket API
   Considerations" section.

   The SCTP socket API supports two kinds of UDP is provided in[RFC5405].  UDP 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 widely implemented a 1:1
   relationship between the sockets and
   widely used by common applications, especially DNS.

3.4.1.  Protocol Description the SCTP associations (except
   for listening sockets).  One-to-many style SCTP sockets are similar
   to unconnected UDP sockets as there 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 1:n relationship between the loss of
   messages from other protocols (e.g., TCP) when sharing
   sockets and the same
   network paths.  UDP traffic SCTP associations.

   The SCTP stack can also cause provide information to the loss applications about
   state changes 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 individual paths and the association whenever
   they occur.  These events are discarded before being delivered similar to an
   application.  There user messages but
   are some exceptions specifically marked as notifications.

   A couple of new functions have been introduced to this general rule,
   however.  UDP-Lite (see [RFC3828], and below) provides support the ability
   for portions use of the message contents to be exempt from checksum
   coverage.  It is also possible
   multiple local and remote addresses.  Additional SCTP-specific send
   and receive calls have been defined to create UDP datagrams allow dealing 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 SCTP
   specific information without using ancillary data 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,
   additional cmsgs, 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, are also defined.  These functions provide
   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 detecting partial delivery of (IP address, port number).
   Demultiplexing using multiple abstract endpoints (sockets) on the
   same IP address are supported. user messages and
   notifications.

   The same SCTP socket may be used by a
   single server to interact with multiple clients (note: this behavior
   differs from TCP, which uses API allows a pair fine-grained control of tuples to identify a
   connection).  Multiple server instances (processes) binding the same protocol
   behaviour through an extensive set of socket can cooperate to service multiple clients- options.

   The SCTP kernel implementations of FreeBSD, Linux and Solaris follow
   mostly the socket
   implementation arranges specified extension 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, BSD Sockets API for UDP, this is only a local
   operation that serves to simplify the local send/receive functions base
   protocol and to filter the traffic for the specified addresses and ports
   [RFC5405].

3.4.3.  Transport Protocol Components corresponding supported protocol extensions.

3.3.3.  Transport Protocol Components

   The transport protocol components provided by UDP SCTP are:

   o  unidirectional  unicast

   o  connection setup with feature negotiation and application-to-port
      mapping

   o  port multiplexing

   o  2-tuple endpoints  reliable or partially reliable delivery

   o  IPv4 broadcast, multicast  ordered and anycast unordered delivery within a stream

   o  IPv6 multicast and anycast  support for multiple concurrent streams

   o  IPv6 jumbograms  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 (checksum) (CRC32C)

   o  checksum optional

3.5.  Lightweight  transport layer multihoming for resilience

   o  transport layer mobility

   [EDITOR'S NOTE: update this list.]

3.4.  User Datagram Protocol (UDP-Lite) (UDP)

   The Lightweight User Datagram Protocol (UDP-Lite) [RFC3828] (UDP) [RFC0768] [RFC2460] is an IETF
   standards track transport protocol.  UDP-Lite  It provides a
   bidirectional set uni-directional,
   datagram protocol which preserves message boundaries.  It provides
   none of logical unicast the following transport features: error correction,
   congestion control, or flow control.  It can be used to send
   broadcast datagrams (IPv4) or multicast message streams
   over a datagram protocol. datagrams (IPv4 and IPv6), in
   addition to unicast (and anycast) datagrams.  IETF guidance on the
   use of UDP-Lite UDP is provided in [RFC5405].

3.5.1. in[RFC5405].  UDP is widely implemented and
   widely used by common applications, especially DNS.

3.4.1.  Protocol Description

   UDP-Lite

   UDP is a connection-less datagram protocol, protocol which maintains message boundaries,
   with no connection setup or feature negotiation.  The protocol use uses
   independent messages, rather than
   a byte-stream.  Each stream ordinarily called datagrams.  The lack of error
   control and flow control implies messages is independently managed,
   therefore retransmission does not hold back data sent using other
   logical streams.

   It provides multiplexing may be damaged, re-ordered,
   lost, or duplicated in transit.  A receiving application unable to multiple sockets on each host using port
   numbers.  An active UDP-Lite session is identified by its four-tuple
   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
   run sufficiently fast or frequently may miss messages.  The lack of IP packet.

