draft-ietf-taps-transports-14.txt   rfc8095.txt 
Network Working Group G. Fairhurst, Ed. Internet Engineering Task Force (IETF) G. Fairhurst, Ed.
Internet-Draft University of Aberdeen Request for Comments: 8095 University of Aberdeen
Intended status: Informational B. Trammell, Ed. Category: Informational B. Trammell, Ed.
Expires: June 9, 2017 M. Kuehlewind, Ed. ISSN: 2070-1721 M. Kuehlewind, Ed.
ETH Zurich ETH Zurich
December 06, 2016 March 2017
Services provided by IETF transport protocols and congestion control Services Provided by
mechanisms IETF Transport Protocols and Congestion Control Mechanisms
draft-ietf-taps-transports-14
Abstract Abstract
This document describes, surveys, and classifies the protocol This document describes, surveys, and classifies the protocol
mechanisms provided by existing IETF protocols, as background for mechanisms provided by existing IETF protocols, as background for
determining a common set of transport services. It examines the determining a common set of transport services. It examines the
Transmission Control Protocol (TCP), Multipath TCP, the Stream Transmission Control Protocol (TCP), Multipath TCP, the Stream
Control Transmission Protocol (SCTP), the User Datagram Protocol Control Transmission Protocol (SCTP), the User Datagram Protocol
(UDP), UDP-Lite, the Datagram Congestion Control Protocol (DCCP), the (UDP), UDP-Lite, the Datagram Congestion Control Protocol (DCCP), the
Internet Control Message Protocol (ICMP), the Realtime Transport Internet Control Message Protocol (ICMP), the Real-Time Transport
Protocol (RTP), File Delivery over Unidirectional Transport/ Protocol (RTP), File Delivery over Unidirectional Transport /
Asynchronous Layered Coding Reliable Multicast (FLUTE/ALC), and NACK- Asynchronous Layered Coding (FLUTE/ALC) for Reliable Multicast, NACK-
Oriented Reliable Multicast (NORM), Transport Layer Security (TLS), Oriented Reliable Multicast (NORM), Transport Layer Security (TLS),
Datagram TLS (DTLS), and the Hypertext Transport Protocol (HTTP), Datagram TLS (DTLS), and the Hypertext Transport Protocol (HTTP),
when HTTP is used as a pseudotransport. This survey provides when HTTP is used as a pseudotransport. This survey provides
background for the definition of transport services within the TAPS background for the definition of transport services within the TAPS
working group. working group.
Status of This Memo Status of This Memo
This Internet-Draft is submitted in full conformance with the This document is not an Internet Standards Track specification; it is
provisions of BCP 78 and BCP 79. published for informational purposes.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months This document is a product of the Internet Engineering Task Force
and may be updated, replaced, or obsoleted by other documents at any (IETF). It represents the consensus of the IETF community. It has
time. It is inappropriate to use Internet-Drafts as reference received public review and has been approved for publication by the
material or to cite them other than as "work in progress." Internet Engineering Steering Group (IESG). Not all documents
approved by the IESG are a candidate for any level of Internet
Standard; see Section 2 of RFC 7841.
This Internet-Draft will expire on June 9, 2017. Information about the current status of this document, any errata,
and how to provide feedback on it may be obtained at
http://www.rfc-editor.org/info/rfc8095.
Copyright Notice Copyright Notice
Copyright (c) 2016 IETF Trust and the persons identified as the Copyright (c) 2017 IETF Trust and the persons identified as the
document authors. All rights reserved. document authors. All rights reserved.
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Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3 1. Introduction ....................................................4
1.1. Overview of Transport Features . . . . . . . . . . . . . 4 1.1. Overview of Transport Features .............................4
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 5 2. Terminology .....................................................5
3. Existing Transport Protocols . . . . . . . . . . . . . . . . 6 3. Existing Transport Protocols ....................................6
3.1. Transport Control Protocol (TCP) . . . . . . . . . . . . 6 3.1. Transport Control Protocol (TCP) ...........................6
3.1.1. Protocol Description . . . . . . . . . . . . . . . . 6 3.1.1. Protocol Description ................................6
3.1.2. Interface description . . . . . . . . . . . . . . . . 8 3.1.2. Interface Description ...............................8
3.1.3. Transport Features . . . . . . . . . . . . . . . . . 8 3.1.3. Transport Features ..................................9
3.2. Multipath TCP (MPTCP) . . . . . . . . . . . . . . . . . . 9 3.2. Multipath TCP (MPTCP) .....................................10
3.2.1. Protocol Description . . . . . . . . . . . . . . . . 9 3.2.1. Protocol Description ...............................10
3.2.2. Interface Description . . . . . . . . . . . . . . . . 10 3.2.2. Interface Description ..............................10
3.2.3. Transport features . . . . . . . . . . . . . . . . . 10 3.2.3. Transport Features .................................11
3.3. User Datagram Protocol (UDP) . . . . . . . . . . . . . . 11 3.3. User Datagram Protocol (UDP) ..............................11
3.3.1. Protocol Description . . . . . . . . . . . . . . . . 11 3.3.1. Protocol Description ...............................11
3.3.2. Interface Description . . . . . . . . . . . . . . . . 12 3.3.2. Interface Description ..............................12
3.3.3. Transport Features . . . . . . . . . . . . . . . . . 12 3.3.3. Transport Features .................................13
3.4. Lightweight User Datagram Protocol (UDP-Lite) . . . . . . 13 3.4. Lightweight User Datagram Protocol (UDP-Lite) .............13
3.4.1. Protocol Description . . . . . . . . . . . . . . . . 13 3.4.1. Protocol Description ...............................13
3.4.2. Interface Description . . . . . . . . . . . . . . . . 13 3.4.2. Interface Description ..............................14
3.4.3. Transport Features . . . . . . . . . . . . . . . . . 14 3.4.3. Transport Features .................................14
3.5. Stream Control Transmission Protocol (SCTP) . . . . . . . 14 3.5. Stream Control Transmission Protocol (SCTP) ...............14
3.5.1. Protocol Description . . . . . . . . . . . . . . . . 14 3.5.1. Protocol Description ...............................15
3.5.2. Interface Description . . . . . . . . . . . . . . . . 16 3.5.2. Interface Description ..............................17
3.5.3. Transport Features . . . . . . . . . . . . . . . . . 19 3.5.3. Transport Features .................................19
3.6. Datagram Congestion Control Protocol (DCCP) . . . . . . . 20 3.6. Datagram Congestion Control Protocol (DCCP) ...............20
3.6.1. Protocol Description . . . . . . . . . . . . . . . . 20 3.6.1. Protocol Description ...............................21
3.6.2. Interface Description . . . . . . . . . . . . . . . . 21 3.6.2. Interface Description ..............................22
3.6.3. Transport Features . . . . . . . . . . . . . . . . . 22 3.6.3. Transport Features .................................22
3.7. Transport Layer Security (TLS) and Datagram TLS (DTLS) as
a pseudotransport . . . . . . . . . . . . . . . . . . . . 22 3.7. Transport Layer Security (TLS) and Datagram TLS
3.7.1. Protocol Description . . . . . . . . . . . . . . . . 23 (DTLS) as a Pseudotransport ...............................23
3.7.2. Interface Description . . . . . . . . . . . . . . . . 24 3.7.1. Protocol Description ...............................23
3.7.3. Transport Features . . . . . . . . . . . . . . . . . 24 3.7.2. Interface Description ..............................24
3.8. Realtime Transport Protocol (RTP) . . . . . . . . . . . . 25 3.7.3. Transport Features .................................25
3.8.1. Protocol Description . . . . . . . . . . . . . . . . 25 3.8. Real-Time Transport Protocol (RTP) ........................26
3.8.2. Interface Description . . . . . . . . . . . . . . . . 26 3.8.1. Protocol Description ...............................26
3.8.3. Transport Features . . . . . . . . . . . . . . . . . 27 3.8.2. Interface Description ..............................27
3.9. Hypertext Transport Protocol (HTTP) over TCP as a 3.8.3. Transport Features .................................27
pseudotransport . . . . . . . . . . . . . . . . . . . . . 27 3.9. Hypertext Transport Protocol (HTTP) over TCP as a
3.9.1. Protocol Description . . . . . . . . . . . . . . . . 28 Pseudotransport ...........................................28
3.9.2. Interface Description . . . . . . . . . . . . . . . . 29 3.9.1. Protocol Description ...............................28
3.9.3. Transport features . . . . . . . . . . . . . . . . . 29 3.9.2. Interface Description ..............................29
3.10. File Delivery over Unidirectional Transport/Asynchronous 3.9.3. Transport Features .................................30
Layered Coding Reliable Multicast (FLUTE/ALC) . . . . . . 30 3.10. File Delivery over Unidirectional Transport /
3.10.1. Protocol Description . . . . . . . . . . . . . . . . 31 Asynchronous Layered Coding (FLUTE/ALC) for
3.10.2. Interface Description . . . . . . . . . . . . . . . 32 Reliable Multicast .......................................31
3.10.3. Transport Features . . . . . . . . . . . . . . . . . 32 3.10.1. Protocol Description ..............................31
3.11. NACK-Oriented Reliable Multicast (NORM) . . . . . . . . . 33 3.10.2. Interface Description .............................33
3.11.1. Protocol Description . . . . . . . . . . . . . . . . 33 3.10.3. Transport Features ................................33
3.11.2. Interface Description . . . . . . . . . . . . . . . 35 3.11. NACK-Oriented Reliable Multicast (NORM) ..................34
3.11.3. Transport Features . . . . . . . . . . . . . . . . . 35 3.11.1. Protocol Description ..............................34
3.12. Internet Control Message Protocol (ICMP) . . . . . . . . 35 3.11.2. Interface Description .............................35
3.12.1. Protocol Description . . . . . . . . . . . . . . . . 36 3.11.3. Transport Features ................................36
3.12.2. Interface Description . . . . . . . . . . . . . . . 37 3.12. Internet Control Message Protocol (ICMP) .................36
3.12.3. Transport Features . . . . . . . . . . . . . . . . . 37 3.12.1. Protocol Description ..............................37
4. Congestion Control . . . . . . . . . . . . . . . . . . . . . 37 3.12.2. Interface Description .............................37
5. Transport Features . . . . . . . . . . . . . . . . . . . . . 38 3.12.3. Transport Features ................................38
6. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 41 4. Congestion Control .............................................38
7. Security Considerations . . . . . . . . . . . . . . . . . . . 41 5. Transport Features .............................................39
8. Contributors . . . . . . . . . . . . . . . . . . . . . . . . 41 6. IANA Considerations ............................................42
9. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 42 7. Security Considerations ........................................42
10. Informative References . . . . . . . . . . . . . . . . . . . 42 8. Informative References .........................................42
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 53 Acknowledgments ...................................................53
Contributors ......................................................53
Authors' Addresses ................................................54
1. Introduction 1. Introduction
Internet applications make use of the Services provided by a Internet applications make use of the services provided by a
Transport protocol, such as TCP (a reliable, in-order stream transport protocol, such as TCP (a reliable, in-order stream
protocol) or UDP (an unreliable datagram protocol). We use the term protocol) or UDP (an unreliable datagram protocol). We use the term
"Transport Service" to mean the end-to-end service provided to an "transport service" to mean the end-to-end service provided to an
application by the transport layer. That service can only be application by the transport layer. That service can only be
provided correctly if information about the intended usage is provided correctly if information about the intended usage is
supplied from the application. The application may determine this supplied from the application. The application may determine this
information at design time, compile time, or run time, and may information at design time, compile time, or run time, and may
include guidance on whether a feature is required, a preference by include guidance on whether a feature is required, a preference by
the application, or something in between. Examples of features of the application, or something in between. Examples of features of
Transport Services are reliable delivery, ordered delivery, content transport services are reliable delivery, ordered delivery, content
privacy to in-path devices, and integrity protection. privacy to in-path devices, and integrity protection.
The IETF has defined a wide variety of transport protocols beyond TCP The IETF has defined a wide variety of transport protocols beyond TCP
and UDP, including SCTP, DCCP, MPTCP, and UDP-Lite. Transport and UDP, including SCTP, DCCP, MPTCP, and UDP-Lite. Transport
services may be provided directly by these transport protocols, or services may be provided directly by these transport protocols or
layered on top of them using protocols such as WebSockets (which runs 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 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 over SCTP over DTLS over UDP or TCP). Services built on top of UDP
or UDP-Lite typically also need to specify additional mechanisms, or UDP-Lite typically also need to specify additional mechanisms,
including a congestion control mechanism (such as NewReno [RFC6582], including a congestion control mechanism (such as NewReno [RFC6582],
TFRC [RFC5348] or LEDBAT [RFC6817]). This extends the set of TCP-Friendly Rate Control (TFRC) [RFC5348], or Low Extra Delay
available Transport Services beyond those provided to applications by Background Transport (LEDBAT) [RFC6817]). This extends the set of
available transport services beyond those provided to applications by
TCP and UDP. TCP and UDP.
The transport protocols described in this document provide a basis The transport protocols described in this document provide a basis
for the definition of transport services provided by common for the definition of transport services provided by common
protocols, as background for the TAPS working group. The protocols protocols, as background for the TAPS working group. The protocols
listed here were chosen to help expose as many potential transport listed here were chosen to help expose as many potential transport
services as possible, and are not meant to be a comprehensive survey services as possible and are not meant to be a comprehensive survey
or classification of all transport protocols. or classification of all transport protocols.
1.1. Overview of Transport Features 1.1. Overview of Transport Features
Transport protocols can be differentiated by the features of the Transport protocols can be differentiated by the features of the
services they provide. services they provide.
Some of these provided features are closely related to basic control Some of these provided features are closely related to basic control
function that a protocol needs to work over a network path, such as function that a protocol needs to work over a network path, such as
addressing. The number of participants in a given association also addressing. The number of participants in a given association also
determines its applicability: if a connection is between endpoints determines its applicability: a connection can be between endpoints
(unicast), to one of multiple endpoints (anycast), or simultaneously (unicast), to one of multiple endpoints (anycast), or simultaneously
to multiple endpoints (multicast). Unicast protocols usually support to multiple endpoints (multicast). Unicast protocols usually support
bidirectional communication, while multicast is generally bidirectional communication, while multicast is generally
unidirectional. Another feature is whether a transport requires a unidirectional. Another feature is whether a transport requires a
control exchange across the network at setup (e.g., TCP), or whether control exchange across the network at setup (e.g., TCP) or whether
it is connection-less (e.g., UDP). it is connectionless (e.g., UDP).
For packet delivery itself, reliability and integrity protection, For packet delivery itself, reliability and integrity protection,
ordering, and framing are basic features. However, these features ordering, and framing are basic features. However, these features
are implemented with different levels of assurance in different are implemented with different levels of assurance in different
protocols. As an example, a transport service may provide full protocols. As an example, a transport service may provide full
reliability, providing detection of loss and retransmission (e.g., reliability, with detection of loss and retransmission (e.g., TCP).
TCP). SCTP offers a message-based service that can provide full or SCTP offers a message-based service that can provide full or partial
partial reliability, and allows the protocol to minimize the head of reliability and allows the protocol to minimize the head-of-line
line blocking due to the support of ordered and unordered message blocking due to the support of ordered and unordered message delivery
delivery within multiple streams. UDP-Lite and DCCP can provide within multiple streams. UDP-Lite and DCCP can provide partial
partial integrity protection to enable corruption tolerance. integrity protection to enable corruption tolerance.
Usually a protocol has been designed to support one specific type of Usually, a protocol has been designed to support one specific type of
delivery/framing: data either needs to be divided into transmission delivery/framing: either data needs to be divided into transmission
units based on network packets (datagram service), a data stream is units based on network packets (datagram service) or a data stream is
segmented and re-combined across multiple packets (stream service), segmented and re-combined across multiple packets (stream service).
or whole objects such as files are handled accordingly. This Whole objects such as files are handled accordingly. This decision
decision strongly influences the interface that is provided to the strongly influences the interface that is provided to the upper
upper layer. layer.
In addition, transport protocols offer a certain support for In addition, transport protocols offer a certain support for
transmission control. For example, a transport service can provide transmission control. For example, a transport service can provide
flow control to allow a receiver to regulate the transmission rate of flow control to allow a receiver to regulate the transmission rate of
a sender. Further a transport service can provide congestion control a sender. Further, a transport service can provide congestion
(see Section 4). As an example TCP and SCTP provide congestion control (see Section 4). As an example, TCP and SCTP provide
control for use in the Internet, whereas UDP leaves this function to congestion control for use in the Internet, whereas UDP leaves this
the upper layer protocol that uses UDP. function to the upper-layer protocol that uses UDP.
Security features are often provided independent of the transport Security features are often provided independently of the transport
protocol, via Transport Layer Security (TLS, see Section 3.7) or by protocol, via Transport Layer Security (TLS) (see Section 3.7) or by
the application layer protocol itself. The security properties TLS the application-layer protocol itself. The security properties TLS
provides to the application (such as confidentiality, integrity, and provides to the application (such as confidentiality, integrity, and
authenticity) are also features of the transport layer, even though authenticity) are also features of the transport layer, even though
they are often presently implemented in a separate protocol. they are often presently implemented in a separate protocol.
2. Terminology 2. Terminology
The following terms are used throughout this document, and in The following terms are used throughout this document and in
subsequent documents produced by TAPS that describe the composition subsequent documents produced by the TAPS working group that describe
and decomposition of transport services. the composition and decomposition of transport services.
Transport Service Feature: a specific end-to-end feature that the Transport Feature: a specific end-to-end feature that the transport
transport layer provides to an application. Examples include layer provides to an application. Examples include
confidentiality, reliable delivery, ordered delivery, message- confidentiality, reliable delivery, ordered delivery, message-
versus-stream orientation, etc. versus-stream orientation, etc.
Transport Service: a set of Transport Features, without an Transport Service: a set of transport features, without an
association to any given framing protocol, which provides a association to any given framing protocol, that provides a
complete service to an application. complete service to an application.
Transport Protocol: an implementation that provides one or more Transport Protocol: an implementation that provides one or more
different transport services using a specific framing and header different transport services using a specific framing and header
format on the wire. format on the wire.
Transport Service Instance: an arrangement of transport protocols
with a selected set of features and configuration parameters that
implements a single transport service, e.g., a protocol stack (RTP
over UDP).
Application: an entity that uses the transport layer for end-to-end Application: an entity that uses the transport layer for end-to-end
delivery data across the network (this may also be an upper layer delivery data across the network (this may also be an upper-layer
protocol or tunnel encapsulation). protocol or tunnel encapsulation).
3. Existing Transport Protocols 3. Existing Transport Protocols
This section provides a list of known IETF transport protocols and This section provides a list of known IETF transport protocols and
transport protocol frameworks. It does not make an assessment about transport protocol frameworks. It does not make an assessment about
whether specific implementations of protocols are fully compliant to whether specific implementations of protocols are fully compliant to
current IETF specifications. current IETF specifications.
3.1. Transport Control Protocol (TCP) 3.1. Transport Control Protocol (TCP)
TCP is an IETF standards track transport protocol. [RFC0793] TCP is an IETF Standards Track transport protocol. [RFC793]
introduces TCP as follows: "The Transmission Control Protocol (TCP) introduces TCP as follows:
is intended for use as a highly reliable host-to-host protocol
between hosts in packet-switched computer communication networks, and The Transmission Control Protocol (TCP) is intended for use as a
in interconnected systems of such networks." Since its introduction, highly reliable host-to-host protocol between hosts in packet-
TCP has become the default connection- oriented, stream-based switched computer communication networks, and in interconnected
transport protocol in the Internet. It is widely implemented by systems of such networks.
endpoints and widely used by common applications.
Since its introduction, TCP has become the default connection-
oriented, stream-based transport protocol in the Internet. It is
widely implemented by endpoints and widely used by common
applications.
3.1.1. Protocol Description 3.1.1. Protocol Description
TCP is a connection-oriented protocol, providing a three way TCP is a connection-oriented protocol that provides a three-way
handshake to allow a client and server to set up a connection and handshake to allow a client and server to set up a connection and
negotiate features, and mechanisms for orderly completion and negotiate features and provides mechanisms for orderly completion and
immediate teardown of a connection. TCP is defined by a family of immediate teardown of a connection [RFC793] [TCP-SPEC]. TCP is
RFCs [RFC7414]. defined by a family of RFCs (see [RFC7414]).
TCP provides multiplexing to multiple sockets on each host using port TCP provides multiplexing to multiple sockets on each host using port
numbers. A similar approach is adopted by other IETF-defined numbers. A similar approach is adopted by other IETF-defined
transports. An active TCP session is identified by its four-tuple of transports. An active TCP session is identified by its four-tuple of
local and remote IP addresses and local port and remote port numbers. local and remote IP addresses and local and remote port numbers. The
The destination port during connection setup is often used to destination port during connection setup is often used to indicate
indicate the requested service. the requested service.
TCP partitions a continuous stream of bytes into segments, sized to TCP partitions a continuous stream of bytes into segments, sized to
fit in IP packets based on a negotiated maximum segment size and fit in IP packets based on a negotiated maximum segment size and
further constrained by the effective Maximum Transmission Unit (MTU) further constrained by the effective Maximum Transmission Unit (MTU)
from Path MTU Discovery (PMTUD). ICMP-based Path MTU discovery from Path MTU Discovery (PMTUD). ICMP-based PMTUD [RFC1191]
[RFC1191][RFC1981] as well as Packetization Layer Path MTU Discovery [RFC1981] as well as Packetization Layer PMTUD (PLPMTUD) [RFC4821]
(PMTUD) [RFC4821] have been defined by the IETF. have been defined by the IETF.
