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<rfc xmlns:xi="http://www.w3.org/2001/XInclude" ipr="trust200902" docName="draft-ietf-taps-interface-16" category="std" consensus="true" tocInclude="true" sortRefs="true" symRefs="true" version="3">
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  <front>
    <title abbrev="TAPS Interface">An Abstract Application Layer Interface to Transport Services</title>
    <seriesInfo name="Internet-Draft" value="draft-ietf-taps-interface-16"/>
    <author initials="B." surname="Trammell" fullname="Brian Trammell" role="editor">
      <organization>Google Switzerland GmbH</organization>
      <address>
        <postal>
          <street>Gustav-Gull-Platz 1</street>
          <city>8004 Zurich</city>
          <country>Switzerland</country>
        </postal>
        <email>ietf@trammell.ch</email>
      </address>
    </author>
    <author initials="M." surname="Welzl" fullname="Michael Welzl" role="editor">
      <organization>University of Oslo</organization>
      <address>
        <postal>
          <street>PO Box 1080 Blindern</street>
          <city>0316  Oslo</city>
          <country>Norway</country>
        </postal>
        <email>michawe@ifi.uio.no</email>
      </address>
    </author>
    <author initials="T." surname="Enghardt" fullname="Theresa Enghardt">
      <organization>Netflix</organization>
      <address>
        <postal>
          <street>121 Albright Way</street>
          <city>Los Gatos, CA 95032</city>
          <country>United States of America</country>
        </postal>
        <email>ietf@tenghardt.net</email>
      </address>
    </author>
    <author initials="G." surname="Fairhurst" fullname="Godred Fairhurst">
      <organization>University of Aberdeen</organization>
      <address>
        <postal>
          <street>Fraser Noble Building</street>
          <city>Aberdeen, AB24 3UE</city>
          <country>Scotland</country>
        </postal>
        <email>gorry@erg.abdn.ac.uk</email>
        <uri>http://www.erg.abdn.ac.uk/</uri>
      </address>
    </author>
    <author initials="M." surname="Kuehlewind" fullname="Mirja Kuehlewind">
      <organization>Ericsson</organization>
      <address>
        <postal>
          <street>Ericsson-Allee 1</street>
          <city>Herzogenrath</city>
          <country>Germany</country>
        </postal>
        <email>mirja.kuehlewind@ericsson.com</email>
      </address>
    </author>
    <author initials="C." surname="Perkins" fullname="Colin Perkins">
      <organization>University of Glasgow</organization>
      <address>
        <postal>
          <street>School of Computing Science</street>
          <city>Glasgow  G12 8QQ</city>
          <country>United Kingdom</country>
        </postal>
        <email>csp@csperkins.org</email>
      </address>
    </author>
    <author initials="P." surname="Tiesel" fullname="Philipp S. Tiesel">
      <organization>SAP SE</organization>
      <address>
        <postal>
          <street>Konrad-Zuse-Ring 10</street>
          <city>14469 Potsdam</city>
          <country>Germany</country>
        </postal>
        <email>philipp@tiesel.net</email>
      </address>
    </author>
    <author initials="T." surname="Pauly" fullname="Tommy Pauly">
      <organization>Apple Inc.</organization>
      <address>
        <postal>
          <street>One Apple Park Way</street>
          <city>Cupertino, California 95014</city>
          <country>United States of America</country>
        </postal>
        <email>tpauly@apple.com</email>
      </address>
    </author>
    <date year="2022" month="September" day="01"/>
    <workgroup>TAPS Working Group</workgroup>
    <keyword>Internet-Draft</keyword>
    <abstract>
      <t>This document describes an abstract application programming interface, API, to the transport
layer that enables the selection of transport protocols and
network paths dynamically at runtime. This API enables faster deployment
of new protocols and protocol features without requiring changes to the
applications. The specified API follows the Transport Services architecture
by providing asynchronous, atomic transmission of messages. It is intended to replace the
BSD sockets API as the common interface to the
transport layer, in an environment where endpoints could select from
multiple interfaces and potential transport protocols.</t>
    </abstract>
  </front>
  <middle>
    <section anchor="introduction">
      <name>Introduction</name>
      <t>This document specifies an abstract application programming interface (API) that specifies the interface component of
the high-level Transport Services architecture defined in
<xref target="I-D.ietf-taps-arch"/>. A Transport Services system supports
asynchronous, atomic transmission of messages over transport protocols and
network paths dynamically selected at runtime, in environments where an endpoint
selects from multiple interfaces and potential transport protocols.</t>
      <t>Applications that adopt this API will benefit from a wide set of
transport features that can evolve over time. This protocol-independent API ensures that the system
providing the API can optimize its behavior based on the application
requirements and network conditions, without requiring changes to the
applications.  This flexibility enables faster deployment of new features and
protocols, and can support applications by offering racing and fallback
mechanisms, which otherwise need to be separately implemented in each application.</t>
      <t>The Transport Services system derives specific path and protocol selection
properties and supported transport features from the analysis provided in
<xref target="RFC8095"/>, <xref target="RFC8923"/>, and
<xref target="RFC8922"/>. The Transport Services API enables an implementation
to dynamically choose a transport protocol rather
than statically binding applications to a protocol at
compile time. The Transport Services API also provides
applications with a way to override transport selection and instantiate a specific stack,
e.g., to support servers wishing to listen to a specific protocol. However, forcing a
choice to use a specific transport stack is discouraged for general use, because it can reduce portability.</t>
      <section anchor="notation">
        <name>Terminology and Notation</name>
        <t>The Transport Services API is described in terms of</t>
        <ul spacing="normal">
          <li>Objects with which an application can interact;</li>
          <li>Actions the application can perform on these Objects;</li>
          <li>Events, which an Object can send to an application to be processed aynchronously; and</li>
          <li>Parameters associated with these Actions and Events.</li>
        </ul>
        <t>The following notations, which can be combined, are used in this document:</t>
        <ul spacing="normal">
          <li>An Action that creates an Object:</li>
        </ul>
        <artwork><![CDATA[
      Object := Action()
]]></artwork>
        <ul spacing="normal">
          <li>An Action that creates an array of Objects:</li>
        </ul>
        <artwork><![CDATA[
      []Object := Action()
]]></artwork>
        <ul spacing="normal">
          <li>An Action that is performed on an Object:</li>
        </ul>
        <artwork><![CDATA[
      Object.Action()
]]></artwork>
        <ul spacing="normal">
          <li>An Object sends an Event:</li>
        </ul>
        <artwork><![CDATA[
      Object -> Event<>
]]></artwork>
        <ul spacing="normal">
          <li>An Action takes a set of Parameters; an Event contains a set of Parameters.
Action and Event parameters whose names are suffixed with a question mark are optional.</li>
        </ul>
        <artwork><![CDATA[
      Action(param0, param1?, ...) / Event<param0, param1, ...>
]]></artwork>
        <t>Objects that are passed as parameters to Actions use call-by-value behavior.
Actions associated with no Object are Actions on the API; they are equivalent to Actions on a per-application global context.</t>
        <t>Events are sent to the application or application-supplied code (e.g. framers,
see <xref target="framing"/>) for processing; the details of event interfaces are platform-
and implementation-specific, and may be implemented using
other forms of asynchronous processing, as idiomatic for the
implementing platform.</t>
        <t>We also make use of the following basic types:</t>
        <ul spacing="normal">
          <li>Boolean: Instances take the value <tt>true</tt> or <tt>false</tt>.</li>
          <li>Integer: Instances take positive or negative integer values.</li>
          <li>Numeric: Instances take positive or negative real number values.</li>
          <li>Enumeration: A family of types in which each instance takes one of a fixed,
predefined set of values specific to a given enumerated type.</li>
          <li>Tuple: An ordered grouping of multiple value types, represented as a
comma-separated list in parentheses, e.g., <tt>(Enumeration, Preference)</tt>.
Instances take a sequence of values each valid for the corresponding value
type.</li>
          <li>Array: Denoted []Type, an instance takes a value for each of zero or more
elements in a sequence of the given Type. An array may be of fixed or
variable length.</li>
          <li>Collection: An unordered grouping of one or more values of the same type.</li>
        </ul>
        <t>For guidance on how these abstract concepts may be implemented in languages
in accordance with native design patterns and language and platform features,
see <xref target="implmapping"/>.</t>
      </section>
      <section anchor="specification-of-requirements">
        <name>Specification of Requirements</name>
        <t>The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD",
"SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in BCP 14 <xref target="RFC2119"/> <xref target="RFC8174"/>
when, and only when, they appear in all capitals, as shown here.</t>
      </section>
    </section>
    <section anchor="principles">
      <name>Overview of the API Design</name>
      <t>The design of the API specified in this document is based on a set of
principles, themselves an elaboration on the architectural design principles
defined in <xref target="I-D.ietf-taps-arch"/>. The API defined in this document
provides:</t>
      <ul spacing="normal">
        <li>A Transport Services system
can offer a variety of transport protocols, independent
of the Protocol Stacks that will be used at
runtime. All common features of these protocol
stacks are made available to the application in a
transport-independent way to the degree possible.
This enables applications written to a single API
to make use of transport protocols in terms of the features
they provide.</li>
        <li>A unified API to datagram and stream-oriented transports, allowing
use of a common API for connection establishment and closing.</li>
        <li>Message-orientation, as opposed to stream-orientation, using
application-assisted framing and deframing where the underlying transport
does not provide these.</li>
        <li>Asynchronous Connection establishment, transmission, and reception.
This allows concurrent operations during establishment and event-driven
application interactions with the transport layer;</li>
        <li>Selection between alternate network paths, using additional information about the
networks over which a connection can operate (e.g. Provisioning Domain (PvD)
information <xref target="RFC7556"/>) where available.</li>
        <li>Explicit support for transport-specific features to be applied, should that
particular transport be part of a chosen Protocol Stack.</li>
        <li>Explicit support for security properties as first-order transport features.</li>
        <li>Explicit support for configuration of cryptographic identities and transport
security parameters persistent across multiple Connections.</li>
        <li>Explicit support for multistreaming and multipath transport protocols, and
the grouping of related Connections into Connection Groups through "cloning"
of Connections (see <xref target="groups"/>). This function allows applications to take full advantage of new
transport protocols supporting these features.</li>
      </ul>
    </section>
    <section anchor="api-summary">
      <name>API Summary</name>
      <t>An application primarily interacts with this API through two Objects:
Preconnections and Connections. A Preconnection object (<xref target="pre-establishment"/>)
represents a set of properties and constraints on the selection and configuration of
paths and protocols to establish a Connection with an Endpoint. A Connection object
represents an instance of a transport Protocol Stack on which data can be sent to and/or
received from a Remote Endpoint (i.e., a logical connection that, depending on the
kind of transport, can be bi-directional or unidirectional, and that can use a stream
protocol or a datagram protocol). Connections are presented consistently to the
application, irrespective of whether the underlying transport is connection-less or
connection-oriented. Connections can be created from Preconnections in three ways:</t>
      <ul spacing="normal">
        <li>by initiating the Preconnection (i.e., actively opening, as in a client; <xref target="initiate"/>),</li>
        <li>through listening on the Preconnection (i.e., passively opening, as in a server <xref target="listen"/>),</li>
        <li>or rendezvousing on the Preconnection (i.e., peer to peer establishment; <xref target="rendezvous"/>).</li>
      </ul>
      <t>Once a Connection is established, data can be sent and received on it in the form of
Messages. The API supports the preservation of message boundaries both
via explicit Protocol Stack support, and via application support through a
Message Framer that finds message boundaries in a stream. Messages are
received asynchronously through event handlers registered by the application.
Errors and other notifications also happen asynchronously on the Connection.
It is not necessary for an application to handle all Events; some Events may
have implementation-specific default handlers. The application should not
assume that ignoring Events (e.g., Errors) is always safe.</t>
      <section anchor="usage-examples">
        <name>Usage Examples</name>
        <t>The following usage examples illustrate how an application might use the
Transport Services API to:</t>
        <ul spacing="normal">
          <li>Act as a server, by listening for incoming connections, receiving requests,
and sending responses, see <xref target="server-example"/>.</li>
          <li>Act as a client, by connecting to a Remote Endpoint using Initiate, sending
requests, and receiving responses, see <xref target="client-example"/>.</li>
          <li>Act as a peer, by connecting to a Remote Endpoint using Rendezvous while
simultaneously waiting for incoming Connections, sending Messages, and
receiving Messages, see <xref target="peer-example"/>.</li>
        </ul>
        <t>The examples in this section presume that a transport protocol is available
between the Local and Remote Endpoints that provides Reliable Data Transfer, Preservation of
Data Ordering, and Preservation of Message Boundaries. In this case, the
application can choose to receive only complete messages.</t>
        <t>If none of the available transport protocols provides Preservation of Message
Boundaries, but there is a transport protocol that provides a reliable ordered
byte stream, an application could receive this byte stream as partial
Messages and transform it into application-layer Messages.  Alternatively,
an application might provide a Message Framer, which can transform a
sequence of Messages into a byte stream and vice versa (<xref target="framing"/>).</t>
        <section anchor="server-example">
          <name>Server Example</name>
          <t>This is an example of how an application might listen for incoming Connections
using the Transport Services API, and receive a request, and send a response.</t>
          <artwork><![CDATA[
LocalSpecifier := NewLocalEndpoint()
LocalSpecifier.WithInterface("any")
LocalSpecifier.WithService("https")

TransportProperties := NewTransportProperties()
TransportProperties.Require(preserve-msg-boundaries)
// Reliable Data Transfer and Preserve Order are Required by default

SecurityParameters := NewSecurityParameters()
SecurityParameters.Set(identity, myIdentity)
SecurityParameters.Set(key-pair, myPrivateKey, myPublicKey)

// Specifying a Remote Endpoint is optional when using Listen()
Preconnection := NewPreconnection(LocalSpecifier,
                                  TransportProperties,
                                  SecurityParameters)

Listener := Preconnection.Listen()

Listener -> ConnectionReceived<Connection>

// Only receive complete messages in a Conn.Received handler
Connection.Receive()

Connection -> Received<messageDataRequest, messageContext>

//---- Receive event handler begin ----
Connection.Send(messageDataResponse)
Connection.Close()

// Stop listening for incoming Connections
// (this example supports only one Connection)
Listener.Stop()
//---- Receive event handler end ----

]]></artwork>
        </section>
        <section anchor="client-example">
          <name>Client Example</name>
          <t>This is an example of how an application might open two Connections to a remote application
using the Transport Services API, and send a request as well as receive a response on each of them.</t>
          <artwork><![CDATA[
RemoteSpecifier := NewRemoteEndpoint()
RemoteSpecifier.WithHostname("example.com")
RemoteSpecifier.WithService("https")

TransportProperties := NewTransportProperties()
TransportProperties.Require(preserve-msg-boundaries)
// Reliable Data Transfer and Preserve Order are Required by default

SecurityParameters := NewSecurityParameters()
TrustCallback := NewCallback({
  // Verify identity of the Remote Endpoint, return the result
})
SecurityParameters.SetTrustVerificationCallback(TrustCallback)

// Specifying a local endpoint is optional when using Initiate()
Preconnection := NewPreconnection(RemoteSpecifier,
                                  TransportProperties,
                                  SecurityParameters)

Connection := Preconnection.Initiate()
Connection2 := Connection.Clone()

Connection -> Ready<>
Connection2 -> Ready<>

//---- Ready event handler for any Connection C begin ----
C.Send(messageDataRequest)

// Only receive complete messages
C.Receive()
//---- Ready event handler for any Connection C end ----

Connection -> Received<messageDataResponse, messageContext>
Connection2 -> Received<messageDataResponse, messageContext>

// Close the Connection in a Receive event handler
Connection.Close()
Connection2.Close()
]]></artwork>
          <t>Preconnections are reusable after being used to initiate a Connection. Hence, for example, after the Connections were closed,
the following would be correct:</t>
          <artwork><![CDATA[
//.. carry out adjustments to the Preconnection, if desire
Connection := Preconnection.Initiate()
]]></artwork>
        </section>
        <section anchor="peer-example">
          <name>Peer Example</name>
          <t>This is an example of how an application might establish a connection with a
peer using Rendezvous(), send a Message, and receive a Message.</t>
          <artwork><![CDATA[
// Configure local candidates: a port on the Local Endpoint
// and via a STUN server
HostCandidate := NewLocalEndpoint()
HostCandidate.WithPort(9876)

StunCandidate := NewLocalEndpoint()
StunCandidate.WithStunServer(address, port, credentials)

LocalCandidates = [HostCandidate, StunCandidate]

// Configure transport and security properties
TransportProperties := ...
SecurityParameters  := ...