   UDP-Lite changes
   congestion handling implies UDP traffic may cause the semantics loss of
   messages from other protocols (e.g., TCP) when sharing the same
   network paths.  UDP "payload length" field to
   that traffic can also cause the loss of a "checksum coverage length" field.  Otherwise, UDP-Lite is
   semantically identical 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 UDP.  Applications using an
   application.  There are some exceptions to this general rule,
   however.  UDP-Lite therefore
   can not make assumptions regarding (see [RFC3828], and below) provides the correctness ability
   for portions of the data
   received 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 insensitive part relatively weak form of the UDP-Lite payload.

   As for checksum
   used by UDP, mechanisms for 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,
   PMTU or PLPMTU discovery, PathMTU discovery/PLPMTUD, 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. than UDP [RFC5405].

3.4.2.  Interface Description

   There is no current

   [RFC0768] describes basic requirements for an API specified in the RFC Series, but guidance for UDP.  Guidance
   on use of common APIs is provided in [RFC5405].

   The interface

   A UDP endpoint consists of UDP-Lite differs from that a tuple of UDP by (IP address, port number).
   Demultiplexing using multiple abstract endpoints (sockets) on the addition of
   same IP address are supported.  The same socket may be used by a
   single (socket) option that communicates a checksum coverage length
   value: at the sender, this specifies the intended checksum coverage, server to interact with the remaining unprotected part multiple clients (note: this behavior
   differs from TCP, which uses a pair of tuples to identify a
   connection).  Multiple server instances (processes) binding the payload called same
   socket can cooperate to service multiple clients- the "error-
   insensitive part".  The checksum coverage may 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 made visible "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 application via the UDP-Lite MIB module [RFC5097].

3.5.3. 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-Lite UDP are:

   o  unicast  unidirectional

   o  port multiplexing

   o  2-tuple endpoints

   o  IPv4 broadcast, multicast and anycast

   o  port multiplexing  IPv6 multicast and anycast

   o  non-reliable, non-ordered delivery  IPv6 jumbograms

   o  message-oriented delivery

   o  partial integrity protection

3.6.  error detection (checksum)

   o  checksum optional

3.5.  Lightweight User Datagram Congestion Control Protocol (DCCP) (UDP-Lite)

   The Lightweight User Datagram Congestion Control Protocol (DCCP) [RFC4340] (UDP-Lite) [RFC3828] is an
   IETF standards track bidirectional transport protocol that protocol.  UDP-Lite 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
   bidirectional set of data and that can benefit from control logical unicast or multicast message streams
   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 datagram protocol.  IETF guidance on the 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. UDP-Lite is
   provided in [RFC5405].

3.5.1.  Protocol Description

   DCCP

   UDP-Lite is a connection-oriented connection-less 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. with no connection
   setup or feature negotiation.  The protocol is defined by use messages, rather than
   a family byte-stream.  Each stream of RFCs. 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 DCCP UDP-Lite 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

   UDP-Lite fragments packets into IP packets, constrained by the
   maximum size of IP packet.

   UDP-Lite changes the service code
   [RFC5595] mechanism to allow transport instantiations to indicate semantics of the
   service treatment UDP "payload length" field to
   that of a "checksum coverage length" field.  Otherwise, UDP-Lite is expected from
   semantically identical to UDP.  Applications using UDP-Lite therefore
   can not make assumptions regarding the correctness of the network.

   The protocol segments data into messages, typically sized to fit
   received 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
   size allowed by insensitive part of the current UDP-Lite payload.

   As for UDP, mechanisms for receiver flow control, congestion control mechanism (CCMPS)
   [RFC4340].

   Each message is identified control,
   PMTU or PLPMTU discovery, support for ECN, etc need to be provided by
   upper layer protocols [RFC5405].

   Examples of use include a sequence number.  The sequence number class of applications that can derive
   benefit from having partially-damaged payloads delivered, rather than
   discarded.  One use 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 error tolerate payload corruption
   when used over paths that include error-prone links, another
   application is a Data Checksum option, which
   contains a strong CRC, lets endpoints detect application data
   corruption.  It also supports reduced checksum coverage, a partial when header integrity checks are required, but payload
   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. provided by some other mechanism (e.g.  [RFC6936].

