Internet Engineering Task Force (IETF) G. Fairhurst, Ed.
Request for Comments: 8095 University of Aberdeen
Category: Informational B. Trammell, Ed.
ISSN: 2070-1721 M. Kuehlewind, Ed.
ETH Zurich
March 2017
Services Provided by
IETF Transport Protocols and Congestion Control Mechanisms
Abstract
This document describes, surveys, and classifies the protocol
mechanisms provided by existing IETF protocols, as background for
determining a common set of transport services. It examines the
Transmission Control Protocol (TCP), Multipath TCP, the Stream
Control Transmission Protocol (SCTP), the User Datagram Protocol
(UDP), UDP-Lite, the Datagram Congestion Control Protocol (DCCP), the
Internet Control Message Protocol (ICMP), the Real-Time Transport
Protocol (RTP), File Delivery over Unidirectional Transport /
Asynchronous Layered Coding (FLUTE/ALC) for Reliable Multicast, NACK-
Oriented Reliable Multicast (NORM), Transport Layer Security (TLS),
Datagram TLS (DTLS), and the Hypertext Transport Protocol (HTTP),
when HTTP is used as a pseudotransport. This survey provides
background for the definition of transport services within the TAPS
working group.
Status of This Memo
This document is not an Internet Standards Track specification; it is
published for informational purposes.
This document is a product of the Internet Engineering Task Force
(IETF). It represents the consensus of the IETF community. It has
received public review and has been approved for publication by the
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.
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.
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Copyright Notice
Copyright (c) 2017 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
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the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Table of Contents
1. Introduction ....................................................4
1.1. Overview of Transport Features .............................4
2. Terminology .....................................................5
3. Existing Transport Protocols ....................................6
3.1. Transport Control Protocol (TCP) ...........................6
3.1.1. Protocol Description ................................6
3.1.2. Interface Description ...............................8
3.1.3. Transport Features ..................................9
3.2. Multipath TCP (MPTCP) .....................................10
3.2.1. Protocol Description ...............................10
3.2.2. Interface Description ..............................10
3.2.3. Transport Features .................................11
3.3. User Datagram Protocol (UDP) ..............................11
3.3.1. Protocol Description ...............................11
3.3.2. Interface Description ..............................12
3.3.3. Transport Features .................................13
3.4. Lightweight User Datagram Protocol (UDP-Lite) .............13
3.4.1. Protocol Description ...............................13
3.4.2. Interface Description ..............................14
3.4.3. Transport Features .................................14
3.5. Stream Control Transmission Protocol (SCTP) ...............14
3.5.1. Protocol Description ...............................15
3.5.2. Interface Description ..............................17
3.5.3. Transport Features .................................19
3.6. Datagram Congestion Control Protocol (DCCP) ...............20
3.6.1. Protocol Description ...............................21
3.6.2. Interface Description ..............................22
3.6.3. Transport Features .................................22
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3.7. Transport Layer Security (TLS) and Datagram TLS
(DTLS) as a Pseudotransport ...............................23
3.7.1. Protocol Description ...............................23
3.7.2. Interface Description ..............................24
3.7.3. Transport Features .................................25
3.8. Real-Time Transport Protocol (RTP) ........................26
3.8.1. Protocol Description ...............................26
3.8.2. Interface Description ..............................27
3.8.3. Transport Features .................................27
3.9. Hypertext Transport Protocol (HTTP) over TCP as a
Pseudotransport ...........................................28
3.9.1. Protocol Description ...............................28
3.9.2. Interface Description ..............................29
3.9.3. Transport Features .................................30
3.10. File Delivery over Unidirectional Transport /
Asynchronous Layered Coding (FLUTE/ALC) for
Reliable Multicast .......................................31
3.10.1. Protocol Description ..............................31
3.10.2. Interface Description .............................33
3.10.3. Transport Features ................................33
3.11. NACK-Oriented Reliable Multicast (NORM) ..................34
3.11.1. Protocol Description ..............................34
3.11.2. Interface Description .............................35
3.11.3. Transport Features ................................36
3.12. Internet Control Message Protocol (ICMP) .................36
3.12.1. Protocol Description ..............................37
3.12.2. Interface Description .............................37
3.12.3. Transport Features ................................38
4. Congestion Control .............................................38
5. Transport Features .............................................39
6. IANA Considerations ............................................42
7. Security Considerations ........................................42
8. Informative References .........................................42
Acknowledgments ...................................................53
Contributors ......................................................53
Authors' Addresses ................................................54
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1. Introduction
Internet applications make use of the services provided by a
transport protocol, such as TCP (a reliable, in-order stream
protocol) or UDP (an unreliable datagram protocol). We use the term
"transport service" to mean the end-to-end service provided to an
application by the transport layer. That service can only be
provided correctly if information about the intended usage is
supplied from the application. The application may determine this
information at design time, compile time, or run time, and may
include guidance on whether a feature is required, a preference by
the application, or something in between. Examples of features of
transport services are reliable delivery, ordered delivery, content
privacy to in-path devices, and integrity protection.
The IETF has defined a wide variety of transport protocols beyond TCP
and UDP, including SCTP, DCCP, MPTCP, and UDP-Lite. Transport
services may be provided directly by these transport protocols or
layered on top of them using protocols such as WebSockets (which runs
over TCP), RTP (over TCP or UDP) or WebRTC data channels (which run
over SCTP over DTLS over UDP or TCP). Services built on top of UDP
or UDP-Lite typically also need to specify additional mechanisms,
including a congestion control mechanism (such as NewReno [RFC6582],
TCP-Friendly Rate Control (TFRC) [RFC5348], or Low Extra Delay
Background Transport (LEDBAT) [RFC6817]). This extends the set of
available transport services beyond those provided to applications by
TCP and UDP.
The transport protocols described in this document provide a basis
for the definition of transport services provided by common
protocols, as background for the TAPS working group. The protocols
listed here were chosen to help expose as many potential transport
services as possible and are not meant to be a comprehensive survey
or classification of all transport protocols.
1.1. Overview of Transport Features
Transport protocols can be differentiated by the features of the
services they provide.
Some of these provided features are closely related to basic control
function that a protocol needs to work over a network path, such as
addressing. The number of participants in a given association also
determines its applicability: a connection can be between endpoints
(unicast), to one of multiple endpoints (anycast), or simultaneously
to multiple endpoints (multicast). Unicast protocols usually support
bidirectional communication, while multicast is generally
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unidirectional. Another feature is whether a transport requires a
control exchange across the network at setup (e.g., TCP) or whether
it is connectionless (e.g., UDP).
For packet delivery itself, reliability and integrity protection,
ordering, and framing are basic features. However, these features
are implemented with different levels of assurance in different
protocols. As an example, a transport service may provide full
reliability, with detection of loss and retransmission (e.g., TCP).
SCTP offers a message-based service that can provide full or partial
reliability and allows the protocol to minimize the head-of-line
blocking due to the support of ordered and unordered message delivery
within multiple streams. UDP-Lite and DCCP can provide partial
integrity protection to enable corruption tolerance.
Usually, a protocol has been designed to support one specific type of
delivery/framing: either data needs to be divided into transmission
units based on network packets (datagram service) or a data stream is
segmented and re-combined across multiple packets (stream service).
Whole objects such as files are handled accordingly. This decision
strongly influences the interface that is provided to the upper
layer.
In addition, transport protocols offer a certain support for
transmission control. For example, a transport service can provide
flow control to allow a receiver to regulate the transmission rate of
a sender. Further, a transport service can provide congestion
control (see Section 4). As an example, TCP and SCTP provide
congestion control for use in the Internet, whereas UDP leaves this
function to the upper-layer protocol that uses UDP.
Security features are often provided independently of the transport
protocol, via Transport Layer Security (TLS) (see Section 3.7) or by
the application-layer protocol itself. The security properties TLS
provides to the application (such as confidentiality, integrity, and
authenticity) are also features of the transport layer, even though
they are often presently implemented in a separate protocol.
2. Terminology
The following terms are used throughout this document and in
subsequent documents produced by the TAPS working group that describe
the composition and decomposition of transport services.
Transport Feature: a specific end-to-end feature that the transport
layer provides to an application. Examples include
confidentiality, reliable delivery, ordered delivery, message-
versus-stream orientation, etc.
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Transport Service: a set of transport features, without an
association to any given framing protocol, that provides a
complete service to an application.
Transport Protocol: an implementation that provides one or more
different transport services using a specific framing and header
format on the wire.
Application: an entity that uses the transport layer for end-to-end
delivery data across the network (this may also be an upper-layer
protocol or tunnel encapsulation).
3. Existing Transport Protocols
This section provides a list of known IETF transport protocols and
transport protocol frameworks. It does not make an assessment about
whether specific implementations of protocols are fully compliant to
current IETF specifications.
3.1. Transport Control Protocol (TCP)
TCP is an IETF Standards Track transport protocol. [RFC793]
introduces TCP as follows:
The Transmission Control Protocol (TCP) is intended for use as a
highly reliable host-to-host protocol between hosts in packet-
switched computer communication networks, and in interconnected
systems of such networks.
Since its introduction, TCP has become the default connection-
oriented, stream-based transport protocol in the Internet. It is
widely implemented by endpoints and widely used by common
applications.
3.1.1. Protocol Description
TCP is a connection-oriented protocol that provides a three-way
handshake to allow a client and server to set up a connection and
negotiate features and provides mechanisms for orderly completion and
immediate teardown of a connection [RFC793] [TCP-SPEC]. TCP is
defined by a family of RFCs (see [RFC7414]).
TCP provides multiplexing to multiple sockets on each host using port
numbers. A similar approach is adopted by other IETF-defined
transports. An active TCP session is identified by its four-tuple of
local and remote IP addresses and local and remote port numbers. The
destination port during connection setup is often used to indicate
the requested service.
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TCP partitions a continuous stream of bytes into segments, sized to
fit in IP packets based on a negotiated maximum segment size and
further constrained by the effective Maximum Transmission Unit (MTU)
from Path MTU Discovery (PMTUD). ICMP-based PMTUD [RFC1191]
[RFC1981] as well as Packetization Layer PMTUD (PLPMTUD) [RFC4821]
have been defined by the IETF.
Each byte in the stream is identified by a sequence number. The
sequence number is used to order segments on receipt, to identify
segments in acknowledgments, and to detect unacknowledged segments
for retransmission. This is the basis of the reliable, ordered
delivery of data in a TCP stream. TCP Selective Acknowledgment
(SACK) [RFC2018] extends this mechanism by making it possible to
provide earlier identification of which segments are missing,
allowing faster retransmission. SACK-based methods (e.g., Duplicate
Selective ACK) can also result in less spurious retransmission.
Receiver flow control is provided by a sliding window, which limits
the amount of unacknowledged data that can be outstanding at a given
time. The window scale option [RFC7323] allows a receiver to use
windows greater than 64 KB.
All TCP senders provide congestion control, such as that described in
[RFC5681]. TCP uses a sequence number with a sliding receiver window
for flow control. The TCP congestion control mechanism also utilizes
this TCP sequence number to manage a separate congestion window
[RFC5681]. The sending window at a given point in time is the
minimum of the receiver window and the congestion window. The
congestion window is increased in the absence of congestion and
decreased if congestion is detected. Often, loss is implicitly
handled as a congestion indication, which is detected in TCP (also as
input for retransmission handling) based on two mechanisms: a
retransmission timer with exponential back-off or the reception of
three acknowledgments for the same segment, so called "duplicated
ACKs" (fast retransmit). In addition, Explicit Congestion
Notification (ECN) [RFC3168] can be used in TCP and, if supported by
both endpoints, allows a network node to signal congestion without
inducing loss. Alternatively, a delay-based congestion control
scheme that reacts to changes in delay as an early indication of
congestion can be used in TCP. This is further described in
Section 4. Examples of different kinds of congestion control schemes
are provided in Section 4.
TCP protocol instances can be extended (see [RFC7414]). Some
protocol features may also be tuned to optimize for a specific
deployment scenario. Some features are sender-side only, requiring
no negotiation with the receiver; some are receiver-side only; and
some are explicitly negotiated during connection setup.
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TCP may buffer data, e.g., to optimize processing or capacity usage.
