wdiff draft-ietf-taps-transports-04.txt draft-ietf-taps-transports-05.txt

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

Services provided by IETF transport protocols and congestion control
mechanisms
draft-ietf-taps-transports-04
draft-ietf-taps-transports-05

Abstract

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

Status of This Memo

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Table of Contents

1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4
3. Existing Transport Protocols . . . . . . . . . . . . . . . . 4
3.1. Transport Control Protocol (TCP) . . . . . . . . . . . . 4
3.1.1. Protocol Description . . . . . . . . . . . . . . . . 5
3.1.2. Interface description . . . . . . . . . . . . . . . . 6
3.1.3. Transport Protocol Components . . . . . . . . . . . . 6
3.2. Multipath TCP (MP-TCP) (MPTCP) . . . . . . . . . . . . . . . . . . 7
3.2.1. Protocol Description . . . . . . . . . . . . . . . . 7
3.2.2. Interface Description . . . . . . . . . . . . . . . . 7
3.2.3. Transport Protocol Components . . . . . . . . . . . . 8
3.3. Stream Control Transmission Protocol (SCTP) . . . . . . . 7 9
3.3.1. Protocol Description . . . . . . . . . . . . . . . . 8 9
3.3.2. Interface Description . . . . . . . . . . . . . . . . 10 11
3.3.3. Transport Protocol Components . . . . . . . . . . . . 11 13
3.4. User Datagram Protocol (UDP) . . . . . . . . . . . . . . 12 13
3.4.1. Protocol Description . . . . . . . . . . . . . . . . 12 14
3.4.2. Interface Description . . . . . . . . . . . . . . . . 13 14
3.4.3. Transport Protocol Components . . . . . . . . . . . . 13 15
3.5. Lightweight User Datagram Protocol (UDP-Lite) . . . . . . 14 15
3.5.1. Protocol Description . . . . . . . . . . . . . . . . 14 15
3.5.2. Interface Description . . . . . . . . . . . . . . . . 15 16
3.5.3. Transport Protocol Components . . . . . . . . . . . . 15 16
3.6. Datagram Congestion Control Protocol (DCCP) . . . . . . . 15 17
3.6.1. Protocol Description . . . . . . . . . . . . . . . . 16 17
3.6.2. Interface Description . . . . . . . . . . . . . . . . 17 19
3.6.3. Transport Protocol Components . . . . . . . . . . . . 17 19
3.7. Realtime Transport Protocol (RTP) . . . . . . . . . . . . 18 19
3.8. NACK-Oriented Reliable Multicast (NORM) . . . . . . . . . 18 20
3.8.1. Protocol Description . . . . . . . . . . . . . . . . 18 20
3.8.2. Interface Description . . . . . . . . . . . . . . . . 19 21
3.8.3. Transport Protocol Components . . . . . . . . . . . . 20 21
3.9. Transport Layer Security (TLS) and Datagram TLS (DTLS) as
a pseudotransport . . . . . . . . . . . . . . . . . . . . 20 22
3.9.1. Protocol Description . . . . . . . . . . . . . . . . 21 23
3.9.2. Interface Description . . . . . . . . . . . . . . . . 21
3.9.3. Transport Protocol Components . . . . . . . . . . . . 21 23
3.10. Hypertext Transport Protocol (HTTP) over TCP as a
pseudotransport . . . . . . . . . . . . . . . . . . . . . 21 24
3.10.1. Protocol Description . . . . . . . . . . . . . . . . 21 25
3.10.2. Interface Description . . . . . . . . . . . . . . . 22 26
3.10.3. Transport Protocol Components . . . . . . . . . . . 23 26
3.11. WebSockets . . . . . . . . . . . . . . . . . . . . . . . 23 27
3.11.1. Protocol Description . . . . . . . . . . . . . . . . 23 27
3.11.2. Interface Description . . . . . . . . . . . . . . . 24 27
3.11.3. Transport Protocol Components . . . . . . . . . . . 24 27
4. Transport Service Features . . . . . . . . . . . . . . . . . 24 27
4.1. Complete Protocol Feature Matrix . . . . . . . . . . . . 26 29
5. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 28 31
6. Security Considerations . . . . . . . . . . . . . . . . . . . 28 31
7. Contributors . . . . . . . . . . . . . . . . . . . . . . . . 28 31
8. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 28 31
9. References . . . . . . . . . . . . . . . . . . . . . . . . . 28 32
9.1. Normative References . . . . . . . . . . . . . . . . . . 28 32
9.2. Informative References . . . . . . . . . . . . . . . . . 29 32
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 34 38

1. Introduction

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

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

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

2. Terminology

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

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

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

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

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

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

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

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

3. Existing Transport Protocols

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

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

3.1. Transport Control Protocol (TCP)

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

3.1.1. Protocol Description

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

TCP provides multiplexing to multiple sockets on each host using port
numbers. An active TCP session is identified by its four-tuple of
local and remote IP addresses and local port and remote port numbers.
The destination port during connection setup has a different role as
it is often used to indicate the requested service.

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

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

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

All TCP senders provide Congestion Control: This uses a separate
window, where each time congestion is detected, this congestion
window is reduced. A receiver detects congestion using one of three
mechanisms: A retransmission timer, detection of loss (interpreted as
a congestion signal), or Explicit Congestion Notification (ECN)
[RFC3168] to provide early signaling (see
[I-D.ietf-aqm-ecn-benefits])

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

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

[EDITOR'S NOTE: add URGENT and PUSH flag (note [RFC6093] says SHOULD
NOT use due to the range of TCP implementations that process TCP
urgent indications differently.) ]

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

A TCP service is unicast.

