One document matched: draft-ietf-taps-transports-03.xml

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<rfc ipr="trust200902" docName="draft-ietf-taps-transports-03" category="info">


  <front>
    <title abbrev="TAPS Transports">Services provided by IETF transport protocols and congestion control mechanisms</title>

    <author initials="G." surname="Fairhurst" fullname="Godred Fairhurst" role="editor">
      <organization>University of Aberdeen</organization>
      <address>
        <postal>
          <street>School of Engineering, Fraser Noble Building</street>
          <city>Aberdeen AB24 3UE</city>
        </postal>
        <email>gorry@erg.abdn.ac.uk</email>
      </address>
    </author>
    <author initials="B." surname="Trammell" fullname="Brian Trammell" role="editor">
      <organization>ETH Zurich</organization>
      <address>
        <postal>
          <street>Gloriastrasse 35</street>
          <city>8092 Zurich</city>
          <country>Switzerland</country>
        </postal>
        <email>ietf@trammell.ch</email>
      </address>
    </author>
    <author initials="M." surname="Kuehlewind" fullname="Mirja Kuehlewind" role="editor">
      <organization>ETH Zurich</organization>
      <address>
        <postal>
          <street>Gloriastrasse 35</street>
          <city>8092 Zurich</city>
          <country>Switzerland</country>
        </postal>
        <email>mirja.kuehlewind@tik.ee.ethz.ch</email>
      </address>
    </author>

    <date year="2015" month="February" day="27"/>

    
    
    

    <abstract>


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



    </abstract>


  </front>

  <middle>


<section anchor="introduction" title="Introduction">

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

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

<t>Transport protocols can also be differentiated by the features of the
services they provide: for instance, SCTP offers a message-based service
that does not suffer head-of-line blocking when used with multiple stream,
because it can accept blocks of data out of order, UDP-Lite provides
partial integrity protection, and LEDBAT can provide low-priority
“scavenger” communication.</t>

</section>
<section anchor="terminology" title="Terminology">

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

<t>[NOTE: The terminology below was presented at the TAPS WG meeting
in Honolulu. While the factoring of the terminology seems uncontroversial,
there may be some entities which still require names (e.g. information
about the interface between the transport and lower layers which could
lead to the availability or unavailability of certain transport protocol
features). Comments are welcome via the TAPS mailing list.]</t>

<t><list style="hanging">
  <t hangText='Transport Service Feature:'>
  a specific end-to-end feature that a transport service provides to its
clients. Examples include confidentiality, reliable delivery, ordered
delivery, message-versus-stream orientation, etc.</t>
  <t hangText='Transport Service:'>
  a set of transport service features, without an association to any given
framing protocol, which provides a complete service to an application.</t>
  <t hangText='Transport Protocol:'>
  an implementation that provides one or more different transport services
using a specific framing and header format on the wire.</t>
  <t hangText='Transport Protocol Component:'>
  an implementation of a transport service feature within a protocol.</t>
  <t hangText='Transport Service Instance:'>
  an arrangement of transport protocols with a selected set of features
and configuration parameters that implements a single transport service,
e.g. a protocol stack (RTP over UDP).</t>
  <t hangText='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).</t>
</list></t>

</section>
<section anchor="existing-transport-protocols" title="Existing Transport Protocols">

<t>This section provides a list of known IETF transport protocol and
transport protocol frameworks.</t>

<t>[EDITOR’S NOTE: Contributions to the subsections below are welcome]</t>

<section anchor="transport-control-protocol-tcp" title="Transport Control Protocol (TCP)">

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

<section anchor="protocol-description" title="Protocol Description">

<t>TCP is a connection-oriented protocol, providing a three way handshake to allow a client and server to set up a connection, and mechanisms for orderly completion and immediate teardown of a connection. TCP is defined by a family of RFCs <xref target="RFC4614"/>.</t>

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

<t>TCP partitions a continuous stream of bytes into segments, sized to fit in IP packets. ICMP-based PathMTU discovery <xref target="RFC1191"/><xref target="RFC1981"/> as well as Packetization Layer Path MTU Discovery (PMTUD) <xref target="RFC4821"/> are supported.</t>

