wdiff draft-ietf-taps-transports-07.txt draft-ietf-taps-transports-08.txt

Network Working Group G. Fairhurst, Ed.
Internet-Draft University of Aberdeen
Intended status: Informational B. Trammell, Ed.
Expires: April 9, June 10, 2016 M. Kuehlewind, Ed.
ETH Zurich
October 07,
December 08, 2015

Services provided by IETF transport protocols and congestion control
mechanisms
draft-ietf-taps-transports-07
draft-ietf-taps-transports-08

Abstract

This document describes transport services provided by existing IETF protocols
and congestion control mechanisms.
protocols. 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 Service Features . . . . . . . . . . . . . . . . 5 . 4
3.1. Transport Congestion Control Protocol (TCP) . . . . . . . . . . . . 5
3.1.1. Protocol Description . . . . . . . . . . . . . . . . 5
3.1.2. Interface description
4. Existing Transport Protocols . . . . . . . . . . . . . . . . 6
3.1.3.
4.1. Transport Features . . . . . . . Control Protocol (TCP) . . . . . . . . . . 7
3.2. Multipath TCP (MPTCP) . . 6
4.1.1. Protocol Description . . . . . . . . . . . . . . . . 8
3.2.1. Protocol Description 6
4.1.2. Interface description . . . . . . . . . . . . . . . . 8
3.2.2. Interface Description . . . . .
4.1.3. Transport Features . . . . . . . . . . . 8
3.2.3. Transport features . . . . . . 8
4.2. Multipath TCP (MPTCP) . . . . . . . . . . . 8
3.3. Stream Control Transmission Protocol (SCTP) . . . . . . . 9
3.3.1.
4.2.1. Protocol Description . . . . . . . . . . . . . . . . 9
3.3.2.
4.2.2. Interface Description . . . . . . . . . . . . . . . . 12
3.3.3. 9
4.2.3. Transport Features features . . . . . . . . . . . . . . . . . 14
3.4. User Datagram 10
4.3. Stream Control Transmission Protocol (UDP) . . . . . . . (SCTP) . . . . . . . 15
3.4.1. 10
4.3.1. Protocol Description . . . . . . . . . . . . . . . . 15
3.4.2. 11
4.3.2. Interface Description . . . . . . . . . . . . . . . . 16
3.4.3. 13
4.3.3. Transport Features . . . . . . . . . . . . . . . . . 16
3.5. Lightweight 15
4.4. User Datagram Protocol (UDP-Lite) (UDP) . . . . . . 17
3.5.1. . . . . . . . . 16
4.4.1. Protocol Description . . . . . . . . . . . . . . . . 17
3.5.2. 16
4.4.2. Interface Description . . . . . . . . . . . . . . . . 18
3.5.3. 17
4.4.3. Transport Features . . . . . . . . . . . . . . . . . 18
3.6.
4.5. Lightweight User Datagram Congestion Control Protocol (DCCP) . (UDP-Lite) . . . . . . 19
3.6.1. 18
4.5.1. Protocol Description . . . . . . . . . . . . . . . . 19
3.6.2. 18
4.5.2. Interface Description . . . . . . . . . . . . . . . . 20
3.6.3. 19
4.5.3. Transport Features . . . . . . . . . . . . . . . . . 21
3.7. Lightweight User 19
4.6. Datagram Congestion Control Protocol (UDP-Lite) (DCCP) . . . . . . 21
3.7.1. . 20
4.6.1. Protocol Description . . . . . . . . . . . . . . . . 21
3.7.2. 20
4.6.2. Interface Description . . . . . . . . . . . . . . . . 22
3.7.3. 21
4.6.3. Transport Features . . . . . . . . . . . . . . . . . 22
3.8.
4.7. Internet Control Message Protocol (ICMP) . . . . . . . . 23
3.8.1. 22
4.7.1. Protocol Description . . . . . . . . . . . . . . . . 23
3.8.2.
4.7.2. Interface Description . . . . . . . . . . . . . . . . 24
3.8.3.
4.7.3. Transport Features . . . . . . . . . . . . . . . . . 24
3.9.
4.8. Realtime Transport Protocol (RTP) . . . . . . . . . . . . 25
3.9.1. 24
4.8.1. Protocol Description . . . . . . . . . . . . . . . . 25
3.9.2. 24
4.8.2. Interface Description . . . . . . . . . . . . . . . . 26
3.9.3. 25
4.8.3. Transport Features . . . . . . . . . . . . . . . . . 26
3.10.
4.9. File Delivery over Unidirectional Transport/Asynchronous
Layered Coding Reliable Multicast (FLUTE/ALC) . . . . . . 26
3.10.1.
4.9.1. Protocol Description . . . . . . . . . . . . . . . . 27
3.10.2.
4.9.2. Interface Description . . . . . . . . . . . . . . . . 29
3.10.3.
4.9.3. Transport Features . . . . . . . . . . . . . . . . . 29
3.11.
4.10. NACK-Oriented Reliable Multicast (NORM) . . . . . . . . . 30
3.11.1.
4.10.1. Protocol Description . . . . . . . . . . . . . . . . 30
3.11.2.
4.10.2. Interface Description . . . . . . . . . . . . . . . 31
3.11.3.
4.10.3. Transport Features . . . . . . . . . . . . . . . . . 32
3.12. 31
4.11. Transport Layer Security (TLS) and Datagram TLS (DTLS) as
a pseudotransport . . . . . . . . . . . . . . . . . . . . 32
3.12.1.
4.11.1. Protocol Description . . . . . . . . . . . . . . . . 33
3.12.2. 32
4.11.2. Interface Description . . . . . . . . . . . . . . . 34
3.12.3. 33
4.11.3. Transport Features . . . . . . . . . . . . . . . . . 34
3.13.
4.12. Hypertext Transport Protocol (HTTP) over TCP as a
pseudotransport . . . . . . . . . . . . . . . . . . . . . 35
3.13.1.
4.12.1. Protocol Description . . . . . . . . . . . . . . . . 36
3.13.2. 35
4.12.2. Interface Description . . . . . . . . . . . . . . . 37
3.13.3. 36
4.12.3. Transport features . . . . . . . . . . . . . . . . . 37
4.
5. Transport Service Features . . . . . . . . . . . . . . . . . 38
4.1. Complete Protocol Feature Matrix . . . . . . . . . . . . 40
5. 37
6. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 42
6. 41
7. Security Considerations . . . . . . . . . . . . . . . . . . . 42
7. 41
8. Contributors . . . . . . . . . . . . . . . . . . . . . . . . 42
8. 41
9. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 43
9. 42
10. Informative References . . . . . . . . . . . . . . . . . . . 43 42
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 52

1. Introduction

Most

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

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

[GF: Ledbat is a mechanism, not protocol - hence use the work
"support"

2. Terminology

The following terms are defined throughout this document, and in para below.]

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

2. Terminology

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

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

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

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

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

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

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

Transport Service Instance: an arrangement of transport protocols
with a selected set of features and configuration parameters that
implements a single transport service, e.g. 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.

3.1. Service Features

Transport Control Protocol (TCP)

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

One fundamental feature is an IETF standards track whether a transport protocol. [RFC0793]
introduces TCP as follows: "The Transmission Control Protocol (TCP)
is intended for use offers a service that
divides the data into transmission units based on network packets
(known as a highly reliable host-to-host protocol
between hosts in packet-switched computer communication networks, Datagram service), or whether it combines and
in interconnected systems of such networks." Since its introduction,
TCP has become the default connection-oriented, stream-based
transport protocol in segments
data across multiple packets (e.g., the Internet. It is widely implemented by
endpoints and widely used Stream service provided by common applications.

3.1.1. Protocol Description

TCP
TCP).

Another fundamental feature is whether a connection-oriented protocol, providing transport requires a three way
handshake to allow control
exchange across the network at setup (e.g., TCP), or whether it
connection-less (e.g., UDP).

A transport service can also offer reliability, for instance, SCTP
offers a client message-based service providing full or partial reliability
and server allowing to set up a connection minimize the head of line blocking due to the support
of unordered and
negotiate features, unordered message delivery within multiple streams,
UDP-Lite and mechanisms DCCP provide partial integrity protection.

A transport service can provide congestion control (see Section 3.1).
TCP and SCTP provide congestion control for orderly completion use in the Internet,
whereas UDP leaves this function to the upper layer protocol that
uses UDP. DCCP offers a range of congestion control approaches and
immediate teardown
LEDBAT can support low-priority "scavenger" communication, intending
to defer use of capacity to other Internet flows sharing a connection. TCP is defined by congested
bottleneck.

Transport services may be unidirectional or bidirectional, to a family
single a single endpoint, to one of
RFCs [RFC4614].

TCP provides multiplexing multiple endpoints, or multicast
simultaneously to multiple sockets on each host using port
numbers.] A similar approach is adopted endpoints.

The service offered by transport protocols and frameworks can also be
differentiated in many other IETF-defined
transports. An active TCP session ways.

3.1. Congestion Control

Congestion control is identified by its four-tuple critical to the stable operation of
local and remote IP addresses and local port the
Internet, applications and remote port numbers.
The destination port during connection setup is often used other protocols that choose to
indicate the requested service.

