wdiff draft-ietf-taps-transports-06.txt draft-ietf-taps-transports-07.txt

Network Working Group G. Fairhurst, Ed.
Internet-Draft University of Aberdeen
Intended status: Informational B. Trammell, Ed.
Expires: January 7, April 9, 2016 M. Kuehlewind, Ed.
ETH Zurich
July 06,
October 07, 2015

Services provided by IETF transport protocols and congestion control
mechanisms
draft-ietf-taps-transports-06
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

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Table of Contents

1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4
3. Existing Transport Protocols . . . . . . . . . . . . . . . . 4 5
3.1. Transport Control Protocol (TCP) . . . . . . . . . . . . 4 5
3.1.1. Protocol Description . . . . . . . . . . . . . . . . 5
3.1.2. Interface description . . . . . . . . . . . . . . . . 6
3.1.3. Transport Protocol Components Features . . . . . . . . . . . . 6 . . . . . 7
3.2. Multipath TCP (MPTCP) . . . . . . . . . . . . . . . . . . 7 8
3.2.1. Protocol Description . . . . . . . . . . . . . . . . 8
3.2.2. Interface Description . . . . . . . . . . . . . . . . 8
3.2.3. Transport Protocol Components features . . . . . . . . . . . . . . . . . 8
3.3. Stream Control Transmission Protocol (SCTP) . . . . . . . 9
3.3.1. Protocol Description . . . . . . . . . . . . . . . . 9
3.3.2. Interface Description . . . . . . . . . . . . . . . . 11 12
3.3.3. Transport Protocol Components Features . . . . . . . . . . . . 13 . . . . . 14
3.4. User Datagram Protocol (UDP) . . . . . . . . . . . . . . 13 15
3.4.1. Protocol Description . . . . . . . . . . . . . . . . 14 15
3.4.2. Interface Description . . . . . . . . . . . . . . . . 14 16
3.4.3. Transport Protocol Components Features . . . . . . . . . . . . 15 . . . . . 16
3.5. Lightweight User Datagram Protocol (UDP-Lite) . . . . . . 15 17
3.5.1. Protocol Description . . . . . . . . . . . . . . . . 15 17
3.5.2. Interface Description . . . . . . . . . . . . . . . . 16 18
3.5.3. Transport Protocol Components Features . . . . . . . . . . . . 16 . . . . . 18
3.6. Datagram Congestion Control Protocol (DCCP) . . . . . . . 17 19
3.6.1. Protocol Description . . . . . . . . . . . . . . . . 17 19
3.6.2. Interface Description . . . . . . . . . . . . . . . . 19 20
3.6.3. Transport Protocol Components . . . . . . . . . . Features . . 19
3.7. Realtime Transport Protocol (RTP) . . . . . . . . . . . . 19
3.8. NACK-Oriented Reliable Multicast (NORM) . . . 21
3.7. Lightweight User Datagram Protocol (UDP-Lite) . . . . . . 20
3.8.1. 21
3.7.1. Protocol Description . . . . . . . . . . . . . . . . 20
3.8.2. 21
3.7.2. Interface Description . . . . . . . . . . . . . . . . 21
3.8.3. 22
3.7.3. Transport Protocol Components . . . . . . . Features . . . . . 21
3.9. Transport Layer Security (TLS) and Datagram TLS (DTLS) as
a pseudotransport . . . . . . . . . . . . 22
3.8. Internet Control Message Protocol (ICMP) . . . . . . . . 22
3.9.1. 23
3.8.1. Protocol Description . . . . . . . . . . . . . . . . 23
3.9.2.
3.8.2. Interface Description . . . . . . . . . . . . . . . . 24
3.9.3.
3.8.3. Transport Protocol Components . . . . Features . . . . . . . . 24
3.10. Hypertext Transport Protocol (HTTP) over TCP as a
pseudotransport . . . . . . . . . 24
3.9. Realtime Transport Protocol (RTP) . . . . . . . . . . . . 25
3.10.1.
3.9.1. Protocol Description . . . . . . . . . . . . . . . . 25
3.10.2.
3.9.2. Interface Description . . . . . . . . . . . . . . . . 26
3.10.3.
3.9.3. Transport Protocol Components Features . . . . . . . . . . . 27
3.11. WebSockets . . . . . . 26
3.10. File Delivery over Unidirectional Transport/Asynchronous
Layered Coding Reliable Multicast (FLUTE/ALC) . . . . . . 26
3.10.1. Protocol Description . . . . . . . . . . . 27
3.11.1. Protocol Description . . . . . 27
3.10.2. Interface Description . . . . . . . . . . . 27
3.11.2. Interface Description . . . . 29
3.10.3. Transport Features . . . . . . . . . . . 27
3.11.3. Transport Protocol Components . . . . . . 29
3.11. NACK-Oriented Reliable Multicast (NORM) . . . . . 28
4. Transport Service Features . . . . 30
3.11.1. Protocol Description . . . . . . . . . . . . . 28
4.1. Complete Protocol Feature Matrix . . . 30
3.11.2. Interface Description . . . . . . . . . 30
5. IANA Considerations . . . . . . 31
3.11.3. Transport Features . . . . . . . . . . . . . . . 31
6. . . 32
3.12. Transport Layer Security Considerations (TLS) and Datagram TLS (DTLS) as
a pseudotransport . . . . . . . . . . . . . . . . . . . 31
7. Contributors . 32
3.12.1. Protocol Description . . . . . . . . . . . . . . . . 33
3.12.2. Interface Description . . . . . . . 32
8. Acknowledgments . . . . . . . . 34
3.12.3. Transport Features . . . . . . . . . . . . . . . . 32
9. References . 34
3.13. Hypertext Transport Protocol (HTTP) over TCP as a
pseudotransport . . . . . . . . . . . . . . . . . . . . . 35
3.13.1. Protocol Description . . . 32
9.1. Normative References . . . . . . . . . . . . . 36
3.13.2. Interface Description . . . . . 33
9.2. . . . . . . . . . . 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 . . . . . . . . . . . . . . . . . 33 . . 43
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 39 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, integrity protection, and minimal latency. 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 a windowed
congestion control, NewReno, TFRC or LEDBAT congestion control mechanism).
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
provides and
DCCP provide partial integrity protection, and LEDBAT can provide low-
priority 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).