<t>Each byte in the stream is identified by a sequence number. The sequence number is used to order segments on receipt, to identify segments in acknowledgments, and to detect unacknowledged segments for retransmission. This is the basis of TCP’s reliable, ordered delivery of data in a stream. TCP Selective Acknowledgment <xref target="RFC2018"/> extends this mechanism by making it possible to identify missing segments more precisely, reducing spurious retransmission.</t>

<t>Receiver flow control is provided by a sliding window: limiting the amount of unacknowledged data that can be outstanding at a given time. The window scale option <xref target="RFC7323"/> allows a receiver to use windows greater than
64KB.</t>

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

<t>A TCP protocol instance can be extended <xref target="RFC4614"/> and tuned. Some features are sender-side only, requiring no negotiation with the receiver; some are receiver-side only, some are explicitly negotiated during connection setup.</t>

<t>By default, TCP segment partitioning uses Nagle’s algorithm <xref target="RFC0896"/> to buffer data at the sender into large segments, potentially incurring sender-side buffering delay; this algorithm can be disabled by the sender to transmit more immediately, e.g. to enable smoother interactive sessions.</t>

<t>[EDITOR’S NOTE: add URGENT and PUSH flag (note <xref target="RFC6093"/> says SHOULD NOT use due to the range of TCP implementations that process TCP urgent indications differently.) ]</t>

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

<t>A TCP service is unicast.</t>

</section>
<section anchor="interface-description" title="Interface description">

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

<t>In API implementations derived from the BSD Sockets API, TCP sockets are created using the <spanx style="verb">SOCK_STREAM</spanx> socket type.</t>

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

<t>(more on the API goes here)</t>

</section>
<section anchor="transport-protocol-components" title="Transport Protocol Components">

<t>The transport protocol components provided by TCP are:</t>

<t><list style="symbols">
  <t>unicast</t>
  <t>connection setup with feature negotiation and application-to-port mapping</t>
  <t>port multiplexing</t>
  <t>reliable delivery</t>
  <t>ordered delivery for each byte stream</t>
  <t>error detection (checksum)</t>
  <t>segmentation</t>
  <t>stream-oriented delivery in a single stream</t>
  <t>data bundling (Nagle’s algorithm)</t>
  <t>flow control</t>
  <t>congestion control</t>
</list></t>

<t>[EDITOR’S NOTE: discussion of how to map this to features and TAPS: what does the higher
layer need to decide? what can the transport layer decide based on global
settings? what must the transport layer decide based on network
characteristics?]</t>

</section>
</section>
<section anchor="multipath-tcp-mp-tcp" title="Multipath TCP (MP-TCP)">

<t>[EDITOR’S NOTE: a few sentences describing Multipath TCP <xref target="RFC6824"/> go
here. Note that this adds transport-layer multihoming to the components
TCP provides]</t>

</section>
<section anchor="stream-control-transmission-protocol-sctp" title="Stream Control Transmission Protocol (SCTP)">

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

<t>SCTP was originally developed for transporting telephony signalling messages
and is deployed in telephony signalling networks, especially in mobile telephony
networks.
Additionally, it is used in the WebRTC framework for data channels and is therefore deployed in all WEB-browsers supporting WebRTC.</t>

<t>[EDITOR’S NOTE: Michael Tuexen and Karen Nielsen signed up as contributors for these sections.]</t>

<section anchor="protocol-description-1" title="Protocol Description">

<t>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 do and
only supports unicast.</t>

<t>SCTP uses the 32-bit CRC32c for protecting SCTP packets against bit errors.
This is stronger than the 16-bit checksums used by TCP or UDP.
However, a partial checksum coverage as provided by DCCP or UDP-Lite is not
supported.</t>

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

<t>SCTP provides a message-oriented service. Multiple small user messages can
be bundled into a single SCTP packet to improve the efficiency. User messages
which would result in IP packets larger than the MTU will be fragmented at
the sender side and reassembled at the receiver side. There is no protocol
limit on the user message size.
<xref target="RFC4821"/> defines a method to perform packetization layer path MTU discovery
with probe packets using the padding chunks defined the <xref target="RFC4820"/>.</t>