TCP therefore provides mechanisms to control this, including an
optional "PUSH" function [RFC793] that explicitly requests the
transport service not to delay data. By default, TCP segment
partitioning uses Nagle's algorithm [TCP-SPEC] to buffer data at the
sender into large segments, potentially incurring sender-side
buffering delay; this algorithm can be disabled by the sender to
transmit more immediately, e.g., to reduce latency for interactive
sessions.
TCP provides an "urgent data" function for limited out-of-order
delivery of the data. This function is deprecated [RFC6093].
A TCP Reset (RST) control message may be used to force a TCP endpoint
to close a session [RFC793], aborting the connection.
A mandatory checksum provides a basic integrity check against
misdelivery and data corruption over the entire packet. Applications
that require end-to-end integrity of data are recommended to include
a stronger integrity check of their payload data. The TCP checksum
[RFC1071] [RFC2460] does not support partial payload protection (as
in DCCP/UDP-Lite).
TCP supports only unicast connections.
3.1.2. Interface Description
The User/TCP Interface defined in [RFC793] provides six user
commands: Open, Send, Receive, Close, Status, and Abort. This
interface does not describe configuration of TCP options or
parameters aside from the use of the PUSH and URGENT flags.
[RFC1122] describes extensions of the TCP/application-layer interface
for:
o reporting soft errors such as reception of ICMP error messages,
extensive retransmission, or urgent pointer advance,
o providing a possibility to specify the Differentiated Services
Code Point (DSCP) [RFC3260] (formerly, the Type-of-Service (TOS))
for segments,
o providing a flush call to empty the TCP send queue, and
o multihoming support.
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In API implementations derived from the BSD Sockets API, TCP sockets
are created using the "SOCK_STREAM" socket type as described in the
IEEE Portable Operating System Interface (POSIX) Base Specifications
[POSIX]. The features used by a protocol instance may be set and
tuned via this API. There are currently no documents in the RFC
Series that describe this interface.
3.1.3. Transport Features
The transport features provided by TCP are:
o connection-oriented transport with feature negotiation and
application-to-port mapping (implemented using SYN segments and
the TCP Option field to negotiate features),
o unicast transport (though anycast TCP is implemented, at risk of
instability due to rerouting),
o port multiplexing,
o unidirectional or bidirectional communication,
o stream-oriented delivery in a single stream,
o fully reliable delivery (implemented using ACKs sent from the
receiver to confirm delivery),
o error detection (implemented using a segment checksum to verify
delivery to the correct endpoint and integrity of the data and
options),
o segmentation,
o data bundling (optional; uses Nagle's algorithm to coalesce data
sent within the same RTT into full-sized segments),
o flow control (implemented using a window-based mechanism where the
receiver advertises the window that it is willing to buffer), and
o congestion control (usually implemented using a window-based
mechanism and four algorithms for different phases of the
transmission: slow start, congestion avoidance, fast retransmit,
and fast recovery [RFC5681]).
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3.2. Multipath TCP (MPTCP)
Multipath TCP [RFC6824] is an extension for TCP to support
multihoming for resilience, mobility, and load balancing. It is
designed to be as indistinguishable to middleboxes from non-multipath
TCP as possible. It does so by establishing regular TCP flows
between a pair of source/destination endpoints and multiplexing the
application's stream over these flows. Sub-flows can be started over
IPv4 or IPv6 for the same session.
3.2.1. Protocol Description
MPTCP uses TCP options for its control plane. They are used to
signal multipath capabilities, as well as to negotiate data sequence
numbers, advertise other available IP addresses, and establish new
sessions between pairs of endpoints.
By multiplexing one byte stream over separate paths, MPTCP can
achieve a higher throughput than TCP in certain situations. However,
if coupled congestion control [RFC6356] is used, it might limit this
benefit to maintain fairness to other flows at the bottleneck. When
aggregating capacity over multiple paths, and depending on the way
packets are scheduled on each TCP subflow, additional delay and
higher jitter might be observed before in-order delivery of data to
the applications.
3.2.2. Interface Description
By default, MPTCP exposes the same interface as TCP to the
application. [RFC6897], however, describes a richer API for MPTCP-
aware applications.
This Basic API describes how an application can:
o enable or disable MPTCP.
o bind a socket to one or more selected local endpoints.
o query local and remote endpoint addresses.
o get a unique connection identifier (similar to an address-port
pair for TCP).
The document also recommends the use of extensions defined for SCTP
[RFC6458] (see Section 3.5) to support multihoming for resilience and
mobility.
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3.2.3. Transport Features
As an extension to TCP, MPTCP provides mostly the same features. By
establishing multiple sessions between available endpoints, it can
additionally provide soft failover solutions in the case that one of
the paths becomes unusable.
Therefore, the transport features provided by MPTCP in addition to
TCP are:
o multihoming for load balancing, with endpoint multiplexing of a
single byte stream, using either coupled congestion control or
throughput maximization,
o address family multiplexing (using IPv4 and IPv6 for the same
session), and
o resilience to network failure and/or handover.
3.3. User Datagram Protocol (UDP)
The User Datagram Protocol (UDP) [RFC768] [RFC2460] is an IETF
Standards Track transport protocol. It provides a unidirectional
datagram protocol that preserves message boundaries. It provides no
error correction, congestion control, or flow control. It can be
used to send broadcast datagrams (IPv4) or multicast datagrams (IPv4
and IPv6), in addition to unicast and anycast datagrams. IETF
guidance on the use of UDP is provided in [RFC8085]. UDP is widely
implemented and widely used by common applications, including DNS.
3.3.1. Protocol Description
UDP is a connectionless protocol that maintains message boundaries,
with no connection setup or feature negotiation. The protocol uses
independent messages, ordinarily called "datagrams". It provides
detection of payload errors and misdelivery of packets to an
unintended endpoint, both of which result in discard of received
datagrams, with no indication to the user of the service.
It is possible to create IPv4 UDP datagrams with no checksum, and
while this is generally discouraged [RFC1122] [RFC8085], certain
special cases permit this use. These datagrams rely on the IPv4
header checksum to protect from misdelivery to an unintended
endpoint. IPv6 does not permit UDP datagrams with no checksum,
although in certain cases [RFC6936], this rule may be relaxed
[RFC6935].
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UDP does not provide reliability and does not provide retransmission.
Messages may be reordered, lost, or duplicated in transit. Note that
due to the relatively weak form of checksum used by UDP, applications
that require end-to-end integrity of data are recommended to include
a stronger integrity check of their payload data.
Because UDP provides no flow control, a receiving application that is
unable to run sufficiently fast, or frequently, may miss messages.
The lack of congestion handling implies UDP traffic may experience
loss when using an overloaded path and may cause the loss of messages
from other protocols (e.g., TCP) when sharing the same network path.
On transmission, UDP encapsulates each datagram into a single IP
packet or several IP packet fragments. This allows a datagram to be
larger than the effective path MTU. Fragments are reassembled before
delivery to the UDP receiver, making this transparent to the user of
the transport service. When jumbograms are supported, larger
messages may be sent without performing fragmentation.
UDP on its own does not provide support for segmentation, receiver
flow control, congestion control, PMTUD/PLPMTUD, or ECN.
Applications that require these features need to provide them on
their own or use a protocol over UDP that provides them [RFC8085].
3.3.2. Interface Description
[RFC768] describes basic requirements for an API for UDP. Guidance
on the use of common APIs is provided in [RFC8085].
A UDP endpoint consists of a tuple of (IP address, port number).
De-multiplexing using multiple abstract endpoints (sockets) on the
same IP address is supported. The same socket may be used by a
single server to interact with multiple clients. (Note: This
behavior differs from TCP, which uses a pair of tuples to identify a
connection). Multiple server instances (processes) that bind to the
same socket can cooperate to service multiple clients. The socket
implementation arranges to not duplicate the same received unicast
message to multiple server processes.
Many operating systems also allow a UDP socket to be "connected",
i.e., to bind a UDP socket to a specific (remote) UDP endpoint.
Unlike TCP's connect primitive, for UDP, this is only a local
operation that serves to simplify the local send/receive functions
and to filter the traffic for the specified addresses and ports
[RFC8085].
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3.3.3. Transport Features
The transport features provided by UDP are:
o unicast, multicast, anycast, or IPv4 broadcast transport,
o port multiplexing (where a receiving port can be configured to
receive datagrams from multiple senders),
o message-oriented delivery,
o unidirectional or bidirectional communication where the
transmissions in each direction are independent,
o non-reliable delivery,
o unordered delivery, and
o error detection (implemented using a segment checksum to verify
delivery to the correct endpoint and integrity of the data;
optional for IPv4 and optional under specific conditions for IPv6
where all or none of the payload data is protected).
3.4. Lightweight User Datagram Protocol (UDP-Lite)
The Lightweight User Datagram Protocol (UDP-Lite) [RFC3828] is an
IETF Standards Track transport protocol. It provides a
unidirectional, datagram protocol that preserves message boundaries.
IETF guidance on the use of UDP-Lite is provided in [RFC8085]. A
UDP-Lite service may support IPv4 broadcast, multicast, anycast, and
unicast, as well as IPv6 multicast, anycast, and unicast.
Examples of use include a class of applications that can derive
benefit from having partially damaged payloads delivered rather than
discarded. One use is to provide header integrity checks but allow
delivery of corrupted payloads to error-tolerant applications or to
applications that use some other mechanism to provide payload
integrity (see [RFC6936]).
3.4.1. Protocol Description
Like UDP, UDP-Lite is a connectionless datagram protocol, with no
connection setup or feature negotiation. It changes the semantics of
the UDP Payload Length field to that of a Checksum Coverage Length
field and is identified by a different IP protocol/next-header value.
The Checksum Coverage Length field specifies the intended checksum
coverage, with the remaining unprotected part of the payload called
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the "error-insensitive part". Therefore, applications using UDP-Lite
cannot make assumptions regarding the correctness of the data
received in the insensitive part of the UDP-Lite payload.
Otherwise, UDP-Lite is semantically identical to UDP. In the same
way as for UDP, mechanisms for receiver flow control, congestion
control, PMTU or PLPMTU discovery, support for ECN, etc., need to be
provided by upper-layer protocols [RFC8085].
3.4.2. Interface Description
There is no API currently specified in the RFC Series, but guidance
on use of common APIs is provided in [RFC8085].
The interface of UDP-Lite differs from that of UDP by the addition of
a single (socket) option that communicates a checksum coverage length
value. The checksum coverage may also be made visible to the
application via the UDP-Lite MIB module [RFC5097].
3.4.3. Transport Features
The transport features provided by UDP-Lite are:
o unicast, multicast, anycast, or IPv4 broadcast transport (same as
for UDP),
o port multiplexing (same as for UDP),
o message-oriented delivery (same as for UDP),
o unidirectional or bidirectional communication where the
transmissions in each direction are independent (same as for UDP),
o non-reliable delivery (same as for UDP),
o non-ordered delivery (same as for UDP), and
o partial or full payload error detection (where the Checksum
Coverage field indicates the size of the payload data covered by
the checksum).
3.5. Stream Control Transmission Protocol (SCTP)
SCTP is a message-oriented IETF Standards Track transport protocol.
The base protocol is specified in [RFC4960]. It supports multihoming
and path failover to provide resilience to path failures. An SCTP
association has multiple streams in each direction, providing
in-sequence delivery of user messages within each stream. This
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allows it to minimize head-of-line blocking. SCTP supports multiple
stream- scheduling schemes controlling stream multiplexing, including
priority and fair weighting schemes.
SCTP was originally developed for transporting telephony signaling
messages and is deployed in telephony signaling networks, especially
in mobile telephony networks. It can also be used for other
services, for example, in the WebRTC framework for data channels.
3.5.1. Protocol Description
SCTP is a connection-oriented protocol using a four-way handshake to
establish an SCTP association and a three-way message exchange to
gracefully shut it down. It uses the same port number concept as
DCCP, TCP, UDP, and UDP-Lite. SCTP only supports unicast.
SCTP uses the 32-bit CRC32c for protecting SCTP packets against bit
errors and misdelivery of packets to an unintended endpoint. This is
stronger than the 16-bit checksums used by TCP or UDP. However,
partial payload checksum coverage as provided by DCCP or UDP-Lite is
not supported.