TCP partitions use a continuous stream
datagram protocol (e.g., UDP or UDP-Lite) need to employ mechanisms
to prevent congestion collapse and to establish some degree of bytes into segments, sized
fairness with concurrent traffic.

A variety of techniques are used to
fit provide congestion control in IP packets. ICMP-based PathMTU discovery [RFC1191][RFC1981]
as well as Packetization Layer Path MTU Discovery (PMTUD) [RFC4821]
are supported. the
Internet. Each byte in technique requires that the stream is identified by protocol provide a sequence number. The
sequence number is used to order segments on receipt, to identify
segments in acknowledgments, and to detect unacknowledged segments method
for retransmission. This is deriving the basis of metric the reliable, ordered
delivery congestion control algorithm uses to
detect congestion and the property of data in a TCP stream. TCP Selective Acknowledgment
[RFC2018] extends this mechanism by making packet it possible uses to identify
missing segments more precisely, reducing spurious retransmission.

Receiver flow control is provided by a sliding window: limiting determine
when to send. Given these relatively wide constraints, the
amount of unacknowledged data
congestion control techniques that can be outstanding at a given
time. The window scale option [RFC7323] allows a receiver applied by different
transport protocols are largely orthogonal to use
windows greater than 64KB.

All TCP senders provide Congestion Control [RFC5681]: the choice of transport
protocols themselves. This uses a
separate window, where each time congestion is detected, this
congestion window is reduced. Most section provides an overview of the used
congestion control techniques available to the protocols described in
Section 4.

Most commonly deployed congestion control mechanisms use one of three
mechanisms to detect congestion: A
retransmission timer (with exponential back-up),

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

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

o a congestion retransmission timer with exponential back-off.

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

A TCP protocol instance can be extended [RFC4614] datagram congestion control.

Rate-based methods have also been defined based on the loss ratio and tuned. Some
features are sender-side only, requiring no negotiation with the
receiver; some are receiver-side only, some are explicitly negotiated
during connection setup.

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

TCP provides a push
observed round trip time, such as TFRC [RFC5348] and TFRC-SP
[RFC4828]. These methods utlise a urgent function to enable data throughput equation to be
directly accessed by determine
the receiver wihout having maximum acceptable rate. Such methods are used with DCCP CCID-3
[RFC4342] and CCID-4 [RFC5622], WEBRC [RFC3738], and other
applications.

In addition, a congestion control mechanism may react to wait changes in
delay as an indication for in-order
delivery of the data. However, [RFC6093] does not recommend the use
of the urgent flag due congestion. Delay-based congestion
detection methods tend to the range of TCP implementations that
process TCP urgent indications differently.

A checksum provides an Integrity Check induce less loss than loss-based methods,
and is mandatory therefore generally do not compete well with them across shared
bottleneck links. However, such methods, such as LEDBAT [RFC6824],
are are deployed in the
entire packet. Internet for scavenger traffic, which will
use unused capacity but readily yield to presumably interactive or
otherwise higher-priority, loss-based congestion-controlled traffic.

4. Existing Transport Protocols

This check protects from delivery section provides a list of corrupted data known IETF transport protocols and miselivery of packets to the wrong endpoint. This check is
relatively weak, applications that require end to end integrity
transport protocol frameworks. It does not make an assessment about
whether specific implementations of
data protocols are recommended fully compliant to include a stronger integrity check of their
payload data. The
current IETF specifications.

4.1. Transport Control Protocol (TCP)

TCP checksum does not support partial corruption
protection (as in DCCP/UDP-Lite).

TCP only supports unicast connections.

3.1.2. Interface description

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

[RFC1122] describes extensions of the TCP/application layer interface
for 1) reporting soft errors such as reception fo ICMP error
messages, extensive retransmission or urgent pointer advance, 2)
providing a possibility to specify the Type-of-Service (TOS) follows: "The Transmission Control Protocol (TCP)
is intended for
segments, 3) providing use as a fush call to empty the TCP send queue, highly reliable host-to-host protocol
between hosts in packet-switched computer communication networks, and
4) multihoming support.

In API implementations derived from the BSD Sockets API,
in interconnected systems of such networks." Since its introduction,
TCP sockets
are created using has become the "SOCK_STREAM" socket type as described default connection- oriented, stream-based
transport protocol in the
IEEE Portable Operating System Interface (POSIX) Base Specifications
[POSIX]. The features Internet. It is widely implemented by
endpoints and widely used by common applications.

4.1.1. Protocol Description

TCP is a protocol instance may be connection-oriented protocol, providing a three way
handshake to allow a client and server to set up a connection and
tuned via this API. However, there
negotiate features, and mechanisms for orderly completion and
immediate teardown of a connection. TCP is no RFC that documents this
interface.

3.1.3. Transport Features

The transport features provided defined by TCP are:

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

o unicast transport

o connection setup with feature negotiation and application-to-port
mapping, implemented using SYN segments and the
RFCs [RFC4614].

TCP option field provides multiplexing to negotiate features.

o port multiplexing: multiple sockets on each host using port
numbers. A similar approach is adopted by other IETF-defined
transports. An active TCP session is uniquely identified by a
combination its four-tuple of the ports
local and remote IP address fields.

o Uni-or bidirectional communication

o stream-oriented delivery in addresses and local port and remote port numbers.
The destination port during connection setup is often used to
indicate the requested service.

TCP partitions a single continuous stream

o fully reliable delivery, implemented using ACKs sent from the
receiver of bytes into segments, sized to confirm delivery.

o error detection:
fit in IP packets. ICMP-based Path MTU discovery [RFC1191][RFC1981]
as well as Packetization Layer Path MTU Discovery (PMTUD) [RFC4821]
have been defined by the IETF.

Each byte in the stream is identified by a segment checksum verifies delivery sequence number. The
sequence number is used to the
correct endpoint order segments on receipt, to identify
segments in acknowledgments, and integrity to detect unacknowledged segments
for retransmission. This is the basis of the reliable, ordered
delivery of data and options.

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

o data bundling, an optional TCP stream. TCP Selective Acknowledgment
[RFC2018] extends this mechanism that uses Nagle's algorithm by making it possible to coalesce data sent within the same RTT into full-sized
segments.

o identify
missing segments more precisely, reducing spurious retransmission.

Receiver flow control using is provided by a window-based mechanism, where the receiver
advertises sliding window: limiting the window
amount of unacknowledged data that it is willing to buffer.

o congestion control: can be outstanding at a window-based method that uses AIMD to
control the sending rate and to conservatively choose given
time. The window scale option [RFC7323] allows a rate after
congestion is detected.

3.2. Multipath TCP (MPTCP)

Multipath receiver to use
windows greater than 64KB.

TCP [RFC6824] is an extension for provides congestion control [RFC5681], described further in
Section 3.1 below.

TCP to support multi-
homing. It is designed to protocol instances can be as transparent as possible to middle-
boxes. It does so by establishing regular TCP flows between a pair
of source/destination endpoints, extended [RFC4614] and multiplexing tuned. Some
features are sender-side only, requiring no negotiation with the application's
stream over these flows.

3.2.1. Protocol Description

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

3.2.2. Interface Description receiver-side only, some are explicitly negotiated
during connection setup.

By default, MPTCP exposes the same interface as TCP segment partitioning uses Nagle's algorithm [RFC0896]
to buffer data at the
application. [RFC6897] however describes a richer API for MPTCP-
aware applications.

This Basic API describes how an application sender into large segments, potentially
incurring sender-side buffering delay; this algorithm can

o enable or disable MPTCP;

o bind a socket be disabled
by the sender to one or transmit more selected local endpoints;

o query local and remote endpoint addresses;

o get a unique connection identifier (similar immediately, e.g., to reduce latency
for interactive sessions.

TCP provides an address-port
pair "urgent data" function for TCP).

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

3.2.3. Transport features

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

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

o congestion control with load balancing over mutiple connections.

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

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

o resilience to network failure and/or handover.

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

3.3. Stream Control Transmission Protocol (SCTP)

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

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

3.3.1. Protocol Description

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

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

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

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

[RFC4960] specifies a TCP friendly congestion control to protect the
network against overload. SCTP also uses a sliding window flow
control to protect receivers against overflow. Similar to TCP, SCTP
also supports delaying acknowledgements. [RFC7053] deprecated [RFC6093].

A mandatory checksum provides a way
for basic integrity check against
misdelivery and data corruption over the sender entire packet. Applications
that require end to end integrity of user messages data are recommended to request the immediate sending include
a stronger integrity check of
the corresponding acknowledgements.

Each SCTP association has between 1 and 65536 uni-directional streams
in each direction. their payload data. The number of streams can be different TCP checksum
does not support partial corruption protection (as in each
direction. Every user-message DCCP/UDP-Lite).

TCP supports only unicast connections.

4.1.2. Interface description

A User/TCP Interface is sent on a particular stream. User
messages can be sent un-ordered or ordered upon request by the upper
layer. Un-ordered messages can be delivered as soon as they are
completely received. Ordered messages sent on the same stream are
delivered at the receiver in the same order as sent by the sender.
For user messages not requiring fragmentation, this minimises head of
line blocking.