3.1.2. Interface description

A User/TCP Interface is defined in [RFC0793] providing six user
commands: Open, Send, Receive, Close, Status. This interface does
not describe configuration of TCP options or parameters beside use of
the PUSH and URGENT flags.

In API implementations derived from the BSD Sockets API, TCP sockets
are created using the "SOCK_STREAM" socket type.

The features used by a protocol instance may be set and tuned via
this API.

(more on the API goes here)

3.1.3. Transport Protocol Components

The transport protocol components provided by TCP are:

o unicast

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

o port multiplexing

o reliable delivery
o ordered delivery for each byte stream

o error detection (checksum)

o segmentation

o stream-oriented delivery in a single stream

o data bundling (Nagle's algorithm)

o flow control

o congestion control

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

3.2. Multipath TCP (MP-TCP)

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

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

3.3. Stream Control Transmission Protocol (SCTP)

SCTP is a message oriented standards track transport protocol and the
base protocol is specified in [RFC4960]. an extension for TCP to support multi-
homing. It supports multi-homing is designed to
handle path failures. An SCTP association has multiple
unidirectional streams in each direction and provides in-sequence
delivery of user messages only within each stream. This allows be as transparent as possible to
minimize head middle-
boxes. It does so by establishing regular TCP flows between a pair
of line blocking. SCTP is extensible source/destination endpoints, and multiplexing the currently
defined extensions include mechanisms for dynamic re-configurations
of streams [RFC6525] and IP-addresses [RFC5061]. Furthermore, the
extension specified in [RFC3758] introduces the concept of partial
reliability for user messages.

SCTP was originally developed application's
stream over these flows.

3.2.1. Protocol Description

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

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

3.3.1. Protocol Description

SCTP is a connection oriented protocol using a four way handshake to establish an SCTP association and a three way message exchange to
gracefully shut it down. It uses new
sessions between pairs of endpoints.

3.2.2. Interface Description

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

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

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

SCTP has been designed with extensibility in mind. Each more selected local endpoints; -
query local and remote endpoint addresses; - get a unique connection
identifier (similar to an address-port pair for TCP).

The document also recommend the use of extensions defined for SCTP packet
starts
[RFC6458] (see next section) to deal with a single common header containing multihoming.

[AUTHOR'S NOTE: research work, and some implementation, also suggest
that the port numbers, a
verification tag and the CRC32c checksum. This common header is
followed by a sequence of chunks. Each chunk consists of a type
field, flags, a length field and a value. [RFC4960] defines how a
receiver processes chunks with an unknown chunk type. The support of
extensions can be negotiated during scheduling algorithm, as well as the SCTP handshake.

SCTP provides a message-oriented service. Multiple small user
messages can path manager, are
configurable options that should be bundled into a single SCTP packet exposed to improve the
efficiency. For example, higher layer. Should
this bundling may be done by delaying user
messages at the sender side similar discussed here?]

3.2.3. Transport Protocol Components

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

As an extension to TCP, MPTCP provides mostly the Nagle algorithm used same components.
By establishing multiple sessions between available endpoints, it can
additionally provide soft failover solutions should one of the paths
become unusable. In addition, by
TCP. User messages which would result in IP packets larger multiplexing one byte stream over
separate paths, it can achieve a higher throughput than the
MTU will be fragmented TCP in
certain situations (note however that coupled congestion control
[RFC6356] might limit this benefit to maintain fairness to other
flows at the sender side bottleneck). When aggregating capacity over multiple
paths, and reassembled at the
receiver side. There is no protocol limit depending on the user message size.
ICMP-based path MTU discovery as specified for IPv4 in [RFC1191] and
for IPv6 in [RFC1981] as well as packetization layer path MTU
discovery as specified in [RFC4821] with probe way packets using the
padding chunks defined the [RFC4820] are supported.

[RFC4960] specifies a scheduled on each TCP friendly congestion control
subflow, an additional delay and higher jitter might be observed
observed before in-order delivery of data to protect the
network against overload. SCTP also uses a sliding window flow
control to protect receivers against overflow.

Each SCTP association has between 1 applications.

The transport protocol components provided by MPTCP therefore are:

o unicast

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

o port multiplexing

o reliable delivery

o error detection (checksum)

o segmentation

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

o flow control

o congestion control

o endpoint multiplexing of streams can be different in each
direction. Every user-message is sent on a particular stream. User
messages single byte stream (higher throughput)

o resilience to network failure and/or handovers

[AUTHOR'S NOTE: it is unclear whether MPTCP has to provide data
bundling.] [AUTHOR'S NOTE: AF muliplexing? sub-flows can be sent un-ordered started
over IPv4 or ordered upon request by the upper
layer. Un-ordered messages can be delivered as soon as they are
completely received. Only all ordered messages sent on IPv6 for the same
stream are delivered at session]

3.3. Stream Control Transmission Protocol (SCTP)

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

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

3.3.1. Protocol Description

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

SCTP handshake is
not sufficient.

According to [RFC4960], each user message sent is either delivered to
the receiver or, in case of excessive retransmissions, uses the
association 32-bit CRC32c for protecting SCTP packets against bit
errors. This is terminated in a non-graceful way, similar to stronger than the 16-bit checksums used by TCP
behaviour. In addition to this reliable transfer, the partial
reliability extension defined in [RFC3758] allows the sender to
abandon user messages. The application can specify the policy for
abandoning user messages. Examples for these policies include:

o Limiting the time or
UDP. However, a user message partial checksum coverage as provided by DCCP or
UDP-Lite is dealt not supported.