   A DCCP protocol instance can be extended [RFC4340] UDP-Lite service may support IPv4 broadcast, multicast, anycast and tuned.  Some
   features are sender-side only, requiring
   unicast.

3.5.2.  Interface Description

   There is no negotiation with current API specified in the
   receiver; some are receiver-side only, some are explicitly negotiated
   during connection setup.

   DCCP supports negotiation 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 congestion control profile, to
   provide Plug and Play congestion control mechanisms.  examples addition 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 single (socket) option that allow each client to choose features it wishes to
   support.  Simultaneous open [RFC5596], as in TCP, can enable
   interoperability in communicates a checksum coverage length
   value: at the presence sender, this specifies the intended checksum coverage,
   with the remaining unprotected part of middleboxes. the payload called the "error-
   insensitive part".  The Connect packet
   includes a Service Code field [RFC5595] designed to allow middle
   boxes and endpoints checksum coverage may also be made visible to identify
   the characteristics required application via the UDP-Lite MIB module [RFC5097].

3.5.3.  Transport Protocol Components

   The transport protocol components provided 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).

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 UDP-Lite are:

   o  unicast

   o  connection setup with feature negotiation  IPv4 broadcast, multicast and application-to-port
      mapping

   o  Service Codes anycast

   o  port multiplexing
   o  non-reliable, ordered non-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

3.6.  Datagram Congestion Control Protocol (RTP)

   RTP provides (DCCP)

   Datagram Congestion Control Protocol (DCCP) [RFC4340] is an end-to-end network IETF
   standards track bidirectional transport service, suitable for
   applications transmitting real-time data, such as audio, video or
   data, over multicast or protocol that provides
   unicast network services, including TCP, UDP,
   UDP-Lite, DCCP. connections of congestion-controlled unreliable messages.

   [EDITOR'S NOTE: Varun Singh Gorry Fairhurst signed up as a contributor for this
   section.]

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 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
   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).

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.]

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 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 data dissemination
   includes discrete "objects" such as file or computer memory-based "objects" in-memory content but
   also can treat a stream of data as well 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
   byte- needed.  Application content is
   carried and message-streaming. identified by the NORM protocol with encoding symbol
   identifiers depending upon the Forward Error Correction (FEC) Scheme
   [RFC3452] configured.  NORM uses NACK-based selective ARQ to reliably
   deliver the application content to the receiver(s).  NORM proactively
   measures round-trip timing information to scale ARQ timers
   appropriately and to support congestion control.  For multicast
   operation, timer-based feedback suppression is uses to achieve group
   size scaling with low feedback traffic levels.  The feedback
   suppression is not applied for unicast operation.

   NORM uses rate-based congestion control based upon the TCP-Friendly
   Rate Control (TFRC) [RFC4324] principles that are also used in DCCP
   [RFC4340].  NORM uses control messages to measure RTT and collect
   congestion event (e..g, loss event, ECN event, etc) information from
   the receiver(s) to support dynamic rate control adjustment.  The TCP-
   Friendly Multicast Congestion Control (TFMCC) [RFC4654] used provides
   some extra features to support multicast but is designed functionally
   equivalent to incorporate packet
   erasure coding as an inherent part of its selective ARQ TFRC in response the unicast case.

   NORM's reliability mechanism is decoupled from congestion control.
   This allows alternative arrangements of transport services to receiver negative acknowledgements.  The be
   invoked.  For example, fixed-rate reliable delivery can be supported
   or unreliable (but optionally "better than best effort" via packet
   erasure coding coding) delivery with rate-control per TFRC can
   also be proactively applied for forward protection from packet loss.
   NORM transmissions are governed by TCP-friendly achieved.
   Additionally, alternative congestion control.
   NORM's reliability, control techniques may be
   applied.  For example, TFRC rate control with congestion control, event
   detection based on ECN for links with high packet loss (e.g.,
   wireless) has been implemented and flow control demonstrated with NORM.