Each byte in the stream is identified by a sequence number. The Each byte in the stream is identified by a sequence number. The
sequence number is used to order segments on receipt, to identify sequence number is used to order segments on receipt, to identify
segments in acknowledgments, and to detect unacknowledged segments segments in acknowledgments, and to detect unacknowledged segments
for retransmission. This is the basis of the reliable, ordered for retransmission. This is the basis of the reliable, ordered
delivery of data in a TCP stream. TCP Selective Acknowledgment delivery of data in a TCP stream. TCP Selective Acknowledgment
(SACK) [RFC2018] extends this mechanism by making it possible to (SACK) [RFC2018] extends this mechanism by making it possible to
provide earlier identification of which segments are missing, provide earlier identification of which segments are missing,
allowing faster retransmission. SACK-based methods (e.g. Duplicate allowing faster retransmission. SACK-based methods (e.g., Duplicate
Selective ACK) can also result in less spurious retransmission. Selective ACK) can also result in less spurious retransmission.
Receiver flow control is provided by a sliding window: limiting the Receiver flow control is provided by a sliding window, which limits
amount of unacknowledged data that can be outstanding at a given the amount of unacknowledged data that can be outstanding at a given
time. The window scale option [RFC7323] allows a receiver to use time. The window scale option [RFC7323] allows a receiver to use
windows greater than 64KB. windows greater than 64 KB.
All TCP senders provide congestion control, such as described in All TCP senders provide congestion control, such as that described in
[RFC5681]. TCP uses a sequence number with a sliding receiver window [RFC5681]. TCP uses a sequence number with a sliding receiver window
for flow control. The TCP congestion control mechanism also utilises for flow control. The TCP congestion control mechanism also utilizes
this TCP sequence number to manage a separate congestion window this TCP sequence number to manage a separate congestion window
[RFC5681]. The sending window at a given point in time is the [RFC5681]. The sending window at a given point in time is the
minimum of the receiver window and the congestion window. The minimum of the receiver window and the congestion window. The
congestion window is increased in the absence of congestion and, congestion window is increased in the absence of congestion and
respectively, decreased if congestion is detected. Often loss is decreased if congestion is detected. Often, loss is implicitly
implicitly handled as a congestion indication which is detected in handled as a congestion indication, which is detected in TCP (also as
TCP (also as input for retransmission handling) based on two input for retransmission handling) based on two mechanisms: a
mechanisms: A retransmission timer with exponential back-off or the retransmission timer with exponential back-off or the reception of
reception of three acknowledgment for the same segment, so called three acknowledgments for the same segment, so called "duplicated
duplicated ACKs (Fast retransmit). In addition, Explicit Congestion ACKs" (fast retransmit). In addition, Explicit Congestion
Notification (ECN) [RFC3168] can be used in TCP, if supported by both Notification (ECN) [RFC3168] can be used in TCP and, if supported by
endpoints, that allows a network node to signal congestion without both endpoints, allows a network node to signal congestion without
inducing loss. Alternatively, a delay-based congestion control inducing loss. Alternatively, a delay-based congestion control
scheme can be used in TCP that reacts to changes in delay as an early scheme that reacts to changes in delay as an early indication of
indication of congestion as also further described in Section 4. congestion can be used in TCP. This is further described in
Examples for different kind of congestion control schemes are given Section 4. Examples of different kinds of congestion control schemes
in Section 4. are provided in Section 4.
TCP protocol instances can be extended [RFC7414] and tuned. Some TCP protocol instances can be extended (see [RFC7414]). Some
features are sender-side only, requiring no negotiation with the protocol features may also be tuned to optimize for a specific
receiver; some are receiver-side only, some are explicitly negotiated deployment scenario. Some features are sender-side only, requiring
during connection setup. no negotiation with the receiver; some are receiver-side only; and
some are explicitly negotiated during connection setup.
TCP may buffer data, e.g., to optimize processing or capacity usage. TCP may buffer data, e.g., to optimize processing or capacity usage.
TCP can therefore provides mechanisms to control this, including an TCP therefore provides mechanisms to control this, including an
optional "PUSH" function [RFC0793] that explicitly requests the optional "PUSH" function [RFC793] that explicitly requests the
transport service not to delay data. By default, TCP segment transport service not to delay data. By default, TCP segment
partitioning uses Nagle's algorithm [RFC0896] to buffer data at the partitioning uses Nagle's algorithm [TCP-SPEC] to buffer data at the
sender into large segments, potentially incurring sender-side sender into large segments, potentially incurring sender-side
buffering delay; this algorithm can be disabled by the sender to buffering delay; this algorithm can be disabled by the sender to
transmit more immediately, e.g., to reduce latency for interactive transmit more immediately, e.g., to reduce latency for interactive
sessions. sessions.
TCP provides an "urgent data" function for limited out-of-order TCP provides an "urgent data" function for limited out-of-order
delivery of the data. This function is deprecated [RFC6093]. delivery of the data. This function is deprecated [RFC6093].
A TCP Reset (RST) control message may be used to force a TCP endpoint A TCP Reset (RST) control message may be used to force a TCP endpoint
to close a session [RFC0793], aborting the connection. to close a session [RFC793], aborting the connection.
A mandatory checksum provides a basic integrity check against A mandatory checksum provides a basic integrity check against
misdelivery and data corruption over the entire packet. Applications misdelivery and data corruption over the entire packet. Applications
that require end to end integrity of data are recommended to include that require end-to-end integrity of data are recommended to include
a stronger integrity check of their payload data. The TCP checksum a stronger integrity check of their payload data. The TCP checksum
[RFC1071] [RFC2460] does not support partial payload protection (as [RFC1071] [RFC2460] does not support partial payload protection (as
in DCCP/UDP-Lite). in DCCP/UDP-Lite).
TCP supports only unicast connections. TCP supports only unicast connections.
3.1.2. Interface description 3.1.2. Interface Description
A User/TCP Interface is defined in [RFC0793] providing six user The User/TCP Interface defined in [RFC793] provides six user
commands: Open, Send, Receive, Close, Status. This interface does commands: Open, Send, Receive, Close, Status, and Abort. This
not describe configuration of TCP options or parameters beside use of interface does not describe configuration of TCP options or
the PUSH and URGENT flags. parameters aside from the use of the PUSH and URGENT flags.
[RFC1122] describes extensions of the TCP/application layer interface [RFC1122] describes extensions of the TCP/application-layer interface
for: for:
o reporting soft errors such as reception of ICMP error messages, o reporting soft errors such as reception of ICMP error messages,
extensive retransmission or urgent pointer advance, extensive retransmission, or urgent pointer advance,
o providing a possibility to specify the Differentiated Services o providing a possibility to specify the Differentiated Services
Code Point (DSCP) [RFC3260] (formerly, the Type-of-Service, TOS) Code Point (DSCP) [RFC3260] (formerly, the Type-of-Service (TOS))
for segments, for segments,
o providing a flush call to empty the TCP send queue, and o providing a flush call to empty the TCP send queue, and
o multihoming support. o multihoming support.
In API implementations derived from the BSD Sockets API, TCP sockets In API implementations derived from the BSD Sockets API, TCP sockets
are created using the "SOCK_STREAM" socket type as described in the are created using the "SOCK_STREAM" socket type as described in the
IEEE Portable Operating System Interface (POSIX) Base Specifications IEEE Portable Operating System Interface (POSIX) Base Specifications
[POSIX]. The features used by a protocol instance may be set and [POSIX]. The features used by a protocol instance may be set and
tuned via this API. There are currently no documents in the RFC tuned via this API. There are currently no documents in the RFC
Series that describe this interface. Series that describe this interface.
3.1.3. Transport Features 3.1.3. Transport Features
The transport features provided by TCP are: The transport features provided by TCP are:
o connection-oriented transport with feature negotiation and o connection-oriented transport with feature negotiation and
application-to-port mapping (implemented using SYN segments and application-to-port mapping (implemented using SYN segments and
the TCP option field to negotiate features), the TCP Option field to negotiate features),
o unicast transport (though anycast TCP is implemented, at risk of o unicast transport (though anycast TCP is implemented, at risk of
instability due to rerouting), instability due to rerouting),
o port multiplexing, o port multiplexing,
o uni- or bidirectional communication, o unidirectional or bidirectional communication,
o stream-oriented delivery in a single stream, o stream-oriented delivery in a single stream,
o fully reliable delivery (implemented using ACKs sent from the o fully reliable delivery (implemented using ACKs sent from the
receiver to confirm delivery), receiver to confirm delivery),
o error detection (implemented using a segment checksum to verify o error detection (implemented using a segment checksum to verify
delivery to the correct endpoint and integrity of the data and delivery to the correct endpoint and integrity of the data and
options), options),
o segmentation, o segmentation,
o data bundling (optional; uses Nagle's algorithm to coalesce data o data bundling (optional; uses Nagle's algorithm to coalesce data
sent within the same RTT into full-sized segments), sent within the same RTT into full-sized segments),
o flow control (implemented using a window-based mechanism where the o flow control (implemented using a window-based mechanism where the
receiver advertises the window that it is willing to buffer), receiver advertises the window that it is willing to buffer), and
o congestion control (usually implemented using a window-based o congestion control (usually implemented using a window-based
mechanism and four algorithms for different phases of the mechanism and four algorithms for different phases of the
transmission: slow start, congestion avoidance, fast retransmit, transmission: slow start, congestion avoidance, fast retransmit,
and fast recovery [RFC5681]). and fast recovery [RFC5681]).
3.2. Multipath TCP (MPTCP) 3.2. Multipath TCP (MPTCP)
Multipath TCP [RFC6824] is an extension for TCP to support multi- Multipath TCP [RFC6824] is an extension for TCP to support
homing for resilience, mobility and load-balancing. It is designed multihoming for resilience, mobility, and load balancing. It is
to be as indistinguishable to middleboxes from non-multipath TCP as designed to be as indistinguishable to middleboxes from non-multipath
possible. It does so by establishing regular TCP flows between a TCP as possible. It does so by establishing regular TCP flows
pair of source/destination endpoints, and multiplexing the between a pair of source/destination endpoints and multiplexing the
application's stream over these flows. Sub- flows can be started application's stream over these flows. Sub-flows can be started over
over IPv4 or IPv6 for the same session. IPv4 or IPv6 for the same session.
3.2.1. Protocol Description 3.2.1. Protocol Description
MPTCP uses TCP options for its control plane. They are used to MPTCP uses TCP options for its control plane. They are used to
signal multipath capabilities, as well as to negotiate data sequence signal multipath capabilities, as well as to negotiate data sequence
numbers, and advertise other available IP addresses and establish new numbers, advertise other available IP addresses, and establish new
sessions between pairs of endpoints. sessions between pairs of endpoints.
By multiplexing one byte stream over separate paths, MPTCP can By multiplexing one byte stream over separate paths, MPTCP can
achieve a higher throughput than TCP in certain situations. However, achieve a higher throughput than TCP in certain situations. However,
if coupled congestion control [RFC6356] is used, it might limit this if coupled congestion control [RFC6356] is used, it might limit this
benefit to maintain fairness to other flows at the bottleneck. When benefit to maintain fairness to other flows at the bottleneck. When
aggregating capacity over multiple paths, and depending on the way aggregating capacity over multiple paths, and depending on the way
packets are scheduled on each TCP subflow, additional delay and packets are scheduled on each TCP subflow, additional delay and
higher jitter might be observed observed before in-order delivery of higher jitter might be observed before in-order delivery of data to
data to the applications. the applications.
3.2.2. Interface Description 3.2.2. Interface Description
By default, MPTCP exposes the same interface as TCP to the By default, MPTCP exposes the same interface as TCP to the
application. [RFC6897] however describes a richer API for MPTCP- application. [RFC6897], however, describes a richer API for MPTCP-
aware applications. aware applications.
This Basic API describes how an application can: This Basic API describes how an application can:
o enable or disable MPTCP. o enable or disable MPTCP.
o bind a socket to one or more selected local endpoints. o bind a socket to one or more selected local endpoints.
o query local and remote endpoint addresses. o query local and remote endpoint addresses.
o get a unique connection identifier (similar to an address-port o get a unique connection identifier (similar to an address-port
pair for TCP). pair for TCP).
The document also recommends the use of extensions defined for SCTP The document also recommends the use of extensions defined for SCTP
[RFC6458] (see next section) to support multihoming for resilience [RFC6458] (see Section 3.5) to support multihoming for resilience and
and mobility. mobility.
3.2.3. Transport features 3.2.3. Transport Features
As an extension to TCP, MPTCP provides mostly the same features. By As an extension to TCP, MPTCP provides mostly the same features. By
establishing multiple sessions between available endpoints, it can establishing multiple sessions between available endpoints, it can
additionally provide soft failover solutions in the case that one of additionally provide soft failover solutions in the case that one of
the paths become unusable. the paths becomes unusable.
The transport features provided by MPTCP in addition to TCP therefore Therefore, the transport features provided by MPTCP in addition to
are: TCP are:
o multihoming for load-balancing, with endpoint multiplexing of a o multihoming for load balancing, with endpoint multiplexing of a
single byte stream, using either coupled congestion control or for single byte stream, using either coupled congestion control or
throughput maximization, throughput maximization,
o address family multiplexing (using IPv4 and IPv6 for the same o address family multiplexing (using IPv4 and IPv6 for the same
session), session), and
o resilience to network failure and/or handover. o resilience to network failure and/or handover.
3.3. User Datagram Protocol (UDP) 3.3. User Datagram Protocol (UDP)
The User Datagram Protocol (UDP) [RFC0768] [RFC2460] is an IETF The User Datagram Protocol (UDP) [RFC768] [RFC2460] is an IETF
standards track transport protocol. It provides a unidirectional Standards Track transport protocol. It provides a unidirectional
datagram protocol that preserves message boundaries. It provides no datagram protocol that preserves message boundaries. It provides no
error correction, congestion control, or flow control. It can be error correction, congestion control, or flow control. It can be
used to send broadcast datagrams (IPv4) or multicast datagrams (IPv4 used to send broadcast datagrams (IPv4) or multicast datagrams (IPv4
and IPv6), in addition to unicast and anycast datagrams. IETF and IPv6), in addition to unicast and anycast datagrams. IETF
guidance on the use of UDP is provided in guidance on the use of UDP is provided in [RFC8085]. UDP is widely
[I-D.ietf-tsvwg-rfc5405bis]. UDP is widely implemented and widely implemented and widely used by common applications, including DNS.
used by common applications, including DNS.
3.3.1. Protocol Description 3.3.1. Protocol Description
UDP is a connection-less protocol that maintains message boundaries, UDP is a connectionless protocol that maintains message boundaries,
with no connection setup or feature negotiation. The protocol uses with no connection setup or feature negotiation. The protocol uses
independent messages, ordinarily called datagrams. It provides independent messages, ordinarily called "datagrams". It provides
detection of payload errors and misdelivery of packets to an detection of payload errors and misdelivery of packets to an
unintended endpoint, either of which result in discard of received unintended endpoint, both of which result in discard of received
datagrams, with no indication to the user of the service. datagrams, with no indication to the user of the service.
It is possible to create IPv4 UDP datagrams with no checksum, and It is possible to create IPv4 UDP datagrams with no checksum, and
while this is generally discouraged [RFC1122] while this is generally discouraged [RFC1122] [RFC8085], certain
[I-D.ietf-tsvwg-rfc5405bis], certain special cases permit this use. special cases permit this use. These datagrams rely on the IPv4
These datagrams rely on the IPv4 header checksum to protect from header checksum to protect from misdelivery to an unintended
misdelivery to an unintended endpoint. IPv6 does not permit UDP endpoint. IPv6 does not permit UDP datagrams with no checksum,
datagrams with no checksum, although in certain cases [RFC6936] this although in certain cases [RFC6936], this rule may be relaxed
rule may be relaxed [RFC6935]. [RFC6935].
UDP does not provide reliability and does not provide retransmission. UDP does not provide reliability and does not provide retransmission.
Messages may be re-ordered, lost, or duplicated in transit. Note Messages may be reordered, lost, or duplicated in transit. Note that
that due to the relatively weak form of checksum used by UDP, due to the relatively weak form of checksum used by UDP, applications
applications that require end to end integrity of data are that require end-to-end integrity of data are recommended to include
recommended to include a stronger integrity check of their payload a stronger integrity check of their payload data.
data.
Because UDP provides no flow control, a receiving application that is Because UDP provides no flow control, a receiving application that is
unable to run sufficiently fast, or frequently, may miss messages. unable to run sufficiently fast, or frequently, may miss messages.
The lack of congestion handling implies UDP traffic may experience The lack of congestion handling implies UDP traffic may experience
loss when using an overloaded path, and may cause the loss of loss when using an overloaded path and may cause the loss of messages
messages from other protocols (e.g., TCP) when sharing the same from other protocols (e.g., TCP) when sharing the same network path.
network path.
On transmission, UDP encapsulates each datagram into a single IP On transmission, UDP encapsulates each datagram into a single IP
packet or several IP packet fragments. This allows a datagram to be packet or several IP packet fragments. This allows a datagram to be
larger than the effective path MTU. Fragments are reassembled before larger than the effective path MTU. Fragments are reassembled before
delivery to the UDP receiver, making this transparent to the user of delivery to the UDP receiver, making this transparent to the user of
the transport service. When the jumbograms are supported, larger the transport service. When jumbograms are supported, larger
messages may be sent without performing fragmentation. messages may be sent without performing fragmentation.
UDP on its own does not provide support for segmentation, receiver UDP on its own does not provide support for segmentation, receiver
flow control, congestion control, PathMTU discovery/PLPMTUD, or ECN. flow control, congestion control, PMTUD/PLPMTUD, or ECN.
Applications that require these features need to provide them on Applications that require these features need to provide them on
their own, or by using a protocol over UDP that provides them their own or use a protocol over UDP that provides them [RFC8085].
[I-D.ietf-tsvwg-rfc5405bis].
3.3.2. Interface Description 3.3.2. Interface Description
[RFC0768] describes basic requirements for an API for UDP. Guidance [RFC768] describes basic requirements for an API for UDP. Guidance
on use of common APIs is provided in [I-D.ietf-tsvwg-rfc5405bis]. on the use of common APIs is provided in [RFC8085].
A UDP endpoint consists of a tuple of (IP address, port number). De- A UDP endpoint consists of a tuple of (IP address, port number).
multiplexing using multiple abstract endpoints (sockets) on the same De-multiplexing using multiple abstract endpoints (sockets) on the
IP address is supported. The same socket may be used by a single same IP address is supported. The same socket may be used by a
server to interact with multiple clients (note: this behavior differs single server to interact with multiple clients. (Note: This
from TCP, which uses a pair of tuples to identify a connection). behavior differs from TCP, which uses a pair of tuples to identify a
Multiple server instances (processes) that bind to the same socket connection). Multiple server instances (processes) that bind to the
can cooperate to service multiple clients. The socket implementation same socket can cooperate to service multiple clients. The socket
arranges to not duplicate the same received unicast message to implementation arranges to not duplicate the same received unicast
multiple server processes. message to multiple server processes.
Many operating systems also allow a UDP socket to be "connected", Many operating systems also allow a UDP socket to be "connected",
i.e., to bind a UDP socket to a specific (remote) UDP endpoint. 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 Unlike TCP's connect primitive, for UDP, this is only a local
operation that serves to simplify the local send/receive functions operation that serves to simplify the local send/receive functions
and to filter the traffic for the specified addresses and ports and to filter the traffic for the specified addresses and ports
[I-D.ietf-tsvwg-rfc5405bis]. [RFC8085].
3.3.3. Transport Features 3.3.3. Transport Features
The transport features provided by UDP are: The transport features provided by UDP are:
o unicast, multicast, anycast, or IPv4 broadcast transport, o unicast, multicast, anycast, or IPv4 broadcast transport,
o port multiplexing (where a receiving port can be configured to o port multiplexing (where a receiving port can be configured to
receive datagrams from multiple senders), receive datagrams from multiple senders),
o message-oriented delivery, o message-oriented delivery,
o uni- or bidirectional communication where the transmissions in o unidirectional or bidirectional communication where the
each direction are independent, transmissions in each direction are independent,
o non-reliable delivery, o non-reliable delivery,
o unordered delivery, o unordered delivery, and
o error detection (implemented using a segment checksum to verify o error detection (implemented using a segment checksum to verify
delivery to the correct endpoint and integrity of the data; delivery to the correct endpoint and integrity of the data;
optional for IPv4 and optional under specific conditions for IPv6 optional for IPv4 and optional under specific conditions for IPv6
where all or none of the payload data is protected), where all or none of the payload data is protected).
3.4. Lightweight User Datagram Protocol (UDP-Lite) 3.4. Lightweight User Datagram Protocol (UDP-Lite)
The Lightweight User Datagram Protocol (UDP-Lite) [RFC3828] is an The Lightweight User Datagram Protocol (UDP-Lite) [RFC3828] is an
IETF standards track transport protocol. It provides a IETF Standards Track transport protocol. It provides a
unidirectional, datagram protocol that preserves message boundaries. unidirectional, datagram protocol that preserves message boundaries.