The base protocol defined in [RFC4960] [RFC0793] providing six user
commands: Open, Send, Receive, Close, Status. This interface does
not allow interleaving describe configuration of TCP options or parameters beside use of
user-messages, which results in sending a large message on one stream
can block
the sending PUSH and URGENT flags.

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

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

o providing a possibility to send data from supporting specify the Differentiated Services
Code Point (DSCP) (formerly, the Type-of-Service, TOS) for
segments,

o providing a
variety of scheduling algorithms including priority based methods.
The stream re-configuration extension defined in [RFC6525] allows
streams flush call to be reset during empty the lifetime of an association TCP send queue, and to
increase

o multihoming support.

In API implementations derived from the number of streams, if BSD Sockets API, TCP sockets
are created using the number of streams negotiated "SOCK_STREAM" socket type as described in the SCTP handshake becomes insufficient.

Each user message sent is either delivered to the receiver or,
IEEE Portable Operating System Interface (POSIX) Base Specifications
[POSIX]. The features used by a protocol instance may be set and
tuned via this API. There are current no documents in
case of excessive retransmissions, the association RFC Series
that describe this interface.

4.1.3. Transport Features

The transport features provided by TCP are:

o unicast transport

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

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

o Uni-or bidirectional communication.

o stream-oriented delivery in a
non-graceful way [RFC4960], similar to TCP behaviour. In addition to
this single stream.

o fully reliable transfer, delivery, implemented using ACKs sent from the partial reliability extension [RFC3758]
allows
receiver to confirm delivery.

o error detection: a sender segment checksum verifies delivery to abandon user messages. The application can
specify the policy for abandoning user messages. Examples for these
policies defined in [RFC3758]
correct endpoint and [RFC7496] are:

o Limiting integrity of the time data and options.

o segmentation: packets are fragmented to a user message is dealt with negotiated maximum
segment size, further constrained by the sender. effective MTU from PMTUD.

o Limiting data bundling, an optional mechanism that uses Nagle's algorithm
to coalesce data sent within the number of retransmissions for each fragment of same RTT into full-sized
segments.

o flow control using a user
message. If window-based mechanism, where the number of retransmissions receiver
advertises the window that it is limited to 0, one
gets a service similar willing to UDP. buffer.

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

SCTP supports multi-homing. Each SCTP endpoint window-based method that uses a list of IP-
addresses Additive
Increase Multiplicative Decrease (AIMD) to control the sending
rate and to conservatively choose a single port number. These addresses can rate after congestion is
detected.

4.2. Multipath TCP (MPTCP)

Multipath TCP [RFC6824] is an extension for TCP to support multi-
homing. It is designed to be any
mixture as transparent as possible to middle-
boxes. It does so by establishing regular TCP flows between a pair
of IPv4 and IPv6 addresses. These addresses are negotiated
during the handshake source/destination endpoints, and multiplexing the address re-configuration extension
specified in [RFC5061] in combination with [RFC4895] can be used to
change application's
stream over these addresses in an authenticated way during the livetime of
an SCTP association. This allows flows.

4.2.1. Protocol Description

MPTCP uses TCP options for transport layer mobility.
Multiple addresses its control plane. They are used for improved resilience. If a remote
address becomes unreachable, to
signal multipath capabilities, as well as to negotiate data sequence
numbers, and advertise other available IP addresses and establish new
sessions between pairs of endpoints.

4.2.2. Interface Description

By default, MPTCP exposes the traffic is switched over same interface as TCP to the
application. [RFC6897] however describes a
reachable one, if richer API for MPTCP-
aware applications.

This Basic API describes how an application can:

o enable or disable MPTCP.

o bind a socket to one exists. Each SCTP end-point supervises
continuously the reachability of all peer addresses using or more selected local endpoints.

o query local and remote endpoint addresses.

o get a heartbeat
mechanism.

For securing user messages, unique connection identifier (similar to an address-port
pair for TCP).

The document also recommends the use of TLS over extensions defined for SCTP has been
specified in [RFC3436]. However, this solution does not
[RFC6458] (see next section) to support all
services provided by SCTP (for example un-ordered delivery or partial
reliability), and therefore multihoming.

4.2.3. Transport features

As an extension to TCP, MPTCP provides mostly the use same features. By
establishing multiple sessions between available endpoints, it can
additionally provide soft failover solutions should one of DTLS the paths
become unusable. In addition, by multiplexing one byte stream over SCTP has been
specified
separate paths, it can achieve a higher throughput than TCP in [RFC6083]
certain situations. Note, however, that coupled congestion control
[RFC6356] might limit this benefit to overcome these limitations. maintain fairness to other
flows at the bottleneck. When using
DTLS aggregating capacity over SCTP, multiple
paths, and depending on the application can use almost all services way packets are scheduled on each TCP
subflow, an additional delay and higher jitter might be observed
observed before in-order delivery of data to the applications.

The transport features provided by SCTP.

[I-D.ietf-tsvwg-natsupp] defines methods for endpoints and
middleboxes MPTCP in addition to provide support NAT for SCTP TCP therefore
are:

o congestion control with load balancing over IPv4. For legacy
NAT traversal, [RFC6951] defines the UDP encapsulation multiple connections.

o endpoint multiplexing of SCTP-
packets. Alternatively, SCTP packets a single byte stream (higher throughput).

o address family multiplexing: sub-flows can be encapsulated in DTLS
packets as specified in [I-D.ietf-tsvwg-sctp-dtls-encaps]. The
latter encapsulation is used within the WebRTC context.

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

3.3.2. Interface Description

[RFC4960] defines an abstract API started over IPv4 or
IPv6 for the base protocol. This API
describes the following functions callable by the upper layer of
SCTP: Initialize, Associate, Send, Receive, Receive Unsent Message,
Receive Unacknowledged Message, Shutdown, Abort, SetPrimary, Status,
Change Heartbeat, Request Heartbeat, Get SRTT Report, Set Failure
Threshold, Set same session.

o resilience to network failure and/or handover.

4.3. Stream Control Transmission Protocol Parameters, and Destroy. The following
notifications are provided by the (SCTP)

SCTP stack to the upper layer:
COMMUNICATION UP, DATA ARRIVE, SHUTDOWN COMPLETE, COMMUNICATION LOST,
COMMUNICATION ERROR, RESTART, SEND FAILURE, NETWORK STATUS CHANGE.

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

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

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

[RFC5061]. Furthermore, the primary path.
Furthermore the application has fine control about parameters like
retransmission thresholds, the path supervision parameters, extension specified in [RFC3758]
introduces the
delayed acknowledgement timeout, concept of partial reliability for user messages.

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

4.3.1. Protocol Description

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

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

SCTP has requested them.
These notifications provide Information about status changes of been designed with extensibility in mind. Each SCTP packet
starts with a single common header containing the association port numbers, a
verification tag and each of the peer addresses. In case CRC32c checksum. This common header is
followed by a sequence of send
failures that application can also be notified chunks. Each chunk consists of a type
field, flags, a length field and user messages
can be returned to the application. When sending user messages,
the stream id, a payload protocol identifier, value. [RFC4960] defines how a
receiver processes chunks with an indication
whether ordered delivery is requested or not. These parameters unknown chunk type. The support of
extensions can also be provided on message reception. Additionally negotiated during the SCTP handshake.

SCTP provides a context message-oriented service. Multiple small user
messages can be provided when sending, which can bundled into a single SCTP packet to improve
efficiency. For example, this bundling may be use in case of send
failures. The sending of arbitrary large done by delaying user
messages is
supported.

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

o the SCTP Authentication extension defined Nagle's algorithm used by TCP.
User messages which would result in [RFC4895] allowing to
manage the shared keys, IP packets larger than the HMAC to use, set MTU
will be fragmented at the chunk types which
are only accepted in an authenticated way, sender and get reassembled at the list of
chunks which are accepted by receiver.
There is no protocol limit on the local user message size. ICMP-based path
MTU discovery as specified for IPv4 in [RFC1191] and remote end point for IPv6 in an
authenticated way.

o
[RFC1981] as well as packetization layer path MTU discovery as
specified in [RFC4821] with probe packets using the SCTP Dynamic Address Reconfiguration extension padding chunks
defined in
[RFC5061]. It allows [RFC4820] are supported.

[RFC4960] specifies TCP-friendly congestion control to manually add and delete local addresses protect the
network against overload; see Section 3.1 for more. SCTP associations and also uses
sliding window flow control to protect receivers against overflow.
Similar to TCP, SCTP also supports delaying acknowledgments.
[RFC7053] provides a way for the enabling sender of automatic address
addition and deletion. Furthermore user messages to request
the peer immediate sending of the corresponding acknowledgments.

Each SCTP association has between 1 and 65536 uni-directional streams
in each direction. The number of streams can be given different in each
direction. Every user message is sent on a hint
for choosing its primary path.

For particular stream. User
messages can be sent un- ordered, or ordered upon request by the following SCTP protocol extensions
upper layer. Un-ordered messages can be delivered as soon as they
are completely received. Ordered messages sent on the BSD Sockets API
extension is defined same stream
are delivered at the receiver in the document specifying same order as sent by the
sender. For user messages not requiring fragmentation, this
minimizes head of line blocking.