   While NORM is NACK-based for reliability transfer, it also supports a
   positive acknowledgment (ACK) mechanism
   are distinct components and that can be separately controlled to meet used for receiver
   flow control.  Again, since this mechanism is decoupled from the
   reliability and congestion control, applications that have different application needs.

3.8.1.  Protocol Description

   [EDITOR'S NOTE:
   needs to be more clear about in this aspect can use the application protocol differently.  One example
   is the use of FEC NORM for quasi-reliable delivery where timely delivery
   of newer content may be favored over completely reliable delivery of
   older content within buffering and packet erasure coding; expand ARQ.] RTT constraints.

3.8.2.  Interface Description

   The NORM specification does not describe a specific application
   programming interface (API) to control protocol is encapsulated in UDP datagrams and thus provides
   multiplexing for multiple sockets on hosts using port numbers.  For
   purposes operation.  A freely-
   available, open source reference implementation of loosely coordinated IP Multicast, NORM is not strictly
   connection-oriented although per-sender state available
   at https://www.nrl.navy.mil/itd/ncs/products/norm, and a documented
   API is maintained by
   receivers provided for protocol operation.  [RFC5740] does not specify this implementation.  While a
   handshake protocol sockets-like API is
   not currently documented, the existing API supports the necessary
   functions for connection establishment and separate session
   initiation can be used that to coordinate port numbers.  However, in-band
   "client-server" style connection establishment can be accomplished
   with the implemented.

3.8.3.  Transport Protocol Components

   The transport protocol components provided by NORM congestion control signaling messages using are:

   o  unicast

   o  multicast

   o  port
   binding techniques like those for TCP client-server connections.

   NORM supports bulk "objects" such as file or multiplexing (UDP ports)
   o  reliable delivery

   o  unordered delivery of in-memory data or file bulk content but
   also can treat 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 logical bulk object for purposes
      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 packet erasure coding.  In TCP, DTLS is designed to run on top of
   UDP.  At the case time of stream transport, NORM can
   support either byte streams or message streams where application- writing, the current version of TLS is 1.2; it
   is defined message boundary information in [RFC5246].  DTLS provides nearly identical
   functionality; it is carried defined in the NORM protocol
   messages.  This allows the receiver(s) to join/re-join {RFC6347}} and recover
   message boundaries mid-stream as needed.  Application content 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
   carried a Best Current Practices (BCP) document
   that describes secure configurations for TLS 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 DTLS to reliably
   deliver counter
   these attacks.  The recommendations are applicable for the application content to vast
   majority of use cases.

   [NOTE: The Logjam authors (weakdh.org) give (inconclusive) evidence
   that one of the receiver(s).  NORM proactively
   measures round-trip timing information to scale ARQ timers
   appropriately and recommendations of [RFC7525], namely use to support congestion control.  For multicast
   operation, timer-based feedback suppression is uses DHE-1024
   as a fallback, may not be sufficient in all cases to achieve group
   size scaling counter an
   attacker with low feedback traffic levels.  The feedback
   suppression the resources of a nation-state.  It is not applied for unicast operation.

   NORM uses rate-based congestion control based upon unclear at this
   time if the TCP-Friendly
   Rate Control (TFRC) [RFC4324] principles that are also used in DCCP
   [RFC4340].  NORM uses control messages RFC is going to measure RTT be updated as a result or whether there
   will be an RFC7525bis.]

3.9.1.  Protocol Description

   Both TLS and collect
   congestion event (e..g, loss event, ECN event, etc) information from DTLS provide the receiver(s) to support dynamic rate control adjustment. same security features and can thus be
   discussed together.  The TCP-
   Friendly Multicast Congestion Control (TFMCC) [RFC4654] used provides
   some extra features to support multicast but 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 functionally
   equivalent to TFRC implicit in the unicast
   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.

   NORM's reliability mechanism  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 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 authenticated.  This is
   part of the application logic; i.e. the authentication decision rests
   with rate-control per TFRC can be achieved.
   Additionally, alternative congestion control techniques may be
   applied.  For the application.  As an 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 in the common use case of
   authentication by means of an X.509 certificate, it is NACK-based the
   application's decision whether the certificate of the peering entity
   is acceptable for reliability transfer, it also supports a
   positive acknowledgment (ACK) mechanism that can the purposes of the application or whether the
   handshake should be used for receiver
   flow control.  Again, since this mechanism aborted.