IETF guidance on the use of UDP- Lite is provided in IETF guidance on the use of UDP-Lite is provided in [RFC8085]. A
[I-D.ietf-tsvwg-rfc5405bis]. A UDP-Lite service may support IPv4 UDP-Lite service may support IPv4 broadcast, multicast, anycast, and
broadcast, multicast, anycast and unicast, and IPv6 multicast, unicast, as well as IPv6 multicast, anycast, and unicast.
anycast and unicast.
Examples of use include a class of applications that can derive Examples of use include a class of applications that can derive
benefit from having partially-damaged payloads delivered, rather than benefit from having partially damaged payloads delivered rather than
discarded. One use is to provider header integrity checks but allow discarded. One use is to provide header integrity checks but allow
delivery of corrupted payloads to error-tolerant applications, or delivery of corrupted payloads to error-tolerant applications or to
when payload integrity is provided by some other mechanism (see applications that use some other mechanism to provide payload
[RFC6936]). integrity (see [RFC6936]).
3.4.1. Protocol Description 3.4.1. Protocol Description
Like UDP, UDP-Lite is a connection-less datagram protocol, with no Like UDP, UDP-Lite is a connectionless datagram protocol, with no
connection setup or feature negotiation. It changes the semantics of connection setup or feature negotiation. It changes the semantics of
the UDP "payload length" field to that of a "checksum coverage the UDP Payload Length field to that of a Checksum Coverage Length
length" field, and is identified by a different IP protocol/next- field and is identified by a different IP protocol/next-header value.
header value. The "checksum coverage length" field specifies the The Checksum Coverage Length field specifies the intended checksum
intended checksum coverage, with the remaining unprotected part of coverage, with the remaining unprotected part of the payload called
the payload called the "error-insensitive part". Applications using the "error-insensitive part". Therefore, applications using UDP-Lite
UDP-Lite therefore cannot make assumptions regarding the correctness cannot make assumptions regarding the correctness of the data
of the data received in the insensitive part of the UDP-Lite payload. received in the insensitive part of the UDP-Lite payload.
Otherwise, UDP-Lite is semantically identical to UDP. In the same Otherwise, UDP-Lite is semantically identical to UDP. In the same
way as for UDP, mechanisms for receiver flow control, congestion way as for UDP, mechanisms for receiver flow control, congestion
control, PMTU or PLPMTU discovery, support for ECN, etc. needs to be control, PMTU or PLPMTU discovery, support for ECN, etc., need to be
provided by upper layer protocols [I-D.ietf-tsvwg-rfc5405bis]. provided by upper-layer protocols [RFC8085].
3.4.2. Interface Description 3.4.2. Interface Description
There is no API currently specified in the RFC Series, but guidance There is no API currently specified in the RFC Series, but guidance
on use of common APIs is provided in [I-D.ietf-tsvwg-rfc5405bis]. on use of common APIs is provided in [RFC8085].
The interface of UDP-Lite differs from that of UDP by the addition of The interface of UDP-Lite differs from that of UDP by the addition of
a single (socket) option that communicates a checksum coverage length a single (socket) option that communicates a checksum coverage length
value. The checksum coverage may also be made visible to the value. The checksum coverage may also be made visible to the
application via the UDP-Lite MIB module [RFC5097]. application via the UDP-Lite MIB module [RFC5097].
3.4.3. Transport Features 3.4.3. Transport Features
The transport features provided by UDP-Lite are: The transport features provided by UDP-Lite are:
o unicast, multicast, anycast, or IPv4 broadcast transport (as for o unicast, multicast, anycast, or IPv4 broadcast transport (same as
UDP), for UDP),
o port multiplexing (as for UDP), o port multiplexing (same as for UDP),
o message-oriented delivery (as for UDP), o message-oriented delivery (same as for UDP),
o Uni- or bidirectional communication where the transmissions in o unidirectional or bidirectional communication where the
each direction are independent (as for UDP), transmissions in each direction are independent (same as for UDP),
o non-reliable delivery (as for UDP), o non-reliable delivery (same as for UDP),
o non-ordered delivery (as for UDP), o non-ordered delivery (same as for UDP), and
o partial or full payload error detection (where the checksum o partial or full payload error detection (where the Checksum
coverage field indicates the size of the payload data covered by Coverage field indicates the size of the payload data covered by
the checksum). the checksum).
3.5. Stream Control Transmission Protocol (SCTP) 3.5. Stream Control Transmission Protocol (SCTP)
SCTP is a message-oriented IETF standards track transport protocol. SCTP is a message-oriented IETF Standards Track transport protocol.
The base protocol is specified in [RFC4960]. It supports multi- The base protocol is specified in [RFC4960]. It supports multihoming
homing and path failover to provide resilience to path failures. An and path failover to provide resilience to path failures. An SCTP
SCTP association has multiple streams in each direction, providing association has multiple streams in each direction, providing
in-sequence delivery of user messages within each stream. This in-sequence delivery of user messages within each stream. This
allows it to minimize head of line blocking. SCTP supports multiple allows it to minimize head-of-line blocking. SCTP supports multiple
stream scheduling schemes controlling stream multiplexing, including stream- scheduling schemes controlling stream multiplexing, including
priority and fair weighting schemes. priority and fair weighting schemes.
SCTP was originally developed for transporting telephony signaling SCTP was originally developed for transporting telephony signaling
messages and is deployed in telephony signaling networks, especially messages and is deployed in telephony signaling networks, especially
in mobile telephony networks. It can also be used for other in mobile telephony networks. It can also be used for other
services, for example, in the WebRTC framework for data channels. services, for example, in the WebRTC framework for data channels.
3.5.1. Protocol Description 3.5.1. Protocol Description
SCTP is a connection-oriented protocol using a four way handshake to SCTP is a connection-oriented protocol using a four-way handshake to
establish an SCTP association, and a three way message exchange to establish an SCTP association and a three-way message exchange to
gracefully shut it down. It uses the same port number concept as gracefully shut it down. It uses the same port number concept as
DCCP, TCP, UDP, and UDP-Lite. SCTP only supports unicast. DCCP, TCP, UDP, and UDP-Lite. SCTP only supports unicast.
SCTP uses the 32-bit CRC32c for protecting SCTP packets against bit SCTP uses the 32-bit CRC32c for protecting SCTP packets against bit
errors and misdelivery of packets to an unintended endpoint. This is errors and misdelivery of packets to an unintended endpoint. This is
stronger than the 16-bit checksums used by TCP or UDP. However, stronger than the 16-bit checksums used by TCP or UDP. However,
partial payload checksum coverage as provided by DCCP or UDP-Lite is partial payload checksum coverage as provided by DCCP or UDP-Lite is
not supported. not supported.
SCTP has been designed with extensibility in mind. A common header SCTP has been designed with extensibility in mind. A common header
is followed by a sequence of chunks. [RFC4960] defines how a is followed by a sequence of chunks. [RFC4960] defines how a
receiver processes chunks with an unknown chunk type. The support of receiver processes chunks with an unknown chunk type. The support of
extensions can be negotiated during the SCTP handshake. Currently extensions can be negotiated during the SCTP handshake. Currently
defined extensions include mechanisms for dynamic re-configuration of defined extensions include mechanisms for dynamic reconfiguration of
streams [RFC6525] and IP addresses [RFC5061]. Furthermore, the streams [RFC6525] and IP addresses [RFC5061]. Furthermore, the
extension specified in [RFC3758] introduces the concept of partial extension specified in [RFC3758] introduces the concept of partial
reliability for user messages. reliability for user messages.
SCTP provides a message-oriented service. Multiple small user SCTP provides a message-oriented service. Multiple small user
messages can be bundled into a single SCTP packet to improve messages can be bundled into a single SCTP packet to improve
efficiency. For example, this bundling may be done by delaying user efficiency. For example, this bundling may be done by delaying user
messages at the sender, similar to Nagle's algorithm used by TCP. messages at the sender, similar to Nagle's algorithm used by TCP.
User messages which would result in IP packets larger than the MTU User messages that would result in IP packets larger than the MTU
will be fragmented at the sender and reassembled at the receiver. will be fragmented at the sender and reassembled at the receiver.
There is no protocol limit on the user message size. For MTU There is no protocol limit on the user message size. For MTU
discovery the same mechanism than for TCP can be used discovery, the same mechanism as for TCP can be used [RFC1981]
[RFC1981][RFC4821], as well as utilizing probe packets with padding [RFC4821], as well as utilization of probe packets with padding
chunks, as defined in [RFC4820]. chunks, as defined in [RFC4820].
[RFC4960] specifies TCP-friendly congestion control to protect the [RFC4960] specifies TCP-friendly congestion control to protect the
network against overload. SCTP also uses sliding window flow control network against overload. SCTP also uses sliding window flow control
to protect receivers against overflow. Similar to TCP, SCTP also to protect receivers against overflow. Similar to TCP, SCTP also
supports delaying acknowledgments. [RFC7053] provides a way for the supports delaying acknowledgments. [RFC7053] provides a way for the
sender of user messages to request the immediate sending of the sender of user messages to request immediate sending of the
corresponding acknowledgments. corresponding acknowledgments.
Each SCTP association has between 1 and 65536 uni-directional streams Each SCTP association has between 1 and 65536 unidirectional streams
in each direction. The number of streams can be different in each in each direction. The number of streams can be different in each
direction. Every user message is sent on a particular stream. User direction. Every user message is sent on a particular stream. User
messages can be sent un- ordered, or ordered upon request by the messages can be sent unordered or ordered upon request by the upper
upper layer. Un-ordered messages can be delivered as soon as they layer. Unordered messages can be delivered as soon as they are
are completely received. For user messages not requiring completely received. For user messages not requiring fragmentation,
fragmentation, this minimizes head of line blocking. On the other this minimizes head-of-line blocking. On the other hand, ordered
hand, ordered messages sent on the same stream are delivered at the messages sent on the same stream are delivered at the receiver in the
receiver in the same order as sent by the sender. same order as sent by the sender.
The base protocol defined in [RFC4960] does not allow interleaving of The base protocol defined in [RFC4960] does not allow interleaving of
user- messages. Large messages on one stream can therefore block the user messages. Large messages on one stream can therefore block the
sending of user messages on other streams. sending of user messages on other streams. [SCTP-NDATA] describes a
[I-D.ietf-tsvwg-sctp-ndata] overcomes this limitation. This draft method to overcome this limitation. This document also specifies
also specifies multiple algorithms for the sender side selection of multiple algorithms for the sender-side selection of which streams to
which streams to send data from, supporting a variety of scheduling send data from, supporting a variety of scheduling algorithms
algorithms including priority based methods. The stream re- including priority-based methods. The stream reconfiguration
configuration extension defined in [RFC6525] allows streams to be extension defined in [RFC6525] allows streams to be reset during the
reset during the lifetime of an association and to increase the lifetime of an association and to increase the number of streams, if
number of streams, if the number of streams negotiated in the SCTP the number of streams negotiated in the SCTP handshake becomes
handshake becomes insufficient. insufficient.
Each user message sent is either delivered to the receiver or, in Each user message sent is delivered to the receiver or, in case of
case of excessive retransmissions, the association is terminated in a excessive retransmissions, the association is terminated in a
non-graceful way [RFC4960], similar to TCP behavior. In addition to non-graceful way [RFC4960], similar to TCP behavior. In addition to
this reliable transfer, the partial reliability extension [RFC3758] this reliable transfer, the partial reliability extension [RFC3758]
allows a sender to abandon user messages. The application can allows a sender to abandon user messages. The application can
specify the policy for abandoning user messages. specify the policy for abandoning user messages.
SCTP supports multi-homing. Each SCTP endpoint uses a list of IP- SCTP supports multihoming. Each SCTP endpoint uses a list of IP
addresses and a single port number. These addresses can be any addresses and a single port number. These addresses can be any
mixture of IPv4 and IPv6 addresses. These addresses are negotiated mixture of IPv4 and IPv6 addresses. These addresses are negotiated
during the handshake and the address re-configuration extension during the handshake, and the address reconfiguration extension
specified in [RFC5061] in combination with [RFC4895] can be used to specified in [RFC5061] in combination with [RFC4895] can be used to
change these addresses in an authenticated way during the lifetime of change these addresses in an authenticated way during the lifetime of
an SCTP association. This allows for transport layer mobility. an SCTP association. This allows for transport-layer mobility.
Multiple addresses are used for improved resilience. If a remote Multiple addresses are used for improved resilience. If a remote
address becomes unreachable, the traffic is switched over to a address becomes unreachable, the traffic is switched over to a
reachable one, if one exists. reachable one, if one exists.
For securing user messages, the use of TLS over SCTP has been For securing user messages, the use of TLS over SCTP has been
specified in [RFC3436]. However, this solution does not support all specified in [RFC3436]. However, this solution does not support all
services provided by SCTP, such as un-ordered delivery or partial services provided by SCTP, such as unordered delivery or partial
reliability. Therefore, the use of DTLS over SCTP has been specified reliability. Therefore, the use of DTLS over SCTP has been specified
in [RFC6083] to overcome these limitations. When using DTLS over in [RFC6083] to overcome these limitations. When using DTLS over
SCTP, the application can use almost all services provided by SCTP. SCTP, the application can use almost all services provided by SCTP.
[I-D.ietf-tsvwg-natsupp] defines methods for endpoints and [NAT-SUPP] defines methods for endpoints and middleboxes to provide
middleboxes to provide NAT traversal for SCTP over IPv4. For legacy NAT traversal for SCTP over IPv4. For legacy NAT traversal,
NAT traversal, [RFC6951] defines the UDP encapsulation of SCTP- [RFC6951] defines the UDP encapsulation of SCTP packets.
packets. Alternatively, SCTP packets can be encapsulated in DTLS Alternatively, SCTP packets can be encapsulated in DTLS packets as
packets as specified in [I-D.ietf-tsvwg-sctp-dtls-encaps]. The specified in [SCTP-DTLS-ENCAPS]. The latter encapsulation is used
latter encapsulation is used within the WebRTC within the WebRTC [WEBRTC-TRANS] context.
[I-D.ietf-rtcweb-transports] context.
An SCTP ABORT chunk may be used to force a SCTP endpoint to close a An SCTP ABORT chunk may be used to force a SCTP endpoint to close a
session [RFC4960], aborting the connection. session [RFC4960], aborting the connection.
SCTP has a well-defined API, described in the next subsection. SCTP has a well-defined API, described in the next subsection.
3.5.2. Interface Description 3.5.2. Interface Description
[RFC4960] defines an abstract API for the base protocol. This API [RFC4960] defines an abstract API for the base protocol. This API
describes the following functions callable by the upper layer of describes the following functions callable by the upper layer of
SCTP: Initialize, Associate, Send, Receive, Receive Unsent Message, SCTP: Initialize, Associate, Send, Receive, Receive Unsent Message,
Receive Unacknowledged Message, Shutdown, Abort, SetPrimary, Status, Receive Unacknowledged Message, Shutdown, Abort, SetPrimary, Status,
Change Heartbeat, Request Heartbeat, Get SRTT Report, Set Failure Change Heartbeat, Request Heartbeat, Get SRTT Report, Set Failure
Threshold, Set Protocol Parameters, and Destroy. The following Threshold, Set Protocol Parameters, and Destroy. The following
notifications are provided by the SCTP stack to the upper layer: notifications are provided by the SCTP stack to the upper layer:
COMMUNICATION UP, DATA ARRIVE, SHUTDOWN COMPLETE, COMMUNICATION LOST, COMMUNICATION UP, DATA ARRIVE, SHUTDOWN COMPLETE, COMMUNICATION LOST,
COMMUNICATION ERROR, RESTART, SEND FAILURE, NETWORK STATUS CHANGE. COMMUNICATION ERROR, RESTART, SEND FAILURE, and NETWORK STATUS
CHANGE.
An extension to the BSD Sockets API is defined in [RFC6458] and An extension to the BSD Sockets API is defined in [RFC6458] and
covers: covers:
o the base protocol defined in [RFC4960]. The API allows control o the base protocol defined in [RFC4960]. The API allows control
over local addresses and port numbers and the primary path. over local addresses and port numbers and the primary path.
Furthermore the application has fine control about parameters like Furthermore, the application has fine control of parameters like
retransmission thresholds, the path supervision parameters, the retransmission thresholds, the path supervision, the delayed
delayed acknowledgment timeout, and the fragmentation point. The acknowledgment timeout, and the fragmentation point. The API
API provides a mechanism to allow the SCTP stack to notify the provides a mechanism to allow the SCTP stack to notify the
application about events if the application has requested them. application about events if the application has requested them.
These notifications provide information about status changes of These notifications provide information about status changes of
the association and each of the peer addresses. In case of send the association and each of the peer addresses. In case of send
failures, including drop of messages sent unreliably, the failures, including drop of messages sent unreliably, the
application can also be notified and user messages can be returned application can also be notified, and user messages can be
to the application. When sending user messages, the stream id, a returned to the application. When sending user messages, the
payload protocol identifier, an indication whether ordered application can indicate a stream id, a payload protocol
delivery is requested or not. These parameters can also be identifier, and an indication of whether ordered delivery is
provided on message reception. Additionally a context can be requested. These parameters can also be provided on message
provided when sending, which can be use in case of send failures. reception. Additionally, a context can be provided when sending,
The sending of arbitrary large user messages is supported. which can be used in case of send failures. The sending of
arbitrarily large user messages is supported.
o the SCTP Partial Reliability extension defined in [RFC3758] to o the SCTP Partial Reliability extension defined in [RFC3758] to
specify for a user message the PR-SCTP policy and the policy specify for a user message the Partially Reliable SCTP (PR-SCTP)
specific parameter. Examples of these policies defined in policy and the policy-specific parameter. Examples of these
[RFC3758] and [RFC7496] are: policies defined in [RFC3758] and [RFC7496] are:
o Limiting the time a user message is dealt with by the sender. * limiting the time a user message is dealt with by the sender.
o Limiting the number of retransmissions for each fragment of a user * limiting the number of retransmissions for each fragment of a
message. If the number of retransmissions is limited to 0, one user message. If the number of retransmissions is limited to
gets a service similar to UDP. 0, one gets a service similar to UDP.
o Abandoning messages of lower priority in case of a send buffer * abandoning messages of lower priority in case of a send buffer
shortage. shortage.
o the SCTP Authentication extension defined in [RFC4895] allowing to o the SCTP Authentication extension defined in [RFC4895] allowing
manage the shared keys, the HMAC to use, set the chunk types which management of the shared keys and allowing the HMAC to use and set
are only accepted in an authenticated way, and get the list of the chunk types (which are only accepted in an authenticated way)
chunks which are accepted by the local and remote end point in an and get the list of chunks that are accepted by the local and
authenticated way. remote endpoints in an authenticated way.
o the SCTP Dynamic Address Reconfiguration extension defined in o the SCTP Dynamic Address Reconfiguration extension defined in
[RFC5061]. It allows to manually add and delete local addresses [RFC5061]. It allows the manual addition and deletion of local
for SCTP associations and the enabling of automatic address addresses for SCTP associations, as well as the enabling of
addition and deletion. Furthermore the peer can be given a hint automatic address addition and deletion. Furthermore, the peer
for choosing its primary path. can be given a hint for choosing its primary path.
A BSD Sockets API extension has been defined in the documents that A BSD Sockets API extension has been defined in the documents that
specify the following SCTP protocol extensions: specify the following SCTP extensions:
o the SCTP Stream Reconfiguration extension defined in [RFC6525]. o the SCTP Stream Reconfiguration extension defined in [RFC6525].
The API allows to trigger the reset operation for incoming and The API allows triggering of the reset operation for incoming and
outgoing streams and the whole association. It provides also a outgoing streams and the whole association. It also provides a
way to notify the association about the corresponding events. way to notify the association about the corresponding events.
Furthermore the application can increase the number of streams. Furthermore, the application can increase the number of streams.
o the UDP Encapsulation of SCTP packets extension defined in o the UDP Encapsulation of SCTP packets extension defined in
[RFC6951]. The API allows the management of the remote UDP [RFC6951]. The API allows the management of the remote UDP
encapsulation port. encapsulation port.
o the SCTP SACK-IMMEDIATELY extension defined in [RFC7053]. The API o the SCTP SACK-IMMEDIATELY extension defined in [RFC7053]. The API
allows the sender of a user message to request the receiver to allows the sender of a user message to request the receiver to
send the corresponding acknowledgment immediately. send the corresponding acknowledgment immediately.
o the additional PR-SCTP policies defined in [RFC7496]. The API o the additional PR-SCTP policies defined in [RFC7496]. The API
allows to enable/disable the PR-SCTP extension, choose the PR-SCTP allows enabling/disabling the PR-SCTP extension, choosing the
policies defined in the document and provide statistical PR-SCTP policies defined in the document, and providing
information about abandoned messages. statistical information about abandoned messages.
Future documents describing SCTP protocol extensions are expected to Future documents describing SCTP extensions are expected to describe
describe the corresponding BSD Sockets API extension in a "Socket API the corresponding BSD Sockets API extension in a "Socket API
Considerations" section. Considerations" section.
