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Versions: (draft-fairhurst-taps-transports) 00 01 02 03 04 05 06 07 08 09 10 11 12 13 14 RFC 8095

Network Working Group                                  G. Fairhurst, Ed.
Internet-Draft                                    University of Aberdeen
Intended status: Informational                          B. Trammell, Ed.
Expires: April 9, 2016                                M. Kuehlewind, Ed.
                                                              ETH Zurich
                                                        October 07, 2015


  Services provided by IETF transport protocols and congestion control
                               mechanisms
                     draft-ietf-taps-transports-07

Abstract

   This document describes services provided by existing IETF protocols
   and congestion control mechanisms.  It is designed to help
   application and network stack programmers and to inform the work of
   the IETF TAPS Working Group.

Status of This Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on April 9, 2016.

Copyright Notice

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

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of



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   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   3
   2.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . .   4
   3.  Existing Transport Protocols  . . . . . . . . . . . . . . . .   5
     3.1.  Transport Control Protocol (TCP)  . . . . . . . . . . . .   5
       3.1.1.  Protocol Description  . . . . . . . . . . . . . . . .   5
       3.1.2.  Interface description . . . . . . . . . . . . . . . .   6
       3.1.3.  Transport Features  . . . . . . . . . . . . . . . . .   7
     3.2.  Multipath TCP (MPTCP) . . . . . . . . . . . . . . . . . .   8
       3.2.1.  Protocol Description  . . . . . . . . . . . . . . . .   8
       3.2.2.  Interface Description . . . . . . . . . . . . . . . .   8
       3.2.3.  Transport features  . . . . . . . . . . . . . . . . .   8
     3.3.  Stream Control Transmission Protocol (SCTP) . . . . . . .   9
       3.3.1.  Protocol Description  . . . . . . . . . . . . . . . .   9
       3.3.2.  Interface Description . . . . . . . . . . . . . . . .  12
       3.3.3.  Transport Features  . . . . . . . . . . . . . . . . .  14
     3.4.  User Datagram Protocol (UDP)  . . . . . . . . . . . . . .  15
       3.4.1.  Protocol Description  . . . . . . . . . . . . . . . .  15
       3.4.2.  Interface Description . . . . . . . . . . . . . . . .  16
       3.4.3.  Transport Features  . . . . . . . . . . . . . . . . .  16
     3.5.  Lightweight User Datagram Protocol (UDP-Lite) . . . . . .  17
       3.5.1.  Protocol Description  . . . . . . . . . . . . . . . .  17
       3.5.2.  Interface Description . . . . . . . . . . . . . . . .  18
       3.5.3.  Transport Features  . . . . . . . . . . . . . . . . .  18
     3.6.  Datagram Congestion Control Protocol (DCCP) . . . . . . .  19
       3.6.1.  Protocol Description  . . . . . . . . . . . . . . . .  19
       3.6.2.  Interface Description . . . . . . . . . . . . . . . .  20
       3.6.3.  Transport Features  . . . . . . . . . . . . . . . . .  21
     3.7.  Lightweight User Datagram Protocol (UDP-Lite) . . . . . .  21
       3.7.1.  Protocol Description  . . . . . . . . . . . . . . . .  21
       3.7.2.  Interface Description . . . . . . . . . . . . . . . .  22
       3.7.3.  Transport Features  . . . . . . . . . . . . . . . . .  22
     3.8.  Internet Control Message Protocol (ICMP)  . . . . . . . .  23
       3.8.1.  Protocol Description  . . . . . . . . . . . . . . . .  23
       3.8.2.  Interface Description . . . . . . . . . . . . . . . .  24
       3.8.3.  Transport Features  . . . . . . . . . . . . . . . . .  24
     3.9.  Realtime Transport Protocol (RTP) . . . . . . . . . . . .  25
       3.9.1.  Protocol Description  . . . . . . . . . . . . . . . .  25
       3.9.2.  Interface Description . . . . . . . . . . . . . . . .  26
       3.9.3.  Transport Features  . . . . . . . . . . . . . . . . .  26
     3.10. File Delivery over Unidirectional Transport/Asynchronous
           Layered Coding Reliable Multicast (FLUTE/ALC) . . . . . .  26
       3.10.1.  Protocol Description . . . . . . . . . . . . . . . .  27
       3.10.2.  Interface Description  . . . . . . . . . . . . . . .  29



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       3.10.3.  Transport Features . . . . . . . . . . . . . . . . .  29
     3.11. NACK-Oriented Reliable Multicast (NORM) . . . . . . . . .  30
       3.11.1.  Protocol Description . . . . . . . . . . . . . . . .  30
       3.11.2.  Interface Description  . . . . . . . . . . . . . . .  31
       3.11.3.  Transport Features . . . . . . . . . . . . . . . . .  32
     3.12. Transport Layer Security (TLS) and Datagram TLS (DTLS) as
           a pseudotransport . . . . . . . . . . . . . . . . . . . .  32
       3.12.1.  Protocol Description . . . . . . . . . . . . . . . .  33
       3.12.2.  Interface Description  . . . . . . . . . . . . . . .  34
       3.12.3.  Transport Features . . . . . . . . . . . . . . . . .  34
     3.13. Hypertext Transport Protocol (HTTP) over TCP as a
           pseudotransport . . . . . . . . . . . . . . . . . . . . .  35
       3.13.1.  Protocol Description . . . . . . . . . . . . . . . .  36
       3.13.2.  Interface Description  . . . . . . . . . . . . . . .  37
       3.13.3.  Transport features . . . . . . . . . . . . . . . . .  37
   4.  Transport Service Features  . . . . . . . . . . . . . . . . .  38
     4.1.  Complete Protocol Feature Matrix  . . . . . . . . . . . .  40
   5.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  42
   6.  Security Considerations . . . . . . . . . . . . . . . . . . .  42
   7.  Contributors  . . . . . . . . . . . . . . . . . . . . . . . .  42
   8.  Acknowledgments . . . . . . . . . . . . . . . . . . . . . . .  43
   9.  Informative References  . . . . . . . . . . . . . . . . . . .  43
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  52

1.  Introduction

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

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




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   LEDBAT).  This extends the set of available Transport Services beyond
   those provided to applications by TCP and UDP.