SCTP has been designed with extensibility in mind. A common header
is followed by a sequence of chunks. [RFC4960] defines how a
receiver processes chunks with an unknown chunk type. The support of
extensions can be negotiated during the SCTP handshake. Currently
defined extensions include mechanisms for dynamic reconfiguration of
streams [RFC6525] and IP addresses [RFC5061]. Furthermore, the
extension specified in [RFC3758] introduces the concept of partial
reliability for user messages.
SCTP provides a message-oriented service. Multiple small user
messages can be bundled into a single SCTP packet to improve
efficiency. For example, this bundling may be done by delaying user
messages at the sender, similar to Nagle's algorithm used by TCP.
User messages that would result in IP packets larger than the MTU
will be fragmented at the sender and reassembled at the receiver.
There is no protocol limit on the user message size. For MTU
discovery, the same mechanism as for TCP can be used [RFC1981]
[RFC4821], as well as utilization of probe packets with padding
chunks, as defined in [RFC4820].
[RFC4960] specifies TCP-friendly congestion control to protect the
network against overload. SCTP also uses sliding window flow control
to protect receivers against overflow. Similar to TCP, SCTP also
supports delaying acknowledgments. [RFC7053] provides a way for the
sender of user messages to request immediate sending of the
corresponding acknowledgments.
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Each SCTP association has between 1 and 65536 unidirectional streams
in each direction. The number of streams can be different in each
direction. Every user message is sent on a particular stream. User
messages can be sent unordered or ordered upon request by the upper
layer. Unordered messages can be delivered as soon as they are
completely received. For user messages not requiring fragmentation,
this minimizes head-of-line blocking. On the other hand, ordered
messages sent on the same stream are delivered at the receiver in the
same order as sent by the sender.
The base protocol defined in [RFC4960] does not allow interleaving of
user messages. Large messages on one stream can therefore block the
sending of user messages on other streams. [SCTP-NDATA] describes a
method to overcome this limitation. This document also specifies
multiple algorithms for the sender-side selection of which streams to
send data from, supporting a variety of scheduling algorithms
including priority-based methods. The stream reconfiguration
extension defined in [RFC6525] allows streams to be reset during the
lifetime of an association and to increase the number of streams, if
the number of streams negotiated in the SCTP handshake becomes
insufficient.
Each user message sent is delivered to the receiver or, in case of
excessive retransmissions, the association is terminated in a
non-graceful way [RFC4960], similar to TCP behavior. In addition to
this reliable transfer, the partial reliability extension [RFC3758]
allows a sender to abandon user messages. The application can
specify the policy for abandoning user messages.
SCTP supports multihoming. Each SCTP endpoint uses a list of IP
addresses and a single port number. These addresses can be any
mixture of IPv4 and IPv6 addresses. These addresses are negotiated
during the handshake, and the address reconfiguration extension
specified in [RFC5061] in combination with [RFC4895] can be used to
change these addresses in an authenticated way during the lifetime of
an SCTP association. This allows for transport-layer mobility.
Multiple addresses are used for improved resilience. If a remote
address becomes unreachable, the traffic is switched over to a
reachable one, if one exists.
For securing user messages, the use of TLS over SCTP has been
specified in [RFC3436]. However, this solution does not support all
services provided by SCTP, such as unordered delivery or partial
reliability. Therefore, the use of DTLS over SCTP has been specified
in [RFC6083] to overcome these limitations. When using DTLS over
SCTP, the application can use almost all services provided by SCTP.
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[NAT-SUPP] defines methods for endpoints and middleboxes to provide
NAT traversal for SCTP over IPv4. For legacy NAT traversal,
[RFC6951] defines the UDP encapsulation of SCTP packets.
Alternatively, SCTP packets can be encapsulated in DTLS packets as
specified in [SCTP-DTLS-ENCAPS]. The latter encapsulation is used
within the WebRTC [WEBRTC-TRANS] context.
An SCTP ABORT chunk may be used to force a SCTP endpoint to close a
session [RFC4960], aborting the connection.
SCTP has a well-defined API, described in the next subsection.
3.5.2. Interface Description
[RFC4960] defines an abstract API for the base protocol. This API
describes the following functions callable by the upper layer of
SCTP: Initialize, Associate, Send, Receive, Receive Unsent Message,
Receive Unacknowledged Message, Shutdown, Abort, SetPrimary, Status,
Change Heartbeat, Request Heartbeat, Get SRTT Report, Set Failure
Threshold, Set Protocol Parameters, and Destroy. The following
notifications are provided by the SCTP stack to the upper layer:
COMMUNICATION UP, DATA ARRIVE, SHUTDOWN COMPLETE, COMMUNICATION LOST,
COMMUNICATION ERROR, RESTART, SEND FAILURE, and NETWORK STATUS
CHANGE.
An extension to the BSD Sockets API is defined in [RFC6458] and
covers:
o the base protocol defined in [RFC4960]. The API allows control
over local addresses and port numbers and the primary path.
Furthermore, the application has fine control of parameters like
retransmission thresholds, the path supervision, the delayed
acknowledgment timeout, and the fragmentation point. The API
provides a mechanism to allow the SCTP stack to notify the
application about events if the application has requested them.
These notifications provide information about status changes of
the association and each of the peer addresses. In case of send
failures, including drop of messages sent unreliably, the
application can also be notified, and user messages can be
returned to the application. When sending user messages, the
application can indicate a stream id, a payload protocol
identifier, and an indication of whether ordered delivery is
requested. These parameters can also be provided on message
reception. Additionally, a context can be provided when sending,
which can be used in case of send failures. The sending of
arbitrarily large user messages is supported.
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o the SCTP Partial Reliability extension defined in [RFC3758] to
specify for a user message the Partially Reliable SCTP (PR-SCTP)
policy and the policy-specific parameter. Examples of these
policies defined in [RFC3758] and [RFC7496] are:
* limiting the time a user message is dealt with by the sender.
* limiting the number of retransmissions for each fragment of a
user message. If the number of retransmissions is limited to
0, one gets a service similar to UDP.
* abandoning messages of lower priority in case of a send buffer
shortage.
o the SCTP Authentication extension defined in [RFC4895] allowing
management of the shared keys and allowing the HMAC to use and set
the chunk types (which are only accepted in an authenticated way)
and get the list of chunks that are accepted by the local and
remote endpoints in an authenticated way.
o the SCTP Dynamic Address Reconfiguration extension defined in
[RFC5061]. It allows the manual addition and deletion of local
addresses for SCTP associations, as well as the enabling of
automatic address addition and deletion. Furthermore, the peer
can be given a hint for choosing its primary path.
A BSD Sockets API extension has been defined in the documents that
specify the following SCTP extensions:
o the SCTP Stream Reconfiguration extension defined in [RFC6525].
The API allows triggering of the reset operation for incoming and
outgoing streams and the whole association. It also provides a
way to notify the association about the corresponding events.
Furthermore, the application can increase the number of streams.
o the UDP Encapsulation of SCTP packets extension defined in
[RFC6951]. The API allows the management of the remote UDP
encapsulation port.
o the SCTP SACK-IMMEDIATELY extension defined in [RFC7053]. The API
allows the sender of a user message to request the receiver to
send the corresponding acknowledgment immediately.
o the additional PR-SCTP policies defined in [RFC7496]. The API
allows enabling/disabling the PR-SCTP extension, choosing the
PR-SCTP policies defined in the document, and providing
statistical information about abandoned messages.
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Future documents describing SCTP extensions are expected to describe
the corresponding BSD Sockets API extension in a "Socket API
Considerations" section.
The SCTP Socket API supports two kinds of sockets:
o one-to-one style sockets (by using the socket type "SOCK_STREAM").
o one-to-many style socket (by using the socket type
"SOCK_SEQPACKET").
One-to-one style sockets are similar to TCP sockets; there is a 1:1
relationship between the sockets and the SCTP associations (except
for listening sockets). One-to-many style SCTP sockets are similar
to unconnected UDP sockets, where there is a 1:n relationship between
the sockets and the SCTP associations.
The SCTP stack can provide information to the applications about
state changes of the individual paths and the association whenever
they occur. These events are delivered similarly to user messages
but are specifically marked as notifications.
New functions have been introduced to support the use of multiple
local and remote addresses. Additional SCTP-specific send and
receive calls have been defined to permit SCTP-specific information
to be sent without using ancillary data in the form of additional
Control Message (cmsg) calls. These functions provide support for
detecting partial delivery of user messages and notifications.
The SCTP Socket API allows a fine-grained control of the protocol
behavior through an extensive set of socket options.
The SCTP kernel implementations of FreeBSD, Linux, and Solaris follow
mostly the specified extension to the BSD Sockets API for the base
protocol and the corresponding supported protocol extensions.
3.5.3. Transport Features
The transport features provided by SCTP are:
o connection-oriented transport with feature negotiation and
application-to-port mapping,
o unicast transport,
o port multiplexing,
o unidirectional or bidirectional communication,
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o message-oriented delivery with durable message framing supporting
multiple concurrent streams,
o fully reliable, partially reliable, or unreliable delivery (based
on user-specified policy to handle abandoned user messages) with
drop notification,
o ordered and unordered delivery within a stream,
o support for stream scheduling prioritization,
o segmentation,
o user message bundling,
o flow control using a window-based mechanism,
o congestion control using methods similar to TCP,
o strong error detection (CRC32c), and
o transport-layer multihoming for resilience and mobility.
3.6. Datagram Congestion Control Protocol (DCCP)
The Datagram Congestion Control Protocol (DCCP) [RFC4340] is an IETF
Standards Track bidirectional transport protocol that provides
unicast connections of congestion-controlled messages without
providing reliability.
The DCCP Problem Statement [RFC4336] describes the goals that DCCP
sought to address. It is suitable for applications that transfer
fairly large amounts of data and that can benefit from control over
the trade-off between timeliness and reliability [RFC4336].
DCCP offers low overhead, and many characteristics common to UDP, but
can avoid "re-inventing the wheel" each time a new multimedia
application emerges. Specifically, it includes core transport
functions (feature negotiation, path state management, RTT
calculation, PMTUD, etc.): DCCP applications select how they send
packets and, where suitable, choose common algorithms to manage their
functions. Examples of applications that can benefit from such
transport services include interactive applications, streaming media,
or on-line games [RFC4336].
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3.6.1. Protocol Description
DCCP is a connection-oriented datagram protocol that provides a
three-way handshake to allow a client and server to set up a
connection and provides mechanisms for orderly completion and
immediate teardown of a connection.
A DCCP protocol instance can be extended [RFC4340] and tuned using
additional features. Some features are sender-side only, requiring
no negotiation with the receiver; some are receiver-side only; and
some are explicitly negotiated during connection setup.
DCCP uses a Connect packet to initiate a session and permits each
endpoint to choose the features it wishes to support. Simultaneous
open [RFC5596], as in TCP, can enable interoperability in the
presence of middleboxes. The Connect packet includes a Service Code
[RFC5595] that identifies the application or protocol using DCCP,
providing middleboxes with information about the intended use of a
connection.
The DCCP service is unicast-only.
It provides multiplexing to multiple sockets at each endpoint using
port numbers. An active DCCP session is identified by its four-tuple
of local and remote IP addresses and local and remote port numbers.
The protocol segments data into messages that are typically sized to
fit in IP packets but may be fragmented if they are smaller than the
maximum packet size. A DCCP interface allows applications to request
fragmentation for packets larger than PMTU, but not larger than the
maximum packet size allowed by the current congestion control
mechanism (Congestion Control Maximum Packet Size (CCMPS)) [RFC4340].
Each message is identified by a sequence number. The sequence number
is used to identify segments in acknowledgments, to detect
unacknowledged segments, to measure RTT, etc. The protocol may
support unordered delivery of data and does not itself provide
retransmission. DCCP supports reduced checksum coverage, a partial
payload protection mechanism similar to UDP-Lite. There is also a
Data Checksum option, which when enabled, contains a strong Cyclic
Redundancy Check (CRC), to enable endpoints to detect application
data corruption.
Receiver flow control is supported, which limits the amount of
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
session [RFC4340], aborting the connection.
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DCCP supports negotiation of the congestion control profile between
endpoints, to provide plug-and-play congestion control mechanisms.
