Network Working Group                                  G. Fairhurst, Ed.
Internet-Draft                                    University of Aberdeen
Intended status: Informational                          B. Trammell, Ed.
Expires: June 10, July 31, 2016                                M. Kuehlewind, Ed.
                                                              ETH Zurich
                                                       December 08, 2015
                                                        January 28, 2016

  Services provided by IETF transport protocols and congestion control
                               mechanisms
                     draft-ietf-taps-transports-08
                     draft-ietf-taps-transports-09

Abstract

   This document describes transport services describes, surveys, classifies and compares the
   protocol mechanisms provided by existing IETF
   protocols. protocols, as
   background for determining a common set of transport services.  It is designed to help application
   examines the Transmission Control Protocol (TCP), Multipath TCP, the
   Stream Control Transmission Protocol (SCTP), the User Datagram
   Protocol (UDP), UDP-Lite, the Datagram Congestion Control Protocol
   (DCCP), the Internet Control Message Protocol (ICMP), the Realtime
   Transport Protocol (RTP), File Delivery over Unidirectional
   Transport/Asynchronous Layered Coding Reliable Multicast (FLUTE/ALC),
   and network stack
   programmers NACK-Oriented Reliable Multicast (NORM), Transport Layer Security
   (TLS), Datagram TLS (DTLS), and to inform the work of the IETF TAPS Working Group. Hypertext Transport Protocol
   (HTTP) when used as a pseudotransport.

Status of This Memo

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   This Internet-Draft will expire on June 10, July 31, 2016.

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   Copyright (c) 2015 2016 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

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Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   3
   2.  Terminology . . . . . . . . . . . .
     1.1.  Overview of Transport Features  . . . . . . . . . . . . .   4
   3.  Transport Service Features  . . . . . . . . . . .
   2.  Terminology . . . . . .   4
     3.1.  Congestion Control . . . . . . . . . . . . . . . . . . .   5
   4.
   3.  Existing Transport Protocols  . . . . . . . . . . . . . . . .   6
     4.1.   5
     3.1.  Transport Control Protocol (TCP)  . . . . . . . . . . . .   6
       4.1.1.
       3.1.1.  Protocol Description  . . . . . . . . . . . . . . . .   6
       4.1.2.
       3.1.2.  Interface description . . . . . . . . . . . . . . . .   8
       4.1.3.
       3.1.3.  Transport Features  . . . . . . . . . . . . . . . . .   8
     4.2.
     3.2.  Multipath TCP (MPTCP) . . . . . . . . . . . . . . . . . .   9
       4.2.1.
       3.2.1.  Protocol Description  . . . . . . . . . . . . . . . .   9
       4.2.2.
       3.2.2.  Interface Description . . . . . . . . . . . . . . . .   9
       4.2.3.
       3.2.3.  Transport features  . . . . . . . . . . . . . . . . .  10
     4.3.
     3.3.  Stream Control Transmission Protocol (SCTP) . . . . . . .  10
       4.3.1.
       3.3.1.  Protocol Description  . . . . . . . . . . . . . . . .  11
       4.3.2.
       3.3.2.  Interface Description . . . . . . . . . . . . . . . .  13
       4.3.3.
       3.3.3.  Transport Features  . . . . . . . . . . . . . . . . .  15
     4.4.
     3.4.  User Datagram Protocol (UDP)  . . . . . . . . . . . . . .  16
       4.4.1.
       3.4.1.  Protocol Description  . . . . . . . . . . . . . . . .  16
       4.4.2.
       3.4.2.  Interface Description . . . . . . . . . . . . . . . .  17
       4.4.3.
       3.4.3.  Transport Features  . . . . . . . . . . . . . . . . .  18
     4.5.  17
     3.5.  Lightweight User Datagram Protocol (UDP-Lite) . . . . . .  18
       4.5.1.
       3.5.1.  Protocol Description  . . . . . . . . . . . . . . . .  18
       4.5.2.
       3.5.2.  Interface Description . . . . . . . . . . . . . . . .  19
       4.5.3.
       3.5.3.  Transport Features  . . . . . . . . . . . . . . . . .  19
     4.6.
     3.6.  Datagram Congestion Control Protocol (DCCP) . . . . . . .  20
       4.6.1.  19
       3.6.1.  Protocol Description  . . . . . . . . . . . . . . . .  20
       4.6.2.
       3.6.2.  Interface Description . . . . . . . . . . . . . . . .  21
       4.6.3.
       3.6.3.  Transport Features  . . . . . . . . . . . . . . . . .  22
     4.7.  21
     3.7.  Internet Control Message Protocol (ICMP)  . . . . . . . .  22
       4.7.1.
       3.7.1.  Protocol Description  . . . . . . . . . . . . . . . .  23
       4.7.2.  22
       3.7.2.  Interface Description . . . . . . . . . . . . . . . .  24
       4.7.3.  23
       3.7.3.  Transport Features  . . . . . . . . . . . . . . . . .  24
     4.8.  23
     3.8.  Realtime Transport Protocol (RTP) . . . . . . . . . . . .  24
       4.8.1.  23
       3.8.1.  Protocol Description  . . . . . . . . . . . . . . . .  24
       4.8.2.
       3.8.2.  Interface Description . . . . . . . . . . . . . . . .  25
       4.8.3.
       3.8.3.  Transport Features  . . . . . . . . . . . . . . . . .  26
     4.9.  25
     3.9.  File Delivery over Unidirectional Transport/Asynchronous
           Layered Coding Reliable Multicast (FLUTE/ALC) . . . . . .  26
       4.9.1.  25
       3.9.1.  Protocol Description  . . . . . . . . . . . . . . . .  27
       4.9.2.  26
       3.9.2.  Interface Description . . . . . . . . . . . . . . . .  29
       4.9.3.  28
       3.9.3.  Transport Features  . . . . . . . . . . . . . . . . .  29
     4.10.  28
     3.10. NACK-Oriented Reliable Multicast (NORM) . . . . . . . . .  30
       4.10.1.  29
       3.10.1.  Protocol Description . . . . . . . . . . . . . . . .  30
       4.10.2.  29
       3.10.2.  Interface Description  . . . . . . . . . . . . . . .  31
       4.10.3.  30
       3.10.3.  Transport Features . . . . . . . . . . . . . . . . .  31
     4.11.  30
     3.11. Transport Layer Security (TLS) and Datagram TLS (DTLS) as
           a pseudotransport . . . . . . . . . . . . . . . . . . . .  32
       4.11.1.  31
       3.11.1.  Protocol Description . . . . . . . . . . . . . . . .  32
       4.11.2.  31
       3.11.2.  Interface Description  . . . . . . . . . . . . . . .  33
       4.11.3.  32
       3.11.3.  Transport Features . . . . . . . . . . . . . . . . .  34
     4.12.  33
     3.12. Hypertext Transport Protocol (HTTP) over TCP as a
           pseudotransport . . . . . . . . . . . . . . . . . . . . .  35
       4.12.1.  34
       3.12.1.  Protocol Description . . . . . . . . . . . . . . . .  35
       4.12.2.
       3.12.2.  Interface Description  . . . . . . . . . . . . . . .  36
       4.12.3.  35
       3.12.3.  Transport features . . . . . . . . . . . . . . . . .  36
   4.  Congestion Control  . . . . . . . . . . . . . . . . . . . . .  37
   5.  Transport Service Features  . . . . . . . . . . . . . . . . .  37 . . . .  38
   6.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  41  42
   7.  Security Considerations . . . . . . . . . . . . . . . . . . .  41  42
   8.  Contributors  . . . . . . . . . . . . . . . . . . . . . . . .  41  42
   9.  Acknowledgments . . . . . . . . . . . . . . . . . . . . . . .  42  43
   10. Informative References  . . . . . . . . . . . . . . . . . . .  42  43
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  52  53

1.  Introduction

   Internet applications make use of the Services provided by a
   Transport protocol, such as TCP (a reliable, in-order stream
   protocol) or UDP (an unreliable datagram protocol).  We use the term
   "Transport Service" to mean the end-to-end service provided to an
   application by the transport layer.  That service can only be
   provided correctly if information about the intended usage is
   supplied from the application.  The application may determine this
   information at design time, compile time, or run time, and may
   include guidance on whether a feature is required, a preference by
   the application, or something in between.  Examples of features of
   Transport Services are reliable delivery, ordered delivery, content
   privacy to in-path devices, and integrity protection.

   The IETF has defined a wide variety of transport protocols beyond TCP
   and UDP, including SCTP, DCCP, MP-TCP, MPTCP, and UDP-Lite.  Transport
   services may be provided directly by these transport protocols, or
   layered on top of them using protocols such as WebSockets (which runs
   over TCP), RTP (over TCP or UDP) or WebRTC data channels (which run
   over SCTP over DTLS over UDP or TCP).  Services built on top of UDP
   or UDP-Lite typically also need to specify additional mechanisms,
   including a congestion control mechanism (such as NewReno, TFRC or
   LEDBAT).  This extends the set of available Transport Services beyond
   those provided to applications by TCP and UDP.

2.  Terminology

   The following terms are defined throughout this document, and in
   subsequent documents produced

1.1.  Overview of Transport Features

   Transport protocols can be differentiated by TAPS describing the composition and
   decomposition features of transport services.

   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
   services they provide.

   Some of these provided features and configuration parameters are closely related to basic control
   function that
      implements a single transport service, e.g., a protocol stack (RTP needs to work 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.  Transport Service Features

   Transport protocols can be differentiated by the features of the
   services they provide.

   One fundamental feature is whether a transport offers a service that
   divides the data into transmission units based on network packets
   (known path, such as
   addressing.  The number of participants in a Datagram service), or whether it combines and segments
   data across given association also
   determines its applicability: if a connection is between endpoints
   (unicast), to one of multiple packets (e.g., the Stream service provided by
   TCP). endpoints (anycast), and/or
   simultaneously to multiple endpoints (multicast).  Unicast protocols
   usually support bidirectional communication, while multicast is
   generally unidirectional.  Another fundamental feature is whether a transport
   requires a control exchange across the network at setup (e.g., TCP),
   or whether it connection-less (e.g., UDP).

   A transport service can also offer reliability, for instance, SCTP
   offers a message-based service providing full or partial reliability
   and allowing to minimize

   For the head delivery of line blocking due to the support
   of unordered packets itself, reliability and integrity
   protection, ordering, and framing are basic features.  However, these
   features are implemented with different levels of assurance in
   different protocols.  As an example, a transport service may provide
   full reliability, providing detection of loss and retransmission
   (e.g., TCP).  SCTP offers a message-based service that can provide
   full or partial reliability, and allows the protocol to minimize the
   head of line blocking due to the support of ordered and unordered
   message delivery within multiple streams, streams.  UDP-Lite and DCCP can
   provide partial integrity protection.

   A protection to enable corruption tolerance.

   Usually a protocol has been designed to support one specific type of
   delivery/framing: data either needs to be divided into transmission
   units based on network packets (datagram service), a data stream is
   segmented and re-combined across multiple packets (stream service),
   or whole objects such as files are handled accordingly.  This
   decision strongly influences the interface that is provided to the
   upper layer.

   In addition, transport protocols offer a certain support on
   transmission control.  For example, a transport service can provide
   flow control to allow a receiver to regulate the transmission rate of
   a sender.  Further a transport service can provide congestion control
   (see Section 3.1). 4).  As an example TCP and SCTP provide congestion
   control for use in the Internet, whereas UDP leaves this function to
   the upper layer protocol that uses UDP.  DCCP offers a range of congestion control approaches and
   LEDBAT can support low-priority "scavenger" communication, intending
   to defer use

   Security features are often provided independent of capacity to other Internet flows sharing a congested
   bottleneck. the transport
   protocol, via Transport services may be unidirectional or bidirectional, to a
   single a single endpoint, to one of multiple endpoints, Layer Security (TLS, see {{transport-layer-
   security-tls-and- datagram-tls-dtls-as-a-pseudotransport}}) or multicast
   simultaneously to multiple endpoints.

   The service offered by transport protocols the
   application layer protocol itself.  The security properties TLS
   provides to the application (such as confidentiality, integrity, and frameworks can
   authenticity) are also be
   differentiated features of the transport layer, even though
   they are often presently implemented in many other ways.

3.1.  Congestion Control

   Congestion control is critical to the stable operation of the
   Internet, applications and other protocols that choose to use a
   datagram protocol (e.g., UDP or UDP-Lite) need to employ mechanisms
   to prevent congestion collapse and to establish some degree of
   fairness with concurrent traffic.

   A variety of techniques separate protocol.

2.  Terminology

   The following terms are used to provide congestion control throughout this document, and in the
   Internet.  Each technique requires
   subsequent documents produced by TAPS that describe the protocol provide composition
   and decomposition of transport services.

   Transport Service Feature:  a method
   for deriving the metric specific end-to-end feature that the congestion control algorithm uses
      transport layer provides to
   detect congestion and the property of an application.  Examples include
      confidentiality, reliable delivery, ordered delivery, message-
      versus-stream orientation, etc.

   Transport Service:  a packet it uses set of Transport Features, without an
      association to determine
   when any given framing protocol, which provides a
      complete service to send.  Given these relatively wide constraints, the
   congestion control techniques an application.

   Transport Protocol:  an implementation that can be applied by provides one or more
      different transport protocols are largely orthogonal to services using a specific framing and header
      format on the choice wire.

   Transport Service Instance:  an arrangement of transport protocols themselves.
      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 an overview a list of the
   congestion control techniques available to the protocols described in
   Section 4.

   Most commonly deployed congestion control mechanisms use one of three
   mechanisms to detect congestion:

   o  detection of loss, which is interpreted as a congestion signal;

   o  Explicit Congestion Notification (ECN) [RFC3168] to provide
      explicit signaling of congestion without inducing loss (see
      [I-D.ietf-aqm-ecn-benefits]); and/or

   o  a retransmission timer with exponential back-off.

   Protocols such as SCTP and TCP [RFC5681] that use sliding-window-
   based receiver flow control commonly use a separate congestion window
   for congestion control.  Each time congestion is detected, this
   separate congestion window is reduced.  Data in flight is capped to
   the minimum of the two windows.  This approach is also used by DCCP
   CCID-2 for datagram congestion control.

   Rate-based methods have also been defined based on the loss ratio and
   observed round trip time, such as TFRC [RFC5348] and TFRC-SP
   [RFC4828].  These methods utlise a throughput equation to determine
   the maximum acceptable rate.  Such methods are used with DCCP CCID-3
   [RFC4342] and CCID-4 [RFC5622], WEBRC [RFC3738], and other
   applications.

   In addition, a congestion control mechanism may react to changes in
   delay as an indication for congestion.  Delay-based congestion
   detection methods tend to induce less loss than loss-based methods,
   and therefore generally do not compete well with them across shared
   bottleneck links.  However, such methods, such as LEDBAT [RFC6824],
   are are deployed in the Internet for scavenger traffic, which will
   use unused capacity but readily yield to presumably interactive or
   otherwise higher-priority, loss-based congestion-controlled traffic.

4.  Existing Transport Protocols

   This section provides a list of known IETF transport known IETF transport protocols and
   transport protocol frameworks.  It does not make an assessment about
   whether specific implementations of protocols are fully compliant to
   current IETF specifications.

4.1.

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.

4.1.1.

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
   negotiate features, and mechanisms for orderly completion and
   immediate teardown of a connection.  TCP is defined by a family of
   RFCs [RFC4614]. [RFC7414].

   TCP provides multiplexing to multiple sockets on each host using port
   numbers.  A similar approach is adopted by other IETF-defined
   transports.  An active TCP session is identified by its four-tuple of
   local and remote IP addresses and local port and remote port numbers.
   The destination port during connection setup is often used to
   indicate the requested service.

   TCP partitions a continuous stream of bytes into segments, sized to
   fit in IP packets. packets based on a negotiated maximum segment size and
   further constrained by the effective MTU from PMTUD.  ICMP-based Path
   MTU discovery [RFC1191][RFC1981] as well as Packetization Layer Path
   MTU Discovery (PMTUD) [RFC4821] have been defined by the IETF.

   Each byte in the stream is identified by a sequence number.  The
   sequence number is used to order segments on receipt, to identify
   segments in acknowledgments, and to detect unacknowledged segments
   for retransmission.  This is the basis of the reliable, ordered
   delivery of data in a TCP stream.  TCP Selective Acknowledgment
   (SACK) [RFC2018] extends this mechanism by making it possible to identify
   missing
   provide earlier identification of which segments more precisely, reducing are missing,
   allowing faster retransmission.  SACK-based methods (e.g.  DSACK) can
   also result in less 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 provides senders provide congestion control [RFC5681], control, such as described in
   [RFC5681].  TCP's congestion control mechanism is tied to a sliding
   window as well [RFC5681].  Examples for different kind of congestion
   control schemes are given in Section 4.  The sending window at a
   given point in time is the minimum of the receiver window and the
   congestion window.  The congestion window is increased in the absence
   of congestion and, respectively, decreased if congestion is detected.
   Often loss is implicitly handled as a congestion indication which is
   detected in TCP (also as input for retransmission handling) based on
   two mechanisms: A retransmission timer with exponential back-up or
   the reception of three acknowledgment for the same segment, so called
   duplicated ACKs (Fast retransmit).  In addition, Explicit Congestion
   Notification (ECN) [RFC3168] can be used in TCP, if supported by both
   endpoints, that allows a network node to signal congestion without
   inducing loss.  Alternatively, a delay-based congestion control
   scheme can be used in TCP that reacts to changes in delay as an early
   indication of congestion as also further described in Section 3.1 below. 4.