<t><xref target="RFC4960"/> specifies a TCP friendly congestion control to protect the network
against overload. SCTP also uses a sliding window flow control to protect
receivers against overflow.</t>

<t>Each SCTP association has between 1 and 65536 uni-directional streams in
each direction. The number of streams can be different in each direction.
Every user-message is sent on a particular stream.
User messages can be sent ordered or un-ordered upon request by the upper layer.
Only all ordered messages sent on the same stream are delivered at the receiver
in the same order as sent by the sender. For user messages not requiring
fragmentation, this minimises head of line blocking. The base protocol defined
in <xref target="RFC4960"/> doesn’t allow interleaving of user-messages, which results in
sending a large message on one stream can block the sending of user messages
on other streams. <xref target="I-D.ietf-tsvwg-sctp-ndata"/> overcomes this limitation and
also allows to specify a scheduler for the sender side streams selection.
The stream re-configuration extension defined in <xref target="RFC6525"/> allows to reset
streams during the lifetime of an association and to increase the number of
streams, if the number of streams negotiated in the SCTP handshake is not
sufficient.</t>

<t>According to <xref target="RFC4960"/>, each user message sent is either delivered to
the receiver or, in case of excessive retransmissions, the association is
terminated in a non-graceful way, similar to the TCP behaviour.
In addition to this reliable transfer, the partial reliability extension
defined in <xref target="RFC3758"/> allows the sender to abandon user messages.
The application can specify the policy for abandoning user messages.
Examples for these policies include:</t>

<t><list style="symbols">
  <t>Limiting the time a user message is dealt with by the sender.</t>
  <t>Limiting the number of retransmissions for each fragment of a user message.</t>
  <t>Abandoning messages of lower priority in case of a send buffer shortage.</t>
</list></t>

<t>SCTP supports multi-homing. Each SCTP end-point uses a list of IP-addresses
and a single port number. These addresses can be any mixture of IPv4 and IPv6
addresses.
These addresses are negotiated during the handshake and the address
re-configuration extension specified in <xref target="RFC5061"/> can be used to change
these addresses during the livetime of an SCTP association.
This allows for transport layer mobility.
Multiple addresses are used for improved resilience.
If a remote address becomes unreachable, the traffic is switched over to a
reachable one, if one exists.
Each SCTP end-point supervises continuously the reachability of all peer
addresses using a heartbeat mechanism.</t>

<t>For securing user messages, the use of TLS over SCTP has been specified in <xref target="RFC3436"/>. However, this solution does not support all services provided
by SCTP (for example un-ordered delivery or partial reliability), and therefore
the use of DTLS over SCTP has been specified in <xref target="RFC6083"/> to overcome these
limitations. When using DTLS over SCTP, the application can use almost all
services provided by SCTP.</t>

<t>For legacy NAT traversal, <xref target="RFC6951"/> defines the UDP encapsulation of
SCTP-packets. Alternatively, SCTP packets can be encapsulated in DTLS packets
as specified in <xref target="I-D.ietf-tsvwg-sctp-dtls-encaps"/>. The latter encapsulation
is used with in the WebRTC context.</t>

<t>Having a well defined API is also a feature provided by SCTP as described in
the next subsection.</t>

</section>
<section anchor="interface-description-1" title="Interface Description">

<t><xref target="RFC4960"/> defines an abstract API for the base protocol.
An extension to the BSD Sockets API is defined in <xref target="RFC6458"/> and covers:</t>

<t><list style="symbols">
  <t>the base protocol defined in <xref target="RFC4960"/>.</t>
  <t>the SCTP Partial Reliability extension defined in <xref target="RFC3758"/>.</t>
  <t>the SCTP Authentication extension defined in <xref target="RFC4895"/>.</t>
  <t>the SCTP Dynamic Address Reconfiguration extension defined in <xref target="RFC5061"/>.</t>
</list></t>