Preconnection := NewPreconnection(LocalCandidates,
                                  [], // No remote candidates yet
                                  TransportProperties,
                                  SecurityParameters)

// Resolve the LocalCandidates. The Preconnection.Resolve() call
// resolves both local and remote candidates but, since the remote
// candidates have not yet been specified, the ResolvedRemote list
// returned will be empty and is not used.
ResolvedLocal, ResolvedRemote = Preconnection.Resolve()

// ...Send the ResolvedLocal list to peer via signalling channel
// ...Receive a list of RemoteCandidates from peer via
//    signalling channel

Preconnection.AddRemote(RemoteCandidates)
Preconnection.Rendezvous()

Preconnection -> RendezvousDone<Connection>

//---- RendezvousDone event handler begin ----
Connection.Send(messageDataRequest)
Connection.Receive()
//---- RendezvousDone event handler end ----

Connection -> Received<messageDataResponse, messageContext>

// If new remote endpoint candidates are received from the peer over
// the signalling channel, for example if using Trickle ICE, then add
// them to the Connection:
Connection.AddRemote(NewRemoteCandidates)

// On a PathChange<> events, resolve the local endpoints to see if a
// new local endpoint has become available and, if so, send to the peer
// as a new candidate and add to the connection:
Connection -> PathChange<>

//---- PathChange event handler begin ----
ResolvedLocal, ResolvedRemote = Preconnection.Resolve()
if ResolvedLocal has changed:
  // ...Send the ResolvedLocal list to peer via signalling channel
  // Add the new local endpoints to the connection:
  Connection.AddLocal(ResolvedLocal)
//---- PathChange event handler end ----


// Close the Connection in a Receive event handler
Connection.Close()
]]></artwork>
        </section>
      </section>
    </section>
    <section anchor="transport-properties">
      <name>Transport Properties</name>
      <t>Each application using the Transport Services API declares its preferences
for how the Transport Services system should operate. This is done by using Transport Properties, as defined in
<xref target="I-D.ietf-taps-arch"/>, at each stage of the lifetime of a connection.</t>
      <t>Transport Properties are divided into Selection, Connection, and Message
Properties. Selection Properties (see <xref target="selection-props"/>) can only be set during pre-establishment. They are only used to specify which paths and protocol stacks can be used and are preferred by the application.
Although Connection Properties (see <xref target="connection-props"/>) can be set during pre-establishment, they may be changed later. They are used to inform decisions made during establishment and to fine-tune the established connection. Calling Initiate on a Preconnection creates an outbound Connection or a Listener, and the Selection Properties remain readable from the Connection or Listener, but become immutable.</t>
      <t>The behavior of the selected protocol stack(s) when
sending Messages is controlled by Message Properties (see <xref target="message-props"/>).</t>
      <t>Selection Properties can be set on Preconnections, and the effect of
Selection Properties can be queried on Connections and Messages.
Connection Properties can be set on Connections and Preconnections;
when set on Preconnections, they act as an initial default for the
resulting Connections. Message Properties can be set on Messages,
Connections, and Preconnections; when set on the latter two, they act as
an initial default for the Messages sent over those Connections,</t>
      <t>Note that configuring Connection Properties and Message Properties on
Preconnections is preferred over setting them later. Early specification of
Connection Properties allows their use as additional input to the selection
process. Protocol Specific Properties, which enable configuration of specialized
features of a specific protocol, see Section 3.2 of <xref target="I-D.ietf-taps-arch"/>, are not
used as an input to the selection process, but only support configuration if
the respective protocol has been selected.</t>
      <section anchor="property-names">
        <name>Transport Property Names</name>
        <t>Transport Properties are referred to by property names. For the purposes of this document, these names are
alphanumeric strings in which words may be separated by hyphens. Specifically, the following characters are allowed: lowercase letters <tt>a-z</tt>, uppercase letters <tt>A-Z</tt>, digits <tt>0-9</tt>, the hyphen <tt>-</tt>, and the underscore <tt>_</tt>.
These names serve two purposes:</t>
        <ul spacing="normal">
          <li>Allowing different components of a Transport Services implementation to pass Transport
Properties, e.g., between a language frontend and a policy manager,
or as a representation of properties retrieved from a file or other storage.</li>
          <li>Making the code of different Transport Services implementations look similar. While individual programming languages may preclude strict adherence to the aforementioned naming convention (for instance, by prohibiting the use of hyphens in symbols), users interacting with multiple implementations will still benefit from the consistency resulting from the use of visually similar symbols.</li>
        </ul>
        <t>Transport Property Names are hierarchically organized in the
form [&lt;Namespace&gt;.]&lt;PropertyName&gt;.</t>
        <ul spacing="normal">
          <li>The Namespace component MUST be empty for well-known, generic properties, i.e., for
properties that are not specific to a protocol and are defined in an RFC.</li>
          <li>Protocol Specific Properties MUST use the protocol acronym as the Namespace, e.g.,
<tt>tcp</tt> for TCP specific Transport Properties. For IETF protocols, property
names under these namespaces SHOULD be defined in an RFC.</li>
          <li>Vendor or implementation specific properties MUST use a string identifying
the vendor or implementation as the Namespace.</li>
        </ul>
        <t>Namespaces for each of the keywords provided in the IANA protocol numbers registry
(see https://www.iana.org/assignments/protocol-numbers/protocol-numbers.xhtml) are reserved
for Protocol Specific Properties and MUST NOT be used for vendor or implementation-specific properties.
Avoid using any of the terms listed as keywords in the protocol numbers registry as any part of a vendor- or
implementation-specific property name.</t>
      </section>
      <section anchor="property-types">
        <name>Transport Property Types</name>
        <t>Each Transport Property has a one of the basic types described in <xref target="notation"/>.</t>
        <t>Most Selection Properties (see <xref target="selection-props"/>) are of the Enumeration type,
and use the Preference Enumeration, which takes one of five possible values
(Prohibit, Avoid, Ignore,  Prefer, or Require) denoting the level of preference
for a given property during protocol selection.</t>
      </section>
    </section>
    <section anchor="scope-of-interface-defn">
      <name>Scope of the API Definition</name>
      <t>This document defines a language- and platform-independent API of a
Transport Services system. Given the wide variety of languages and language
conventions used to write applications that use the transport layer to connect
to other applications over the Internet, this independence makes this API
necessarily abstract.</t>
      <t>There is no interoperability benefit in tightly defining how the API is
presented to application programmers across diverse platforms. However,
maintaining the "shape" of the abstract API across different platforms reduces
the effort for programmers who learn to use the Transport Services API to then
apply their knowledge to another platform.</t>
      <t>We therefore make the following recommendations:</t>
      <ul spacing="normal">
        <li>Actions, Events, and Errors in implementations of the Transport Services API SHOULD use
the names given for them in the document, subject to capitalization,
punctuation, and other typographic conventions in the language of the
implementation, unless the implementation itself uses different names for
substantially equivalent objects for networking by convention.</li>
        <li>Transport Services systems SHOULD implement each Selection Property,
Connection Property, and Message Context Property specified in this document.
The Transport Services API SHOULD be implemented even when in a specific implementation/platform it
will always result in no operation, e.g. there is no action when the API
specifies a Property that is not available in a transport protocol implemented
on a specific platform. For example, if TCP is the only underlying transport protocol,
the Message Property <tt>msgOrdered</tt> can be implemented (trivially, as a no-op) as
disabling the requirement for ordering will not have any effect on delivery order
for Connections over TCP. Similarly, the <tt>msgLifetime</tt> Message Property can be
implemented but ignored, as the description of this Property states that "it is not
guaranteed that a Message will not be sent when its Lifetime has expired".</li>
        <li>Implementations may use other representations for Transport Property Names,
e.g., by providing constants, but should provide a straight-forward mapping
between their representation and the property names specified here.</li>
      </ul>
    </section>
    <section anchor="pre-establishment">
      <name>Pre-Establishment Phase</name>
      <t>The Pre-Establishment phase allows applications to specify properties for
the Connections that they are about to make, or to query the API about potential
Connections they could make.</t>
      <t>A Preconnection Object represents a potential Connection. It is a passive Object
(a data structure) that merely maintains the state that
describes the properties of a Connection that might exist in the future.  This state
comprises Local Endpoint and Remote Endpoint Objects that denote the endpoints
of the potential Connection (see <xref target="endpointspec"/>), the Selection Properties
(see <xref target="selection-props"/>), any preconfigured Connection Properties
(<xref target="connection-props"/>), and the security parameters (see
<xref target="security-parameters"/>):</t>
      <artwork><![CDATA[
   Preconnection := NewPreconnection([]LocalEndpoint,
                                     []RemoteEndpoint,
                                     TransportProperties,
                                     SecurityParameters)
]]></artwork>
      <t>At least one Local Endpoint MUST be specified if the Preconnection is used to Listen()
for incoming Connections, but the list of Local Endpoints MAY be empty if
the Preconnection is used to Initiate()
connections. If no Local Endpoint is specified, the Transport Services system will
assign an ephemeral local port to the Connection on the appropriate interface(s).
At least one Remote Endpoint MUST be specified if the Preconnection is used
to Initiate() Connections, but the list of Remote Endpoints MAY be empty if
the Preconnection is used to Listen() for incoming Connections.
At least one Local Endpoint and one Remote Endpoint MUST be specified if a
peer-to-peer Rendezvous() is to occur based on the Preconnection.</t>
      <t>If more than one Local Endpoint is specified on a Preconnection, then all
the Local Endpoints on the Preconnection MUST represent the same host. For
example, they might correspond to different interfaces on a multi-homed
host, of they might correspond to local interfaces and a STUN server that
can be resolved to a server reflexive address for a Preconnection used to
make a peer-to-peer Rendezvous().</t>
      <t>If more than one Remote Endpoint is specified on the Preconnection, then
all the Remote Endpoints on the Preconnection SHOULD represent the same
service. For example, the Remote Endpoints might represent various network
interfaces of a host, or a server reflexive address that can be used to
reach a host, or a set of hosts that provide equivalent local balanced
service.</t>
      <t>In most cases, it is expected that a single Remote Endpoint will be
specified by name, and a later call to  Initiate() on the Preconnection
(see <xref target="initiate"/>) will internally resolve that name to a list of concrete
endpoints. Specifying multiple Remote Endpoints on a Preconnection allows
applications to override this for more detailed control.</t>
      <t>If Message Framers are used (see <xref target="framing"/>), they MUST be added to the
Preconnection during pre-establishment.</t>
      <section anchor="endpointspec">
        <name>Specifying Endpoints</name>
        <t>The transport services API uses the Local Endpoint and Remote Endpoint Objects
to refer to the endpoints of a transport connection. Endpoints can be created
as either Remote or Local:</t>
        <artwork><![CDATA[
RemoteSpecifier := NewRemoteEndpoint()
LocalSpecifier := NewLocalEndpoint()
]]></artwork>
        <t>A single Endpoint Object represents the identity of a network host. That endpoint
can be more or less specific depending on which identifiers are set. For example,
an Endpoint that only specifies a hostname may in fact end up corresponding
to several different IP addresses on different hosts.</t>
        <t>An Endpoint Object can be configured with the following identifiers:</t>
        <ul spacing="normal">
          <li>Hostname (string):</li>
        </ul>
        <artwork><![CDATA[
RemoteSpecifier.WithHostname("example.com")
]]></artwork>
        <ul spacing="normal">
          <li>Port (a 16-bit integer):</li>
        </ul>
        <artwork><![CDATA[
RemoteSpecifier.WithPort(443)
]]></artwork>
        <ul spacing="normal">
          <li>Service (an identifier that maps to a port; either a the name of a well-known service, or a DNS SRV service name to be resolved):</li>
        </ul>
        <artwork><![CDATA[
RemoteSpecifier.WithService("https")
]]></artwork>
        <ul spacing="normal">
          <li>IP address (IPv4 or IPv6 address):</li>
        </ul>
        <artwork><![CDATA[
RemoteSpecifier.WithIPv4Address(192.0.2.21)
]]></artwork>
        <artwork><![CDATA[
RemoteSpecifier.WithIPv6Address(2001:db8:4920:e29d:a420:7461:7073:0a)
]]></artwork>
        <ul spacing="normal">
          <li>Interface name (string), e.g., to qualify link-local or multicast addresses (see <xref target="ifspec"/> for details):</li>
        </ul>
        <artwork><![CDATA[
LocalSpecifier.WithInterface("en0")
]]></artwork>
        <t>Note that an IPv6 address specified with a scope (e.g. <tt>2001:db8:4920:e29d:a420:7461:7073:0a%en0</tt>)
is equivalent to <tt>WithIPv6Address</tt> with an unscoped address and <tt>WithInterface </tt> together.</t>
        <t>An Endpoint cannot have multiple identifiers of a same type set. That is,
an endpoint cannot have two IP addresses specified. Two separate IP addresses
are represented as two Endpoint Objects. If a Preconnection specifies a Remote
Endpoint with a specific IP address set, it will only establish Connections to
that IP address. If, on the other hand, the Remote Endpoint specifies a hostname
but no addresses, the Connection can perform name resolution and attempt
using any address derived from the original hostname of the Remote Endpoint.
Note that multiple Remote Endpoints can be added to a Preconnection, as discussed
in <xref target="add-endpoints"/>.</t>
        <t>The Transport Services system resolves names internally, when the Initiate(),
Listen(), or Rendezvous() method is called to establish a Connection. Privacy
considerations for the timing of this resolution are given in <xref target="privacy-security"/>.</t>
        <t>The Resolve() action on a Preconnection can be used by the application to force
early binding when required, for example with some Network Address Translator
(NAT) traversal protocols (see <xref target="rendezvous"/>).</t>
        <section anchor="using-multicast-endpoints">
          <name>Using Multicast Endpoints</name>
          <t>Specifying a multicast group address on a Local Endpoint will indicate to the Transport
Services system that the resulting connection will be used to receive multicast messages. The
Remote Endpoint can be used to filter incoming multicast from specific senders. Such
a Preconnection will only support calling Listen(), not Initiate(). Calling Listen()
will cause the Transport Services system to register for receiving multicast, such
as issuing an IGMP join <xref target="RFC3376"/> or using MLD for IPV6 <xref target="RFC4604"/>. Any Connections
that are accepted from this Listener are receive-only.</t>
          <t>Similarly, specifying a multicast group address on the Remote Endpoint will indicate that the
resulting connection will be used to send multicast messages, and that the Preconnection will
support Initiate() but not Listen(). Any Connections created this way are send-only.</t>
          <t>A Rendezvous() call on Preconnections containing group addresses results in an
EstablishmentError as described in <xref target="rendezvous"/>.</t>
          <t>See <xref target="multicast-examples"/> for more examples.</t>
        </section>
        <section anchor="ifspec">
          <name>Constraining Interfaces for Endpoints</name>
          <t>Note that this API has multiple ways to constrain and prioritize endpoint candidates based on the network interface:</t>
          <ul spacing="normal">
            <li>Specifying an interface on a RemoteEndpoint qualifies the scope of the remote endpoint, e.g., for link-local addresses.</li>
            <li>Specifying an interface on a LocalEndpoint explicitly binds all candidates derived from this endpoint to use the specified interface.</li>
            <li>Specifying an interface using the <tt>interface</tt> Selection Property (<xref target="prop-interface"/>) or indirectly via the <tt>pvd</tt> Selection Property (<xref target="prop-pvd"/>) influences the selection among the available candidates.</li>
          </ul>
          <t>While specifying an interface on an endpoint restricts the candidates available for connection establishment in the Pre-Establishment Phase, the Selection Properties prioritize and constrain the connection establishment.</t>
        </section>
        <section anchor="endpoint-aliases">
          <name>Endpoint Aliases</name>
          <t>An Endpoint can have an alternative definition when using different protocols.
For example, a server that supports both TLS/TCP and QUIC may be accessible
on two different port numbers depending on which protocol is used.</t>
          <t>To support this, Endpoint Objects can specify "aliases". An Endpoint can have
multiple aliases set.</t>
          <artwork><![CDATA[
RemoteSpecifier.AddAlias(AlternateRemoteSpecifier)
]]></artwork>
          <t>In order to scope an alias to a specific transport protocol, an Endpoint can
specify a protocol identifier.</t>
          <artwork><![CDATA[
RemoteSpecifier.WithProtocol(QUIC)
]]></artwork>
          <t>The following example shows a case where "example.com" has a server
running on port 443, with an alternate port of 8443 for QUIC.</t>
          <artwork><![CDATA[
RemoteSpecifier := NewRemoteEndpoint()
RemoteSpecifier.WithHostname("example.com")
RemoteSpecifier.WithPort(443)

QUICRemoteSpecifier := NewRemoteEndpoint()
QUICRemoteSpecifier.WithHostname("example.com")
QUICRemoteSpecifier.WithPort(8443)
QUICRemoteSpecifier.WithProtocol(QUIC)