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

3. Existing Transport Protocols

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

[EDITOR'S NOTE: Contributions to the subsections below are welcome]

3.1. Transport Control Protocol (TCP)

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

3.1.1. Protocol Description

TCP is a connection-oriented protocol, providing a three way
handshake to allow a client and server to set up a connection, 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.
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 has a different role as
it is often used to
indicate the requested service.

TCP partitions a continuous stream of bytes into segments, sized to
fit in IP packets. ICMP-based PathMTU discovery [RFC1191][RFC1981]
as well as Packetization Layer Path MTU Discovery (PMTUD) [RFC4821]
are supported.

Each byte in the stream is identified by a sequence number. The
sequence number is used to order segments on receipt, to identify
segments in acknowledgments, and to detect unacknowledged segments
for retransmission. This is the basis of TCP's 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.

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: Control [RFC5681]: This uses a
separate window, where each time congestion is detected, this
congestion window is reduced. A receiver detects Most of the used congestion using control
mechanisms use one of three
mechanisms: mechanisms to detect congestion: A
retransmission timer, 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])
[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. e.g., to enable smoother reduce latency
for interactive sessions.

[EDITOR'S NOTE: add URGENT

TCP provides a push and PUSH flag (note 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] says SHOULD
NOT does not recommend the use
of the urgent flag due to the range of TCP implementations that
process TCP urgent indications differently.) ] differently.

A checksum provides an Integrity Check and is mandatory across the
entire packet. The TCP checksum does not support partial corruption
protection as in DCCP/UDP-Lite). This check protects from
misdelivery delivery of data corrupted data, but data
and miselivery of packets to the wrong endpoint. This check is
relatively weak, and applications that require end to end integrity of
data are recommended to include a stronger integrity check of their
payload data.

A The TCP service is unicast. 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
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. 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.

(more on the API goes here) However, there is no RFC that documents this
interface.