The SCTP socket API supports two kinds of sockets: The SCTP Socket API supports two kinds of sockets:
o one-to-one style sockets (by using the socket type "SOCK_STREAM"). o one-to-one style sockets (by using the socket type "SOCK_STREAM").
o one-to-many style socket (by using the socket type o one-to-many style socket (by using the socket type
"SOCK_SEQPACKET"). "SOCK_SEQPACKET").
One-to-one style sockets are similar to TCP sockets, there is a 1:1 One-to-one style sockets are similar to TCP sockets; there is a 1:1
relationship between the sockets and the SCTP associations (except relationship between the sockets and the SCTP associations (except
for listening sockets). One-to-many style SCTP sockets are similar for listening sockets). One-to-many style SCTP sockets are similar
to unconnected UDP sockets, where there is a 1:n relationship between to unconnected UDP sockets, where there is a 1:n relationship between
the sockets and the SCTP associations. the sockets and the SCTP associations.
The SCTP stack can provide information to the applications about The SCTP stack can provide information to the applications about
state changes of the individual paths and the association whenever state changes of the individual paths and the association whenever
they occur. These events are delivered similar to user messages but they occur. These events are delivered similarly to user messages
are specifically marked as notifications. but are specifically marked as notifications.
New functions have been introduced to support the use of multiple New functions have been introduced to support the use of multiple
local and remote addresses. Additional SCTP-specific send and local and remote addresses. Additional SCTP-specific send and
receive calls have been defined to permit SCTP-specific information receive calls have been defined to permit SCTP-specific information
to be sent without using ancillary data in the form of additional to be sent without using ancillary data in the form of additional
cmsgs. These functions provide support for detecting partial Control Message (cmsg) calls. These functions provide support for
delivery of user messages and notifications. detecting partial delivery of user messages and notifications.
The SCTP socket API allows a fine-grained control of the protocol The SCTP Socket API allows a fine-grained control of the protocol
behavior through an extensive set of socket options. behavior through an extensive set of socket options.
The SCTP kernel implementations of FreeBSD, Linux and Solaris follow The SCTP kernel implementations of FreeBSD, Linux, and Solaris follow
mostly the specified extension to the BSD Sockets API for the base mostly the specified extension to the BSD Sockets API for the base
protocol and the corresponding supported protocol extensions. protocol and the corresponding supported protocol extensions.
3.5.3. Transport Features 3.5.3. Transport Features
The transport features provided by SCTP are: The transport features provided by SCTP are:
o connection-oriented transport with feature negotiation and o connection-oriented transport with feature negotiation and
application-to-port mapping, application-to-port mapping,
o unicast transport, o unicast transport,
o port multiplexing, o port multiplexing,
o uni- or bidirectional communication, o unidirectional or bidirectional communication,
o message-oriented delivery with durable message framing supporting o message-oriented delivery with durable message framing supporting
multiple concurrent streams, multiple concurrent streams,
o fully reliable, partially reliable, or unreliable delivery (based o fully reliable, partially reliable, or unreliable delivery (based
on user specified policy to handle abandoned user messages) with on user-specified policy to handle abandoned user messages) with
drop notification, drop notification,
o ordered and unordered delivery within a stream, o ordered and unordered delivery within a stream,
o support for stream scheduling prioritization, o support for stream scheduling prioritization,
o segmentation, o segmentation,
o user message bundling, o user message bundling,
o flow control using a window-based mechanism, o flow control using a window-based mechanism,
o congestion control using methods similar to TCP, o congestion control using methods similar to TCP,
o strong error detection (CRC32c),
o transport layer multihoming for resilience and mobility. o strong error detection (CRC32c), and
o transport-layer multihoming for resilience and mobility.
3.6. Datagram Congestion Control Protocol (DCCP) 3.6. Datagram Congestion Control Protocol (DCCP)
Datagram Congestion Control Protocol (DCCP) [RFC4340] is an IETF The Datagram Congestion Control Protocol (DCCP) [RFC4340] is an IETF
standards track bidirectional transport protocol that provides Standards Track bidirectional transport protocol that provides
unicast connections of congestion-controlled messages without unicast connections of congestion-controlled messages without
providing reliability. providing reliability.
The DCCP Problem Statement describes the goals that DCCP sought to The DCCP Problem Statement [RFC4336] describes the goals that DCCP
address [RFC4336]: It is suitable for applications that transfer sought to address. It is suitable for applications that transfer
fairly large amounts of data and that can benefit from control over fairly large amounts of data and that can benefit from control over
the trade off between timeliness and reliability [RFC4336]. the trade-off between timeliness and reliability [RFC4336].
DCCP offers low overhead, and many characteristics common to UDP, but DCCP offers low overhead, and many characteristics common to UDP, but
can avoid "re-inventing the wheel" each time a new multimedia can avoid "re-inventing the wheel" each time a new multimedia
application emerges. Specifically it includes core transport application emerges. Specifically, it includes core transport
functions (feature negotiation, path state management, RTT functions (feature negotiation, path state management, RTT
calculation, PMTUD, etc.): DCCP applications select how they send calculation, PMTUD, etc.): DCCP applications select how they send
packets and, where suitable, choose common algorithms to manage their packets and, where suitable, choose common algorithms to manage their
functions. Examples of applications that can benefit from such functions. Examples of applications that can benefit from such
transport services include interactive applications, streaming media, transport services include interactive applications, streaming media,
or on-line games [RFC4336]. or on-line games [RFC4336].
3.6.1. Protocol Description 3.6.1. Protocol Description
DCCP is a connection-oriented datagram protocol, providing a three- DCCP is a connection-oriented datagram protocol that provides a
way handshake to allow a client and server to set up a connection, three-way handshake to allow a client and server to set up a
and mechanisms for orderly completion and immediate teardown of a connection and provides mechanisms for orderly completion and
connection. immediate teardown of a connection.
A DCCP protocol instance can be extended [RFC4340] and tuned using A DCCP protocol instance can be extended [RFC4340] and tuned using
additional features. Some features are sender-side only, requiring additional features. Some features are sender-side only, requiring
no negotiation with the receiver; some are receiver-side only; and no negotiation with the receiver; some are receiver-side only; and
some are explicitly negotiated during connection setup. some are explicitly negotiated during connection setup.
DCCP uses a Connect packet to initiate a session, and permits each DCCP uses a Connect packet to initiate a session and permits each
endpoint to choose the features it wishes to support. Simultaneous endpoint to choose the features it wishes to support. Simultaneous
open [RFC5596], as in TCP, can enable interoperability in the open [RFC5596], as in TCP, can enable interoperability in the
presence of middleboxes. The Connect packet includes a Service Code presence of middleboxes. The Connect packet includes a Service Code
[RFC5595] that identifies the application or protocol using DCCP, [RFC5595] that identifies the application or protocol using DCCP,
providing middleboxes with information about the intended use of a providing middleboxes with information about the intended use of a
connection. connection.
The DCCP service is unicast-only. The DCCP service is unicast-only.
It provides multiplexing to multiple sockets at each endpoint using It provides multiplexing to multiple sockets at each endpoint using
port numbers. An active DCCP session is identified by its four-tuple port numbers. An active DCCP session is identified by its four-tuple
of local and remote IP addresses and local port and remote port of local and remote IP addresses and local and remote port numbers.
numbers.
The protocol segments data into messages, typically sized to fit in The protocol segments data into messages that are typically sized to
IP packets, but which may be fragmented providing they are smaller fit in IP packets but may be fragmented if they are smaller than the
than the maximum packet size. A DCCP interface allows applications maximum packet size. A DCCP interface allows applications to request
to request fragmentation for packets larger than PMTU, but not larger fragmentation for packets larger than PMTU, but not larger than the
than the maximum packet size allowed by the current congestion maximum packet size allowed by the current congestion control
control mechanism (CCMPS) [RFC4340]. mechanism (Congestion Control Maximum Packet Size (CCMPS)) [RFC4340].
Each message is identified by a sequence number. The sequence number Each message is identified by a sequence number. The sequence number
is used to identify segments in acknowledgments, to detect is used to identify segments in acknowledgments, to detect
unacknowledged segments, to measure RTT, etc. The protocol may unacknowledged segments, to measure RTT, etc. The protocol may
support unordered delivery of data, and does not itself provide support unordered delivery of data and does not itself provide
retransmission. DCCP supports reduced checksum coverage, a partial retransmission. DCCP supports reduced checksum coverage, a partial
payload protection mechanism similar to UDP-Lite. There is also a payload protection mechanism similar to UDP-Lite. There is also a
Data Checksum option, which when enabled, contains a strong CRC, to Data Checksum option, which when enabled, contains a strong Cyclic
enable endpoints to detect application data corruption. Redundancy Check (CRC), to enable endpoints to detect application
data corruption.
Receiver flow control is supported, which limits the amount of Receiver flow control is supported, which limits the amount of
unacknowledged data that can be outstanding at a given time. unacknowledged data that can be outstanding at a given time.
A DCCP Reset packet may be used to force a DCCP endpoint to close a A DCCP Reset packet may be used to force a DCCP endpoint to close a
session [RFC4340], aborting the connection. session [RFC4340], aborting the connection.
DCCP supports negotiation of the congestion control profile between DCCP supports negotiation of the congestion control profile between
endpoints, to provide plug-and-play congestion control mechanisms. endpoints, to provide plug-and-play congestion control mechanisms.
Examples of specified profiles include "TCP-like" [RFC4341], "TCP- Examples of specified profiles include "TCP-like" [RFC4341], "TCP-
friendly" [RFC4342], and "TCP-friendly for small packets" [RFC5622]. friendly" [RFC4342], and "TCP-friendly for small packets" [RFC5622].
Additional mechanisms are recorded in an IANA registry. Additional mechanisms are recorded in an IANA registry (see
<http://www.iana.org/assignments/dccp-parameters>).
A lightweight UDP-based encapsulation (DCCP-UDP) has been defined A lightweight UDP-based encapsulation (DCCP-UDP) has been defined
[RFC6773] that permits DCCP to be used over paths where DCCP is not [RFC6773] that permits DCCP to be used over paths where DCCP is not
natively supported. Support for DCCP in NAPT/NATs is defined in natively supported. Support for DCCP in NAPT/NATs is defined in
[RFC4340] and [RFC5595]. Upper layer protocols specified on top of [RFC4340] and [RFC5595]. Upper-layer protocols specified on top of
DCCP include DTLS [RFC5595], RTP [RFC5672], ICE/SDP [RFC6773]. DCCP include DTLS [RFC5238], RTP [RFC5762], and Interactive
Connectivity Establishment / Session Description Protocol (ICE/SDP)
[RFC6773].
3.6.2. Interface Description 3.6.2. Interface Description
Functions expected for a DCCP API include: Open, Close and Management Functions expected for a DCCP API include: Open, Close, and
of the progress a DCCP connection. The Open function provides Management of the progress a DCCP connection. The Open function
feature negotiation, selection of an appropriate CCID for congestion provides feature negotiation, selection of an appropriate Congestion
control and other parameters associated with the DCCP connection. A Control Identifier (CCID) for congestion control, and other
function allows an application to send DCCP datagrams, including parameters associated with the DCCP connection. A function allows an
setting the required checksum coverage, and any required options. application to send DCCP datagrams, including setting the required
(DCCP permits sending datagrams with a zero-length payload.) A checksum coverage and any required options. (DCCP permits sending
function allows reception of data, including indicating if the data datagrams with a zero-length payload.) A function allows reception
was used or dropped. Functions can also make the status of a of data, including indicating if the data was used or dropped.
connection visible to an application, including detection of the Functions can also make the status of a connection visible to an
maximum packet size and the ability to perform flow control by application, including detection of the maximum packet size and the
detecting a slow receiver at the sender. ability to perform flow control by detecting a slow receiver at the
sender.
There is no API currently specified in the RFC Series. There is no API currently specified in the RFC Series.
3.6.3. Transport Features 3.6.3. Transport Features
The transport features provided by DCCP are: The transport features provided by DCCP are:
o unicast transport, o unicast transport,
o connection-oriented communication with feature negotiation and o connection-oriented communication with feature negotiation and
application-to-port mapping, application-to-port mapping,
o signaling of application class for middlebox support (implemented o signaling of application class for middlebox support (implemented
using Service Codes), using Service Codes),
o port multiplexing, o port multiplexing,
o uni-or bidirectional communication, o unidirectional or bidirectional communication,
o message-oriented delivery, o message-oriented delivery,
o unreliable delivery with drop notification, o unreliable delivery with drop notification,
o unordered delivery, o unordered delivery,
o flow control (implemented using the slow receiver function) o flow control (implemented using the slow receiver function), and
o partial and full payload error detection (with optional strong o partial and full payload error detection (with optional strong
integrity check). integrity check).
3.7. Transport Layer Security (TLS) and Datagram TLS (DTLS) as a 3.7. Transport Layer Security (TLS) and Datagram TLS (DTLS) as a
pseudotransport Pseudotransport
Transport Layer Security (TLS) [RFC5246] and Datagram TLS (DTLS) Transport Layer Security (TLS) [RFC5246] and Datagram TLS (DTLS)
[RFC6347] are IETF protocols that provide several security-related [RFC6347] are IETF protocols that provide several security-related
features to applications. TLS is designed to run on top of a features to applications. TLS is designed to run on top of a
reliable streaming transport protocol (usually TCP), while DTLS is reliable streaming transport protocol (usually TCP), while DTLS is
designed to run on top of a best-effort datagram protocol (UDP or 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 DCCP [RFC5238]). At the time of writing, the current version of TLS
is 1.2, defined in [RFC5246]; work on TLS version 1.3 is 1.2, defined in [RFC5246]; work on TLS version is 1.3 [TLS-1.3]
[I-D.ietf-tls-tls13] nearing completion. DTLS provides nearly nearing completion. DTLS provides nearly identical functionality to
identical functionality to applications; it is defined in [RFC6347] applications; it is defined in [RFC6347] and its current version is
and its current version is also 1.2. The TLS protocol evolved from also 1.2. The TLS protocol evolved from the Secure Sockets Layer
the Secure Sockets Layer (SSL) [RFC6101] protocols developed in the (SSL) [RFC6101] protocols developed in the mid-1990s to support
mid-1990s to support protection of HTTP traffic. protection of HTTP traffic.
While older versions of TLS and DTLS are still in use, they provide While older versions of TLS and DTLS are still in use, they provide
weaker security guarantees. [RFC7457] outlines important attacks on weaker security guarantees. [RFC7457] outlines important attacks on
TLS and DTLS. [RFC7525] is a Best Current Practices (BCP) document TLS and DTLS. [RFC7525] is a Best Current Practices (BCP) document
that describes secure configurations for TLS and DTLS to counter that describes secure configurations for TLS and DTLS to counter
these attacks. The recommendations are applicable for the vast these attacks. The recommendations are applicable for the vast
majority of use cases. majority of use cases.
3.7.1. Protocol Description 3.7.1. Protocol Description
skipping to change at page 23, line 44 skipping to change at page 24, line 21
application's decision whether the certificate of the peering entity application's decision whether the certificate of the peering entity
is acceptable for authorization decisions. is acceptable for authorization decisions.
Perfect forward secrecy, if enabled and supported by the selected Perfect forward secrecy, if enabled and supported by the selected
algorithms, ensures that traffic encrypted and captured during a algorithms, ensures that traffic encrypted and captured during a
session at time t0 cannot be later decrypted at time t1 (t1 > t0), 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 even if the long-term secrets of the communicating peers are later
compromised. compromised.
As DTLS is generally used over an unreliable datagram transport such As DTLS is generally used over an unreliable datagram transport such
as UDP, applications will need to tolerate lost, re-ordered, or as UDP, applications will need to tolerate lost, reordered, or
duplicated datagrams. Like TLS, DTLS conveys application data in a duplicated datagrams. Like TLS, DTLS conveys application data in a
sequence of independent records. However, because records are mapped sequence of independent records. However, because records are mapped
to unreliable datagrams, there are several features unique to DTLS to unreliable datagrams, there are several features unique to DTLS
that are not applicable to TLS: that are not applicable to TLS:
o Record replay detection (optional). o Record replay detection (optional).
o Record size negotiation (estimates of PMTU and record size o Record size negotiation (estimates of PMTU and record size
expansion factor). expansion factor).
skipping to change at page 24, line 33 skipping to change at page 25, line 9
TLS is commonly invoked using an API provided by packages such as TLS is commonly invoked using an API provided by packages such as
OpenSSL, wolfSSL, or GnuTLS. Using such APIs entails the OpenSSL, wolfSSL, or GnuTLS. Using such APIs entails the
manipulation of several important abstractions, which fall into the manipulation of several important abstractions, which fall into the
following categories: long-term keys and algorithms, session state, following categories: long-term keys and algorithms, session state,
and communications/connections. and communications/connections.
Considerable care is required in the use of TLS APIs to ensure Considerable care is required in the use of TLS APIs to ensure
creation of a secure application. The programmer should have at creation of a secure application. The programmer should have at
least a basic understanding of encryption and digital signature least a basic understanding of encryption and digital signature
algorithms and their strengths, public key infrastructure (including algorithms and their strengths, public key infrastructure (including
X.509 certificates and certificate revocation), and the sockets API. X.509 certificates and certificate revocation), and the Sockets API.
See [RFC7525] and [RFC7457], as mentioned above. See [RFC7525] and [RFC7457], as mentioned above.
As an example, in the case of OpenSSL, the primary abstractions are As an example, in the case of OpenSSL, the primary abstractions are
the library itself and method (protocol), session, context, cipher the library itself, method (protocol), session, context, cipher, and
and connection. After initializing the library and setting the connection. After initializing the library and setting the method, a
method, a cipher suite is chosen and used to configure a context cipher suite is chosen and used to configure a context object.
object. Session objects may then be minted according to the Session objects may then be minted according to the parameters
parameters present in a context object and associated with individual present in a context object and associated with individual
connections. Depending on how precisely the programmer wishes to connections. Depending on how precisely the programmer wishes to
select different algorithmic or protocol options, various levels of select different algorithmic or protocol options, various levels of
details may be required. details may be required.
3.7.3. Transport Features 3.7.3. Transport Features
Both TLS and DTLS employ a layered architecture. The lower layer is Both TLS and DTLS employ a layered architecture. The lower layer is
commonly called the record protocol. It is responsible for: commonly called the "record protocol". It is responsible for:
o message fragmentation, o message fragmentation,
o authentication and integrity via message authentication codes o authentication and integrity via message authentication codes
(MAC), (MACs),
o data encryption, o data encryption, and
o scheduling transmission using the underlying transport protocol. o scheduling transmission using the underlying transport protocol.
DTLS augments the TLS record protocol with: DTLS augments the TLS record protocol with:
o ordering and replay protection, implemented using sequence o ordering and replay protection, implemented using sequence
numbers. numbers.
Several protocols are layered on top of the record protocol. These Several protocols are layered on top of the record protocol. These
include the handshake, alert, and change cipher spec protocols. include the handshake, alert, and change cipher spec protocols.
skipping to change at page 25, line 32 skipping to change at page 26, line 9
capability and a cookie-like mechanism to resist DoS attacks. (TLS capability and a cookie-like mechanism to resist DoS attacks. (TLS
compression is not recommended at present). The alert protocol is compression is not recommended at present). The alert protocol is
used to inform the peer of various conditions, most of which are used to inform the peer of various conditions, most of which are
terminal for the connection. The change cipher spec protocol is used terminal for the connection. The change cipher spec protocol is used
to synchronize changes in cryptographic parameters for each peer. to synchronize changes in cryptographic parameters for each peer.
The data protocol, when used with an appropriate cipher, provides: The data protocol, when used with an appropriate cipher, provides:
o authentication of one end or both ends of a connection, o authentication of one end or both ends of a connection,
o confidentiality, o confidentiality, and
o cryptographic integrity protection. o cryptographic integrity protection.
Both TLS and DTLS are unicast-only. Both TLS and DTLS are unicast-only.
3.8. Realtime Transport Protocol (RTP) 3.8. Real-Time Transport Protocol (RTP)
RTP provides an end-to-end network transport service, suitable for RTP provides an end-to-end network transport service, suitable for
applications transmitting real-time data, such as audio, video or applications transmitting real-time data, such as audio, video or
data, over multicast or unicast transport services, including TCP, data, over multicast or unicast transport services, including TCP,
UDP, UDP-Lite, DCCP, TLS and DTLS. UDP, UDP-Lite, DCCP, TLS, and DTLS.
3.8.1. Protocol Description 3.8.1. Protocol Description
The RTP standard [RFC3550] defines a pair of protocols, RTP and the The RTP standard [RFC3550] defines a pair of protocols: RTP and the
RTP control protocol, RTCP. The transport does not provide RTP Control Protocol (RTCP). The transport does not provide
connection setup, instead relying on out-of-band techniques or connection setup, instead relying on out-of-band techniques or
associated control protocols to setup, negotiate parameters or tear associated control protocols to setup, negotiate parameters, or tear
down a session. down a session.
An RTP sender encapsulates audio/video data into RTP packets to An RTP sender encapsulates audio/video data into RTP packets to
transport media streams. The RFC-series specifies RTP payload transport media streams. The RFC Series specifies RTP payload
formats that allow packets to carry a wide range of media, and formats that allow packets to carry a wide range of media and
specifies a wide range of multiplexing, error control and other specifies a wide range of multiplexing, error control, and other
support mechanisms. support mechanisms.