RemoteSpecifier.AddAlias(QUICRemoteSpecifier)
]]></artwork>
        </section>
        <section anchor="endpoint-examples">
          <name>Endpoint Examples</name>
          <t>The following examples of Endpoints show common usage patterns.</t>
          <t>Specify a Remote Endpoint using a hostname and service name:</t>
          <artwork><![CDATA[
RemoteSpecifier := NewRemoteEndpoint()
RemoteSpecifier.WithHostname("example.com")
RemoteSpecifier.WithService("https")
]]></artwork>
          <t>Specify a Remote Endpoint using an IPv6 address and remote port:</t>
          <artwork><![CDATA[
RemoteSpecifier := NewRemoteEndpoint()
RemoteSpecifier.WithIPv6Address(2001:db8:4920:e29d:a420:7461:7073:0a)
RemoteSpecifier.WithPort(443)
]]></artwork>
          <t>Specify a Remote Endpoint using an IPv4 address and remote port:</t>
          <artwork><![CDATA[
RemoteSpecifier := NewRemoteEndpoint()
RemoteSpecifier.WithIPv4Address(192.0.2.21)
RemoteSpecifier.WithPort(443)
]]></artwork>
          <t>Specify a Local Endpoint using a local interface name and local port:</t>
          <artwork><![CDATA[
LocalSpecifier := NewLocalEndpoint()
LocalSpecifier.WithInterface("en0")
LocalSpecifier.WithPort(443)
]]></artwork>
          <t>As an alternative to specifying an interface name for the Local Endpoint, an application
can express more fine-grained preferences using the <tt>Interface Instance or Type</tt>
Selection Property, see <xref target="prop-interface"/>. However, if the application specifies Selection
Properties that are inconsistent with the Local Endpoint, this will result in an Error once the
application attempts to open a Connection.</t>
          <t>Specify a Local Endpoint using a STUN server:</t>
          <artwork><![CDATA[
LocalSpecifier := NewLocalEndpoint()
LocalSpecifier.WithStunServer(address, port, credentials)
]]></artwork>
        </section>
        <section anchor="multicast-examples">
          <name>Multicast Examples</name>
          <t>Specify a Local Endpoint using an Any-Source Multicast group to join on a named local interface:</t>
          <artwork><![CDATA[
LocalSpecifier := NewLocalEndpoint()
LocalSpecifier.WithIPv4Address(233.252.0.0)
LocalSpecifier.WithInterface("en0")
]]></artwork>
          <t>Source-Specific Multicast requires setting both a Local and Remote Endpoint:</t>
          <artwork><![CDATA[
LocalSpecifier := NewLocalEndpoint()
LocalSpecifier.WithIPv4Address(232.1.1.1)
LocalSpecifier.WithInterface("en0")

RemoteSpecifier := NewRemoteEndpoint()
RemoteSpecifier.WithIPv4Address(192.0.2.22)
]]></artwork>
          <t>One common pattern for multicast is to both send and receive multicast. For such
cases, an application can set up both a Listener and a Connection. The Listener
is only used to accept Connections that receive inbound multicast. The initiated
Connection is only used to send multicast.</t>
          <artwork><![CDATA[
// Prepare multicast Listener
LocalMulticastSpecifier := NewLocalEndpoint()
LocalMulticastSpecifier.WithIPv4Address(233.252.0.0)
LocalMulticastSpecifier.WithPort(5353)
LocalMulticastSpecifier.WithInterface("en0")

TransportProperties := NewTransportProperties()
TransportProperties.Require(preserve-msg-boundaries)
// Reliable Data Transfer and Preserve Order are Required by default

// Specifying a Remote Endpoint is optional when using Listen()
Preconnection := NewPreconnection(LocalMulticastSpecifier,
                                  TransportProperties,
                                  SecurityParameters)

MulticastListener := Preconnection.Listen()

// Handle inbound messages sent to the multicast group
MulticastListener -> ConnectionReceived<MulticastReceiverConnection>
MulticastReceiverConnection.Receive()
MulticastReceiverConnection -> Received<messageDataRequest, messageContext>

// Prepare Connection to send multicast
LocalSpecifier := NewLocalEndpoint()
LocalSpecifier.WithPort(5353)
LocalSpecifier.WithInterface("en0")
RemoteMulticastSpecifier := NewRemoteEndpoint()
RemoteMulticastSpecifier.WithIPv4Address(233.252.0.0)
RemoteMulticastSpecifier.WithPort(5353)
RemoteMulticastSpecifier.WithInterface("en0")

Preconnection2 := NewPreconnection(LocalSpecifier,
                                   RemoteMulticastSpecifier,
                                   TransportProperties,
                                   SecurityParameters)

// Send outbound messages to the multicast group
MulticastSenderConnection := Preconnection.Initiate()
MulticastSenderConnection.Send(messageData)
]]></artwork>
        </section>
      </section>
      <section anchor="selection-props">
        <name>Specifying Transport Properties</name>
        <t>A Preconnection Object holds properties reflecting the application's
requirements and preferences for the transport. These include Selection
Properties for selecting protocol stacks and paths, as well as Connection
Properties and Message Properties for configuration of the detailed operation
of the selected Protocol Stacks on a per-Connection and Message level.</t>
        <t>The protocol(s) and path(s) selected as candidates during establishment are
determined and configured using these properties. Since there could be paths
over which some transport protocols are unable to operate, or remote endpoints
that support only specific network addresses or transports, transport protocol
selection is necessarily tied to path selection. This may involve choosing
between multiple local interfaces that are connected to different access
networks.</t>
        <t>When additional information (such as Provisioning Domain (PvD) information
<xref target="RFC7556"/>) is available about the networks over which an endpoint can operate,
this can inform the selection between alternate network paths.
Path information can include network segment PMTU, set of supported DSCPs,
expected usage, cost, etc. The usage of this information by the Transport
Services System is generally independent of the specific mechanism/protocol
used to receive the information (e.g. zero-conf, DHCP, or IPv6 RA).</t>
        <t>Most Selection Properties are represented as Preferences, which can
take one of five values:</t>
        <table anchor="tab-pref-levels">
          <name>Selection Property Preference Levels</name>
          <thead>
            <tr>
              <th align="left">Preference</th>
              <th align="left">Effect</th>
            </tr>
          </thead>
          <tbody>
            <tr>
              <td align="left">Require</td>
              <td align="left">Select only protocols/paths providing the property, fail otherwise</td>
            </tr>
            <tr>
              <td align="left">Prefer</td>
              <td align="left">Prefer protocols/paths providing the property, proceed otherwise</td>
            </tr>
            <tr>
              <td align="left">Ignore</td>
              <td align="left">No preference</td>
            </tr>
            <tr>
              <td align="left">Avoid</td>
              <td align="left">Prefer protocols/paths not providing the property, proceed otherwise</td>
            </tr>
            <tr>
              <td align="left">Prohibit</td>
              <td align="left">Select only protocols/paths not providing the property, fail otherwise</td>
            </tr>
          </tbody>
        </table>
        <t>The implementation MUST ensure an outcome that is consistent with all application
requirements expressed using Require and Prohibit. While preferences
expressed using Prefer and Avoid influence protocol and path selection as well,
outcomes can vary given the same Selection Properties, because the available
protocols and paths can differ across systems and contexts. However,
implementations are RECOMMENDED to seek to provide a consistent outcome
to an application, given the same set of Selection Properties.</t>
        <t>Note that application preferences can conflict with each other. For
example, if an application indicates a preference for a specific path by
specifying an interface, but also a preference for a protocol, a situation
might occur in which the preferred protocol is not available on the preferred
path. In such cases, applications can expect properties that determine path
selection to be prioritized over properties that determine protocol selection.
The transport system SHOULD determine the preferred path first, regardless of
protocol preferences. This ordering is chosen to provide consistency across
implementations, based on the fact that it is more common for the use of a
given network path to determine cost to the user (i.e., an interface type
preference might be based on a user's preference to avoid being charged
more for a cellular data plan).</t>
        <t>Selection and Connection Properties, as well as defaults for Message
Properties, can be added to a Preconnection to configure the selection process
and to further configure the eventually selected protocol stack(s). They are
collected into a TransportProperties object to be passed into a Preconnection
object:</t>
        <artwork><![CDATA[
TransportProperties := NewTransportProperties()
]]></artwork>
        <t>Individual properties are then set on the TransportProperties Object.
Setting a Transport Property to a value overrides the previous value of this Transport Property.</t>
        <artwork><![CDATA[
TransportProperties.Set(property, value)
]]></artwork>
        <t>To aid readability, implementations MAY provide additional convenience functions to simplify use of Selection Properties: see <xref target="preference-conv"/> for examples.
In addition, implementations MAY provide a mechanism to create TransportProperties objects that
are preconfigured for common use cases, as outlined in <xref target="property-profiles"/>.</t>
        <t>Transport Properties for an established connection can be queried via the Connection object, as outlined in <xref target="introspection"/>.</t>
        <t>A Connection gets its Transport Properties either by being explicitly configured
via a Preconnection, by configuration after establishment, or by inheriting them
from an antecedent via cloning; see <xref target="groups"/> for more.</t>
        <t><xref target="connection-props"/> provides a list of Connection Properties, while Selection
Properties are listed in the subsections below. Selection Properties are
only considered during establishment, and can not be changed after a Connection
is established. After a Connection is established, Selection Properties can only
be read to check the properties used by the Connection. Upon reading, the
Preference type of a Selection Property changes into Boolean, where <tt>true</tt> means
that the selected Protocol Stack supports the feature or uses the path associated with the Selection Property, and <tt>false</tt> means that the Protocol Stack does not support the feature or use the path. Implementations
of Transport Services systems may alternatively use the two Preference values <tt>Require</tt>
and <tt>Prohibit</tt> to represent <tt>true</tt> and <tt>false</tt>, respectively.</t>
        <t>An implementation of the Transport Services API must provide sensible defaults for Selection
Properties. The default values for each property below represent a
configuration that can be implemented over TCP. If these default values are used
and TCP is not supported by a Transport Services system, then an application using the
default set of Properties might not succeed in establishing a connection. Using
the same default values for independent Transport Services implementations can be beneficial
when applications are ported between different implementations/platforms, even if this
default could lead to a connection failure when TCP is not available. If default
values other than those suggested below are used, it is RECOMMENDED to clearly
document any differences.</t>
        <section anchor="prop-reliable">
          <name>Reliable Data Transfer (Connection)</name>
          <dl>
            <dt>Name:</dt>
            <dd>
              <t>reliability</t>
            </dd>
            <dt>Type:</dt>
            <dd>
              <t>Preference</t>
            </dd>
            <dt>Default:</dt>
            <dd>
              <t>Require</t>
            </dd>
          </dl>
          <t>This property specifies whether the application needs to use a transport
protocol that ensures that all data is received at the Remote Endpoint without
corruption. When reliable data transfer is enabled, this
also entails being notified when a Connection is closed or aborted.</t>
        </section>
        <section anchor="prop-boundaries">
          <name>Preservation of Message Boundaries</name>
          <dl>
            <dt>Name:</dt>
            <dd>
              <t>preserveMsgBoundaries</t>
            </dd>
            <dt>Type:</dt>
            <dd>
              <t>Preference</t>
            </dd>
            <dt>Default:</dt>
            <dd>
              <t>Ignore</t>
            </dd>
          </dl>
          <t>This property specifies whether the application needs or prefers to use a transport
protocol that preserves message boundaries.</t>
        </section>
        <section anchor="prop-partially-reliable">
          <name>Configure Per-Message Reliability</name>
          <dl>
            <dt>Name:</dt>
            <dd>
              <t>perMsgReliability</t>
            </dd>
            <dt>Type:</dt>
            <dd>
              <t>Preference</t>
            </dd>
            <dt>Default:</dt>
            <dd>
              <t>Ignore</t>
            </dd>
          </dl>
          <t>This property specifies whether an application considers it useful to specify different
reliability requirements for individual Messages in a Connection.</t>
        </section>
        <section anchor="prop-ordering">
          <name>Preservation of Data Ordering</name>
          <dl>
            <dt>Name:</dt>
            <dd>
              <t>preserveOrder</t>
            </dd>
            <dt>Type:</dt>
            <dd>
              <t>Preference</t>
            </dd>
            <dt>Default:</dt>
            <dd>
              <t>Require</t>
            </dd>
          </dl>
          <t>This property specifies whether the application wishes to use a transport
protocol that can ensure that data is received by the application on the other
end in the same order as it was sent.</t>
        </section>
        <section anchor="prop-0rtt">
          <name>Use 0-RTT Session Establishment with a Safely Replayable Message</name>
          <dl>
            <dt>Name:</dt>
            <dd>
              <t>zeroRttMsg</t>
            </dd>
            <dt>Type:</dt>
            <dd>
              <t>Preference</t>
            </dd>
            <dt>Default:</dt>
            <dd>
              <t>Ignore</t>
            </dd>
          </dl>
          <t>This property specifies whether an application would like to supply a Message to
the transport protocol before Connection establishment that will then be
reliably transferred to the other side before or during Connection
establishment. This Message can potentially be received multiple times (i.e.,
multiple copies of the message data may be passed to the Remote Endpoint).
See also <xref target="msg-safelyreplayable"/>.</t>
        </section>
        <section anchor="prop-multistream">
          <name>Multistream Connections in Group</name>
          <dl>
            <dt>Name:</dt>
            <dd>
              <t>multistreaming</t>
            </dd>
            <dt>Type:</dt>
            <dd>
              <t>Preference</t>
            </dd>
            <dt>Default:</dt>
            <dd>
              <t>Prefer</t>
            </dd>
          </dl>
          <t>This property specifies that the application would prefer multiple Connections
within a Connection Group to be provided by streams of a single underlying
transport connection where possible.</t>
        </section>
        <section anchor="prop-checksum-control-send">
          <name>Full Checksum Coverage on Sending</name>
          <dl>
            <dt>Name:</dt>
            <dd>
              <t>fullChecksumSend</t>
            </dd>
            <dt>Type:</dt>
            <dd>
              <t>Preference</t>
            </dd>
            <dt>Default:</dt>
            <dd>
              <t>Require</t>
            </dd>
          </dl>
          <t>This property specifies the application's need for protection against
corruption for all data transmitted on this Connection. Disabling this property could enable
later control of the sender checksum coverage (see <xref target="msg-checksum"/>).</t>
        </section>
        <section anchor="prop-checksum-control-receive">
          <name>Full Checksum Coverage on Receiving</name>
          <dl>
            <dt>Name:</dt>
            <dd>
              <t>fullChecksumRecv</t>
            </dd>
            <dt>Type:</dt>
            <dd>
              <t>Preference</t>
            </dd>
            <dt>Default:</dt>
            <dd>
              <t>Require</t>
            </dd>
          </dl>
          <t>This property specifies the application's need for protection against
corruption for all data received on this Connection. Disabling this property could enable
later control of the required minimum receiver checksum coverage (see <xref target="conn-recv-checksum"/>).</t>
        </section>
        <section anchor="prop-cc">
          <name>Congestion control</name>
          <dl>
            <dt>Name:</dt>
            <dd>
              <t>congestionControl</t>
            </dd>
            <dt>Type:</dt>
            <dd>
              <t>Preference</t>
            </dd>
            <dt>Default:</dt>
            <dd>
              <t>Require</t>
            </dd>
          </dl>
          <t>This property specifies whether the application would like the Connection to be
congestion controlled or not. Note that if a Connection is not congestion
controlled, an application using such a Connection SHOULD itself perform
congestion control in accordance with <xref target="RFC2914"/> or use a circuit breaker in
accordance with <xref target="RFC8084"/>, whichever is appropriate. Also note that reliability
is usually combined with congestion control in protocol implementations,
rendering "reliable but not congestion controlled" a request that is unlikely to