   [GF: Ledbat is a mechanism, not protocol - hence use the work
   "support" in para below.]

   Transport protocols can also be differentiated by the features of the
   services they provide: for instance, SCTP offers a message-based
   service providing full or partial reliability and allowing to
   minimize the head of line blocking due to the support of unordered
   and unordered message delivery within multiple streams, UDP-Lite and
   DCCP provide partial integrity protection, and LEDBAT can support
   low-priority "scavenger" communication.

2.  Terminology

   The following terms are defined throughout this document, and in
   subsequent documents produced by TAPS describing the composition and
   decomposition of transport services.

   [EDITOR'S NOTE: we may want to add definitions for the different
   kinds of interfaces that are important here.]

   [GF: Interfaces may be covered by Micahel Welzl's companion
   document?]

   Transport Service Feature:  a specific end-to-end feature that a
      transport service provides to its clients.  Examples include
      confidentiality, reliable delivery, ordered delivery, message-
      versus-stream orientation, etc.

   Transport Service:  a set of transport service features, without an
      association to any given framing protocol, which provides a
      complete service to an application.

   Transport Protocol:  an implementation that provides one or more
      different transport services using a specific framing and header
      format on the wire.

   Transport Protocol Component:  an implementation of a transport
      service feature within a protocol.

   Transport Service Instance:  an arrangement of transport protocols
      with a selected set of features and configuration parameters that
      implements a single transport service, e.g. a protocol stack (RTP
      over UDP).





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   Application:  an entity that uses the transport layer for end-to-end
      delivery data across the network (this may also be an upper layer
      protocol or tunnel encapsulation).

3.  Existing Transport Protocols

   This section provides a list of known IETF transport protocol and
   transport protocol frameworks.

3.1.  Transport Control Protocol (TCP)

   TCP is an IETF standards track transport protocol.  [RFC0793]
   introduces TCP as follows: "The Transmission Control Protocol (TCP)
   is intended for use as a highly reliable host-to-host protocol
   between hosts in packet-switched computer communication networks, and
   in interconnected systems of such networks."  Since its introduction,
   TCP has become the default connection-oriented, stream-based
   transport protocol in the Internet.  It is widely implemented by
   endpoints and widely used by common applications.

3.1.1.  Protocol Description

   TCP is a connection-oriented protocol, providing a three way
   handshake to allow a client and server to set up a connection and
   negotiate features, and mechanisms for orderly completion and
   immediate teardown of a connection.  TCP is defined by a family of
   RFCs [RFC4614].

   TCP provides multiplexing to multiple sockets on each host using port
   numbers.]  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.  ICMP-based PathMTU discovery [RFC1191][RFC1981]
   as well as Packetization Layer Path MTU Discovery (PMTUD) [RFC4821]
   are supported.

   Each byte in the stream is identified by a sequence number.  The
   sequence number is used to order segments on receipt, to identify
   segments in acknowledgments, and to detect unacknowledged segments
   for retransmission.  This is the basis of the reliable, ordered
   delivery of data in a TCP stream.  TCP Selective Acknowledgment
   [RFC2018] extends this mechanism by making it possible to identify
   missing segments more precisely, reducing spurious retransmission.




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   Receiver flow control is provided by a sliding window: limiting the
   amount of unacknowledged data that can be outstanding at a given
   time.  The window scale option [RFC7323] allows a receiver to use
   windows greater than 64KB.

   All TCP senders provide Congestion Control [RFC5681]: This uses a
   separate window, where each time congestion is detected, this
   congestion window is reduced.  Most of the used congestion control
   mechanisms use one of three mechanisms to detect congestion: A
   retransmission timer (with exponential back-up), detection of loss
   (interpreted as a congestion signal), or Explicit Congestion
   Notification (ECN) [RFC3168] to provide early signaling (see
   [I-D.ietf-aqm-ecn-benefits]).  In addition, a congestion control
   mechanism may react to changes in delay as an early indication for
   congestion.

   A TCP protocol instance can be extended [RFC4614] and tuned.  Some
   features are sender-side only, requiring no negotiation with the
   receiver; some are receiver-side only, some are explicitly negotiated
   during connection setup.

   By default, TCP segment partitioning uses Nagle's algorithm [RFC0896]
   to buffer data at the sender into large segments, potentially
   incurring sender-side buffering delay; this algorithm can be disabled
   by the sender to transmit more immediately, e.g., to reduce latency
   for interactive sessions.

   TCP provides a push and a urgent function to enable data to be
   directly accessed by the receiver wihout having to wait for in-order
   delivery of the data.  However, [RFC6093] does not recommend the use
   of the urgent flag due to the range of TCP implementations that
   process TCP urgent indications differently.

   A checksum provides an Integrity Check and is mandatory across the
   entire packet.  This check protects from delivery of corrupted data
   and miselivery of packets to the wrong endpoint.  This check is
   relatively weak, 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
   protection (as in DCCP/UDP-Lite).

   TCP only supports unicast connections.

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




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   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 1) reporting soft errors such as reception fo ICMP error
   messages, extensive retransmission or urgent pointer advance, 2)
   providing a possibility to specify the Type-of-Service (TOS) for
   segments, 3) providing a fush call to empty the TCP send queue, and
   4) 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.  However, there is no RFC that documents this
   interface.