If a frame of media data is large, it will be fragmented into several If a frame of media data is large, it will be fragmented into several
RTP packets. Likewise, several small frames may be bundled into a RTP packets. Likewise, several small frames may be bundled into a
single RTP packet. single RTP packet.
An RTP receiver collects RTP packets from the network, validates them An RTP receiver collects RTP packets from the network, validates them
for correctness, and sends them to the media decoder input-queue. for correctness, and sends them to the media decoder input queue.
Missing packet detection is performed by the channel decoder. The Missing packet detection is performed by the channel decoder. The
play-out buffer is ordered by time stamp and is used to reorder playout buffer is ordered by time stamp and is used to reorder
packets. Damaged frames may be repaired before the media payloads packets. Damaged frames may be repaired before the media payloads
are decompressed to display or store the data. Some uses of RTP are are decompressed to display or store the data. Some uses of RTP are
able to exploit the partial payload protection features offered by able to exploit the partial payload protection features offered by
DCCP and UDP-Lite. DCCP and UDP-Lite.
RTCP is a control protocol that works alongside an RTP flow. Both RTCP is a control protocol that works alongside an RTP flow. Both
the RTP sender and receiver will send RTCP report packets. This is the RTP sender and receiver will send RTCP report packets. This is
used to periodically send control information and report performance. used to periodically send control information and report performance.
Based on received RTCP feedback, an RTP sender can adjust the Based on received RTCP feedback, an RTP sender can adjust the
transmission, e.g., perform rate adaptation at the application layer transmission, e.g., perform rate adaptation at the application layer
in the case of congestion. in the case of congestion.
An RTCP receiver report (RTCP RR) is returned to the sender An RTCP receiver report (RTCP RR) is returned to the sender
periodically to report key parameters (e.g, the fraction of packets periodically to report key parameters (e.g., the fraction of packets
lost in the last reporting interval, the cumulative number of packets lost in the last reporting interval, the cumulative number of packets
lost, the highest sequence number received, and the inter-arrival lost, the highest sequence number received, and the inter-arrival
jitter). The RTCP RR packets also contain timing information that jitter). The RTCP RR packets also contain timing information that
allows the sender to estimate the network round trip time (RTT) to allows the sender to estimate the network round-trip time (RTT) to
the receivers. the receivers.
The interval between reports sent from each receiver tends to be on 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 the order of a few seconds on average, although this varies with the
session rate, and sub-second reporting intervals are possible for session rate, and sub-second reporting intervals are possible for
high rate sessions. The interval is randomized to avoid high rate sessions. The interval is randomized to avoid
synchronization of reports from multiple receivers. synchronization of reports from multiple receivers.
3.8.2. Interface Description 3.8.2. Interface Description
There is no standard application programming interface defined for There is no standard API defined for RTP or RTCP. Implementations
RTP or RTCP. Implementations are typically tightly integrated with a are typically tightly integrated with a particular application and
particular application, and closely follow the principles of closely follow the principles of application-level framing and
application level framing and integrated layer processing [ClarkArch] integrated layer processing [ClarkArch] in media processing
in media processing [RFC2736], error recovery and concealment, rate [RFC2736], error recovery and concealment, rate adaptation, and
adaptation, and security [RFC7202]. Accordingly, RTP implementations security [RFC7202]. Accordingly, RTP implementations tend to be
tend to be targeted at particular application domains (e.g., voice- targeted at particular application domains (e.g., voice-over-IP,
over-IP, IPTV, or video conferencing), with a feature set optimized IPTV, or video conferencing), with a feature set optimized for that
for that domain, rather than being general purpose implementations of domain, rather than being general purpose implementations of the
the protocol. protocol.
3.8.3. Transport Features 3.8.3. Transport Features
The transport features provided by RTP are: The transport features provided by RTP are:
o unicast, multicast or IPv4 broadcast (provided by lower layer o unicast, multicast, or IPv4 broadcast (provided by lower-layer
protocol), protocol),
o port multiplexing (provided by lower layer protocol), o port multiplexing (provided by lower-layer protocol),
o uni- or bidirectional communication (provided by lower layer o unidirectional or bidirectional communication (provided by lower-
protocol), layer protocol),
o message-oriented delivery with support for media types and other o message-oriented delivery with support for media types and other
extensions, extensions,
o reliable delivery when using erasure coding or unreliable delivery o reliable delivery when using erasure coding or unreliable delivery
with drop notification (if supported by lower layer protocol), with drop notification (if supported by lower-layer protocol),
o connection setup with feature negotiation (using associated o connection setup with feature negotiation (using associated
protocols) and application-to-port mapping (provided by lower protocols) and application-to-port mapping (provided by lower-
layer protocol), layer protocol),
o segmentation, o segmentation, and
o performance metric reporting (using associated protocols). o performance metric reporting (using associated protocols).
3.9. Hypertext Transport Protocol (HTTP) over TCP as a pseudotransport 3.9. Hypertext Transport Protocol (HTTP) over TCP as a Pseudotransport
The Hypertext Transfer Protocol (HTTP) is an application-level The Hypertext Transfer Protocol (HTTP) is an application-level
protocol widely used on the Internet. It provides object-oriented protocol widely used on the Internet. It provides object-oriented
delivery of discrete data or files. Version 1.1 of the protocol is delivery of discrete data or files. Version 1.1 of the protocol is
specified in [RFC7230] [RFC7231] [RFC7232] [RFC7233] [RFC7234] specified in [RFC7230] [RFC7231] [RFC7232] [RFC7233] [RFC7234]
[RFC7235], and version 2 in [RFC7540]. HTTP is usually transported [RFC7235], and version 2 is specified in [RFC7540]. HTTP is usually
over TCP using port 80 and 443, although it can be used with other transported over TCP using ports 80 and 443, although it can be used
transports. When used over TCP it inherits TCP's properties. with other transports. When used over TCP, it inherits TCP's
properties.
Application layer protocols may use HTTP as a substrate with an Application-layer protocols may use HTTP as a substrate with an
existing method and data formats, or specify new methods and data existing method and data formats, or specify new methods and data
formats. There are various reasons for this practice listed in formats. There are various reasons for this practice listed in
[RFC3205]; these include being a well-known and well-understood [RFC3205]; these include being a well-known and well-understood
protocol, reusability of existing servers and client libraries, easy protocol, reusability of existing servers and client libraries, easy
use of existing security mechanisms such as HTTP digest use of existing security mechanisms such as HTTP digest
authentication [RFC2617] and TLS [RFC5246], the ability of HTTP to authentication [RFC7235] and TLS [RFC5246], and the ability of HTTP
traverse firewalls makes it work over many types of infrastructure, to traverse firewalls, which allows it to work over many types of
and in cases where an application server often needs to support HTTP infrastructure and in cases where an application server often needs
anyway. to support HTTP anyway.
Depending on application need, the use of HTTP as a substrate Depending on application need, the use of HTTP as a substrate
protocol may add complexity and overhead in comparison to a special- protocol may add complexity and overhead in comparison to a special-
purpose protocol (e.g., HTTP headers, suitability of the HTTP purpose protocol (e.g., HTTP headers, suitability of the HTTP
security model, etc.). [RFC3205] addresses this issue and provides security model, etc.). [RFC3205] addresses this issue, provides some
some guidelines and identifies concerns about the use of HTTP guidelines, and identifies concerns about the use of HTTP standard
standard port 80 and 443, the use of HTTP URL scheme and interaction ports 80 and 443, the use of the HTTP URL scheme, and interaction
with existing firewalls, proxies and NATs. with existing firewalls, proxies, and NATs.
Representational State Transfer (REST) [REST] is another example of Representational State Transfer (REST) [REST] is another example of
how applications can use HTTP as transport protocol. REST is an how applications can use HTTP as a transport protocol. REST is an
architecture style that may be used to build applications using HTTP architecture style that may be used to build applications using HTTP
as a communication protocol. as a communication protocol.
3.9.1. Protocol Description 3.9.1. Protocol Description
Hypertext Transfer Protocol (HTTP) is a request/response protocol. A The Hypertext Transfer Protocol (HTTP) is a request/response
client sends a request containing a request method, URI and protocol protocol. A client sends a request containing a request method, URI,
version followed message whose design is inspired by MIME (see and protocol version followed by message whose design is inspired by
[RFC7231] for the differences between an HTTP object and a MIME MIME (see [RFC7231] for the differences between an HTTP object and a
message), containing information about the client and request MIME message), containing information about the client and request
modifiers. The message can also contain a message body carrying modifiers. The message can also contain a message body carrying
application data. The server responds with a status or error code application data. The server responds with a status or error code
followed by a message containing information about the server and followed by a message containing information about the server and
information about the data. This may include a message body. It is information about the data. This may include a message body. It is
possible to specify a data format for the message body using MIME possible to specify a data format for the message body using MIME
media types [RFC2045]. The protocol has additional features, some media types [RFC2045]. The protocol has additional features; some
relevant to pseudo-transport are described below. relevant to pseudotransport are described below.
Content negotiation, specified in [RFC7231], is a mechanism provided Content negotiation, specified in [RFC7231], is a mechanism provided
by HTTP to allow selection of a representation for a requested by HTTP to allow selection of a representation for a requested
resource. The client and server negotiate acceptable data formats, resource. The client and server negotiate acceptable data formats,
character sets, data encoding (e.g., data can be transferred character sets, and data encoding (e.g., data can be transferred
compressed using gzip). HTTP can accommodate exchange of messages as compressed using gzip). HTTP can accommodate exchange of messages as
well as data streaming (using chunked transfer encoding [RFC7230]). well as data streaming (using chunked transfer encoding [RFC7230]).
It is also possible to request a part of a resource using an object It is also possible to request a part of a resource using an object
range request [RFC7233]. The protocol provides powerful cache range request [RFC7233]. The protocol provides powerful cache
control signaling defined in [RFC7234]. control signaling defined in [RFC7234].
The persistent connections of HTTP 1.1 and HTTP 2.0 allow multiple The persistent connections of HTTP 1.1 and HTTP 2.0 allow multiple
request- response transactions (streams) during the life-time of a request/response transactions (streams) during the lifetime of a
single HTTP connection. This reduces overhead during connection single HTTP connection. This reduces overhead during connection
establishment and mitigates transport layer slow-start that would establishment and mitigates transport-layer slow-start that would
have otherwise been incurred for each transaction. HTTP 2.0 have otherwise been incurred for each transaction. HTTP 2.0
connections can multiplex many request/response pairs in parallel on connections can multiplex many request/response pairs in parallel on
a single transport connection. Both are important to reduce latency a single transport connection. Both are important to reduce latency
for HTTP's primary use case. for HTTP's primary use case.
HTTP can be combined with security mechanisms, such as TLS (denoted HTTP can be combined with security mechanisms, such as TLS (denoted
by HTTPS). This adds protocol properties provided by such a by HTTPS). This adds protocol properties provided by such a
mechanism (e.g., authentication, encryption). The TLS Application- mechanism (e.g., authentication and encryption). The TLS
Layer Protocol Negotiation (ALPN) extension [RFC7301] can be used to Application-Layer Protocol Negotiation (ALPN) extension [RFC7301] can
negotiate the HTTP version within the TLS handshake, eliminating the be used to negotiate the HTTP version within the TLS handshake,
latency incurred by additional round-trip exchanges. Arbitrary eliminating the latency incurred by additional round-trip exchanges.
cookie strings, included as part of the request headers, are often Arbitrary cookie strings, included as part of the request headers,
used as bearer tokens in HTTP. are often used as bearer tokens in HTTP.
3.9.2. Interface Description 3.9.2. Interface Description
There are many HTTP libraries available exposing different APIs. The There are many HTTP libraries available exposing different APIs. The
APIs provide a way to specify a request by providing a URI, a method, APIs provide a way to specify a request by providing a URI, a method,
request modifiers and optionally a request body. For the response, request modifiers, and, optionally, a request body. For the
callbacks can be registered that will be invoked when the response is response, callbacks can be registered that will be invoked when the
received. If HTTPS is used, the API exposes a registration of response is received. If HTTPS is used, the API exposes a
callbacks for a server that requests client authentication and when registration of callbacks when a server requests client
certificate verification is needed. authentication and when certificate verification is needed.
The World Wide Web Consortium (W3C) has standardized the The World Wide Web Consortium (W3C) has standardized the
XMLHttpRequest API [XHR]. This API can be used for sending HTTP/ XMLHttpRequest API [XHR]. This API can be used for sending HTTP/
HTTPS requests and receiving server responses. Besides the XML data HTTPS requests and receiving server responses. Besides the XML data
format, the request and response data format can also be JSON, HTML, format, the request and response data format can also be JSON, HTML,
and plain text. JavaScript and XMLHttpRequest are ubiquitous and plain text. JavaScript and XMLHttpRequest are ubiquitous
programming models for websites, and more general applications, where programming models for websites and more general applications where
native code is less attractive. native code is less attractive.
3.9.3. Transport features 3.9.3. Transport Features
The transport features provided by HTTP, when used as a pseudo- The transport features provided by HTTP, when used as a
transport, are: pseudotransport, are:
o unicast transport (provided by the lower layer protocol, usually o unicast transport (provided by the lower-layer protocol, usually
TCP), TCP),
o uni- or bidirectional communication, o unidirectional or bidirectional communication,
o transfer of objects in multiple streams with object content type o transfer of objects in multiple streams with object content type
negotiation, supporting partial transmission of object ranges, negotiation, supporting partial transmission of object ranges,
o ordered delivery (provided by the lower layer protocol, usually o ordered delivery (provided by the lower-layer protocol, usually
TCP), TCP),
o fully reliable delivery (provided by the lower layer protocol, o fully reliable delivery (provided by the lower-layer protocol,
usually TCP), usually TCP),
o flow control (provided by the lower layer protocol, usually TCP). o flow control (provided by the lower-layer protocol, usually TCP),
and
o congestion control (provided by the lower layer protocol, usually o congestion control (provided by the lower-layer protocol, usually
TCP). TCP).
HTTPS (HTTP over TLS) additionally provides the following features HTTPS (HTTP over TLS) additionally provides the following features
(as provided by TLS): (as provided by TLS):
o authentication (of one or both ends of a connection), o authentication (of one or both ends of a connection),
o confidentiality, o confidentiality, and
o integrity protection. o integrity protection.
3.10. File Delivery over Unidirectional Transport/Asynchronous Layered 3.10. File Delivery over Unidirectional Transport / Asynchronous
Coding Reliable Multicast (FLUTE/ALC) Layered Coding (FLUTE/ALC) for Reliable Multicast
FLUTE/ALC is an IETF standards track protocol specified in [RFC6726] FLUTE/ALC is an IETF Standards Track protocol specified in [RFC6726]
and [RFC5775]. It provides object-oriented delivery of discrete data and [RFC5775]. It provides object-oriented delivery of discrete data
or files. Asynchronous Layer Coding (ALC) provides an underlying or files. Asynchronous Layer Coding (ALC) provides an underlying
reliable transport service and FLUTE a file-oriented specialization reliable transport service and FLUTE a file-oriented specialization
of the ALC service (e.g., to carry associated metadata). The of the ALC service (e.g., to carry associated metadata). [RFC6726]
[RFC6726] and [RFC5775] protocols are non-backward-compatible updates and [RFC5775] are non-backward-compatible updates of [RFC3926] and
of the [RFC3926] and [RFC3450] experimental protocols; these [RFC3450], which are Experimental protocols; these Experimental
experimental protocols are currently largely deployed in the 3GPP protocols are currently largely deployed in the 3GPP Multimedia
Multimedia Broadcast and Multicast Services (MBMS) (see [MBMS], Broadcast / Multicast Service (MBMS) (see [MBMS], Section 7) and
section 7) and similar contexts (e.g., the Japanese ISDB-Tmm similar contexts (e.g., the Japanese ISDB-Tmm standard).
standard).
The FLUTE/ALC protocol has been designed to support massively The FLUTE/ALC protocol has been designed to support massively
scalable reliable bulk data dissemination to receiver groups of scalable reliable bulk data dissemination to receiver groups of
arbitrary size using IP Multicast over any type of delivery network, arbitrary size using IP Multicast over any type of delivery network,
including unidirectional networks (e.g., broadcast wireless including unidirectional networks (e.g., broadcast wireless
channels). However, the FLUTE/ALC protocol also supports point-to- channels). However, the FLUTE/ALC protocol also supports point-to-
point unicast transmissions. point unicast transmissions.
FLUTE/ALC bulk data dissemination has been designed for discrete file FLUTE/ALC bulk data dissemination has been designed for discrete file
or memory-based "objects". Although FLUTE/ALC is not well adapted to or memory-based "objects". Although FLUTE/ALC is not well adapted to
byte- and message-streaming, there is an exception: FLUTE/ALC is used byte and message streaming, there is an exception: FLUTE/ALC is used
to carry 3GPP Dynamic Adaptive Streaming over HTTP (DASH) when to carry 3GPP Dynamic Adaptive Streaming over HTTP (DASH) when
scalability is a requirement (see [MBMS], section 5.6). scalability is a requirement (see [MBMS], Section 5.6).
FLUTE/ALC's reliability, delivery mode, congestion control, and flow/ FLUTE/ALC's reliability, delivery mode, congestion control, and flow/
rate control mechanisms can be separately controlled to meet rate control mechanisms can be separately controlled to meet
different application needs. Section 4.1 of different application needs. Section 4.1 of [RFC8085] describes
[I-D.ietf-tsvwg-rfc5405bis] describes multicast congestion control multicast congestion control requirements for UDP.
requirements for UDP.
3.10.1. Protocol Description 3.10.1. Protocol Description
The FLUTE/ALC protocol works on top of UDP (though it could work on The FLUTE/ALC protocol works on top of UDP (though it could work on
top of any datagram delivery transport protocol), without requiring top of any datagram delivery transport protocol), without requiring
any connectivity from receivers to the sender. Purely unidirectional any connectivity from receivers to the sender. Purely unidirectional
networks are therefore supported by FLUTE/ALC. This guarantees networks are therefore supported by FLUTE/ALC. This guarantees
scalability to an unlimited number of receivers in a session, since scalability to an unlimited number of receivers in a session, since
the sender behaves exactly the same regardless of the number of the sender behaves exactly the same regardless of the number of
receivers. receivers.
FLUTE/ALC supports the transfer of bulk objects such as file or in- FLUTE/ALC supports the transfer of bulk objects such as file or
memory content, using either a push or an on-demand mode. in push 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, mode, content is sent once to the receivers, while in on-demand mode,
content is sent continuously during periods of time that can greatly content is sent continuously during periods of time that can greatly
exceed the average time required to download the session objects (see exceed the average time required to download the session objects (see
[RFC5651], section 4.2). [RFC5651], Section 4.2).
This enables receivers to join a session asynchronously, at their own This enables receivers to join a session asynchronously, at their own
discretion, receive the content and leave the session. In this case, discretion, receive the content, and leave the session. In this
data content is typically sent continuously, in loops (also known as case, data content is typically sent continuously, in loops (also
"carousels"). FLUTE/ALC also supports the transfer of an object known as "carousels"). FLUTE/ALC also supports the transfer of an
stream, with loose real-time constraints. This is particularly object stream, with loose real-time constraints. This is
useful to carry 3GPP DASH when scalability is a requirement and particularly useful to carry 3GPP DASH when scalability is a
unicast transmissions over HTTP cannot be used ([MBMS], section 5.6). requirement and unicast transmissions over HTTP cannot be used
In this case, packets are sent in sequence using push mode. FLUTE/ ([MBMS], Section 5.6). In this case, packets are sent in sequence
ALC is not well adapted to byte- and message-streaming and other using push mode. FLUTE/ALC is not well adapted to byte and message
solutions could be preferred (e.g., FECFRAME [RFC6363] with real-time streaming, and other solutions could be preferred (e.g., FECFRAME
flows). [RFC6363] with real-time flows).
The FLUTE file delivery instantiation of ALC provides a metadata The FLUTE file delivery instantiation of ALC provides a metadata
delivery service. Each object of the FLUTE/ALC session is described delivery service. Each object of the FLUTE/ALC session is described
in a dedicated entry of a File Delivery Table (FDT), using an XML in a dedicated entry of a File Delivery Table (FDT), using an XML
format (see [RFC6726], section 3.2). This metadata can include, but format (see [RFC6726], Section 3.2). This metadata can include, but
is not restricted to, a URI attribute (to identify and locate the is not restricted to, a URI attribute (to identify and locate the
object), a media type attribute, a size attribute, an encoding object), a media type attribute, a size attribute, an encoding
attribute, or a message digest attribute. Since the set of objects attribute, or a message digest attribute. Since the set of objects
sent within a session can be dynamic, with new objects being added 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 and old ones removed, several instances of the FDT can be sent, and a
mechanism is provided to identify a new FDT Instance. mechanism is provided to identify a new FDT instance.
Error detection and verification of the protocol control information Error detection and verification of the protocol control information
relies on the on the underlying transport (e.g., UDP checksum). relies on the underlying transport (e.g., UDP checksum).