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

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

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

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

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

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

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

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

Each SCTP as
described association has between 1 and 65536 uni-directional streams
in each direction. The number of streams can be different in each
direction. Every user-message is sent on a particular stream. User
messages can be sent un-ordered or ordered upon request by the next subsection.

3.3.2. Interface Description

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

o the same order as sent by the
sender. For user messages not requiring fragmentation, this
minimises head of line blocking. The base protocol defined in [RFC4960].

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

o sending a large message on one stream can block the SCTP Authentication sending of
user messages on other streams. [I-D.ietf-tsvwg-sctp-ndata]
overcomes this limitation. Furthermore, [I-D.ietf-tsvwg-sctp-ndata]
specifies multiple algorithms for the sender side selection of which
streams to send data from supporting a variety of scheduling
algorithms including priority based ones. The stream re-
configuration extension defined in [RFC4895].

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

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

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

o TCP
behaviour. In addition to this reliable transfer, the SCTP SACK-IMMEDIATELY partial
reliability extension defined in [RFC7053]. [RFC3758] allows the sender to
abandon user messages. The application can specify the policy for
abandoning user messages. Examples for these policies include:

o Limiting the SCTP Stream Reconfiguration extension defined in [RFC6525]. time a user message is dealt with by the sender.

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

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

Future documents describing SCTP protocol extensions are expected number of retransmissions is limited to
describe the corresponding BSD Sockets API extension in 0, one
gets a "Socket API
Considerations" section.

The SCTP socket API supports two kinds of sockets:

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

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

One-to-one style sockets are service similar to TCP sockets, there is UDP.

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

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

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

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

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

The SCTP kernel implementations of FreeBSD, Linux and Solaris follow
mostly traffic is switched over to a
reachable one, if one exists. Each SCTP end-point supervises
continuously the reachability of all peer addresses using a heartbeat
mechanism.

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

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

3.3.3. Transport Protocol Components UDP encapsulation of
SCTP-packets. Alternatively, SCTP packets can be encapsulated in
DTLS packets as specified in [I-D.ietf-tsvwg-sctp-dtls-encaps]. The transport protocol components
latter encapsulation is used with in the WebRTC context.

Having a well defined API is also a feature provided by SCTP are:

o unicast

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

o port multiplexing

o reliable or partially reliable delivery

o ordered as
described in the next subsection.

3.3.2. Interface Description

[RFC4960] defines an abstract API for the base protocol. An
extension to the BSD Sockets API is defined in [RFC6458] and unordered delivery within a stream covers:

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

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

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

o message-oriented delivery
o congestion control

o user message bundling

o user message fragmentation and reassembly the SCTP Dynamic Address Reconfiguration extension defined in
[RFC5061].

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

o strong error detection (CRC32C) the SCTP SACK-IMMEDIATELY extension defined in [RFC7053].

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

o transport layer mobility

[EDITOR'S NOTE: update this list.]

3.4. User Datagram Protocol (UDP)

The User Datagram Protocol (UDP) [RFC0768] [RFC2460] is an IETF
standards track transport protocol. It provides a uni-directional,
datagram protocol which preserves message boundaries. It provides
none the UDP Encapsulation of SCTP packets extension defined in
[RFC6951].

o the following transport features: error correction,
congestion control, or flow control. It can be used to send
broadcast datagrams (IPv4) or multicast datagrams (IPv4 and IPv6), additional PR-SCTP policies defined in
addition
[I-D.ietf-tsvwg-sctp-prpolicies].

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

The SCTP socket API supports two kinds of UDP is provided in[RFC5405]. UDP sockets:

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

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

One-to-one style sockets are similar to TCP sockets, there is widely implemented a 1:1
relationship between the sockets and
widely used by common applications, especially DNS.

3.4.1. Protocol Description the SCTP associations (except
for listening sockets). One-to-many style SCTP sockets are similar
to unconnected UDP sockets as there is a connection-less protocol which maintains message boundaries,
with no connection setup or feature negotiation. The protocol uses
independent messages, ordinarily called datagrams. The lack of error
control and flow control implies messages may be damaged, re-ordered,
lost, or duplicated in transit. A receiving application unable to
run sufficiently fast or frequently may miss messages. The lack of
congestion handling implies UDP traffic may cause 1:n relationship between the loss of
messages from other protocols (e.g., TCP) when sharing
sockets and the same
network paths. UDP traffic SCTP associations.

The SCTP stack can also cause provide information to the loss applications about
state changes of other UDP
traffic in the same or other flows for the same reasons.

Messages with bit errors are ordinarily detected by an invalid end-
to-end checksum individual paths and the association whenever
they occur. These events are discarded before being delivered similar to an
application. There user messages but
are some exceptions specifically marked as notifications.

A couple of new functions have been introduced to this general rule,
however. UDP-Lite (see [RFC3828], and below) provides support the ability
for portions use of the message contents to be exempt from checksum
coverage. It is also possible
multiple local and remote addresses. Additional SCTP-specific send
and receive calls have been defined to create UDP datagrams allow dealing with no
checksum, and while this is generally discouraged [RFC1122]
[RFC5405], certain special cases permit its use [RFC6935]. The
checksum support considerations for omitting the checksum are defined SCTP
specific information without using ancillary data in [RFC6936]. Note that due to the relatively weak form of checksum
used by UDP, applications that require end to end integrity of data
are recommended to include a stronger integrity check of their
payload data.