3.1.3.  Transport Features

   The transport features provided by TCP are:

   [EDITOR'S NOTE: expand each of these slightly]

   o  unicast transport

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

   o  port multiplexing: each TCP session is uniquely identified by a
      combination of the ports and IP address fields.

   o  Uni-or bidirectional communication

   o  stream-oriented delivery in a single stream

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

   o  error detection: a segment checksum verifies delivery to the
      correct endpoint and integrity of the data and options.

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

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



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

   o  congestion control: a window-based method that uses AIMD to
      control the sending rate and to conservatively choose a rate after
      congestion is detected.

3.2.  Multipath TCP (MPTCP)

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

3.2.1.  Protocol Description

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

3.2.2.  Interface Description

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

   This Basic API describes how an application can

   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.

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



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   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 with load balancing over mutiple connections.

   o  endpoint multiplexing of a single byte stream (higher throughput).

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

   o  resilience to network failure and/or handover.

   [AUTHOR'S NOTE: it is unclear whether MPTCP has to provide data
   bundling.]

3.3.  Stream Control Transmission Protocol (SCTP)

   SCTP is a message-oriented standards track transport protocol.  The
   base protocol is specified in [RFC4960].  It supports multi-homing to
   handle path failures.  It also optionally supports path failover to
   provide resilliance to path failures.  An SCTP association has
   multiple unidirectional streams in each direction and provides in-
   sequence delivery of user messages only within each stream.  This
   allows it to minimize head of line blocking.  SCTP is extensible and
   the currently defined extensions include mechanisms for dynamic re-
   configurations of streams [RFC6525] and IP-addresses [RFC5061].
   Furthermore, the extension specified in [RFC3758] introduces the
   concept of partial reliability for user messages.

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

3.3.1.  Protocol Description

   SCTP is a connection-oriented protocol using a four way handshake to
   establish an SCTP association and a three way message exchange to




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   gracefully shut it down.  It uses the same port number concept as
   DCCP, TCP, UDP, and UDP-Lite, and only supports unicast.

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

   SCTP has been designed with extensibility in mind.  Each SCTP packet
   starts with a single common header containing the port numbers, a
   verification tag and the CRC32c checksum.  This common header is
   followed by a sequence of chunks.  Each chunk consists of a type
   field, flags, a length field and a value.  [RFC4960] defines how a
   receiver processes chunks with an unknown chunk type.  The support of
   extensions can be negotiated during the SCTP handshake.

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

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

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

   The base protocol defined in [RFC4960] does not allow interleaving of
   user-messages, which results in sending a large message on one stream



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   can block the sending of user messages on other streams.
   [I-D.ietf-tsvwg-sctp-ndata] overcomes this limitation.  Furthermore,
   [I-D.ietf-tsvwg-sctp-ndata] specifies multiple algorithms for the
   sender side selection of which streams to send data from supporting a
   variety of scheduling algorithms including priority based 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.  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 for 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 of
   an SCTP association.  This allows for transport layer mobility.
   Multiple addresses are used for improved resilience.  If a remote
   address becomes unreachable, the traffic is switched over to a
   reachable one, if one exists.  Each SCTP end-point supervises
   continuously the reachability of all peer addresses using a heartbeat
   mechanism.

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




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   [I-D.ietf-tsvwg-natsupp] defines methods for endpoints and
   middleboxes to provide support NAT 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.

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 to control
      the 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 acknowledgement timeout, and the fragmentation point.  The
      API provides a mechanism to allow the SCTP stack to notify the
      application about event if the application has requested them.
      These notifications provide Information about status changes of
      the association and each of the peer addresses.  In case of send
      failures that 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 defined in [RFC4895] allowing to
      manage the shared keys, the HMAC to use, set the chunk types which



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      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 acknowledgement 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




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   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 snet 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 through an extensive set of socket options.

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

3.3.3.  Transport Features

   The transport features provided by SCTP are:

   [GF: This needs to be harmonised with the components for TCP]

   o  unicast.

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

   o  port multiplexing.

   o  message-oriented delivery.

   o  fully reliable or partially reliable delivery.

   o  ordered and unordered delivery within a stream.

   o  support for multiple concurrent streams.

   o  support for stream scheduling prioritization.

   o  flow control.

   o  congestion control.



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   o  user message bundling.

   o  user message fragmentation and reassembly.

   o  strong error/misdelivery detection (CRC32c).

   o  transport layer multihoming for resilience.

   o  transport layer 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
   none of the following transport features: error correction,
   congestion control, or flow control.  It can be used to send
   broadcast datagrams (IPv4) or multicast datagrams (IPv4 and IPv6), in
   addition to unicast (and anycast) datagrams.  IETF guidance on the
   use of UDP is provided in[I-D.ietf-tsvwg-rfc5405bis].  UDP is widely
   implemented and widely used by common applications, including DNS.

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 the wrong
   endpoint, either of which result in discard of received datagrams.

   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 its use.
   These datagrams relie on the IPv4 header checksum to protect from
   misdelivery to the wrong 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].  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.

   It does not provide reliability and does not provide retransmission.
   This implies messages may be re-ordered, lost, or duplicated in
   transit.




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   A receiving application that is unable to run sufficiently fast, or
   frequently, may miss messages since there is no flow control.  The
   lack of congestion handling implies UDP traffic may experience loss
   when using an overlaoded path and may cause the loss of messages from
   other protocols (e.g., TCP) when sharing the same network path.

   [GF: This para isn't needed": Messages with payload errors are
   ordinarily detected by an invalid end- to-end checksum and are
   discarded before being delivered to an application.  UDP-Lite (see
   [RFC3828], and below) provides the ability for portions of the
   message contents to be exempt from checksum coverage.]

   On transmission, UDP encapsulates each datagram into an IP packet,
   which may in turn be fragmented by IP and are reassembled before
   delivery to the UDP receiver.

   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 need these to be
   provided by protocols operating over UDP [I-D.ietf-tsvwg-rfc5405bis].