3.1.3. Transport Protocol Components Features

The transport protocol components features provided by TCP (new version) are:

[EDITOR'S NOTE: discussion expand each of how to map this to features and TAPS:
what does the higher layer need to decide? what can the transport
layer decide based on global settings? what must the these slightly]

o unicast transport layer
decide based on network characteristics?]

o Connection-oriented bidirectional communication using three-way
handshake connection setup with feature negotiation and an
explicit distinction between passive and active open: This implies
both unicast addressing 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 guarantee
combination of return routability.

o Single stream-oriented transmission: The stream abstraction atop the datagram service provided by ports and IP is implemented by dividing the
stream into segments. address fields.

o Limited control over segment transmission scheduling (Nagle's
algorithm): This allows for delay minimization in interactive
applications by preventing the transport to add additional delays
(by deactivating Nagle's algorithm). Uni-or bidirectional communication

o Port multiplexing, with application-to-port mapping during
connection setup: Note that in the presence of network address and
port translation (NAPT), TCP ports are stream-oriented delivery in effect part of the
endpoint address for forwarding purposes. a single stream

o Full reliability using (S)ACK- and RTO-based loss detection and
retransmissions: Loss is sensed fully reliable delivery, implemented using duplicated ACKs ("fast
retransmit"), which places a lower bound on sent from the delay inherent in
this approach
receiver to reliability. The retransmission timeout
determines the upper bound on the delay (expect if also
exponential back-off is performed). The use of selective
acknowlegdements further reduces the latency for retransmissions
if multiple packets are lost during one congestion event. confirm delivery.

o Error detection based on error detection: a segment checksum covering verifies delivery to the network
correct endpoint and
transport headers as well as payload: Packets that are detected as
corrupted integrity of the data and options.

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

o data bundling, an optional mechanism that uses Nagle's algorithm
to
retransmit them.

o Window-based flow control, with receiver-side window management
and signaling of available window: Scaling coalesce data sent within the same RTT into full-sized
segments.

o flow control window
beyond 64kB requires the use of an optional feature, which has
performance implications in environments using a window-based mechanism, where this option is not
supported; this can be the case either if the receiver does not
implement window scaling or if a network node on the path strips
advertises the window scaling option. that it is willing to buffer.

o Window-based congestion control: a window-based method that uses AIMD to
control reacting to loss, delay,
retransmission timeout, or an explicit congestion signal (ECN):
Most commonly used is a loss signal from the reliability
component's retransmission mechanism. TCP reacts the sending rate and to conservatively choose a rate after
congestion
signal by reducing the size of the congestion window;
retransmission timeout is generally handled with a larger reaction
than other signals. 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 recommend recommends the use of extensions defined for SCTP
[RFC6458] (see next section) to deal with support multihoming.

[AUTHOR'S NOTE: research work, and some implementation, also suggest
that the scheduling algorithm, as well as the path manager, are
configurable options that should be exposed to higher layer. Should
this be discussed here?]

3.2.3. Transport Protocol Components

[AUTHOR'S NOTE: shouldn't it be "service feature"?] features

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

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

o congestion control with load balancing over mutiple connections connections.

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

[AUTHOR'S NOTE: it is unclear whether MPTCP has to provide data
bundling.] [AUTHOR'S NOTE: AF muliplexing? sub-flows can be started
over IPv4 or IPv6 for the same session]

3.3. Stream Control Transmission Protocol (SCTP)

SCTP is a message oriented message-oriented standards track transport protocol and the 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 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 re-
configurations of streams [RFC6525] and IP-addresses [RFC5061].
Furthermore, the extension specified in [RFC3758] introduces the
concept of partial reliability for user messages.

SCTP was originally developed for transporting telephony signalling
messages and is deployed in telephony signalling networks, especially
in mobile telephony networks. Additionally, it is 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 WEB-browsers supporting WebRTC.

3.3.1. Protocol Description

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

SCTP uses the 32-bit CRC32c for protecting SCTP packets against bit
errors.
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 coverage, as provided by DCCP or UDP-Lite is not
supported.

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

SCTP provides a message-oriented service. Multiple small user
messages can be bundled into a single SCTP packet to improve the
efficiency. For example, this bundling may be done by delaying user
messages at the sender side similar to the Nagle algorithm used by TCP.
User messages which would result in IP packets larger than the MTU
will be fragmented at the sender side and reassembled at the
receiver side. 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. Only all ordered Ordered messages sent on the same stream are
delivered at the receiver in the same order as sent by the sender.
For user messages not requiring fragmentation, this minimises head of
line blocking.