To provide robustness against packet loss and improve the efficiency To provide robustness against packet loss and improve the efficiency
of the on-demand mode, FLUTE/ALC relies on packet erasure coding (AL- of the on-demand mode, FLUTE/ALC relies on packet erasure coding
FEC). AL-FEC encoding is proactive (since there is no feedback and (Application-Layer Forward Error Correction (AL-FEC)). AL-FEC
therefore no (N)ACK-based retransmission) and ALC packets containing encoding is proactive (since there is no feedback and therefore no
repair data are sent along with ALC packets containing source data. (N)ACK-based retransmission), and ALC packets containing repair data
are sent along with ALC packets containing source data. Several FEC
Several FEC Schemes have been standardized; FLUTE/ALC does not Schemes have been standardized; FLUTE/ALC does not mandate the use of
mandate the use of any particular one. Several strategies concerning any particular one. Several strategies concerning the transmission
the transmission order of ALC source and repair packets are possible, order of ALC source and repair packets are possible, in particular,
in particular in on-demand mode where it can deeply impact the in on-demand mode where it can deeply impact the service provided
service provided (e.g., to favor the recovery of objects in sequence, (e.g., to favor the recovery of objects in sequence or, at the other
or at the other extreme, to favor the recovery of all objects in extreme, to favor the recovery of all objects in parallel), and
parallel), and FLUTE/ALC does not mandate nor recommend the use of FLUTE/ALC does not mandate nor recommend the use of any particular
any particular one. one.
A FLUTE/ALC session is composed of one or more channels, associated A FLUTE/ALC session is composed of one or more channels, associated
to different destination unicast and/or multicast IP addresses. ALC to different destination unicast and/or multicast IP addresses. ALC
packets are sent in those channels at a certain transmission rate, packets are sent in those channels at a certain transmission rate,
with a rate that often differs depending on the channel. FLUTE/ALC with a rate that often differs depending on the channel. FLUTE/ALC
does not mandate nor recommend any strategy to select which 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 packet to send on which channel. FLUTE/ALC can use a multiple rate
congestion control building block (e.g., WEBRC) to provide congestion congestion control building block (e.g., Wave and Equation Based Rate
control that is feedback free, where receivers adjust their reception Control (WEBRC)) to provide congestion control that is feedback free,
rates individually by joining and leaving channels associated with where receivers adjust their reception rates individually by joining
the session. To that purpose, the ALC header provides a specific and leaving channels associated with the session. To that purpose,
field to carry congestion control specific information. However the ALC header provides a specific field to carry congestion-control-
FLUTE/ALC does not mandate the use of a particular congestion control specific information. However, FLUTE/ALC does not mandate the use of
mechanism although WEBRC is mandatory to support for the Internet a particular congestion control mechanism although WEBRC is mandatory
([RFC6726], section 1.1.4). FLUTE/ALC is often used over a network to support for the Internet ([RFC6726], Section 1.1.4). FLUTE/ALC is
path with pre-provisioned capacity [I-D.ietf-tsvwg-rfc5405bis] where often used over a network path with pre-provisioned capacity
there are no flows competing for capacity. In this case, a sender- [RFC8085] where there are no flows competing for capacity. In this
based rate control mechanism and a single channel is sufficient. case, a sender-based rate control mechanism and a single channel are
sufficient.
[RFC6584] provides per-packet authentication, integrity, and anti- [RFC6584] provides per-packet authentication, integrity, and anti-
replay protection in the context of the ALC and NORM protocols. replay protection in the context of the ALC and NORM protocols.
Several mechanisms are proposed that seamlessly integrate into these Several mechanisms are proposed that seamlessly integrate into these
protocols using the ALC and NORM header extension mechanisms. protocols using the ALC and NORM header extension mechanisms.
3.10.2. Interface Description 3.10.2. Interface Description
The FLUTE/ALC specification does not describe a specific application The FLUTE/ALC specification does not describe a specific API to
programming interface (API) to control protocol operation. Although control protocol operation. Although open source and commercial
open source and commercial implementations have specified APIs, there implementations have specified APIs, there is no IETF-specified API
is no IETF- specified API for FLUTE/ALC. for FLUTE/ALC.
3.10.3. Transport Features 3.10.3. Transport Features
The transport features provided by FLUTE/ALC are: The transport features provided by FLUTE/ALC are:
o unicast, multicast, anycast or IPv4 broadcast transmission, o unicast, multicast, anycast, or IPv4 broadcast transmission,
o object-oriented delivery of discrete data or files and associated o object-oriented delivery of discrete data or files and associated
metadata, metadata,
o fully reliable or partially reliable delivery (of file or in- o fully reliable or partially reliable delivery (of file or in-
memory objects), using proactive packet erasure coding (AL-FEC) to memory objects), using proactive packet erasure coding (AL-FEC) to
recover from packet erasures, recover from packet erasures,
o ordered or unordered delivery (of file or in-memory objects), o ordered or unordered delivery (of file or in-memory objects),
o error detection (based on the UDP checksum), o error detection (based on the UDP checksum),
o per-packet authentication, o per-packet authentication,
o per-packet integrity, o per-packet integrity,
o per-packet replay protection, o per-packet replay protection, and
o congestion control for layered flows (e.g., with WEBRC). o congestion control for layered flows (e.g., with WEBRC).
3.11. NACK-Oriented Reliable Multicast (NORM) 3.11. NACK-Oriented Reliable Multicast (NORM)
NORM is an IETF standards track protocol specified in [RFC5740]. It NORM is an IETF Standards Track protocol specified in [RFC5740]. It
provides object-oriented delivery of discrete data or files. provides object-oriented delivery of discrete data or files.
The protocol was designed to support reliable bulk data dissemination The protocol was designed to support reliable bulk data dissemination
to receiver groups using IP Multicast but also provides for point-to- to receiver groups using IP Multicast but also provides for point-to-
point unicast operation. Support for bulk data dissemination point unicast operation. Support for bulk data dissemination
includes discrete file or computer memory-based "objects" as well as includes discrete file or computer memory-based "objects" as well as
byte- and message-streaming. byte and message streaming.
NORM can incorporate packet erasure coding as a part of its selective NORM can incorporate packet erasure coding as a part of its selective
ARQ in response to negative acknowledgments from the receiver. The Automatic Repeat reQuest (ARQ) in response to negative
packet erasure coding can also be proactively applied for forward acknowledgments from the receiver. The packet erasure coding can
protection from packet loss. NORM transmissions are governed by TCP- also be proactively applied for forward protection from packet loss.
friendly multicast congestion control (TFMCC, [RFC4654]). The NORM transmissions are governed by TCP-Friendly Multicast Congestion
reliability, congestion control and flow control mechanisms can be Control (TFMCC) [RFC4654]. The reliability, congestion control, and
separately controlled to meet different application needs. flow control mechanisms can be separately controlled to meet
different application needs.
3.11.1. Protocol Description 3.11.1. Protocol Description
The NORM protocol is encapsulated in UDP datagrams and thus provides The NORM protocol is encapsulated in UDP datagrams and thus provides
multiplexing for multiple sockets on hosts using port numbers. For multiplexing for multiple sockets on hosts using port numbers. For
loosely coordinated IP Multicast, NORM is not strictly connection- loosely coordinated IP Multicast, NORM is not strictly connection-
oriented although per-sender state is maintained by receivers for oriented although per-sender state is maintained by receivers for
protocol operation. [RFC5740] does not specify a handshake protocol protocol operation. [RFC5740] does not specify a handshake protocol
for connection establishment. Separate session initiation can be for connection establishment. Separate session initiation can be
used to coordinate port numbers. However, in-band "client-server" used to coordinate port numbers. However, in-band "client-server"
style connection establishment can be accomplished with the NORM style connection establishment can be accomplished with the NORM
congestion control signaling messages using port binding techniques congestion control signaling messages using port binding techniques
like those for TCP client-server connections. like those for TCP client-server connections.
NORM supports bulk "objects" such as file or in-memory content but 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 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 of packet erasure coding. In the case of stream transport, NORM can
support either byte streams or message streams where application- support either byte streams or message streams where application-
defined message boundary information is carried in the NORM protocol defined message boundary information is carried in the NORM protocol
messages. This allows the receiver(s) to join/re-join and recover messages. This allows the receiver(s) to join/rejoin and recover
message boundaries mid-stream as needed. Application content is message boundaries mid-stream as needed. Application content is
carried and identified by the NORM protocol with encoding symbol carried and identified by the NORM protocol with encoding symbol
identifiers depending upon the Forward Error Correction (FEC) Scheme identifiers depending upon the Forward Error Correction (FEC) Scheme
[RFC3452] configured. NORM uses NACK-based selective ARQ to reliably [RFC5052] configured. NORM uses NACK-based selective ARQ to reliably
deliver the application content to the receiver(s). NORM proactively deliver the application content to the receiver(s). NORM proactively
measures round-trip timing information to scale ARQ timers measures round-trip timing information to scale ARQ timers
appropriately and to support congestion control. For multicast appropriately and to support congestion control. For multicast
operation, timer-based feedback suppression is uses to achieve group operation, timer-based feedback suppression is used to achieve group
size scaling with low feedback traffic levels. The feedback size scaling with low feedback traffic levels. The feedback
suppression is not applied for unicast operation. suppression is not applied for unicast operation.
NORM uses rate-based congestion control based upon the TCP-Friendly NORM uses rate-based congestion control based upon the TCP-Friendly
Rate Control (TFRC) [RFC4324] principles that are also used in DCCP Rate Control (TFRC) [RFC5348] principles that are also used in DCCP
[RFC4340]. NORM uses control messages to measure RTT and collect [RFC4340]. NORM uses control messages to measure RTT and collect
congestion event information (e.g., reflecting a loss event or ECN congestion event information (e.g., reflecting a loss event or ECN
event) from the receiver(s) to support dynamic adjustment or the event) from the receiver(s) to support dynamic adjustment or the
rate. The TCP-Friendly Multicast Congestion Control (TFMCC) rate. TCP-Friendly Multicast Congestion Control (TFMCC) [RFC4654]
[RFC4654] provides extra features to support multicast, but is provides extra features to support multicast but is functionally
functionally equivalent to TFRC for unicast. equivalent to TFRC for unicast.
Error detection and verification of the protocol control information Error detection and verification of the protocol control information
relies on the on the underlying transport(e.g., UDP checksum). relies on the on the underlying transport (e.g., UDP checksum).
The reliability mechanism is decoupled from congestion control. This The reliability mechanism is decoupled from congestion control. This
allows invocation of alternative arrangements of transport services. allows invocation of alternative arrangements of transport services,
For example, to support, fixed-rate reliable delivery or unreliable for example, to support, fixed-rate reliable delivery or unreliable
delivery (that may optionally be "better than best effort" via packet delivery (that may optionally be "better than best effort" via packet
erasure coding) using TFRC. Alternative congestion control erasure coding) using TFRC. Alternative congestion control
techniques may be applied. For example, TFRC rate control with techniques may be applied, for example, TFRC with congestion event
congestion event detection based on ECN. detection based on ECN.
While NORM provides NACK-based reliability, it also supports a While NORM provides NACK-based reliability, it also supports a
positive acknowledgment (ACK) mechanism that can be used for receiver positive acknowledgment (ACK) mechanism that can be used for receiver
flow control. This mechanism is decoupled from the reliability and flow control. This mechanism is decoupled from the reliability and
congestion control, supporting applications with different needs. congestion control, supporting applications with different needs.
One example is use of NORM for quasi-reliable delivery, where timely One example is use of NORM for quasi-reliable delivery, where timely
delivery of newer content may be favored over completely reliable delivery of newer content may be favored over completely reliable
delivery of older content within buffering and RTT constraints. delivery of older content within buffering and RTT constraints.
3.11.2. Interface Description 3.11.2. Interface Description
The NORM specification does not describe a specific application The NORM specification does not describe a specific API to control
programming interface (API) to control protocol operation. A freely- protocol operation. A freely available, open-source reference
available, open source reference implementation of NORM is available implementation of NORM is available at
at https://www.nrl.navy.mil/itd/ncs/products/norm, and a documented <https://www.nrl.navy.mil/itd/ncs/products/norm>, and a documented
API is provided for this implementation. While a sockets-like API is API is provided for this implementation. While a sockets-like API is
not currently documented, the existing API supports the necessary not currently documented, the existing API supports the necessary
functions for that to be implemented. functions for that to be implemented.
3.11.3. Transport Features 3.11.3. Transport Features
The transport features provided by NORM are: The transport features provided by NORM are:
o unicast or multicast transport, o unicast or multicast transport,
skipping to change at page 35, line 37 skipping to change at page 36, line 27
coding both proactively and as part of ARQ) delivery, coding both proactively and as part of ARQ) delivery,
o unordered delivery, o unordered delivery,
o error detection (relies on UDP checksum), o error detection (relies on UDP checksum),
o segmentation, o segmentation,
o data bundling (using Nagle's algorithm), o data bundling (using Nagle's algorithm),
o flow control (timer-based and/or ack-based), o flow control (timer-based and/or ACK-based), and
o congestion control (also supporting fixed rate reliable or o congestion control (also supporting fixed-rate reliable or
unreliable delivery). unreliable delivery).
3.12. Internet Control Message Protocol (ICMP) 3.12. Internet Control Message Protocol (ICMP)
The Internet Control Message Protocol (ICMP) [RFC0792] for IPv4 and The Internet Control Message Protocol (ICMP) [RFC792] for IPv4 and
ICMP for IPv6 [RFC4443] are IETF standards track protocols. It is a ICMP for IPv6 [RFC4443] are IETF Standards Track protocols. It is a
connection-less unidirectional protocol that delivers individual connectionless unidirectional protocol that delivers individual
messages, without error correction, congestion control, or flow messages, without error correction, congestion control, or flow
control. Messages may be sent as unicast, IPv4 broadcast or control. Messages may be sent as unicast, IPv4 broadcast, or
multicast datagrams (IPv4 and IPv6), in addition to anycast multicast datagrams (IPv4 and IPv6), in addition to anycast
datagrams. datagrams.
While ICMP is not typically described as a transport protocol, it While ICMP is not typically described as a transport protocol, it
does position itself over the network layer, and the operation of does position itself over the network layer, and the operation of
other transport protocols can be closely linked to the functions other transport protocols can be closely linked to the functions
provided by ICMP. provided by ICMP.
Transport Protocols and upper layer protocols can use received ICMP Transport protocols and upper-layer protocols can use received ICMP
messages to help them take appropriate decisions when network or messages to help them make appropriate decisions when network or
endpoint errors are reported. For example, to implement, ICMP-based endpoint errors are reported, for example, to implement ICMP-based
Path MTU discovery [RFC1191][RFC1981] or assist in Packetization Path MTU Discovery (PMTUD) [RFC1191] [RFC1981] or assist in
Layer Path MTU Discovery (PMTUD) [RFC4821]. Such reactions to Packetization Layer PMTUD (PLPMTUD) [RFC4821]. Such reactions to
received messages need to protect from off-path data injection received messages need to protect from off-path data injection
[I-D.ietf-tsvwg-rfc5405bis], to avoid an application receiving
packets created by an unauthorized third party. An application [RFC8085] to avoid an application receiving packets created by an
therefore needs to ensure that all messages are appropriately unauthorized third party. An application therefore needs to ensure
validated, by checking the payload of the messages to ensure these that all messages are appropriately validated by checking the payload
are received in response to actually transmitted traffic (e.g., a of the messages to ensure they are received in response to actually
reported error condition that corresponds to a UDP datagram or TCP transmitted traffic (e.g., a reported error condition that
segment was actually sent by the application). This requires context corresponds to a UDP datagram or TCP segment was actually sent by the
[RFC6056], such as local state about communication instances to each application). This requires context [RFC6056], such as local state
destination (e.g., in the TCP, DCCP, or SCTP protocols). This state about communication instances to each destination (e.g., in TCP,
is not always maintained by UDP-based applications DCCP, or SCTP). This state is not always maintained by UDP-based
[I-D.ietf-tsvwg-rfc5405bis]. applications [RFC8085].
3.12.1. Protocol Description 3.12.1. Protocol Description
ICMP is a connection-less unidirectional protocol, It delivers ICMP is a connectionless unidirectional protocol. It delivers
independent messages, called datagrams. Each message is required to independent messages, called "datagrams". Each message is required
carry a checksum as an integrity check and to protect from mis- to carry a checksum as an integrity check and to protect from
delivery to an unintended endpoint. misdelivery to an unintended endpoint.
ICMP messages typically relay diagnostic information from an endpoint ICMP messages typically relay diagnostic information from an endpoint
[RFC1122] or network device [RFC1812] addressed to the sender of a [RFC1122] or network device [RFC1812] addressed to the sender of a
flow. This usually contains the network protocol header of a packet flow. This usually contains the network protocol header of a packet
that encountered a reported issue. Some formats of messages can also that encountered a reported issue. Some formats of messages can also
carry other payload data. Each message carries an integrity check carry other payload data. Each message carries an integrity check
calculated in the same way as for UDP, this checksum is not optional. calculated in the same way as for UDP; this checksum is not optional.
The RFC series defines additional IPv6 message formats to support a The RFC Series defines additional IPv6 message formats to support a
range of uses. In the case of IPv6 the protocol incorporates range of uses. In the case of IPv6, the protocol incorporates
neighbor discovery [RFC2461] [RFC3971] (provided by ARP for IPv4) and neighbor discovery [RFC4861] [RFC3971] (provided by ARP for IPv4) and
the Multicast Listener Discovery (MLD) [RFC2710] group management Multicast Listener Discovery (MLD) [RFC2710] group management
functions (provided by IGMP for IPv4). functions (provided by IGMP for IPv4).
Reliable transmission can not be assumed. A receiving application Reliable transmission cannot be assumed. A receiving application
that is unable to run sufficiently fast, or frequently, may miss that is unable to run sufficiently fast, or frequently, may miss
messages since there is no flow or congestion control. In addition messages since there is no flow or congestion control. In addition,
some network devices rate-limit ICMP messages. some network devices rate-limit ICMP messages.
3.12.2. Interface Description 3.12.2. Interface Description
ICMP processing is integrated in many connection-oriented transports, ICMP processing is integrated in many connection-oriented transports
but like other functions needs to be provided by an upper-layer but, like other functions, needs to be provided by an upper-layer
protocol when using UDP and UDP-Lite. protocol when using UDP and UDP-Lite.
On some stacks, a bound socket also allows a UDP application to be On some stacks, a bound socket also allows a UDP application to be
notified when ICMP error messages are received for its transmissions notified when ICMP error messages are received for its transmissions
[I-D.ietf-tsvwg-rfc5405bis]. [RFC8085].
Any response to ICMP error messages ought to be robust to temporary Any response to ICMP error messages ought to be robust to temporary
routing failures (sometimes called "soft errors"), e.g., transient routing failures (sometimes called "soft errors"), e.g., transient
ICMP "unreachable" messages ought to not normally cause a ICMP "unreachable" messages ought to not normally cause a
communication abort [RFC5461] [I-D.ietf-tsvwg-rfc5405bis]. communication abort [RFC5461] [RFC8085].
3.12.3. Transport Features 3.12.3. Transport Features
ICMP does not provide any transport service directly to applications. ICMP does not provide any transport service directly to applications.
Used together with other transport protocols, it provides Used together with other transport protocols, it provides
transmission of control, error, and measurement data between transmission of control, error, and measurement data between
endpoints, or from devices along the path to one endpoint. endpoints or from devices along the path to one endpoint.
4. Congestion Control 4. Congestion Control
Congestion control is critical to the stable operation of the Congestion control is critical to the stable operation of the
Internet. A variety of mechanisms are used to provide the congestion Internet. A variety of mechanisms are used to provide the congestion
control needed by many Internet transport protocols. Congestion is control needed by many Internet transport protocols. Congestion is
detected based on sensing of network conditions, whether through detected based on sensing of network conditions, whether through
explicit or implicit feedback. The congestion control mechanisms explicit or implicit feedback. The congestion control mechanisms
that can be applied by different transport protocols are largely that can be applied by different transport protocols are largely
orthogonal to the choice of transport protocol. This section orthogonal to the choice of transport protocol. This section
provides an overview of the congestion control mechanisms available provides an overview of the congestion control mechanisms available
to the protocols described in Section 3. to the protocols described in Section 3.
Many protocols use a separate window to determine the maximum sending Many protocols use a separate window to determine the maximum sending
rate that is allowed by the congestion control. The used congestion rate that is allowed by the congestion control. The used congestion
control mechanism will increase the congestion window if feedback is control mechanism will increase the congestion window if feedback is
received that indicates that the currently used network path is not received that indicates that the currently used network path is not
congested, and will reduce the window otherwise. Window-based congested and will reduce the window otherwise. Window-based
mechanisms often increase their window slowing over multiple RTTs, mechanisms often increase their window slowing over multiple RTTs,
while decreasing strongly when the first indication of congestion is while decreasing strongly when the first indication of congestion is
received. One example is an Additive Increase Multiplicative received. One example is an Additive Increase Multiplicative
Decrease (AIMD) scheme, where the window is increased by a certain Decrease (AIMD) scheme, where the window is increased by a certain
number of packets/bytes for each data segment that has been number of packets/bytes for each data segment that has been
successfully transmitted, while the window decreases multiplicatively successfully transmitted, while the window decreases multiplicatively
on the occurrence of a congestion event. This can lead to a rather on the occurrence of a congestion event. This can lead to a rather
unstable, oscillating sending rate, but will resolve a congestion unstable, oscillating sending rate but will resolve a congestion
situation quickly. TCP New Reno [RFC5681] which is one of the situation quickly. Examples of window-based AIMD schemes include TCP
initial proposed schemes for TCP as well as TCP Cubic NewReno [RFC5681], TCP Cubic [CUBIC] (the default mechanism for TCP
[I-D.ietf-tcpm-cubic] which is the default mechanism for TCP in Linux in Linux), and CCID 2 specified for DCCP [RFC4341].
are two examples for window-based AIMD schemes. This approach is
also used by DCCP CCID-2 for datagram congestion control.