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

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

3.4.3.  Transport Features

   The transport features provided by UDP are:

   o  unicast.



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   o  multicast, anycast, or IPv4 broadcast.

   o  port multiplexing.  A receiving port can be configured to receive
      datagrams from multiple senders.

   o  message-oriented delivery.

   o  unidirectional or bidirectional.  Transmission in each direction
      is independent.

   o  non-reliable delivery.

   o  non-ordered delivery.

   o  IPv6 jumbograms.

   o  error and misdelivery detection (checksum).

   o  optional checksum.  All or none of the payload data is 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].

3.5.1.  Protocol Description

   UDP-Lite is a connection-less datagram protocol, with no connection
   setup or feature negotiation.  The protocol use messages, rather than
   a byte-stream.  Each stream of messages is independently managed,
   therefore retransmission does not hold back data sent using other
   logical streams.

   It provides multiplexing to multiple sockets on each host using port
   numbers, and its operation follows that for UDP.  An active UDP-Lite
   session is identified by its four-tuple of local and remote IP
   addresses and local port and remote port numbers.

   UDP-Lite 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.  Applications using UDP-Lite therefore
   can not make assumptions regarding the correctness of the data
   received in the insensitive part of the UDP-Lite payload.




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   As for UDP, mechanisms for receiver flow control, congestion control,
   PMTU or PLPMTU discovery, support for ECN, etc need to be provided by
   upper layer protocols [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.

3.5.2.  Interface Description

   There is no current API specified in the RFC Series, but guidance on
   use of common APIs is provided in [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".  The checksum coverage may also be made visible to
   the application via the UDP-Lite MIB module [RFC5097].

3.5.3.  Transport Features

   The transport features provided by UDP-Lite are:

   o  unicast.

   o  multicast, anycast, or IPv4 broadcast.

   o  port multiplexing (as for UDP).

   o  message-oriented delivery (as for UDP).

   o  non-reliable delivery (as for UDP).

   o  non-ordered delivery (as for UDP).

   o  error and misdelivery detection (checksum).

   o  partialor full integrity protection.  The checksum coverage field
      indicates the size of the payload data covered by the checksum.





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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].  It is suitable for applications that transfer
   fairly large amounts of data and that can benefit from control over
   the trade off between timeliness and reliability [RFC4336].

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

3.6.1.  Protocol Description

   DCCP is a connection-oriented datagram protocol, providing a three
   way handshake to allow a client and server to set up a connection,
   and mechanisms for orderly completion and immediate teardown of a
   connection.  The protocol is defined by a family of RFCs.

   It provides multiplexing to multiple sockets 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 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 mechanisms similar to UDP-lIte.  There is also a



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   Data Checksum option that when enabled, contains a strong CRC, to
   enable endpoints to detect application data corruption.

   Receiver flow control is supported: limiting the amount of
   unacknowledged data that can be outstanding at a given time.

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

   A DCCP service is unicast.

   DCCP supports negotiation of the congestion control profile, to
   provide Plug and Play congestion control mechanisms.  Examples of
   specified profiles include [RFC4341] [RFC4342] [RFC5662].  All IETF-
   defined methods provide Congestion Control.

   DCCP use a Connect packet to 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 middle
   boxes and endpoints to identify the characteristics required by a
   session.

   A lightweight UDP-based encapsulation (DCCP-UDP) has been defined
   [RFC6773] that permits DCCP to be used over paths where it is not
   natively supported.  Support in NAPT/NATs is defined in [RFC4340] and
   [RFC5595].

   Upper layer protocols specified on top of DCCP include: DTLS
   [RFC5595], RTP [RFC5672], ICE/SDP [RFC6773].

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

3.6.2.  Interface Description

   API characteristics include: - Datagram transmission.  - Notification
   of the current maximum packet size.  - Send and reception of zero-
   length payloads.  - Slow Receiver flow control at a receiver.  -
   Detect a Slow receiver at the sender.

   There is no current API curremntly specified in the RFC Series.






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3.6.3.  Transport Features

   The transport features provided by DCCP are:

   o  unicast.

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

   o  Service Codes.  Identifies the upper layer service to the endpoint
      and network.

   o  port multiplexing.

   o  message-oriented delivery.

   o  non-reliable delivery.

   o  ordered delivery.

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

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

   o  timestamps.

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

3.7.  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].

3.7.1.  Protocol Description

   UDP-Lite is a connection-less datagram protocol, with no connection
   setup or feature negotiation.  The protocol use messages, rather than
   a byte-stream.  Each stream of messages is independently managed,
   therefore retransmission does not hold back data sent using other
   logical streams.





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   It provides multiplexing to multiple sockets on each host using port
   numbers, and its operation follows that for UDP.  An active UDP-Lite
   session is identified by its four-tuple of local and remote IP
   addresses and local port and remote port numbers.

   UDP-Lite 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.  Applications using UDP-Lite therefore
   can not make assumptions regarding the correctness of the data
   received in the insensitive part of the UDP-Lite payload.

   As for UDP, mechanisms for receiver flow control, congestion control,
   PMTU or PLPMTU discovery, support for ECN, etc need to be provided by
   upper layer protocols [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.

3.7.2.  Interface Description

   There is no current API 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".  The checksum coverage may also be made visible to
   the application via the UDP-Lite MIB module [RFC5097].

3.7.3.  Transport Features

   The transport features provided by UDP-Lite are:

   o  unicast

   o  multicast, anycast, or IPv4 broadcast.

   o  port multiplexing (as for UDP).




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   o  message-oriented delivery (as for UDP).

   o  non-reliable delivery(as for UDP).

   o  non-ordered delivery (as for UDP).

   o  partial or full integrity protection.