   TCP protocol instances can be extended [RFC4614] [RFC7414] 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.

   TCP may buffer data, e.g., to optimize processing or capacity usage.
   TCP can therefore provides mechanisms to control this, including an
   optional "PUSH" function [RFC0793] that explicitly requests the
   transport service not to delay data.  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 reduce latency for interactive
   sessions.

   TCP provides an "urgent data" function for limited out-of-order
   delivery of the data.  This function is deprecated [RFC6093].

   A mandatory checksum provides a basic integrity check against
   misdelivery and data corruption over the entire packet.  Applications
   that require end to end integrity of data are recommended to include
   a stronger integrity check of their payload data.  The TCP checksum
   does not support partial corruption payload protection (as in DCCP/UDP-Lite).

   TCP supports only unicast connections.

4.1.2.

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.

   [RFC1122] describes extensions of the TCP/application layer interface
   for:

   o  reporting soft errors such as reception of ICMP error messages,
      extensive retransmission or urgent pointer advance,

   o  providing a possibility to specify the Differentiated Services
      Code Point (DSCP) [RFC3260] (formerly, the Type-of-Service, TOS)
      for segments,

   o  providing a flush call to empty the TCP send queue, and

   o  multihoming support.

   In API implementations derived from the BSD Sockets API, TCP sockets
   are created using the "SOCK_STREAM" socket type as described in the
   IEEE Portable Operating System Interface (POSIX) Base Specifications
   [POSIX].  The features used by a protocol instance may be set and
   tuned via this API.  There are current currently no documents in the RFC
   Series that describe this interface.

4.1.3.

3.1.3.  Transport Features

   The transport features provided by TCP are:

   o  unicast  connection-oriented transport

   o  connection setup with feature negotiation and
      application-to-port
      mapping, implemented mapping (implemented using SYN segments and
      the TCP option field to negotiate features. features),

   o  port multiplexing: each  unicast transport (though anycast TCP session is uniquely identified by a
      combination implemented, at risk of the ports and IP address fields.
      instability due to rerouting),

   o  port multiplexing,

   o  Uni-or  uni- or bidirectional communication. communication,

   o  stream-oriented delivery in a single stream. stream,

   o  fully reliable delivery, implemented delivery (implemented using ACKs sent from the
      receiver to confirm delivery. delivery),

   o  error detection: detection (implemented using a segment checksum verifies to verify
      delivery to the correct endpoint and integrity of the data and options.
      options),

   o  segmentation: packets are fragmented to a negotiated maximum
      segment size, further constrained by the effective MTU from PMTUD.  segmentation,

   o  data bundling, an optional mechanism that bundling (optional; uses Nagle's algorithm to coalesce data
      sent within the same RTT into full-sized
      segments. segments),

   o  flow control (implemented using a window-based mechanism, mechanism where the
      receiver advertises the window that it is willing to buffer. buffer),

   o  congestion control: control (usually implemented using a window-based method that uses Additive
      Increase Multiplicative Decrease (AIMD) to control the sending
      rate
      mechanism and to conservatively choose a rate after four algorithm for different phases of the
      transmission: slow start, congestion is
      detected.

4.2. avoidance, fast retransmit,
      and fast recovery [RFC5681]).

3.2.  Multipath TCP (MPTCP)

   Multipath TCP [RFC6824] is an extension for TCP to support multi-
   homing.
   homing for resilience, mobility and load-balancing.  It is designed
   to be as transparent as possible to middle-
   boxes. middleboxes.  It does so by
   establishing regular TCP flows between a pair of source/destination
   endpoints, and multiplexing the application's stream over these
   flows.

4.2.1.  Sub-flows can be started over IPv4 or IPv6 for the same
   session.

3.2.1.  Protocol Description

   MPTCP uses TCP options for its control plane.  They are used to
   signal multipath capabilities, as well as to negotiate data sequence
   numbers, and advertise other available IP addresses and establish new
   sessions between pairs of endpoints.

4.2.2.

   By multiplexing one byte stream over separate paths, MPTCP can
   achieve a higher throughput than TCP in certain situations.  However,
   if coupled congestion control [RFC6356] is used, it might limit this
   benefit to maintain fairness to other flows at the bottleneck.  When
   aggregating capacity over multiple paths, and depending on the way
   packets are scheduled on each TCP subflow, additional delay and
   higher jitter might be observed observed before in-order delivery of
   data to the applications.

3.2.2.  Interface Description

   By default, MPTCP exposes the same interface as TCP to the
   application.  [RFC6897] however describes a richer API for MPTCP-
   aware applications.

   This Basic API describes how an application can:

   o  enable or disable MPTCP.

   o  bind a socket to one or more selected local endpoints.

   o  query local and remote endpoint addresses.

   o  get a unique connection identifier (similar to an address-port
      pair for TCP).

   The document also recommends the use of extensions defined for SCTP
   [RFC6458] (see next section) to support multihoming.

4.2.3. multihoming for resilience
   and mobility.

3.2.3.  Transport features

   As an extension to TCP, MPTCP provides mostly the same features.  By
   establishing multiple sessions between available endpoints, it can
   additionally provide soft failover solutions should one of the paths
   become unusable.  In addition, by multiplexing one byte stream over
   separate paths, it can achieve a higher throughput than TCP in
   certain situations.  Note, however, that coupled congestion control
   [RFC6356] might limit this benefit to maintain fairness to other
   flows at the bottleneck.  When aggregating capacity over multiple
   paths, and depending on the way packets are scheduled on each TCP
   subflow, an additional delay and higher jitter might be observed
   observed before in-order delivery of data to the applications.

   The transport features provided by MPTCP in addition to TCP therefore
   are:

   o  congestion control  multihoming for load-balancing, with load balancing over multiple connections.

   o endpoint multiplexing of a
      single byte stream (higher throughput). stream, using either coupled congestion control or for
      throughput maximization,

   o  address family multiplexing: sub-flows can be started over multiplexing (using IPv4 or and IPv6 for the same session.
      session),

   o  resilience to network failure and/or handover.

4.3.

3.3.  Stream Control Transmission Protocol (SCTP)

   SCTP is a message-oriented IETF standards track transport protocol.
   The base protocol is specified in [RFC4960].  It supports multi-
   homing and path failover to provide resilience to path failures.  An
   SCTP association has multiple streams in each direction, providing
   in-sequence delivery of user messages within each stream.  This
   allows it to minimize head of line blocking.  SCTP supports multiple
   stream scheduling schemes controlling stream multiplexing, including
   priority and fair weighting schemes.

   SCTP is extensible.  Currently defined extensions include mechanisms
   for dynamic re-configuration of streams [RFC6525] and IP addresses

   [RFC5061].  Furthermore, the extension specified in [RFC3758]
   introduces the concept of partial reliability for user messages.

   SCTP was originally developed for transporting telephony signalling signaling
   messages and is deployed in telephony signalling signaling networks, especially
   in mobile telephony networks.  It can also be used for other
   services, for example example, in the WebRTC framework for data channels.  It
   is therefore deployed in all Web browsers supporting WebRTC.

4.3.1.

3.3.1.  Protocol Description

   SCTP is a connection-oriented protocol using a four way handshake to
   establish an SCTP association, and a three way message exchange to
   gracefully shut it down.  It uses the same port number concept as
   DCCP, TCP, UDP, and UDP-Lite.  SCTP only supports unicast.

   SCTP uses the 32-bit CRC32c for protecting SCTP packets against bit
   errors and misdelivery of packets to an unintended endpoint.  This is
   stronger than the 16-bit checksums used by TCP or UDP.  However,
   partial payload checksum coverage as provided by DCCP or UDP-Lite is
   not supported.

   SCTP has been designed with extensibility in mind.  Each SCTP packet
   starts with a single common header containing the port numbers, a
   verification tag and the CRC32c checksum.  This  A common header
   is followed by a sequence of chunks.  Each chunk consists of a type
   field, flags, a length field and a value.  [RFC4960] defines how a
   receiver processes chunks with an unknown chunk type.  The support of
   extensions can be negotiated during the SCTP handshake.  Currently
   defined extensions include mechanisms for dynamic re-configuration of
   streams [RFC6525] and IP addresses [RFC5061].  Furthermore, the
   extension specified in [RFC3758] introduces the concept of partial
   reliability for user messages.

   SCTP provides a message-oriented service.  Multiple small user
   messages can be bundled into a single SCTP packet to improve
   efficiency.  For example, this bundling may be done by delaying user
   messages at the sender, similar to Nagle's algorithm used by TCP.
   User messages which would result in IP packets larger than the MTU
   will be fragmented at the sender and reassembled at the receiver.
   There is no protocol limit on the user message size.  ICMP-based path  For MTU
   discovery as specified for IPv4 in [RFC1191] and the same mechanism than for IPv6 in
   [RFC1981] TCP can be used
   [RFC1981][RFC4821], as well as packetization layer path MTU discovery as
   specified in [RFC4821] with utilizing probe packets using the with padding chunks
   chunks, as defined in [RFC4820] are supported. [RFC4820].

   [RFC4960] specifies TCP-friendly congestion control to protect the
   network against overload; see Section 3.1 for more. overload.  SCTP also uses sliding window flow control
   to protect receivers against overflow.  Similar to TCP, SCTP also
   supports delaying acknowledgments.  [RFC7053] provides a way for the
   sender of user messages to request the immediate sending of the
   corresponding acknowledgments.

   Each SCTP association has between 1 and 65536 uni-directional streams
   in each direction.  The number of streams can be different in each
   direction.  Every user message is sent on a particular stream.  User
   messages can be sent un- ordered, un-ordered, or ordered upon request by the upper
   layer.  Un-ordered messages can be delivered as soon as they are
   completely received.  Ordered messages sent on the same stream are
   delivered at the receiver in the same order as sent by the sender.
   For user messages not requiring fragmentation, this minimizes head of
   line blocking.

   The base protocol defined in [RFC4960] does not allow interleaving of
   user- messages.  Large messages on one stream can therefore block the
   sending of user messages on other streams.
   [I-D.ietf-tsvwg-sctp-ndata] overcomes this limitation.  This draft
   also specifies multiple algorithms for the sender side selection of
   which streams to send data from, supporting a variety of scheduling
   algorithms including priority based methods.  The stream re-
   configuration extension defined in [RFC6525] allows streams to be
   reset during the lifetime of an association and to increase the
   number of streams, if the number of streams negotiated in the SCTP
   handshake becomes insufficient.

   Each user message sent is either delivered to the receiver or, in
   case of excessive retransmissions, the association is terminated in a
   non-graceful way [RFC4960], similar to TCP behaviour. behavior.  In addition to
   this reliable transfer, the partial reliability extension [RFC3758]
   allows a sender to abandon user messages.  The application can
   specify the policy for abandoning user messages.  Examples of these
   policies defined in [RFC3758] and [RFC7496] are:

   o  Limiting the time a user message is dealt with by the sender.

   o  Limiting the number of retransmissions for each fragment of a user
      message.  If the number of retransmissions is limited to 0, one
      gets a service similar to UDP.

   o  Abandoning messages of lower priority in case of a send buffer
      shortage.

   SCTP supports multi-homing.  Each SCTP endpoint uses a list of IP-
   addresses and a single port number.  These addresses can be any
   mixture of IPv4 and IPv6 addresses.  These addresses are negotiated
   during the handshake and the address re-configuration extension
   specified in [RFC5061] in combination with [RFC4895] can be used to
   change these addresses in an authenticated way during the livetime lifetime of
   an SCTP association.  This allows for transport layer mobility.
   Multiple addresses are used for improved resilience.  If a remote
   address becomes unreachable, the traffic is switched over to a
   reachable one, if one exists.  [I-D.ietf-tsvwg-sctp-failover]
   specifies a quicker failover operation reducing the latency of the
   failover.

   For securing user messages, the use of TLS over SCTP has been
   specified in [RFC3436].  However, this solution does not support all
   services provided by SCTP, such as un-ordered delivery or partial
   reliability.  Therefore, the use of DTLS over SCTP has been specified
   in [RFC6083] to overcome these limitations.  When using DTLS over
   SCTP, the application can use almost all services provided by SCTP.

   [I-D.ietf-tsvwg-natsupp] defines methods for endpoints and
   middleboxes to provide support NAT traversal for SCTP over IPv4.  For legacy
   NAT traversal, [RFC6951] defines the UDP encapsulation of SCTP-
   packets.  Alternatively, SCTP packets can be encapsulated in DTLS
   packets as specified in [I-D.ietf-tsvwg-sctp-dtls-encaps].  The
   latter encapsulation is used within the WebRTC context.

   SCTP has a well-defined API, described in the next subsection.

4.3.2.

3.3.2.  Interface Description

   [RFC4960] defines an abstract API for the base protocol.  This API
   describes the following functions callable by the upper layer of
   SCTP: Initialize, Associate, Send, Receive, Receive Unsent Message,
   Receive Unacknowledged Message, Shutdown, Abort, SetPrimary, Status,
   Change Heartbeat, Request Heartbeat, Get SRTT Report, Set Failure
   Threshold, Set Protocol Parameters, and Destroy.  The following
   notifications are provided by the SCTP stack to the upper layer:
   COMMUNICATION UP, DATA ARRIVE, SHUTDOWN COMPLETE, COMMUNICATION LOST,
   COMMUNICATION ERROR, RESTART, SEND FAILURE, NETWORK STATUS CHANGE.

   An extension to the BSD Sockets API is defined in [RFC6458] and
   covers:

   o  the base protocol defined in [RFC4960].  The API allows control
      over local addresses and port numbers and the primary path.
      Furthermore the application has fine control about parameters like
      retransmission thresholds, the path supervision parameters, the
      delayed acknowledgment timeout, and the fragmentation point.  The
      API provides a mechanism to allow the SCTP stack to notify the
      application about event events if the application has requested them.
      These notifications provide Information information about status changes of
      the association and each of the peer addresses.  In case of send
      failures, including drop of messages sent unreliably, the
      application can also be notified and user messages can be returned
      to the application.  When sending user messages, the stream id, a
      payload protocol identifier, an indication whether ordered
      delivery is requested or not.  These parameters can also be
      provided on message reception.  Additionally a context can be
      provided when sending, which can be use in case of send failures.
      The sending of arbitrary large user messages is supported.

   o  the SCTP Partial Reliability extension defined in [RFC3758] to
      specify for a user message the PR-SCTP policy and the policy
      specific parameter.

   o  the SCTP Authentication extension  Examples of these policies defined in [RFC4895] allowing to
      manage
      [RFC3758] and [RFC7496] are:

      *  Limiting the shared keys, time a user message is dealt with by the HMAC to use, set sender.

      *  Limiting the chunk types which
      are only accepted in an authenticated way, and get number of retransmissions for each fragment of a
         user message.  If the list number of retransmissions is limited to
         0, one gets a service similar to UDP.

      *  Abandoning messages of lower priority in case of a send buffer
         shortage.

   o  the SCTP Authentication extension defined in [RFC4895] allowing to
      manage the shared keys, the HMAC to use, set the chunk types which
      are only accepted in an authenticated way, and get the list of
      chunks which are accepted by the local and remote end point in an
      authenticated way.

   o  the SCTP Dynamic Address Reconfiguration extension defined in
      [RFC5061].  It allows to manually add and delete local addresses
      for SCTP associations and the enabling of automatic address
      addition and deletion.  Furthermore the peer can be given a hint
      for choosing its primary path.

   For the following SCTP protocol extensions the BSD Sockets API
   extension is defined in the document specifying the protocol
   extensions:

   o  the SCTP Stream Reconfiguration extension defined in [RFC6525].
      The API allows to trigger the reset operation for incoming and
      outgoing streams and the whole association.  It provides also a
      way to notify the association about the corresponding events.
      Furthermore the application can increase the number of streams.

   o  the UDP Encapsulation of SCTP packets extension defined in
      [RFC6951].  The API allows the management of the remote UDP
      encapsulation port.

   o  the SCTP SACK-IMMEDIATELY extension defined in [RFC7053].  The API
      allows the sender of a user message to request the receiver to
      send the corresponding acknowledgment immediately.

   o  the additional PR-SCTP policies defined in [RFC7496].  The API
      allows to enable/disable the PR-SCTP extension, choose the PR-SCTP
      policies defined in the document and provide statistical
      information about abandoned messages.

   Future documents describing SCTP protocol extensions are expected to
   describe the corresponding BSD Sockets API extension in a "Socket API
   Considerations" section.