The base protocol
extensions:

o defined in [RFC4960] does not allow interleaving of
user- messages. Large messages on one stream can therefore block the SCTP Stream Reconfiguration
sending of user messages on other streams.
[I-D.ietf-tsvwg-sctp-ndata] overcomes this limitation. This draft
also specifies multiple algorithms for the sender side selection of
which streams to send data from, supporting a variety of scheduling
algorithms including priority based methods. The stream re-
configuration extension defined in [RFC6525].
The API [RFC6525] allows to trigger the reset operation for incoming and
outgoing streams and the whole association. It provides also a
way to notify be
reset during the lifetime of an association about the corresponding events.
Furthermore the application can and to increase the
number of streams.

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

o the SCTP SACK-IMMEDIATELY extension defined in [RFC7053]. The API
allows the sender of a
handshake becomes insufficient.

Each user message sent is either delivered to request the receiver to
send the corresponding acknowledgement immediately.

o or, in
case of excessive retransmissions, the additional PR-SCTP policies defined association is terminated in [RFC7496]. The API
allows a
non-graceful way [RFC4960], similar to enable/disable TCP behaviour. In addition to
this reliable transfer, the PR-SCTP extension, choose partial reliability extension [RFC3758]
allows a sender to abandon user messages. The application can
specify the PR-SCTP policy for abandoning user messages. Examples of these
policies defined in the document [RFC3758] and provide statistical
information about abandoned messages.

Future documents describing SCTP protocol extensions are expected to
describe [RFC7496] are:

o Limiting the corresponding BSD Sockets API extension in time a "Socket API
Considerations" section.

The SCTP socket API supports two kinds of sockets:

o one-to-one style sockets (by using user message is dealt with by the socket type "SOCK_STREAM"). sender.

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

One-to-one style sockets are number of retransmissions for each fragment of a user
message. If the number of retransmissions is limited to 0, one
gets a service similar to 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, where there is endpoint uses a 1:n relationship between list of IP-
addresses and a single port number. These addresses can be any
mixture of IPv4 and IPv6 addresses. These addresses are negotiated
during the sockets handshake and the SCTP associations.

The SCTP stack address re-configuration extension
specified in [RFC5061] in combination with [RFC4895] can provide information be used to
change these addresses in an authenticated way during the applications about
state changes livetime of the individual paths and the association whenever
they occur. These events
an SCTP association. This allows for transport layer mobility.
Multiple addresses are delivered similar used for improved resilience. If a remote
address becomes unreachable, the traffic is switched over to a
reachable one, if one exists. [I-D.ietf-tsvwg-sctp-failover]
specifies a quicker failover operation reducing the latency of the
failover.

For securing user messages but
are specifically marked as notifications.

New functions have messages, the use of TLS over SCTP has been introduced to
specified in [RFC3436]. However, this solution does not support all
services provided by SCTP, such as un-ordered delivery or partial
reliability. Therefore, the use of multiple
local and remote addresses. Additional SCTP-specific send and
receive calls have DTLS over SCTP has been defined specified
in [RFC6083] to permit SCTP-specific information overcome these limitations. When using DTLS over
SCTP, the application can use almost all services provided by SCTP.

[I-D.ietf-tsvwg-natsupp] defines methods for endpoints and
middleboxes to be snet without using ancillary data in the form of additional
cmsgs. These functions provide support NAT for detecting partial
delivery of user messages and notifications.

The SCTP socket API allows a fine-grained control of over IPv4. For legacy
NAT traversal, [RFC6951] defines the protocol
behaviour through an extensive set UDP encapsulation of socket options.

The SCTP-
packets. Alternatively, SCTP kernel implementations of FreeBSD, Linux and Solaris follow
mostly the packets can be encapsulated in DTLS
packets as specified extension to in [I-D.ietf-tsvwg-sctp-dtls-encaps]. The
latter encapsulation is used within the BSD Sockets WebRTC context.

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

4.3.2. Interface Description

[RFC4960] defines an abstract API for the base
protocol and protocol. This API
describes the corresponding supported protocol extensions.

3.3.3. Transport Features following functions callable by the upper layer of
SCTP: Initialize, Associate, Send, Receive, Receive Unsent Message,
Receive Unacknowledged Message, Shutdown, Abort, SetPrimary, Status,
Change Heartbeat, Request Heartbeat, Get SRTT Report, Set Failure
Threshold, Set Protocol Parameters, and Destroy. The transport features following
notifications are provided by the SCTP are:

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

o unicast.

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

o port multiplexing.

o message-oriented delivery.

o fully reliable or partially reliable delivery.

o ordered and unordered delivery within a stream.

o support for multiple concurrent streams.

o support for stream scheduling prioritization.

o flow control.

o congestion control.

o user message bundling.

o user message fragmentation upper layer:
COMMUNICATION UP, DATA ARRIVE, SHUTDOWN COMPLETE, COMMUNICATION LOST,
COMMUNICATION ERROR, RESTART, SEND FAILURE, NETWORK STATUS CHANGE.

An extension to the BSD Sockets API is defined in [RFC6458] and reassembly.

o strong error/misdelivery detection (CRC32c).

o transport layer multihoming for resilience.
covers:

o transport layer mobility.

3.4. User Datagram Protocol (UDP) the base protocol defined in [RFC4960]. The User Datagram Protocol (UDP) [RFC0768] [RFC2460] is an IETF
standards track transport protocol. It API allows control
over local addresses and port numbers and the primary path.
Furthermore the application has fine control about parameters like
retransmission thresholds, the path supervision parameters, the
delayed acknowledgment timeout, and the fragmentation point. The
API provides a unidirectional,
datagram protocol that preserves message boundaries. It provides
none of the following transport features: error correction,
congestion control, or flow control. It can be used mechanism to send
broadcast datagrams (IPv4) or multicast datagrams (IPv4 and IPv6), in
addition allow the SCTP stack to unicast (and anycast) datagrams. IETF guidance on notify the
use
application about event if the application has requested them.
These notifications provide Information about status changes of UDP is provided in[I-D.ietf-tsvwg-rfc5405bis]. UDP is widely
implemented
the association and widely used by common applications, each of the peer addresses. In case of send
failures, including DNS.

3.4.1. Protocol Description

UDP is a connection-less protocol that maintains message boundaries,
with no connection setup or feature negotiation. The protocol uses
independent messages, ordinarily called datagrams. Each stream drop of messages is independently managed, therefore retransmission does not
hold back data sent using other logical streams. It provides
detection of payload errors unreliably, the
application can also be notified and misdelivery of packets user messages can be returned
to the wrong
endpoint, either of application. When sending user messages, the stream id, a
payload protocol identifier, an indication whether ordered
delivery is requested or not. These parameters can also be
provided on message reception. Additionally a context can be
provided when sending, which result can be use in discard case of received datagrams.

It is possible to create IPv4 UDP datagrams with no checksum, and
while this send failures.
The sending of arbitrary large user messages is generally discouraged [RFC1122]
[I-D.ietf-tsvwg-rfc5405bis], certain special cases permit its use.
These datagrams relie on the IPv4 header checksum to protect from
misdelivery to supported.

o the wrong endpoint. IPv6 does not by permit UDP
datagrams with no checksum, although SCTP Partial Reliability extension defined in certain cases this rule may
be relaxed [RFC6935]. The checksum support considerations [RFC3758] to
specify for
omitting a user message the checksum are PR-SCTP policy and the policy
specific parameter.

o the SCTP Authentication extension defined in [RFC6936]. Note that due [RFC4895] allowing to
manage the
relatively weak form of checksum used by UDP, applications that
require end shared keys, the HMAC to end integrity of data use, set the chunk types which
are recommended to include a
stronger integrity check only accepted in an authenticated way, and get the list of their payload data.

It does not provide reliability
chunks which are accepted by the local and does not provide retransmission.
This implies messages may be re-ordered, lost, or duplicated remote end point in
transit.

A receiving application that is unable to run sufficiently fast, or
frequently, may miss messages since there is no flow control. The
lack of congestion handling implies UDP traffic may experience loss
when using an overlaoded path
authenticated way.

o the SCTP Dynamic Address Reconfiguration extension defined in
[RFC5061]. It allows to manually add and delete local addresses
for SCTP associations and may cause the loss enabling of messages from
other protocols (e.g., TCP) when sharing automatic address
addition and deletion. Furthermore the same network peer can be given a hint
for choosing its primary path.

[GF: This para isn't needed": Messages with payload errors are
ordinarily detected by an invalid end- to-end checksum and are
discarded before being delivered

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

o the SCTP Stream Reconfiguration extension defined in [RFC6525].
The API allows to an application. UDP-Lite (see
[RFC3828], trigger the reset operation for incoming and
outgoing streams and below) the whole association. It provides also a
way to notify the ability for portions association about the corresponding events.
Furthermore the application can increase the number of streams.

o the
message contents to be exempt from checksum coverage.]

On transmission, UDP encapsulates each datagram into an IP packet,
which may Encapsulation of SCTP packets extension defined in turn be fragmented by IP and are reassembled before
delivery to
[RFC6951]. The API allows the management of the remote UDP receiver.