3.8.  Internet Control Message Protocol (ICMP)

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

   It provides a conection-less unidirectional protocol that delivers
   individual messages.  It provides none of the following transport
   features: error correction, congestion control, or flow control.
   Some messages may be sent as broadcast datagrams (IPv4) or multicast
   datagrams (IPv4 and IPv6), in addition to unicast (and anycast)
   datagrams.

3.8.1.  Protocol Description

   ICMP is a conection-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 and to protect from misdelivery to the
   wrong 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 the reported issue.  Some formats of messages may
   also carry other payload data.  Each message carries an integrity
   check calculated in the same way as UDP.

   The RFC series defines additional IPv6 message formats to support a
   range of uses.  In the case of IPv6 the protocol incorporates
   neighbour 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 ICMP messages
   to help them take appropriate decisions when network or endpoint



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   errors are reported.  For example to implement, ICMP-based PathMTU
   discovery [RFC1191][RFC1981] or assist in Packetization Layer Path
   MTU Discovery (PMTUD) [RFC4821].  Such reactions to received messages
   needs to protects from off-path data injection
   [I-D.ietf-tsvwg-rfc5405bis], avoiding an application receiving
   packets that were created by an unauthorized third party.  An
   application therefore needs to ensure that aLL messaged 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].

3.8.2.  Interface Description

   ICMP processing is integrated into many connection-oriented
   transports, but like other functions needs to be provided by an
   upper-layer protocol when using UDP and UDP-Lite.  On some stacks, a
   bound socket also allows a UDP application to be notified when ICMP
   error messages are received for its transmissions
   [I-D.ietf-tsvwg-rfc5405bis].

3.8.3.  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).






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3.9.  Realtime Transport Protocol (RTP)

   RTP provides an end-to-end network transport service, suitable for
   applications transmitting real-time data, such as audio, video or
   data, over multicast or unicast network services, including TCP, UDP,
   UDP-Lite, or DCCP.

   [EDITOR'S NOTE: Varun Singh signed up as contributor for this
   section.  Given the complexity of RTP, suggest to have an abbreviated
   section here contrasting RTP with other transports, and focusing on
   those features that are RTP-unique.  Gorry Fairhurst contributed this
   stub section]

3.9.1.  Protocol Description

   The RTP standard [RFC3550] defines a pair of protocols, RTP and the
   Real Time Control Protocol, RTCP.  The transport does not provide
   connection setup, but relies on out-of-band techniques or associated
   control protocols to setup, negotiate parameters or tear-down a
   session.

   An RTP sender encapsulates audio/video data into RTP packets to
   transport media streams.  The RFC-series specifies RTP media formats
   allow packets to carry a wide range of media, and specifies a wide
   range of mulriplexing, error control and other support mechanisms.

   If a frame of media data is large, it will be fragment this into
   several RTP packets.  If small, several frames may be bundled into a
   single RTP packet.  RTP may runs over a congestion-controlled or non-
   congestion-controlled transport protocol.

   An RTP receiver collects RTP packets from network, validates them for
   correctness, and sends them to the media decoder input-queue.
   Missing packet detection is performed by the channel decoder.  The
   play-out buffer is ordered by time stamp and is used to reorder
   packets.  Damaged frames may be repaired before the media payloads
   are decompressed to display or store the data.

   RTCP is an associated control protocol that works with RTP.  Both the
   RTP sender and receiver can send RTCP report packets.  This 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
   in the case of congestion.

   An RTCP receiver report (RTCP RR) is returned to the sender
   periodically to report key parameters (e.g, the fraction of packets
   lost in the last reporting interval, the cumulative number of packets



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   lost, the highest sequence number received, and the inter-arrival
   jitter).  The RTCP RR packets also contain timing information that
   allows the sender to estimate the network round trip time (RTT) to
   the receivers.

   The interval between reports sent from each receiver tends to be on
   the order of a few seconds on average, although this varies with the
   session rate, and sub-second reporting intervals are possible for
   high rate sessions.  The interval is randomised to avoid
   synchronization of reports from multiple receivers.

3.9.2.  Interface Description

   [EDITOR'S NOTE: to do]

3.9.3.  Transport Features

   The transport features provided by RTP are:

   o  unicast.

   o  multicast, anycast or IPv4 broadcast.

   o  port multiplexing.

   o  message-oriented delivery.

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

   o  support for media types and other extensions.

   o  segmentation and aggregation.

   o  performance reporting.

   o  drop notification.

   o  timestamps.

3.10.  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],. 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]



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   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".  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 that can largely exceed the
   average time required to download the session objects (see [RFC5651],
   section 4.2).

   Altough 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).  In that case, each Audio/Video segment 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, or AL-FEC) in a proactive way.  The goal of using
   AL-FEC is both to increase the robustness in front of packet erasures
   and to improve the efficiency of the on-demand service.  FLUTE/ALC
   transmissions can be governed by a congestion control mechanism such
   as the "Wave and Equation Based Rate Control" (WEBRC) [RFC3738] when
   FLUTE/ALC is used in a layered transmission manner, with several
   session channels over which ALC packets are sent.  However many
   FLUTE/ALC deployments involve only Constant Bit Rate (CBR) channels
   with no competing flows, for which a sender-based rate control
   mechanism is sufficient.  In any case, FLUTE/ALC's reliability,
   delivery mode, congestion control, and flow/rate control mechanisms
   are distinct components that can be separately controlled to meet
   different application needs.

3.10.1.  Protocol Description

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



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   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.

   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.

   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



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   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 in case of Internet
   ([RFC6726], section 1.1.4).  FLUTE/ALC is often used over a network
   path with pre-provisoned capacity [RFC5404] whete theres 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.

3.10.2.  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.

3.10.3.  Transport Features

   The transport features provided by FLUTE/ALC are:

   o  unicast

   o  multicast, anycast or IPv4 broadcast.

   o  per-object dynamic meta-data delivery.

   o  push delivery or on-demand delivery service.

   o  fully reliable or partially reliable delivery (of file or in-
      memory objects).