   The SCTP socket API supports two kinds of sockets:

   o  one-to-one style sockets (by using the socket type "SOCK_STREAM").

   o  one-to-many style socket (by using the socket type
      "SOCK_SEQPACKET").

   One-to-one style sockets are similar to TCP sockets, there is a 1:1
   relationship between the sockets and the SCTP associations (except
   for listening sockets).  One-to-many style SCTP sockets are similar
   to unconnected UDP sockets, where there is a 1:n relationship between
   the sockets and the SCTP associations.

   The SCTP stack can provide information to the applications about
   state changes of the individual paths and the association whenever
   they occur.  These events are delivered similar to user messages but
   are specifically marked as notifications.

   New functions have been introduced to support the use of multiple
   local and remote addresses.  Additional SCTP-specific send and
   receive calls have been defined to permit SCTP-specific information
   to be sent without using ancillary data in the form of additional
   cmsgs.  These functions provide support for detecting partial
   delivery of user messages and notifications.

   The SCTP socket API allows a fine-grained control of the protocol
   behaviour
   behavior through an extensive set of socket options.

   The SCTP kernel implementations of FreeBSD, Linux and Solaris follow
   mostly the specified extension to the BSD Sockets API for the base
   protocol and the corresponding supported protocol extensions.

4.3.3.

3.3.3.  Transport Features

   The transport features provided by SCTP are:

   o  unicast.

   o  connection setup  connection-oriented transport with feature negotiation and
      application-to-port
      mapping. mapping,

   o  unicast transport,

   o  port multiplexing. multiplexing,

   o  Uni-or  uni- or bidirectional communication. communication,

   o  message-oriented delivery with durable message framing supporting
      multiple concurrent streams. streams,

   o  fully reliable, partially reliable, or unreliable delivery. delivery (based
      on user specified policy to handle abandoned user messages) with
      drop notification,

   o  ordered and unordered delivery within a stream.

   o  user message fragmentation and reassembly. stream,

   o  support for stream scheduling prioritization. prioritization,

   o  segmentation,
   o  user message bundling. bundling,

   o  flow control using a window-based mechanism. mechanism,

   o  congestion control using methods similar to TCP. TCP,

   o  strong error/misdelivery error detection (CRC32c). (CRC32c),

   o  transport layer multihoming for resilience.

   o  transport layer mobility.

   o resilience to network failure and/or handover.

4.4. and mobility.

3.4.  User Datagram Protocol (UDP)

   The User Datagram Protocol (UDP) [RFC0768] [RFC2460] is an IETF
   standards track transport protocol.  It provides a unidirectional
   datagram protocol that preserves message boundaries.  It provides no
   error correction,congestion correction, congestion control, or flow control.  It can be
   used to send broadcast datagrams (IPv4) or multicast datagrams (IPv4
   and IPv6), in addition to unicast and anycast datagrams.  IETF
   guidance on the use of UDP is provided in {{I-D.ietf-tsvwg- rfc5405bis}}.
   [I-D.ietf-tsvwg-rfc5405bis].  UDP is widely implemented and widely
   used by common applications, including DNS.

4.4.1.

3.4.1.  Protocol Description

   UDP is a connection-less protocol that maintains message boundaries,
   with no connection setup or feature negotiation.  The protocol uses
   independent messages, ordinarily called datagrams.  Each stream of
   messages is independently managed, therefore retransmission does not
   hold back data sent using other logical streams.  It provides
   detection of payload errors and misdelivery of packets to an
   unintended endpoint, either of which result in discard of received
   datagrams, with no indication to the user of the service.

   It is possible to create IPv4 UDP datagrams with no checksum, and
   while this is generally discouraged [RFC1122]
   [I-D.ietf-tsvwg-rfc5405bis], certain special cases permit this use.
   These datagrams rely on the IPv4 header checksum to protect from
   misdelivery to an unintended endpoint.  IPv6 does not by permit UDP
   datagrams with no checksum, although in certain cases this rule may
   be relaxed [RFC6935].  The checksum support considerations for
   omitting the checksum are defined in [RFC6936].

   UDP does not provide reliability and does not provide retransmission.
   This implies messages may be re-ordered, lost, or duplicated in
   transit.  Note that due to the relatively weak form of checksum used
   by UDP, applications that require end to end integrity of data are
   recommended to include a stronger integrity check of their payload
   data.

   Because UDP provides no flow control, a receiving application that is
   unable to run sufficiently fast, or frequently, may miss messages.

   The lack of congestion handling implies UDP traffic may experience
   loss when using an overloaded path, and may cause the loss of
   messages from other protocols (e.g., TCP) when sharing the same
   network path.

   On transmission, UDP encapsulates each datagram into an a single IP packet,
   which may in turn
   packet or several IP packet fragments.  This allows a datagram to be fragmented by IP.
   larger than the effective path MTU.  Fragments are reassembled before
   delivery to the UDP receiver. receiver, making this transparent to the user of
   the transport service.  When the jumbograms are supported, larger
   messages may be sent without performing fragmentation.

   Applications that need to provide fragmentation or that have other
   requirements such as receiver flow control, congestion control,
   PathMTU discovery/PLPMTUD, support for ECN, etc etc. need these to be
   provided by protocols operating over UDP [I-D.ietf-tsvwg-rfc5405bis].

4.4.2.

3.4.2.  Interface Description

   [RFC0768] describes basic requirements for an API for UDP.  Guidance
   on use of common APIs is provided in [I-D.ietf-tsvwg-rfc5405bis].

   A UDP endpoint consists of a tuple of (IP address, port number).
   Demultiplexing  De-
   multiplexing using multiple abstract endpoints (sockets) on the same
   IP address are is supported.  The same socket may be used by a single
   server to interact with multiple clients (note: this behavior differs
   from TCP, which uses a pair of tuples to identify a connection).
   Multiple server instances (processes) that bind to the same socket
   can cooperate to service multiple clients- the clients.  The socket implementation
   arranges to not duplicate the same received unicast message to
   multiple server processes.

   Many operating systems also allow a UDP socket to be "connected",
   i.e., to bind a UDP socket to a specific (remote) UDP endpoint.
   Unlike TCP's connect primitive, for UDP, this is only a local
   operation that serves to simplify the local send/receive functions
   and to filter the traffic for the specified addresses and ports
   [I-D.ietf-tsvwg-rfc5405bis].

4.4.3.

3.4.3.  Transport Features

   The transport features provided by UDP are:

   o  unicast.

   o  unicast, multicast, anycast, or IPv4 broadcast. broadcast transport,

   o  port multiplexing.  A multiplexing (where a receiving port can be configured to
      receive datagrams from multiple senders. senders),

   o  message-oriented delivery. delivery,

   o  Uni-or  uni- or bidirectional communication.  Transmission communication where the transmissions in
      each direction is independent. are independent,

   o  non-reliable delivery. delivery,

   o  non-ordered delivery.  unordered delivery,

   o  error detection: detection (implemented using a segment checksum verifies to verify
      delivery to the correct endpoint and integrity of the data.  This checksum is data;
      optional for IPv4, IPv4 and optional under specific conditions for IPv6
      where all or none of the payload data is protected.

   o  IPv6 jumbograms.

4.5. protected),

3.5.  Lightweight User Datagram Protocol (UDP-Lite)

   The Lightweight User Datagram Protocol (UDP-Lite) [RFC3828] is an
   IETF standards track transport protocol.  It provides a
   unidirectional, datagram protocol that preserves message boundaries.
   IETF guidance on the use of UDP- Lite is provided in
   [I-D.ietf-tsvwg-rfc5405bis].

4.5.1.  Protocol Description

   Like UDP,  A UDP-Lite is service may support IPv4
   broadcast, multicast, anycast and unicast, and IPv6 multicast,
   anycast and unicast.

   Examples of use include a connection-less datagram protocol, 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]).

3.5.1.  Protocol Description

   Like UDP, UDP-Lite is a connection-less datagram protocol, with no
   connection setup or feature negotiation.  It changes the semantics of
   the UDP "payload length" field to that of a "checksum coverage
   length" field, and is identified by a different IP protocol/next-
   header value.  Otherwise, UDP-Lite is semantically identical to UDP.  The "checksum coverage length" field specifies the
   intended checksum coverage, with the remaining unprotected part of
   the payload called the "error-insensitive part".  Applications using
   UDP-Lite therefore cannot make assumptions regarding the correctness
   of the data received in the insensitive part of the UDP-Lite payload.

   Otherwise, UDP-Lite is semantically identical to UDP.  In the same
   way as for UDP, mechanisms for receiver flow control, congestion
   control, PMTU or PLPMTU discovery, support for ECN, etc
   need etc. needs to be
   provided by upper layer protocols [I-D.ietf-tsvwg-rfc5405bis].

   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, and IPv6 multicast, anycast and unicast.

4.5.2.

3.5.2.  Interface Description

   There is no API currently specified in the RFC Series, but guidance
   on use of common APIs is provided in [I-D.ietf-tsvwg-rfc5405bis].

   The interface of UDP-Lite differs from that of UDP by the addition of
   a single (socket) option that communicates a checksum coverage length
   value: at the sender, this specifies the intended checksum coverage,
   with the remaining unprotected part of the payload called the "error-
   insensitive part".
   value.  The checksum coverage may also be made visible to the
   application via the UDP-Lite MIB module [RFC5097].

4.5.3.

3.5.3.  Transport Features

   The transport features provided by UDP-Lite are:

   o  unicast.

   o  unicast, multicast, anycast, or IPv4 broadcast. broadcast transport (as for
      UDP),

   o  port multiplexing (as for UDP). UDP),

   o  message-oriented delivery (as for UDP). UDP),

   o  Uni-or  Uni- or bidirectional communication.  Transmission communication where the transmissions in
      each direction is independent. are independent (as for UDP),

   o  non-reliable delivery (as for UDP). UDP),

   o  non-ordered delivery (as for UDP).

   o  misdelivery detection (the checksum always provides protection
      from misdelivery). UDP),

   o  partial or full integrity protection.  The payload error detection (where the checksum
      coverage field indicates the size of the payload data covered by
      the checksum.

4.6. checksum).

3.6.  Datagram Congestion Control Protocol (DCCP)

   Datagram Congestion Control Protocol (DCCP) [RFC4340] is an IETF
   standards track bidirectional transport protocol that provides
   unicast connections of congestion-controlled messages without
   providing reliability.

   The DCCP Problem Statement describes the goals that DCCP sought to
   address [RFC4336]. [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].

   DCCP offers low overhead, and many characteristics common to UDP, but
   can avoid "re-inventing the wheel" each time a new multimedia
   application emerges.  Specifically it includes core transport
   functions (feature negotiation, path state management, RTT
   calculation, PMTUD,
   etc): This allows etc.): DCCP applications to use a compatible method defining select how they send
   packets and and, where suitable to suitable, choose common algorithms to manage their
   functions.  Examples of suitable applications that can benefit from such
   transport services include interactive applications, streaming media media,
   or on-line games [RFC4336].

4.6.1.

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

   A DCCP protocol is defined by instance can be extended [RFC4340] and tuned using
   additional features.  Some features are sender-side only, requiring
   no negotiation with the receiver; some are receiver-side only; and
   some are explicitly negotiated during connection setup.

   DCCP uses a family Connect packet to initiate a session, and permits each
   endpoint to choose the features it wishes to support.  Simultaneous
   open [RFC5596], as in TCP, can enable interoperability in the
   presence of middleboxes.  The Connect packet includes a Service Code
   [RFC5595] that identifies the application or protocol using DCCP,
   providing middleboxes with information about the intended use of RFCs. a
   connection.

   DCCP service is unicast-only.

   It provides multiplexing to multiple sockets at each endpoint using
   port numbers.  An active DCCP session is identified by its four-tuple
   of local and remote IP addresses and local port and remote port
   numbers.  At connection setup, DCCP also exchanges the service code
   [RFC5595], a mechanism that allows 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 smaller
   than the maximum packet size.  A DCCP interface allows applications
   to request fragmentation for packets larger than PMTU, but not larger
   than the maximum packet size allowed by the current congestion
   control mechanism (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.  DCCP supports reduced checksum coverage, a partial integrity
   payload protection mechanism similar to UDP-Lite.  There is also a
   Data Checksum option that option, which when enabled, contains a strong CRC, to
   enable endpoints to detect application data corruption - similar to
   SCTP. corruption.

   Receiver flow control is supported, which limits the amount of
   unacknowledged data that can be outstanding at a given time.

   A DCCP protocol instance can be extended [RFC4340] and tuned using
   additional features.  Some features are sender-side only, requiring
   no negotiation with the receiver; some are receiver-side only; and
   some are explicitly negotiated during connection setup.

   DCCP service is unicast-only.

   It supports negotiation of the congestion control profile, profile between
   endpoints, to provide
   plug- and-play plug-and-play congestion control mechanisms.
   Examples of specified profiles include "TCP-like" [RFC4341], "TCP-friendly" "TCP-
   friendly" [RFC4342], and "TCP-friendly for small packets" [RFC5622].
   Additional mechanisms are recorded in an IANA registry.

   DCCP uses a Connect packet to initiate a session, and permits half-
   connections that allow each client to choose the features it wishes
   to support.  Simultaneous open [RFC5596], as in TCP, can enable
   interoperability in the presence of middleboxes.  The Connect packet
   includes a Service Code field [RFC5595] designed to allow middleboxes
   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 DCCP is not
   natively supported.  Support for DCCP in NAPT/NATs is defined in
   [RFC4340] and [RFC5595].  Upper layer protocols specified on top of
   DCCP include DTLS [RFC5595], RTP [RFC5672], ICE/SDP [RFC6773].

   A common packet format has allowed tools to evolve that can read and
   interpret DCCP packets (e.g., Wireshark).

4.6.2.

3.6.2.  Interface Description

   Functions expected for a DCCP API characteristics include: - Datagram transmission.  - Notification Open, Close and Management
   of the current maximum packet size.  - Send progress a DCCP connection.  The Open function provides
   feature negotiation, selection of an appropriate CCID for congestion
   control and other parameters associated with the DCCP connection.  A
   function allows an application to send DCCP datagrams, including
   setting the required checksum coverage, and any required options.
   (DCCP permits sending datagrams with a zero-length payload.)  A
   function allows reception of zero-
   length payloads.  - Slow Receiver flow control at data, including indicating if the data
   was used or dropped.  Functions can also make the status of a receiver.  -
   connection visible to an application, including detection of the
   maximum packet size and the ability to detect perform flow control by
   detecting a slow receiver at the sender.

   There is no API currently specified in the RFC Series.

4.6.3.

3.6.3.  Transport Features

   The transport features provided by DCCP are:

   o  unicast transport. transport,

   o  connection setup  connection-oriented communication with feature negotiation and
      application-to-port
      mapping. mapping,

   o  signaling of application class for middlebox support (implemented
      using Service Codes.  Identifies the upper layer service to the endpoint
      and network. Codes),

   o  port multiplexing. multiplexing,

   o  Uni-or  uni-or bidirectional communication. communication,
   o  message-oriented delivery. delivery,

   o  non-reliable delivery.  unreliable delivery with drop notification,

   o  ordered delivery.  unordered delivery,

   o  flow control.  The slow receiver function allows a receiver to control (implemented using the rate of the sender.

   o  drop notification.  Allows a slow receiver to notify which datagrams
      were not delivered to the peer upper layer protocol.

   o  timestamps. function)

   o  partial and full integrity protection payload error detection (with optional strong
      integrity check).

4.7.

3.7.  Internet Control Message Protocol (ICMP)

   The Internet Control Message Protocol (ICMP) [RFC0792] for IPv4 and
   [RFC4433]
   ICMP for IPv6 [RFC4433] are IETF standards track protocols.

   ICMP  It is a
   connection-less unidirectional protocol that delivers individual
   messages, without error correction, congestion control, or flow
   control.  Messages may be sent as unicast, IPv4 broadcast or
   multicast datagrams (IPv4 and IPv6), in addition to anycast
   datagrams.

4.7.1.  Protocol Description

   ICMP is a connection-less unidirectional protocol that delivers
   individual messages.  The protocol uses independent messages,
   ordinarily called datagrams.  Each message is required to carry a
   checksum as an integrity check

   Transport Protocols and to protect from misdelivery to an
   unintended endpoint.

   ICMP messages typically relay diagnostic information from an endpoint
   [RFC1122] or network device [RFC1716] addressed to the sender of a
   flow.  This usually contains the network protocol header of a packet
   that encountered a reported issue.  Some formats of messages can also
   carry other payload data.  Each message carries an integrity check
   calculated in the same way as for UDP, this checksum is not optional.

   The RFC series defines additional IPv6 message formats to support a
   range of uses.  In the case of IPv6 the protocol incorporates
   neighbor discovery [RFC2461] [RFC3971]} (provided by ARP for IPv4)
   and the Multicast Listener Discovery (MLD) [RFC2710] group management
   functions (provided by IGMP for IPv4).