Applications that need
encapsulation port.

o the SCTP SACK-IMMEDIATELY extension defined in [RFC7053]. The API
allows the sender of a user message to provide fragmentation or that have other
requirements such as request the receiver flow control, congestion control,
PathMTU discovery/PLPMTUD, support for ECN, etc need these to be
provided by protocols operating over UDP [I-D.ietf-tsvwg-rfc5405bis].

3.4.2. Interface Description

[RFC0768] describes basic requirements for an
send the corresponding acknowledgment immediately.

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

Future documents describing SCTP protocol extensions are expected to
describe the corresponding BSD Sockets API for UDP. Guidance
on use of common APIs is provided extension in [I-D.ietf-tsvwg-rfc5405bis].

A UDP endpoint consists of a tuple "Socket API
Considerations" section.

The SCTP socket API supports two kinds of (IP address, port number).
Demultiplexing sockets:

o one-to-one style sockets (by using multiple abstract endpoints (sockets) on the
same IP address are supported. The same socket may be used by a
single server type "SOCK_STREAM").

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

One-to-one style sockets are similar to interact with multiple clients (note: this behavior
differs from TCP, which uses TCP sockets, there is a pair of tuples 1:1
relationship between the sockets and the SCTP associations (except
for listening sockets). One-to-many style SCTP sockets are similar
to identify unconnected UDP sockets, where there is a
connection). Multiple server instances (processes) that bind 1:n relationship between
the
same socket sockets and the SCTP associations.

The SCTP stack can cooperate provide information to service multiple clients- the socket
implementation arranges to not duplicate applications about
state changes of the same received unicast
message to multiple server processes.

Many operating systems also allow a UDP socket to be "connected",
i.e., individual paths and the association whenever
they occur. These events are delivered similar to bind a UDP socket user messages but
are specifically marked as notifications.

New functions have been introduced to a specific (remote) UDP endpoint.
Unlike TCP's connect primitive, for UDP, this is only a support the use of multiple
local
operation that serves and remote addresses. Additional SCTP-specific send and
receive calls have been defined to simplify permit SCTP-specific information
to be sent without using ancillary data in the local send/receive form of additional
cmsgs. These functions provide support for detecting partial
delivery of user messages and notifications.

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

The SCTP kernel implementations of FreeBSD, Linux and Solaris follow
mostly the specified extension to filter the traffic BSD Sockets API for the specified addresses base
protocol and ports
[I-D.ietf-tsvwg-rfc5405bis].

3.4.3. the corresponding supported protocol extensions.

4.3.3. Transport Features

The transport features provided by UDP SCTP are:

o unicast.

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

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

o Uni-or bidirectional communication.

o message-oriented delivery. delivery supporting multiple concurrent streams.

o unidirectional fully reliable, partially reliable, or bidirectional. Transmission in each direction
is independent.

o non-reliable delivery.

o non-ordered unreliable delivery.

o IPv6 jumbograms. ordered and unordered delivery within a stream.

o error user message fragmentation and misdelivery reassembly.

o support for stream scheduling prioritization.

o user message bundling.

o flow control using a window-based mechanism.

o congestion control using methods similar to TCP.

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

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

3.5. Lightweight transport layer multihoming for resilience.

o transport layer mobility.

o resilience to network failure and/or handover.

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

The Lightweight User Datagram Protocol (UDP-Lite) [RFC3828] (UDP) [RFC0768] [RFC2460] is an IETF
standards track transport protocol. It provides a
unidirectional, unidirectional
datagram protocol that preserves message boundaries. It provides no
error correction,congestion control, or flow control. It can be used
to send broadcast datagrams (IPv4) or multicast datagrams (IPv4 and
IPv6), in addition to unicast and anycast datagrams. IETF guidance
on the use of UDP-Lite UDP is provided in
[I-D.ietf-tsvwg-rfc5405bis].

3.5.1. {{I-D.ietf-tsvwg- rfc5405bis}}. UDP
is widely implemented and widely used by common applications,
including DNS.

4.4.1. Protocol Description

UDP-Lite

UDP is a connection-less datagram protocol, protocol that maintains message boundaries,
with no connection setup or feature negotiation. The protocol use uses
independent messages, rather than
a byte-stream. ordinarily called datagrams. Each stream of
messages is independently managed, therefore retransmission does not
hold back data sent using other logical streams. It provides multiplexing to multiple sockets on each host using port
numbers, and its operation follows that for UDP. An active UDP-Lite
session is identified by its four-tuple
detection of local and remote IP
addresses and local port payload errors and remote port numbers.

UDP-Lite changes misdelivery of packets to an
unintended endpoint, either of which result in discard of received
datagrams, with no indication to the semantics user of the service.

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

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

Because UDP provides no flow control, a
different IP protocol/next-header value. Otherwise, UDP-Lite receiving application that is
semantically identical
unable to UDP. Applications run sufficiently fast, or frequently, may miss messages.
The lack of congestion handling implies UDP traffic may experience
loss when using UDP-Lite therefore
can not make assumptions regarding an overloaded path, and may cause the correctness loss of
messages from other protocols (e.g., TCP) when sharing the data
received same
network path.

On transmission, UDP encapsulates each datagram into an IP packet,
which may in turn be fragmented by IP. Fragments are reassembled
before delivery to the insensitive part of the UDP-Lite payload.

As for UDP, mechanisms for UDP receiver.

Applications that need to provide 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 these to be
provided by
upper layer protocols [I-D.ietf-tsvwg-rfc5405bis].

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

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

3.5.2. UDP [I-D.ietf-tsvwg-rfc5405bis].

4.4.2. 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 [I-D.ietf-tsvwg-rfc5405bis].

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) that bind 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 local send/receive functions
and to filter the UDP-Lite MIB module [RFC5097].

3.5.3. traffic for the specified addresses and ports
[I-D.ietf-tsvwg-rfc5405bis].

4.4.3. Transport Features

The transport features provided by UDP-Lite UDP are:

o unicast.

o multicast, anycast, or IPv4 broadcast.

o port multiplexing (as for UDP). multiplexing. A receiving port can be configured to receive
datagrams from multiple senders.

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

o Uni-or bidirectional communication. Transmission in each
direction is independent.

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

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

o error detection: a segment checksum verifies delivery to the
correct endpoint and misdelivery detection (checksum).

o partialor full integrity protection. The of the data. This checksum coverage is
optional for IPv4, and optional under specific conditions for IPv6
where all or none of the payload data is protected.

o IPv6 jumbograms.

4.5. Lightweight User Datagram Protocol (UDP-Lite)

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

4.5.1. Protocol Description

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

3.6. Datagram Congestion Control Protocol (DCCP)

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

The DCCP Problem Statement describes the goals that DCCP sought to
address [RFC4336]. It is suitable UDP-Lite payload.

In the same way as for applications that transfer
fairly large amounts UDP, mechanisms for receiver flow control,
congestion control, PMTU or PLPMTU discovery, support for ECN, etc
need to be provided by upper layer protocols
[I-D.ietf-tsvwg-rfc5405bis].

Examples of data and use include a class of applications that can derive
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 having partially-damaged payloads delivered, rather than
discarded. One use a compatible method defining
how they send packets and where suitable to choose common algorithms is to manage their functions. Examples of suitable applications support error tolerate payload corruption
when used over paths that include
interactive applications, streaming media or on-line games [RFC4336].

3.6.1. Protocol Description

DCCP error-prone links, another
application is when header integrity checks are required, but payload
integrity is a connection-oriented datagram protocol, providing a three
way handshake to allow a client provided by some other mechanism (e.g., [RFC6936]).

A UDP-Lite service may support IPv4 broadcast, multicast, anycast and server to set up a connection,
unicast, and mechanisms for orderly completion IPv6 multicast, anycast and immediate teardown of a
connection. The protocol unicast.

4.5.2. Interface Description

There is defined by a family no API currently specified in the RFC Series, but guidance
on use of RFCs.

It provides multiplexing to multiple sockets at each endpoint using
port numbers. An active DCCP session common APIs is identified by its four-tuple provided in [I-D.ietf-tsvwg-rfc5405bis].

The interface of local and remote IP addresses and local port and remote port
numbers. At connection setup, DCCP also exchanges UDP-Lite differs from that of UDP by the service code
[RFC5595], addition of
a mechanism single (socket) option that allows transport instantiations to
indicate communicates a checksum coverage length
value: at the service treatment that is expected from sender, this specifies the network. intended checksum coverage,
with the remaining unprotected part of the payload called the "error-
insensitive part". The protocol segments data into messages, typically sized to fit in
IP packets, but which checksum coverage may also be fragmented providing they are less than
the maximum packet size. A DCCP interface allows applications made visible to
request fragmentation
the application via the UDP-Lite MIB module [RFC5097].

4.5.3. Transport Features

The transport features provided by UDP-Lite are:

o unicast.

o multicast, anycast, or IPv4 broadcast.

o port multiplexing (as for UDP).

o message-oriented delivery (as for UDP).

o Uni-or bidirectional communication. Transmission in each
direction is independent.

o non-reliable delivery (as for packets larger than PMTU, but not larger
than UDP).

o non-ordered delivery (as for UDP).

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

o partial or full integrity protection. The checksum coverage field
indicates the maximum packet size allowed by of the current congestion
control mechanism (CCMPS) [RFC4340].