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

According to [RFC4960], each 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, way [RFC4960], similar to the TCP behaviour. In addition to
this reliable transfer, the partial reliability extension defined in [RFC3758]
allows the a sender to abandon user messages. The application can
specify the policy for abandoning user messages. Examples for these
policies include: 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 end-point 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.

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

Having a well defined API is also a feature provided by

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

3.3.2. Interface Description

[RFC4960] defines an abstract API for the base protocol. An
extension to the BSD Sockets This API is defined in [RFC6458]
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]. [RFC3758] to
specify for a user message the PR-SCTP policy and the policy
specific parameter.

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

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

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

o the SCTP SACK-IMMEDIATELY extension defined in [RFC7053].

o the SCTP Stream Reconfiguration extension defined in [RFC6525].

o the UDP
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
[I-D.ietf-tsvwg-sctp-prpolicies]. [RFC7496]. The API
allows to enable/disable the PR-SCTP extension, choose the PR-SCTP
policies defined in the document and provide statistical
information about abandoned messages.

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

The SCTP socket API supports two kinds of sockets:

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

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

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

A couple of new

New functions have been introduced to support the use of multiple
local and remote addresses. Additional SCTP-specific send and
receive calls have been defined to allow dealing with the SCTP
specific permit SCTP-specific information
to be snet without using ancillary data in the form of additional cmsgs, which are also defined.
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 Protocol Components Features

The transport protocol components features provided by SCTP are:

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

o unicast unicast.

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

o port multiplexing multiplexing.

o message-oriented delivery.

o fully reliable or partially reliable delivery delivery.

o ordered and unordered delivery within a stream stream.

o support for multiple concurrent streams streams.

o support for stream scheduling prioritization prioritization.

o flow control

o message-oriented delivery control.

o congestion control control.

o user message bundling bundling.

o user message fragmentation and reassembly reassembly.

o strong error error/misdelivery detection (CRC32C) (CRC32c).

o transport layer multihoming for resilience resilience.

o transport layer mobility

[EDITOR'S NOTE: update this list.] mobility.

3.4. User Datagram Protocol (UDP)

The User Datagram Protocol (UDP) [RFC0768] [RFC2460] is an IETF
standards track transport protocol. It provides a uni-directional, unidirectional,
datagram protocol which 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[RFC5405]. in[I-D.ietf-tsvwg-rfc5405bis]. UDP is widely
implemented and widely used by common applications, especially including DNS.

3.4.1. Protocol Description

UDP is a connection-less protocol which that maintains message boundaries,
with no connection setup or feature negotiation. The protocol uses
independent messages, ordinarily called datagrams. The lack Each stream of error
control and flow control implies
messages may be damaged, re-ordered,
lost, or duplicated in transit. A receiving application unable to
run sufficiently fast or frequently may miss messages. The lack is independently managed, therefore retransmission does not
hold back data sent using other logical streams. It provides
detection of
congestion handling implies UDP traffic may cause the loss payload errors and misdelivery of
messages from other protocols (e.g., TCP) when sharing the same
network paths. UDP traffic can also cause packets to the loss wrong
endpoint, either of other UDP
traffic which result in the same or other flows for the same reasons.

Messages with bit errors are ordinarily detected by an invalid end-
to-end checksum and are discarded before being delivered to an
application. There are some exceptions to this general rule,
however. UDP-Lite (see [RFC3828], and below) provides the ability
for portions discard of the message contents to be exempt from checksum
coverage. received datagrams.

It is also possible to create IPv4 UDP datagrams with no checksum, and
while this is generally discouraged [RFC1122]
[RFC5405],
[I-D.ietf-tsvwg-rfc5405bis], certain special cases permit its use 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.

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. IP and are reassembled before
delivery to the UDP receiver.

Applications concerned with 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 other than operating over UDP [RFC5405]. [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 [RFC5405]. [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) binding 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
[RFC5405].
[I-D.ietf-tsvwg-rfc5405bis].