Some classes of applications prefer to use a transport service that Some classes of applications prefer to use a transport service that
allows sending at a more stable rate, that is slowly varied in allows sending at a more stable rate that is slowly varied in
response to congestion. Rate-based methods offer this type of response to congestion. Rate-based methods offer this type of
congestion control and have been defined based on the loss ratio and congestion control and have been defined based on the loss ratio and
observed round trip time, such as TFRC [RFC5348] and TFRC-SP observed round-trip time, such as TFRC [RFC5348] and TFRC-SP
[RFC4828]. These methods utilize a throughput equation to determine [RFC4828]. These methods utilize a throughput equation to determine
the maximum acceptable rate. Such methods are used with DCCP CCID-3 the maximum acceptable rate. Such methods are used with DCCP CCID 3
[RFC4342] and CCID-4 [RFC5622], WEBRC [RFC3738], and other [RFC4342], CCID 4 [RFC5622], WEBRC [RFC3738], and other applications.
applications.
Another class of applications prefer a transport service that yields Another class of applications prefers a transport service that yields
to other (higher-priority) traffic, such as interactive to other (higher-priority) traffic, such as interactive
transmissions. While most traffic in the Internet uses loss-based transmissions. While most traffic in the Internet uses loss-based
congestion control and therefore tends to fill the network buffers congestion control and therefore tends to fill the network buffers
(to a certain level if Active Queue Management (AQM) is used), low- (to a certain level if Active Queue Management (AQM) is used), low-
priority congestion control methods often react to changes in delay priority congestion control methods often react to changes in delay
as an earlier indication of congestion. This approach tends to as an earlier indication of congestion. This approach tends to
induce less loss than a loss-based method but does generally not induce less loss than a loss-based method but does generally not
compete well with loss-based traffic across shared bottleneck links. compete well with loss-based traffic across shared bottleneck links.
Therefore, methods such as LEDBAT [RFC6824], are deployed in the Therefore, methods such as LEDBAT [RFC6817] are deployed in the
Internet for scavenger traffic that aim to only utilize otherwise Internet for scavenger traffic that aims to only utilize otherwise
unused capacity. unused capacity.
5. Transport Features 5. Transport Features
The transport protocol features described in this document can be The transport protocol features described in this document can be
used as a basis for defining common transport features, listed below used as a basis for defining common transport features. These are
with the protocols supporting them: listed below with the protocols supporting them:
o Control Functions o Control Functions
* Addressing * Addressing
+ unicast (TCP, MPTCP, UDP, UDP-Lite, SCTP, DCCP, TLS, RTP, + unicast (TCP, MPTCP, UDP, UDP-Lite, SCTP, DCCP, TLS, RTP,
HTTP, ICMP) HTTP, ICMP)
+ multicast (UDP, UDP-Lite, RTP, ICMP, FLUTE/ALC, NORM). Note + multicast (UDP, UDP-Lite, RTP, ICMP, FLUTE/ALC, NORM). Note
that, as TLS and DTLS are unicast-only, there is no widely that, as TLS and DTLS are unicast-only, there is no widely
deployed mechanism for supporting the features in the deployed mechanism for supporting the features listed under
Security section below when using multicast addressing. the Security bullet (below) when using multicast addressing.
+ IPv4 broadcast (UDP, UDP-Lite, ICMP) + IPv4 broadcast (UDP, UDP-Lite, ICMP)
+ anycast (UDP, UDP-Lite). Connection-oriented protocols such + anycast (UDP, UDP-Lite). Connection-oriented protocols such
as TCP and DCCP have also been deployed using anycast as TCP and DCCP have also been deployed using anycast
addressing, with the risk that routing changes may cause addressing, with the risk that routing changes may cause
connection failure. connection failure.
* Association type * Association type
+ connection-oriented (TCP, MPTCP, DCCP, SCTP, TLS, RTP, HTTP, + connection-oriented (TCP, MPTCP, DCCP, SCTP, TLS, RTP, HTTP,
NORM) NORM)
skipping to change at page 39, line 20 skipping to change at page 40, line 9
+ connection-oriented (TCP, MPTCP, DCCP, SCTP, TLS, RTP, HTTP, + connection-oriented (TCP, MPTCP, DCCP, SCTP, TLS, RTP, HTTP,
NORM) NORM)
+ connectionless (UDP, UDP-Lite, FLUTE/ALC) + connectionless (UDP, UDP-Lite, FLUTE/ALC)
* Multihoming support * Multihoming support
+ resilience and mobility (MPTCP, SCTP) + resilience and mobility (MPTCP, SCTP)
+ load-balancing (MPTCP) + load balancing (MPTCP)
+ address family multiplexing (MPTCP, SCTP) + address family multiplexing (MPTCP, SCTP)
* Middlebox cooperation * Middlebox cooperation
+ application-class signaling to middleboxes (DCCP) + application-class signaling to middleboxes (DCCP)
+ error condition signaling from middleboxes and routers to + error condition signaling from middleboxes and routers to
endpoints (ICMP) endpoints (ICMP)
skipping to change at page 40, line 4 skipping to change at page 40, line 40
+ fully reliable delivery (TCP, MPTCP, SCTP, TLS, HTTP, FLUTE/ + fully reliable delivery (TCP, MPTCP, SCTP, TLS, HTTP, FLUTE/
ALC, NORM) ALC, NORM)
+ partially reliable delivery (SCTP, NORM) + partially reliable delivery (SCTP, NORM)
- using packet erasure coding (RTP, FLUTE/ALC, NORM) - using packet erasure coding (RTP, FLUTE/ALC, NORM)
- with specified policy for dropped messages (SCTP) - with specified policy for dropped messages (SCTP)
+ unreliable delivery (SCTP, UDP, UDP-Lite, DCCP, RTP) + unreliable delivery (SCTP, UDP, UDP-Lite, DCCP, RTP)
- with drop notification to sender (SCTP, DCCP, RTP) - with drop notification to sender (SCTP, DCCP, RTP)
+ error detection + error detection
- checksum for error detection (TCP, MPTCP, UDP, UDP-Lite, - checksum for error detection (TCP, MPTCP, UDP, UDP-Lite,
SCTP, DCCP, TLS, DTLS, FLUTE/ALC, NORM, ICMP) SCTP, DCCP, TLS, DTLS, FLUTE/ALC, NORM, ICMP)
- partial payload checksum protection (UDP-Lite, DCCP). - partial payload checksum protection (UDP-Lite, DCCP).
Some uses of RTP can exploit partial payload checksum Some uses of RTP can exploit partial payload checksum
protection feature to provide a corruption tolerant protection feature to provide a corruption-tolerant
transport service. transport service.
- checksum optional (UDP). Possible with IPv4 and in - checksum optional (UDP). Possible with IPv4 and, in
certain cases with IPv6. certain cases, with IPv6.
* Ordering * Ordering
+ ordered delivery (TCP, MPTCP, SCTP, TLS, RTP, HTTP, FLUTE) + ordered delivery (TCP, MPTCP, SCTP, TLS, RTP, HTTP, FLUTE)
+ unordered delivery permitted (UDP, UDP-Lite, SCTP, DCCP, + unordered delivery permitted (UDP, UDP-Lite, SCTP, DCCP,
RTP, NORM) RTP, NORM)
* Type/framing * Type/framing
skipping to change at page 41, line 4 skipping to change at page 41, line 38
* Directionality * Directionality
+ unidirectional (UDP, UDP-Lite, DCCP, RTP, FLUTE/ALC, NORM) + unidirectional (UDP, UDP-Lite, DCCP, RTP, FLUTE/ALC, NORM)
+ bidirectional (TCP, MPTCP, SCTP, TLS, HTTP) + bidirectional (TCP, MPTCP, SCTP, TLS, HTTP)
o Transmission control o Transmission control
* flow control (TCP, MPTCP, SCTP, DCCP, TLS, RTP, HTTP) * flow control (TCP, MPTCP, SCTP, DCCP, TLS, RTP, HTTP)
* congestion control (TCP, MPTCP, SCTP, DCCP, RTP, FLUTE/ALC, * congestion control (TCP, MPTCP, SCTP, DCCP, RTP, FLUTE/ALC,
NORM). Congestion control can also provided by the transport NORM). Congestion control can also provided by the transport
supporting an upper later transport (e.g., TLS, RTP, HTTP). supporting an upper-layer transport (e.g., TLS, RTP, HTTP).
* segmentation (TCP, MPTCP, SCTP, TLS, RTP, HTTP, FLUTE/ALC, * segmentation (TCP, MPTCP, SCTP, TLS, RTP, HTTP, FLUTE/ALC,
NORM) NORM)
* data/message bundling (TCP, MPTCP, SCTP, TLS, HTTP) * data/message bundling (TCP, MPTCP, SCTP, TLS, HTTP)
* stream scheduling prioritization (SCTP, HTTP2) * stream scheduling prioritization (SCTP, HTTP2)
* endpoint multiplexing (MPTCP) * endpoint multiplexing (MPTCP)
skipping to change at page 41, line 32 skipping to change at page 42, line 20
* authentication of both ends of a connection (TLS, DTLS) * authentication of both ends of a connection (TLS, DTLS)
* confidentiality (TLS, DTLS) * confidentiality (TLS, DTLS)
* cryptographic integrity protection (TLS, DTLS) * cryptographic integrity protection (TLS, DTLS)
* replay protection (TLS, DTLS, FLUTE/ALC) * replay protection (TLS, DTLS, FLUTE/ALC)
6. IANA Considerations 6. IANA Considerations
This document has no considerations for IANA. This document does not require any IANA actions.
7. Security Considerations 7. Security Considerations
This document surveys existing transport protocols and protocols This document surveys existing transport protocols and protocols
providing transport-like services. Confidentiality, integrity, and providing transport-like services. Confidentiality, integrity, and
authenticity are among the features provided by those services. This authenticity are among the features provided by those services. This
document does not specify any new features or mechanisms for document does not specify any new features or mechanisms for
providing these features. Each RFC referenced by this document providing these features. Each RFC referenced by this document
discusses the security considerations of the specification it discusses the security considerations of the specification it
contains. contains.
8. Contributors 8. Informative References
In addition to the editors, this document is the work of Brian
Adamson, Dragana Damjanovic, Kevin Fall, Simone Ferlin-Oliviera,
Ralph Holz, Olivier Mehani, Karen Nielsen, Colin Perkins, Vincent
Roca, and Michael Tuexen.
o Section 3.2 on MPTCP was contributed by Simone Ferlin-Oliviera
(ferlin@simula.no) and Olivier Mehani
(olivier.mehani@nicta.com.au)
o Section 3.3 on UDP was contributed by Kevin Fall (kfall@kfall.com)
o Section 3.5 on SCTP was contributed by Michael Tuexen (tuexen@fh-
muenster.de) and Karen Nielsen (karen.nielsen@tieto.com)
o Section 3.7 on TLS and DTLS was contributed by Ralph Holz
(ralph.holz@nicta.com.au) and Olivier Mehani
(olivier.mehani@nicta.com.au)
o Section 3.8 on RTP contains contributions from Colin Perkins
(csp@csperkins.org)
o Section 3.9 on HTTP was contributed by Dragana Damjanovic [ClarkArch]
(ddamjanovic@mozilla.com) Clark, D. and D. Tennenhouse, "Architectural
Considerations for a New Generation of Protocols",
Proceedings of ACM SIGCOMM, DOI 10.1145/99517.99553, 1990.
o Section 3.10 on FLUTE/ALC was contributed by Vincent Roca [CUBIC] Rhee, I., Xu, L., Ha, S., Zimmermann, A., Eggert, L., and
(vincent.roca@inria.fr) R. Scheffenegger, "CUBIC for Fast Long-Distance Networks",
Work in Progress, draft-ietf-tcpm-cubic-04, February 2017.
o Section 3.11 on NORM was contributed by Brian Adamson [MBMS] 3GPP, "Multimedia Broadcast/Multicast Service (MBMS);
(brian.adamson@nrl.navy.mil) Protocols and codecs", 3GPP TS 26.346, 2015,
<http://www.3gpp.org/DynaReport/26346.htm>.
9. Acknowledgments [NAT-SUPP] Stewart, R., Tuexen, M., and I. Ruengeler, "Stream Control
Transmission Protocol (SCTP) Network Address Translation
Support", Work in Progress, draft-ietf-tsvwg-natsupp-09,
May 2016.
Thanks to Joe Touch, Michael Welzl, Spencer Dawkins, and the TAPS [POSIX] IEEE, "Standard for Information Technology -- Portable
Working Group for the comments, feedback, and discussion. This work Operating System Interface (POSIX(R)) Base Specifications,
is supported by the European Commission under grant agreement No. Issue 7", IEEE 1003.1, DOI 10.1109/ieeestd.2016.7582338,
318627 mPlane and from the Horizon 2020 research and innovation <http://ieeexplore.ieee.org/document/7582338/>.
program under grant agreements No. 644334 (NEAT) and No. 688421
(MAMI). This support does not imply endorsement.
10. Informative References [REST] Fielding, R., "Architectural Styles and the Design of
Network-based Software Architectures, Chapter 5:
Representational State Transfer", Ph.D.
Dissertation, University of California, Irvine, 2000.
[RFC0768] Postel, J., "User Datagram Protocol", STD 6, RFC 768, [RFC768] Postel, J., "User Datagram Protocol", STD 6, RFC 768,
DOI 10.17487/RFC0768, August 1980, DOI 10.17487/RFC0768, August 1980,
<http://www.rfc-editor.org/info/rfc768>. <http://www.rfc-editor.org/info/rfc768>.
[RFC0792] Postel, J., "Internet Control Message Protocol", STD 5, [RFC792] Postel, J., "Internet Control Message Protocol", STD 5,
RFC 792, DOI 10.17487/RFC0792, September 1981, RFC 792, DOI 10.17487/RFC0792, September 1981,
<http://www.rfc-editor.org/info/rfc792>. <http://www.rfc-editor.org/info/rfc792>.
[RFC0793] Postel, J., "Transmission Control Protocol", STD 7, [RFC793] Postel, J., "Transmission Control Protocol", STD 7,
RFC 793, DOI 10.17487/RFC0793, September 1981, RFC 793, DOI 10.17487/RFC0793, September 1981,
<http://www.rfc-editor.org/info/rfc793>. <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>.
[RFC1071] Braden, R., Borman, D., and C. Partridge, "Computing the [RFC1071] Braden, R., Borman, D., and C. Partridge, "Computing the
Internet checksum", RFC 1071, DOI 10.17487/RFC1071, Internet checksum", RFC 1071, DOI 10.17487/RFC1071,
September 1988, <http://www.rfc-editor.org/info/rfc1071>. September 1988, <http://www.rfc-editor.org/info/rfc1071>.
[RFC1122] Braden, R., Ed., "Requirements for Internet Hosts - [RFC1122] Braden, R., Ed., "Requirements for Internet Hosts -
Communication Layers", STD 3, RFC 1122, Communication Layers", STD 3, RFC 1122,
DOI 10.17487/RFC1122, October 1989, DOI 10.17487/RFC1122, October 1989,
<http://www.rfc-editor.org/info/rfc1122>. <http://www.rfc-editor.org/info/rfc1122>.
[RFC1191] Mogul, J. and S. Deering, "Path MTU discovery", RFC 1191, [RFC1191] Mogul, J. and S. Deering, "Path MTU discovery", RFC 1191,
skipping to change at page 43, line 44 skipping to change at page 44, line 14
[RFC2045] Freed, N. and N. Borenstein, "Multipurpose Internet Mail [RFC2045] Freed, N. and N. Borenstein, "Multipurpose Internet Mail
Extensions (MIME) Part One: Format of Internet Message Extensions (MIME) Part One: Format of Internet Message
Bodies", RFC 2045, DOI 10.17487/RFC2045, November 1996, Bodies", RFC 2045, DOI 10.17487/RFC2045, November 1996,
<http://www.rfc-editor.org/info/rfc2045>. <http://www.rfc-editor.org/info/rfc2045>.
[RFC2460] Deering, S. and R. Hinden, "Internet Protocol, Version 6 [RFC2460] Deering, S. and R. Hinden, "Internet Protocol, Version 6
(IPv6) Specification", RFC 2460, DOI 10.17487/RFC2460, (IPv6) Specification", RFC 2460, DOI 10.17487/RFC2460,
December 1998, <http://www.rfc-editor.org/info/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 [RFC2710] Deering, S., Fenner, W., and B. Haberman, "Multicast
Listener Discovery (MLD) for IPv6", RFC 2710, Listener Discovery (MLD) for IPv6", RFC 2710,
DOI 10.17487/RFC2710, October 1999, DOI 10.17487/RFC2710, October 1999,
<http://www.rfc-editor.org/info/rfc2710>. <http://www.rfc-editor.org/info/rfc2710>.
[RFC2736] Handley, M. and C. Perkins, "Guidelines for Writers of RTP [RFC2736] Handley, M. and C. Perkins, "Guidelines for Writers of RTP
Payload Format Specifications", BCP 36, RFC 2736, Payload Format Specifications", BCP 36, RFC 2736,
DOI 10.17487/RFC2736, December 1999, DOI 10.17487/RFC2736, December 1999,
<http://www.rfc-editor.org/info/rfc2736>. <http://www.rfc-editor.org/info/rfc2736>.
skipping to change at page 44, line 44 skipping to change at page 44, line 47
[RFC3436] Jungmaier, A., Rescorla, E., and M. Tuexen, "Transport [RFC3436] Jungmaier, A., Rescorla, E., and M. Tuexen, "Transport
Layer Security over Stream Control Transmission Protocol", Layer Security over Stream Control Transmission Protocol",
RFC 3436, DOI 10.17487/RFC3436, December 2002, RFC 3436, DOI 10.17487/RFC3436, December 2002,
<http://www.rfc-editor.org/info/rfc3436>. <http://www.rfc-editor.org/info/rfc3436>.
[RFC3450] Luby, M., Gemmell, J., Vicisano, L., Rizzo, L., and J. [RFC3450] Luby, M., Gemmell, J., Vicisano, L., Rizzo, L., and J.
Crowcroft, "Asynchronous Layered Coding (ALC) Protocol Crowcroft, "Asynchronous Layered Coding (ALC) Protocol
Instantiation", RFC 3450, DOI 10.17487/RFC3450, December Instantiation", RFC 3450, DOI 10.17487/RFC3450, December
2002, <http://www.rfc-editor.org/info/rfc3450>. 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. [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550, Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
July 2003, <http://www.rfc-editor.org/info/rfc3550>. July 2003, <http://www.rfc-editor.org/info/rfc3550>.
[RFC3738] Luby, M. and V. Goyal, "Wave and Equation Based Rate [RFC3738] Luby, M. and V. Goyal, "Wave and Equation Based Rate
Control (WEBRC) Building Block", RFC 3738, Control (WEBRC) Building Block", RFC 3738,
DOI 10.17487/RFC3738, April 2004, DOI 10.17487/RFC3738, April 2004,
<http://www.rfc-editor.org/info/rfc3738>. <http://www.rfc-editor.org/info/rfc3738>.
skipping to change at page 45, line 31 skipping to change at page 45, line 31
[RFC3926] Paila, T., Luby, M., Lehtonen, R., Roca, V., and R. Walsh, [RFC3926] Paila, T., Luby, M., Lehtonen, R., Roca, V., and R. Walsh,
"FLUTE - File Delivery over Unidirectional Transport", "FLUTE - File Delivery over Unidirectional Transport",
RFC 3926, DOI 10.17487/RFC3926, October 2004, RFC 3926, DOI 10.17487/RFC3926, October 2004,
<http://www.rfc-editor.org/info/rfc3926>. <http://www.rfc-editor.org/info/rfc3926>.
[RFC3971] Arkko, J., Ed., Kempf, J., Zill, B., and P. Nikander, [RFC3971] Arkko, J., Ed., Kempf, J., Zill, B., and P. Nikander,
"SEcure Neighbor Discovery (SEND)", RFC 3971, "SEcure Neighbor Discovery (SEND)", RFC 3971,
DOI 10.17487/RFC3971, March 2005, DOI 10.17487/RFC3971, March 2005,
<http://www.rfc-editor.org/info/rfc3971>. <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 [RFC4336] Floyd, S., Handley, M., and E. Kohler, "Problem Statement
for the Datagram Congestion Control Protocol (DCCP)", for the Datagram Congestion Control Protocol (DCCP)",
RFC 4336, DOI 10.17487/RFC4336, March 2006, RFC 4336, DOI 10.17487/RFC4336, March 2006,
<http://www.rfc-editor.org/info/rfc4336>. <http://www.rfc-editor.org/info/rfc4336>.