   Reliable transmission can not be assumed.  A receiving application
   that is unable to run sufficiently fast, or frequently, may miss
   messages since there is no flow or congestion control.  In addition
   some network devices rate-limit ICMP messages.

   Transport Protocols and upper layer protocols can use received upper layer protocols can use received ICMP
   messages to help them take appropriate decisions when network or
   endpoint errors are reported.  For example example, to implement, ICMP-based
   Path MTU discovery [RFC1191][RFC1981] or assist in Packetization
   Layer Path MTU Discovery (PMTUD) [RFC4821].  Such reactions to
   received messages need to protects protect from off-path data injection
   [I-D.ietf-tsvwg-rfc5405bis], avoiding to avoid an application receiving
   packets that were created by an unauthorized third party.  An application
   therefore needs to ensure that all messages are appropriately
   validated, by checking the payload of the messages to ensure these
   are received in response to actually transmitted traffic (e.g., a
   reported error condition that corresponds to a UDP datagram or TCP
   segment was actually sent by the application).  This requires context
   [RFC6056], such as local state about communication instances to each
   destination (e.g., in the TCP, DCCP, or SCTP protocols).  This state
   is not always maintained by UDP-based applications
   [I-D.ietf-tsvwg-rfc5405bis].

   Any response to ICMP error messages ought to be robust to temporary
   routing failures (sometimes called "soft errors"), e.g., transient
   ICMP "unreachable" messages ought to not normally cause a
   communication abort [RFC5461] [I-D.ietf-tsvwg-rfc5405bis].

4.7.2.  Interface

3.7.1.  Protocol Description

   ICMP processing is integrated into many connection-oriented
   transports, but like other functions needs a connection-less unidirectional protocol, It delivers
   independent messages, called datagrams.  Each message is required to be provided by
   carry a checksum as an
   upper-layer protocol when using UDP integrity check and UDP-Lite.  On some stacks, a
   bound socket also allows a UDP application to be notified when protect from mis-
   delivery to an unintended endpoint.

   ICMP
   error messages are received for its transmissions
   [I-D.ietf-tsvwg-rfc5405bis].

4.7.3.  Transport Features

   The transport features provided by ICMP are:

   o  unidirectional.

   o  multicast, anycast and IP4 broadcast.

   o  message-oriented delivery.

   o  non-reliable delivery.

   o  non-ordered delivery.

   o  error and misdelivery detection (checksum).

4.8.  Realtime Transport Protocol (RTP)

   RTP provides typically relay diagnostic information from an end-to-end network transport service, suitable for
   applications transmitting real-time data, such as audio, video or
   data, over multicast endpoint
   [RFC1122] or unicast network services, including TCP, UDP,
   UDP-Lite, or DCCP.

4.8.1.  Protocol Description

   The RTP standard [RFC3550] defines device [RFC1716] addressed to the sender of a pair
   flow.  This usually contains the network protocol header of protocols, RTP and a packet
   that encountered a reported issue.  Some formats of messages can also
   carry other payload data.  Each message carries an integrity check
   calculated in the
   Real Time Control Protocol, RTCP.  The transport does same way as for UDP, this checksum is not provide
   connection setup, instead relying on out-of-band techniques or
   associated control protocols to setup, negotiate parameters or tear
   down a session.

   An RTP sender encapsulates audio/video data into RTP packets to
   transport media streams. optional.

   The RFC-series specifies RTP media RFC series defines additional IPv6 message formats
   allow packets to carry a wide range of media, and specifies support a wide
   range of multiplexing, error control and other support mechanisms.

   If a frame uses.  In the case of media data is large, it will be fragmented into several
   RTP packets.  Likewise, several small frames may be bundled into a
   single RTP packet.  RTP may run over a congestion-controlled or non-
   congestion-controlled transport protocol.

   An RTP receiver collects RTP packets from network, validates them IPv6 the protocol incorporates
   neighbor discovery [RFC2461] [RFC3971]} (provided by ARP for
   correctness, IPv4)
   and sends them to the media decoder input-queue.
   Missing packet detection is performed by the channel decoder.  The
   play-out buffer is ordered Multicast Listener Discovery (MLD) [RFC2710] group management
   functions (provided by time stamp and is used to reorder
   packets.  Damaged frames may IGMP for IPv4).

   Reliable transmission can not be repaired before the media payloads
   are decompressed assumed.  A receiving application
   that is unable to display run sufficiently fast, or store the data.

   RTCP frequently, may miss
   messages since there is a control protocol that works alongside a RTP flow.  Both the
   RTP sender and receiver can send RTCP report packets.  This no flow or congestion control.  In addition
   some network devices rate-limit ICMP messages.

3.7.2.  Interface Description

   ICMP processing is used
   to periodically send control information and report performance.
   Based on received RTCP feedback, an RTP sender can adjust the
   transmission, e.g., perform rate adaptation at the application layer integrated in the case of congestion.

   An RTCP receiver report (RTCP RR) is returned to the sender
   periodically many connection-oriented transports,
   but like other functions needs to report key parameters (e.g, the fraction of packets
   lost in the last reporting interval, the cumulative number of packets
   lost, the highest sequence number received, be provided by an upper-layer
   protocol when using UDP and the inter-arrival
   jitter).  The RTCP RR packets UDP-Lite.

   On some stacks, a bound socket also contain timing information that allows the sender a UDP application to estimate the network round trip time (RTT) be
   notified when ICMP error messages are received for its transmissions
   [I-D.ietf-tsvwg-rfc5405bis].

   Any response to
   the receivers.

   The interval between reports sent from each receiver tends ICMP error messages ought to be on
   the order of a few seconds on average, although this varies with the
   session rate, and sub-second reporting intervals are possible for
   high rate sessions.  The interval is randomized robust to avoid
   synchronization temporary
   routing failures (sometimes called "soft errors"), e.g., transient
   ICMP "unreachable" messages ought to not normally cause a
   communication abort [RFC5461] [I-D.ietf-tsvwg-rfc5405bis].

3.7.3.  Transport Features

   ICMP does not provide any transport service directly to applications.
   Used together with other transport protocols, it provides
   transmission of reports control, error, and measurement data between
   endpoints, or from multiple receivers.

4.8.2.  Interface Description

   There is no standard application programming interface defined for devices along the path to one endpoint.

3.8.  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 RTCP.  Implementations are typically tightly integrated with a
   particular application, and closely follow the principles of
   application level framing and integrated layer processing [ClarkArch]
   in media processing [RFC2736], error recovery and concealment, rate
   adaptation,
   data, over multicast or unicast transport services, including TCP,
   UDP, UDP-Lite, DCCP, TLS and security [RFC7202].  Accordingly, DTLS.

3.8.1.  Protocol Description

   The RTP implementations
   tend to be targeted at particular application domains (e.g., voice-
   over-IP, IPTV, or video conferencing), with standard [RFC3550] defines a feature set optimised
   for that domain, rather than being general purpose implementations pair of protocols, RTP and the protocol.

4.8.3.  Transport Features
   RTP control protocol, RTCP.  The transport features provided by RTP are:

   o  unicast transport.

   o  multicast, anycast does not provide
   connection setup, instead relying on out-of-band techniques or IPv4 broadcast.

   o  port multiplexing.

   o  Uni-or bidirectional communication.

   o  message-oriented delivery.

   o
   associated control protocols for connection setup with feature negotiation
      and application-to-port mapping.

   o  support for media types and other extensions.

   o to setup, negotiate parameters or tear
   down a session.

   An RTP sender encapsulates audio/video data into RTP packets to
   transport media streams.  The RFC-series specifies RTP payload
   formats that allow packets to carry a wide range of reliability functions, including the possibility of
      using packet erasure coding.

   o  segmentation and aggregation.

   o  performance reporting.

   o  drop notification.

   o  timestamps.

4.9.  File Delivery over Unidirectional Transport/Asynchronous Layered
      Coding Reliable Multicast (FLUTE/ALC)

   FLUTE/ALC is an IETF standards track protocol specified in [RFC6726] media, and [RFC5775].  Asynchronous Layer Coding (ALC) provides an
   underlying reliable transport service
   specifies a wide range of multiplexing, error control and FLUTE other
   support mechanisms.

   If a file-oriented
   specialization frame of media data is large, it will be fragmented into several
   RTP packets.  Likewise, several small frames may be bundled into a
   single RTP packet.

   An RTP receiver collects RTP packets from the ALC service (e.g., network, validates them
   for correctness, and sends them to carry associated
   metadata). the media decoder input-queue.
   Missing packet detection is performed by the channel decoder.  The [RFC6726]
   play-out buffer is ordered by time stamp and [RFC5775] protocols is used to reorder
   packets.  Damaged frames may be repaired before the media payloads
   are non-backward-
   compatible updates decompressed to display or store the data.  Some uses of RTP are
   able to exploit the [RFC3926] partial payload protection features offered by
   DCCP and [RFC3450] experimental
   protocols; these experimental protocols are currently largely
   deployed in UDP-Lite.

   RTCP is a control protocol that works alongside an RTP flow.  Both
   the 3GPP Multimedia Broadcast RTP sender and Multicast Services
   (MBMS) (see [MBMS], section 7) receiver will send RTCP report packets.  This is
   used to periodically send control information and similar contexts (e.g., report performance.
   Based on received RTCP feedback, an RTP sender can adjust the
   Japanese ISDB-Tmm standard).

   The FLUTE/ALC protocol has been designed
   transmission, e.g., perform rate adaptation at the application layer
   in the case of congestion.

   An RTCP receiver report (RTCP RR) is returned to support massively
   scalable reliable bulk data dissemination the sender
   periodically to receiver groups report key parameters (e.g, the fraction of
   arbitrary size using IP Multicast over any type packets
   lost in the last reporting interval, the cumulative number of delivery network,
   including unidirectional networks (e.g., broadcast wireless
   channels).  However, packets
   lost, the FLUTE/ALC protocol highest sequence number received, and the inter-arrival
   jitter).  The RTCP RR packets also supports point-to-
   point unicast transmissions.

   FLUTE/ALC bulk data dissemination has been designed for discrete file
   or memory-based "objects".  Transmissions happen either in push mode,
   where content is sent once, or in on-demand mode, where content is
   continuously sent during periods of time contain timing information that can largely exceed
   allows the
   average sender to estimate the network round trip time required (RTT) to download
   the session objects (see [RFC5651],
   section 4.2).

   Although FLUTE/ALC is not well adapted receivers.

   The interval between reports sent from each receiver tends to byte- be on
   the order of a few seconds on average, although this varies with the
   session rate, and message-
   streaming, there is an exception: FLUTE/ALC sub-second reporting intervals are possible for
   high rate sessions.  The interval is used randomized to carry 3GPP
   Dynamic Adaptive Streaming over HTTP (DASH) when scalability is a
   requirement (see [MBMS], section 5.6).  In that case, each Audio/
   Video segment avoid
   synchronization of reports from multiple receivers.

3.8.2.  Interface Description

   There is transmitted as a distinct FLUTE/ALC object in push
   mode.  FLUTE/ALC uses packet erasure coding (also known as
   Application-Level Forward Erasure Correction, no standard application programming interface defined for
   RTP or AL-FEC) in RTCP.  Implementations are typically tightly integrated with a
   proactive way.  The goal of using AL-FEC is both to increase the
   robustness in front of packet erasures
   particular application, and to improve closely follow the efficiency principles of the on-demand service.  FLUTE/ALC transmissions can be governed by
   a congestion control mechanism such as the "Wave
   application level framing and Equation Based
   Rate Control" (WEBRC) [RFC3738] when FLUTE/ALC is used integrated layer processing [ClarkArch]
   in a layered
   transmission manner, with several session channels over which ALC
   packets are sent.  However many FLUTE/ALC deployments target pre-
   provisioned networks media processing [RFC2736], error recovery and involve only Constant Bit Rate (CBR)
   channels with no competing flows, for which a sender-based concealment, rate
   control mechanism is sufficient.  In any case, FLUTE/ALC's
   reliability, delivery mode, congestion control,
   adaptation, and flow/rate control
   mechanisms are distinct components that can be separately controlled security [RFC7202].  Accordingly, RTP implementations
   tend to meet different be targeted at particular application needs.  Section 4.1 of
   [I-D.ietf-tsvwg-rfc5405bis] describes multicast congestion control
   requirements domains (e.g., voice-
   over-IP, IPTV, or video conferencing), with a feature set optimized
   for UDP.

4.9.1.  Protocol Description

   The FLUTE/ALC protocol works on top of UDP (though it could work on
   top that domain, rather than being general purpose implementations of any datagram delivery
   the protocol.

3.8.3.  Transport Features

   The transport features provided by RTP are:

   o  unicast, multicast or IPv4 broadcast (provided by lower layer
      protocol), without requiring
   any connectivity from receivers to the sender.  Purely unidirectional
   networks are therefore supported

   o  port multiplexing (provided by FLUTE/ALC.  This guarantees
   scalability to an unlimited number of receivers in a session, since
   the sender lower layer protocol),

   o  uni- or bidirectional communication (provided by lower layer
      protocol),

   o  message-oriented delivery with support for media types and other
      extensions,

   o  reliable delivery when using erasure coding or unreliable delivery
      with drop notification (if supported by lower layer protocol),

   o  connection setup with feature negotiation (using associated
      protocols) and application-to-port mapping (provided by lower
      layer protocol),

   o  segmentation,

   o  performance metric reporting (using associated protocols).

3.9.  File Delivery over Unidirectional Transport/Asynchronous Layered
      Coding Reliable Multicast (FLUTE/ALC)

   FLUTE/ALC is an IETF standards track protocol specified in [RFC6726]
   and [RFC5775].  It provides object-oriented delivery of discrete data
   or files.  Asynchronous Layer Coding (ALC) provides an underlying
   reliable transport service and FLUTE a file-oriented specialization
   of the ALC service (e.g., to carry associated metadata).  The
   [RFC6726] and [RFC5775] protocols are non-backward-compatible updates
   of the [RFC3926] and [RFC3450] experimental protocols; these
   experimental protocols are currently largely deployed in the 3GPP
   Multimedia Broadcast and Multicast Services (MBMS) (see [MBMS],
   section 7) and similar contexts (e.g., the Japanese ISDB-Tmm
   standard).

   The FLUTE/ALC protocol has been designed to support massively
   scalable reliable bulk data dissemination to receiver groups of
   arbitrary size using IP Multicast over any type of delivery network,
   including unidirectional networks (e.g., broadcast wireless
   channels).  However, the FLUTE/ALC protocol also supports point-to-
   point unicast transmissions.

   FLUTE/ALC bulk data dissemination has been designed for discrete file
   or memory-based "objects".  Although FLUTE/ALC is not well adapted to
   byte- and message-streaming, there is an exception: FLUTE/ALC is used
   to carry 3GPP Dynamic Adaptive Streaming over HTTP (DASH) when
   scalability is a requirement (see [MBMS], section 5.6).

   FLUTE/ALC's reliability, delivery mode, congestion control, and flow/
   rate control mechanisms can be separately controlled to meet
   different application needs.  Section 4.1 of
   [I-D.ietf-tsvwg-rfc5405bis] describes multicast congestion control
   requirements for UDP.

3.9.1.  Protocol Description

   The FLUTE/ALC protocol works on top of UDP (though it could work on
   top of any datagram delivery transport protocol), without requiring
   any connectivity from receivers to the sender.  Purely unidirectional
   networks are therefore supported by FLUTE/ALC.  This guarantees
   scalability to an unlimited number of receivers in a session, since
   the sender behaves exactly the same regardless of the number of
   receivers.

   FLUTE/ALC supports the transfer of bulk objects such as file or in-
   memory content, using either a push or an on-demand mode. in push
   mode, content is sent once to the receivers, while in on-demand mode,
   content is sent continuously during periods of time that can greatly
   exceed the average time required to download the session objects. objects (see
   [RFC5651], section 4.2).

   This enables receivers to join a session asynchronously, at their own
   discretion, receive the content and leave the session.  In this case,
   data content is typically sent continuously, in loops (also known as
   "carousels").  FLUTE/ALC also supports the transfer of an object
   stream, with loose real-time constraints.  This is particularly
   useful to carry 3GPP DASH when scalability is a requirement and
   unicast transmissions over HTTP cannot be used ([MBMS], section 5.6).
   In this case, packets are sent in sequence using push mode.  FLUTE/
   ALC is not well adapted to byte- and message-streaming and other
   solutions could be preferred (e.g., FECFRAME [RFC6363] with real-time
   flows).

   The FLUTE file delivery instantiation of ALC provides a metadata
   delivery service.  Each object of the FLUTE/ALC session is described
   in a dedicated entry of a File Delivery Table (FDT), using an XML
   format (see [RFC6726], section 3.2).  This metadata can include, but
   is not restricted to, a URI attribute (to identify and locate the
   object), a media type attribute, a size attribute, an encoding
   attribute, or a message digest attribute.  Since the set of objects
   sent within a session can be dynamic, with new objects being added
   and old ones removed, several instances of the FDT can be sent and a
   mechanism is provided to identify a new FDT Instance.