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

<t><list style="symbols">
  <t>the SCTP SACK-IMMEDIATELY extension defined in <xref target="RFC7053"/>.</t>
  <t>the SCTP Stream Reconfiguration extension defined in <xref target="RFC6525"/>.</t>
  <t>the UDP Encapsulation of SCTP packets extension defined in <xref target="RFC6951"/>.</t>
  <t>the additional PR-SCTP policies defined in <xref target="I-D.ietf-tsvwg-sctp-prpolicies"/>.</t>
</list></t>

<t>Future documents describing SCTP protocol extensions are expected to describe
the corresponding BSD Sockets API extension in a <spanx style="verb">Socket API Considerations</spanx> section.</t>

<t>The SCTP socket API supports two kinds of sockets:</t>

<t><list style="symbols">
  <t>one-to-one style sockets (by using the socket type <spanx style="verb">SOCK_STREAM</spanx>).</t>
  <t>one-to-many style socket (by using the socket type <spanx style="verb">SOCK_SEQPACKET</spanx>).</t>
</list></t>

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

<t>The SCTP stack can provide information to the applications about state
changes of the individual paths and the association whenever they occur.
These events are delivered similar to user messages but are specifically
marked as notifications.</t>

<t>A couple of new functions have been introduced to support the use of
multiple local and remote addresses.
Additional SCTP-specific send and receive calls have been defined to allow
dealing with the SCTP specific information without using ancillary data
in the form of additional cmsgs, which are also defined.
These functions provide support for detecting partial delivery of
user messages and notifications.</t>

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

<t>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.</t>

</section>
<section anchor="transport-protocol-components-1" title="Transport Protocol Components">

<t>The transport protocol components provided by SCTP are:</t>

<t><list style="symbols">
  <t>unicast</t>
  <t>connection setup with feature negotiation and application-to-port mapping</t>
  <t>port multiplexing</t>
  <t>reliable or partially reliable delivery</t>
  <t>ordered and unordered delivery within a stream</t>
  <t>support for multiple prioritised streams</t>
  <t>flow control (slow receiver function)</t>
  <t>message-oriented delivery</t>
  <t>congestion control</t>
  <t>application PDU bundling</t>
  <t>application PDU fragmentation and reassembly</t>
  <t>integrity check</t>
  <t>transport layer multihoming for resilience</t>
  <t>transport layer mobility</t>
</list></t>

<t>[EDITOR’S NOTE: update this list.]</t>

</section>
</section>
<section anchor="user-datagram-protocol-udp" title="User Datagram Protocol (UDP)">

<t>The User Datagram Protocol (UDP) <xref target="RFC0768"/> <xref target="RFC2460"/> is an IETF
 standards track transport protocol.  It provides a uni-directional,
 datagram protocol which preserves message boundaries.  It provides
 none of the following transport features: error correction,
 congestion control, or flow control.  It can be used to send
 broadcast datagrams (IPv4) or multicast datagrams (IPv4 and IPv6), in
 addition to unicast (and anycast) datagrams.  IETF guidance on the
 use of UDP is provided in<xref target="RFC5405"/>.  UDP is widely implemented and
 widely used by common applications, especially DNS.</t>

<t>[EDITOR’S NOTE: Kevin Fall signed up as a contributor for this section.]</t>

<section anchor="protocol-description-2" title="Protocol Description">

<t>UDP is a connection-less protocol which maintains message boundaries,
with no connection setup or feature negotiation.  The protocol uses
independent messages, ordinarily called datagrams.  The lack of error
control and flow control implies messages may be damaged, re-ordered,
lost, or duplicated in transit.  A receiving application unable to
run sufficiently fast or frequently may miss messages.  The lack of
congestion handling implies UDP traffic may cause the loss of
messages from other protocols (e.g., TCP) when sharing the same
network paths.  UDP traffic can also cause the loss of other UDP
traffic in the same or other flows for the same reasons.</t>