3.4.3. Transport Protocol Components Features

The transport protocol components features provided by UDP are:

o unidirectional

o port multiplexing

o 2-tuple endpoints unicast.

o multicast, anycast, or IPv4 broadcast, multicast and anycast

o IPv6 multicast and anycast broadcast.

o IPv6 jumbograms port multiplexing. A receiving port can be configured to receive
datagrams from multiple senders.

o message-oriented delivery 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 error detection (checksum)

o checksum 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. UDP-Lite It provides a
bidirectional set of logical unicast or multicast message streams
over a
unidirectional, datagram protocol. protocol that preserves message boundaries.
IETF guidance on the use of UDP-Lite is provided in [RFC5405].
[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.
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 fragments packets into IP packets, constrained by the
maximum size of IP packet.

UDP-Lite changes the semantics of the UDP "payload length" field to
that of a "checksum coverage length" field. 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 [RFC5405]. [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. (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 [RFC5405]. [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 Protocol Components Features

The transport protocol components features provided by UDP-Lite are:

o unicast unicast.

o multicast, anycast, or IPv4 broadcast, multicast and anycast broadcast.

o port multiplexing (as for UDP).

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

o message-oriented non-reliable delivery (as for UDP).

o partial non-ordered delivery (as for UDP).

o error and misdelivery detection (checksum).

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

3.6. Datagram Congestion Control Protocol (DCCP)

Datagram Congestion Control Protocol (DCCP) [RFC4340] is an IETF
standards track bidirectional transport protocol that provides
unicast connections of congestion-controlled unreliable messages.

[EDITOR'S NOTE: Gorry Fairhurst signed up as a contributor for this
section.] 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 on at each host 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 the service code

[RFC5595]
[RFC5595], a mechanism to allow 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 MAY allow 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
Data Checksum option, which option that when enabled, contains a strong CRC, lets to
enable endpoints to detect application data corruption. It also supports reduced checksum coverage, a partial
integrity mechanisms similar to UDP-lIte.

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

A DCCP protocol instance can be extended [RFC4340] and tuned. 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 Examples of
specified profiles include [RFC4341] [RFC4342] [RFC5662]. All IETF-
defined methods provide Congestion Control.

DCCP use a Connect packet to start 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 DCCP service is unicast.

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

3.6.2. Interface Description

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

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

3.6.3. RFC Series.

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

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].

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].

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).

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
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).

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
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]
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.

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
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 Protocol Components Features

The transport protocol components features provided by DCCP FLUTE/ALC are:

o unicast

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

o Service Codes multicast, anycast or IPv4 broadcast.

o port multiplexing per-object dynamic meta-data delivery.

o non-reliable, ordered push delivery or on-demand delivery service.

o flow control (slow receiver function) fully reliable or partially reliable delivery (of file or in-
memory objects).

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

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

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

o partial integrity protection

3.7. Realtime Transport Protocol (RTP)

[EDITOR'S NOTE: Varun Singh signed up as contributor

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

o congestion control for this
section. Given the complexity of RTP, suggest to have an abbreviated
section here contrasting RTP layered flows (e.g., with other transports, and focusing on
those features that are RTP-unique.]

3.8. 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.

3.8.1.

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
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.

3.8.2.

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.

3.8.3.

3.11.3. Transport Protocol Components Features

The transport protocol components features provided by NORM are:

o unicast or multicast.

o multicast stream-oriented delivery in a single stream.

o port multiplexing (UDP ports) object-oriented delivery of discrete data or file items.

o reliable delivery delivery.

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

o error detection (UDP checksum)

o segmentation

o stream-oriented delivery in a single stream checksum).

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

o data bundling (Nagle's algorithm) algorithm).

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

o congestion control control.

o packet erasure coding (both proactively and as part of ARQ)

3.9. 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 (usually UDP). (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.

[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.]

3.9.1.

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 TCP, 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) (optional).

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

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

o An anti-DoS stateless cookie mechanism (optional) (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 bans forbids the use of
stream ciphers, which are essentially incompatible when operating on
independent encrypted records.

3.9.2.

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.

3.9.3.