   As DTLS is decoupled from used over the
   reliability and congestion control, applications that have different unreliable UDP transport, it needs in 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 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. achieved?]

3.9.2.  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

   TLS is available
   at https://www.nrl.navy.mil/itd/ncs/products/norm, and commonly used with a documented
   API socket-like interface, although details
   can vary between implementations.  This is provided particularly true for this implementation.  While a sockets-like API is
   not currently documented, the existing API supports the necessary
   functions
   choice which cryptographic algorithms to use, see below.

   [TODO: DTLS interface]
   Both TLS and DTLS allow to employ a multitude of cipher suites for that
   encryption, hashing and applying message integrity.  It is no easy
   task to be implemented.

3.8.3. 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 transport protocol components provided by NORM are:

   o  unicast

   o  multicast

   o  port multiplexing (UDP ports)

   o  reliable delivery

   o  ordered delivery lower layer is
   commonly called the record protocol.  It is responsible for each byte or
   fragmenting messages, applying message stream

   o  unordered delivery of in-memory data or file bulk content objects

   o  error detection (UDP checksum)

   o  segmentation

   o  stream-oriented delivery in a single stream

   o  object-oriented delivery of discrete data or file items

   o  data bundling (Nagle's algorithm)

   o  flow control (timer-based and/or ack-based)

   o  congestion control

   o  packet erasure coding (both proactively authentication codes (MACs),
   encrypting data, and as part sending it via the underlying transport
   protocol.  Several essential protocols run on top of ARQ)

3.9.  Transport Layer Security (TLS) the record
   protocol in order to carry out the handshake and Datagram TLS (DTLS) as establish a
      pseudotransport

   [NOTE: A few words on secure
   session.

   [EDITOR'S NOTE: TLS [RFC5246] and DTLS [RFC6347] here, and how
   they get used by other protocols can also compress, but this has been found to meet be
   a security goals as an add-on
   interlayer above transport.  Kevin Fall will contribute text to this
   section.]

3.9.1.  Protocol Description

3.9.2.  Interface Description

3.9.3.  Transport Protocol Components 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.  Also, though

   Though not strictly bound to TCP, HTTP is almost exclusively run over
   TCP, and therefore inherits its properties when used in this way.  This can have disadvantages, when
   the application does not naturally require single-streamed, reliable
   transport.

3.10.1.  Protocol Description

   Hypertext Transfer Protocol (HTTP) is a request/response protocol.  A
   client sends a request containing a request method, URI and protocol
   version followed by a MIME-like message (see [RFC7231] for the
   differences between an HTTP object and a MIME message), containing
   information about the client and request modifiers.  The message can
   contain a message body carrying application data as well.  The server
   responds with a status or error code followed by a MIME-like message
   containing information about the server and information about carried
   data and it can include a message body.  It is possible to specify a
   data format for the message body using MIME media types [RFC2045].
   Furthermore, the protocol has numerous additional features; features
   relevant to pseudotransport are described below.

   Content negotiation, specified in [RFC7231], is a mechanism provided
   by HTTP for selecting a representation on a requested resource.  The
   client and server negotiate acceptable data formats, charsets, data
   encoding (e.g. data can be transferred compressed, gzip), etc.  HTTP
   can accommodate exchange of messages as well as data streaming (using
   chunked transfer encoding [RFC7230]).  It is also possible to request
   a part of a resource using range requests specified in [RFC7233].
   The protocol provides powerful cache control signalling defined in
   [RFC7234].

   HTTP 1.1's and HTTP 2.0's persistent connections can be use to
   perform multiple request-response transactions during the life-time
   of a single HTTP connection.  Moreover, HTTP 2.0 connections can
   multiplex many request/response pairs in parallel on a single
   connection.  This reduces connection establishment overhead and the
   effect of TCP slow-start on each transaction, important for HTTP's
   primary use case.