succeed. If the Connection is congestion controlled, performing additional congestion control
in the application can have negative performance implications.</t>
        </section>
        <section anchor="keep-alive">
          <name>Keep alive</name>
          <dl>
            <dt>Name:</dt>
            <dd>
              <t>keepAlive</t>
            </dd>
            <dt>Type:</dt>
            <dd>
              <t>Preference</t>
            </dd>
            <dt>Default:</dt>
            <dd>
              <t>Ignore</t>
            </dd>
          </dl>
          <t>This property specifies whether the application would like the Connection to send
keep-alive packets or not. Note that if a Connection determines that keep-alive
packets are being sent, the applicaton should itself avoid generating additional keep alive
messages. Note that when supported, the system will use the default period
for generation of the keep alive-packets. (See also <xref target="keep-alive-timeout"/>).</t>
        </section>
        <section anchor="prop-interface">
          <name>Interface Instance or Type</name>
          <dl>
            <dt>Name:</dt>
            <dd>
              <t>interface</t>
            </dd>
            <dt>Type:</dt>
            <dd>
              <t>Collection of (Preference, Enumeration)</t>
            </dd>
            <dt>Default:</dt>
            <dd>
              <t>Empty (not setting a preference for any interface)</t>
            </dd>
          </dl>
          <t>This property allows the application to select any specific network interfaces
or categories of interfaces it wants to <tt>Require</tt>, <tt>Prohibit</tt>, <tt>Prefer</tt>, or
<tt>Avoid</tt>. Note that marking a specific interface as <tt>Require</tt> strictly limits path
selection to that single interface, and often leads to less flexible and resilient
connection establishment.</t>
          <t>In contrast to other Selection Properties, this property is a tuple of an
(Enumerated) interface identifier and a preference, and can either be
implemented directly as such, or for making one preference available for each
interface and interface type available on the system.</t>
          <t>The set of valid interface types is implementation- and system-specific. For
example, on a mobile device, there may be <tt>Wi-Fi</tt> and <tt>Cellular</tt> interface types
available; whereas on a desktop computer,  <tt>Wi-Fi</tt> and <tt>Wired
Ethernet</tt> interface types might be available. An implementation should provide all types
that are supported on the local system, to allow
applications to be written generically. For example, if a single implementation
is used on both mobile devices and desktop devices, it should define the
<tt>Cellular</tt> interface type for both systems, since an application might wish to
always prohibit cellular.</t>
          <t>The set of interface types is expected to change over time as new access
technologies become available. The taxonomy of interface types on a given
Transport Services system is implementation-specific.</t>
          <t>Interface types should not be treated as a proxy for properties of interfaces
such as metered or unmetered network access. If an application needs to prohibit
metered interfaces, this should be specified via Provisioning Domain attributes
(see <xref target="prop-pvd"/>) or another specific property.</t>
          <t>Note that this property is not used to specify an interface scope for a particular endpoint. <xref target="ifspec"/> provides details about how to qualify endpoint candidates on a per-interface basis.</t>
        </section>
        <section anchor="prop-pvd">
          <name>Provisioning Domain Instance or Type</name>
          <dl>
            <dt>Name:</dt>
            <dd>
              <t>pvd</t>
            </dd>
            <dt>Type:</dt>
            <dd>
              <t>Collection of (Preference, Enumeration)</t>
            </dd>
            <dt>Default:</dt>
            <dd>
              <t>Empty (not setting a preference for any PvD)</t>
            </dd>
          </dl>
          <t>Similar to <tt>interface</tt> (see <xref target="prop-interface"/>), this property
allows the application to control path selection by selecting which specific
Provisioning Domain (PvD) or categories of PVDs it wants to
<tt>Require</tt>, <tt>Prohibit</tt>, <tt>Prefer</tt>, or <tt>Avoid</tt>. Provisioning Domains define
consistent sets of network properties that may be more specific than network
interfaces <xref target="RFC7556"/>.</t>
          <t>As with interface instances and types, this property is a tuple of an (Enumerated)
PvD identifier and a preference, and can either be implemented directly as such,
or for making one preference available for each interface and interface type
available on the system.</t>
          <t>The identification of a specific PvD is
implementation- and system-specific, because there is currently no portable standard
format for a PvD identifier. For example, this identifier might be a string name
or an integer. As with requiring specific interfaces, requiring a specific PvD
strictly limits the path selection.</t>
          <t>Categories or types of PvDs are also defined to be implementation- and
system-specific. These can be useful to identify a service that is provided by a
PvD. For example, if an application wants to use a PvD that provides a
Voice-Over-IP service on a Cellular network, it can use the relevant PvD type to
require a PvD that provides this service, without needing to look up a
particular instance. While this does restrict path selection, it is broader than
requiring specific PvD instances or interface instances, and should be preferred
over these options.</t>
        </section>
        <section anchor="use-temporary-local-address">
          <name>Use Temporary Local Address</name>
          <dl>
            <dt>Name:</dt>
            <dd>
              <t>useTemporaryLocalAddress</t>
            </dd>
            <dt>Type:</dt>
            <dd>
              <t>Preference</t>
            </dd>
            <dt>Default:</dt>
            <dd>
              <t>Avoid for Listeners and Rendezvous Connections. Prefer for other Connections.</t>
            </dd>
          </dl>
          <t>This property allows the application to express a preference for the use of
temporary local addresses, sometimes called "privacy" addresses <xref target="RFC8981"/>.
Temporary addresses are generally used to prevent linking connections over time
when a stable address, sometimes called "permanent" address, is not needed.
There are some caveats to note when specifying this property. First, if an
application Requires the use of temporary addresses, the resulting Connection
cannot use IPv4, because temporary addresses do not exist in IPv4. Second,
temporary local addresses might involve trading off privacy for performance.
For instance, temporary addresses can interfere with resumption mechanisms
that some protocols rely on to reduce initial latency.</t>
        </section>
        <section anchor="multipath-mode">
          <name>Multipath Transport</name>
          <dl>
            <dt>Name:</dt>
            <dd>
              <t>multipath</t>
            </dd>
            <dt>Type:</dt>
            <dd>
              <t>Enumeration</t>
            </dd>
            <dt>Default:</dt>
            <dd>
              <t>Disabled for connections created through initiate and rendezvous, Passive for listeners</t>
            </dd>
          </dl>
          <t>This property specifies whether and how applications want to take advantage of
transferring data across multiple paths between the same end hosts.
Using multiple paths allows connections to migrate between interfaces or aggregate bandwidth
as availability and performance properties change.
Possible values are:</t>
          <dl>
            <dt>Disabled:</dt>
            <dd>
              <t>The connection will not use multiple paths once established, even if the chosen transport supports using multiple paths.</t>
            </dd>
            <dt>Active:</dt>
            <dd>
              <t>The connection will negotiate the use of multiple paths if the chosen transport supports this.</t>
            </dd>
            <dt>Passive:</dt>
            <dd>
              <t>The connection will support the use of multiple paths if the Remote Endpoint requests it.</t>
            </dd>
          </dl>
          <t>The policy for using multiple paths is specified using the separate <tt>multipathPolicy</tt> property, see <xref target="multipath-policy"/> below.
To enable the peer endpoint to initiate additional paths towards a local address other than the one initially used, it is necessary to set the <tt>advertisesAltaddr</tt> property (see <xref target="altaddr"/> below).</t>
          <t>Setting this property to "Active", can have privacy implications: It enables the transport to establish connectivity using alternate paths that might result in users being linkable across the multiple paths, even if the <tt>advertisesAltaddr</tt> property (see <xref target="altaddr"/> below) is set to false.</t>
          <t>Note that Multipath Transport has no corresponding Selection Property of type Preference.
Enumeration values other than "Disabled" are interpreted as a preference for choosing protocols that can make use of multiple paths.
The "Disabled" value implies a requirement not to use multiple paths in parallel but does not prevent choosing a protocol that is capable of using multiple paths, e.g., it does not prevent choosing TCP, but prevents sending the <tt>MP_CAPABLE</tt> option in the TCP handshake.</t>
        </section>
        <section anchor="altaddr">
          <name>Advertisement of Alternative Addresses</name>
          <dl>
            <dt>Name:</dt>
            <dd>
              <t>advertisesAltaddr</t>
            </dd>
            <dt>Type:</dt>
            <dd>
              <t>Boolean</t>
            </dd>
            <dt>Default:</dt>
            <dd>
              <t>False</t>
            </dd>
          </dl>
          <t>This property specifies whether alternative addresses, e.g., of other interfaces, should be advertised to the
peer endpoint by the protocol stack. Advertising these addresses enables the peer-endpoint to establish additional connectivity, e.g., for connection migration or using multiple paths.</t>
          <t>Note that this can have privacy implications because it might result in users being linkable across the multiple paths.
Also, note that setting this to false does not prevent the local Transport Services system from <em>establishing</em> connectivity using alternate paths (see <xref target="multipath-mode"/> above); it only prevents <em>proactive advertisement</em> of addresses.</t>
        </section>
        <section anchor="direction-of-communication">
          <name>Direction of communication</name>
          <dl>
            <dt>Name:</dt>
            <dd>
              <t>direction</t>
            </dd>
            <dt>Type:</dt>
            <dd>
              <t>Enumeration</t>
            </dd>
            <dt>Default:</dt>
            <dd>
              <t>Bidirectional</t>
            </dd>
          </dl>
          <t>This property specifies whether an application wants to use the connection for sending and/or receiving data.  Possible values are:</t>
          <dl>
            <dt>Bidirectional:</dt>
            <dd>
              <t>The connection must support sending and receiving data</t>
            </dd>
            <dt>Unidirectional send:</dt>
            <dd>
              <t>The connection must support sending data, and the application cannot use the connection to receive any data</t>
            </dd>
            <dt>Unidirectional receive:</dt>
            <dd>
              <t>The connection must support receiving data, and the application cannot use the connection to send any data</t>
            </dd>
          </dl>
          <t>Since unidirectional communication can be
supported by transports offering bidirectional communication, specifying
unidirectional communication may cause a transport stack that supports
bidirectional communication to be selected.</t>
        </section>
        <section anchor="prop-soft-error">
          <name>Notification of ICMP soft error message arrival</name>
          <dl>
            <dt>Name:</dt>
            <dd>
              <t>softErrorNotify</t>
            </dd>
            <dt>Type:</dt>
            <dd>
              <t>Preference</t>
            </dd>
            <dt>Default:</dt>
            <dd>
              <t>Ignore</t>
            </dd>
          </dl>
          <t>This property specifies whether an application considers it useful to be
informed when an ICMP error message arrives that does not force termination of a
connection. When set to true, received ICMP errors are available as
SoftErrors, see <xref target="soft-errors"/>. Note that even if a protocol supporting this property is selected,
not all ICMP errors will necessarily be delivered, so applications cannot rely
upon receiving them <xref target="RFC8085"/>.</t>
        </section>
        <section anchor="active-read-before-send">
          <name>Initiating side is not the first to write</name>
          <dl>
            <dt>Name:</dt>
            <dd>
              <t>activeReadBeforeSend</t>
            </dd>
            <dt>Type:</dt>
            <dd>
              <t>Preference</t>
            </dd>
            <dt>Default:</dt>
            <dd>
              <t>Ignore</t>
            </dd>
          </dl>
          <t>The most common client-server communication pattern involves the
client actively opening a connection, then sending data to the server. The
server listens (passive open), reads, and then answers. This property
specifies whether an application wants to diverge from this pattern -- either
by actively opening with Initiate(), immediately followed by reading, or passively opening with Listen(),
immediately followed by writing. This property is ignored when establishing
connections using Rendezvous().
Requiring this property limits the choice of mappings to underlying protocols,
which can reduce
efficiency. For example, it prevents the Transport Services system from mapping
Connections to SCTP streams, where
the first transmitted data takes the role of an active open signal <xref target="I-D.ietf-taps-impl"/>.</t>
        </section>
      </section>
      <section anchor="security-parameters">
        <name>Specifying Security Parameters and Callbacks</name>
        <t>Most security parameters, e.g., TLS ciphersuites, local identity and private key, etc.,
may be configured statically. Others are dynamically configured during connection establishment.
Security parameters and callbacks are partitioned based on their place in the lifetime
of connection establishment. Similar to Transport Properties, both parameters and callbacks
are inherited during cloning (see <xref target="groups"/>).</t>
        <section anchor="specifying-security-parameters-on-a-pre-connection">
          <name>Specifying Security Parameters on a Pre-Connection</name>
          <t>Common security parameters such as TLS ciphersuites are known to implementations. Clients should
use common safe defaults for these values whenever possible. However, as discussed in
<xref target="RFC8922"/>, many transport security protocols require specific
security parameters and constraints from the client at the time of configuration and
actively during a handshake. These configuration parameters need to be specified in the
pre-connection phase and are created as follows:</t>
          <artwork><![CDATA[
SecurityParameters := NewSecurityParameters()
]]></artwork>
          <t>Security configuration parameters and sample usage follow:</t>
          <ul spacing="normal">
            <li>Local identity and private keys: Used to perform private key operations and prove one's
identity to the Remote Endpoint. (Note, if private keys are not available, e.g., since they are
stored in hardware security modules (HSMs), handshake callbacks must be used. See below for details.)</li>
          </ul>
          <artwork><![CDATA[
SecurityParameters.Set(identity, myIdentity)
SecurityParameters.Set(key-pair, myPrivateKey, myPublicKey)
]]></artwork>
          <ul spacing="normal">
            <li>Supported algorithms: Used to restrict what parameters are used by underlying transport security protocols.
When not specified, these algorithms should use known and safe defaults for the system. Parameters include:
ciphersuites, supported groups, and signature algorithms. These parameters take a collection of supported algorithms as parameter.</li>
          </ul>
          <artwork><![CDATA[
SecurityParameters.Set(supported-group, "secp256r1")
SecurityParameters.Set(ciphersuite, "TLS_AES_128_GCM_SHA256")
SecurityParameters.Set(signature-algorithm, "ecdsa_secp256r1_sha256")
]]></artwork>
          <ul spacing="normal">
            <li>Pre-Shared Key import: Used to install pre-shared keying material established
out-of-band. Each pre-shared keying material is associated with some identity that typically identifies
its use or has some protocol-specific meaning to the Remote Endpoint.</li>
          </ul>
          <artwork><![CDATA[
SecurityParameters.Set(pre-shared-key, key, identity)
]]></artwork>
          <ul spacing="normal">
            <li>Session cache management: Used to tune session cache capacity, lifetime, and
other policies.</li>
          </ul>
          <artwork><![CDATA[
SecurityParameters.Set(max-cached-sessions, 16)
SecurityParameters.Set(cached-session-lifetime-seconds, 3600)
]]></artwork>
          <t>Connections that use Transport Services SHOULD use security in general. However, for
compatibility with endpoints that do not support transport security protocols (such
as a TCP endpoint that does not support TLS), applications can initialize their
security parameters to indicate that security can be disabled, or can be opportunistic.
If security is disabled, the Transport Services system will not attempt to add
transport security automatically. If security is opportunistic, it will allow
Connections without transport security, but will still attempt to use security if
available.</t>
          <artwork><![CDATA[
SecurityParameters := NewDisabledSecurityParameters()