<t>Messages with bit errors are ordinarily detected by an invalid end-
to-end checksum and are discarded before being delivered to an
application.  There are some exceptions to this general rule,
however.  UDP-Lite (see <xref target="RFC3828"/>, and below) provides the ability for
portions of the message contents to be exempt from checksum coverage.
It is also possible to create UDP datagrams with no checksum, and
while this is generally discouraged <xref target="RFC1122"/> <xref target="RFC5405"/>, certain
special cases permit its use <xref target="RFC6935"/>.  The checksum support
considerations for omitting the checksum are defined in <xref target="RFC6936"/>.
Note that due to the relatively weak form of checksum used by UDP,
applications that require end to end integrity of data are
recommended to include a stronger integrity check of their payload
data.</t>

<t>On transmission, UDP encapsulates each datagram into an IP packet,
which may in turn be fragmented by IP.  Applications concerned with
fragmentation or that have other requirements such as receiver flow
control, congestion control, PathMTU discovery/PLPMTUD, support for
ECN, etc need to be provided by protocols other than UDP <xref target="RFC5405"/>.</t>

</section>
<section anchor="interface-description-2" title="Interface Description">

<t><xref target="RFC0768"/> describes basic requirements for an API for UDP.
Guidance on use of common APIs is provided in <xref target="RFC5405"/>.</t>

<t>A UDP endpoint consists of a tuple of (IP address, port number).
Demultiplexing using multiple abstract endpoints (sockets) on the
same IP address are supported.  The same socket may be used by a
single server to interact with multiple clients (note: this behavior
differs from TCP, which uses a pair of tuples to identify a
connection).  Multiple server instances (processes) binding 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.</t>

<t>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
<xref target="RFC5405"/>.</t>

</section>
<section anchor="transport-protocol-components-2" title="Transport Protocol Components">

<t>The transport protocol components provided by UDP are:</t>

<t><list style="symbols">
  <t>unidirectional</t>
  <t>port multiplexing</t>
  <t>2-tuple endpoints</t>
  <t>IPv4 broadcast, multicast and anycast</t>
  <t>IPv6 multicast and anycast</t>
  <t>IPv6 jumbograms</t>
  <t>message-oriented delivery</t>
  <t>error detection (checksum)</t>
  <t>checksum optional</t>
</list></t>

</section>
</section>
<section anchor="lightweight-user-datagram-protocol-udp-lite" title="Lightweight User Datagram Protocol (UDP-Lite)">

<t>The Lightweight User Datagram Protocol (UDP-Lite) <xref target="RFC3828"/> is an IETF
standards track transport protocol.
UDP-Lite provides a bidirectional set of logical unicast or
multicast message streams over
a datagram protocol. IETF guidance on the use of UDP-Lite is provided in
<xref target="RFC5405"/>.</t>

<t>[EDITOR’S NOTE: Gorry Fairhurst signed up as a contributor for this
section.]</t>

<section anchor="protocol-description-3" title="Protocol Description">

<t>UDP-Lite is a connection-less datagram protocol,
with no connection setup or feature negotiation.
The protocol use messages,
rather than a byte-stream.  Each stream of messages is independently
managed, therefore retransmission does not hold back data sent using
other logical streams.</t>

<t>It provides multiplexing to multiple sockets on each host using port
numbers.
An active UDP-Lite session is identified by its four-tuple of local and
remote IP
addresses and local port and remote port numbers.</t>

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

<t>UDP-Lite changes the semantics of the UDP “payload length” field to
that of a “checksum coverage length” field.  Otherwise, UDP-Lite is
semantically identical to UDP.  Applications using UDP-Lite therefore
can not make
assumptions regarding the correctness of the data received in the
insensitive part of the UDP-Lite payload.</t>

<t>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 <xref target="RFC5405"/>.</t>

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

<t>A UDP-Lite service may support IPv4 broadcast, multicast, anycast and
unicast.</t>

</section>
<section anchor="interface-description-3" title="Interface Description">

<t>There is no current API specified in the RFC Series, but guidance on
use of common APIs is provided in <xref target="RFC5405"/>.</t>

<t>The interface of UDP-Lite differs
from that of UDP by the addition of a single (socket) option that
communicates a checksum coverage length value: at the sender, this
specifies the intended checksum coverage, with the remaining
unprotected part of the payload called the “error-insensitive part”.
The checksum coverage may also be made visible to the application
via the UDP-Lite MIB module <xref target="RFC5097"/>.</t>