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

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




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   o  proactive packet erasure coding (AL-FEC) to recover from packet
      erasures and improve the on-demand delivery service,

   o  error detection (through UDP and lower level checksums).

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

   o  rate control transmission in a given channel.

3.11.  NACK-Oriented Reliable Multicast (NORM)

   NORM is an IETF standards track protocol specified in [RFC5740].  The
   protocol was designed to support reliable bulk data dissemination to
   receiver groups using IP Multicast but also provides for point-to-
   point unicast operation.  Its support for bulk data dissemination
   includes discrete file or computer memory-based "objects" as well as
   byte- and message-streaming.  NORM is designed to incorporate packet
   erasure coding as an inherent part of its selective ARQ in response
   to receiver negative acknowledgements.  The packet erasure coding can
   also be proactively applied for forward protection from packet loss.
   NORM transmissions are governed by the TCP-friendly congestion
   control.  NORM's reliability, congestion control, and flow control
   mechanism are distinct components and can be separately controlled to
   meet different application needs.

3.11.1.  Protocol Description

   [EDITOR'S NOTE: needs to be more clear about the application of FEC
   and packet erasure coding; expand ARQ.]

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

   NORM supports bulk "objects" such as file or in-memory content but
   also can treat a stream of data as a logical bulk object for purposes
   of packet erasure coding.  In the case of stream transport, NORM can
   support either byte streams or message streams where application-
   defined message boundary information is carried in the NORM protocol
   messages.  This allows the receiver(s) to join/re-join and recover
   message boundaries mid-stream as needed.  Application content is



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   carried and identified by the NORM protocol with encoding symbol
   identifiers depending upon the Forward Error Correction (FEC) Scheme
   [RFC3452] configured.  NORM uses NACK-based selective ARQ to reliably
   deliver the application content to the receiver(s).  NORM proactively
   measures round-trip timing information to scale ARQ timers
   appropriately and to support congestion control.  For multicast
   operation, timer-based feedback suppression is uses to achieve group
   size scaling with low feedback traffic levels.  The feedback
   suppression is not applied for unicast operation.

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

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

   While NORM is NACK-based for reliability transfer, it also supports a
   positive acknowledgment (ACK) mechanism that can be used for receiver
   flow control.  Again, since this mechanism is decoupled from the
   reliability and congestion control, applications that have different
   needs in this aspect can use the protocol differently.  One example
   is the use of NORM for quasi-reliable delivery where timely delivery
   of newer content may be favored over completely reliable delivery of
   older content within buffering and RTT constraints.

3.11.2.  Interface Description

   The NORM specification does not describe a specific application
   programming interface (API) to control protocol operation.  A freely-
   available, open source reference implementation of NORM is available
   at https://www.nrl.navy.mil/itd/ncs/products/norm, and a documented
   API is provided for this implementation.  While a sockets-like API is
   not currently documented, the existing API supports the necessary
   functions for that to be implemented.




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3.11.3.  Transport Features

   The transport features provided by NORM are:

   o  unicast or multicast.

   o  stream-oriented delivery 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 of ARQ).

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

   Transport Layer Security (TLS) and Datagram TLS (DTLS) are IETF
   protocols that provide several security-related features to
   applications.  TLS is designed to run on top of a reliable streaming
   transport protocol (usually TCP), while DTLS is designed to run on
   top of a best-effort datagram protocol (UDP or DCCP [RFC5238]).  At
   the time of writing, the current version of TLS is 1.2; it is defined
   in [RFC5246].  DTLS provides nearly identical functionality to
   applications; it is defined in [RFC6347] and its current version is
   also 1.2.  The TLS protocol evolved from the Secure Sockets Layer
   (SSL) protocols developed in the mid 90s to support protection of
   HTTP traffic.

   While older versions of TLS and DTLS are still in use, they provide
   weaker security guarantees.  [RFC7457] outlines important attacks on
   TLS and DTLS.  [RFC7525] is a Best Current Practices (BCP) document
   that describes secure configurations for TLS and DTLS to counter
   these attacks.  The recommendations are applicable for the vast
   majority of use cases.



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   [NOTE: The Logjam authors (weakdh.org) give (inconclusive) evidence
   that one of the recommendations of [RFC7525], namely the use of
   DHE-1024 as a fallback, may not be sufficient in all cases to counter
   an attacker with the resources of a nation-state.  It is unclear at
   this time if the RFC is going to be updated as a result, or whether
   there will be an RFC7525bis.]

3.12.1.  Protocol Description

   Both TLS and DTLS provide the same security features and can thus be
   discussed together.  The features they provide are:

   o  Confidentiality

   o  Data integrity

   o  Peer authentication (optional)

   o  Perfect forward secrecy (optional)

   The authentication of the peer entity can be omitted; a common web
   use case is where the server is authenticated and the client is not.
   TLS also provides a completely anonymous operation mode in which
   neither peer's identity is authenticated.  It is important to note
   that TLS itself does not specify how a peering entity's identity
   should be interpreted.  For example, in the common use case of
   authentication by means of an X.509 certificate, it is the
   application's decision whether the certificate of the peering entity
   is acceptable for authorization decisions.  Perfect forward secrecy,
   if enabled and supported by the selected algorithms, ensures that
   traffic encrypted and captured during a session at time t0 cannot be
   later decrypted at time t1 (t1 > t0), even if the long-term secrets
   of the communicating peers are later compromised.

   As DTLS is generally used over an unreliable datagram transport such
   as UDP, applications will need to tolerate loss, re-ordered, or
   duplicated datagrams.  Like TLS, DTLS conveys application data in a
   sequence of independent records.  However, because records are mapped
   to unreliable datagrams, there are several features unique to DTLS
   that are not applicable to TLS:

   o  Record replay detection (optional).

   o  Record size negotiation (estimates of PMTU and record size
      expansion factor).