3.12.3. Transport Protocol Components Features

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

o message fragmentation

o authentication codes (MACs),
encrypting data, and invoking integrity via message authentication codes
(MAC)

o data encryption

o scheduling transmission from using the underlying transport protocol. protocol

DTLS augments the TLS record protocol with
sequence numbers used for with:

o ordering and replay detection. 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.

3.10.

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.

3.10.1.

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.

3.10.2.

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.

3.10.3.

3.13.3. Transport Protocol Components features

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

o unicast

o reliable delivery

o ordered delivery unicast.

o message and stream-oriented transfer.

o bi- or unidirectional transmission.

o ordered delivery.

o fully reliable delivery.

o object range request request.

o message content type negotiation negotiation.

o congestion control flow control.

HTTPS (HTTP over TLS) additionally provides the following components:

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

o confidentiality confidentiality.

o integrity protection

3.11. WebSockets

[RFC6455]

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

3.11.1. Protocol Description

3.11.2. Interface Description
3.11.3. Transport Protocol Components 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

+ broadcast (IPv4 only)

+ multicast

+ multicast, anycast

+ something on ports and NAT 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
- 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
* 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

The next

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.

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

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

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

5. IANA Considerations

This document has no considerations for IANA.

6. Security Considerations

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

7. Contributors

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

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

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

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

o Section 3.8 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 3.9 3.12 on MPTCP TLS and DTLS was contributed by Ralph Holz
(ralph.holz@nicta.com.au) and Olivier Mehani
(olivier.mehani@nicta.com.au)

o Section 3.10 3.13 on HTTP was contributed by Dragana Damjanovic
(ddamjanovic@mozilla.com)

8. Acknowledgments

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

[EDITOR'S NOTE: add H2020-NEAT ack].

9. References
9.1. Normative References

[RFC0791] Postel, J., "Internet Protocol", STD 5, RFC 791, September
1981.

9.2. Informative References

[RFC0768] Postel, J., "User Datagram Protocol", STD 6, RFC 768, DOI
10.17487/RFC0768, August 1980. 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. 1981,
<http://www.rfc-editor.org/info/rfc793>.

[RFC0896] Nagle, J., "Congestion control Control in IP/TCP internetworks", Internetworks",
RFC 896, DOI 10.17487/RFC0896, January 1984. 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. 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 1990.
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.
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. 1996,
<http://www.rfc-editor.org/info/rfc2018>.

[RFC2045] Freed, N. and N. Borenstein, "Multipurpose Internet Mail
Extensions (MIME) Part One: Format of Internet Message
Bodies", RFC 2045, DOI 10.17487/RFC2045, November 1996. 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. 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. 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. 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.

[RFC3390] Allman, M., Floyd, S., and C. Partridge, "Increasing TCP's
Initial Window", RFC 3390, October 2002. 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.
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.
2002, <http://www.rfc-editor.org/info/rfc3452>.

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

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

[RFC3758] Stewart, R., Ramalho, M., Xie, Q., Tuexen, M., and P.
Conrad, "Stream Control Transmission Protocol (SCTP)
Partial Reliability Extension", RFC 3758, DOI 10.17487/
RFC3758, May 2004. 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.
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.
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. 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. 2006,
<http://www.rfc-editor.org/info/rfc4340>.

[RFC4341] Floyd, S. and E. Kohler, "Profile for Datagram Congestion
Control Protocol (DCCP) Congestion Control ID 2: TCP-like
Congestion Control", RFC 4341, DOI 10.17487/RFC4341, March 2006.
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. 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.
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. 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. 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. 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.
2007, <http://www.rfc-editor.org/info/rfc4895>.

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

[RFC5061] Stewart, R., Xie, Q., Tuexen, M., Maruyama, S., and M.
Kozuka, "Stream Control Transmission Protocol (SCTP)
Dynamic Address Reconfiguration", RFC 5061, DOI 10.17487/
RFC5061, September
2007. 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. 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, August 2008.

[RFC5348] Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP
Friendly Rate 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 (TFRC): Protocol Specification", (DCCP)", RFC
5348, September 2008.

[RFC5405] Eggert, L.
5238, DOI 10.17487/RFC5238, May 2008,
<http://www.rfc-editor.org/info/rfc5238>.