SecurityParameters := NewOpportunisticSecurityParameters()
]]></artwork>
          <t>Representation of security parameters in implementations should parallel
that chosen for Transport Property names as sugggested in <xref target="scope-of-interface-defn"/>.</t>
        </section>
        <section anchor="connection-establishment-callbacks">
          <name>Connection Establishment Callbacks</name>
          <t>Security decisions, especially pertaining to trust, are not static. Once configured,
parameters may also be supplied during connection establishment. These are best
handled as client-provided callbacks.
Callbacks block the progress of the connection establishment, which distinguishes them from other Events in the transport system. How callbacks and events are implemented is specific to each implementation.
Security handshake callbacks that may be invoked during connection establishment include:</t>
          <ul spacing="normal">
            <li>Trust verification callback: Invoked when a Remote Endpoint's trust must be verified before the
handshake protocol can continue. For example, the application could verify an X.509 certificate
as described in <xref target="RFC5280"/>.</li>
          </ul>
          <artwork><![CDATA[
TrustCallback := NewCallback({
  // Handle trust, return the result
})
SecurityParameters.SetTrustVerificationCallback(trustCallback)
]]></artwork>
          <ul spacing="normal">
            <li>Identity challenge callback: Invoked when a private key operation is required, e.g., when
local authentication is requested by a Remote Endpoint.</li>
          </ul>
          <artwork><![CDATA[
ChallengeCallback := NewCallback({
  // Handle challenge
})
SecurityParameters.SetIdentityChallengeCallback(challengeCallback)
]]></artwork>
        </section>
      </section>
    </section>
    <section anchor="establishment">
      <name>Establishing Connections</name>
      <t>Before a Connection can be used for data transfer, it needs to be established.
Establishment ends the pre-establishment phase; all transport properties and
cryptographic parameter specification must be complete before establishment,
as these will be used to select candidate Paths and Protocol Stacks
for the Connection. Establishment may be active, using the Initiate() Action;
passive, using the Listen() Action; or simultaneous for peer-to-peer, using
the Rendezvous() Action. These Actions are described in the subsections below.</t>
      <section anchor="initiate">
        <name>Active Open: Initiate</name>
        <t>Active open is the Action of establishing a Connection to a Remote Endpoint presumed
to be listening for incoming Connection requests. Active open is used by clients in
client-server interactions. Active open is supported by the Transport Services API through the
Initiate Action:</t>
        <artwork><![CDATA[
Connection := Preconnection.Initiate(timeout?)
]]></artwork>
        <t>The timeout parameter specifies how long to wait before aborting Active open.
Before calling Initiate, the caller must have populated a Preconnection
Object with a Remote Endpoint specifier, optionally a Local Endpoint
specifier (if not specified, the system will attempt to determine a
suitable Local Endpoint), as well as all properties
necessary for candidate selection.</t>
        <t>The Initiate() Action returns a Connection object. Once Initiate() has been
called, any changes to the Preconnection MUST NOT have any effect on the
Connection. However, the Preconnection can be reused, e.g., to Initiate
another Connection.</t>
        <t>Once Initiate is called, the candidate Protocol Stack(s) may cause one or more
candidate transport-layer connections to be created to the specified Remote
Endpoint. The caller may immediately begin sending Messages on the Connection
(see <xref target="sending"/>) after calling Initiate(); note that any data marked as "safely replayable" that is sent
while the Connection is being established may be sent multiple times or on
multiple candidates.</t>
        <t>The following Events may be sent by the Connection after Initiate() is called:</t>
        <artwork><![CDATA[
Connection -> Ready<>
]]></artwork>
        <t>The Ready Event occurs after Initiate has established a transport-layer
connection on at least one usable candidate Protocol Stack over at least one
candidate Path. No Receive Events (see <xref target="receiving"/>) will occur before
the Ready Event for Connections established using Initiate.</t>
        <artwork><![CDATA[
Connection -> EstablishmentError<reason?>
]]></artwork>
        <t>An EstablishmentError occurs either when the set of transport properties and security
parameters cannot be fulfilled on a Connection for initiation (e.g., the set of
available Paths and/or Protocol Stacks meeting the constraints is empty) or
reconciled with the Local and/or Remote Endpoints; when the remote specifier
cannot be resolved; or when no transport-layer connection can be established to
the Remote Endpoint (e.g., because the Remote Endpoint is not accepting
connections, the application is prohibited from opening a Connection by the
operating system, or the establishment attempt has timed out for any other reason).</t>
        <t>Connection establishment
and transmission of the first message can be combined in a single action <xref target="initiate-and-send"/>.</t>
      </section>
      <section anchor="listen">
        <name>Passive Open: Listen</name>
        <t>Passive open is the Action of waiting for Connections from Remote Endpoints,
commonly used by servers in client-server interactions. Passive open is
supported by the Transport Services API through the Listen Action and returns a Listener object:</t>
        <artwork><![CDATA[
Listener := Preconnection.Listen()
]]></artwork>
        <t>Before calling Listen, the caller must have initialized the Preconnection
during the pre-establishment phase with a Local Endpoint specifier, as well
as all properties necessary for Protocol Stack selection. A Remote Endpoint
may optionally be specified, to constrain what Connections are accepted.</t>
        <t>The Listen() Action returns a Listener object. Once Listen() has been called,
any changes to the Preconnection MUST NOT have any effect on the Listener. The
Preconnection can be disposed of or reused, e.g., to create another Listener.</t>
        <artwork><![CDATA[
Listener.Stop()
]]></artwork>
        <t>Listening continues until the global context shuts down, or until the Stop
action is performed on the Listener object.</t>
        <artwork><![CDATA[
Listener -> ConnectionReceived<Connection>
]]></artwork>
        <t>The ConnectionReceived Event occurs when a Remote Endpoint has established a
transport-layer connection to this Listener (for Connection-oriented
transport protocols), or when the first Message has been received from the
Remote Endpoint (for Connectionless protocols), causing a new Connection to be
created. The resulting Connection is contained within the ConnectionReceived
Event, and is ready to use as soon as it is passed to the application via the
event.</t>
        <artwork><![CDATA[
Listener.SetNewConnectionLimit(value)
]]></artwork>
        <t>If the caller wants to rate-limit the number of inbound Connections that will be delivered,
it can set a cap using SetNewConnectionLimit(). This mechanism allows a server to
protect itself from being drained of resources. Each time a new Connection is delivered
by the ConnectionReceived Event, the value is automatically decremented. Once the
value reaches zero, no further Connections will be delivered until the caller sets the
limit to a higher value. By default, this value is Infinite. The caller is also able to reset
the value to Infinite at any point.</t>
        <artwork><![CDATA[
Listener -> EstablishmentError<reason?>
]]></artwork>
        <t>An EstablishmentError occurs either when the Properties and security parameters of the Preconnection cannot be fulfilled for listening or cannot be reconciled with the Local Endpoint (and/or Remote Endpoint, if specified), when the Local Endpoint (or Remote Endpoint, if specified) cannot
be resolved, or when the application is prohibited from listening by policy.</t>
        <artwork><![CDATA[
Listener -> Stopped<>
]]></artwork>
        <t>A Stopped Event occurs after the Listener has stopped listening.</t>
      </section>
      <section anchor="rendezvous">
        <name>Peer-to-Peer Establishment: Rendezvous</name>
        <t>Simultaneous peer-to-peer Connection establishment is supported by the
Rendezvous() Action:</t>
        <artwork><![CDATA[
Preconnection.Rendezvous()
]]></artwork>
        <t>A Preconnection Object used in a Rendezvous() MUST have both the
Local Endpoint candidates and the Remote Endpoint candidates specified,
along with the Transport Properties and security parameters needed for
Protocol Stack selection, before the Rendezvous() Action is initiated.</t>
        <t>The Rendezvous() Action listens on the Local Endpoint
candidates for an incoming Connection from the Remote Endpoint candidates,
while also simultaneously trying to establish a Connection from the Local
Endpoint candidates to the Remote Endpoint candidates.</t>
        <t>If there are multiple Local Endpoints or Remote Endpoints configured, then
initiating a Rendezvous() action will systematically probe the reachability
of those endpoint candidates following an approach such as that used in
Interactive Connectivity Establishment (ICE) <xref target="RFC8445"/>.</t>
        <t>If the endpoints are suspected to be behind a NAT, Rendezvous() can be
initiated using Local Endpoints that support a method of discovering NAT
bindings such as Session Traversal Utilities for NAT (STUN) <xref target="RFC8489"/> or
Traversal Using Relays around NAT (TURN) <xref target="RFC8656"/>.  In this case, the
Local Endpoint will resolve to a mixture of local and server reflexive
addresses. The Resolve() action on the Preconnection can be used to
discover these bindings:</t>
        <artwork><![CDATA[
[]LocalEndpoint, []RemoteEndpoint := Preconnection.Resolve()
]]></artwork>
        <t>The Resolve() call returns lists of Local Endpoints and Remote Endpoints,
that represent the concrete addresses, local and server reflexive, on which
a Rendezvous() for the Preconnection will listen for incoming Connections,
and to which it will attempt to establish connections.</t>
        <t>Note that the set of LocalEndpoints returned by Resolve() might or might not
contain information about all possible local interfaces; it is valid only
for a Rendezvous happening at the same time as the resolution. Care should
be taken in using these values in any other context.</t>
        <t>An application that uses Rendezvous() to establish a peer-to-peer connection
in the presence of NATs will configure the Preconnection object with at least
one a Local Endpoint that supports NAT binding discovery. It will then Resolve()
the Preconnection, and pass the resulting list of Local Endpoint candidates to
the peer via a signalling protocol, for example as part of an ICE <xref target="RFC5245"/>
exchange within SIP <xref target="RFC3261"/> or WebRTC <xref target="RFC7478"/>.  The peer will then,
via the same signalling channel, return the Remote Endpoint candidates.
The set of Remote Endpoint candidates are then configured onto the Preconnection:</t>
        <artwork><![CDATA[
Preconnection.AddRemote([]RemoteEndpoint)
]]></artwork>
        <t>The Rendezvous() Action can be initiated once both the Local Endpoint
candidates and the Remote Endpoint candidates retrieved from the peer via
the signalling channel have been added to the Preconnection.</t>
        <t>If successful, the Rendezvous() Action returns a Connection object via a
RendezvousDone&lt;&gt; Event:</t>
        <artwork><![CDATA[
Preconnection -> RendezvousDone<Connection>
]]></artwork>
        <t>The RendezvousDone&lt;&gt; Event occurs when a Connection is established with the
Remote Endpoint. For Connection-oriented transports, this occurs when the
transport-layer connection is established; for Connectionless transports,
it occurs when the first Message is received from the Remote Endpoint. The
resulting Connection is contained within the RendezvousDone&lt;&gt; Event, and is
ready to use as soon as it is passed to the application via the Event.
Changes made to a Preconnection after Rendezvous() has been called do
not have any effect on existing Connections.</t>
        <t>An EstablishmentError occurs either when the Properties and Security
Parameters of the Preconnection cannot be fulfilled for rendezvous or
cannot be reconciled with the Local and/or Remote Endpoints, when the Local
Endpoint or Remote Endpoint cannot be resolved, when no transport-layer
connection can be established to the Remote Endpoint, or when the
application is prohibited from rendezvous by policy:</t>
        <artwork><![CDATA[
Preconnection -> EstablishmentError<reason?>
]]></artwork>
      </section>
      <section anchor="groups">
        <name>Connection Groups</name>
        <t>Connection Groups can be created using the Clone Action:</t>
        <artwork><![CDATA[
Connection := Connection.Clone(framer?, connectionProperties?)
]]></artwork>
        <t>Calling Clone on a Connection yields a Connection Group containing two Connections: the parent
Connection on which Clone was called, and a resulting cloned Connection.
The new Connection is actively openend, and it will send a Ready Event or an EstablishmentError Event.
Calling Clone on any of these Connections adds another Connection to
the Connection Group. Connections in a Connection Group share all
Connection Properties except <tt>connPriority</tt> (see <xref target="conn-priority"/>),
and these Connection Properties are entangled: Changing one of the
Connection Properties on one Connection in the Connection Group
automatically changes the Connection Property for all others. For example, changing
<tt>connTimeout</tt> (see
<xref target="conn-timeout"/>) on one Connection in a Connection Group will automatically
make the same change to this Connection Property for all other Connections in the Connection Group.
Like all other Properties, <tt>connPriority</tt> is copied
to the new Connection when calling Clone(), but in this case, a later change to the
<tt>connPriority</tt> on one Connection does not change it on the
other Connections in the same Connection Group.</t>
        <t>The optional <tt>connectionProperties</tt> parameter allows passing
Transport Properties that control the behavior of the underlying stream or connection to be created, e.g., protocol-specific properties to request specific stream IDs for SCTP or QUIC.</t>
        <t>Message Properties set on a Connection also apply only to that Connection.</t>
        <t>A new Connection created by Clone can have a Message Framer assigned via the optional
<tt>framer</tt> parameter of the Clone Action. If this parameter is not supplied, the
stack of Message Framers associated with a Connection is copied to
the cloned Connection when calling Clone. Then, a cloned Connection
has the same stack of Message Framers as the Connection from which they
are Cloned, but these Framers may internally maintain per-Connection state.</t>
        <t>It is also possible to check which Connections belong to the same Connection Group.
Calling GroupedConnections() on a specific Connection returns a set of all Connections
in the same group.</t>
        <artwork><![CDATA[
[]Connection := Connection.GroupedConnections()
]]></artwork>
        <t>Connections will belong to the same group if the application previously called Clone.
Passive Connections can also be added to the same group -- e.g., when a Listener
receives a new Connection that is just a new stream of an already active multi-streaming
protocol instance.</t>
        <t>If the underlying protocol supports multi-streaming, it is natural to use this
functionality to implement Clone. In that case, Connections in a Connection Group are
multiplexed together, giving them similar treatment not only inside endpoints,
but also across the end-to-end Internet path.</t>
        <t>Note that calling Clone() can result in on-the-wire signaling, e.g., to open a new
transport connection, depending on the underlying Protocol Stack. When Clone() leads to
the opening of multiple such connections,
the Transport Services system will ensure consistency of
Connection Properties by uniformly applying them to all underlying connections
in a group. Even in such a case, there are possibilities for a Transport Services system
to implement prioritization within a Connection Group <xref target="TCP-COUPLING"/> <xref target="RFC8699"/>.</t>
        <t>Attempts to clone a Connection can result in a CloneError:</t>
        <artwork><![CDATA[
Connection -> CloneError<reason?>
]]></artwork>
        <t>The <tt>connPriority</tt> Connection Property operates on Connections in a Connection Group
using the same approach as in <xref target="msg-priority"/>: when allocating available network
capacity among Connections in a Connection Group, sends on Connections with
higher Priority values will be prioritized over sends on Connections that have
lower Priority values. Capacity will be shared among these Connections according to
the <tt>connScheduler</tt> property (<xref target="conn-scheduler"/>).
See <xref target="priority-in-taps"/> for more.</t>
      </section>
      <section anchor="add-endpoints">
        <name>Adding and Removing Endpoints on a Connection</name>
        <t>Transport protocols that are explicitly multipath aware are expected to automatically
manage the set of Remote Endpoints that they are communicating with, and the paths to
those endpoints. A <tt>PathChange&lt;&gt;</tt> event, described in <xref target="conn-path-change"/>, will be
generated when the path changes.</t>
        <t>In some cases, however, it is necessary to explicitly indicate to a Connection that
a new remote endpoint has become available for use, or to indicate that some remote
endpoint is no longer available. This is most common in the case of peer to peer
connections using Trickle ICE <xref target="RFC8838"/>.</t>
        <t>The <tt>AddRemote()</tt> action can be used to add one or more new remote endpoints
to a Connection:</t>
        <artwork><![CDATA[
Connection.AddRemote([]RemoteEndpoint)
]]></artwork>
        <t>Endpoints that are already known to the Connection are ignored. A call to
<tt>AddRemote()</tt> makes the new remote endpoints available to the connection,
but whether the Connection makes use of those endpoints will depend on the
underlying transport protocol.</t>
        <t>Similarly, the <tt>RemoveRemote()</tt> action can be used to tell a connection to
stop using one or more remote endpoints:</t>
        <artwork><![CDATA[
Connection.RemoveRemote([]RemoteEndpoint)
]]></artwork>
        <t>Removing all known remote endpoints can have the effect of aborting the
connection. The effect of removing the active remote endpoint(s) depends
on the underlying transport: multipath aware transports might be able to
switch to a new path if other reachable remote endpoints exist, or the
connection might abort.</t>
        <t>Similarly, the <tt>AddLocal()</tt> and <tt>RemoveLocal()</tt> actions can be used to add
and remove local endpoints to/from a Connection.</t>
      </section>
    </section>
    <section anchor="introspection">
      <name>Managing Connections</name>
      <t>During pre-establishment and after establishment, connections can be configured and queried using Connection
Properties, and asynchronous information may be available about the state of the
connection via Soft Errors.</t>
      <t>Connection Properties represent the configuration and state of the selected
Protocol Stack(s) backing a Connection. These Connection Properties may be
Generic, applying regardless of transport protocol, or Specific, applicable to a
single implementation of a single transport protocol stack. Generic Connection
Properties are defined in <xref target="connection-props"/> below.</t>
      <t>Protocol Specific Properties are defined in a transport- and
implementation-specific way to
permit more specialized protocol features to be used.
Too much reliance by an application on Protocol Specific Properties can significantly reduce the flexibility
of a transport services implementation to make appropriate
selection and configuration choices. Therefore, it is RECOMMENDED that
Protocol Properties are used for properties common across different protocols and that
Protocol Specific Properties are only used where specific protocols or properties are necessary.</t>
      <t>The application can set and query Connection Properties on a per-Connection
basis. Connection Properties that are not read-only can be set during
pre-establishment (see <xref target="selection-props"/>), as well as on connections directly using
the SetProperty action:</t>
      <artwork><![CDATA[
Connection.SetProperty(property, value)
]]></artwork>
      <t>Note that changing one of the Connection Properties on one Connection in a Connection Group
will also change it for all other Connections of that group; see further <xref target="groups"/>.</t>
      <t>At any point, the application can query Connection Properties.</t>
      <artwork><![CDATA[
ConnectionProperties := Connection.GetProperties()
value := ConnectionProperties.Get(property)
if ConnectionProperties.Has(boolean_or_preference_property) then ...
]]></artwork>
      <t>Depending on the status of the connection, the queried Connection
Properties will include different information:</t>
      <ul spacing="normal">
        <li>The connection state, which can be one of the following:
Establishing, Established, Closing, or Closed.</li>
        <li>Whether the connection can be used to send data. A connection can not be used
for sending if the connection was created with the Selection Property
<tt>direction</tt> set to <tt>unidirectional receive</tt> or if a Message
marked as <tt>Final</tt> was sent over this connection. See also <xref target="msg-final"/>.</li>
        <li>Whether the connection can be used to receive data. A connection cannot be
used for reading if the connection was created with the Selection Property
<tt>direction</tt> set to <tt>unidirectional send</tt> or if a Message
marked as <tt>Final</tt> was received. See <xref target="receiving-final-messages"/>. The latter
is only supported by certain transport protocols, e.g., by TCP as half-closed
connection.</li>
        <li>For Connections that are Established, Closing, or Closed:
Connection Properties (<xref target="connection-props"/>) of the
actual protocols that were selected and instantiated, and Selection
Properties that the application specified on the Preconnection.
Selection Properties of type <tt>Preference</tt> will be exposed as boolean values
indicating whether or not the property applies to the selected transport.
Note that the instantiated protocol stack might not match all
Protocol Selection Properties that the application specified on the
Preconnection.</li>
        <li>For Connections that are Established: information concerning the
path(s) used by the Protocol Stack. This can be derived from local PVD information,
measurements by the Protocol Stack, or other sources.
For example, a TAPS system that is configured to receive and process PVD information
<xref target="RFC7556"/> could also provide network configuration information for the chosen path(s).</li>
      </ul>
      <section anchor="connection-props">
        <name>Generic Connection Properties</name>
        <t>Generic Connection Properties are defined independent of the chosen protocol stack
and therefore available on all Connections.</t>
        <t>Many Connection Properties have a corresponding Selection Property that
enables applications to express their preference for protocols providing a supporting transport feature.</t>
        <section anchor="conn-recv-checksum">
          <name>Required Minimum Corruption Protection Coverage for Receiving</name>
          <dl>
            <dt>Name:</dt>
            <dd>
              <t>recvChecksumLen</t>
            </dd>
            <dt>Type:</dt>
            <dd>
              <t>Integer (non-negative) or <tt>Full Coverage</tt></t>
            </dd>
            <dt>Default:</dt>
            <dd>
              <t><tt>Full Coverage</tt></t>
            </dd>
          </dl>
          <t>If this property is an Integer, it specifies the minimum number of bytes in a received
message that need to be covered by a checksum.
A receiving endpoint will not forward messages that have less coverage
to the application. The application is responsible for handling
any corruption within the non-protected part of the message <xref target="RFC8085"/>.
A special value of 0 means that a received packet may also have a zero checksum field.</t>
        </section>
        <section anchor="conn-priority">
          <name>Connection Priority</name>
          <dl>
            <dt>Name:</dt>
            <dd>
              <t>connPriority</t>
            </dd>
            <dt>Type:</dt>
            <dd>
              <t>Integer (non-negative)</t>
            </dd>
            <dt>Default:</dt>
            <dd>