Each message is identified payload data covered by a sequence number. The sequence number the checksum.

4.6. Datagram Congestion Control Protocol (DCCP)

Datagram Congestion Control Protocol (DCCP) [RFC4340] is used to identify segments in acknowledgments, to detect
unacknowledged segments, to measure RTT, etc. The an IETF
standards track bidirectional transport protocol may
support ordered or unordered delivery that provides
unicast connections of data, and does not itself
provide retransmission. congestion-controlled messages without
providing reliability.

The DCCP supports reduced checksum coverage, a
partial integrity mechanisms similar to UDP-lIte. There is also a
Data Checksum option Problem Statement describes the goals that when enabled, contains a strong CRC, to
enable endpoints DCCP sought to detect application data corruption.

Receiver flow control
address [RFC4336]. It is supported: limiting the amount suitable for applications that transfer
fairly large amounts of
unacknowledged data and that can be outstanding at a given time.

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

A DCCP service is unicast.

DCCP supports negotiation of the congestion benefit from control profile, to
provide Plug over
the trade off between timeliness and Play congestion control mechanisms. Examples of
specified profiles include [RFC4341] [RFC4342] [RFC5662]. All IETF-
defined methods provide Congestion Control. reliability [RFC4336].

DCCP use offers low overhead, and many characteristics common to UDP, but
can avoid "re-inventing the wheel" each time a Connect packet new multimedia
application emerges. Specifically it includes core functions
(feature negotiation, path state management, RTT calculation, PMTUD,
etc): This allows applications to initiate use a session, compatible method defining
how they send packets and permits half-
connections that allow each client where suitable to choose the features it wishes common algorithms
to support. Simultaneous open [RFC5596], as in TCP, can enable
interoperability in the presence manage their functions. Examples of middleboxes. The Connect packet
includes suitable applications include
interactive applications, streaming media or on-line games [RFC4336].

4.6.1. Protocol Description

DCCP is a Service Code field [RFC5595] designed connection-oriented datagram protocol, providing a three-
way handshake 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] client and
[RFC5595].

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

A common packet format has allowed tools server to evolve that can read set up a connection,
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 mechanisms for orderly completion and reception immediate teardown of zero-
length payloads. - Slow Receiver flow control at a receiver. -
Detect a Slow receiver at the sender.

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

3.6.3. Transport Features
connection. The transport features provided protocol is defined by a family of RFCs.

It provides multiplexing to multiple sockets at each endpoint using
port numbers. An active DCCP are:

o unicast.

o connection setup with feature negotiation session is identified by its four-tuple
of local and application-to-port
mapping.

o Service Codes. Identifies remote IP addresses and local port and remote port
numbers. At connection setup, DCCP also exchanges the upper layer service code
[RFC5595], a mechanism that allows transport instantiations to
indicate the service treatment that is expected from the endpoint
and network.

o port multiplexing.

o message-oriented delivery.

o non-reliable delivery.

o ordered delivery.

o flow control.

The slow receiver function allows a receiver protocol segments data into messages, typically sized to
control the rate of fit in
IP packets, but which may be fragmented providing they are less than
the sender.

o drop notification. Allows a receiver maximum packet size. A DCCP interface allows applications to notify which datagrams
were
request fragmentation for packets larger than PMTU, but not delivered to larger
than the peer upper layer protocol.

o timestamps.

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

3.7. Lightweight User Datagram Protocol (UDP-Lite)

The Lightweight User Datagram Protocol (UDP-Lite) [RFC3828] maximum packet size allowed by the current congestion
control mechanism (CCMPS) [RFC4340].

Each message is an
IETF standards track transport protocol. It provides identified by a
unidirectional, datagram protocol that preserves message boundaries.
IETF guidance on the use of UDP-Lite sequence number. The sequence number
is provided used to identify segments in
[I-D.ietf-tsvwg-rfc5405bis].

3.7.1. Protocol Description

UDP-Lite is a connection-less datagram protocol, with no connection
setup or feature negotiation. acknowledgments, to detect
unacknowledged segments, to measure RTT, etc. The protocol use messages, rather than
a byte-stream. Each stream may
support ordered or unordered delivery of messages is independently managed,
therefore retransmission data, and does not hold back data sent using other
logical streams.

It provides multiplexing itself
provide retransmission. DCCP supports reduced checksum coverage, a
partial integrity mechanism similar to multiple sockets on each host using port
numbers, and its operation follows UDP-Lite. There is also a
Data Checksum option that for UDP. An active UDP-Lite
session when enabled, contains a strong CRC, to
enable endpoints to detect application data corruption - similar to
SCTP.

Receiver flow control is identified by its four-tuple of local and remote IP
addresses and local port and remote port numbers.

UDP-Lite changes supported, which limits the semantics amount of the UDP "payload length" field to
unacknowledged data that of can be outstanding at a "checksum coverage length" field, given time.

A DCCP protocol instance can be extended [RFC4340] and is identified by a
different IP protocol/next-header value. Otherwise, UDP-Lite is
semantically identical to UDP. Applications tuned using UDP-Lite therefore
can not make assumptions regarding
additional features. Some features are sender-side only, requiring
no negotiation with the correctness receiver; some are receiver-side only; and
some are explicitly negotiated during connection setup.

DCCP service is unicast-only.

It supports negotiation of the data
received in the insensitive part congestion control profile, to provide
plug- and-play congestion control mechanisms. Examples of the UDP-Lite payload.

As specified
profiles include "TCP-like" [RFC4341], "TCP-friendly" [RFC4342], and
"TCP-friendly for UDP, small packets" [RFC5622]. Additional mechanisms for receiver flow control, congestion control,
PMTU or PLPMTU discovery, support for ECN, etc need
are recorded in an IANA registry.

DCCP uses a Connect packet to initiate a session, and permits half-
connections that allow each client to choose the features it wishes
to support. Simultaneous open [RFC5596], as in TCP, can enable
interoperability in the presence of middleboxes. The Connect packet
includes a Service Code field [RFC5595] designed to be provided allow middleboxes
and endpoints to identify the characteristics required by
upper layer protocols [I-D.ietf-tsvwg-rfc5405bis].

Examples of use include a class of applications session.

A lightweight UDP-based encapsulation (DCCP-UDP) has been defined
[RFC6773] that can derive
benefit from having partially-damaged payloads delivered, rather than
discarded. One use is permits DCCP to support error tolerate payload corruption
when be used over paths that include error-prone links, another
application where DCCP is when header integrity checks are required, but payload
integrity not
natively supported. Support in NAPT/NATs is provided by some other mechanism (e.g., [RFC6936].

A UDP-Lite service may support IPv4 broadcast, multicast, anycast and
unicast, and IPv6 multicast, anycast defined in [RFC4340] and unicast.

3.7.2. Interface Description

There is no current API
[RFC5595].

Upper layer protocols specified in the RFC Series, but guidance on
use top of DCCP include DTLS
[RFC5595], RTP [RFC5672], ICE/SDP [RFC6773].

A common APIs is provided in [I-D.ietf-tsvwg-rfc5405bis].

The interface of UDP-Lite differs from packet format has allowed tools to evolve that can read and
interpret DCCP packets (e.g., Wireshark).

4.6.2. Interface Description

API characteristics include: - Datagram transmission. - Notification
of UDP by the addition current maximum packet size. - Send and reception of
a single (socket) option that communicates a checksum coverage zero-
length
value: payloads. - Slow Receiver flow control at the sender, this specifies the intended checksum coverage,
with the remaining unprotected part of the payload called the "error-
insensitive part". The checksum coverage may also be made visible a receiver. -
ability to detect a slow receiver at the application via sender.

There is no API currently specified in the UDP-Lite MIB module [RFC5097].

3.7.3. RFC Series.

4.6.3. Transport Features

The transport features provided by UDP-Lite DCCP are:

o unicast transport.

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

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

o port multiplexing (as for UDP). multiplexing.

o Uni-or bidirectional communication.

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

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

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

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

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

o timestamps.

o partial or and full integrity protection.

3.8. protection (with optional strong
integrity check).

4.7. Internet Control Message Protocol (ICMP)

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

It provides

ICMP is a conection-less connection-less unidirectional protocol that delivers
individual messages. It provides none of the following transport
features: messages, without error correction, congestion control, or
flow control.
Some messages Messages may be sent as unicast, IPv4 broadcast datagrams (IPv4) or
multicast datagrams (IPv4 and IPv6), in addition to unicast (and anycast) anycast
datagrams.

3.8.1.

4.7.1. Protocol Description

ICMP is a conection-less connection-less unidirectional protocol that delivers
individual messages. The protocol uses independent messages,
ordinarily called datagrams. Each message is required to carry a
checksum as an integrity check and to protect from misdelivery to the
wrong an
unintended endpoint.

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

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

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

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

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

3.8.2.

4.7.2. Interface Description

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

3.8.3.

4.7.3. Transport Features

The transport features provided by ICMP are:

o unidirectional.

o multicast, anycast and IP4 broadcast.

o message-oriented delivery.

o non-reliable delivery.

o non-ordered delivery.

o error and misdelivery detection (checksum).

3.9.

4.8. Realtime Transport Protocol (RTP)

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

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

3.9.1.