   Error detection and verification of the protocol control information
   relies on the on the underlying transport (e.g., UDP checksum).

   To provide robustness against packet loss and improve the efficiency
   of the on-demand mode, FLUTE/ALC relies on packet erasure coding (AL-
   FEC).  AL-FEC encoding is proactive (since there is no feedback and
   therefore no (N)ACK-based retransmission) and ALC packets containing
   repair data are sent along with ALC packets containing source data.
   Several FEC Schemes have been standardized; FLUTE/ALC does not
   mandate the use of any particular one.  Several strategies concerning
   the transmission order of ALC source and repair packets are possible,
   in particular in on-demand mode where it can deeply impact the
   service provided (e.g., to favor the recovery of objects in sequence,
   or at the other extreme, to favor the recovery of all objects in
   parallel), and FLUTE/ALC does not mandate nor recommend the use of
   any particular one.

   A FLUTE/ALC session is composed of one or more channels, associated
   to different destination unicast and/or multicast IP addresses.  ALC
   packets are sent in those channels at a certain transmission rate,
   with a rate that often differs depending on the channel.  FLUTE/ALC
   does not mandate nor recommend any strategy to select which ALC
   packet to send on which channel.  FLUTE/ALC can use a multiple rate
   congestion control building block (e.g., WEBRC) to provide congestion
   control that is feedback free, where receivers adjust their reception
   rates individually by joining and leaving channels associated with
   the session.  To that purpose, the ALC header provides a specific
   field to carry congestion control specific information.  However
   FLUTE/ALC does not mandate the use of a particular congestion control
   mechanism although WEBRC is mandatory to support for the Internet
   ([RFC6726], section 1.1.4).  FLUTE/ALC is often used over a network
   path with pre-provisioned capacity [I-D.ietf-tsvwg-rfc5405bis] where
   there are no flows competing for capacity.  In this case, a sender-
   based rate control mechanism and a single channel is sufficient.

   [RFC6584] provides per-packet authentication, integrity, and anti-
   replay protection in the context of the ALC and NORM protocols.
   Several mechanisms are proposed that seamlessly integrate into these
   protocols using the ALC and NORM header extension mechanisms.

4.9.2.

3.9.2.  Interface Description

   The FLUTE/ALC specification does not describe a specific application
   programming interface (API) to control protocol operation.

   Open source reference implementations of FLUTE/ALC are available at
   http://planete-bcast.inrialpes.fr/ (no longer maintained) and
   http://mad.cs.tut.fi/ (no longer maintained), and these
   implementations specify and document their own APIs.  Commercial
   versions are also available, some derived from the above
   implementations, with their own API.

4.9.3.

3.9.3.  Transport Features

   The transport features provided by FLUTE/ALC are:

   o  unicast

   o  unicast, multicast, anycast or IPv4 broadcast.

   o  per-object dynamic meta-data delivery. broadcast transmission,

   o  push  object-oriented delivery of discrete data or on-demand delivery service. files and associated
      metadata,

   o  fully reliable or partially reliable delivery (of file or in-
      memory objects). objects), using proactive packet erasure coding (AL-FEC) to
      recover from packet erasures,

   o  ordered or unordered delivery (of file or in-memory objects).

   o  per-packet authentication, integrity, and anti-replay services.

   o  proactive objects),

   o  error detection (based on the UDP checksum),

   o  per-packet authentication,

   o  per-packet integrity,

   o  per-packet replay protection,

   o  congestion control for layered flows (e.g., with WEBRC).

3.10.  NACK-Oriented Reliable Multicast (NORM)

   NORM is an IETF standards track protocol specified in [RFC5740].  It
   provides object-oriented delivery of discrete data or files.

   The protocol was designed to support reliable bulk data dissemination
   to receiver groups using IP Multicast but also provides for point-to-
   point unicast operation.  Support for bulk data dissemination
   includes discrete file or computer memory-based "objects" as well as
   byte- and message-streaming.

   NORM can incorporate packet erasure coding as a part of its selective
   ARQ in response to negative acknowledgments from the receiver.  The
   packet erasure coding can also be proactively applied for forward
   protection from packet loss.  NORM transmissions are governed by the
   TCP-friendly congestion control.  The reliability, congestion control
   and flow control mechanisms can be separately controlled to meet
   different application needs.

3.10.1.  Protocol Description

   The NORM protocol is encapsulated in UDP datagrams and thus provides
   multiplexing for multiple sockets on hosts using port numbers.  For
   loosely coordinated IP Multicast, NORM is not strictly connection-
   oriented although per-sender state is maintained by receivers for
   protocol operation.  [RFC5740] does not specify a handshake protocol
   for connection establishment.  Separate session initiation can be
   used to coordinate port numbers.  However, in-band "client-server"
   style connection establishment can be accomplished with the NORM
   congestion control signaling messages using port binding techniques
   like those for TCP client-server connections.

   NORM supports bulk "objects" such as file or in-memory content but
   also can treat a stream of data as a logical bulk object for purposes
   of packet erasure coding (AL-FEC) 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 from packet
      erasures
   message boundaries mid-stream as needed.  Application content is
   carried and improve identified by the on-demand delivery service,

   o  error detection (through UDP).

   o  congestion control for layered flows (e.g., NORM protocol with WEBRC).

4.10.  NACK-Oriented Reliable Multicast (NORM) 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 an IETF standards track protocol specified in [RFC5740]. uses to achieve group
   size scaling with low feedback traffic levels.  The
   protocol was designed 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 support reliable bulk data dissemination measure RTT and collect
   congestion event information (e.g., reflecting a loss event or ECN
   event) from the receiver(s) to
   receiver groups using IP support dynamic adjustment or the
   rate.  The TCP-Friendly Multicast Congestion Control (TFMCC)
   [RFC4654] provides extra features to support multicast, but also provides for point-to-
   point unicast operation.  Support is
   functionally equivalent to TFRC for bulk data dissemination
   includes discrete file or computer memory-based "objects" as well as
   byte- unicast.

   Error detection and message-streaming.  NORM verification of the protocol control information
   relies on the on the underlying transport(e.g., UDP checksum).

   The reliability mechanism is designed to incorporate packet
   erasure coding as an inherent part decoupled from congestion control.  This
   allows invocation of its selective ARQ in response alternative arrangements of transport services.
   For example, to receiver negative acknowledgments.  The support, fixed-rate reliable delivery or unreliable
   delivery (that may optionally be "better than best effort" via packet
   erasure coding can coding) using TFRC.  Alternative congestion control
   techniques may be applied.  For example, TFRC rate control with
   congestion event detection based on ECN.

   While NORM provides NACK-based reliability, it also supports a
   positive acknowledgment (ACK) mechanism that can be proactively applied used for forward protection receiver
   flow control.  This mechanism is decoupled from packet loss.
   NORM transmissions are governed by the TCP-friendly congestion
   control.  NORM's reliability, reliability and
   congestion control, and flow control
   mechanism are distinct components and can be separately controlled to
   meet supporting applications with different application needs.

4.10.1.  Protocol Description

   The NORM protocol
   One example is encapsulated in UDP datagrams and thus provides
   multiplexing use of NORM for multiple sockets on hosts using port numbers.  For
   loosely coordinated IP Multicast, quasi-reliable delivery, where timely
   delivery of newer content may be favored over completely reliable
   delivery of older content within buffering and RTT constraints.

3.10.2.  Interface Description

   The NORM is specification does not strictly connection-
   oriented although per-sender state describe a specific application
   programming interface (API) to control protocol operation.  A freely-
   available, open source reference implementation of NORM is maintained by receivers available
   at https://www.nrl.navy.mil/itd/ncs/products/norm, and a documented
   API is 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 NORM
   congestion control signaling messages using port binding techniques
   like those for TCP client-server connections. implemented.

3.10.3.  Transport Features

   The transport features provided by NORM supports bulk "objects" such as file are:

   o  unicast or in-memory content but
   also can treat multicast transport,

   o  unidirectional communication,

   o  stream-oriented delivery in a single stream or object-oriented
      delivery of in-memory data as a logical or file bulk object for purposes
   of packet erasure coding.  In the case of stream transport, NORM can
   support either byte streams content objects,

   o  fully reliable (NACK-based) 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 partially reliable (using erasure
      coding both proactively and recover
   message boundaries mid-stream as needed.  Application content is
   carried part of ARQ) delivery,

   o  unordered delivery,

   o  error detection (relies on UDP checksum),

   o  segmentation,

   o  data bundling (using Nagle's algorithm),

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

   o  congestion control (also supporting fixed rate reliable or
      unreliable delivery).

3.11.  Transport Layer Security (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 to reliably
   deliver the application content to the receiver(s).  NORM proactively
   measures round- trip timing information to scale ARQ timers
   appropriately Datagram TLS (DTLS) as a
       pseudotransport

   Transport Layer Security (TLS) [RFC5246]} and Datagram TLS (DTLS)
   [RFC6347]} are IETF protocols that provide several security-related
   features to support congestion control.  For multicast
   operation, timer-based feedback suppression applications.  TLS is uses designed to achieve group
   size scaling with low feedback traffic levels.  The feedback
   suppression run on top of a
   reliable streaming transport protocol (usually TCP), while DTLS 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
   designed to measure RTT and collect
   congestion event (e..g, loss event, ECN event, etc) information from run on top of a best-effort datagram protocol (UDP or
   DCCP [RFC5238]).  At the receiver(s) to support dynamic rate control adjustment.  The TCP-
   Friendly Multicast Congestion Control (TFMCC) [RFC4654] used time of writing, the current version of TLS
   is 1.2; which is defined in [RFC5246].  DTLS provides
   some extra features nearly
   identical functionality to support multicast but applications; it is functionally
   equivalent to TFRC defined in the unicast case.

   NORM's reliability mechanism [RFC6347]
   and its current version is decoupled also 1.2.  The TLS protocol evolved from congestion control.
   This allows alternative arrangements of transport services
   the Secure Sockets Layer (SSL) protocols developed in the mid-1990s
   to be
   invoked.  For example, fixed-rate reliable delivery can be supported
   or unreliable (but optionally "better than best effort" via packet
   erasure coding) delivery with rate- control per TFRC can be achieved.
   Additionally, alternative congestion control techniques may be
   applied.  For example, TFRC rate control with congestion event
   detection based support protection of HTTP traffic.

   While older versions of TLS and DTLS are still in use, they provide
   weaker security guarantees.  [RFC7457] outlines important attacks on ECN for links with high packet loss (e.g.,
   wireless) has been implemented
   TLS and demonstrated with NORM.

   While NORM DTLS.  [RFC7525] is NACK-based for reliability transfer, it also supports a
   positive acknowledgment (ACK) mechanism Best Current Practices (BCP) document
   that can be used describes secure configurations for TLS and DTLS to counter
   these attacks.  The recommendations are applicable for receiver
   flow control.  Again, since this mechanism is decoupled from the
   reliability vast
   majority of use cases.

3.11.1.  Protocol Description

   Both TLS and congestion control, applications that have different
   needs in this aspect DTLS provide the same security features and can thus be
   discussed together.  The features they provide are:

   o  Confidentiality

   o  Data integrity

   o  Peer authentication (optional)
   o  Perfect forward secrecy (optional)

   The authentication of the peer entity can be omitted; a common web
   use case is where the protocol differently.  One example server is authenticated and the use of NORM for quasi-reliable delivery where timely delivery
   of newer content may be favored over client is not.
   TLS also provides a completely reliable delivery of
   older content within buffering and RTT constraints.

4.10.2.  Interface Description

   The NORM specification anonymous operation mode in which
   neither peer's identity is authenticated.  It is important to note
   that TLS itself does not describe specify how a specific application
   programming interface (API) to control protocol operation.  A freely-
   available, open source reference implementation peering entity's identity
   should be interpreted.  For example, in the common use case of NORM is available
   at https://www.nrl.navy.mil/itd/ncs/products/norm, and a documented
   API is provided for this implementation.  While a sockets-like API
   authentication by means of an X.509 certificate, it is
   not currently documented, the existing API supports
   application's decision whether the certificate of the necessary
   functions peering entity
   is acceptable for authorization decisions.

   Perfect forward secrecy, if enabled and supported by the selected
   algorithms, ensures that to traffic encrypted and captured during a
   session at time t0 cannot be implemented.

4.10.3.  Transport Features

   The later decrypted at time t1 (t1 > t0),
   even if the long-term secrets of the communicating peers are later
   compromised.

   As DTLS is generally used over an unreliable datagram transport features provided by NORM are:

   o  unicast such
   as UDP, applications will need to tolerate lost, re-ordered, or multicast transport.

   o  stream-oriented delivery
   duplicated datagrams.  Like TLS, DTLS conveys application data in a single stream.

   o  object-oriented delivery of discrete data or file items.

   o  reliable delivery.

   o  unordered unidirectional delivery (of in-memory data or file bulk
      content objects).

   o  error detection (UDP checksum).

   o  segmentation.

   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
   sequence of ARQ).

4.11.  Transport Layer Security (TLS) and Datagram TLS (DTLS) as a
       pseudotransport

   Transport Layer Security (TLS) and Datagram TLS (DTLS) independent records.  However, because records are mapped
   to unreliable datagrams, there are IETF
   protocols that provide several security-related features unique to
   applications.  TLS is designed to run on top of a reliable streaming
   transport protocol (usually TCP), while DTLS is designed
   that are not applicable to run on
   top TLS:

   o  Record replay detection (optional).

   o  Record size negotiation (estimates of a best-effort datagram protocol (UDP or DCCP [RFC5238]).  At
   the time PMTU and record size
      expansion factor).

   o  Coveyance of writing, IP don't fragment (DF) bit settings by application.

   o  An anti-DoS stateless cookie mechanism (optional).

   Generally, DTLS follows the current version of TLS is 1.2; which is
   defined in [RFC5246]. design as closely as possible.  To
   operate over datagrams, DTLS provides nearly identical functionality
   to applications; it is defined in [RFC6347] includes a sequence number and limited
   forms of retransmission and fragmentation for its current version
   is also 1.2. internal
   operations.  The TLS protocol evolved from sequence number may be used for detecting replayed
   information, according to the Secure Sockets Layer
   (SSL) protocols developed windowing procedure described in the mid 90s to support protection of
   HTTP traffic.

   While older versions
   Section 4.1.2.6 of TLS and [RFC6347].  DTLS forbids the use of stream
   ciphers, which are still in use, they provide
   weaker security guarantees.  [RFC7457] outlines important attacks essentially incompatible when operating on
   independent encrypted records.

3.11.2.  Interface Description

   TLS and DTLS.  [RFC7525] is a Best Current Practices (BCP) document
   that describes secure configurations for TLS and DTLS to counter
   these attacks.  The recommendations are applicable for commonly invoked using an API provided by packages such as
   OpenSSL, wolfSSL, or GnuTLS.  Using such APIs entails the vast
   majority
   manipulation of use cases.

4.11.1.  Protocol Description

   Both TLS and DTLS provide several important abstractions, which fall into the same security features
   following categories: long-term keys and can thus algorithms, session state,
   and communications/connections.  There may also be
   discussed together.  The features they provide are:

   o  Confidentiality

   o  Data integrity

   o  Peer authentication (optional)
   o  Perfect forward secrecy (optional)

   The authentication special APIs
   required to deal with time and/or random numbers, both of the peer entity can be omitted; which are
   needed by a common web
   use case is where the server is authenticated variety of encryption algorithms and the client protocols.

   Considerable care is not.
   TLS also provides a completely anonymous operation mode required in which
   neither peer's identity is authenticated.  It is important to note
   that the use of TLS itself does not specify how APIs to ensure
   creation of a peering entity's identity secure application.  The programmer should be interpreted.  For have at
   least a basic understanding of encryption and digital signature
   algorithms and their strengths, public key infrastructure (including
   X.509 certificates and certificate revocation), and the sockets API.
   See [RFC7525] and [RFC7457], as mentioned above.

   As an example, in the common use case of
   authentication by means of an X.509 certificate, it is OpenSSL, the
   application's decision whether primary abstractions are
   the certificate of library itself and method (protocol), session, context, cipher
   and connection.  After initializing the peering entity
   is acceptable for authorization decisions.  Perfect forward secrecy,
   if enabled library and supported by setting the selected algorithms, ensures that
   traffic encrypted
   method, a cipher suite is chosen and captured during used to configure a session at time t0 cannot context
   object.  Session objects may then be
   later decrypted at time t1 (t1 > t0), even if minted according to the long-term secrets
   of
   parameters present in a context object and associated with individual
   connections.  Depending on how precisely the communicating peers are later compromised.

   As DTLS is generally used over an unreliable datagram transport such
   as UDP, applications will need programmer wishes to tolerate lost, re-ordered,
   select different algorithmic or
   duplicated datagrams.  Like TLS, DTLS conveys application data in a
   sequence protocol options, various levels of independent records.  However, because records are mapped
   to unreliable datagrams, there are several features unique to
   details may be required.