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




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

   Generally, DTLS follows the TLS design as closely as possible.  To
   operate over datagrams, DTLS includes a sequence number and limited
   forms of retransmission and fragmentation for its internal
   operations.  The sequence number may be used for detecting replayed
   information, according to the windowing procedure described in
   Section 4.1.2.6 of [RFC6347].  Note also that DTLS forbids the use of
   stream ciphers, which are essentially incompatible when operating on
   independent encrypted records.

3.12.2.  Interface Description

   TLS is commonly invoked using an API provided by packages such as
   OpenSSL, wolfSSL, or GnuTLS.  Using such APIs entails the
   manipulation of several important abstractions, which fall into the
   following categories: long-term keys and algorithms, session state,
   and communications/connections.  There may also be special APIs
   required to deal with time and/or random numbers, both of which are
   needed by a variety of encryption algorithms and protocols.

   Considerable care is required in the use of TLS APIs in order to
   create a secure application.  The programmer should have at least a
   basic understanding of encryption and digital signature algorithms
   and their strengths, public key infrastructure (including X.509
   certificates and certificate revocation), and the sockets API.  See
   [RFC7525] and [RFC7457], as mentioned above.

   As an example, in the case of OpenSSL, the primary abstractions are
   the library itself and method (protocol), session, context, cipher
   and connection.  After initializing the library and setting the
   method, a cipher suite is chosen and used to configure a context
   object.  Session objects may then be minted according to the
   parameters present in a context object and associated with individual
   connections.  Depending on how precisely the programmer wishes to
   select different algorithmic or protocol options, various levels of
   details may be required.

3.12.3.  Transport Features

   Both TLS and DTLS employ a layered architecture.  The lower layer is
   commonly called the record protocol.  It is responsible for:

   o  message fragmentation

   o  authentication and integrity via message authentication codes
      (MAC)




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   o  data encryption

   o  scheduling transmission using the underlying transport protocol

   DTLS augments the TLS record protocol with:

   o  ordering and replay protection, implemented using sequence
      numbers.

   Several protocols are layered on top of the record protocol.  These
   include the handshake, alert, and change cipher spec protocols.
   There is also the data protocol, used to carry application traffic.
   The handshake protocol is used to establish cryptographic and
   compression parameters when a connection is first set up.  In DTLS,
   this protocol also has a basic fragmentation and retransmission
   capability and a cookie-like mechanism to resist DoS attacks.  (TLS
   compression is not recommended at present).  The alert protocol is
   used to inform the peer of various conditions, most of which are
   terminal for the connection.  The change cipher spec protocol is used
   to synchronize changes in cryptographic parameters for each peer.

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

   Hypertext Transfer Protocol (HTTP) is an application-level protocol
   widely used on the Internet.  Version 1.1 of the protocol is
   specified in [RFC7230] [RFC7231] [RFC7232] [RFC7233] [RFC7234]
   [RFC7235], and version 2 in [RFC7540].  Furthermore, HTTP is used as
   a substrate for other application-layer protocols.  There are various
   reasons for this practice listed in [RFC3205]; these include being a
   well-known and well-understood protocol, reusability of existing
   servers and client libraries, easy use of existing security
   mechanisms such as HTTP digest authentication [RFC2617] and TLS
   [RFC5246], the ability of HTTP to traverse firewalls which makes it
   work with a lot of infrastructure, and cases where a application
   server often needs to support HTTP anyway.

   Depending on application's needs, the use of HTTP as a substrate
   protocol may add complexity and overhead in comparison to a special-
   purpose protocol (e.g.  HTTP headers, suitability of the HTTP
   security model etc.).  [RFC3205] address this issues and provides
   some guidelines and concerns about the use of HTTP standard port 80
   and 443, the use of HTTP URL scheme and interaction with existing
   firewalls, proxies and NATs.

   Though not strictly bound to TCP, HTTP is almost exclusively run over
   TCP, and therefore inherits its properties when used in this way.





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3.13.1.  Protocol Description

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

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

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

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

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




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3.13.2.  Interface Description

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

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

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

3.13.3.  Transport features

   The transport features provided by HTTP, when used as a
   pseudotransport, are:

   o  unicast.

   o  message 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 the following components:

   o  authentication (of one or both ends of a connection).




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   o  confidentiality.

   o  integrity protection.

4.  Transport Service Features

   [EDITOR'S NOTE: This section is still work-in-progress.  This list is
   probably not complete and/or too detailed.]

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

   o  Control Functions

      *  Addressing

         +  unicast

         +  multicast, anycast and IPv4 broadcast

         +  use of NAPT-compatible port numbers

      *  Multihoming support

         +  multihoming for resilience

         +  multihoming for mobility

            -  specify handover latency?

         +  multihoming for load-balancing

            -  specify interleaving delay?

      *  Multiplexing

         +  application to port mapping

         +  single vs. multiple streaming

   o  Delivery

      *  reliability

         +  fully reliable delivery

         +  partially reliable delivery



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            -  packet erasure coding

         +  unreliable delivery

            -  drop notification

            -  Integrity protection

               o  checksum for error detection

               o  partial payload checksum protection

               o  checksum optional

      *  ordering

         +  ordered delivery

         +  unordered delivery

            -  unordered delivery of in-memory data

      *  type/framing

         +  stream-oriented delivery

         +  message-oriented delivery

         +  object-oriented delivery of discrete data or file items

            -  object content type negotiation

         +  range-based partical object transmission

         +  file bulk content objects

   o  Transmission control

      *  rate control

         +  timer-based

         +  ACK-based

      *  congestion control

      *  flow control




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      *  segmentation

      *  data/message bundling (Nagle's algorithm)

      *  stream scheduling prioritization

   o  Security

      *  authentication of one end of a connection

      *  authentication of both ends of a connection

      *  confidentiality

      *  cryptographic integrity protection

   A future revision of this document will define transport service
   features based upon this list.