</section>
<section anchor="transport-protocol-components-3" title="Transport Protocol Components">

<t>The transport protocol components provided by UDP-Lite are:</t>

<t><list style="symbols">
  <t>unicast</t>
  <t>IPv4 broadcast, multicast and anycast</t>
  <t>port multiplexing</t>
  <t>non-reliable, non-ordered delivery</t>
  <t>message-oriented delivery</t>
  <t>partial integrity protection</t>
</list></t>

</section>
</section>
<section anchor="datagram-congestion-control-protocol-dccp" title="Datagram Congestion Control Protocol (DCCP)">

<t>Datagram Congestion Control Protocol (DCCP) <xref target="RFC4340"/> is an
IETF standards track
bidirectional transport protocol that provides unicast connections of
congestion-controlled unreliable messages.  </t>

<t>[EDITOR’S NOTE: Gorry Fairhurst signed up as a contributor for this
section.]</t>

<t>The DCCP Problem Statement describes the goals that
DCCP sought to address <xref target="RFC4336"/>. It is suitable for
applications that transfer fairly large amounts of data and that can
benefit from control over the trade off between timeliness and
reliability <xref target="RFC4336"/>.</t>

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

<section anchor="protocol-description-4" title="Protocol Description">

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

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

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

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

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

<t>A DCCP protocol instance can be extended <xref target="RFC4340"/> and tuned.
Some features are sender-side only, requiring no negotiation with the
receiver;
some are receiver-side only, some are explicitly negotiated during
connection setup.</t>

<t>DCCP supports negotiation of the congestion control profile,
to provide Plug and Play congestion control mechanisms.
examples of specified profiles include
<xref target="RFC4341"/> <xref target="RFC4342"/> <xref target="RFC5662"/>.
All IETF-defined methods provide Congestion Control.</t>

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

<t>Upper layer protocols specified on top of DCCP 
include: DTLS <xref target="RFC5595"/>, RTP <xref target="RFC5672"/>, 
ICE/SDP <xref target="RFC6773"/>.</t>

<t>A DCCP service is unicast.</t>

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

</section>
<section anchor="interface-description-4" title="Interface Description">

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

<t>There is no current API specified in the RFC Series.</t>

</section>
<section anchor="transport-protocol-components-4" title="Transport Protocol Components">

<t>The transport protocol components provided by DCCP are:</t>

<t><list style="symbols">
  <t>unicast</t>
  <t>connection setup with feature negotiation and application-to-port mapping</t>
  <t>Service Codes</t>
  <t>port multiplexing</t>
  <t>non-reliable, ordered delivery</t>
  <t>flow control (slow receiver function)</t>
  <t>drop notification</t>
  <t>timestamps</t>
  <t>message-oriented delivery</t>
  <t>partial integrity protection</t>
</list></t>

</section>
</section>
<section anchor="realtime-transport-protocol-rtp" title="Realtime Transport Protocol (RTP)">

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

<t>[EDITOR’S NOTE: Varun Singh signed up as contributor for this section.]</t>

</section>
<section anchor="transport-layer-security-tls-and-datagram-tls-dtls-as-a" title="Transport Layer Security (TLS) and Datagram TLS (DTLS) as a">
<t>pseudo transport</t>

<t>[NOTE: A few words on TLS <xref target="RFC5246"/> and DTLS <xref target="RFC6347"/> here, 
and how they get used by other protocols to meet security 
goals as an add-on interlayer above transport.]</t>

<section anchor="protocol-description-5" title="Protocol Description">

</section>
<section anchor="interface-description-5" title="Interface Description">

</section>
<section anchor="transport-protocol-components-5" title="Transport Protocol Components">

</section>
</section>
<section anchor="hypertext-transport-protocol-http-as-a-pseudotransport" title="Hypertext Transport Protocol (HTTP) as a pseudotransport">