3.11.3.  Transport Features

   Both TLS and DTLS
   that are not applicable to TLS: employ a layered architecture.  The lower layer is
   commonly called the record protocol.  It is responsible for:

   o  Record replay detection (optional).  message fragmentation,

   o  Record size negotiation (estimates of PMTU  authentication and record size
      expansion factor). integrity via message authentication codes
      (MAC),

   o  Coveyance of IP don't fragment (DF) bit settings by application.  data encryption,

   o  An anti-DoS stateless cookie mechanism (optional).

   Generally,  scheduling transmission using the underlying transport protocol.

   DTLS follows augments the TLS design as closely as possible.  To
   operate over datagrams, DTLS includes a sequence number record protocol with:

   o  ordering and limited
   forms replay protection, implemented using sequence
      numbers.

   Several protocols are layered on top of retransmission the record protocol.  These
   include the handshake, alert, and change cipher spec protocols.
   There is also the data protocol, used to carry application traffic.
   The handshake protocol is used to establish cryptographic and
   compression parameters when a connection is first set up.  In DTLS,
   this protocol also has a basic fragmentation for its internal
   operations. and retransmission
   capability and a cookie-like mechanism to resist DoS attacks.  (TLS
   compression is not recommended at present).  The sequence number may be alert protocol is
   used for detecting replayed
   information, according to inform the windowing procedure described in
   Section 4.1.2.6 peer of [RFC6347].  DTLS forbids the use various conditions, most of stream
   ciphers, which are essentially incompatible
   terminal for the connection.  The change cipher spec protocol is used
   to synchronize changes in cryptographic parameters for each peer.

   The data protocol, when operating on
   independent encrypted records.

4.11.2.  Interface Description used with an appropriate cipher, provides:

   o  authentication of one end or both ends of a connection,

   o  confidentiality,

   o  cryptographic integrity protection.

   Both TLS and DTLS are unicast-only.

3.12.  Hypertext Transport Protocol (HTTP) over TCP as a pseudotransport

   The Hypertext Transfer Protocol (HTTP) is commonly invoked using an API provided by packages such as
   OpenSSL, wolfSSL, or GnuTLS.  Using such APIs entails application-level
   protocol widely used on the
   manipulation Internet.  It provides object-oriented
   delivery of discrete data or files.  Version 1.1 of several important abstractions, which fall into the
   following categories: long-term keys protocol is
   specified in [RFC7230] [RFC7231] [RFC7232] [RFC7233] [RFC7234]
   [RFC7235], and algorithms, session state, version 2 in [RFC7540].  HTTP is usually transported
   over TCP using port 80 and communications/connections.  There may also 443, although it can be special APIs
   required to deal used with time and/or random numbers, both of which other
   transports.  When used over TCP it inherits its properties.

   Application layer protocols may use HTTP as a substrate with an
   existing method and data formats, or specify new methods and data
   formats.  There are
   needed by various reasons for this practice listed in
   [RFC3205]; these include being a variety well-known and well-understood
   protocol, reusability of encryption algorithms existing servers and protocols.

   Considerable care is required in the client libraries, easy
   use of existing security mechanisms such as HTTP digest
   authentication [RFC2617] and TLS APIs to ensure
   creation [RFC5246], the ability of a secure application.  The programmer should have at
   least a basic understanding HTTP to
   traverse firewalls makes it work over many types of encryption and digital signature
   algorithms and their strengths, public key infrastructure (including
   X.509 certificates and certificate revocation), infrastructure,
   and in cases where an application server often needs to support HTTP
   anyway.

   Depending on application need, the sockets API.
   See [RFC7525] and [RFC7457], use of HTTP as mentioned above.

   As an example, a substrate
   protocol may add complexity and overhead in the case comparison to a special-
   purpose protocol (e.g., HTTP headers, suitability of OpenSSL, the primary abstractions are the library itself and method (protocol), session, context, cipher HTTP
   security model, etc.).  [RFC3205] addresses this issue and connection.  After initializing the library provides
   some guidelines and setting identifies concerns about the
   method, a cipher suite is chosen use of HTTP
   standard port 80 and used to configure a context
   object.  Session objects may then be minted according to 443, the
   parameters present in a context object use of HTTP URL scheme and associated interaction
   with individual
   connections.  Depending on how precisely the programmer wishes to
   select different algorithmic or protocol options, various levels existing firewalls, proxies and NATs.

   Representational State Transfer (REST) [REST] is another example of
   details
   how applications can use HTTP as transport protocol.  REST is an
   architecture style that may be required.

4.11.3.  Transport Features

   Both TLS and DTLS employ used to build applications using HTTP
   as a layered architecture.  The lower layer is
   commonly called the record communication protocol.  It

3.12.1.  Protocol Description

   Hypertext Transfer Protocol (HTTP) is responsible for:

   o  message fragmentation.

   o  authentication and integrity via message authentication codes
      (MAC).

   o  data encryption.

   o  scheduling transmission using the underlying transport a request/response protocol.

   DTLS augments the TLS record protocol with:

   o  ordering  A
   client sends a request containing a request method, URI and replay protection, implemented using sequence
      numbers.

   Several protocols are layered on top of protocol
   version followed by a MIME-like message (see [RFC7231] for the record protocol.  These
   include
   differences between an HTTP object and a MIME message), containing
   information about the handshake, alert, client and change cipher spec protocols.
   There is request modifiers.  The message can
   also the data protocol, used to carry contain a message body carrying application traffic. data.  The handshake protocol is used to establish cryptographic and
   compression parameters when server
   responds with a connection is first set up.  In DTLS,
   this protocol also has status or error code followed by a basic fragmentation and retransmission
   capability MIME-like message
   containing information about the server and information about the
   data.  This may include a cookie-like mechanism to resist DoS attacks.  (TLS
   compression is not recommended at present).  The alert protocol message body.  It is
   used possible to inform the peer of various conditions, most of which are
   terminal specify a
   data format for the connection. message body using MIME media types [RFC2045].
   The change cipher spec protocol is used has additional features, some relevant to synchronize changes pseudo-
   transport are described below.

   Content negotiation, specified in cryptographic parameters for each peer.

   The data protocol, when used with an appropriate cipher, provides:

   o  authentication of one end or both ends [RFC7231], is a mechanism provided
   by HTTP to allow selection of a connection.

   o  confidentiality.

   o  cryptographic integrity protection.

4.12.  Hypertext Transport Protocol (HTTP) over TCP as representation for a pseudotransport requested
   resource.  The 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].  HTTP is usually transported
   over TCP using port 80 client and 443, although it server negotiate acceptable data formats,
   character sets, data encoding (e.g., data can be used with other
   transports.  When used over TCP it inherits its properties. transferred
   compressed using gzip).  HTTP is used can accommodate exchange of messages as
   well as data streaming (using chunked transfer encoding [RFC7230]).
   It is also possible to request 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 part of existing servers and client libraries, easy use a resource using an object
   range request [RFC7233].  The protocol provides powerful cache
   control signaling defined in [RFC7234].

   The persistent connections of
   existing security mechanisms such as HTTP digest authentication
   [RFC2617] 1.1 and TLS [RFC5246], HTTP 2.0 allow multiple
   request- response transactions (streams) during the ability life-time of a
   single HTTP to traverse
   firewalls makes it work over connection.  HTTP 2.0 connections can multiplex many types of infrastructure, and
   request/response pairs in
   cases where parallel on a application server often needs single transport connection.
   This reduces overhead during connection establishment and mitigates
   transport layer slow-start that would have otherwise been incurred
   for each transaction.  Both are important to support HTTP anyway.

   Depending on application need, the reduce latency for
   HTTP's primary use of case.

   HTTP can be combined with security mechanisms, such as a substrate TLS (denoted
   by HTTPS).  This adds protocol may add complexity and overhead in comparison to properties provided by such a special-
   purpose protocol
   mechanism (e.g., HTTP headers, suitability of authentication, encryption).  The TLS Application-
   Layer Protocol Negotiation (ALPN) extension [RFC7301] can be used to
   negotiate the HTTP
   security model, etc.).  [RFC3205] addresses this issue and provides
   some guidelines and concerns about version within the use of HTTP standard port 80
   and 443, TLS handshake, eliminating the use
   latency incurred by additional round-trip exchanges.  Arbitrary
   cookie strings, included as part of HTTP URL scheme and interaction with existing
   firewalls, proxies and NATs.

4.12.1.  Protocol the MIME headers, are often used
   as bearer tokens in HTTP.

3.12.2.  Interface Description

   Hypertext Transfer Protocol (HTTP) is

   There are many HTTP libraries available exposing different APIs.  The
   APIs provide a request/response protocol.  A
   client sends way to specify a request containing by providing a URI, a request method, URI
   request modifiers and protocol
   version followed by optionally a MIME-like message (see [RFC7231] for request body.  For the
   differences between an HTTP object and a MIME message), containing
   information about response,
   callbacks can be registered that will be invoked when the response is
   received.  If TLS is used, the API exposes a registration of
   callbacks for a server that requests client authentication and request modifiers. when
   certificate verification is needed.

   The message World Wide Web Consortium (W3C) has standardized the
   XMLHttpRequest API [XHR].  This API can
   contain a message body carrying application be used for sending HTTP/
   HTTPS requests and receiving server responses.  Besides the XML data as well.
   format, the request and response data format can also be JSON, HTML,
   and plain text.  JavaScript and XMLHttpRequest are ubiquitous
   programming models for websites, and more general applications, where
   native code is less attractive.

3.12.3.  Transport features

   The server
   responds with transport features provided by HTTP, when used as a status pseudo-
   transport, are:

   o  unicast transport (provided by the lower layer protocol, usually
      TCP),

   o  uni- or error code followed bidirectional communication,

   o  transfer of objects in multiple streams with object content type
      negotiation, supporting partial transmission of object ranges,

   o  ordered delivery (provided by the lower layer protocol, usually
      TCP),

   o  fully reliable delivery (provided by 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 lower layer protocol,
      usually TCP),

   o  flow control (provided by the message body using MIME media types [RFC2045].
   Furthermore, lower layer protocol, usually TCP).

   o  congestion control (provided by the protocol has numerous additional features; lower layer protocol, usually
      TCP).

   HTTPS (HTTP over TLS) additionally provides the following features
   relevant to pseudotransport are described below.

   Content negotiation, specified in [RFC7231], is a mechanism
   (as 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 using 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 TLS):

   o  authentication (of one or both ends of a resource using range requests specified in
   [RFC7233].  The protocol provides powerful cache connection),

   o  confidentiality,

   o  integrity protection.

4.  Congestion Control

   Congestion control signalling
   defined in [RFC7234].

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

   It is possible mechanisms are used to combine HTTP with security mechanisms, like TLS
   (denoted by HTTPS), which adds protocol properties provided provide the congestion
   control needed by such a
   mechanism (e.g., authentication, encryption). many Internet transport protocols.  Congestion is
   detected based on sensing of network conditions, whether through
   explicit or implicit feedback.  The TLS Application-
   Layer Protocol Negotiation (ALPN) extension [RFC7301] congestion control mechanisms
   that can be used for
   HTTP version negotiation within the TLS handshake, which eliminates applied by different transport protocols are largely
   orthogonal to the latency choice of addition round-trips.  Arbitrary cookie strings,
   included as part transport protocol.  This section
   provides an overview of the MIME headers, are often used as bearer tokens congestion control mechanisms available
   to the protocols described in HTTP.

   Application layer Section 3.

   Many protocols using HTTP as substrate may use an
   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 separate window to send messages (e.g., [RFC6546]).
   Because HTTP works in many restricted infrastructures, it determine the maximum sending
   rate that is also
   used to tunnel other application-layer protocols.

4.12.2.  Interface Description

   There are many HTTP libraries available exposing different APIs.  The
   APIs provide a way to specify a request allowed by providing a URI, a method,
   request modifiers and optionally a request body.  For the response,
   callbacks can be registered congestion control.  The used congestion
   control mechanism will increase the congestion window if feedback is
   received that indicates that the currently used network path is not
   congested, and will be invoked reduce the window otherwise.  Window-based
   mechanisms often increase their window slowing over multiple RTTs,
   while decreasing strongly when the response is
   received.  If TLS first indication of congestion is
   received.  One example are Additive Increase Multiplicative Decrease
   (AIMD) schemes, where the window is increased by a certain number of
   packets/bytes for each data segment that has been successfully
   transmitted, while the window is used, API expose a registration multiplicatively decrease on the
   occurrence 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 congestion event.  This can be use for lead to a rather
   unstable, oscillating 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 rate, but will resolve a ubiquitous
   programming model congestion
   situation quickly.  TCP New Reno [RFC5681] which is one of the
   initial proposed schemes for websites, and more general applications, where
   native code TCP as well as TCP Cubic
   [I-D.ietf-tcpm-cubic] which is less attractive.

   Representational State Transfer (REST) [REST] the default mechanism for TCP in Linux
   are two examples for window-based AIMD schemes.  This approach is another example how
   also used by DCCP CCID-2 for datagram congestion control.

   Some classes of applications can prefer to use HTTP as a transport protocol.  REST service that
   allows sending at a more stable rate, that is an
   architecture style for building application slowly varied in
   response to congestion.  Rate-based methods offer this type of
   congestion control and have been defined based on the Internet.  It uses
   HTTP loss ratio and
   observed round trip time, such as TFRC [RFC5348] and TFRC-SP
   [RFC4828].  These methods utilize a communication protocol.

4.12.3.  Transport features

   The transport features provided by HTTP, when throughput equation to determine
   the maximum acceptable rate.  Such methods are used as with DCCP CCID-3
   [RFC4342] and CCID-4 [RFC5622], WEBRC [RFC3738], and other
   applications.

   Another class of applications prefer a
   pseudotransport, are:

   o  unicast.

   o  message transport service that yields
   to other (higher-priority) traffic, such as interactive
   transmissions.  While most traffic in the Internet uses loss-based
   congestion control and stream-oriented transfer.

   o  bi- or unidirectional transmission.

   o  ordered delivery.

   o  fully reliable delivery.

   o  object range request.

   o  message content type negotiation.

   o  flow control.

   HTTPS (HTTP over TLS) additionally provides therefore need to fill the following components:

   o  authentication (of one or both ends network buffers (to
   a certain level if Active Queue Management (AQM) is used), low-
   priority congestion control methods often react to changes in delay
   as an earlier indication of congestion.  This approach tends to
   induce less loss than a connection).

   o  confidentiality.

   o  integrity protection. loss-based method but does generally not
   compete well with loss-based traffic across shared bottleneck links.
   Therefore, methods such as LEDBAT [RFC6824], are deployed in the
   Internet for scavenger traffic that aim to only utilize otherwise
   unused capacity.

5.  Transport Service Features

   The tables below summarize some key features to illustrate the range
   of functions provided across the IETF-specified transports.  Figure 1
   considers transports that may be directly layered over the network,
   and Figure 2 considers transports layered over another transport
   service.  Features that are permitted, but not required, are marked
   as "Poss" indicating that it is possible for the transport service to
   offer this feature.

   +---------------+------+------+------+------+------+------+------+
   | Feature       | TCP  | MPTCP| SCTP | UDP  | UDP-L|DCCP  |ICMP  |
   +---------------+------+------+------+------+------+------+------+
   | Datagram      | No   | No   | Yes  | Yes  | Yes  | Yes  | Yes  |
   +---------------+------+------+------+------+------+------+------+
   | Conn. Oriented| Yes  | Yes  | Yes  | No   | No   | Yes  | No   |
   +---------------+------+------+------+------+------+------+------+
   | Reliability   | Yes  | Yes  | Yes  | No   | No   | No   | No   |
   +---------------+------+------+------+------+------+------+------+
   | Partial Rel.  | No   | No   | Pos Poss | N/A  | N/A  | Yes  | N/A  |
   +---------------+------+------+------+------+------+------+------+
   | Corupt. Tol   | No   | No   | No   | No   | Yes  | Yes  | No   |
   +---------------+------+------+------+------+------+------+------+
   | Cong.Control  | Yes  | Yes  | Yes  | No   | No   | Yes  | No   |
   +---------------+------+------+------+------+------+------+------+
   | Endpoint      |  1   | >=1  | >=1  |  1   |  1   |  1   |  1   |
   +---------------+------+------+------+------+------+------+------+
   | Multicast Cap.| No   | No   | No   | Yes  | Yes  | No   | No   |
   +---------------+------+------+------+------+------+------+------+

             Figure 1: Summary comparison: Transport protocols

   +---------------+------+------+------+------+------+
   | Feature       | RTP  | FLUTE| NORM |(D)TLS| HTTP |
   +---------------+------+------+------+------+------+
   | Datagram      | Yes  | No   | Both | Both | No   |
   +---------------+------+------+------+------+------+
   | Conn. Oriented| No   | Yes  | Yes  | Yes  | Yes  |
   +---------------+------+------+------+------+------+
   | Reliability   | No   | Yes  | Pos Poss | Pos Poss | Yes  |
   +---------------+------+------+------+------+------+
   | Partial R     | Pos Poss | No   | Pos Poss | No   | No   |
   +---------------+------+------+------+------+------+
   | Corupt. Tol   | Poss | No   | No   | No   | No   |
   +---------------+------+------+------+------+------+
   | Cong.Control  | Poss | Poss | Poss | N/A  | N/A  |
   +---------------+------+------+------+------+------+
   | Endpoint      | >=1  | >=1  | >=1  |  1   |  1   |
   +---------------+------+------+------+------+------+
   | Multicast Cap.| Yes  | Yes  | Yes  | No   | No   |
   +---------------+------+------+------+------+------+

              Figure 2: Upper layer transports and frameworks

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

   o  Control Functions

      *  Addressing

         +  unicast (TCP, MPTCP, SCTP, UDP, UDP-Lite, DCCP, ICMP, RTP,
            TLS, HTTP)

         +  multicast (UDP, UDP-Lite, DCCP, ICMP, RTP, FLUTE/ALC, NORM) NORM).
            Note that, as TLS and DTLS are unicast-only, there is no
            widely deployed mechanism for supporting the features in the
            Security section below when using multicast addressing.