On transmission, UDP encapsulates each datagram into an IP packet,
additional cmsgs, which may in turn be fragmented by IP. Applications concerned with
fragmentation or that have other requirements such as receiver flow
control, congestion control, PathMTU discovery/PLPMTUD, are also defined. These functions provide
support for
ECN, etc need to be provided by protocols other than UDP [RFC5405].

3.4.2. Interface Description

[RFC0768] describes basic requirements for an API for UDP. Guidance
on use of common APIs is provided in [RFC5405].

A UDP endpoint consists of a tuple detecting partial delivery of (IP address, port number).
Demultiplexing using multiple abstract endpoints (sockets) on the
same IP address are supported. user messages and
notifications.

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

The SCTP kernel implementations of FreeBSD, Linux and Solaris follow
mostly the socket
implementation arranges specified extension to not duplicate the same received unicast
message to multiple server processes.

Many operating systems also allow a UDP socket to be "connected",
i.e., to bind a UDP socket to a specific (remote) UDP endpoint.
Unlike TCP's connect primitive, BSD Sockets API for UDP, this is only a local
operation that serves to simplify the local send/receive functions base
protocol and to filter the traffic for the specified addresses and ports
[RFC5405].

3.4.3. Transport Protocol Components corresponding supported protocol extensions.

3.3.3. Transport Protocol Components

The transport protocol components provided by UDP SCTP are:

o unidirectional unicast

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

o port multiplexing

o 2-tuple endpoints reliable or partially reliable delivery

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

o IPv6 multicast and anycast support for multiple concurrent streams

o IPv6 jumbograms support for stream scheduling prioritization

o flow control

o message-oriented delivery

o congestion control

o user message bundling

o user message fragmentation and reassembly

o strong error detection (checksum) (CRC32C)

o checksum optional

3.5. Lightweight transport layer multihoming for resilience

o transport layer mobility

[EDITOR'S NOTE: update this list.]

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

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

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

3.4.1. Protocol Description

UDP-Lite

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

It provides multiplexing may be damaged, re-ordered,
lost, or duplicated in transit. A receiving application unable to multiple sockets on each host using port
numbers. An active UDP-Lite session is identified by its four-tuple
of local and remote IP addresses and local port and remote port
numbers.

UDP-Lite fragments packets into IP packets, constrained by the
maximum size
run sufficiently fast or frequently may miss messages. The lack of IP packet.

UDP-Lite changes
congestion handling implies UDP traffic may cause the semantics loss of
messages from other protocols (e.g., TCP) when sharing the same
network paths. UDP "payload length" field to
that traffic can also cause the loss of a "checksum coverage length" field. Otherwise, UDP-Lite is
semantically identical other UDP
traffic in the same or other flows for the same reasons.

Messages with bit errors are ordinarily detected by an invalid end-
to-end checksum and are discarded before being delivered to UDP. Applications using an
application. There are some exceptions to this general rule,
however. UDP-Lite therefore
can not make assumptions regarding (see [RFC3828], and below) provides the correctness ability
for portions of the data
received message contents to be exempt from checksum
coverage. It is also possible to create UDP datagrams with no
checksum, and while this is generally discouraged [RFC1122]
[RFC5405], certain special cases permit its use [RFC6935]. The
checksum support considerations for omitting the checksum are defined
in [RFC6936]. Note that due to the insensitive part relatively weak form of the UDP-Lite payload.

As for checksum
used by UDP, mechanisms for applications that require end to end integrity of data
are recommended to include a stronger integrity check of their
payload data.

On transmission, UDP encapsulates each datagram into an IP packet,
which may in turn be fragmented by IP. Applications concerned with
fragmentation or that have other requirements such as receiver flow
control, congestion control,
PMTU or PLPMTU discovery, PathMTU discovery/PLPMTUD, support for
ECN, etc need to be provided by
upper layer protocols [RFC5405].

Examples of use include a class of applications that can derive
benefit from having partially-damaged payloads delivered, rather than
discarded. One use is to support error tolerate payload corruption
when used over paths that include error-prone links, another
application is when header integrity checks are required, but payload
integrity is provided by some other mechanism (e.g. [RFC6936].

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

3.5.2. than UDP [RFC5405].

3.4.2. Interface Description

There is no current

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

The interface

A UDP endpoint consists of UDP-Lite differs from that a tuple of UDP by (IP address, port number).
Demultiplexing using multiple abstract endpoints (sockets) on the addition of
same IP address are supported. The same socket may be used by a
single (socket) option that communicates a checksum coverage length
value: at the sender, this specifies the intended checksum coverage, server to interact with the remaining unprotected part multiple clients (note: this behavior
differs from TCP, which uses a pair of tuples to identify a
connection). Multiple server instances (processes) binding the payload called same
socket can cooperate to service multiple clients- the "error-
insensitive part". The checksum coverage may socket
implementation arranges to not duplicate the same received unicast
message to multiple server processes.

Many operating systems also allow a UDP socket to be made visible "connected",
i.e., to bind a UDP socket to a specific (remote) UDP endpoint.
Unlike TCP's connect primitive, for UDP, this is only a local
operation that serves to simplify the application via the UDP-Lite MIB module [RFC5097].

3.5.3. local send/receive functions
and to filter the traffic for the specified addresses and ports
[RFC5405].