Examples of specified profiles include "TCP-like" [RFC4341], "TCP-
friendly" [RFC4342], and "TCP-friendly for small packets" [RFC5622].
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
[RFC6773] that permits DCCP to be used over paths where DCCP is not
natively supported. Support for DCCP in NAPT/NATs is defined in
[RFC4340] and [RFC5595]. Upper-layer protocols specified on top of
DCCP include DTLS [RFC5238], RTP [RFC5762], and Interactive
Connectivity Establishment / Session Description Protocol (ICE/SDP)
[RFC6773].
3.6.2. Interface Description
Functions expected for a DCCP API include: Open, Close, and
Management of the progress a DCCP connection. The Open function
provides feature negotiation, selection of an appropriate Congestion
Control Identifier (CCID) for congestion control, and other
parameters associated with the DCCP connection. A function allows an
application to send DCCP datagrams, including setting the required
checksum coverage and any required options. (DCCP permits sending
datagrams with a zero-length payload.) A function allows reception
of data, including indicating if the data was used or dropped.
Functions can also make the status of a connection visible to an
application, including detection of the maximum packet size and the
ability to perform flow control by detecting a slow receiver at the
sender.
There is no API currently specified in the RFC Series.
3.6.3. Transport Features
The transport features provided by DCCP are:
o unicast transport,
o connection-oriented communication with feature negotiation and
application-to-port mapping,
o signaling of application class for middlebox support (implemented
using Service Codes),
o port multiplexing,
o unidirectional or bidirectional communication,
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o message-oriented delivery,
o unreliable delivery with drop notification,
o unordered delivery,
o flow control (implemented using the slow receiver function), and
o partial and full payload error detection (with optional strong
integrity check).
3.7. Transport Layer Security (TLS) and Datagram TLS (DTLS) as a
Pseudotransport
Transport Layer Security (TLS) [RFC5246] and Datagram TLS (DTLS)
[RFC6347] are IETF protocols that provide several security-related
features to applications. TLS is designed to run on top of a
reliable streaming transport protocol (usually TCP), while DTLS is
designed to run on top of a best-effort datagram protocol (UDP or
DCCP [RFC5238]). At the time of writing, the current version of TLS
is 1.2, defined in [RFC5246]; work on TLS version is 1.3 [TLS-1.3]
nearing completion. DTLS provides nearly identical functionality to
applications; it is defined in [RFC6347] and its current version is
also 1.2. The TLS protocol evolved from the Secure Sockets Layer
(SSL) [RFC6101] protocols developed in the mid-1990s to support
protection of HTTP traffic.
While older versions of TLS and DTLS are still in use, they provide
weaker security guarantees. [RFC7457] outlines important attacks on
TLS and DTLS. [RFC7525] is a Best Current Practices (BCP) document
that describes secure configurations for TLS and DTLS to counter
these attacks. The recommendations are applicable for the vast
majority of use cases.
3.7.1. Protocol Description
Both TLS and DTLS provide the same security features and can thus be
discussed together. The features they provide are:
o Confidentiality
o Data integrity
o Peer authentication (optional)
o Perfect forward secrecy (optional)
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The authentication of the peer entity can be omitted; a common web
use case is where the server is authenticated and the client is not.
TLS also provides a completely anonymous operation mode in which
neither peer's identity is authenticated. It is important to note
that TLS itself does not specify how a peering entity's identity
should be interpreted. For example, in the common use case of
authentication by means of an X.509 certificate, it is the
application's decision whether the certificate of the peering entity
is acceptable for authorization decisions.
Perfect forward secrecy, if enabled and supported by the selected
algorithms, ensures that traffic encrypted and captured during a
session at time t0 cannot be later decrypted at time t1 (t1 > t0),
even if the long-term secrets of the communicating peers are later
compromised.
As DTLS is generally used over an unreliable datagram transport such
as UDP, applications will need to tolerate lost, reordered, or
duplicated datagrams. Like TLS, DTLS conveys application data in a
sequence of independent records. However, because records are mapped
to unreliable datagrams, there are several features unique to DTLS
that are not applicable to TLS:
o Record replay detection (optional).
o Record size negotiation (estimates of PMTU and record size
expansion factor).
o Conveyance of IP don't fragment (DF) bit settings by application.
o An anti-DoS stateless cookie mechanism (optional).
Generally, DTLS follows the TLS design as closely as possible. To
operate over datagrams, DTLS includes a sequence number and limited
forms of retransmission and fragmentation for its internal
operations. The sequence number may be used for detecting replayed
information, according to the windowing procedure described in
Section 4.1.2.6 of [RFC6347]. DTLS forbids the use of stream
ciphers, which are essentially incompatible when operating on
independent encrypted records.
3.7.2. Interface Description
TLS is commonly invoked using an API provided by packages such as
OpenSSL, wolfSSL, or GnuTLS. Using such APIs entails the
manipulation of several important abstractions, which fall into the
following categories: long-term keys and algorithms, session state,
and communications/connections.
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Considerable care is required in the use of TLS APIs to ensure
creation of a secure application. The programmer should have at
least a basic understanding of encryption and digital signature
algorithms and their strengths, public key infrastructure (including
X.509 certificates and certificate revocation), and the Sockets API.
See [RFC7525] and [RFC7457], as mentioned above.
As an example, in the case of OpenSSL, the primary abstractions are
the library itself, method (protocol), session, context, cipher, and
connection. After initializing the library and setting the method, a
cipher suite is chosen and used to configure a context object.
Session objects may then be minted according to the parameters
present in a context object and associated with individual
connections. Depending on how precisely the programmer wishes to
select different algorithmic or protocol options, various levels of
details may be required.
3.7.3. Transport Features
Both TLS and DTLS employ a layered architecture. The lower layer is
commonly called the "record protocol". It is responsible for:
o message fragmentation,
o authentication and integrity via message authentication codes
(MACs),
o data encryption, and
o scheduling transmission using the underlying transport protocol.
DTLS augments the TLS record protocol with:
o ordering and replay protection, implemented using sequence
numbers.
Several protocols are layered on top of the record protocol. These
include the handshake, alert, and change cipher spec protocols.
There is also the data protocol, used to carry application traffic.
The handshake protocol is used to establish cryptographic and
compression parameters when a connection is first set up. In DTLS,
this protocol also has a basic fragmentation and retransmission
capability and a cookie-like mechanism to resist DoS attacks. (TLS
compression is not recommended at present). The alert protocol is
used to inform the peer of various conditions, most of which are
terminal for the connection. The change cipher spec protocol is used
to synchronize changes in cryptographic parameters for each peer.
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The data protocol, when used with an appropriate cipher, provides:
o authentication of one end or both ends of a connection,
o confidentiality, and
o cryptographic integrity protection.
Both TLS and DTLS are unicast-only.
3.8. Real-Time Transport Protocol (RTP)
RTP provides an end-to-end network transport service, suitable for
applications transmitting real-time data, such as audio, video or
data, over multicast or unicast transport services, including TCP,
UDP, UDP-Lite, DCCP, TLS, and DTLS.
3.8.1. Protocol Description
The RTP standard [RFC3550] defines a pair of protocols: RTP and the
RTP Control Protocol (RTCP). The transport does not provide
connection setup, instead relying on out-of-band techniques or
associated control protocols to setup, negotiate parameters, or tear
down a session.
An RTP sender encapsulates audio/video data into RTP packets to
transport media streams. The RFC Series specifies RTP payload
formats that allow packets to carry a wide range of media and
specifies a wide range of multiplexing, error control, and other
support mechanisms.
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
single RTP packet.
An RTP receiver collects RTP packets from the network, validates them
for correctness, and sends them to the media decoder input queue.
Missing packet detection is performed by the channel decoder. The
playout buffer is ordered by time stamp and is used to reorder
packets. Damaged frames may be repaired before the media payloads
are decompressed to display or store the data. Some uses of RTP are
able to exploit the partial payload protection features offered by
DCCP and UDP-Lite.
RTCP is a control protocol that works alongside an RTP flow. Both
the RTP sender and receiver will send RTCP report packets. This is
used to periodically send control information and report performance.
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Based on received RTCP feedback, an RTP sender can adjust the
transmission, e.g., perform rate adaptation at the application layer
in the case of congestion.
An RTCP receiver report (RTCP RR) is returned to the sender
periodically to report key parameters (e.g., the fraction of packets
lost in the last reporting interval, the cumulative number of packets
lost, the highest sequence number received, and the inter-arrival
jitter). The RTCP RR packets also contain timing information that
allows the sender to estimate the network round-trip time (RTT) to
the receivers.
The interval between reports sent from each receiver tends to be on
the order of a few seconds on average, although this varies with the
session rate, and sub-second reporting intervals are possible for
high rate sessions. The interval is randomized to avoid
synchronization of reports from multiple receivers.
3.8.2. Interface Description
There is no standard API defined for RTP or RTCP. Implementations
are typically tightly integrated with a particular application and
closely follow the principles of application-level framing and
integrated layer processing [ClarkArch] in media processing
[RFC2736], error recovery and concealment, rate adaptation, and
security [RFC7202]. Accordingly, RTP implementations tend to be
targeted at particular application domains (e.g., voice-over-IP,
IPTV, or video conferencing), with a feature set optimized for that
domain, rather than being general purpose implementations of the
protocol.
3.8.3. Transport Features
The transport features provided by RTP are:
o unicast, multicast, or IPv4 broadcast (provided by lower-layer
protocol),
o port multiplexing (provided by lower-layer protocol),
o unidirectional or bidirectional communication (provided by lower-
layer protocol),
o message-oriented delivery with support for media types and other
extensions,
o reliable delivery when using erasure coding or unreliable delivery
with drop notification (if supported by lower-layer protocol),
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o connection setup with feature negotiation (using associated
protocols) and application-to-port mapping (provided by lower-
layer protocol),
o segmentation, and
o performance metric reporting (using associated protocols).
3.9. Hypertext Transport Protocol (HTTP) over TCP as a Pseudotransport
The Hypertext Transfer Protocol (HTTP) is an application-level
protocol widely used on the Internet. It provides object-oriented
delivery of discrete data or files. Version 1.1 of the protocol is
specified in [RFC7230] [RFC7231] [RFC7232] [RFC7233] [RFC7234]
[RFC7235], and version 2 is specified in [RFC7540]. HTTP is usually
transported over TCP using ports 80 and 443, although it can be used
with other transports. When used over TCP, it inherits TCP's
properties.
Application-layer protocols may use HTTP as a substrate with an
existing method and data formats, or specify new methods and data
formats. There are various reasons for this practice listed in
[RFC3205]; these include being a well-known and well-understood
protocol, reusability of existing servers and client libraries, easy
use of existing security mechanisms such as HTTP digest
authentication [RFC7235] and TLS [RFC5246], and the ability of HTTP
to traverse firewalls, which allows it to work over many types of
infrastructure and in cases where an application server often needs
to support HTTP anyway.
Depending on application need, the use of HTTP as a substrate
protocol may add complexity and overhead in comparison to a special-
purpose protocol (e.g., HTTP headers, suitability of the HTTP
security model, etc.). [RFC3205] addresses this issue, provides some
guidelines, and identifies concerns about the use of HTTP standard
ports 80 and 443, the use of the HTTP URL scheme, and interaction
with existing firewalls, proxies, and NATs.
Representational State Transfer (REST) [REST] is another example of
how applications can use HTTP as a transport protocol. REST is an
architecture style that may be used to build applications using HTTP
as a communication protocol.
3.9.1. Protocol Description
The Hypertext Transfer Protocol (HTTP) is a request/response
protocol. A client sends a request containing a request method, URI,
and protocol version followed by message whose design is inspired by
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MIME (see [RFC7231] for the differences between an HTTP object and a
MIME message), containing information about the client and request
modifiers. The message can also contain a message body carrying
application data. The server responds with a status or error code
followed by a message containing information about the server and
information about the data. This may include a message body. It is
possible to specify a data format for the message body using MIME
media types [RFC2045]. The protocol has additional features; some
relevant to pseudotransport are described below.
Content negotiation, specified in [RFC7231], is a mechanism provided
by HTTP to allow selection of a representation for a requested
resource. The client and server negotiate acceptable data formats,
character sets, and data encoding (e.g., data can be transferred
compressed using gzip). HTTP can accommodate exchange of messages as
well as data streaming (using chunked transfer encoding [RFC7230]).