         +  IPv4 broadcast (UDP, UDP-Lite, DCCP) ICMP)

         +  anycast (UDP, UDP-Lite, DCCP). UDP-Lite).  Connection-oriented protocols such
            as TCP can be and are used with DCCP have also been deployed using anycast
            routing,
            addressing, with the risk that routing changes may cause
            connection failure.

      *  Association type
         +  connection-oriented (TCP, MPTCP, SCTP, DCCP, RTP, NORM, TLS,
            HTTP)

         +  connectionless (UDP, UDP-Lite, FLUTE/ALC)

      *  Multihoming support

         +  multihoming for  resilience (MPTCP, SCTP)

         +  multihoming for and mobility (MPTCP, SCTP)

         +  multihoming for  load-balancing (MPTCP)

         +  address family multiplexing (MPTCP, SCTP)

      *  Application  Middlebox cooperation

         +  application-class signaling to port mapping (TCP, MPTCP, SCTP, UDP, UDP-Lite,
         DCCP, FLUTE/ALC, NORM, TLS, HTTP) middleboxes (DCCP)

         +  with commonly deployed support in NAPT (TCP, MPTCP, UDP,
            TLS, HTTP)  error condition signaling from middleboxes and routers to
            endpoints (ICMP)

      *  Signaling

         +  control information and error signaling (ICMP)

         +  performance metric reporting (RTP)

   o  Delivery

      *  reliability  Reliability

         +  fully reliable delivery (TCP, MPTCP, SCTP, FLUTE/ALC, NORM,
            TLS, HTTP)

         +  partially reliable delivery (SCTP, NORM)

            -  using packet erasure coding (NORM, FLUTE, (FLUTE/ALC, NORM, RTP)

            -  with specified policy for dropped messages (SCTP)

         +  unreliable delivery (SCTP, UDP, UDP-Lite, DCCP) DCCP, RTP)

            -  with drop notification (SCTP, to sender (RTP, SCTP, DCCP)

         +  Integrity protection  error detection

            -  checksum for error detection (TCP, MPTCP, SCTP, UDP, UDP-
               Lite, DCCP, ICMP, FLUTE/ALC, NORM, TLS, HTTP) DTLS)

            -  partial payload checksum protection (UDP-Lite, DCCP) DCCP).
               Some uses of RTP can exploit partial payload checksum
               protection feature to provide a corruption tolerant
               transport service.

            -  checksum optional (UDP) (UDP).  Possible with IPv4 and in
               certain cases with IPv6.

      *  ordering  Ordering

         +  ordered delivery (TCP, MPTCP, SCTP, RTP, FLUTE, TLS, HTTP)

         +  unordered delivery permitted (SCTP, UDP, UDP-Lite, DCCP,
            RTP, NORM)

      *  type/framing  Type/framing

         +  stream-oriented delivery (TCP, MPTCP, SCTP, TLS) TLS, HTTP)

            -  with multiple streams per association (SCTP) (SCTP, HTTP2)

         +  message-oriented delivery (SCTP, UDP, UDP-Lite, DCCP, RTP,
            DTLS)

         +  object-oriented delivery of discrete data or file items files and
            associated metadata (FLUTE/ALC, NORM, HTTP)

            -  with partial delivery of object ranges (HTTP)

      *  Directionality

         +  unidirectional (TCP, SCTP, UDP, UDP-Lite DCCP, RTP, FLUTE/
            ALC, NORM)

         +  bidirectional (TCP, MPTCP, SCTP, HTTP, TLS)

   o  Transmission control

      *  flow control (TCP, MPTCP, SCTP, DCCP, RTP, TLS, HTTP)

      *  congestion control (TCP, MPTCP, SCTP, DCCP, RTP, FLUTE/ALC, NORM,
         TLS, HTTP)
         NORM).  Congestion control can also provided by the transport
         supporting an upper later transport (e.g., RTP,HTTP, TLS).

      *  segmentation (TCP, MPTCP, SCTP, RTP, FLUTE/ALC, NORM, TLS,
         HTTP)

      *  data/message bundling (TCP, MPTCP, SCTP, TLS, HTTP)
      *  stream scheduling prioritization (SCTP) (SCTP, HTTP2)

      *  endpoint multiplexing (MPTCP)

   o  Security (may be used in combination with other transports)

      *  authentication of one end of a connection (TLS) (FLUTE/ALC, TLS,
         DTLS)

      *  authentication of both ends of a connection (TLS) (TLS, DTLS)

      *  confidentiality (TLS) (TLS, DTLS)

      *  cryptographic integrity protection (TLS) (TLS, DTLS)

      *  replay protection (FLUTE/ALC, DTLS)

6.  IANA Considerations

   This document has no considerations for IANA.

7.  Security Considerations

   This document surveys existing transport protocols and protocols
   providing transport-like services.  Confidentiality, integrity, and
   authenticity are among the features provided by those services.  This
   document does not specify any new components features or mechanisms for
   providing these features.  Each RFC listed in referenced by this document
   discusses the security considerations of the specification it
   contains.

8.  Contributors

   In addition to the editors, this document is the work of Brian
   Adamson, Dragana Damjanovic, Kevin Fall, Simone Ferlin-Oliviera,
   Ralph Holz, Olivier Mehani, Karen Nielsen, Colin Perkins, Vincent
   Roca, and Michael Tuexen.

   o  Section 4.2 3.2 on MPTCP was contributed by Simone Ferlin-Oliviera
      (ferlin@simula.no) and Olivier Mehani
      (olivier.mehani@nicta.com.au)

   o  Section 4.4 3.4 on UDP was contributed by Kevin Fall (kfall@kfall.com)

   o  Section 4.3 3.3 on SCTP was contributed by Michael Tuexen (tuexen@fh-
      muenster.de) and Karen Nielsen (karen.nielsen@tieto.com)

   o  Section 4.8 3.8 on RTP contains contributions from Colin Perlins Perkins
      (csp@csperkins.org)

   o  Section 4.9 3.9 on FLUTE/ALC was contributed by Vincent Roca
      (vincent.roca@inria.fr)

   o  Section 4.10 3.10 on NORM was contributed by Brian Adamson
      (brian.adamson@nrl.navy.mil)

   o  Section 4.11 3.11 on TLS and DTLS was contributed by Ralph Holz
      (ralph.holz@nicta.com.au) and Olivier Mehani
      (olivier.mehani@nicta.com.au)

   o  Section 4.12 3.12 on HTTP was contributed by Dragana Damjanovic
      (ddamjanovic@mozilla.com)

9.  Acknowledgments

   Thanks to Joe Touch, Michael Welzl, and the TAPS Working Group for
   the comments, feedback, and discussion.  This work is partially supported by
   the European Commission under grant agreements
   FP7-ICT-318627 agreement No. 318627 mPlane and
   from the Horizon 2020 research and innovation program under grant agreement
   agreements No. 644334 (NEAT); (NEAT) and No. 688421 (MAMI).  This support
   does not imply endorsement.

10.  Informative References

   [RFC0768]  Postel, J., "User Datagram Protocol", STD 6, RFC 768,
              DOI 10.17487/RFC0768, August 1980,
              <http://www.rfc-editor.org/info/rfc768>.

   [RFC0792]  Postel, J., "Internet Control Message Protocol", STD 5,
              RFC 792, DOI 10.17487/RFC0792, September 1981,
              <http://www.rfc-editor.org/info/rfc792>.

   [RFC0793]  Postel, J., "Transmission Control Protocol", STD 7,
              RFC 793, DOI 10.17487/RFC0793, September 1981,
              <http://www.rfc-editor.org/info/rfc793>.

   [RFC0896]  Nagle, J., "Congestion Control in IP/TCP Internetworks",
              RFC 896, DOI 10.17487/RFC0896, January 1984,
              <http://www.rfc-editor.org/info/rfc896>.

   [RFC1122]  Braden, R., Ed., "Requirements for Internet Hosts -
              Communication Layers", STD 3, RFC 1122,
              DOI 10.17487/RFC1122, October 1989,
              <http://www.rfc-editor.org/info/rfc1122>.

   [RFC1191]  Mogul, J. and S. Deering, "Path MTU discovery", RFC 1191,
              DOI 10.17487/RFC1191, November 1990,
              <http://www.rfc-editor.org/info/rfc1191>.

   [RFC1716]  Almquist, P. and F. Kastenholz, "Towards Requirements for
              IP Routers", RFC 1716, DOI 10.17487/RFC1716, November
              1994, <http://www.rfc-editor.org/info/rfc1716>.

   [RFC1981]  McCann, J., Deering, S., and J. Mogul, "Path MTU Discovery
              for IP version 6", RFC 1981, DOI 10.17487/RFC1981, August
              1996, <http://www.rfc-editor.org/info/rfc1981>.

   [RFC2018]  Mathis, M., Mahdavi, J., Floyd, S., and A. Romanow, "TCP
              Selective Acknowledgment Options", RFC 2018,
              DOI 10.17487/RFC2018, October 1996,
              <http://www.rfc-editor.org/info/rfc2018>.

   [RFC2045]  Freed, N. and N. Borenstein, "Multipurpose Internet Mail
              Extensions (MIME) Part One: Format of Internet Message
              Bodies", RFC 2045, DOI 10.17487/RFC2045, November 1996,
              <http://www.rfc-editor.org/info/rfc2045>.

   [RFC2460]  Deering, S. and R. Hinden, "Internet Protocol, Version 6
              (IPv6) Specification", RFC 2460, DOI 10.17487/RFC2460,
              December 1998, <http://www.rfc-editor.org/info/rfc2460>.

   [RFC2461]  Narten, T., Nordmark, E., and W. Simpson, "Neighbor
              Discovery for IP Version 6 (IPv6)", RFC 2461,
              DOI 10.17487/RFC2461, December 1998,
              <http://www.rfc-editor.org/info/rfc2461>.

   [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, DOI 10.17487/RFC2617, June 1999,
              <http://www.rfc-editor.org/info/rfc2617>.

   [RFC2710]  Deering, S., Fenner, W., and B. Haberman, "Multicast
              Listener Discovery (MLD) for IPv6", RFC 2710,
              DOI 10.17487/RFC2710, October 1999,
              <http://www.rfc-editor.org/info/rfc2710>.

   [RFC2736]  Handley, M. and C. Perkins, "Guidelines for Writers of RTP
              Payload Format Specifications", BCP 36, RFC 2736,
              DOI 10.17487/RFC2736, December 1999,
              <http://www.rfc-editor.org/info/rfc2736>.

   [RFC3168]  Ramakrishnan, K., Floyd, S., and D. Black, "The Addition
              of Explicit Congestion Notification (ECN) to IP",
              RFC 3168, DOI 10.17487/RFC3168, September 2001,
              <http://www.rfc-editor.org/info/rfc3168>.

   [RFC3205]  Moore, K., "On the use of HTTP as a Substrate", BCP 56,
              RFC 3205, DOI 10.17487/RFC3205, February 2002,
              <http://www.rfc-editor.org/info/rfc3205>.

   [RFC3260]  Grossman, D., "New Terminology and Clarifications for
              Diffserv", RFC 3260, DOI 10.17487/RFC3260, April 2002,
              <http://www.rfc-editor.org/info/rfc3260>.

   [RFC3436]  Jungmaier, A., Rescorla, E., and M. Tuexen, "Transport
              Layer Security over Stream Control Transmission Protocol",
              RFC 3436, DOI 10.17487/RFC3436, December 2002,
              <http://www.rfc-editor.org/info/rfc3436>.

   [RFC3450]  Luby, M., Gemmell, J., Vicisano, L., Rizzo, L., and J.
              Crowcroft, "Asynchronous Layered Coding (ALC) Protocol
              Instantiation", RFC 3450, DOI 10.17487/RFC3450, December
              2002, <http://www.rfc-editor.org/info/rfc3450>.

   [RFC3452]  Luby, M., Vicisano, L., Gemmell, J., Rizzo, L., Handley,
              M., and J. Crowcroft, "Forward Error Correction (FEC)
              Building Block", RFC 3452, DOI 10.17487/RFC3452, December
              2002, <http://www.rfc-editor.org/info/rfc3452>.

   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
              July 2003, <http://www.rfc-editor.org/info/rfc3550>.

   [RFC3738]  Luby, M. and V. Goyal, "Wave and Equation Based Rate
              Control (WEBRC) Building Block", RFC 3738,
              DOI 10.17487/RFC3738, April 2004,
              <http://www.rfc-editor.org/info/rfc3738>.

   [RFC3758]  Stewart, R., Ramalho, M., Xie, Q., Tuexen, M., and P.
              Conrad, "Stream Control Transmission Protocol (SCTP)
              Partial Reliability Extension", RFC 3758,
              DOI 10.17487/RFC3758, May 2004,
              <http://www.rfc-editor.org/info/rfc3758>.

   [RFC3828]  Larzon, L-A., Degermark, M., Pink, S., Jonsson, L-E., Ed.,
              and G. Fairhurst, Ed., "The Lightweight User Datagram
              Protocol (UDP-Lite)", RFC 3828, DOI 10.17487/RFC3828, July
              2004, <http://www.rfc-editor.org/info/rfc3828>.

   [RFC3926]  Paila, T., Luby, M., Lehtonen, R., Roca, V., and R. Walsh,
              "FLUTE - File Delivery over Unidirectional Transport",
              RFC 3926, DOI 10.17487/RFC3926, October 2004,
              <http://www.rfc-editor.org/info/rfc3926>.

   [RFC3971]  Arkko, J., Ed., Kempf, J., Zill, B., and P. Nikander,
              "SEcure Neighbor Discovery (SEND)", RFC 3971,
              DOI 10.17487/RFC3971, March 2005,
              <http://www.rfc-editor.org/info/rfc3971>.

   [RFC4324]  Royer, D., Babics, G., and S. Mansour, "Calendar Access
              Protocol (CAP)", RFC 4324, DOI 10.17487/RFC4324, December
              2005, <http://www.rfc-editor.org/info/rfc4324>.

   [RFC4336]  Floyd, S., Handley, M., and E. Kohler, "Problem Statement
              for the Datagram Congestion Control Protocol (DCCP)",
              RFC 4336, DOI 10.17487/RFC4336, March 2006,
              <http://www.rfc-editor.org/info/rfc4336>.

   [RFC4340]  Kohler, E., Handley, M., and S. Floyd, "Datagram
              Congestion Control Protocol (DCCP)", RFC 4340,
              DOI 10.17487/RFC4340, March 2006,
              <http://www.rfc-editor.org/info/rfc4340>.

   [RFC4341]  Floyd, S. and E. Kohler, "Profile for Datagram Congestion
              Control Protocol (DCCP) Congestion Control ID 2: TCP-like
              Congestion Control", RFC 4341, DOI 10.17487/RFC4341, March
              2006, <http://www.rfc-editor.org/info/rfc4341>.

   [RFC4342]  Floyd, S., Kohler, E., and J. Padhye, "Profile for
              Datagram Congestion Control Protocol (DCCP) Congestion
              Control ID 3: TCP-Friendly Rate Control (TFRC)", RFC 4342,
              DOI 10.17487/RFC4342, March 2006,
              <http://www.rfc-editor.org/info/rfc4342>.

   [RFC4433]  Kulkarni, M., Patel, A., and K. Leung, "Mobile IPv4
              Dynamic Home Agent (HA) Assignment", RFC 4433,
              DOI 10.17487/RFC4433, March 2006,
              <http://www.rfc-editor.org/info/rfc4433>.

   [RFC4614]  Duke, M., Braden, R., Eddy, W., and E. Blanton, "A Roadmap
              for Transmission Control Protocol (TCP) Specification
              Documents", RFC 4614, DOI 10.17487/RFC4614, September
              2006, <http://www.rfc-editor.org/info/rfc4614>.

   [RFC4654]  Widmer, J. and M. Handley, "TCP-Friendly Multicast
              Congestion Control (TFMCC): Protocol Specification",
              RFC 4654, DOI 10.17487/RFC4654, August 2006,
              <http://www.rfc-editor.org/info/rfc4654>.