3.4.3. Transport Protocol Components

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

o unicast unidirectional

o port multiplexing

o 2-tuple endpoints

o IPv4 broadcast, multicast and anycast

o port multiplexing IPv6 multicast and anycast

o non-reliable, non-ordered delivery IPv6 jumbograms

o message-oriented delivery

o partial integrity protection

3.6. error detection (checksum)

o checksum optional

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

The Lightweight User Datagram Congestion Control Protocol (DCCP) [RFC4340] (UDP-Lite) [RFC3828] is an
IETF standards track bidirectional transport protocol that protocol. UDP-Lite provides
unicast connections of congestion-controlled unreliable messages.

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

The DCCP Problem Statement describes the goals that DCCP sought to
address [RFC4336]. It is suitable for applications that transfer
fairly large amounts
bidirectional set of data and that can benefit from control logical unicast or multicast message streams
over
the trade off between timeliness and reliability [RFC4336].

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

3.6.1. UDP-Lite is
provided in [RFC5405].

3.5.1. Protocol Description

DCCP

UDP-Lite is a connection-oriented connection-less datagram protocol, providing a three
way handshake to allow a client and server to set up a connection,
and mechanisms for orderly completion and immediate teardown of a
connection. with no connection
setup or feature negotiation. The protocol is defined by use messages, rather than
a family byte-stream. Each stream of RFCs. messages is independently managed,
therefore retransmission does not hold back data sent using other
logical streams.

It provides multiplexing to multiple sockets on each host using port
numbers. An active DCCP UDP-Lite session is identified by its four-tuple
of local and remote IP addresses and local port and remote port
numbers.
At connection setup, DCCP also exchanges

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

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

The protocol segments data into messages, typically sized to fit
received in
IP packets, but which may be fragmented providing they are less than
the A DCCP interface MAY allow applications to request fragmentation
for packets larger than PMTU, but not larger than the maximum packet
size allowed by insensitive part of the current UDP-Lite payload.

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

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

Examples of use include a sequence number. The sequence number class of applications that can derive
benefit from having partially-damaged payloads delivered, rather than
discarded. One use is used to identify segments in acknowledgments, to detect
unacknowledged segments, to measure RTT, etc. The protocol may support ordered or unordered delivery of data, and does not itself
provide retransmission. There error tolerate payload corruption
when used over paths that include error-prone links, another
application is a Data Checksum option, which
contains a strong CRC, lets endpoints detect application data
corruption. It also supports reduced checksum coverage, a partial when header integrity checks are required, but payload
integrity mechanisms similar to UDP-lIte.

Receiver flow control is supported: limiting the amount of
unacknowledged data that can be outstanding at a given time. provided by some other mechanism (e.g. [RFC6936].

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

3.5.2. Interface Description

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

DCCP supports negotiation RFC Series, but guidance on
use of common APIs is provided in [RFC5405].

The interface of UDP-Lite differs from that of UDP by the congestion control profile, to
provide Plug and Play congestion control mechanisms. examples addition of
specified profiles include [RFC4341] [RFC4342] [RFC5662]. All IETF-
defined methods provide Congestion Control.

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

3.5.3. Transport Protocol Components

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

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

A DCCP service is unicast.

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

3.6.2. Interface Description

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

There is no current API specified in the RFC Series.

3.6.3. Transport Protocol Components

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

o unicast

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

o Service Codes anycast

o port multiplexing
o non-reliable, ordered non-ordered delivery

o flow control (slow receiver function)

o drop notification

o timestamps
o message-oriented delivery

o partial integrity protection

3.7. Realtime Transport

3.6. Datagram Congestion Control Protocol (RTP)

RTP provides (DCCP)

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

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

3.8. NACK-Oriented Reliable Multicast (NORM)

NORM is an IETF standards track protocol specified in [RFC5740].

The
protocol was designed to support reliable bulk data dissemination DCCP Problem Statement describes the goals that DCCP sought to
address [RFC4336]. It is suitable for applications that transfer
fairly large amounts of data and that can benefit from control over
the trade off between timeliness and reliability [RFC4336].

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

3.6.1. Protocol Description

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

It provides multiplexing to multiple sockets on each host using port
numbers. An active DCCP session is identified by its four-tuple of
local and remote IP addresses and local port and remote port numbers.
At connection setup, DCCP also exchanges the the service code
[RFC5595] mechanism to allow transport instantiations to indicate the
service treatment that is expected from the network.

The protocol segments data into messages, typically sized to fit in
IP packets, but which may be fragmented providing they are less than
the A DCCP interface MAY allow applications to request fragmentation
for packets larger than PMTU, but not larger than the maximum packet
size allowed by the current congestion control mechanism (CCMPS)
[RFC4340].

Each message is identified by a sequence number. The sequence number
is used to identify segments in acknowledgments, to detect
unacknowledged segments, to measure RTT, etc. The protocol may
support ordered or unordered delivery of data, and does not itself
provide retransmission. There is a Data Checksum option, which
contains a strong CRC, lets endpoints detect application data
corruption. It also supports reduced checksum coverage, a partial
integrity mechanisms similar to UDP-lIte.

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

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

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

DCCP use a Connect packet to start a session, and permits half-
connections that allow each client to choose features it wishes to
support. Simultaneous open [RFC5596], as in TCP, can enable
interoperability in the presence of middleboxes. The Connect packet
includes a Service Code field [RFC5595] designed to allow middle
boxes and endpoints to identify the characteristics required by a
session. A lightweight UDP-based encapsulation (DCCP-UDP) has been
defined [RFC6773] that permits DCCP to be used over paths where it is
not natively supported. Support in NAPT/NATs is defined in [RFC4340]
and [RFC5595].

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

A DCCP service is unicast.