It is also possible to request a part of a resource using an object
range request [RFC7233]. The protocol provides powerful cache
control signaling defined in [RFC7234].
The persistent connections of HTTP 1.1 and HTTP 2.0 allow multiple
request/response transactions (streams) during the lifetime of a
single HTTP connection. This reduces overhead during connection
establishment and mitigates transport-layer slow-start that would
have otherwise been incurred for each transaction. HTTP 2.0
connections can multiplex many request/response pairs in parallel on
a single transport connection. Both are important to reduce latency
for HTTP's primary use case.
HTTP can be combined with security mechanisms, such as TLS (denoted
by HTTPS). This adds protocol properties provided by such a
mechanism (e.g., authentication and encryption). The TLS
Application-Layer Protocol Negotiation (ALPN) extension [RFC7301] can
be used to negotiate the HTTP version within the TLS handshake,
eliminating the latency incurred by additional round-trip exchanges.
Arbitrary cookie strings, included as part of the request headers,
are often used as bearer tokens in HTTP.
3.9.2. Interface Description
There are many HTTP libraries available exposing different APIs. The
APIs provide a way to specify a request by providing a URI, a method,
request modifiers, and, optionally, a request body. For the
response, callbacks can be registered that will be invoked when the
response is received. If HTTPS is used, the API exposes a
registration of callbacks when a server requests client
authentication and when certificate verification is needed.
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The World Wide Web Consortium (W3C) has standardized the
XMLHttpRequest API [XHR]. This API can be used for sending HTTP/
HTTPS requests and receiving server responses. Besides the XML data
format, the request and response data format can also be JSON, HTML,
and plain text. JavaScript and XMLHttpRequest are ubiquitous
programming models for websites and more general applications where
native code is less attractive.
3.9.3. Transport Features
The transport features provided by HTTP, when used as a
pseudotransport, are:
o unicast transport (provided by the lower-layer protocol, usually
TCP),
o unidirectional or bidirectional communication,
o transfer of objects in multiple streams with object content type
negotiation, supporting partial transmission of object ranges,
o ordered delivery (provided by the lower-layer protocol, usually
TCP),
o fully reliable delivery (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
TCP).
HTTPS (HTTP over TLS) additionally provides the following features
(as provided by TLS):
o authentication (of one or both ends of a connection),
o confidentiality, and
o integrity protection.
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3.10. File Delivery over Unidirectional Transport / Asynchronous
Layered Coding (FLUTE/ALC) for Reliable Multicast
FLUTE/ALC is an IETF Standards Track protocol specified in [RFC6726]
and [RFC5775]. It provides object-oriented delivery of discrete data
or files. Asynchronous Layer Coding (ALC) provides an underlying
reliable transport service and FLUTE a file-oriented specialization
of the ALC service (e.g., to carry associated metadata). [RFC6726]
and [RFC5775] are non-backward-compatible updates of [RFC3926] and
[RFC3450], which are Experimental protocols; these Experimental
protocols are currently largely deployed in the 3GPP Multimedia
Broadcast / Multicast Service (MBMS) (see [MBMS], Section 7) and
similar contexts (e.g., the Japanese ISDB-Tmm standard).
The FLUTE/ALC protocol has been designed to support massively
scalable reliable bulk data dissemination to receiver groups of
arbitrary size using IP Multicast over any type of delivery network,
including unidirectional networks (e.g., broadcast wireless
channels). However, the FLUTE/ALC protocol also supports point-to-
point unicast transmissions.
FLUTE/ALC bulk data dissemination has been designed for discrete file
or memory-based "objects". Although FLUTE/ALC is not well adapted to
byte and message streaming, there is an exception: FLUTE/ALC is used
to carry 3GPP Dynamic Adaptive Streaming over HTTP (DASH) when
scalability is a requirement (see [MBMS], Section 5.6).
FLUTE/ALC's reliability, delivery mode, congestion control, and flow/
rate control mechanisms can be separately controlled to meet
different application needs. Section 4.1 of [RFC8085] describes
multicast congestion control requirements for UDP.
3.10.1. Protocol Description
The FLUTE/ALC protocol works on top of UDP (though it could work on
top of any datagram delivery transport protocol), without requiring
any connectivity from receivers to the sender. Purely unidirectional
networks are therefore supported by FLUTE/ALC. This guarantees
scalability to an unlimited number of receivers in a session, since
the sender behaves exactly the same regardless of the number of
receivers.
FLUTE/ALC supports the transfer of bulk objects such as file or
in-memory content, using either a push or an on-demand mode. In push
mode, content is sent once to the receivers, while in on-demand mode,
content is sent continuously during periods of time that can greatly
exceed the average time required to download the session objects (see
[RFC5651], Section 4.2).
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This enables receivers to join a session asynchronously, at their own
discretion, receive the content, and leave the session. In this
case, data content is typically sent continuously, in loops (also
known as "carousels"). FLUTE/ALC also supports the transfer of an
object stream, with loose real-time constraints. This is
particularly useful to carry 3GPP DASH when scalability is a
requirement and unicast transmissions over HTTP cannot be used
([MBMS], Section 5.6). In this case, packets are sent in sequence
using push mode. FLUTE/ALC is not well adapted to byte and message
streaming, and other solutions could be preferred (e.g., FECFRAME
[RFC6363] with real-time flows).
The FLUTE file delivery instantiation of ALC provides a metadata
delivery service. Each object of the FLUTE/ALC session is described
in a dedicated entry of a File Delivery Table (FDT), using an XML
format (see [RFC6726], Section 3.2). This metadata can include, but
is not restricted to, a URI attribute (to identify and locate the
object), a media type attribute, a size attribute, an encoding
attribute, or a message digest attribute. Since the set of objects
sent within a session can be dynamic, with new objects being added
and old ones removed, several instances of the FDT can be sent, and a
mechanism is provided to identify a new FDT instance.
Error detection and verification of the protocol control information
relies on the underlying transport (e.g., UDP checksum).
To provide robustness against packet loss and improve the efficiency
of the on-demand mode, FLUTE/ALC relies on packet erasure coding
(Application-Layer Forward Error Correction (AL-FEC)). AL-FEC
encoding is proactive (since there is no feedback and therefore no
(N)ACK-based retransmission), and ALC packets containing repair data
are sent along with ALC packets containing source data. Several FEC
Schemes have been standardized; FLUTE/ALC does not mandate the use of
any particular one. Several strategies concerning the transmission
order of ALC source and repair packets are possible, in particular,
in on-demand mode where it can deeply impact the service provided
(e.g., to favor the recovery of objects in sequence or, at the other
extreme, to favor the recovery of all objects in parallel), and
FLUTE/ALC does not mandate nor recommend the use of any particular
one.
A FLUTE/ALC session is composed of one or more channels, associated
to different destination unicast and/or multicast IP addresses. ALC
packets are sent in those channels at a certain transmission rate,
with a rate that often differs depending on the channel. FLUTE/ALC
does not mandate nor recommend any strategy to select which ALC
packet to send on which channel. FLUTE/ALC can use a multiple rate
congestion control building block (e.g., Wave and Equation Based Rate
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Control (WEBRC)) to provide congestion control that is feedback free,
where receivers adjust their reception rates individually by joining
and leaving channels associated with the session. To that purpose,
the ALC header provides a specific field to carry congestion-control-
specific information. However, FLUTE/ALC does not mandate the use of
a particular congestion control mechanism although WEBRC is mandatory
to support for the Internet ([RFC6726], Section 1.1.4). FLUTE/ALC is
often used over a network path with pre-provisioned capacity
[RFC8085] where there are no flows competing for capacity. In this
case, a sender-based rate control mechanism and a single channel are
sufficient.
[RFC6584] provides per-packet authentication, integrity, and anti-
replay protection in the context of the ALC and NORM protocols.
Several mechanisms are proposed that seamlessly integrate into these
protocols using the ALC and NORM header extension mechanisms.
3.10.2. Interface Description
The FLUTE/ALC specification does not describe a specific API to
control protocol operation. Although open source and commercial
implementations have specified APIs, there is no IETF-specified API
for FLUTE/ALC.
3.10.3. Transport Features
The transport features provided by FLUTE/ALC are:
o unicast, multicast, anycast, or IPv4 broadcast transmission,
o object-oriented delivery of discrete data or files and associated
metadata,
o fully reliable or partially reliable delivery (of file or in-
memory objects), using proactive packet erasure coding (AL-FEC) to
recover from packet erasures,
o ordered or unordered delivery (of file or in-memory objects),
o error detection (based on the UDP checksum),
o per-packet authentication,
o per-packet integrity,
o per-packet replay protection, and
o congestion control for layered flows (e.g., with WEBRC).
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3.11. NACK-Oriented Reliable Multicast (NORM)
NORM is an IETF Standards Track protocol specified in [RFC5740]. It
provides object-oriented delivery of discrete data or files.
The protocol was designed to support reliable bulk data dissemination
to receiver groups using IP Multicast but also provides for point-to-
point unicast operation. Support for bulk data dissemination
includes discrete file or computer memory-based "objects" as well as
byte and message streaming.
NORM can incorporate packet erasure coding as a part of its selective
Automatic Repeat reQuest (ARQ) in response to negative
acknowledgments from the receiver. The packet erasure coding can
also be proactively applied for forward protection from packet loss.
NORM transmissions are governed by TCP-Friendly Multicast Congestion
Control (TFMCC) [RFC4654]. The reliability, congestion control, and
flow control mechanisms can be separately controlled to meet
different application needs.
3.11.1. Protocol Description
The NORM protocol is encapsulated in UDP datagrams and thus provides
multiplexing for multiple sockets on hosts using port numbers. For
loosely coordinated IP Multicast, NORM is not strictly connection-
oriented although per-sender state is maintained by receivers for
protocol operation. [RFC5740] does not specify a handshake protocol
for connection establishment. Separate session initiation can be
used to coordinate port numbers. However, in-band "client-server"
style connection establishment can be accomplished with the NORM
congestion control signaling messages using port binding techniques
like those for TCP client-server connections.
NORM supports bulk "objects" such as file or in-memory content but
also can treat a stream of data as a logical bulk object for purposes
of packet erasure coding. In the case of stream transport, NORM can
support either byte streams or message streams where application-
defined message boundary information is carried in the NORM protocol
messages. This allows the receiver(s) to join/rejoin and recover
message boundaries mid-stream as needed. Application content is
carried and identified by the NORM protocol with encoding symbol
identifiers depending upon the Forward Error Correction (FEC) Scheme
[RFC5052] configured. NORM uses NACK-based selective ARQ to reliably
deliver the application content to the receiver(s). NORM proactively
measures round-trip timing information to scale ARQ timers
appropriately and to support congestion control. For multicast
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operation, timer-based feedback suppression is used to achieve group
size scaling with low feedback traffic levels. The feedback
suppression is not applied for unicast operation.
NORM uses rate-based congestion control based upon the TCP-Friendly
Rate Control (TFRC) [RFC5348] principles that are also used in DCCP
[RFC4340]. NORM uses control messages to measure RTT and collect
congestion event information (e.g., reflecting a loss event or ECN
event) from the receiver(s) to support dynamic adjustment or the
rate. TCP-Friendly Multicast Congestion Control (TFMCC) [RFC4654]
provides extra features to support multicast but is functionally
equivalent to TFRC for unicast.
Error detection and verification of the protocol control information
relies on the on the underlying transport (e.g., UDP checksum).
The reliability mechanism is decoupled from congestion control. This
allows invocation of alternative arrangements of transport services,
for example, to support, fixed-rate reliable delivery or unreliable
delivery (that may optionally be "better than best effort" via packet
erasure coding) using TFRC. Alternative congestion control
techniques may be applied, for example, TFRC with congestion event
detection based on ECN.
While NORM provides NACK-based reliability, it also supports a
positive acknowledgment (ACK) mechanism that can be used for receiver
flow control. This mechanism is decoupled from the reliability and
congestion control, supporting applications with different needs.
One example is use of NORM for quasi-reliable delivery, where timely
delivery of newer content may be favored over completely reliable
delivery of older content within buffering and RTT constraints.