[RFC5404] Westerlund, M. and G. Fairhurst, "Unicast UDP Usage Guidelines I. Johansson, "RTP Payload Format for Application Designers", BCP 145,
G.719", RFC 5405, November
2008. 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. 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. 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. 2010,
<http://www.rfc-editor.org/info/rfc5662>.

[RFC5672] Crocker, D., Ed., "RFC 4871 DomainKeys Identified Mail
(DKIM) Signatures -- Update", RFC 5672, DOI 10.17487/
RFC5672, August 2009. 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.

[RFC6773] Phelan, T., Fairhurst, G., and C. Perkins, "DCCP-UDP: A
Datagram Congestion Control Protocol UDP Encapsulation for
NAT Traversal", RFC 6773, November 2012.

[RFC5925] Touch, J., Mankin, A., 2009,
<http://www.rfc-editor.org/info/rfc5740>.

[RFC5775] Luby, M., Watson, M., and R. Bonica, "The TCP
Authentication Option", RFC 5925, June 2010. 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. 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. 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. 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. 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.

[RFC6298] Paxson, V., Allman, M., Chu, J., and M. Sargent,
"Computing TCP's Retransmission Timer", RFC 6298, June
2011. 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. 2012, <http://www.rfc-editor.org/info/rfc6347>.

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

[RFC6363] Watson, M., Begen, A., and V. Roca, "Forward Error
Correction (FEC) Framework", RFC 6363, DOI 10.17487/
RFC6363, October 2011. 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. 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.

[RFC6691] Borman, D., "TCP Options 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 Maximum Segment Size (MSS)", R. Lehtonen,
"FLUTE - File Delivery over Unidirectional Transport", RFC 6691, July 2012.
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. 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. 2013, <http://www.rfc-editor.org/info/rfc6897>.

[RFC6935] Eubanks, M., Chimento, P., and M. Westerlund, "IPv6 and
UDP Checksums for Tunneled Packets", RFC 6935, DOI
10.17487/RFC6935, April 2013. 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. 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. 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. 2013,
<http://www.rfc-editor.org/info/rfc7053>.

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

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

[RFC7232] Fielding, R. R., Ed. and J. Reschke, Ed., "Hypertext Transfer
Protocol (HTTP/1.1): Conditional Requests", RFC 7232, DOI
10.17487/RFC7232, June 2014. 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. 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. 2014,
<http://www.rfc-editor.org/info/rfc7234>.

[RFC7235] Fielding, R. R., Ed. and J. Reschke, Ed., "Hypertext Transfer
Protocol (HTTP/1.1): Authentication", RFC 7235, DOI
10.17487/RFC7235, June 2014. 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. 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. 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. 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.
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-05
ecn-benefits-06 (work in progress), June July 2015.

[I-D.ietf-tsvwg-sctp-dtls-encaps]
Tuexen, M., Stewart, R., Jesup, R., and S. Loreto, "DTLS
Encapsulation of SCTP Packets", draft-ietf-tsvwg-sctp-
dtls-encaps-09 (work in progress), January 2015.

[I-D.ietf-tsvwg-sctp-prpolicies]
Tuexen, M., Seggelmann, R., Stewart, R., and S. Loreto,
"Additional Policies for the Partial Reliability Extension
of the Stream Control Transmission Protocol", draft-ietf-
tsvwg-sctp-prpolicies-07 (work in progress), February
2015.

[I-D.ietf-tsvwg-sctp-ndata]
Stewart, R., Tuexen, M., Loreto, S., and R. Seggelmann,
"Stream Schedulers and User Message Interleaving for the
Stream Control Transmission Protocol", draft-ietf-tsvwg-
sctp-ndata-03
sctp-ndata-04 (work in progress), March 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-07 draft-ietf-tsvwg-natsupp-08 (work in progress),
February
July 2015.

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

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Authors' Addresses

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

Email: gorry@erg.abdn.ac.uk
Brian Trammell (editor)
ETH Zurich
Gloriastrasse 35
8092 Zurich
Switzerland

Email: ietf@trammell.ch

Mirja Kuehlewind (editor)
ETH Zurich
Gloriastrasse 35
8092 Zurich
Switzerland

Email: mirja.kuehlewind@tik.ee.ethz.ch