              <t>100</t>
            </dd>
          </dl>
          <t>This property is a non-negative integer representing the
priority of this Connection
relative to other Connections in the same
Connection Group. A higher value reflects a higher priority. It has no effect
on Connections not part of a Connection
Group. As noted in <xref target="groups"/>, this property is not entangled when Connections
are cloned, i.e., changing the Priority on one Connection in a Connection Group
does not change it on the other Connections in the same Connection Group.
No guarantees of a specific behavior regarding Connection Priority are given;
a Transport Services system may ignore this property. See <xref target="priority-in-taps"/> for more details.</t>
        </section>
        <section anchor="conn-timeout">
          <name>Timeout for Aborting Connection</name>
          <dl>
            <dt>Name:</dt>
            <dd>
              <t>connTimeout</t>
            </dd>
            <dt>Type:</dt>
            <dd>
              <t>Numeric (non-negative) or <tt>Disabled</tt></t>
            </dd>
            <dt>Default:</dt>
            <dd>
              <t><tt>Disabled</tt></t>
            </dd>
          </dl>
          <t>If this property is Numeric, it specifies how long to wait before deciding that an active Connection has
failed when trying to reliably deliver data to the Remote Endpoint. Adjusting this property
will only take effect when the underlying stack supports reliability. If this property has the enumerated
value <tt>Disabled</tt>, it means that no timeout is scheduled.</t>
        </section>
        <section anchor="keep-alive-timeout">
          <name>Timeout for keep alive packets</name>
          <dl>
            <dt>Name:</dt>
            <dd>
              <t>keepAliveTimeout</t>
            </dd>
            <dt>Type:</dt>
            <dd>
              <t>Numeric (non-negative) or <tt>Disabled</tt></t>
            </dd>
            <dt>Default:</dt>
            <dd>
              <t>Implementation-defined</t>
            </dd>
          </dl>
          <t>A Transport Services API can request a protocol that supports sending keep alive packets <xref target="keep-alive"/>.
If this property is Numeric, it specifies the maximum length of time an idle connection (one for which no transport
packets have been sent) should wait before
the Local Endpoint sends a keep-alive packet to the Remote Endpoint. Adjusting this property
will only take effect when the underlying stack supports sending keep-alive packets.
Guidance on setting this value for connection-less transports is
provided in <xref target="RFC8085"/>.
A value greater than the connection timeout (<xref target="conn-timeout"/>) or the enumerated value <tt>Disabled</tt> will disable the sending of keep-alive packets.</t>
        </section>
        <section anchor="conn-scheduler">
          <name>Connection Group Transmission Scheduler</name>
          <dl>
            <dt>Name:</dt>
            <dd>
              <t>connScheduler</t>
            </dd>
            <dt>Type:</dt>
            <dd>
              <t>Enumeration</t>
            </dd>
            <dt>Default:</dt>
            <dd>
              <t>Weighted Fair Queueing (see Section 3.6 in <xref target="RFC8260"/>)</t>
            </dd>
          </dl>
          <t>This property specifies which scheduler should be used among Connections within
a Connection Group, see <xref target="groups"/>. The set of schedulers can
be taken from <xref target="RFC8260"/>.</t>
        </section>
        <section anchor="prop-cap-profile">
          <name>Capacity Profile</name>
          <dl>
            <dt>Name:</dt>
            <dd>
              <t>connCapacityProfile</t>
            </dd>
            <dt>Type:</dt>
            <dd>
              <t>Enumeration</t>
            </dd>
            <dt>Default:</dt>
            <dd>
              <t>Default Profile (Best Effort)</t>
            </dd>
          </dl>
          <t>This property specifies the desired network treatment for traffic sent by the
application and the tradeoffs the application is prepared to make in path and
protocol selection to receive that desired treatment. When the capacity profile
is set to a value other than Default, z Transport Services system SHOULD select paths
and configure protocols to optimize the tradeoff between delay, delay variation, and
efficient use of the available capacity based on the capacity profile specified. How this is realized
is implementation-specific. The Capacity Profile MAY also be used
to set markings on the wire for Protocol Stacks supporting this.
Recommendations for use with DSCP are provided below for each profile; note that
when a Connection is multiplexed, the guidelines in Section 6 of <xref target="RFC7657"/> apply.</t>
          <t>The following values are valid for the Capacity Profile:</t>
          <dl>
            <dt>Default:</dt>
            <dd>
              <t>The application provides no information about its expected capacity
  profile. Transport Services implementations that
  map the requested capacity profile onto per-connection DSCP signaling
  SHOULD assign the DSCP Default Forwarding <xref target="RFC2474"/> Per Hop Behaviour (PHB).</t>
            </dd>
            <dt>Scavenger:</dt>
            <dd>
              <t>The application is not interactive. It expects to send
  and/or receive data without any urgency. This can, for example, be used to
  select protocol stacks with scavenger transmission control and/or to assign
  the traffic to a lower-effort service. Transport Services implementations that
  map the requested capacity profile onto per-connection DSCP signaling
  SHOULD assign the DSCP Less than Best Effort
  <xref target="RFC8622"/> PHB.</t>
            </dd>
            <dt>Low Latency/Interactive:</dt>
            <dd>
              <t>The application is interactive, and prefers loss to
  latency. Response time should be optimized at the expense of delay variation
  and efficient use of the available capacity when sending on this connection. This can be
  used by the system to disable the coalescing of multiple small Messages into
  larger packets (Nagle's algorithm); to prefer immediate acknowledgment from
  the peer endpoint when supported by the underlying transport; and so on.
  Transport Services implementations that map the requested capacity profile onto per-connection DSCP signaling without multiplexing SHOULD assign a DSCP Assured Forwarding (AF41,AF42,AF43,AF44) <xref target="RFC2597"/> PHB. Inelastic traffic that is expected to conform to the configured network service rate could be mapped to the DSCP Expedited Forwarding <xref target="RFC3246"/> or <xref target="RFC5865"/> PHBs.</t>
            </dd>
            <dt>Low Latency/Non-Interactive:</dt>
            <dd>
              <t>The application prefers loss to latency, but is
  not interactive. Response time should be optimized at the expense of delay
  variation and efficient use of the available capacity when sending on this connection. Transport
  system implementations that map the requested capacity profile onto
  per-connection DSCP signaling without multiplexing SHOULD assign a DSCP
  Assured Forwarding (AF21,AF22,AF23,AF24) <xref target="RFC2597"/> PHB.</t>
            </dd>
            <dt>Constant-Rate Streaming:</dt>
            <dd>
              <t>The application expects to send/receive data at a
  constant rate after Connection establishment. Delay and delay variation should
  be minimized at the expense of efficient use of the available capacity.
  This implies that the Connection might fail if the Path is unable to maintain the desired rate.
  A transport can interpret this capacity profile as preferring a circuit
  breaker <xref target="RFC8084"/> to a rate-adaptive congestion controller. Transport
  system implementations that map the requested capacity profile onto
  per-connection DSCP signaling without multiplexing SHOULD assign a DSCP
  Assured Forwarding (AF31,AF32,AF33,AF34) <xref target="RFC2597"/> PHB.</t>
            </dd>
            <dt>Capacity-Seeking:</dt>
            <dd>
              <t>The application expects to send/receive data at the
  maximum rate allowed by its congestion controller over a relatively long
  period of time. Transport Services implementations that map the requested
  capacity profile onto per-connection DSCP signaling without multiplexing
  SHOULD assign a DSCP Assured Forwarding (AF11,AF12,AF13,AF14) <xref target="RFC2597"/> PHB
  per Section 4.8 of <xref target="RFC4594"/>.</t>
            </dd>
          </dl>
          <t>The Capacity Profile for a selected protocol stack may be modified on a
per-Message basis using the Transmission Profile Message Property; see
<xref target="send-profile"/>.</t>
        </section>
        <section anchor="multipath-policy">
          <name>Policy for using Multipath Transports</name>
          <dl>
            <dt>Name:</dt>
            <dd>
              <t>multipathPolicy</t>
            </dd>
            <dt>Type:</dt>
            <dd>
              <t>Enumeration</t>
            </dd>
            <dt>Default:</dt>
            <dd>
              <t>Handover</t>
            </dd>
          </dl>
          <t>This property specifies the local policy for transferring data across multiple paths between the same end hosts if Multipath Transport is not set to Disabled (see <xref target="multipath-mode"/>). Possible values are:</t>
          <dl>
            <dt>Handover:</dt>
            <dd>
              <t>The connection ought only to attempt to migrate between different paths when the original path is lost or becomes unusable. The thresholds used to declare a path unusable are implementation specific.</t>
            </dd>
            <dt>Interactive:</dt>
            <dd>
              <t>The connection ought only to attempt to minimize the latency for interactive traffic patterns by transmitting data across multiple paths when this is beneficial.
The goal of minimizing the latency will be balanced against the cost of each of these paths. Depending on the cost of the
lower-latency path, the scheduling might choose to use a higher-latency path. Traffic can be scheduled such that data may be transmitted
on multiple paths in parallel to achieve a lower latency. The specific scheduling algorithm is implementation-specific.</t>
            </dd>
            <dt>Aggregate:</dt>
            <dd>
              <t>The connection ought to attempt to use multiple paths in parallel to maximize available capacity and possibly overcome the capacity limitations of the individual paths. The actual strategy is implementation specific.</t>
            </dd>
          </dl>
          <t>Note that this is a local choice - the Remote Endpoint can choose a different policy.</t>
        </section>
        <section anchor="bounds-on-send-or-receive-rate">
          <name>Bounds on Send or Receive Rate</name>
          <dl>
            <dt>Name:</dt>
            <dd>
              <t>minSendRate / minRecvRate / maxSendRate / maxRecvRate</t>
            </dd>
            <dt>Type:</dt>
            <dd>
              <t>Numeric (non-negative) or <tt>Unlimited</tt> / Numeric (non-negative) or <tt>Unlimited</tt> / Numeric (non-negative) or <tt>Unlimited</tt> / Numeric (non-negative) or <tt>Unlimited</tt></t>
            </dd>
            <dt>Default:</dt>
            <dd>
              <t><tt>Unlimited</tt> / <tt>Unlimited</tt> / <tt>Unlimited</tt> / <tt>Unlimited</tt></t>
            </dd>
          </dl>
          <t>Numeric values of this property specify an upper-bound rate that a transfer is not expected to
exceed (even if flow control and congestion control allow higher rates), and/or a
lower-bound rate below which the application does not deem
it will be useful. These are specified in bits per second.
The enumerated value <tt>Unlimited</tt> indicates that no bound is specified.</t>
        </section>
        <section anchor="group-connection-limit">
          <name>Group Connection Limit</name>
          <dl>
            <dt>Name:</dt>
            <dd>
              <t>groupConnLimit</t>
            </dd>
            <dt>Type:</dt>
            <dd>
              <t>Numeric (non-negative) or <tt>Unlimited</tt></t>
            </dd>
            <dt>Default:</dt>
            <dd>
              <t><tt>Unlimited</tt></t>
            </dd>
          </dl>
          <t>If this property is Numeric, it controls the number of Connections that can be accepted from
a peer as new members of the Connection's group. Similar to SetNewConnectionLimit(),
this limits the number of ConnectionReceived Events that will occur, but constrained
to the group of the Connection associated with this property. For a multi-streaming transport,
this limits the number of allowed streams.</t>
        </section>
        <section anchor="isolate-session">
          <name>Isolate Session</name>
          <dl>
            <dt>Name:</dt>
            <dd>
              <t>isolateSession</t>
            </dd>
            <dt>Type:</dt>
            <dd>
              <t>Boolean</t>
            </dd>
            <dt>Default:</dt>
            <dd>
              <t>false</t>
            </dd>
          </dl>
          <t>When set to true, this property will initiate new Connections using as little
cached information (such as session tickets or cookies) as possible from
previous connections that are not in the same Connection Group. Any state generated by this
Connection will only be shared with Connections in the same Connection Group. Cloned Connections
will use saved state from within the Connection Group.
This is used for separating Connection Contexts as specified in <xref target="I-D.ietf-taps-arch"/>.</t>
          <t>Note that this does not guarantee no leakage of information, as
implementations may not be able to fully isolate all caches (e.g. RTT
estimates). Note that this property may degrade connection performance.</t>
        </section>
        <section anchor="read-only-conn-prop">
          <name>Read-only Connection Properties</name>
          <t>The following generic Connection Properties are read-only, i.e. they cannot be changed by an application.</t>
          <section anchor="size-safelyreplayable">
            <name>Maximum Message Size Concurrent with Connection Establishment</name>
            <dl>
              <dt>Name:</dt>
              <dd>
                <t>zeroRttMsgMaxLen</t>
              </dd>
              <dt>Type:</dt>
              <dd>
                <t>Integer (non-negative)</t>
              </dd>
            </dl>
            <t>This property represents the maximum Message size that can be sent
before or during Connection establishment, see also <xref target="msg-safelyreplayable"/>.
It is specified as the number of bytes.</t>
          </section>
          <section anchor="conn-max-msg-notfrag">
            <name>Maximum Message Size Before Fragmentation or Segmentation</name>
            <dl>
              <dt>Name:</dt>
              <dd>
                <t>singularTransmissionMsgMaxLen</t>
              </dd>
              <dt>Type:</dt>
              <dd>
                <t>Integer (non-negative) or <tt>Not applicable</tt></t>
              </dd>
            </dl>
            <t>This property, if applicable, represents the maximum Message size that can be
sent without incurring network-layer fragmentation at the sender.
It is specified as the number of bytes. It exposes a value to the application
based on the Maximum Packet Size (MPS) as described in Datagram PLPMTUD <xref target="RFC8899"/>.
This can allow a sending stack to avoid unwanted fragmentation at the
network-layer or segmentation by the transport layer.</t>
          </section>
          <section anchor="conn-max-msg-send">
            <name>Maximum Message Size on Send</name>
            <dl>
              <dt>Name:</dt>
              <dd>
                <t>sendMsgMaxLen</t>
              </dd>
              <dt>Type:</dt>
              <dd>
                <t>Integer (non-negative)</t>
              </dd>
            </dl>
            <t>This property represents the maximum Message size that an application can send.
It is specified as the number of bytes.</t>
          </section>
          <section anchor="conn-max-msg-recv">
            <name>Maximum Message Size on Receive</name>
            <dl>
              <dt>Name:</dt>
              <dd>
                <t>recvMsgMaxLen</t>
              </dd>
              <dt>Type:</dt>
              <dd>
                <t>Integer (non-negative)</t>
              </dd>
            </dl>
            <t>This property represents the maximum Message size that an application can receive.
It specified as the number of bytes.</t>
          </section>
        </section>
      </section>
      <section anchor="tcp-uto">
        <name>TCP-specific Properties: User Timeout Option (UTO)</name>
        <t>These properties specify configurations for the User Timeout Option (UTO),
in the case that TCP becomes the chosen transport protocol.
Implementation is optional and useful only if TCP is implemented in the Transport Services system.</t>
        <t>These TCP-specific properties are included here because the feature <tt>Suggest
timeout to the peer</tt> is part of the minimal set of transport services
<xref target="RFC8923"/>, where this feature was categorized as "functional".
This means that when an Transport Services implementation offers this feature,
the Transport Services API has to expose an interface to the application. Otherwise, the implementation might
violate assumptions by the application, which could cause the application to
fail.</t>
        <t>All of the below properties are optional (e.g., it is possible to specify <tt>User Timeout Enabled</tt> as true,
but not specify an Advertised User Timeout value; in this case, the TCP default will be used).
These properties reflect the API extension specified in Section 3 of <xref target="RFC5482"/>.</t>
        <section anchor="advertised-user-timeout">
          <name>Advertised User Timeout</name>
          <dl>
            <dt>Name:</dt>
            <dd>
              <t>tcp.userTimeoutValue</t>
            </dd>
            <dt>Type:</dt>
            <dd>
              <t>Integer (non-negative)</t>
            </dd>
            <dt>Default:</dt>
            <dd>
              <t>the TCP default</t>
            </dd>
          </dl>
          <t>This time value is advertised via the TCP User Timeout Option (UTO) <xref target="RFC5482"/> at the Remote Endpoint
to adapt its own <tt>Timeout for aborting Connection</tt> (see <xref target="conn-timeout"/>) value.</t>
        </section>
        <section anchor="user-timeout-enabled">
          <name>User Timeout Enabled</name>
          <dl>
            <dt>Name:</dt>
            <dd>
              <t>tcp.userTimeoutEnabled</t>
            </dd>
            <dt>Type:</dt>
            <dd>
              <t>Boolean</t>
            </dd>
            <dt>Default:</dt>
            <dd>
              <t>false</t>
            </dd>
          </dl>
          <t>This property controls whether the UTO option is enabled for a
connection. This applies to both sending and receiving.</t>
        </section>
        <section anchor="timeout-changeable">
          <name>Timeout Changeable</name>
          <dl>
            <dt>Name:</dt>
            <dd>
              <t>tcp.userTimeoutChangeable</t>
            </dd>
            <dt>Type:</dt>
            <dd>
              <t>Boolean</t>
            </dd>
            <dt>Default:</dt>
            <dd>
              <t>true</t>
            </dd>
          </dl>
          <t>This property controls whether the <tt>connTimeout</tt> (see <xref target="conn-timeout"/>)
may be changed
based on a UTO option received from the remote peer. This boolean becomes false when
<tt>connTimeout</tt> (see <xref target="conn-timeout"/>) is used.</t>
        </section>
      </section>
      <section anchor="connection-lifecycle-events">
        <name>Connection Lifecycle Events</name>
        <t>During the lifetime of a connection there are events that can occur when configured.</t>
        <section anchor="soft-errors">
          <name>Soft Errors</name>
          <t>Asynchronous introspection is also possible, via the SoftError Event. This event
informs the application about the receipt and contents of an ICMP error message related to the Connection. This will only happen if the underlying protocol stack supports access to soft errors; however, even if the underlying stack supports it, there
is no guarantee that a soft error will be signaled.</t>
          <artwork><![CDATA[
Connection -> SoftError<>
]]></artwork>
        </section>
        <section anchor="conn-path-change">
          <name>Path change</name>
          <t>This event notifies the application when at least one of the paths underlying a Connection has changed. Changes occur
on a single path when the PMTU changes as well as when multiple paths are used
and paths are added or removed, the set of local endpoints changes, or a handover has been performed.</t>
          <artwork><![CDATA[
Connection -> PathChange<>
]]></artwork>
        </section>
      </section>
    </section>
    <section anchor="datatransfer">
      <name>Data Transfer</name>
      <t>Data is sent and received as Messages, which allows the application
to communicate the boundaries of the data being transferred.</t>
      <section anchor="msg">
        <name>Messages and Framers</name>
        <t>Each Message has an optional Message Context, which allows to add Message Properties, identify Send Events related to a specific Message or to inspect meta-data related to the Message sent. Framers can be used to extend or modify the message data with additional information that can be processed at the receiver to detect message boundaries.</t>
        <section anchor="msg-ctx">
          <name>Message Contexts</name>
          <t>Using the MessageContext object, the application can set and retrieve meta-data of the message, including Message Properties (see <xref target="message-props"/>) and framing meta-data (see <xref target="framing-meta"/>).
Therefore, a MessageContext object can be passed to the Send action and is returned by each Send and Receive related event.</t>
          <t>Message Properties can be set and queried using the Message Context:</t>
          <artwork><![CDATA[
MessageContext.add(property, value)
PropertyValue := MessageContext.get(property)
]]></artwork>
          <t>These Message Properties may be generic properties or Protocol Specific Properties.</t>
          <t>For MessageContexts returned by send Events (see <xref target="send-events"/>) and receive Events (see <xref target="receive-events"/>), the application can query information about the Local and Remote Endpoint:</t>
          <artwork><![CDATA[
RemoteEndpoint := MessageContext.GetRemoteEndpoint()
LocalEndpoint := MessageContext.GetLocalEndpoint()
]]></artwork>
        </section>
        <section anchor="framing">
          <name>Message Framers</name>
          <t>Although most applications communicate over a network using well-formed
Messages, the boundaries and metadata of the Messages are often not
directly communicated by the transport protocol itself. For example,
HTTP applications send and receive HTTP messages over a byte-stream
transport, requiring that the boundaries of HTTP messages be parsed
from the stream of bytes.</t>
          <t>Message Framers allow extending a Connection's Protocol Stack to define
how to encapsulate or encode outbound Messages, and how to decapsulate
or decode inbound data into Messages. Message Framers allow message
boundaries to be preserved when using a Connection object, even when
using byte-stream transports. This is designed based on the fact
that many of the current application protocols evolved over TCP, which
does not provide message boundary preservation, and since many of these
protocols require message boundaries to function, each application layer
protocol has defined its own framing.</t>
          <t>To use a Message Framer, the application adds it to its Preconnection object.
Then, the Message Framer can intercept all calls to Send() or Receive()
on a Connection to add Message semantics, in addition to interacting with
the setup and teardown of the Connection. A Framer can start sending data
before the application sends data if the framing protocol requires a prefix
or handshake (see <xref target="RFC8229"/> for an example of such a framing protocol).</t>
          <figure anchor="fig-framer-stack">
            <name>Protocol Stack showing a Message Framer</name>
            <artwork><![CDATA[
  Initiate()   Send()   Receive()   Close()
      |          |         ^          |
      |          |         |          |
 +----v----------v---------+----------v-----+
 |                Connection                |
 +----+----------+---------^----------+-----+
      |          |         |          |
      |      +-----------------+      |
      |      |    Messages     |      |
      |      +-----------------+      |
      |          |         |          |
 +----v----------v---------+----------v-----+