3.11.2. Interface Description
The NORM specification does not describe a specific API to control
protocol operation. A freely available, open-source reference
implementation of NORM is available at
<
https://www.nrl.navy.mil/itd/ncs/products/norm>, and a documented
API is provided for this implementation. While a sockets-like API is
not currently documented, the existing API supports the necessary
functions for that to be implemented.
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3.11.3. Transport Features
The transport features provided by NORM are:
o unicast or multicast transport,
o unidirectional communication,
o stream-oriented delivery in a single stream or object-oriented
delivery of in-memory data or file bulk content objects,
o fully reliable (NACK-based) or partially reliable (using erasure
coding both proactively and as part of ARQ) delivery,
o unordered delivery,
o error detection (relies on UDP checksum),
o segmentation,
o data bundling (using Nagle's algorithm),
o flow control (timer-based and/or ACK-based), and
o congestion control (also supporting fixed-rate reliable or
unreliable delivery).
3.12. Internet Control Message Protocol (ICMP)
The Internet Control Message Protocol (ICMP) [RFC792] for IPv4 and
ICMP for IPv6 [RFC4443] are IETF Standards Track protocols. It is a
connectionless unidirectional protocol that delivers individual
messages, without error correction, congestion control, or flow
control. Messages may be sent as unicast, IPv4 broadcast, or
multicast datagrams (IPv4 and IPv6), in addition to anycast
datagrams.
While ICMP is not typically described as a transport protocol, it
does position itself over the network layer, and the operation of
other transport protocols can be closely linked to the functions
provided by ICMP.
Transport protocols and upper-layer protocols can use received ICMP
messages to help them make appropriate decisions when network or
endpoint errors are reported, for example, to implement ICMP-based
Path MTU Discovery (PMTUD) [RFC1191] [RFC1981] or assist in
Packetization Layer PMTUD (PLPMTUD) [RFC4821]. Such reactions to
received messages need to protect from off-path data injection
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[RFC8085] to avoid an application receiving packets created by an
unauthorized third party. An application therefore needs to ensure
that all messages are appropriately validated by checking the payload
of the messages to ensure they are received in response to actually
transmitted traffic (e.g., a reported error condition that
corresponds to a UDP datagram or TCP segment was actually sent by the
application). This requires context [RFC6056], such as local state
about communication instances to each destination (e.g., in TCP,
DCCP, or SCTP). This state is not always maintained by UDP-based
applications [RFC8085].
3.12.1. Protocol Description
ICMP is a connectionless unidirectional protocol. It delivers
independent messages, called "datagrams". Each message is required
to carry a checksum as an integrity check and to protect from
misdelivery to an unintended endpoint.
ICMP messages typically relay diagnostic information from an endpoint
[RFC1122] or network device [RFC1812] addressed to the sender of a
flow. This usually contains the network protocol header of a packet
that encountered a reported issue. Some formats of messages can also
carry other payload data. Each message carries an integrity check
calculated in the same way as for UDP; this checksum is not optional.
The RFC Series defines additional IPv6 message formats to support a
range of uses. In the case of IPv6, the protocol incorporates
neighbor discovery [RFC4861] [RFC3971] (provided by ARP for IPv4) and
Multicast Listener Discovery (MLD) [RFC2710] group management
functions (provided by IGMP for IPv4).
Reliable transmission cannot be assumed. A receiving application
that is unable to run sufficiently fast, or frequently, may miss
messages since there is no flow or congestion control. In addition,
some network devices rate-limit ICMP messages.
3.12.2. Interface Description
ICMP processing is integrated in many connection-oriented transports
but, like other functions, needs to be provided by an upper-layer
protocol when using UDP and UDP-Lite.
On some stacks, a bound socket also allows a UDP application to be
notified when ICMP error messages are received for its transmissions
[RFC8085].
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Any response to ICMP error messages ought to be robust to temporary
routing failures (sometimes called "soft errors"), e.g., transient
ICMP "unreachable" messages ought to not normally cause a
communication abort [RFC5461] [RFC8085].
3.12.3. Transport Features
ICMP does not provide any transport service directly to applications.
Used together with other transport protocols, it provides
transmission of control, error, and measurement data between
endpoints or from devices along the path to one endpoint.
4. Congestion Control
Congestion control is critical to the stable operation of the
Internet. A variety of mechanisms are used to provide the congestion
control needed by many Internet transport protocols. Congestion is
detected based on sensing of network conditions, whether through
explicit or implicit feedback. The congestion control mechanisms
that can be applied by different transport protocols are largely
orthogonal to the choice of transport protocol. This section
provides an overview of the congestion control mechanisms available
to the protocols described in Section 3.
Many protocols use a separate window to determine the maximum sending
rate that is allowed by the congestion control. The used congestion
control mechanism will increase the congestion window if feedback is
received that indicates that the currently used network path is not
congested and will reduce the window otherwise. Window-based
mechanisms often increase their window slowing over multiple RTTs,
while decreasing strongly when the first indication of congestion is
received. One example is an Additive Increase Multiplicative
Decrease (AIMD) scheme, where the window is increased by a certain
number of packets/bytes for each data segment that has been
successfully transmitted, while the window decreases multiplicatively
on the occurrence of a congestion event. This can lead to a rather
unstable, oscillating sending rate but will resolve a congestion
situation quickly. Examples of window-based AIMD schemes include TCP
NewReno [RFC5681], TCP Cubic [CUBIC] (the default mechanism for TCP
in Linux), and CCID 2 specified for DCCP [RFC4341].
Some classes of applications prefer to use a transport service that
allows sending at a more stable rate that is slowly varied in
response to congestion. Rate-based methods offer this type of
congestion control and have been defined based on the loss ratio and
observed round-trip time, such as TFRC [RFC5348] and TFRC-SP
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[RFC4828]. These methods utilize a throughput equation to determine
the maximum acceptable rate. Such methods are used with DCCP CCID 3
[RFC4342], CCID 4 [RFC5622], WEBRC [RFC3738], and other applications.
Another class of applications prefers a transport service that yields
to other (higher-priority) traffic, such as interactive
transmissions. While most traffic in the Internet uses loss-based
congestion control and therefore tends to fill the network buffers
(to a certain level if Active Queue Management (AQM) is used), low-
priority congestion control methods often react to changes in delay
as an earlier indication of congestion. This approach tends to
induce less loss than a loss-based method but does generally not
compete well with loss-based traffic across shared bottleneck links.
Therefore, methods such as LEDBAT [RFC6817] are deployed in the
Internet for scavenger traffic that aims to only utilize otherwise
unused capacity.
5. Transport Features
The transport protocol features described in this document can be
used as a basis for defining common transport features. These are
listed below with the protocols supporting them:
o Control Functions
* Addressing
+ unicast (TCP, MPTCP, UDP, UDP-Lite, SCTP, DCCP, TLS, RTP,
HTTP, ICMP)
+ multicast (UDP, UDP-Lite, RTP, ICMP, FLUTE/ALC, NORM). Note
that, as TLS and DTLS are unicast-only, there is no widely
deployed mechanism for supporting the features listed under
the Security bullet (below) when using multicast addressing.
+ IPv4 broadcast (UDP, UDP-Lite, ICMP)
+ anycast (UDP, UDP-Lite). Connection-oriented protocols such
as TCP and DCCP have also been deployed using anycast
addressing, with the risk that routing changes may cause
connection failure.
* Association type
+ connection-oriented (TCP, MPTCP, DCCP, SCTP, TLS, RTP, HTTP,
NORM)
+ connectionless (UDP, UDP-Lite, FLUTE/ALC)
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* Multihoming support
+ resilience and mobility (MPTCP, SCTP)
+ load balancing (MPTCP)
+ address family multiplexing (MPTCP, SCTP)
* Middlebox cooperation
+ application-class signaling to middleboxes (DCCP)
+ error condition signaling from middleboxes and routers to
endpoints (ICMP)
* Signaling
+ control information and error signaling (ICMP)
+ application performance reporting (RTP)
o Delivery
* Reliability
+ fully reliable delivery (TCP, MPTCP, SCTP, TLS, HTTP, FLUTE/
ALC, NORM)
+ partially reliable delivery (SCTP, NORM)
- using packet erasure coding (RTP, FLUTE/ALC, NORM)
- with specified policy for dropped messages (SCTP)
+ unreliable delivery (SCTP, UDP, UDP-Lite, DCCP, RTP)
- with drop notification to sender (SCTP, DCCP, RTP)
+ error detection
- checksum for error detection (TCP, MPTCP, UDP, UDP-Lite,
SCTP, DCCP, TLS, DTLS, FLUTE/ALC, NORM, ICMP)
- partial payload checksum protection (UDP-Lite, DCCP).
Some uses of RTP can exploit partial payload checksum
protection feature to provide a corruption-tolerant
transport service.
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- checksum optional (UDP). Possible with IPv4 and, in
certain cases, with IPv6.
* Ordering
+ ordered delivery (TCP, MPTCP, SCTP, TLS, RTP, HTTP, FLUTE)
+ unordered delivery permitted (UDP, UDP-Lite, SCTP, DCCP,
RTP, NORM)
* Type/framing
+ stream-oriented delivery (TCP, MPTCP, SCTP, TLS, HTTP)
- with multiple streams per association (SCTP, HTTP2)
+ message-oriented delivery (UDP, UDP-Lite, SCTP, DCCP, DTLS,
RTP)
+ object-oriented delivery of discrete data or files and
associated metadata (HTTP, FLUTE/ALC, NORM)
- with partial delivery of object ranges (HTTP)
* Directionality
+ unidirectional (UDP, UDP-Lite, DCCP, RTP, FLUTE/ALC, NORM)
+ bidirectional (TCP, MPTCP, SCTP, TLS, HTTP)
o Transmission control
* flow control (TCP, MPTCP, SCTP, DCCP, TLS, RTP, HTTP)
* congestion control (TCP, MPTCP, SCTP, DCCP, RTP, FLUTE/ALC,
NORM). Congestion control can also provided by the transport
supporting an upper-layer transport (e.g., TLS, RTP, HTTP).
* segmentation (TCP, MPTCP, SCTP, TLS, RTP, HTTP, FLUTE/ALC,
NORM)
* data/message bundling (TCP, MPTCP, SCTP, TLS, HTTP)
* stream scheduling prioritization (SCTP, HTTP2)
* endpoint multiplexing (MPTCP)
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o Security
* authentication of one end of a connection (TLS, DTLS, FLUTE/
ALC)
* authentication of both ends of a connection (TLS, DTLS)
* confidentiality (TLS, DTLS)
* cryptographic integrity protection (TLS, DTLS)
* replay protection (TLS, DTLS, FLUTE/ALC)
6. IANA Considerations
This document does not require any IANA actions.
7. Security Considerations
This document surveys existing transport protocols and protocols
providing transport-like services. Confidentiality, integrity, and
authenticity are among the features provided by those services. This
document does not specify any new features or mechanisms for
providing these features. Each RFC referenced by this document
discusses the security considerations of the specification it
contains.
8. Informative References
[ClarkArch]
Clark, D. and D. Tennenhouse, "Architectural
Considerations for a New Generation of Protocols",
Proceedings of ACM SIGCOMM, DOI 10.1145/99517.99553, 1990.
[CUBIC] Rhee, I., Xu, L., Ha, S., Zimmermann, A., Eggert, L., and
R. Scheffenegger, "CUBIC for Fast Long-Distance Networks",
Work in Progress, draft-ietf-tcpm-cubic-04, February 2017.
[MBMS] 3GPP, "Multimedia Broadcast/Multicast Service (MBMS);
Protocols and codecs", 3GPP TS 26.346, 2015,
<
http://www.3gpp.org/DynaReport/26346.htm>.
[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.
Fairhurst, et al. Informational [Page 42]
RFC 8095 Transport Services March 2017
[POSIX] IEEE, "Standard for Information Technology -- Portable
Operating System Interface (POSIX(R)) Base Specifications,
Issue 7", IEEE 1003.1, DOI 10.1109/ieeestd.2016.7582338,
<
http://ieeexplore.ieee.org/document/7582338/>.
[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.
[RFC768] Postel, J., "User Datagram Protocol", STD 6, RFC 768,
DOI 10.17487/RFC0768, August 1980,
<
http://www.rfc-editor.org/info/rfc768>.