   [RFC4820]  Tuexen, M., Stewart, R., and P. Lei, "Padding Chunk and
              Parameter for the Stream Control Transmission Protocol
              (SCTP)", RFC 4820, DOI 10.17487/RFC4820, March 2007,
              <http://www.rfc-editor.org/info/rfc4820>.

   [RFC4821]  Mathis, M. and J. Heffner, "Packetization Layer Path MTU
              Discovery", RFC 4821, DOI 10.17487/RFC4821, March 2007,
              <http://www.rfc-editor.org/info/rfc4821>.

   [RFC4828]  Floyd, S. and E. Kohler, "TCP Friendly Rate Control
              (TFRC): The Small-Packet (SP) Variant", RFC 4828,
              DOI 10.17487/RFC4828, April 2007,
              <http://www.rfc-editor.org/info/rfc4828>.

   [RFC4895]  Tuexen, M., Stewart, R., Lei, P., and E. Rescorla,
              "Authenticated Chunks for the Stream Control Transmission
              Protocol (SCTP)", RFC 4895, DOI 10.17487/RFC4895, August
              2007, <http://www.rfc-editor.org/info/rfc4895>.

   [RFC4960]  Stewart, R., Ed., "Stream Control Transmission Protocol",
              RFC 4960, DOI 10.17487/RFC4960, September 2007,
              <http://www.rfc-editor.org/info/rfc4960>.

   [RFC5061]  Stewart, R., Xie, Q., Tuexen, M., Maruyama, S., and M.
              Kozuka, "Stream Control Transmission Protocol (SCTP)
              Dynamic Address Reconfiguration", RFC 5061,
              DOI 10.17487/RFC5061, September 2007,
              <http://www.rfc-editor.org/info/rfc5061>.

   [RFC5097]  Renker, G. and G. Fairhurst, "MIB for the UDP-Lite
              protocol", RFC 5097, DOI 10.17487/RFC5097, January 2008,
              <http://www.rfc-editor.org/info/rfc5097>.

   [RFC5246]  Dierks, T. and E. Rescorla, "The Transport Layer Security
              (TLS) Protocol Version 1.2", RFC 5246,
              DOI 10.17487/RFC5246, August 2008,
              <http://www.rfc-editor.org/info/rfc5246>.

   [RFC5238]  Phelan, T., "Datagram Transport Layer Security (DTLS) over
              the Datagram Congestion Control Protocol (DCCP)",
              RFC 5238, DOI 10.17487/RFC5238, May 2008,
              <http://www.rfc-editor.org/info/rfc5238>.

   [RFC5348]  Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP
              Friendly Rate Control (TFRC): Protocol Specification",
              RFC 5348, DOI 10.17487/RFC5348, September 2008,
              <http://www.rfc-editor.org/info/rfc5348>.

   [RFC5461]  Gont, F., "TCP's Reaction to Soft Errors", RFC 5461,
              DOI 10.17487/RFC5461, February 2009,
              <http://www.rfc-editor.org/info/rfc5461>.

   [RFC5595]  Fairhurst, G., "The Datagram Congestion Control Protocol
              (DCCP) Service Codes", RFC 5595, DOI 10.17487/RFC5595,
              September 2009, <http://www.rfc-editor.org/info/rfc5595>.

   [RFC5596]  Fairhurst, G., "Datagram Congestion Control Protocol
              (DCCP) Simultaneous-Open Technique to Facilitate NAT/
              Middlebox Traversal", RFC 5596, DOI 10.17487/RFC5596,
              September 2009, <http://www.rfc-editor.org/info/rfc5596>.

   [RFC5622]  Floyd, S. and E. Kohler, "Profile for Datagram Congestion
              Control Protocol (DCCP) Congestion ID 4: TCP-Friendly Rate
              Control for Small Packets (TFRC-SP)", RFC 5622,
              DOI 10.17487/RFC5622, August 2009,
              <http://www.rfc-editor.org/info/rfc5622>.

   [RFC5651]  Luby, M., Watson, M., and L. Vicisano, "Layered Coding
              Transport (LCT) Building Block", RFC 5651,
              DOI 10.17487/RFC5651, October 2009,
              <http://www.rfc-editor.org/info/rfc5651>.

   [RFC5672]  Crocker, D., Ed., "RFC 4871 DomainKeys Identified Mail
              (DKIM) Signatures -- Update", RFC 5672,
              DOI 10.17487/RFC5672, August 2009,
              <http://www.rfc-editor.org/info/rfc5672>.

   [RFC5740]  Adamson, B., Bormann, C., Handley, M., and J. Macker,
              "NACK-Oriented Reliable Multicast (NORM) Transport
              Protocol", RFC 5740, DOI 10.17487/RFC5740, November 2009,
              <http://www.rfc-editor.org/info/rfc5740>.

   [RFC5775]  Luby, M., Watson, M., and L. Vicisano, "Asynchronous
              Layered Coding (ALC) Protocol Instantiation", RFC 5775,
              DOI 10.17487/RFC5775, April 2010,
              <http://www.rfc-editor.org/info/rfc5775>.

   [RFC5681]  Allman, M., Paxson, V., and E. Blanton, "TCP Congestion
              Control", RFC 5681, DOI 10.17487/RFC5681, September 2009,
              <http://www.rfc-editor.org/info/rfc5681>.

   [RFC6056]  Larsen, M. and F. Gont, "Recommendations for Transport-
              Protocol Port Randomization", BCP 156, RFC 6056,
              DOI 10.17487/RFC6056, January 2011,
              <http://www.rfc-editor.org/info/rfc6056>.

   [RFC6083]  Tuexen, M., Seggelmann, R., and E. Rescorla, "Datagram
              Transport Layer Security (DTLS) for Stream Control
              Transmission Protocol (SCTP)", RFC 6083,
              DOI 10.17487/RFC6083, January 2011,
              <http://www.rfc-editor.org/info/rfc6083>.

   [RFC6093]  Gont, F. and A. Yourtchenko, "On the Implementation of the
              TCP Urgent Mechanism", RFC 6093, DOI 10.17487/RFC6093,
              January 2011, <http://www.rfc-editor.org/info/rfc6093>.

   [RFC6525]  Stewart, R., Tuexen, M., and P. Lei, "Stream Control
              Transmission Protocol (SCTP) Stream Reconfiguration",
              RFC 6525, DOI 10.17487/RFC6525, February 2012,
              <http://www.rfc-editor.org/info/rfc6525>.

   [RFC6546]  Trammell, B., "Transport of Real-time Inter-network
              Defense (RID) Messages over HTTP/TLS", RFC 6546,
              DOI 10.17487/RFC6546, April 2012,
              <http://www.rfc-editor.org/info/rfc6546>.

   [RFC6347]  Rescorla, E. and N. Modadugu, "Datagram Transport Layer
              Security Version 1.2", RFC 6347, DOI 10.17487/RFC6347,
              January 2012, <http://www.rfc-editor.org/info/rfc6347>.

   [RFC6356]  Raiciu, C., Handley, M., and D. Wischik, "Coupled
              Congestion Control for Multipath Transport Protocols",
              RFC 6356, DOI 10.17487/RFC6356, October 2011,
              <http://www.rfc-editor.org/info/rfc6356>.

   [RFC6363]  Watson, M., Begen, A., and V. Roca, "Forward Error
              Correction (FEC) Framework", RFC 6363,
              DOI 10.17487/RFC6363, October 2011,
              <http://www.rfc-editor.org/info/rfc6363>.

   [RFC6458]  Stewart, R., Tuexen, M., Poon, K., Lei, P., and V.
              Yasevich, "Sockets API Extensions for the Stream Control
              Transmission Protocol (SCTP)", RFC 6458,
              DOI 10.17487/RFC6458, December 2011,
              <http://www.rfc-editor.org/info/rfc6458>.

   [RFC6584]  Roca, V., "Simple Authentication Schemes for the
              Asynchronous Layered Coding (ALC) and NACK-Oriented
              Reliable Multicast (NORM) Protocols", RFC 6584,
              DOI 10.17487/RFC6584, April 2012,
              <http://www.rfc-editor.org/info/rfc6584>.

   [RFC6726]  Paila, T., Walsh, R., Luby, M., Roca, V., and R. Lehtonen,
              "FLUTE - File Delivery over Unidirectional Transport",
              RFC 6726, DOI 10.17487/RFC6726, November 2012,
              <http://www.rfc-editor.org/info/rfc6726>.

   [RFC6773]  Phelan, T., Fairhurst, G., and C. Perkins, "DCCP-UDP: A
              Datagram Congestion Control Protocol UDP Encapsulation for
              NAT Traversal", RFC 6773, DOI 10.17487/RFC6773, November
              2012, <http://www.rfc-editor.org/info/rfc6773>.

   [RFC6824]  Ford, A., Raiciu, C., Handley, M., and O. Bonaventure,
              "TCP Extensions for Multipath Operation with Multiple
              Addresses", RFC 6824, DOI 10.17487/RFC6824, January 2013,
              <http://www.rfc-editor.org/info/rfc6824>.

   [RFC6897]  Scharf, M. and A. Ford, "Multipath TCP (MPTCP) Application
              Interface Considerations", RFC 6897, DOI 10.17487/RFC6897,
              March 2013, <http://www.rfc-editor.org/info/rfc6897>.

   [RFC6935]  Eubanks, M., Chimento, P., and M. Westerlund, "IPv6 and
              UDP Checksums for Tunneled Packets", RFC 6935,
              DOI 10.17487/RFC6935, April 2013,
              <http://www.rfc-editor.org/info/rfc6935>.

   [RFC6936]  Fairhurst, G. and M. Westerlund, "Applicability Statement
              for the Use of IPv6 UDP Datagrams with Zero Checksums",
              RFC 6936, DOI 10.17487/RFC6936, April 2013,
              <http://www.rfc-editor.org/info/rfc6936>.

   [RFC6951]  Tuexen, M. and R. Stewart, "UDP Encapsulation of Stream
              Control Transmission Protocol (SCTP) Packets for End-Host
              to End-Host Communication", RFC 6951,
              DOI 10.17487/RFC6951, May 2013,
              <http://www.rfc-editor.org/info/rfc6951>.

   [RFC7053]  Tuexen, M., Ruengeler, I., and R. Stewart, "SACK-
              IMMEDIATELY Extension for the Stream Control Transmission
              Protocol", RFC 7053, DOI 10.17487/RFC7053, November 2013,
              <http://www.rfc-editor.org/info/rfc7053>.

   [RFC7202]  Perkins, C. and M. Westerlund, "Securing the RTP
              Framework: Why RTP Does Not Mandate a Single Media
              Security Solution", RFC 7202, DOI 10.17487/RFC7202, April
              2014, <http://www.rfc-editor.org/info/rfc7202>.

   [RFC7230]  Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer
              Protocol (HTTP/1.1): Message Syntax and Routing",
              RFC 7230, DOI 10.17487/RFC7230, June 2014,
              <http://www.rfc-editor.org/info/rfc7230>.

   [RFC7231]  Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer
              Protocol (HTTP/1.1): Semantics and Content", RFC 7231,
              DOI 10.17487/RFC7231, June 2014,
              <http://www.rfc-editor.org/info/rfc7231>.

   [RFC7232]  Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer
              Protocol (HTTP/1.1): Conditional Requests", RFC 7232,
              DOI 10.17487/RFC7232, June 2014,
              <http://www.rfc-editor.org/info/rfc7232>.

   [RFC7233]  Fielding, R., Ed., Lafon, Y., Ed., and J. Reschke, Ed.,
              "Hypertext Transfer Protocol (HTTP/1.1): Range Requests",
              RFC 7233, DOI 10.17487/RFC7233, June 2014,
              <http://www.rfc-editor.org/info/rfc7233>.

   [RFC7234]  Fielding, R., Ed., Nottingham, M., Ed., and J. Reschke,
              Ed., "Hypertext Transfer Protocol (HTTP/1.1): Caching",
              RFC 7234, DOI 10.17487/RFC7234, June 2014,
              <http://www.rfc-editor.org/info/rfc7234>.

   [RFC7235]  Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer
              Protocol (HTTP/1.1): Authentication", RFC 7235,
              DOI 10.17487/RFC7235, June 2014,
              <http://www.rfc-editor.org/info/rfc7235>.

   [RFC7301]  Friedl, S., Popov, A., Langley, A., and E. Stephan,
              "Transport Layer Security (TLS) Application-Layer Protocol
              Negotiation Extension", RFC 7301, DOI 10.17487/RFC7301,
              July 2014, <http://www.rfc-editor.org/info/rfc7301>.

   [RFC7323]  Borman, D., Braden, B., Jacobson, V., and R.
              Scheffenegger, Ed., "TCP Extensions for High Performance",
              RFC 7323, DOI 10.17487/RFC7323, September 2014,
              <http://www.rfc-editor.org/info/rfc7323>.

   [RFC7414]  Duke, M., Braden, R., Eddy, W., Blanton, E., and A.
              Zimmermann, "A Roadmap for Transmission Control Protocol
              (TCP) Specification Documents", RFC 7414,
              DOI 10.17487/RFC7414, February 2015,
              <http://www.rfc-editor.org/info/rfc7414>.

   [RFC7457]  Sheffer, Y., Holz, R., and P. Saint-Andre, "Summarizing
              Known Attacks on Transport Layer Security (TLS) and
              Datagram TLS (DTLS)", RFC 7457, DOI 10.17487/RFC7457,
              February 2015, <http://www.rfc-editor.org/info/rfc7457>.

   [RFC7496]  Tuexen, M., Seggelmann, R., Stewart, R., and S. Loreto,
              "Additional Policies for the Partially Reliable Stream
              Control Transmission Protocol Extension", RFC 7496,
              DOI 10.17487/RFC7496, April 2015,
              <http://www.rfc-editor.org/info/rfc7496>.

   [RFC7525]  Sheffer, Y., Holz, R., and P. Saint-Andre,
              "Recommendations for Secure Use of Transport Layer
              Security (TLS) and Datagram Transport Layer Security
              (DTLS)", BCP 195, RFC 7525, DOI 10.17487/RFC7525, May
              2015, <http://www.rfc-editor.org/info/rfc7525>.

   [RFC7540]  Belshe, M., Peon, R., and M. Thomson, Ed., "Hypertext
              Transfer Protocol Version 2 (HTTP/2)", RFC 7540,
              DOI 10.17487/RFC7540, May 2015,
              <http://www.rfc-editor.org/info/rfc7540>.

   [I-D.ietf-tsvwg-rfc5405bis]
              Eggert, L., Fairhurst, G., and G. Shepherd, "UDP Usage
              Guidelines", draft-ietf-tsvwg-rfc5405bis-07 (work in
              progress), November 2015.

   [I-D.ietf-aqm-ecn-benefits]
              Fairhurst, G. and M. Welzl, "The Benefits of using
              Explicit Congestion Notification (ECN)", draft-ietf-aqm-
              ecn-benefits-07
              ecn-benefits-08 (work in progress), November 2015.

   [I-D.ietf-tsvwg-rfc5405bis]
              Eggert, L., Fairhurst, G., and G. Shepherd, "UDP Usage
              Guidelines", draft-ietf-tsvwg-rfc5405bis-07 (work in
              progress), November 2015.

   [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-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-04 (work in progress), July 2015.

   [I-D.ietf-tsvwg-sctp-failover]
              Nishida, Y., Natarajan, P., Caro, A., Amer, P., and K.
              Nielsen, "SCTP-PF: Quick Failover Algorithm in SCTP",
              draft-ietf-tsvwg-sctp-failover-13
              draft-ietf-tsvwg-sctp-failover-14 (work in progress),
              September
              December 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-08 (work in progress),
              July 2015.

   [I-D.ietf-tcpm-cubic]
              Rhee, I., Xu, L., Ha, S., Zimmermann, A., Eggert, L., and
              R. Scheffenegger, "CUBIC for Fast Long-Distance Networks",
              draft-ietf-tcpm-cubic-00 (work in progress), June 2015.

   [XHR]      van Kesteren, A., Aubourg, J., Song, J., and H. Steen,
              "XMLHttpRequest working draft
              (http://www.w3.org/TR/XMLHttpRequest/)", 2000.

   [REST]     Fielding, R., "Architectural Styles and the Design of
              Network-based Software Architectures, Ph. D. (UC Irvine),
              Chapter 5: Representational State Transfer", 2000.

   [POSIX]    1-2008, IEEE., "IEEE Standard for Information Technology
              -- Portable Operating System Interface (POSIX) Base
              Specifications, Issue 7", n.d..

   [MBMS]     3GPP TSG WS S4, ., "3GPP TS 26.346: Multimedia Broadcast/
              Multicast Service (MBMS); Protocols and codecs, release 13
              (http://www.3gpp.org/DynaReport/26346.htm).", 2015.

   [ClarkArch]
              Clark, D. and D. Tennenhouse, "Architectural
              Considerations for a New Generation of Protocols (Proc.
              ACM SIGCOMM)", 1990.

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