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

3.6.2. Interface Description

There is no current API specified in the RFC Series.

3.6.3. Transport Protocol Components

The transport protocol components provided by DCCP are:

o unicast

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

o Service Codes

o port multiplexing

o non-reliable, ordered delivery

o flow control (slow receiver function)

o drop notification

o timestamps

o message-oriented delivery

o partial integrity protection

3.7. Realtime Transport Protocol (RTP)

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

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

3.8. NACK-Oriented Reliable Multicast (NORM)

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

3.8.1. Protocol Description

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

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

NORM supports bulk data dissemination
includes discrete "objects" such as file or computer memory-based "objects" in-memory content but
also can treat a stream of data as well a logical bulk object for purposes
of packet erasure coding. In the case of stream transport, NORM can
support either byte streams or message streams where application-
defined message boundary information is carried in the NORM protocol
messages. This allows the receiver(s) to join/re-join and recover
message boundaries mid-stream as
byte- needed. Application content is
carried and message-streaming. identified by the NORM protocol with encoding symbol
identifiers depending upon the Forward Error Correction (FEC) Scheme
[RFC3452] configured. NORM uses NACK-based selective ARQ to reliably
deliver the application content to the receiver(s). NORM proactively
measures round-trip timing information to scale ARQ timers
appropriately and to support congestion control. For multicast
operation, timer-based feedback suppression is uses to achieve group
size scaling with low feedback traffic levels. The feedback
suppression is not applied for unicast operation.

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

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

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

3.8.1. Protocol Description

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

3.8.2. Interface Description

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

3.8.3. Transport Protocol Components

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

o unicast

o multicast

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

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

o unordered delivery of in-memory data or file bulk content but
also can treat objects

o error detection (UDP checksum)

o segmentation

o stream-oriented delivery in a single stream

o object-oriented delivery of discrete data or file items

o data bundling (Nagle's algorithm)

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

o congestion control

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

3.9. Transport Layer Security (TLS) and Datagram TLS (DTLS) as a logical bulk object for purposes
pseudotransport

Transport Layer Security (TLS) and Datagram TLS are IETF protocols
that provide several security-related features to applications. TLS
is designed to run on top of packet erasure coding. In TCP, DTLS is designed to run on top of
UDP. At the case time of stream transport, NORM can
support either byte streams or message streams where application- writing, the current version of TLS is 1.2; it
is defined message boundary information in [RFC5246]. DTLS provides nearly identical
functionality; it is carried defined in the NORM protocol
messages. This allows the receiver(s) to join/re-join {RFC6347}} and recover
message boundaries mid-stream as needed. Application content also at version 1.2.

While older versions of TLS and DTLS are still in use, they provide
weaker security guarantees. [RFC7457] outlines important attacks on
TLS and DTLS. [RFC7525] is
carried a Best Current Practices (BCP) document
that describes secure configurations for TLS and identified by the NORM protocol with encoding symbol
identifiers depending upon the Forward Error Correction (FEC) Scheme
[RFC3452] configured. NORM uses NACK-based selective ARQ DTLS to reliably
deliver counter
these attacks. The recommendations are applicable for the application content to vast
majority of use cases.

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

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

3.9.1. Protocol Description

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

o Confidentiality

o Data integrity

o Data authenticity

o Optionally authentication of the peer entity

[Note: Both TLS and DTLS provide replay protection, although it is
optional in DTLS. The TLS RFC discusses this only in the security
considerations and thus views it as a feature that is functionally
equivalent to TFRC implicit in the unicast
ones listed above. DTLS mentions it as an explicit feature.]

The authentication of the peer entity can be omitted, although this
is a rare use case.

NORM's reliability mechanism In many use cases (e.g. the Web), authentication
is not mutual, however (e.g. only the Web server is authenticated,
but not the client). It is important to note that TLS itself does
not specify how a peering entity is decoupled from congestion control.
This allows alternative arrangements of transport services to be
invoked. For example, fixed-rate reliable delivery can be supported
or unreliable (but optionally "better than best effort" via packet
erasure coding) delivery authenticated. This is
part of the application logic; i.e. the authentication decision rests
with rate-control per TFRC can be achieved.
Additionally, alternative congestion control techniques may be
applied. For the application. As an example, TFRC rate control with congestion event
detection based on ECN for links with high packet loss (e.g.,
wireless) has been implemented and demonstrated with NORM.

While NORM in the common use case of
authentication by means of an X.509 certificate, it is NACK-based the
application's decision whether the certificate of the peering entity
is acceptable for reliability transfer, it also supports a
positive acknowledgment (ACK) mechanism that can the purposes of the application or whether the
handshake should be used for receiver
flow control. Again, since this mechanism aborted.

As DTLS is decoupled from used over the
reliability and congestion control, applications that have different unreliable UDP transport, it needs in to add
three features to provide the same security guarantees as TLS: *
Message fragmentation * Message reordering * Message loss

As a result, DTLS provides features that UDP lacks.

[EDITOR'S NOTE: Need to describe how this aspect can use the protocol differently. One example is the use of NORM for quasi-reliable delivery where timely delivery
of newer content may be favored over completely reliable delivery of
older content within buffering and RTT constraints.

3.8.2. achieved?]

3.9.2. Interface Description

The NORM specification does not describe a specific application
programming interface (API) to control protocol operation. A freely-
available, open source reference implementation of NORM

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

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

3.8.3. choose safe settings here. [RFC7525] provides guidance.