[RFC792] Postel, J., "Internet Control Message Protocol", STD 5,
RFC 792, DOI 10.17487/RFC0792, September 1981,
<
http://www.rfc-editor.org/info/rfc792>.
[RFC793] Postel, J., "Transmission Control Protocol", STD 7,
RFC 793, DOI 10.17487/RFC0793, September 1981,
<
http://www.rfc-editor.org/info/rfc793>.
[RFC1071] Braden, R., Borman, D., and C. Partridge, "Computing the
Internet checksum", RFC 1071, DOI 10.17487/RFC1071,
September 1988, <
http://www.rfc-editor.org/info/rfc1071>.
[RFC1122] Braden, R., Ed., "Requirements for Internet Hosts -
Communication Layers", STD 3, RFC 1122,
DOI 10.17487/RFC1122, October 1989,
<
http://www.rfc-editor.org/info/rfc1122>.
[RFC1191] Mogul, J. and S. Deering, "Path MTU discovery", RFC 1191,
DOI 10.17487/RFC1191, November 1990,
<
http://www.rfc-editor.org/info/rfc1191>.
[RFC1812] Baker, F., Ed., "Requirements for IP Version 4 Routers",
RFC 1812, DOI 10.17487/RFC1812, June 1995,
<
http://www.rfc-editor.org/info/rfc1812>.
[RFC1981] McCann, J., Deering, S., and J. Mogul, "Path MTU Discovery
for IP version 6", RFC 1981, DOI 10.17487/RFC1981, August
1996, <
http://www.rfc-editor.org/info/rfc1981>.
[RFC2018] Mathis, M., Mahdavi, J., Floyd, S., and A. Romanow, "TCP
Selective Acknowledgment Options", RFC 2018,
DOI 10.17487/RFC2018, October 1996,
<
http://www.rfc-editor.org/info/rfc2018>.
Fairhurst, et al. Informational [Page 43]
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[RFC2045] Freed, N. and N. Borenstein, "Multipurpose Internet Mail
Extensions (MIME) Part One: Format of Internet Message
Bodies", RFC 2045, DOI 10.17487/RFC2045, November 1996,
<
http://www.rfc-editor.org/info/rfc2045>.
[RFC2460] Deering, S. and R. Hinden, "Internet Protocol, Version 6
(IPv6) Specification", RFC 2460, DOI 10.17487/RFC2460,
December 1998, <
http://www.rfc-editor.org/info/rfc2460>.
[RFC2710] Deering, S., Fenner, W., and B. Haberman, "Multicast
Listener Discovery (MLD) for IPv6", RFC 2710,
DOI 10.17487/RFC2710, October 1999,
<
http://www.rfc-editor.org/info/rfc2710>.
[RFC2736] Handley, M. and C. Perkins, "Guidelines for Writers of RTP
Payload Format Specifications", BCP 36, RFC 2736,
DOI 10.17487/RFC2736, December 1999,
<
http://www.rfc-editor.org/info/rfc2736>.
[RFC3168] Ramakrishnan, K., Floyd, S., and D. Black, "The Addition
of Explicit Congestion Notification (ECN) to IP",
RFC 3168, DOI 10.17487/RFC3168, September 2001,
<
http://www.rfc-editor.org/info/rfc3168>.
[RFC3205] Moore, K., "On the use of HTTP as a Substrate", BCP 56,
RFC 3205, DOI 10.17487/RFC3205, February 2002,
<
http://www.rfc-editor.org/info/rfc3205>.
[RFC3260] Grossman, D., "New Terminology and Clarifications for
Diffserv", RFC 3260, DOI 10.17487/RFC3260, April 2002,
<
http://www.rfc-editor.org/info/rfc3260>.
[RFC3436] Jungmaier, A., Rescorla, E., and M. Tuexen, "Transport
Layer Security over Stream Control Transmission Protocol",
RFC 3436, DOI 10.17487/RFC3436, December 2002,
<
http://www.rfc-editor.org/info/rfc3436>.
[RFC3450] Luby, M., Gemmell, J., Vicisano, L., Rizzo, L., and J.
Crowcroft, "Asynchronous Layered Coding (ALC) Protocol
Instantiation", RFC 3450, DOI 10.17487/RFC3450, December
2002, <
http://www.rfc-editor.org/info/rfc3450>.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
July 2003, <
http://www.rfc-editor.org/info/rfc3550>.
Fairhurst, et al. Informational [Page 44]
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[RFC3738] Luby, M. and V. Goyal, "Wave and Equation Based Rate
Control (WEBRC) Building Block", RFC 3738,
DOI 10.17487/RFC3738, April 2004,
<
http://www.rfc-editor.org/info/rfc3738>.
[RFC3758] Stewart, R., Ramalho, M., Xie, Q., Tuexen, M., and P.
Conrad, "Stream Control Transmission Protocol (SCTP)
Partial Reliability Extension", RFC 3758,
DOI 10.17487/RFC3758, May 2004,
<
http://www.rfc-editor.org/info/rfc3758>.
[RFC3828] Larzon, L-A., Degermark, M., Pink, S., Jonsson, L-E., Ed.,
and G. Fairhurst, Ed., "The Lightweight User Datagram
Protocol (UDP-Lite)", RFC 3828, DOI 10.17487/RFC3828, July
2004, <
http://www.rfc-editor.org/info/rfc3828>.
[RFC3926] Paila, T., Luby, M., Lehtonen, R., Roca, V., and R. Walsh,
"FLUTE - File Delivery over Unidirectional Transport",
RFC 3926, DOI 10.17487/RFC3926, October 2004,
<
http://www.rfc-editor.org/info/rfc3926>.
[RFC3971] Arkko, J., Ed., Kempf, J., Zill, B., and P. Nikander,
"SEcure Neighbor Discovery (SEND)", RFC 3971,
DOI 10.17487/RFC3971, March 2005,
<
http://www.rfc-editor.org/info/rfc3971>.
[RFC4336] Floyd, S., Handley, M., and E. Kohler, "Problem Statement
for the Datagram Congestion Control Protocol (DCCP)",
RFC 4336, DOI 10.17487/RFC4336, March 2006,
<
http://www.rfc-editor.org/info/rfc4336>.
[RFC4340] Kohler, E., Handley, M., and S. Floyd, "Datagram
Congestion Control Protocol (DCCP)", RFC 4340,
DOI 10.17487/RFC4340, March 2006,
<
http://www.rfc-editor.org/info/rfc4340>.
[RFC4341] Floyd, S. and E. Kohler, "Profile for Datagram Congestion
Control Protocol (DCCP) Congestion Control ID 2: TCP-like
Congestion Control", RFC 4341, DOI 10.17487/RFC4341, March
2006, <
http://www.rfc-editor.org/info/rfc4341>.
[RFC4342] Floyd, S., Kohler, E., and J. Padhye, "Profile for
Datagram Congestion Control Protocol (DCCP) Congestion
Control ID 3: TCP-Friendly Rate Control (TFRC)", RFC 4342,
DOI 10.17487/RFC4342, March 2006,
<
http://www.rfc-editor.org/info/rfc4342>.
Fairhurst, et al. Informational [Page 45]
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[RFC4443] Conta, A., Deering, S., and M. Gupta, Ed., "Internet
Control Message Protocol (ICMPv6) for the Internet
Protocol Version 6 (IPv6) Specification", RFC 4443,
DOI 10.17487/RFC4443, March 2006,
<
http://www.rfc-editor.org/info/rfc4443>.
[RFC4654] Widmer, J. and M. Handley, "TCP-Friendly Multicast
Congestion Control (TFMCC): Protocol Specification",
RFC 4654, DOI 10.17487/RFC4654, August 2006,
<
http://www.rfc-editor.org/info/rfc4654>.
[RFC4820] Tuexen, M., Stewart, R., and P. Lei, "Padding Chunk and
Parameter for the Stream Control Transmission Protocol
(SCTP)", RFC 4820, DOI 10.17487/RFC4820, March 2007,
<
http://www.rfc-editor.org/info/rfc4820>.
[RFC4821] Mathis, M. and J. Heffner, "Packetization Layer Path MTU
Discovery", RFC 4821, DOI 10.17487/RFC4821, March 2007,
<
http://www.rfc-editor.org/info/rfc4821>.
[RFC4828] Floyd, S. and E. Kohler, "TCP Friendly Rate Control
(TFRC): The Small-Packet (SP) Variant", RFC 4828,
DOI 10.17487/RFC4828, April 2007,
<
http://www.rfc-editor.org/info/rfc4828>.
[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,
"Authenticated Chunks for the Stream Control Transmission
Protocol (SCTP)", RFC 4895, DOI 10.17487/RFC4895, August
2007, <
http://www.rfc-editor.org/info/rfc4895>.
[RFC4960] Stewart, R., Ed., "Stream Control Transmission Protocol",
RFC 4960, DOI 10.17487/RFC4960, September 2007,
<
http://www.rfc-editor.org/info/rfc4960>.
[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>.
Fairhurst, et al. Informational [Page 46]
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[RFC5061] Stewart, R., Xie, Q., Tuexen, M., Maruyama, S., and M.
Kozuka, "Stream Control Transmission Protocol (SCTP)
Dynamic Address Reconfiguration", RFC 5061,
DOI 10.17487/RFC5061, September 2007,
<
http://www.rfc-editor.org/info/rfc5061>.
[RFC5097] Renker, G. and G. Fairhurst, "MIB for the UDP-Lite
protocol", RFC 5097, DOI 10.17487/RFC5097, January 2008,
<
http://www.rfc-editor.org/info/rfc5097>.
[RFC5238] Phelan, T., "Datagram Transport Layer Security (DTLS) over
the Datagram Congestion Control Protocol (DCCP)",
RFC 5238, DOI 10.17487/RFC5238, May 2008,
<
http://www.rfc-editor.org/info/rfc5238>.
[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
Friendly Rate Control (TFRC): Protocol Specification",
RFC 5348, DOI 10.17487/RFC5348, September 2008,
<
http://www.rfc-editor.org/info/rfc5348>.
[RFC5461] Gont, F., "TCP's Reaction to Soft Errors", RFC 5461,
DOI 10.17487/RFC5461, February 2009,
<
http://www.rfc-editor.org/info/rfc5461>.
[RFC5595] Fairhurst, G., "The Datagram Congestion Control Protocol
(DCCP) Service Codes", RFC 5595, DOI 10.17487/RFC5595,
September 2009, <
http://www.rfc-editor.org/info/rfc5595>.
[RFC5596] Fairhurst, G., "Datagram Congestion Control Protocol
(DCCP) Simultaneous-Open Technique to Facilitate NAT/
Middlebox Traversal", RFC 5596, DOI 10.17487/RFC5596,
September 2009, <
http://www.rfc-editor.org/info/rfc5596>.
[RFC5622] Floyd, S. and E. Kohler, "Profile for Datagram Congestion
Control Protocol (DCCP) Congestion ID 4: TCP-Friendly Rate
Control for Small Packets (TFRC-SP)", RFC 5622,
DOI 10.17487/RFC5622, August 2009,
<
http://www.rfc-editor.org/info/rfc5622>.
[RFC5651] Luby, M., Watson, M., and L. Vicisano, "Layered Coding
Transport (LCT) Building Block", RFC 5651,
DOI 10.17487/RFC5651, October 2009,
<
http://www.rfc-editor.org/info/rfc5651>.
Fairhurst, et al. Informational [Page 47]
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[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>.
[RFC5740] Adamson, B., Bormann, C., Handley, M., and J. Macker,
"NACK-Oriented Reliable Multicast (NORM) Transport
Protocol", RFC 5740, DOI 10.17487/RFC5740, November 2009,
<
http://www.rfc-editor.org/info/rfc5740>.
[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
Layered Coding (ALC) Protocol Instantiation", RFC 5775,
DOI 10.17487/RFC5775, April 2010,
<
http://www.rfc-editor.org/info/rfc5775>.
[RFC6056] Larsen, M. and F. Gont, "Recommendations for Transport-
Protocol Port Randomization", BCP 156, RFC 6056,
DOI 10.17487/RFC6056, January 2011,
<
http://www.rfc-editor.org/info/rfc6056>.