   It is possible to combine HTTP with security mechanisms, like TLS
   (denoted by HTTPS), which adds protocol properties provided by such a
   mechanism (e.g. authentication, encryption, etc.).  TLS's
   Application-Layer Protocol Negotiation (ALPN) extension [RFC7301] can
   be used for HTTP version negotiation within TLS handshake which
   eliminates addition round-trip.  Arbitrary cookie strings, included
   as part of the MIME headers, are often used as bearer tokens in HTTP.

   Application layer protocols using HTTP as substrate may use existing
   method and data formats, or specify new methods and data formats.
   Furthermore some protocols may not fit a request/response paradigm
   and instead rely on HTTP to send messages (e.g.  [RFC6546]).  Because
   HTTP is working in many restricted infrastructures, it is also used
   to tunnel other application-layer protocols.

3.10.2.  Interface Description

   There are many HTTP libraries available exposing different APIs.  The
   APIs provide a way to specify a request by providing a URI, a method,
   request modifiers and optionally a request body.  For the response,
   callbacks can be registered that will be invoked when the response is
   received.  If TLS is used, API expose a registration of callbacks in
   case a server requests client authentication and when certificate
   verification is needed.

   World Wide Web Consortium (W3C) standardized the XMLHttpRequest API
   [XHR], an API that can be use for sending HTTP/HTTPS requests and
   receiving server responses.  Besides XML data format, request and
   response data format can also be JSON, HTML and plain text.
   Specifically JavaScript and XMLHttpRequest are a ubiquitous
   programming model for websites, and more general applications, where
   native code is less attractive.

   Representational State Transfer (REST) [REST] is another example how
   applications can use HTTP as transport protocol.  REST is an
   architecture style for building application on the Internet.  It uses
   HTTP as a communication protocol.

3.10.3.  Transport Protocol Components

   The transport protocol components provided by HTTP, when used as a
   pseudotransport over TCP,
   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
   o  integrity protection

3.11.  WebSockets

   [RFC6455]

   [EDITOR'S NOTE: Salvatore Loreto will contribute text for this
   section.]

3.11.1.  Protocol Description

3.11.2.  Interface Description

3.11.3.  Transport Protocol Components

4.  Transport Service Features

   The transport protocol components analyzed in this document which can
   be used as a basis for defining common transport service features,
   normalized and separated into categories, are as follows:

   o  Destination selection

      *  unicast

      *  broadcast (IPv4 only)

      *  multicast

      *  anycast

      *  transport layer multihoming for resilience

      *  transport layer mobility

      *  port multiplexing

      *  service codes

   o  Connection setup

      *  connection setup with feature negotiation and application-to-
         port mapping

   o  Delivery

      *  reliable delivery
      *  partially reliable delivery

      *  unreliable delivery

      *  packet erasure coding

      *  ordered delivery

      *  unordered delivery

      *  stream-oriented delivery

      *  message-oriented delivery

      *  message fragmentation

      *  object-oriented delivery of discrete data or file items

      *  unordered delivery of in-memory data or file bulk content
         objects

      *  object range request

      *  object content type negotiation

      *  single streaming

      *  multiple streaming

      *  stream scheduling prioritization

      *  segmentation

      *  data bundling (Nagle's algorithm)

      *  message bundling

   o  Transmission control

      *  timer-based rate control

      *  ack-based flow control

      *  drop notification

      *  packet erasure coding

      *  congestion control

   o  Integrity protection

      *  checksum for error detection

      *  partial checksum protection

      *  checksum optional

      *  cryptographic integrity protection

   o  Security

      *  authentication of one end of a connection

      *  authentication of both ends of a connection

      *  confidentiality

   The next revision of this document will define transport service
   features based upon this list.

   [EDITOR'S NOTE: this section will drawn from the candidate features
   provided by protocol components in the previous section - please
   discuss on taps@ietf.org list]

4.1.  Complete Protocol Feature Matrix

   [EDITOR'S NOTE: Dave Thaler has signed up as a contributor for this
   section.  Michael Welzl also has a beginning of a matrix which could
   be useful here.]

   [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.

   +-----------------+---------+---------+---------+---------+---------+
   | 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.

   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.

   [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.