4.8.1. Protocol Description

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

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

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

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

RTCP is an associated a control protocol that works with RTP. alongside a RTP flow. Both the
RTP sender and receiver can send RTCP report packets. This is used
to periodically send control information and report performance.
Based on received RTCP feedback, an RTP sender can adjust the
transmission, e.g., perform rate adaptation at the application layer
in the case of congestion.

An RTCP receiver report (RTCP RR) is returned to the sender
periodically to report key parameters (e.g, the fraction of packets
lost in the last reporting interval, the cumulative number of packets
lost, the highest sequence number received, and the inter-arrival
jitter). The RTCP RR packets also contain timing information that
allows the sender to estimate the network round trip time (RTT) to
the receivers.

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

4.8.2. Interface Description

There is no standard application programming interface defined for
RTP or RTCP. Implementations are typically tightly integrated with a
particular application, and closely follow the principles of
application level framing and integrated layer processing [ClarkArch]
in media processing [RFC2736], error recovery and concealment, rate sessions. The interval is randomised
adaptation, and security [RFC7202]. Accordingly, RTP implementations
tend to avoid
synchronization be targeted at particular application domains (e.g., voice-
over-IP, IPTV, or video conferencing), with a feature set optimised
for that domain, rather than being general purpose implementations of reports from multiple receivers.

3.9.2. Interface Description

[EDITOR'S NOTE: to do]

3.9.3.
the protocol.

4.8.3. Transport Features

The transport features provided by RTP are:

o unicast. unicast transport.

o multicast, anycast or IPv4 broadcast.

o port multiplexing.

o Uni-or bidirectional communication.

o message-oriented delivery.

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

o support for media types and other extensions.

o a range of reliability functions, including the possibility of
using packet erasure coding.

o segmentation and aggregation.

o performance reporting.

o drop notification.

o timestamps.

3.10.

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

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

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

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

Altough

Although FLUTE/ALC is not well adapted to byte- and message-streaming, message-
streaming, there is an exception: FLUTE/ALC is used to carry 3GPP
Dynamic Adaptive Streaming over HTTP (DASH) when scalability is a
requirement (see [MBMS], section 5.6). In that case, each Audio/Video Audio/
Video segment is transmitted as a distinct FLUTE/ALC object in push
mode. FLUTE/ALC uses packet erasure coding (also known as
Application-Level Forward Erasure Correction, or AL-FEC) in a
proactive way. The goal of using AL-FEC is both to increase the
robustness in front of packet erasures and to improve the efficiency
of the on-demand service. FLUTE/ALC transmissions can be governed by
a congestion control mechanism such as the "Wave and Equation Based
Rate Control" (WEBRC) [RFC3738] when FLUTE/ALC is used in a layered
transmission manner, with several session channels over which ALC
packets are sent. However many FLUTE/ALC deployments target pre-
provisioned networks and involve only Constant Bit Rate (CBR)
channels with no competing flows, for which a sender-based rate
control mechanism is sufficient. In any case, FLUTE/ALC's
reliability, delivery mode, congestion control, and flow/rate control
mechanisms are distinct components that can be separately controlled
to meet different application needs.

3.10.1. Section 4.1 of
[I-D.ietf-tsvwg-rfc5405bis] describes multicast congestion control
requirements for UDP.

4.9.1. Protocol Description

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

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

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

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

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

A FLUTE/ALC session is composed of one or more channels, associated
to different destination unicast and/or multicast IP addresses. ALC
packets are sent in those channels at a certain transmission rate,
with a rate that often differs depending on the channel. FLUTE/ALC
does not mandate nor recommend any strategy to select which ALC
packet to send on which channel. FLUTE/ALC can use a multiple rate
congestion control building block (e.g., WEBRC) to provide congestion
control that is feedback free, where receivers adjust their reception
rates individually by joining and leaving channels associated with
the session. To that purpose, the ALC header provides a specific
field to carry congestion control specific information. However
FLUTE/ALC does not mandate the use of a particular congestion control
mechanism although WEBRC is mandatory to support in case of for the Internet
([RFC6726], section 1.1.4). FLUTE/ALC is often used over a network
path with pre-provisoned pre-provisioned capacity [RFC5404] whete theres [I-D.ietf-tsvwg-rfc5405bis] where
there are no flows competing for capacity. In this case, a sender-based sender-
based rate control mechanism and a single channel is sufficient.

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

3.10.2.

4.9.2. Interface Description

The FLUTE/ALC specification does not describe a specific application
programming interface (API) to control protocol operation.
Open source reference implementations of FLUTE/ALC are available at
http://planete-bcast.inrialpes.fr/ (no longer maintained) and
http://mad.cs.tut.fi/ (no longer maintained), and these
implementations specify and document their own APIs. Commercial
versions are also available, some derived from the above
implementations, with their own API.

3.10.3.

4.9.3. Transport Features

The transport features provided by FLUTE/ALC are:

o unicast

o multicast, anycast or IPv4 broadcast.

o per-object dynamic meta-data delivery.

o push delivery or on-demand delivery service.

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

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

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

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

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

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

o rate control transmission in a given channel.

3.11.

4.10. 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 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. acknowledgments. The packet erasure coding can
also be proactively applied for forward protection from packet loss.
NORM transmissions are governed by the TCP-friendly congestion
control. NORM's reliability, congestion control, and flow control
mechanism are distinct components and can be separately controlled to
meet different application needs.

3.11.1.

4.10.1. 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 connection-
oriented although per-sender state is maintained by receivers for
protocol operation. [RFC5740] does not specify a handshake protocol
for connection establishment and separate session initiation can be
used to coordinate port numbers. However, in-band "client-server"
style connection establishment can be accomplished with the NORM
congestion control signaling messages using port binding techniques
like those for TCP client-server connections.

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

NORM uses rate-based congestion control based upon the TCP-Friendly
Rate Control (TFRC) [RFC4324] principles that are also used in DCCP

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

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

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

3.11.2.

4.10.2. Interface Description

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

3.11.3.

4.10.3. Transport Features

The transport features provided by NORM are:

o unicast or multicast. multicast transport.

o stream-oriented delivery in a single stream.

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

o reliable delivery.

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

o error detection (UDP checksum).

o segmentation.

o data bundling (Nagle's algorithm).

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

o congestion control.

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

3.12.

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

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

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

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

3.12.1.

4.11.1. Protocol Description

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

o Confidentiality

o Data integrity

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

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

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

o Record replay detection (optional).

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

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

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

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

3.12.2.

4.11.2. Interface Description

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

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

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

3.12.3.

4.11.3. Transport Features

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

o message fragmentation fragmentation.

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

o data encryption encryption.

o scheduling transmission using the underlying transport protocol protocol.

DTLS augments the TLS record protocol with:

o ordering and replay protection, implemented using sequence
numbers.

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

3.13.

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

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

o confidentiality.

o cryptographic integrity protection.

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

The Hypertext Transfer Protocol (HTTP) is an application-level
protocol widely used on the Internet. Version 1.1 of the protocol is
specified in [RFC7230] [RFC7231] [RFC7232] [RFC7233] [RFC7234]
[RFC7235], and version 2 in [RFC7540]. Furthermore, HTTP is usually transported
over TCP using port 80 and 443, although it can be used with other
transports. When used over TCP it inherits its properties.

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 over many types of infrastructure, and in
cases where a application server often needs to support HTTP anyway.

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

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

3.13.1.

4.12.1. 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. (e.g., data can be transferred compressed, compressed using gzip), etc.
HTTP can accommodate exchange of messages as well as data streaming
(using chunked transfer encoding [RFC7230]). It is also possible to
request a part 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
transport connection. This reduces connection establishment overhead
and the effect of TCP the transport layer slow-start on each transaction,
important in reducing latency 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. (e.g., authentication, encryption, etc.). TLS's
Application-Layer encryption). The TLS Application-
Layer Protocol Negotiation (ALPN) extension [RFC7301] can be used for
HTTP version negotiation within the TLS handshake handshake, which eliminates
the latency of addition round-trip. round-trips. 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 an
existing method and data formats, or specify new methods and data
formats. Furthermore some protocols may not fit a request/response
paradigm and instead rely on HTTP to send messages (e.g. (e.g., [RFC6546]).
Because HTTP is working works in many restricted infrastructures, it is also
used to tunnel other application-layer protocols.

3.13.2.

4.12.2. Interface Description

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

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

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

3.13.3. Transport features

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

o unicast.

o message and stream-oriented transfer.

o bi- or unidirectional transmission.

o ordered delivery.

o fully reliable delivery.

o object range request.

o message content type negotiation.

o flow control.

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

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

o confidentiality.

o integrity protection.

4. Transport Service Features

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

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

o Control Functions

* Addressing

+ unicast

+ multicast, anycast and IPv4 broadcast

+ use of NAPT-compatible port numbers

* Multihoming support

+ multihoming for resilience

+ multihoming for mobility

- specify handover latency?

+ multihoming for load-balancing

- specify interleaving delay?

* Multiplexing

+ application to port mapping

+ single vs. multiple streaming

o Delivery

* reliability

+ fully reliable delivery

+ partially reliable delivery
- packet erasure coding

+ unreliable delivery

- drop notification

- Integrity protection

o checksum transport protocol. REST is an
architecture style for error detection building application on the Internet. It uses
HTTP as a communication protocol.