[RFC6083] Tuexen, M., Seggelmann, R., and E. Rescorla, "Datagram
Transport Layer Security (DTLS) for Stream Control
Transmission Protocol (SCTP)", RFC 6083,
DOI 10.17487/RFC6083, January 2011,
<
http://www.rfc-editor.org/info/rfc6083>.
[RFC6093] Gont, F. and A. Yourtchenko, "On the Implementation of the
TCP Urgent Mechanism", RFC 6093, DOI 10.17487/RFC6093,
January 2011, <
http://www.rfc-editor.org/info/rfc6093>.
[RFC6101] Freier, A., Karlton, P., and P. Kocher, "The Secure
Sockets Layer (SSL) Protocol Version 3.0", RFC 6101,
DOI 10.17487/RFC6101, August 2011,
<
http://www.rfc-editor.org/info/rfc6101>.
[RFC6347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer
Security Version 1.2", RFC 6347, DOI 10.17487/RFC6347,
January 2012, <
http://www.rfc-editor.org/info/rfc6347>.
[RFC6356] Raiciu, C., Handley, M., and D. Wischik, "Coupled
Congestion Control for Multipath Transport Protocols",
RFC 6356, DOI 10.17487/RFC6356, October 2011,
<
http://www.rfc-editor.org/info/rfc6356>.
Fairhurst, et al. Informational [Page 48]
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[RFC6363] Watson, M., Begen, A., and V. Roca, "Forward Error
Correction (FEC) Framework", RFC 6363,
DOI 10.17487/RFC6363, October 2011,
<
http://www.rfc-editor.org/info/rfc6363>.
[RFC6458] Stewart, R., Tuexen, M., Poon, K., Lei, P., and V.
Yasevich, "Sockets API Extensions for the Stream Control
Transmission Protocol (SCTP)", RFC 6458,
DOI 10.17487/RFC6458, December 2011,
<
http://www.rfc-editor.org/info/rfc6458>.
[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
NewReno Modification to TCP's Fast Recovery Algorithm",
RFC 6582, DOI 10.17487/RFC6582, April 2012,
<
http://www.rfc-editor.org/info/rfc6582>.
[RFC6584] Roca, V., "Simple Authentication Schemes for the
Asynchronous Layered Coding (ALC) and NACK-Oriented
Reliable Multicast (NORM) Protocols", RFC 6584,
DOI 10.17487/RFC6584, April 2012,
<
http://www.rfc-editor.org/info/rfc6584>.
[RFC6726] Paila, T., Walsh, R., Luby, M., Roca, V., and R. Lehtonen,
"FLUTE - File Delivery over Unidirectional Transport",
RFC 6726, DOI 10.17487/RFC6726, November 2012,
<
http://www.rfc-editor.org/info/rfc6726>.
[RFC6773] Phelan, T., Fairhurst, G., and C. Perkins, "DCCP-UDP: A
Datagram Congestion Control Protocol UDP Encapsulation for
NAT Traversal", RFC 6773, DOI 10.17487/RFC6773, November
2012, <
http://www.rfc-editor.org/info/rfc6773>.
[RFC6817] Shalunov, S., Hazel, G., Iyengar, J., and M. Kuehlewind,
"Low Extra Delay Background Transport (LEDBAT)", RFC 6817,
DOI 10.17487/RFC6817, December 2012,
<
http://www.rfc-editor.org/info/rfc6817>.
[RFC6824] Ford, A., Raiciu, C., Handley, M., and O. Bonaventure,
"TCP Extensions for Multipath Operation with Multiple
Addresses", RFC 6824, DOI 10.17487/RFC6824, January 2013,
<
http://www.rfc-editor.org/info/rfc6824>.
Fairhurst, et al. Informational [Page 49]
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[RFC6897] Scharf, M. and A. Ford, "Multipath TCP (MPTCP) Application
Interface Considerations", RFC 6897, DOI 10.17487/RFC6897,
March 2013, <
http://www.rfc-editor.org/info/rfc6897>.
[RFC6935] Eubanks, M., Chimento, P., and M. Westerlund, "IPv6 and
UDP Checksums for Tunneled Packets", RFC 6935,
DOI 10.17487/RFC6935, April 2013,
<
http://www.rfc-editor.org/info/rfc6935>.
[RFC6936] Fairhurst, G. and M. Westerlund, "Applicability Statement
for the Use of IPv6 UDP Datagrams with Zero Checksums",
RFC 6936, DOI 10.17487/RFC6936, April 2013,
<
http://www.rfc-editor.org/info/rfc6936>.
[RFC6951] Tuexen, M. and R. Stewart, "UDP Encapsulation of Stream
Control Transmission Protocol (SCTP) Packets for End-Host
to End-Host Communication", RFC 6951,
DOI 10.17487/RFC6951, May 2013,
<
http://www.rfc-editor.org/info/rfc6951>.
[RFC7053] Tuexen, M., Ruengeler, I., and R. Stewart, "SACK-
IMMEDIATELY Extension for the Stream Control Transmission
Protocol", RFC 7053, DOI 10.17487/RFC7053, November 2013,
<
http://www.rfc-editor.org/info/rfc7053>.
[RFC7202] Perkins, C. and M. Westerlund, "Securing the RTP
Framework: Why RTP Does Not Mandate a Single Media
Security Solution", RFC 7202, DOI 10.17487/RFC7202, April
2014, <
http://www.rfc-editor.org/info/rfc7202>.
[RFC7230] Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer
Protocol (HTTP/1.1): Message Syntax and Routing",
RFC 7230, DOI 10.17487/RFC7230, June 2014,
<
http://www.rfc-editor.org/info/rfc7230>.
[RFC7231] Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer
Protocol (HTTP/1.1): Semantics and Content", RFC 7231,
DOI 10.17487/RFC7231, June 2014,
<
http://www.rfc-editor.org/info/rfc7231>.
[RFC7232] Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer
Protocol (HTTP/1.1): Conditional Requests", RFC 7232,
DOI 10.17487/RFC7232, June 2014,
<
http://www.rfc-editor.org/info/rfc7232>.
Fairhurst, et al. Informational [Page 50]
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[RFC7233] Fielding, R., Ed., Lafon, Y., Ed., and J. Reschke, Ed.,
"Hypertext Transfer Protocol (HTTP/1.1): Range Requests",
RFC 7233, DOI 10.17487/RFC7233, June 2014,
<
http://www.rfc-editor.org/info/rfc7233>.
[RFC7234] Fielding, R., Ed., Nottingham, M., Ed., and J. Reschke,
Ed., "Hypertext Transfer Protocol (HTTP/1.1): Caching",
RFC 7234, DOI 10.17487/RFC7234, June 2014,
<
http://www.rfc-editor.org/info/rfc7234>.
[RFC7235] Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer
Protocol (HTTP/1.1): Authentication", RFC 7235,
DOI 10.17487/RFC7235, June 2014,
<
http://www.rfc-editor.org/info/rfc7235>.
[RFC7301] Friedl, S., Popov, A., Langley, A., and E. Stephan,
"Transport Layer Security (TLS) Application-Layer Protocol
Negotiation Extension", RFC 7301, DOI 10.17487/RFC7301,
July 2014, <
http://www.rfc-editor.org/info/rfc7301>.
[RFC7323] Borman, D., Braden, B., Jacobson, V., and R.
Scheffenegger, Ed., "TCP Extensions for High Performance",
RFC 7323, DOI 10.17487/RFC7323, September 2014,
<
http://www.rfc-editor.org/info/rfc7323>.
[RFC7414] Duke, M., Braden, R., Eddy, W., Blanton, E., and A.
Zimmermann, "A Roadmap for Transmission Control Protocol
(TCP) Specification Documents", RFC 7414,
DOI 10.17487/RFC7414, February 2015,
<
http://www.rfc-editor.org/info/rfc7414>.
[RFC7457] Sheffer, Y., Holz, R., and P. Saint-Andre, "Summarizing
Known Attacks on Transport Layer Security (TLS) and
Datagram TLS (DTLS)", RFC 7457, DOI 10.17487/RFC7457,
February 2015, <
http://www.rfc-editor.org/info/rfc7457>.
[RFC7496] Tuexen, M., Seggelmann, R., Stewart, R., and S. Loreto,
"Additional Policies for the Partially Reliable Stream
Control Transmission Protocol Extension", RFC 7496,
DOI 10.17487/RFC7496, April 2015,
<
http://www.rfc-editor.org/info/rfc7496>.
[RFC7525] Sheffer, Y., Holz, R., and P. Saint-Andre,
"Recommendations for Secure Use of Transport Layer
Security (TLS) and Datagram Transport Layer Security
(DTLS)", BCP 195, RFC 7525, DOI 10.17487/RFC7525, May
2015, <
http://www.rfc-editor.org/info/rfc7525>.
Fairhurst, et al. Informational [Page 51]
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[RFC7540] Belshe, M., Peon, R., and M. Thomson, Ed., "Hypertext
Transfer Protocol Version 2 (HTTP/2)", RFC 7540,
DOI 10.17487/RFC7540, May 2015,
<
http://www.rfc-editor.org/info/rfc7540>.
[RFC8085] Eggert, L., Fairhurst, G., and G. Shepherd, "UDP Usage
Guidelines", BCP 145, RFC 8085, DOI 10.17487/RFC8085,
March 2017, <
http://www.rfc-editor.org/info/rfc8085>.
[SCTP-DTLS-ENCAPS]
Tuexen, M., Stewart, R., Jesup, R., and S. Loreto, "DTLS
Encapsulation of SCTP Packets", Work in Progress,
draft-ietf-tsvwg-sctp-dtls-encaps-09, January 2015.
[SCTP-NDATA]
Stewart, R., Tuexen, M., Loreto, S., and R. Seggelmann,
"Stream Schedulers and User Message Interleaving for the
Stream Control Transmission Protocol", Work in Progress,
draft-ietf-tsvwg-sctp-ndata-08, October 2016.
[TCP-SPEC] Eddy, W., Ed., "Transmission Control Protocol
Specification", Work in Progress, draft-ietf-tcpm-
rfc793bis-04, December 2016.
[TLS-1.3] Rescorla, E., "The Transport Layer Security (TLS) Protocol
Version 1.3", Work in Progress, draft-ietf-tls-tls13-18,
October 2016.
[WEBRTC-TRANS]
Alvestrand, H., "Transports for WebRTC", Work in
Progress, draft-ietf-rtcweb-transports-17, October 2016.
[XHR] van Kesteren, A., Aubourg, J., Song, J., and H. Steen,
"XMLHttpRequest Level 1", World Wide Web Consortium NOTE-
XMLHttpRequest-20161006, October 2016,
<
http://www.w3.org/TR/XMLHttpRequest/>.
Fairhurst, et al. Informational [Page 52]
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Acknowledgments
Thanks to Joe Touch, Michael Welzl, Spencer Dawkins, and the TAPS
working group for the comments, feedback, and discussion. This work
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.
Contributors
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
(
[email protected]) and Olivier Mehani
(
[email protected]).
o Section 3.3 on UDP was contributed by Kevin Fall
(
[email protected]).
o Section 3.5 on SCTP was contributed by Michael Tuexen (tuexen@fh-
muenster.de) and Karen Nielsen (
[email protected]).
o Section 3.7 on TLS and DTLS was contributed by Ralph Holz
(
[email protected]) and Olivier Mehani
(
[email protected]).
o Section 3.8 on RTP contains contributions from Colin Perkins
(
[email protected]).
o Section 3.9 on HTTP was contributed by Dragana Damjanovic
(
[email protected]).
o Section 3.10 on FLUTE/ALC was contributed by Vincent Roca
(
[email protected]).
o Section 3.11 on NORM was contributed by Brian Adamson
(
[email protected]).
Fairhurst, et al. Informational [Page 53]
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Authors' Addresses
Godred Fairhurst (editor)
University of Aberdeen
School of Engineering, Fraser Noble Building
Aberdeen AB24 3UE
Email:
[email protected]
Brian Trammell (editor)
ETH Zurich
Gloriastrasse 35
8092 Zurich
Switzerland
Email:
[email protected]
Mirja Kuehlewind (editor)
ETH Zurich
Gloriastrasse 35
8092 Zurich
Switzerland
Email:
[email protected]
Fairhurst, et al. Informational [Page 54]