[RFC4340] Kohler, E., Handley, M., and S. Floyd, "Datagram [RFC4340] Kohler, E., Handley, M., and S. Floyd, "Datagram
Congestion Control Protocol (DCCP)", RFC 4340, Congestion Control Protocol (DCCP)", RFC 4340,
DOI 10.17487/RFC4340, March 2006, DOI 10.17487/RFC4340, March 2006,
<http://www.rfc-editor.org/info/rfc4340>. <http://www.rfc-editor.org/info/rfc4340>.
skipping to change at page 46, line 36 skipping to change at page 46, line 30
[RFC4821] Mathis, M. and J. Heffner, "Packetization Layer Path MTU [RFC4821] Mathis, M. and J. Heffner, "Packetization Layer Path MTU
Discovery", RFC 4821, DOI 10.17487/RFC4821, March 2007, Discovery", RFC 4821, DOI 10.17487/RFC4821, March 2007,
<http://www.rfc-editor.org/info/rfc4821>. <http://www.rfc-editor.org/info/rfc4821>.
[RFC4828] Floyd, S. and E. Kohler, "TCP Friendly Rate Control [RFC4828] Floyd, S. and E. Kohler, "TCP Friendly Rate Control
(TFRC): The Small-Packet (SP) Variant", RFC 4828, (TFRC): The Small-Packet (SP) Variant", RFC 4828,
DOI 10.17487/RFC4828, April 2007, DOI 10.17487/RFC4828, April 2007,
<http://www.rfc-editor.org/info/rfc4828>. <http://www.rfc-editor.org/info/rfc4828>.
[RFC4861] Narten, T., Nordmark, E., Simpson, W., and H. Soliman,
"Neighbor Discovery for IP version 6 (IPv6)", RFC 4861,
DOI 10.17487/RFC4861, September 2007,
<http://www.rfc-editor.org/info/rfc4861>.
[RFC4895] Tuexen, M., Stewart, R., Lei, P., and E. Rescorla, [RFC4895] Tuexen, M., Stewart, R., Lei, P., and E. Rescorla,
"Authenticated Chunks for the Stream Control Transmission "Authenticated Chunks for the Stream Control Transmission
Protocol (SCTP)", RFC 4895, DOI 10.17487/RFC4895, August Protocol (SCTP)", RFC 4895, DOI 10.17487/RFC4895, August
2007, <http://www.rfc-editor.org/info/rfc4895>. 2007, <http://www.rfc-editor.org/info/rfc4895>.
[RFC4960] Stewart, R., Ed., "Stream Control Transmission Protocol", [RFC4960] Stewart, R., Ed., "Stream Control Transmission Protocol",
RFC 4960, DOI 10.17487/RFC4960, September 2007, RFC 4960, DOI 10.17487/RFC4960, September 2007,
<http://www.rfc-editor.org/info/rfc4960>. <http://www.rfc-editor.org/info/rfc4960>.
[RFC5052] Watson, M., Luby, M., and L. Vicisano, "Forward Error
Correction (FEC) Building Block", RFC 5052,
DOI 10.17487/RFC5052, August 2007,
<http://www.rfc-editor.org/info/rfc5052>.
[RFC5061] Stewart, R., Xie, Q., Tuexen, M., Maruyama, S., and M. [RFC5061] Stewart, R., Xie, Q., Tuexen, M., Maruyama, S., and M.
Kozuka, "Stream Control Transmission Protocol (SCTP) Kozuka, "Stream Control Transmission Protocol (SCTP)
Dynamic Address Reconfiguration", RFC 5061, Dynamic Address Reconfiguration", RFC 5061,
DOI 10.17487/RFC5061, September 2007, DOI 10.17487/RFC5061, September 2007,
<http://www.rfc-editor.org/info/rfc5061>. <http://www.rfc-editor.org/info/rfc5061>.
[RFC5097] Renker, G. and G. Fairhurst, "MIB for the UDP-Lite [RFC5097] Renker, G. and G. Fairhurst, "MIB for the UDP-Lite
protocol", RFC 5097, DOI 10.17487/RFC5097, January 2008, protocol", RFC 5097, DOI 10.17487/RFC5097, January 2008,
<http://www.rfc-editor.org/info/rfc5097>. <http://www.rfc-editor.org/info/rfc5097>.
[RFC5246] Dierks, T. and E. Rescorla, "The Transport Layer Security
(TLS) Protocol Version 1.2", RFC 5246,
DOI 10.17487/RFC5246, August 2008,
<http://www.rfc-editor.org/info/rfc5246>.
[RFC5238] Phelan, T., "Datagram Transport Layer Security (DTLS) over [RFC5238] Phelan, T., "Datagram Transport Layer Security (DTLS) over
the Datagram Congestion Control Protocol (DCCP)", the Datagram Congestion Control Protocol (DCCP)",
RFC 5238, DOI 10.17487/RFC5238, May 2008, RFC 5238, DOI 10.17487/RFC5238, May 2008,
<http://www.rfc-editor.org/info/rfc5238>. <http://www.rfc-editor.org/info/rfc5238>.
[RFC5246] Dierks, T. and E. Rescorla, "The Transport Layer Security
(TLS) Protocol Version 1.2", RFC 5246,
DOI 10.17487/RFC5246, August 2008,
<http://www.rfc-editor.org/info/rfc5246>.
[RFC5348] Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP [RFC5348] Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP
Friendly Rate Control (TFRC): Protocol Specification", Friendly Rate Control (TFRC): Protocol Specification",
RFC 5348, DOI 10.17487/RFC5348, September 2008, RFC 5348, DOI 10.17487/RFC5348, September 2008,
<http://www.rfc-editor.org/info/rfc5348>. <http://www.rfc-editor.org/info/rfc5348>.
[RFC5461] Gont, F., "TCP's Reaction to Soft Errors", RFC 5461, [RFC5461] Gont, F., "TCP's Reaction to Soft Errors", RFC 5461,
DOI 10.17487/RFC5461, February 2009, DOI 10.17487/RFC5461, February 2009,
<http://www.rfc-editor.org/info/rfc5461>. <http://www.rfc-editor.org/info/rfc5461>.
[RFC5595] Fairhurst, G., "The Datagram Congestion Control Protocol [RFC5595] Fairhurst, G., "The Datagram Congestion Control Protocol
skipping to change at page 47, line 48 skipping to change at page 48, line 5
Control Protocol (DCCP) Congestion ID 4: TCP-Friendly Rate Control Protocol (DCCP) Congestion ID 4: TCP-Friendly Rate
Control for Small Packets (TFRC-SP)", RFC 5622, Control for Small Packets (TFRC-SP)", RFC 5622,
DOI 10.17487/RFC5622, August 2009, DOI 10.17487/RFC5622, August 2009,
<http://www.rfc-editor.org/info/rfc5622>. <http://www.rfc-editor.org/info/rfc5622>.
[RFC5651] Luby, M., Watson, M., and L. Vicisano, "Layered Coding [RFC5651] Luby, M., Watson, M., and L. Vicisano, "Layered Coding
Transport (LCT) Building Block", RFC 5651, Transport (LCT) Building Block", RFC 5651,
DOI 10.17487/RFC5651, October 2009, DOI 10.17487/RFC5651, October 2009,
<http://www.rfc-editor.org/info/rfc5651>. <http://www.rfc-editor.org/info/rfc5651>.
[RFC5672] Crocker, D., Ed., "RFC 4871 DomainKeys Identified Mail [RFC5681] Allman, M., Paxson, V., and E. Blanton, "TCP Congestion
(DKIM) Signatures -- Update", RFC 5672, Control", RFC 5681, DOI 10.17487/RFC5681, September 2009,
DOI 10.17487/RFC5672, August 2009, <http://www.rfc-editor.org/info/rfc5681>.
<http://www.rfc-editor.org/info/rfc5672>.
[RFC5740] Adamson, B., Bormann, C., Handley, M., and J. Macker, [RFC5740] Adamson, B., Bormann, C., Handley, M., and J. Macker,
"NACK-Oriented Reliable Multicast (NORM) Transport "NACK-Oriented Reliable Multicast (NORM) Transport
Protocol", RFC 5740, DOI 10.17487/RFC5740, November 2009, Protocol", RFC 5740, DOI 10.17487/RFC5740, November 2009,
<http://www.rfc-editor.org/info/rfc5740>. <http://www.rfc-editor.org/info/rfc5740>.
[RFC5762] Perkins, C., "RTP and the Datagram Congestion Control
Protocol (DCCP)", RFC 5762, DOI 10.17487/RFC5762, April
2010, <http://www.rfc-editor.org/info/rfc5762>.
[RFC5775] Luby, M., Watson, M., and L. Vicisano, "Asynchronous [RFC5775] Luby, M., Watson, M., and L. Vicisano, "Asynchronous
Layered Coding (ALC) Protocol Instantiation", RFC 5775, Layered Coding (ALC) Protocol Instantiation", RFC 5775,
DOI 10.17487/RFC5775, April 2010, DOI 10.17487/RFC5775, April 2010,
<http://www.rfc-editor.org/info/rfc5775>. <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- [RFC6056] Larsen, M. and F. Gont, "Recommendations for Transport-
Protocol Port Randomization", BCP 156, RFC 6056, Protocol Port Randomization", BCP 156, RFC 6056,
DOI 10.17487/RFC6056, January 2011, DOI 10.17487/RFC6056, January 2011,
<http://www.rfc-editor.org/info/rfc6056>. <http://www.rfc-editor.org/info/rfc6056>.
[RFC6083] Tuexen, M., Seggelmann, R., and E. Rescorla, "Datagram [RFC6083] Tuexen, M., Seggelmann, R., and E. Rescorla, "Datagram
Transport Layer Security (DTLS) for Stream Control Transport Layer Security (DTLS) for Stream Control
Transmission Protocol (SCTP)", RFC 6083, Transmission Protocol (SCTP)", RFC 6083,
DOI 10.17487/RFC6083, January 2011, DOI 10.17487/RFC6083, January 2011,
<http://www.rfc-editor.org/info/rfc6083>. <http://www.rfc-editor.org/info/rfc6083>.
[RFC6093] Gont, F. and A. Yourtchenko, "On the Implementation of the [RFC6093] Gont, F. and A. Yourtchenko, "On the Implementation of the
TCP Urgent Mechanism", RFC 6093, DOI 10.17487/RFC6093, TCP Urgent Mechanism", RFC 6093, DOI 10.17487/RFC6093,
January 2011, <http://www.rfc-editor.org/info/rfc6093>. January 2011, <http://www.rfc-editor.org/info/rfc6093>.
[RFC6101] Freier, A., Karlton, P., and P. Kocher, "The Secure [RFC6101] Freier, A., Karlton, P., and P. Kocher, "The Secure
Sockets Layer (SSL) Protocol Version 3.0", RFC 6101, Sockets Layer (SSL) Protocol Version 3.0", RFC 6101,
DOI 10.17487/RFC6101, August 2011, DOI 10.17487/RFC6101, August 2011,
<http://www.rfc-editor.org/info/rfc6101>. <http://www.rfc-editor.org/info/rfc6101>.
[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>.
[RFC6347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer [RFC6347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer
Security Version 1.2", RFC 6347, DOI 10.17487/RFC6347, Security Version 1.2", RFC 6347, DOI 10.17487/RFC6347,
January 2012, <http://www.rfc-editor.org/info/rfc6347>. January 2012, <http://www.rfc-editor.org/info/rfc6347>.
[RFC6356] Raiciu, C., Handley, M., and D. Wischik, "Coupled [RFC6356] Raiciu, C., Handley, M., and D. Wischik, "Coupled
Congestion Control for Multipath Transport Protocols", Congestion Control for Multipath Transport Protocols",
RFC 6356, DOI 10.17487/RFC6356, October 2011, RFC 6356, DOI 10.17487/RFC6356, October 2011,
<http://www.rfc-editor.org/info/rfc6356>. <http://www.rfc-editor.org/info/rfc6356>.
[RFC6363] Watson, M., Begen, A., and V. Roca, "Forward Error [RFC6363] Watson, M., Begen, A., and V. Roca, "Forward Error
Correction (FEC) Framework", RFC 6363, Correction (FEC) Framework", RFC 6363,
DOI 10.17487/RFC6363, October 2011, DOI 10.17487/RFC6363, October 2011,
<http://www.rfc-editor.org/info/rfc6363>. <http://www.rfc-editor.org/info/rfc6363>.
[RFC6458] Stewart, R., Tuexen, M., Poon, K., Lei, P., and V. [RFC6458] Stewart, R., Tuexen, M., Poon, K., Lei, P., and V.
Yasevich, "Sockets API Extensions for the Stream Control Yasevich, "Sockets API Extensions for the Stream Control
Transmission Protocol (SCTP)", RFC 6458, Transmission Protocol (SCTP)", RFC 6458,
DOI 10.17487/RFC6458, December 2011, DOI 10.17487/RFC6458, December 2011,
<http://www.rfc-editor.org/info/rfc6458>. <http://www.rfc-editor.org/info/rfc6458>.
[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>.
[RFC6582] Henderson, T., Floyd, S., Gurtov, A., and Y. Nishida, "The [RFC6582] Henderson, T., Floyd, S., Gurtov, A., and Y. Nishida, "The
NewReno Modification to TCP's Fast Recovery Algorithm", NewReno Modification to TCP's Fast Recovery Algorithm",
RFC 6582, DOI 10.17487/RFC6582, April 2012, RFC 6582, DOI 10.17487/RFC6582, April 2012,
<http://www.rfc-editor.org/info/rfc6582>. <http://www.rfc-editor.org/info/rfc6582>.
[RFC6584] Roca, V., "Simple Authentication Schemes for the [RFC6584] Roca, V., "Simple Authentication Schemes for the
Asynchronous Layered Coding (ALC) and NACK-Oriented Asynchronous Layered Coding (ALC) and NACK-Oriented
Reliable Multicast (NORM) Protocols", RFC 6584, Reliable Multicast (NORM) Protocols", RFC 6584,
DOI 10.17487/RFC6584, April 2012, DOI 10.17487/RFC6584, April 2012,
<http://www.rfc-editor.org/info/rfc6584>. <http://www.rfc-editor.org/info/rfc6584>.
skipping to change at page 52, line 5 skipping to change at page 52, line 10
"Recommendations for Secure Use of Transport Layer "Recommendations for Secure Use of Transport Layer
Security (TLS) and Datagram Transport Layer Security Security (TLS) and Datagram Transport Layer Security
(DTLS)", BCP 195, RFC 7525, DOI 10.17487/RFC7525, May (DTLS)", BCP 195, RFC 7525, DOI 10.17487/RFC7525, May
2015, <http://www.rfc-editor.org/info/rfc7525>. 2015, <http://www.rfc-editor.org/info/rfc7525>.
[RFC7540] Belshe, M., Peon, R., and M. Thomson, Ed., "Hypertext [RFC7540] Belshe, M., Peon, R., and M. Thomson, Ed., "Hypertext
Transfer Protocol Version 2 (HTTP/2)", RFC 7540, Transfer Protocol Version 2 (HTTP/2)", RFC 7540,
DOI 10.17487/RFC7540, May 2015, DOI 10.17487/RFC7540, May 2015,
<http://www.rfc-editor.org/info/rfc7540>. <http://www.rfc-editor.org/info/rfc7540>.
[I-D.ietf-tsvwg-rfc5405bis] [RFC8085] Eggert, L., Fairhurst, G., and G. Shepherd, "UDP Usage
Eggert, L., Fairhurst, G., and G. Shepherd, "UDP Usage Guidelines", BCP 145, RFC 8085, DOI 10.17487/RFC8085,
Guidelines", draft-ietf-tsvwg-rfc5405bis-16 (work in March 2017, <http://www.rfc-editor.org/info/rfc8085>.
progress), July 2016.
[I-D.ietf-tsvwg-sctp-dtls-encaps] [SCTP-DTLS-ENCAPS]
Tuexen, M., Stewart, R., Jesup, R., and S. Loreto, "DTLS Tuexen, M., Stewart, R., Jesup, R., and S. Loreto, "DTLS
Encapsulation of SCTP Packets", draft-ietf-tsvwg-sctp- Encapsulation of SCTP Packets", Work in Progress,
dtls-encaps-09 (work in progress), January 2015. draft-ietf-tsvwg-sctp-dtls-encaps-09, January 2015.
[I-D.ietf-tsvwg-sctp-ndata] [SCTP-NDATA]
Stewart, R., Tuexen, M., Loreto, S., and R. Seggelmann, Stewart, R., Tuexen, M., Loreto, S., and R. Seggelmann,
"Stream Schedulers and User Message Interleaving for the "Stream Schedulers and User Message Interleaving for the
Stream Control Transmission Protocol", draft-ietf-tsvwg- Stream Control Transmission Protocol", Work in Progress,
sctp-ndata-07 (work in progress), July 2016. draft-ietf-tsvwg-sctp-ndata-08, October 2016.
[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-09 (work in progress),
May 2016.
[I-D.ietf-tcpm-cubic] [TCP-SPEC] Eddy, W., Ed., "Transmission Control Protocol
Rhee, I., Xu, L., Ha, S., Zimmermann, A., Eggert, L., and Specification", Work in Progress, draft-ietf-tcpm-
R. Scheffenegger, "CUBIC for Fast Long-Distance Networks", rfc793bis-04, December 2016.
draft-ietf-tcpm-cubic-02 (work in progress), August 2016.
[I-D.ietf-rtcweb-transports] [TLS-1.3] Rescorla, E., "The Transport Layer Security (TLS) Protocol
Alvestrand, H., "Transports for WebRTC", draft-ietf- Version 1.3", Work in Progress, draft-ietf-tls-tls13-18,
rtcweb-transports-15 (work in progress), August 2016. October 2016.
[I-D.ietf-tls-tls13] [WEBRTC-TRANS]
Rescorla, E., "The Transport Layer Security (TLS) Protocol Alvestrand, H., "Transports for WebRTC", Work in
Version 1.3", draft-ietf-tls-tls13-14 (work in progress), Progress, draft-ietf-rtcweb-transports-17, October 2016.
July 2016.
[XHR] van Kesteren, A., Aubourg, J., Song, J., and H. Steen, [XHR] van Kesteren, A., Aubourg, J., Song, J., and H. Steen,
"XMLHttpRequest working draft "XMLHttpRequest Level 1", World Wide Web Consortium NOTE-
(http://www.w3.org/TR/XMLHttpRequest/)", 2000. XMLHttpRequest-20161006, October 2016,
<http://www.w3.org/TR/XMLHttpRequest/>.
[REST] Fielding, R., "Architectural Styles and the Design of Acknowledgments
Network-based Software Architectures, Ph. D. (UC Irvine),
Chapter 5: Representational State Transfer", 2000.
[POSIX] 1-2008, IEEE., "IEEE Standard for Information Technology Thanks to Joe Touch, Michael Welzl, Spencer Dawkins, and the TAPS
-- Portable Operating System Interface (POSIX) Base working group for the comments, feedback, and discussion. This work
Specifications, Issue 7", n.d.. is supported by the European Commission under grant agreement No.
318627 mPlane and from the Horizon 2020 research and innovation
program under grant agreements No. 644334 (NEAT) and No. 688421
(MAMI). This support does not imply endorsement.
[MBMS] 3GPP TSG WS S4, ., "3GPP TS 26.346: Multimedia Broadcast/ Contributors
Multicast Service (MBMS); Protocols and codecs, release 13
(http://www.3gpp.org/DynaReport/26346.htm).", 2015.
[ClarkArch] In addition to the editors, this document is the work of Brian
Clark, D. and D. Tennenhouse, "Architectural Adamson, Dragana Damjanovic, Kevin Fall, Simone Ferlin-Oliviera,
Considerations for a New Generation of Protocols (Proc. Ralph Holz, Olivier Mehani, Karen Nielsen, Colin Perkins, Vincent
ACM SIGCOMM)", 1990. Roca, and Michael Tuexen.
o Section 3.2 on MPTCP was contributed by Simone Ferlin-Oliviera
(ferlin@simula.no) and Olivier Mehani
(olivier.mehani@nicta.com.au).
o Section 3.3 on UDP was contributed by Kevin Fall
(kfall@kfall.com).
o Section 3.5 on SCTP was contributed by Michael Tuexen (tuexen@fh-
muenster.de) and Karen Nielsen (karen.nielsen@tieto.com).
o Section 3.7 on TLS and DTLS was contributed by Ralph Holz
(ralph.holz@nicta.com.au) and Olivier Mehani
(olivier.mehani@nicta.com.au).
o Section 3.8 on RTP contains contributions from Colin Perkins
(csp@csperkins.org).
o Section 3.9 on HTTP was contributed by Dragana Damjanovic
(ddamjanovic@mozilla.com).
o Section 3.10 on FLUTE/ALC was contributed by Vincent Roca
(vincent.roca@inria.fr).
o Section 3.11 on NORM was contributed by Brian Adamson
(brian.adamson@nrl.navy.mil).
Authors' Addresses Authors' Addresses
Godred Fairhurst (editor) Godred Fairhurst (editor)
University of Aberdeen University of Aberdeen
School of Engineering, Fraser Noble Building School of Engineering, Fraser Noble Building
Aberdeen AB24 3UE Aberdeen AB24 3UE
Email: gorry@erg.abdn.ac.uk Email: gorry@erg.abdn.ac.uk
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