 |                Framer(s)                 |
 +----+----------+---------^----------+-----+
      |          |         |          |
      |      +-----------------+      |
      |      |   Byte-stream   |      |
      |      +-----------------+      |
      |          |         |          |
 +----v----------v---------+----------v-----+
 |         Transport Protocol Stack         |
 +------------------------------------------+
]]></artwork>
          </figure>
          <t>Note that while Message Framers add the most value when placed above
a protocol that otherwise does not preserve message boundaries, they can
also be used with datagram- or message-based protocols. In these cases,
they add an additional transformation to further encode or encapsulate,
and can potentially support packing multiple application-layer Messages
into individual transport datagrams.</t>
          <t>The API to implement a Message Framer can vary depending on the implementation;
guidance on implementing Message Framers can be found in <xref target="I-D.ietf-taps-impl"/>.</t>
          <section anchor="adding-message-framers-to-pre-connections">
            <name>Adding Message Framers to Pre-Connections</name>
            <t>The Message Framer object can be added to one or more Preconnections
to run on top of transport protocols. Multiple Framers may be added to a Preconnection;
in this case, the Framers operate as a framing stack, i.e. the last one added runs
first when framing outbound messages, and last when parsing inbound data.</t>
            <t>The following example adds a basic HTTP Message Framer to a Preconnection:</t>
            <artwork><![CDATA[
framer := NewHTTPMessageFramer()
Preconnection.AddFramer(framer)
]]></artwork>
            <t>Since Message Framers pass from Preconnection to Listener or Connection, addition of
Framers must happen before any operation that may result in the creation of a Connection.</t>
          </section>
          <section anchor="framing-meta">
            <name>Framing Meta-Data</name>
            <t>When sending Messages, applications can add Framer-specific
properties to a MessageContext (<xref target="msg-ctx"/>).
In order to set these properties, the <tt>add</tt> and <tt>get</tt> actions
on the MessageContext. To avoid naming conflicts, the property
names SHOULD be prefixed with a namespace referencing the
framer implementation or the protocol it implements as described
in <xref target="property-names"/>.</t>
            <t>This mechanism can be used, for example, to set the type of a Message for a TLV format.
The namespace of values is custom for each unique Message Framer.</t>
            <artwork><![CDATA[
messageContext := NewMessageContext()
messageContext.add(framer, key, value)
Connection.Send(messageData, messageContext)
]]></artwork>
            <t>When an application receives a MessageContext in a Receive event,
it can also look to see if a value was set by a specific Message Framer.</t>
            <artwork><![CDATA[
messageContext.get(framer, key) -> value
]]></artwork>
            <t>For example, if an HTTP Message Framer is used, the values could correspond
to HTTP headers:</t>
            <artwork><![CDATA[
httpFramer := NewHTTPMessageFramer()
...
messageContext := NewMessageContext()
messageContext.add(httpFramer, "accept", "text/html")
]]></artwork>
          </section>
        </section>
        <section anchor="message-props">
          <name>Message Properties</name>
          <t>Applications needing to annotate the Messages they send with extra information
(for example, to control how data is scheduled and processed by the transport protocols supporting the
Connection) can include this information in the Message Context passed to the Send Action. For other uses of the message context, see <xref target="msg-ctx"/>.</t>
          <t>Message Properties are per-Message, not per-Send if partial Messages
are sent (<xref target="send-partial"/>). All data blocks associated with a single Message
share properties specified in the Message Contexts. For example, it would not
make sense to have the beginning of a Message expire, but allow the end of a
Message to still be sent.</t>
          <t>A MessageContext object contains metadata for the Messages to be sent or received.</t>
          <artwork><![CDATA[
messageData := "hello"
messageContext := NewMessageContext()
messageContext.add(parameter, value)
Connection.Send(messageData, messageContext)
]]></artwork>
          <t>The simpler form of Send, which does not take any messageContext, is equivalent to passing a default MessageContext without adding any Message Properties.</t>
          <t>If an application wants to override Message Properties for a specific message,
it can acquire an empty MessageContext Object and add all desired Message
Properties to that Object. It can then reuse the same messageContext Object
for sending multiple Messages with the same properties.</t>
          <t>Properties can be added to a MessageContext object only before the context is used
for sending. Once a MessageContext has been used with a Send call, further modifications
to the MessageContext object do not have any effect on this Send call. Message Properties
that are not added to a MessageContext object before using the context for sending will either
take a specific default value or be configured based on Selection or Connection Properties
of the Connection that is associated with the Send call.
This initialization behavior is defined per Message Property below.</t>
          <t>The Message Properties could be inconsistent with the properties of the Protocol Stacks
underlying the Connection on which a given Message is sent. For example,
a Protocol Stack must be able to provide ordering if the <tt>msgOrdered</tt>
property of a Message is enabled. Sending a Message with Message Properties
inconsistent with the Selection Properties of the Connection yields an error.</t>
          <t>If a Message Property contradicts a Connection Property, and
if this per-Message behavior can be supported, it overrides the Connection
Property for the specific Message. For example, if
<tt>reliability</tt> is set to <tt>Require</tt> and a protocol
with configurable per-Message reliability is used, setting
<tt>msgReliable</tt> to <tt>false</tt> for a particular Message will
allow this Message to be sent without any reliability guarantees. Changing
the <tt>msgReliable</tt> Message Property is only possible for
Connections that were established enabling the Selection Property
<tt>perMsgReliability</tt>.</t>
          <t>The following Message Properties are supported:</t>
          <section anchor="msg-lifetime">
            <name>Lifetime</name>
            <dl>
              <dt>Name:</dt>
              <dd>
                <t>msgLifetime</t>
              </dd>
              <dt>Type:</dt>
              <dd>
                <t>Numeric (non-negative)</t>
              </dd>
              <dt>Default:</dt>
              <dd>
                <t>infinite</t>
              </dd>
            </dl>
            <t>The Lifetime specifies how long a particular Message can wait to be sent to the
Remote Endpoint before it is irrelevant and no longer needs to be
(re-)transmitted. This is a hint to the Transport Services implementation -- it is not guaranteed
that a Message will not be sent when its Lifetime has expired.</t>
            <t>Setting a Message's Lifetime to infinite indicates that the application does
not wish to apply a time constraint on the transmission of the Message, but it does not express a need for
reliable delivery; reliability is adjustable per Message via the <tt>perMsgReliability</tt>
property (see <xref target="msg-reliable-message"/>). The type and units of Lifetime are implementation-specific.</t>
          </section>
          <section anchor="msg-priority">
            <name>Priority</name>
            <dl>
              <dt>Name:</dt>
              <dd>
                <t>msgPriority</t>
              </dd>
              <dt>Type:</dt>
              <dd>
                <t>Integer (non-negative)</t>
              </dd>
              <dt>Default:</dt>
              <dd>
                <t>100</t>
              </dd>
            </dl>
            <t>This property specifies the priority of a Message, relative to other Messages sent over the
same Connection.</t>
            <t>A Message with Priority 0 will yield to a Message with Priority 1, which will
yield to a Message with Priority 2, and so on. Priorities may be used as a
sender-side scheduling construct only, or be used to specify priorities on the
wire for Protocol Stacks supporting prioritization.</t>
            <t>Note that this property is not a per-message override of <tt>connPriority</tt>
- see <xref target="conn-priority"/>. The priority properties may interact, but can be used
independently and be realized by different mechanisms; see <xref target="priority-in-taps"/>.</t>
          </section>
          <section anchor="msg-ordered">
            <name>Ordered</name>
            <dl>
              <dt>Name:</dt>
              <dd>
                <t>msgOrdered</t>
              </dd>
              <dt>Type:</dt>
              <dd>
                <t>Boolean</t>
              </dd>
              <dt>Default:</dt>
              <dd>
                <t>the queried Boolean value of the Selection Property <tt>preserveOrder</tt> (<xref target="prop-ordering"/>)</t>
              </dd>
            </dl>
            <t>The order in which Messages were submitted for transmission via the Send Action will be preserved on delivery via Receive&lt;&gt; events for all Messages on a Connection that have this Message Property set to true.</t>
            <t>If false, the Message is delivered to the receiving application without preserving the ordering.
This property is used for protocols that support preservation of data ordering,
see <xref target="prop-ordering"/>, but allow out-of-order delivery for certain messages, e.g., by multiplexing independent messages onto
different streams.</t>
            <t>If it is not configured by the application before sending, this property's default value
will be based on the Selection Property <tt>preserveOrder</tt> of the Connection
associated with the Send Action.</t>
          </section>
          <section anchor="msg-safelyreplayable">
            <name>Safely Replayable</name>
            <dl>
              <dt>Name:</dt>
              <dd>
                <t>safelyReplayable</t>
              </dd>
              <dt>Type:</dt>
              <dd>
                <t>Boolean</t>
              </dd>
              <dt>Default:</dt>
              <dd>
                <t>false</t>
              </dd>
            </dl>
            <t>If true, <tt>safelyReplayable</tt> specifies that a Message is safe to send to the Remote Endpoint
more than once for a single Send Action. It marks the data as safe for
certain 0-RTT establishment techniques, where retransmission of the 0-RTT data
may cause the remote application to receive the Message multiple times.</t>
            <t>For protocols that do not protect against duplicated messages,
e.g., UDP, all messages need to be marked as "safely replayable" by enabling this property.
To enable protocol selection to choose such a protocol,
<tt>safelyReplayable</tt> needs to be added to the TransportProperties passed to the
Preconnection. If such a protocol was chosen, disabling <tt>safelyReplayable</tt> on
individual messages MUST result in a SendError.</t>
          </section>
          <section anchor="msg-final">
            <name>Final</name>
            <dl>
              <dt>Name:</dt>
              <dd>
                <t>final</t>
              </dd>
              <dt>Type:</dt>
              <dd>
                <t>Boolean</t>
              </dd>
              <dt>Default:</dt>
              <dd>
                <t>false</t>
              </dd>
            </dl>
            <t>If true, this indicates a Message is the last that
the application will send on a Connection. This allows underlying protocols
to indicate to the Remote Endpoint that the Connection has been effectively
closed in the sending direction. For example, TCP-based Connections can
send a FIN once a Message marked as Final has been completely sent,
indicated by marking endOfMessage. Protocols that do not support signalling
the end of a Connection in a given direction will ignore this property.</t>
            <t>A Final Message must always be sorted to the end of a list of Messages.
The Final property overrides Priority and any other property that would re-order
Messages. If another Message is sent after a Message marked as Final has already
been sent on a Connection, the Send Action for the new Message will cause a SendError Event.</t>
          </section>
          <section anchor="msg-checksum">
            <name>Sending Corruption Protection Length</name>
            <dl>
              <dt>Name:</dt>
              <dd>
                <t>msgChecksumLen</t>
              </dd>
              <dt>Type:</dt>
              <dd>
                <t>Integer (non-negative) or <tt>Full Coverage</tt></t>
              </dd>
              <dt>Default:</dt>
              <dd>
                <t><tt>Full Coverage</tt></t>
              </dd>
            </dl>
            <t>If this property is an Integer, it specifies the minimum length of the section of a sent Message,
starting from byte 0, that the application requires to be delivered without
corruption due to lower layer errors. It is used to specify options for simple
integrity protection via checksums. A value of 0 means that no checksum
needs to be calculated, and the enumerated value <tt>Full Coverage</tt> means
that the entire Message needs to be protected by a checksum. Only <tt>Full Coverage</tt> is
guaranteed, any other requests are advisory, which may result in <tt>Full Coverage</tt> being applied.</t>
          </section>
          <section anchor="msg-reliable-message">
            <name>Reliable Data Transfer (Message)</name>
            <dl>
              <dt>Name:</dt>
              <dd>
                <t>msgReliable</t>
              </dd>
              <dt>Type:</dt>
              <dd>
                <t>Boolean</t>
              </dd>
              <dt>Default:</dt>
              <dd>
                <t>the queried Boolean value of the Selection Property <tt>reliability</tt> (<xref target="prop-reliable"/>)</t>
              </dd>
            </dl>
            <t>When true, this property specifies that a Message should be sent in such a way
that the transport protocol ensures all data is received on the other side
without corruption. Changing the <tt>msgReliable</tt> property on Messages
is only possible for Connections that were established enabling the Selection Property <tt>perMsgReliability</tt>.
When this is not the case, changing <tt>msgReliable</tt> will generate an error.</t>
            <t>Disabling this property indicates that the Transport Services system may disable retransmissions
or other reliability mechanisms for this particular Message, but such disabling is not guaranteed.</t>
            <t>If it is not configured by the application before sending, this property's default value
will be based on the Selection Property <tt>reliability</tt> of the Connection
associated with the Send Action.</t>
          </section>
          <section anchor="send-profile">
            <name>Message Capacity Profile Override</name>
            <dl>
              <dt>Name:</dt>
              <dd>
                <t>msgCapacityProfile</t>
              </dd>
              <dt>Type:</dt>
              <dd>
                <t>Enumeration</t>
              </dd>
              <dt>Default:</dt>
              <dd>
                <t>inherited from the Connection Property <tt>connCapacityProfile</tt> (<xref target="prop-cap-profile"/>)</t>
              </dd>
            </dl>
            <t>This enumerated property specifies the application's preferred tradeoffs for
sending this Message; it is a per-Message override of the <tt>connCapacityProfile</tt>
Connection Property (see <xref target="prop-cap-profile"/>).
If it is not configured by the application before sending, this property's default value
will be based on the Connection Property <tt>connCapacityProfile</tt> of the Connection
associated with the Send Action.</t>
          </section>
          <section anchor="send-singular">
            <name>No Network-Layer Fragmentation</name>
            <dl>
              <dt>Name:</dt>
              <dd>
                <t>noFragmentation</t>
              </dd>
              <dt>Type:</dt>
              <dd>
                <t>Boolean</t>
              </dd>
              <dt>Default:</dt>
              <dd>
                <t>false</t>
              </dd>
            </dl>
            <t>This property specifies that a message should be sent and received
without network-layer fragmentation, if possible. It can be used
to avoid network layer fragmentation when transport segmentation is prefered.</t>
            <t>This only takes effect when the transport uses a network layer that supports this functionality.
When it does take effect, setting this property to
true will cause the sender to avoid network-layer source fragmentation.
When using IPv4, this will result in the Don't Fragment bit being set in the IP header.</t>
            <t>Attempts to send a message with this property that result in a size greater than the
transport's current estimate of its maximum packet size (<tt>singularTransmissionMsgMaxLen</tt>)
can result in transport segmentation when permitted, or in a <tt>SendError</tt>.</t>
            <t>Note: noSegmentation should be used when it is desired to only send a message within
a single network packet.</t>
          </section>
          <section anchor="no-transport-fragmentation">
            <name>No Segmentation</name>
            <dl>
              <dt>Name:</dt>
              <dd>
                <t>noSegmentation</t>
              </dd>
              <dt>Type:</dt>
              <dd>
                <t>Boolean</t>
              </dd>
              <dt>Default:</dt>
              <dd>
                <t>false</t>
              </dd>
            </dl>
            <t>When set to true, this property requests the transport layer
to not provide segmentation of messages larger than the
maximum size permitted by the network layer, and also
to avoid network-layer source fragmentation of messages.
When running over IPv4, setting this property to
true will result in a sending endpount setting the
Don't Fragment bit in the IPv4 header of packets generated by the
transport layer.</t>
            <t>An
attempt to send a message that results in a size greater than the
transport's current estimate of its maximum packet size (<tt>singularTransmissionMsgMaxLen</tt>)
will result in a <tt>SendError</tt>.
This only takes effect when the transport and network layer
support this functionality.</t>
          </section>
        </section>
      </section>
      <section anchor="sending">
        <name>Sending Data</name>
        <t>Once a Connection has been established, it can be used for sending Messages.
By default, Send enqueues a complete Message,
and takes optional per-Message properties (see <xref target="send-basic"/>). All Send actions
are asynchronous, and deliver Events (see <xref target="send-events"/>). Sending partial
Messages for streaming large data is also supported (see <xref target="send-partial"/>).</t>
        <t>Messages are sent on a Connection using the Send action:</t>
        <artwork><![CDATA[
Connection.Send(messageData, messageContext?, endOfMessage?)
]]></artwork>
        <t>where messageData is the data object to send, and messageContext allows
adding Message Properties, identifying Send Events related to a specific
Message or inspecting meta-data related to the Message sent (see <xref target="msg-ctx"/>).</t>
        <t>The optional endOfMessage parameter supports partial sending and is described in
<xref target="send-partial"/>.</t>
        <section anchor="send-basic">
          <name>Basic Sending</name>
          <t>The most basic form of sending on a connection involves enqueuing a single Data
block as a complete Message with default Message Properties.</t>
          <artwork><![CDATA[
messageData := "hello"
Connection.Send(messageData)
]]></artwork>
          <t>The interpretation of a Message to be sent is dependent on the implementation, and
on the constraints on the Protocol Stacks implied by the Connection's transport properties.
For example, a Message may be a single datagram for UDP Connections; or an HTTP
Request for HTTP Connections.</t>
          <t>Some transport protocols can deliver arbitrarily sized Messages, but other
protocols constrain the maximum Message size. Applications can query the
Connection Property <tt>sendMsgMaxLen</tt> (<xref target="conn-max-msg-send"/>) to determine the maximum size
allowed for a single Message. If a Message is too large to fit in the Maximum Message
Size for the Connection, the Send will fail with a SendError event (<xref target="send-error"/>). For
example, it is invalid to send a Message over a UDP connection that is larger than
the available datagram sending size.</t>
        </section>
        <section anchor="send-events">
          <name>Send Events</name>
          <t>Like all Actions in Transport Services API, the Send Action is asynchronous. There are
several Events that can be delivered in response to Sending a Message.
Exactly one Event (Sent, Expired, or SendError) will be delivered in response
to each call to Send.</t>
          <t>Note that if partial Sends are used (<xref target="send-partial"/>), there will still be exactly
one Send Event delivered for each call to Send. For example, if a Message
expired while two requests to Send data for that Message are outstanding,
there will be two Expired events delivered.</t>
          <t>The Transport Services API should allow the application to correlate which Send Action resulted
in a particular Send Event. The manner in which this correlation is indicated
is implementation-specific.</t>
          <section anchor="sent">
            <name>Sent</name>
            <artwork><![CDATA[
Connection -> Sent<messageContext>
]]></artwork>
            <t>The Sent Event occurs when a previous Send Action has completed, i.e., when
the data derived from the Message has been passed down or through the
underlying Protocol Stack and is no longer the responsibility of
the Transport Services API. The exact disposition of the Message (i.e.,
whether it has actually been transmitted, moved into a buffer on the network
interface, moved into a kernel buffer, and so on) when the Sent Event occurs
is implementation-specific. The Sent Event contains a reference to the Message
Context of the Message to which it applies.</t>
            <t>Sent Events allow an application to obtain an understanding of the amount
of buffering it creates. That is, if an application calls the Send Action multiple
times without waiting for a Sent Event, it has created more buffer inside the
Transport Services system than an application that always waits for the Sent Event before
calling the next Send Action.</t>
          </section>
          <section anchor="expired">
            <name>Expired</name>
            <artwork><![CDATA[
Connection -> Expired<messageContext>
]]></artwork>
            <t>The Expired Event occurs when a previous Send Action expired before completion;
i.e. when the Message was not sent before its Lifetime (see <xref target="msg-lifetime"/>)
expired. This is separate from SendError, as it is an expected behavior for
partially reliable transports. The Expired Event contains a reference to the
Message Context of the Message to which it applies.</t>
          </section>
          <section anchor="send-error">
            <name>SendError</name>
            <artwork><![CDATA[
Connection -> SendError<messageContext, reason?>
]]></artwork>
            <t>A SendError occurs when a Message was not sent due to an error condition:
an attempt to send a Message which is too large for the system and
Protocol Stack to handle, some failure of the underlying Protocol Stack, or a
set of Message Properties not consistent with the Connection's transport
properties. The SendError contains a reference to the Message Context of the
Message to which it applies.</t>
          </section>
        </section>
        <section anchor="send-partial">
          <name>Partial Sends</name>
          <t>It is not always possible for an application to send all data associated with
a Message in a single Send Action. The Message data may be too large for
the application to hold in memory at one time, or the length of the Message
may be unknown or unbounded.</t>
          <t>Partial Message sending is supported by passing an endOfMessage boolean
parameter to the Send Action. This value is always true by default, and
the simpler forms of Send are equivalent to passing true for endOfMessage.</t>
          <t>The following example sends a Message in two separate calls to Send.</t>
          <artwork><![CDATA[
messageContext := NewMessageContext()
messageContext.add(parameter, value)