<t><xref target="RFC3205"/></t>

<t>[EDITOR’S NOTE: No identified contributor for this section yet.]</t>

<section anchor="protocol-description-6" title="Protocol Description">

</section>
<section anchor="interface-description-6" title="Interface Description">

</section>
<section anchor="transport-protocol-components-6" title="Transport Protocol Components">

</section>
</section>
<section anchor="websockets" title="WebSockets">

<t><xref target="RFC6455"/></t>

<t>[EDITOR’S NOTE: No identified contributor for this section yet.]</t>

<section anchor="protocol-description-7" title="Protocol Description">

</section>
<section anchor="interface-description-7" title="Interface Description">

</section>
<section anchor="transport-protocol-components-7" title="Transport Protocol Components">

</section>
</section>
</section>
<section anchor="transport-service-features" title="Transport Service Features">

<t>[EDITOR’S NOTE: this section will drawn from the candidate features provided by protocol components in the
previous section – please discuss on taps@ietf.org list]</t>

<section anchor="complete-protocol-feature-matrix" title="Complete Protocol Feature Matrix">

<t>[EDITOR’S NOTE: Dave Thaler has signed up as a contributor for this section. Michael Welzl also has a beginning of a matrix which could be useful here.]</t>

<t>[EDITOR’S NOTE: The below is a strawman proposal below by Gorry Fairhurst for initial discussion]</t>

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

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

<t>Note (1): this feature requires support in an upper layer protocol. </t>

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

</section>
</section>
<section anchor="iana-considerations" title="IANA Considerations">

<t>This document has no considerations for IANA.</t>

</section>
<section anchor="security-considerations" title="Security Considerations">

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

</section>
<section anchor="contributors" title="Contributors">

<t>[Editor’s Note: turn this into a real contributors section with addresses once we figure out how to trick the toolchain into doing so]</t>

<!--
 -
    ins: K. Fall
    name: Kevin Fall
    email: kfall@kfall.com
 -
    ins: M. Tuexen
    name: Michael Tuexen
    org: Muenster University of Applied Sciences
    street: Stegerwaldstrasse 39
    city: 48565 Steinfurt
    country: Germany
    email: tuexen@fh-muenster.de
-->

<t><list style="symbols">
  <t><xref target="user-datagram-protocol-udp"/> on UDP was contributed by Kevin Fall (kfall@kfall.com)</t>
  <t><xref target="stream-control-transmission-protocol-sctp"/> on SCTP was contributed by Michael Tuexen (tuexen@fh-muenster.de)</t>
</list></t>

</section>
<section anchor="acknowledgments" title="Acknowledgments">

<t>This work is partially supported by the European Commission under grant
agreement FP7-ICT-318627 mPlane; support does not imply endorsement.</t>

</section>


  </middle>

  <back>

    <references title='Normative References'>

&RFC0791;


    </references>

    <references title='Informative References'>

&RFC0768;
&RFC0793;
&RFC0896;
&RFC1122;
&RFC1191;
&RFC1981;
&RFC2018;
&RFC2460;
&RFC3168;
&RFC3205;
&RFC3390;
&RFC3436;
&RFC3758;
&RFC3828;
&RFC4336;
&RFC4340;
&RFC4341;
&RFC4342;
&RFC4614;
&RFC4820;
&RFC4821;
&RFC4895;
&RFC4960;
&RFC5061;
&RFC5097;
&RFC5246;
&RFC5348;
&RFC5405;
&RFC5595;
&RFC5596;
&RFC5662;
&RFC5672;
&RFC6773;
&RFC5925;
&RFC5681;
&RFC6083;
&RFC6093;
&RFC6525;
&RFC6298;
&RFC6935;
&RFC6936;
&RFC6455;
&RFC6347;
&RFC6458;
&RFC6691;
&RFC6824;
&RFC6951;
&RFC7053;
&RFC7323;
&I-D.ietf-aqm-ecn-benefits;
&I-D.ietf-tsvwg-sctp-dtls-encaps;
&I-D.ietf-tsvwg-sctp-prpolicies;
&I-D.ietf-tsvwg-sctp-ndata;


    </references>



  </back>
</rfc>