   [RFC3205]  Moore, K., "On the use of HTTP as a Substrate", BCP 56,
              RFC 3205, February 2002.

   [RFC3390]  Allman, M., Floyd, S., and C. Partridge, "Increasing TCP's
              Initial Window", RFC 3390, October 2002.

   [RFC3436]  Jungmaier, A., Rescorla, E., and M. Tuexen, "Transport
              Layer Security over Stream Control Transmission Protocol",
              RFC 3436, December 2002.

   [RFC3452]  Luby, M., Vicisano, L., Gemmell, J., Rizzo, L., Handley,
              M., and J. Crowcroft, "Forward Error Correction (FEC)
              Building Block", RFC 3452, December 2002.

   [RFC3758]  Stewart, R., Ramalho, M., Xie, Q., Tuexen, M., and P.
              Conrad, "Stream Control Transmission Protocol (SCTP)
              Partial Reliability Extension", RFC 3758, May 2004.

   [RFC3828]  Larzon, L-A., Degermark, M., Pink, S., Jonsson, L-E., and
              G. Fairhurst, "The Lightweight User Datagram Protocol
              (UDP-Lite)", RFC 3828, July 2004.

   [RFC4324]  Royer, D., Babics, G., and S. Mansour, "Calendar Access
              Protocol (CAP)", RFC 4324, December 2005.

   [RFC4336]  Floyd, S., Handley, M., and E. Kohler, "Problem Statement
              for the Datagram Congestion Control Protocol (DCCP)", RFC
              4336, March 2006.

   [RFC4340]  Kohler, E., Handley, M., and S. Floyd, "Datagram
              Congestion Control Protocol (DCCP)", RFC 4340, March 2006.

   [RFC4341]  Floyd, S. and E. Kohler, "Profile for Datagram Congestion
              Control Protocol (DCCP) Congestion Control ID 2: TCP-like
              Congestion Control", RFC 4341, March 2006.

   [RFC4342]  Floyd, S., Kohler, E., and J. Padhye, "Profile for
              Datagram Congestion Control Protocol (DCCP) Congestion
              Control ID 3: TCP-Friendly Rate Control (TFRC)", RFC 4342,
              March 2006.

   [RFC4614]  Duke, M., Braden, R., Eddy, W., and E. Blanton, "A Roadmap
              for Transmission Control Protocol (TCP) Specification
              Documents", RFC 4614, September 2006.

   [RFC4654]  Widmer, J. and M. Handley, "TCP-Friendly Multicast
              Congestion Control (TFMCC): Protocol Specification", RFC
              4654, August 2006.

   [RFC4820]  Tuexen, M., Stewart, R., and P. Lei, "Padding Chunk and
              Parameter for the Stream Control Transmission Protocol
              (SCTP)", RFC 4820, March 2007.

   [RFC4821]  Mathis, M. and J. Heffner, "Packetization Layer Path MTU
              Discovery", RFC 4821, March 2007.

   [RFC4895]  Tuexen, M., Stewart, R., Lei, P., and E. Rescorla,
              "Authenticated Chunks for the Stream Control Transmission
              Protocol (SCTP)", RFC 4895, August 2007.

   [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.

   [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.

   [RFC6935]  Eubanks, M., Chimento, P., and M. Westerlund, "IPv6

   [RFC6347]  Rescorla, E. and
              UDP Checksums for Tunneled Packets", N. Modadugu, "Datagram Transport Layer
              Security Version 1.2", RFC 6935, April 2013.

   [RFC6936]  Fairhurst, G. 6347, January 2012.

   [RFC6356]  Raiciu, C., Handley, M., and M. Westerlund, "Applicability Statement D. Wischik, "Coupled
              Congestion Control for the Use of IPv6 UDP Datagrams with Zero Checksums", Multipath Transport Protocols", RFC 6936, April 2013.
              6356, October 2011.

   [RFC6455]  Fette, I. and A. Melnikov, "The WebSocket Protocol", RFC
              6455, December 2011.

   [RFC6347]  Rescorla, E. and N. Modadugu, "Datagram Transport Layer
              Security Version 1.2", RFC 6347, January 2012.

   [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.

   [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.

   [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.

   [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