[TODO: list the RFCs?] [TODO: more detail?] ### Transport Protocol
Components

Both TLS and DTLS employ a layered architecture. The transport protocol components provided by NORM are:

o unicast

o multicast

o port multiplexing (UDP ports)

o reliable delivery

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

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

o error detection (UDP checksum)

o segmentation

o stream-oriented delivery in a single stream

o object-oriented delivery of discrete data or file items

o data bundling (Nagle's algorithm)

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

o congestion control

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

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

[NOTE: A few words on secure
session.

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

3.9.1. Protocol Description

3.9.2. Interface Description

3.9.3. Transport Protocol Components weakness. It is not described here.]

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

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

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

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

3.10.1. Protocol Description

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

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

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

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

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

3.10.2. Interface Description

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

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

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

3.10.3. Transport Protocol Components

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

o unicast

o reliable delivery

o ordered delivery

o message and stream-oriented

o object range request

o message content type negotiation

o congestion control

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

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

o confidentiality
o integrity protection

3.11. WebSockets

[RFC6455]

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

3.11.1. Protocol Description

3.11.2. Interface Description

3.11.3. Transport Protocol Components

4. Transport Service Features

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

o Destination selection

* unicast

* broadcast (IPv4 only)

* multicast

* anycast

* transport layer multihoming for resilience

* transport layer mobility

* port multiplexing

* service codes

o Connection setup

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

o Delivery

* reliable delivery
* partially reliable delivery

* unreliable delivery

* packet erasure coding

* ordered delivery

* unordered delivery

* stream-oriented delivery

* message-oriented delivery

* message fragmentation

* object-oriented delivery of discrete data or file items

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

* object range request

* object content type negotiation

* single streaming

* multiple streaming

* stream scheduling prioritization

* segmentation

* data bundling (Nagle's algorithm)

* message bundling

o Transmission control

* timer-based rate control

* ack-based flow control

* drop notification

* packet erasure coding

* congestion control

o Integrity protection

* checksum for error detection

* partial checksum protection

* checksum optional

* cryptographic integrity protection

o Security

* authentication of one end of a connection

* authentication of both ends of a connection

* confidentiality

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

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

4.1. Complete Protocol Feature Matrix

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

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

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

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

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

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

5. IANA Considerations

This document has no considerations for IANA.

6. Security Considerations

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

7. Contributors

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

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

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

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

o Section 3.8 on NORM was contributed by Brian Adamson
(brian.adamson@nrl.navy.mil)

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

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

8. Acknowledgments

Thanks to Karen Nielsen, Joe Touch, and Michael Welzl for the
comments, feedback, and discussion. This work is partially supported
by the European Commission under grant agreement FP7-ICT-318627
mPlane; support does not imply endorsement.

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

9. References

9.1. Normative References

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

9.2. Informative References

[RFC0768] Postel, J., "User Datagram Protocol", STD 6, RFC 768,
August 1980.

[RFC0793] Postel, J., "Transmission Control Protocol", STD 7, RFC
793, September 1981.

[RFC0896] Nagle, J., "Congestion control in IP/TCP internetworks",
RFC 896, January 1984.

[RFC1122] Braden, R., "Requirements for Internet Hosts -
Communication Layers", STD 3, RFC 1122, October 1989.

[RFC1191] Mogul, J. and S. Deering, "Path MTU discovery", RFC 1191,
November 1990.

[RFC1981] McCann, J., Deering, S., and J. Mogul, "Path MTU Discovery
for IP version 6", RFC 1981, August 1996.

[RFC2018] Mathis, M., Mahdavi, J., Floyd, S., and A. Romanow, "TCP
Selective Acknowledgment Options", RFC 2018, October 1996.

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

[RFC2460] Deering, S. and R. Hinden, "Internet Protocol, Version 6
(IPv6) Specification", RFC 2460, December 1998.

[RFC2617] Franks, J., Hallam-Baker, P., Hostetler, J., Lawrence, S.,
Leach, P., Luotonen, A., and L. Stewart, "HTTP
Authentication: Basic and Digest Access Authentication",
RFC 2617, June 1999.

[RFC3168] Ramakrishnan, K., Floyd, S., and D. Black, "The Addition
of Explicit Congestion Notification (ECN) to IP", RFC
3168, September 2001.

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

[RFC4960] Stewart, R., "Stream Control Transmission Protocol", RFC
4960, September 2007.

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

[RFC5097] Renker, G. and G. Fairhurst, "MIB for the UDP-Lite
protocol", RFC 5097, January 2008.

[RFC5246] Dierks, T. and E. Rescorla, "The Transport Layer Security
(TLS) Protocol Version 1.2", RFC 5246, August 2008.

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

[RFC5405] Eggert, L. and G. Fairhurst, "Unicast UDP Usage Guidelines
for Application Designers", BCP 145, RFC 5405, November
2008.

[RFC5595] Fairhurst, G., "The Datagram Congestion Control Protocol
(DCCP) Service Codes", RFC 5595, September 2009.

[RFC5596] Fairhurst, G., "Datagram Congestion Control Protocol
(DCCP) Simultaneous-Open Technique to Facilitate NAT/
Middlebox Traversal", RFC 5596, September 2009.

[RFC5662] Shepler, S., Eisler, M., and D. Noveck, "Network File
System (NFS) Version 4 Minor Version 1 External Data
Representation Standard (XDR) Description", RFC 5662,
January 2010.

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

[RFC5740] Adamson, B., Bormann, C., Handley, M., and J. Macker,
"NACK-Oriented Reliable Multicast (NORM) Transport
Protocol", RFC 5740, November 2009.