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

4.1.  Complete Protocol Feature Matrix

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

   [EDITOR'S NOTE: The below is a strawman proposal below by Gorry
   Fairhurst for initial discussion]

   The table below summarises protocol mechanisms that have been
   standardised.  It does not make an assessment on whether specific
   implementations are fully compliant to these specifications.
















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   +-----------------+---------+---------+---------+---------+---------+
   | Mechanism       | UDP     | UDP-L   | DCCP    | SCTP    | TCP     |
   +-----------------+---------+---------+---------+---------+---------+
   | Unicast         | Yes     | Yes     | Yes     | Yes     | Yes     |
   |                 |         |         |         |         |         |
   | Mcast/IPv4Bcast | Yes(2)  | Yes     | No      | No      | No      |
   |                 |         |         |         |         |         |
   | Port Mux        | Yes     | Yes     | Yes     | Yes     | Yes     |
   |                 |         |         |         |         |         |
   | Mode            | Dgram   | Dgram   | Dgram   | Dgram   | Stream  |
   |                 |         |         |         |         |         |
   | Connected       | No      | No      | Yes     | Yes     | Yes     |
   |                 |         |         |         |         |         |
   | Data bundling   | No      | No      | No      | Yes     | Yes     |
   |                 |         |         |         |         |         |
   | Feature Nego    | No      | No      | Yes     | Yes     | Yes     |
   |                 |         |         |         |         |         |
   | Options         | No      | No      | Support | Support | Support |
   |                 |         |         |         |         |         |
   | Data priority   | *       | *       | *       | Yes     | No      |
   |                 |         |         |         |         |         |
   | Data bundling   | No      | No      | No      | Yes     | Yes     |
   |                 |         |         |         |         |         |
   | Reliability     | None    | None    | None    | Select  | Full    |
   |                 |         |         |         |         |         |
   | Ordered deliv   | No      | No      | No      | Stream  | Yes     |
   |                 |         |         |         |         |         |
   | Corruption Tol. | No      | Support | Support | No      | No      |
   |                 |         |         |         |         |         |
   | Flow Control    | No      | No      | Support | Yes     | Yes     |
   |                 |         |         |         |         |         |
   | PMTU/PLPMTU     | (1)     | (1)     | Yes     | Yes     | Yes     |
   |                 |         |         |         |         |         |
   | Cong Control    | (1)     | (1)     | Yes     | Yes     | Yes     |
   |                 |         |         |         |         |         |
   | ECN Support     | (1)     | (1)     | Yes     | TBD     | Yes     |
   |                 |         |         |         |         |         |
   | NAT support     | Limited | Limited | Support | TBD     | Support |
   |                 |         |         |         |         |         |
   | Security        | DTLS    | DTLS    | DTLS    | DTLS    | TLS, AO |
   |                 |         |         |         |         |         |
   | UDP encaps      | N/A     | None    | Yes     | Yes     | None    |
   |                 |         |         |         |         |         |
   | RTP support     | Support | Support | Support | ?       | Support |
   +-----------------+---------+---------+---------+---------+---------+

   Note (1): this feature requires support in an upper layer protocol.




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   Note (2): this feature requires support in an upper layer protocol
   when used with IPv6.

5.  IANA Considerations

   This document has no considerations for IANA.

6.  Security Considerations

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

7.  Contributors

   [Editor's Note: turn this into a real contributors section with
   addresses once we figure out how to trick the toolchain into doing
   so]

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

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

   o  Section 3.3 on SCTP was contributed by Michael Tuexen (tuexen@fh-
      muenster.de)

   o  Section 3.10 on FLUTE/ALC was contributed by Vincent Roca
      (vincent.roca@inria.fr)

   o  Section 3.11 on NORM was contributed by Brian Adamson
      (brian.adamson@nrl.navy.mil)

   o  Section 3.12 on TLS and DTLS was contributed by Ralph Holz
      (ralph.holz@nicta.com.au) and Olivier Mehani
      (olivier.mehani@nicta.com.au)

   o  Section 3.13 on HTTP was contributed by Dragana Damjanovic
      (ddamjanovic@mozilla.com)








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8.  Acknowledgments

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

9.  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>.



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   [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>.

   [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>.

   [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>.



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   [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>.







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   [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>.

   [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>.








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   [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>.

   [RFC5404]  Westerlund, M. and I. Johansson, "RTP Payload Format for
              G.719", RFC 5404, DOI 10.17487/RFC5404, January 2009,
              <http://www.rfc-editor.org/info/rfc5404>.

   [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>.

   [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>.

   [RFC5662]  Shepler, S., Ed., Eisler, M., Ed., and D. Noveck, Ed.,
              "Network File System (NFS) Version 4 Minor Version 1
              External Data Representation Standard (XDR) Description",
              RFC 5662, DOI 10.17487/RFC5662, January 2010,
              <http://www.rfc-editor.org/info/rfc5662>.




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   [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>.




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   [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>.

   [RFC6455]  Fette, I. and A. Melnikov, "The WebSocket Protocol", RFC
              6455, DOI 10.17487/RFC6455, December 2011,
              <http://www.rfc-editor.org/info/rfc6455>.

   [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>.







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   [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>.

   [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>.






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   [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>.

   [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-05 (work in
              progress), August 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-06 (work in progress), July 2015.





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   [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-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.

   [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.

Authors' Addresses

   Godred Fairhurst (editor)
   University of Aberdeen
   School of Engineering, Fraser Noble Building
   Aberdeen AB24 3UE

   Email: gorry@erg.abdn.ac.uk










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   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



































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