4.12.3. Transport features

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

o partial payload checksum protection unicast.

o checksum optional

* ordering

+ ordered delivery

+ unordered delivery

- unordered delivery of in-memory data

* type/framing

+ message and stream-oriented delivery

+ message-oriented delivery

+ object-oriented delivery of discrete data transfer.

o bi- or file items

- unidirectional transmission.

o ordered delivery.

o fully reliable delivery.

o object range request.

o message content type negotiation

+ range-based partical object transmission

+ file bulk content objects negotiation.

o Transmission control

* rate control

+ timer-based

+ ACK-based

* congestion control

* flow control
* segmentation

* data/message bundling (Nagle's algorithm)

* stream scheduling prioritization control.

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

o Security

* authentication of (of one end of a connection

* authentication of or both ends of a connection

* confidentiality

* cryptographic connection).

o confidentiality.

o integrity protection

A future revision of this document will define transport service protection.

5. Transport Service Features

The tables below summarize some key features based upon this list.

[EDITOR'S NOTE: this section will drawn from to illustrate the candidate features range
of functions provided by protocol components in across 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 IETF-specified transports. Figure 1
considers transports 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 may be directly layered over the network,
and Figure 2 considers transports layered over another transport
service.

+---------------+------+------+------+------+------+------+------+
| SCTP Feature | TCP |
+-----------------+---------+---------+---------+---------+---------+
| Unicast | Yes | Yes | Yes | Yes | Yes |
| | | | | | | MPTCP| SCTP | Mcast/IPv4Bcast UDP | Yes(2) UDP-L|DCCP |ICMP | Yes
+---------------+------+------+------+------+------+------+------+
| No Datagram | No | No |
| | | | | | |
| Port Mux | Yes | Yes | Yes | Yes | Yes |
+---------------+------+------+------+------+------+------+------+
| | | | | | |
| Mode | Dgram | Dgram | Dgram | Dgram | Stream |
| | | | | | |
| Connected | No | No | Conn. Oriented| Yes | Yes | Yes |
| | | | | | |
| Data bundling | No | No | No | Yes | Yes |
| | | | | | |
| Feature Nego | No | No
+---------------+------+------+------+------+------+------+------+
| Yes Reliability | Yes | Yes |
| | | | | | |
| Options Yes | No | No | Support | Support | Support |
| | | | | | |
| Data priority | * No | * No | *
+---------------+------+------+------+------+------+------+------+
| Yes Partial Rel. | No | No | Pos | N/A | N/A | Yes | N/A |
+---------------+------+------+------+------+------+------+------+
| Corupt. Tol | Data bundling No | No | No | No | Yes | Yes |
| | | | | | |
| Reliability | None | None | None | Select | Full |
| | | | | | |
| Ordered deliv | No | No | No
+---------------+------+------+------+------+------+------+------+
| Stream Cong.Control | Yes | Yes | | | | | | |
| Corruption Tol. Yes | No | Support No | Support Yes | No | No
+---------------+------+------+------+------+------+------+------+
| Endpoint | 1 | >=1 | >=1 | 1 | 1 | 1 | 1 | Flow Control
+---------------+------+------+------+------+------+------+------+
| Multicast Cap.| No | No | Support No | Yes | Yes | No | No |
+---------------+------+------+------+------+------+------+------+

Figure 1: Summary comparison: Transport protocols

+---------------+------+------+------+------+------+
| Feature | RTP | FLUTE| NORM |(D)TLS| HTTP |
+---------------+------+------+------+------+------+
|
| PMTU/PLPMTU | (1) | (1) | Yes | Yes Datagram | Yes | No | Both | Both | No |
+---------------+------+------+------+------+------+
| | |
| Cong Control | (1) | (1) | Yes Conn. Oriented| No | Yes | Yes |
| | | | | | |
| ECN Support | (1) | (1) | Yes | TBD | Yes |
+---------------+------+------+------+------+------+
| Reliability | No | Yes | Pos | Pos | Yes |
+---------------+------+------+------+------+------+
| NAT support | Limited | Limited | Support | TBD | Support |
| | | | | | |
| Security | DTLS Partial R | DTLS Pos | DTLS No | DTLS Pos | TLS, AO No | No |
+---------------+------+------+------+------+------+
| Corupt. Tol | Poss | No | No | No | No | UDP encaps
+---------------+------+------+------+------+------+
| N/A Cong.Control | None Poss | Yes Poss | Yes Poss | None N/A | N/A |
+---------------+------+------+------+------+------+
| Endpoint | >=1 | >=1 | >=1 | 1 | 1 | RTP support
+---------------+------+------+------+------+------+
| Support Multicast Cap.| Yes | Support Yes | Support Yes | ? No | Support No |
+-----------------+---------+---------+---------+---------+---------+

Note (1): this feature requires support in an upper
+---------------+------+------+------+------+------+

Figure 2: Upper layer protocol.

Note (2): transports and frameworks

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

o Control Functions

* Addressing

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

+ multicast (UDP, UDP-Lite, DCCP, FLUTE/ALC, NORM)

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

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

* Multihoming support

+ multihoming for resilience (MPTCP, SCTP)

+ multihoming for mobility (MPTCP, SCTP)

+ multihoming for load-balancing (MPTCP)

* Application to port mapping (TCP, MPTCP, SCTP, UDP, UDP-Lite,
DCCP, FLUTE/ALC, NORM, TLS, HTTP)

+ with commonly deployed support in an upper layer protocol
when NAPT (TCP, MPTCP, UDP,
TLS, HTTP)

o Delivery

* reliability

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

+ partially reliable delivery (SCTP, NORM)

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

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

- with drop notification (SCTP, DCCP)

+ Integrity protection

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

- partial payload checksum protection (UDP-Lite, DCCP)

- checksum optional (UDP)

* ordering

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

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

* type/framing

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

- with multiple streams per association (SCTP)

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

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

o Transmission control

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

* congestion control (TCP, MPTCP, SCTP, DCCP, FLUTE/ALC, NORM,
TLS, HTTP)

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

* data/message bundling (TCP, MPTCP, SCTP, TLS, HTTP)

* stream scheduling prioritization (SCTP)

o Security (may be used in combination with IPv6.

5. other transports)

* authentication of one end of a connection (TLS)

* authentication of both ends of a connection (TLS)

* confidentiality (TLS)

* cryptographic integrity protection (TLS)

6. IANA Considerations

This document has no considerations for IANA.

7. Security Considerations

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

8. Contributors

[Editor's Note: turn this into a real contributors section with
addresses once we figure out how

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

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

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

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

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

o Section 3.10 4.9 on FLUTE/ALC was contributed by Vincent Roca
(vincent.roca@inria.fr)

o Section 3.11 4.10 on NORM was contributed by Brian Adamson
(brian.adamson@nrl.navy.mil)

o Section 3.12 4.11 on TLS and DTLS was contributed by Ralph Holz
(ralph.holz@nicta.com.au) and Olivier Mehani
(olivier.mehani@nicta.com.au)

o Section 3.13 4.12 on HTTP was contributed by Dragana Damjanovic
(ddamjanovic@mozilla.com)

9. Acknowledgments

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

10. Informative References

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

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

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

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

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

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

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

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

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

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

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

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

[RFC2617] Franks, J., Hallam-Baker, P., Hostetler, J., Lawrence, S.,
Leach, P., Luotonen, A., and L. Stewart, "HTTP
Authentication: Basic and Digest Access Authentication",
RFC 2617, DOI 10.17487/RFC2617, June 1999,
<http://www.rfc-editor.org/info/rfc2617>.

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

[RFC5404] Westerlund, M.

[RFC5348] Floyd, S., Handley, M., Padhye, J., and I. Johansson, "RTP Payload Format for
G.719", J. Widmer, "TCP
Friendly Rate Control (TFRC): Protocol Specification",
RFC 5404, 5348, DOI 10.17487/RFC5404, January 2009,
<http://www.rfc-editor.org/info/rfc5404>. 10.17487/RFC5348, September 2008,
<http://www.rfc-editor.org/info/rfc5348>.

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

[I-D.ietf-tsvwg-rfc5405bis]
Eggert, L., Fairhurst, G., and G. Shepherd, "UDP Usage
Guidelines", draft-ietf-tsvwg-rfc5405bis-05 draft-ietf-tsvwg-rfc5405bis-07 (work in
progress), August November 2015.

[I-D.ietf-aqm-ecn-benefits]
Fairhurst, G. and M. Welzl, "The Benefits of using
Explicit Congestion Notification (ECN)", draft-ietf-aqm-
ecn-benefits-06
ecn-benefits-07 (work in progress), July November 2015.

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

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

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

[I-D.ietf-tsvwg-natsupp]
Stewart, R., Tuexen, M., and I. Ruengeler, "Stream Control
Transmission Protocol (SCTP) Network Address Translation
Support", draft-ietf-tsvwg-natsupp-08 (work in progress),
July 2015.

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

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

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

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

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

Authors' Addresses

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

Email: gorry@erg.abdn.ac.uk

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

Email: ietf@trammell.ch

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

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