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

[RFC5925] Touch, J., Mankin, A., and R. Bonica, "The TCP
Authentication Option", RFC 5925, June 2010.

[RFC5681] Allman, M., Paxson, V., and E. Blanton, "TCP Congestion
Control", RFC 5681, September 2009.

[RFC6083] Tuexen, M., Seggelmann, R., and E. Rescorla, "Datagram
Transport Layer Security (DTLS) for Stream Control
Transmission Protocol (SCTP)", RFC 6083, January 2011.

[RFC6093] Gont, F. and A. Yourtchenko, "On the Implementation of the
TCP Urgent Mechanism", RFC 6093, January 2011.

[RFC6525] Stewart, R., Tuexen, M., and P. Lei, "Stream Control
Transmission Protocol (SCTP) Stream Reconfiguration", RFC
6525, February 2012.

[RFC6546] Trammell, B., "Transport of Real-time Inter-network
Defense (RID) Messages over HTTP/TLS", RFC 6546, April
2012.

[RFC6298] Paxson, V., Allman, M., Chu, J., and M. Sargent,
"Computing TCP's Retransmission Timer", RFC 6298, June
2011.

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

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

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

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

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

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

[RFC6458] Stewart, R., Tuexen, M., Poon, K., Lei, P., and V.
Yasevich, "Sockets API Extensions for the Stream Control
Transmission Protocol (SCTP)", RFC 6458, December 2011.

[RFC6691] Borman, D., "TCP Options and Maximum Segment Size (MSS)",
RFC 6691, July 2012.

[RFC6824] Ford, A., Raiciu, C., Handley, M., and O. Bonaventure,
"TCP Extensions for Multipath Operation with Multiple
Addresses", RFC 6824, January 2013.

[RFC6897] Scharf, M. and A. Ford, "Multipath TCP (MPTCP) Application
Interface Considerations", RFC 6897, March 2013.

[RFC6935] Eubanks, M., Chimento, P., and M. Westerlund, "IPv6 and
UDP Checksums for Tunneled Packets", RFC 6935, April 2013.

[RFC6936] Fairhurst, G. and M. Westerlund, "Applicability Statement
for the Use of IPv6 UDP Datagrams with Zero Checksums",
RFC 6936, April 2013.

[RFC6951] Tuexen, M. and R. Stewart, "UDP Encapsulation of Stream
Control Transmission Protocol (SCTP) Packets for End-Host
to End-Host Communication", RFC 6951, May 2013.

[RFC7053] Tuexen, M., Ruengeler, I., and R. Stewart, "SACK-
IMMEDIATELY Extension for the Stream Control Transmission
Protocol", RFC 7053, November 2013.

[RFC7230] Fielding, R. and J. Reschke, "Hypertext Transfer Protocol
(HTTP/1.1): Message Syntax and Routing", RFC 7230, June
2014.

[RFC7231] Fielding, R. and J. Reschke, "Hypertext Transfer Protocol
(HTTP/1.1): Semantics and Content", RFC 7231, June 2014.

[RFC7232] Fielding, R. and J. Reschke, "Hypertext Transfer Protocol
(HTTP/1.1): Conditional Requests", RFC 7232, June 2014.

[RFC7233] Fielding, R., Lafon, Y., and J. Reschke, "Hypertext
Transfer Protocol (HTTP/1.1): Range Requests", RFC 7233,
June 2014.

[RFC7234] Fielding, R., Nottingham, M., and J. Reschke, "Hypertext
Transfer Protocol (HTTP/1.1): Caching", RFC 7234, June
2014.

[RFC7235] Fielding, R. and J. Reschke, "Hypertext Transfer Protocol
(HTTP/1.1): Authentication", RFC 7235, June 2014.

[RFC7301] Friedl, S., Popov, A., Langley, A., and E. Stephan,
"Transport Layer Security (TLS) Application-Layer Protocol
Negotiation Extension", RFC 7301, July 2014.

[RFC7323] Borman, D., Braden, B., Jacobson, V., and R.
Scheffenegger, "TCP Extensions for High Performance", RFC
7323, September 2014.

[RFC7457] Sheffer, Y., Holz, R., and P. Saint-Andre, "Summarizing
Known Attacks on Transport Layer Security (TLS) and
Datagram TLS (DTLS)", RFC 7457, February 2015.

[RFC7525] Sheffer, Y., Holz, R., and P. Saint-Andre,
"Recommendations for Secure Use of Transport Layer
Security (TLS) and Datagram Transport Layer Security
(DTLS)", BCP 195, RFC 7525, May 2015.

[RFC7540] Belshe, M., Peon, R., and M. Thomson, "Hypertext Transfer
Protocol Version 2 (HTTP/2)", RFC 7540, May 2015.

[I-D.ietf-aqm-ecn-benefits]
Welzl, M. and G. Fairhurst, "The Benefits and Pitfalls of
using Explicit Congestion Notification (ECN)", draft-ietf-
aqm-ecn-benefits-00 (work in progress), October 2014.

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

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

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

[I-D.ietf-tsvwg-natsupp]
Stewart, R., Tuexen, M., and I. Ruengeler, "Stream Control
Transmission Protocol (SCTP) Network Address Translation
Support", draft-ietf-tsvwg-natsupp-07 (work in progress),
February 2015.

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

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

Authors' Addresses

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

Email: gorry@erg.abdn.ac.uk

Brian Trammell (editor)
ETH Zurich
Gloriastrasse 35
8092 Zurich
Switzerland

Email: ietf@trammell.ch

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

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