messageData := "hel"
endOfMessage := false
Connection.Send(messageData, messageContext, endOfMessage)

messageData := "lo"
endOfMessage := true
Connection.Send(messageData, messageContext, endOfMessage)
]]></artwork>
          <t>All data sent with the same MessageContext object will be treated as belonging
to the same Message, and will constitute an in-order series until the
endOfMessage is marked.</t>
        </section>
        <section anchor="send-batching">
          <name>Batching Sends</name>
          <t>To reduce the overhead of sending multiple small Messages on a Connection, the
application could batch several Send Actions together. This provides a hint to
the system that the sending of these Messages ought to be coalesced when possible,
and that sending any of the batched Messages can be delayed until the last Message
in the batch is enqueued.</t>
          <t>The semantics for starting and ending a batch can be implementation-specific, but need
to allow multiple Send Actions to be enqueued.</t>
          <artwork><![CDATA[
Connection.StartBatch()
Connection.Send(messageData)
Connection.Send(messageData)
Connection.EndBatch()
]]></artwork>
        </section>
        <section anchor="initiate-and-send">
          <name>Send on Active Open: InitiateWithSend</name>
          <t>For application-layer protocols where the Connection initiator also sends the
first message, the InitiateWithSend() action combines Connection initiation with
a first Message sent:</t>
          <artwork><![CDATA[
Connection := Preconnection.InitiateWithSend(messageData, messageContext?, timeout?)
]]></artwork>
          <t>Whenever possible, a messageContext should be provided to declare the Message passed to InitiateWithSend
as "safely replayable" using the <tt>safelyReplayable</tt> property. This allows the Transport Services system to make use of 0-RTT establishment in case this is supported
by the available protocol stacks. When the selected stack(s) do not support transmitting data upon connection
establishment, InitiateWithSend is identical to Initiate() followed by Send().</t>
          <t>Neither partial sends nor send batching are supported by InitiateWithSend().</t>
          <t>The Events that may be sent after InitiateWithSend() are equivalent to those
that would be sent by an invocation of Initiate() followed immediately by an
invocation of Send(), with the caveat that a send failure that occurs because
the Connection could not be established will not result in a
SendError separate from the EstablishmentError signaling the failure of Connection
establishment.</t>
        </section>
        <section anchor="priority-in-taps">
          <name>Priority and the Transport Services API</name>
          <t>The Transport Services API provides two properties to allow a sender
to signal the relative priority of data transmission: <tt>msgPriority</tt>
            <xref target="msg-priority"/> and <tt>connPriority</tt> <xref target="conn-priority"/>.
These properties are designed to allow the expression
and implementation of a wide variety of approaches to transmission priority in
the transport and application layer, including those which do not appear on
the wire (affecting only sender-side transmission scheduling) as well as those
that do (e.g. <xref target="I-D.ietf-httpbis-priority"/>.</t>
          <t>A Transport Services system gives no guarantees about how its expression of
relative priorities will be realized. However, the Transport Services system will
seek to ensure that performance of relatively-prioritized connections and
messages is not worse with respect to those connections and messages than
an equivalent configuration in which all prioritization properties are left
at their defaults.</t>
          <t>The Transport Services API does order <tt>connPriority</tt> over
<tt>msgPriority</tt>. In the absense of other externalities
(e.g., transport-layer flow control), a priority 1 Message on a priority 0
Connection will be sent before a priority 0 Message on a priority 1
Connection in the same group.</t>
        </section>
      </section>
      <section anchor="receiving">
        <name>Receiving Data</name>
        <t>Once a Connection is established, it can be used for receiving data (unless the
<tt>direction</tt> property is set to <tt>unidirectional send</tt>). As with
sending, the data is received in Messages. Receiving is an asynchronous
operation, in which each call to Receive enqueues a request to receive new
data from the connection. Once data has been received, or an error is encountered,
an event will be delivered to complete any pending Receive requests (see <xref target="receive-events"/>).
If Messages arrive at the Transport Services system before Receive requests are issued,
ensuing Receive requests will first operate on these Messages before awaiting any further Messages.</t>
        <section anchor="enqueuing-receives">
          <name>Enqueuing Receives</name>
          <t>Receive takes two parameters to specify the length of data that an application
is willing to receive, both of which are optional and have default values if not
specified.</t>
          <artwork><![CDATA[
Connection.Receive(minIncompleteLength?, maxLength?)
]]></artwork>
          <t>By default, Receive will try to deliver complete Messages in a single event (<xref target="receive-complete"/>).</t>
          <t>The application can set a minIncompleteLength value to indicate the smallest partial
Message data size in bytes that should be delivered in response to this Receive. By default,
this value is infinite, which means that only complete Messages should be delivered (see <xref target="receive-partial"/>
and <xref target="framing"/> for more information on how this is accomplished).
If this value is set to some smaller value, the associated receive event will be triggered
only when at least that many bytes are available, or the Message is complete with fewer
bytes, or the system needs to free up memory. Applications should always
check the length of the data delivered to the receive event and not assume
it will be as long as minIncompleteLength in the case of shorter complete Messages
or memory issues.</t>
          <t>The maxLength argument indicates the maximum size of a Message in bytes
that the application is currently prepared to receive. The default
value for maxLength is infinite. If an incoming Message is larger than the
minimum of this size and the maximum Message size on receive for
the Connection's Protocol Stack, it will be delivered via ReceivedPartial
events (<xref target="receive-partial"/>).</t>
          <t>Note that maxLength does not guarantee that the application will receive that many
bytes if they are available; the Transport Services API could return ReceivedPartial events with less
data than maxLength according to implementation constraints. Note also that maxLength
and minIncompleteLength are intended only to manage buffering, and are not interpreted
as a receiver preference for message reordering.</t>
        </section>
        <section anchor="receive-events">
          <name>Receive Events</name>
          <t>Each call to Receive will be paired with a single Receive Event, which can be a success
or an error. This allows an application to provide backpressure to the transport stack
when it is temporarily not ready to receive messages.</t>
          <t>The Transport Services API should allow the application to correlate which call to Receive resulted
in a particular Receive Event. The manner in which this correlation is indicated
is implementation-specific.</t>
          <section anchor="receive-complete">
            <name>Received</name>
            <artwork><![CDATA[
Connection -> Received<messageData, messageContext>
]]></artwork>
            <t>A Received event indicates the delivery of a complete Message.
It contains two objects, the received bytes as messageData, and the metadata and properties of the received Message as messageContext.</t>
            <t>The messageData object provides access to the bytes that were received for this Message, along with the length of the byte array.
The messageContext is provided to enable retrieving metadata about the message and referring to the message. The messageContext object ist described in <xref target="msg-ctx"/>.</t>
            <t>See <xref target="framing"/> for handling Message framing in situations where the Protocol
Stack only provides a byte-stream transport.</t>
          </section>
          <section anchor="receive-partial">
            <name>ReceivedPartial</name>
            <artwork><![CDATA[
Connection -> ReceivedPartial<messageData, messageContext, endOfMessage>
]]></artwork>
            <t>If a complete Message cannot be delivered in one event, one part of the Message
can be delivered with a ReceivedPartial event. To continue to receive more
of the same Message, the application must invoke Receive again.</t>
            <t>Multiple invocations of ReceivedPartial deliver data for the same Message by
passing the same MessageContext, until the endOfMessage flag is delivered or a
ReceiveError occurs. All partial blocks of a single Message are delivered in
order without gaps. This event does not support delivering discontiguous partial
Messages. If, for example, Message A is divided into three pieces (A1, A2, A3) and
Message B is divided into three pieces (B1, B2, B3), and preserveOrder is not Required,
the ReceivedPartial may deliver them in a sequence like this: A1, B1, B2, A2, A3, B3,
because the messageContext allows the application to identify the pieces as belonging
to Message A and B, respectively. However, a sequence like: A1, A3 will never occur.</t>
            <t>If the minIncompleteLength in the Receive request was set to be infinite (indicating
a request to receive only complete Messages), the ReceivedPartial event may still be
delivered if one of the following conditions is true:</t>
            <ul spacing="normal">
              <li>the underlying Protocol Stack supports message boundary preservation, and
the size of the Message is larger than the buffers available for a single
message;</li>
              <li>the underlying Protocol Stack does not support message boundary
preservation, and the Message Framer (see <xref target="framing"/>) cannot determine
the end of the message using the buffer space it has available; or</li>
              <li>the underlying Protocol Stack does not support message boundary
preservation, and no Message Framer was supplied by the application</li>
            </ul>
            <t>Note that in the absence of message boundary preservation or
a Message Framer, all bytes received on the Connection will be represented as one
large Message of indeterminate length.</t>
            <t>In the following example, an application only wants to receive up to 1000 bytes
at a time from a Connection. If a 1500-byte message arrives, it would receive
the message in two separate ReceivedPartial events.</t>
            <artwork><![CDATA[
Connection.Receive(1, 1000)

// Receive first 1000 bytes, message is incomplete
Connection -> ReceivedPartial<messageData(1000 bytes), messageContext, false>

Connection.Receive(1, 1000)

// Receive last 500 bytes, message is now complete
Connection -> ReceivedPartial<messageData(500 bytes), messageContext, true>
]]></artwork>
          </section>
          <section anchor="receive-error">
            <name>ReceiveError</name>
            <artwork><![CDATA[
Connection -> ReceiveError<messageContext, reason?>
]]></artwork>
            <t>A ReceiveError occurs when data is received by the underlying Protocol Stack
that cannot be fully retrieved or parsed, and when it is useful for the application
to be notified of such errors. For example, a ReceiveError can
indicate that a Message (identified via the MessageContext)
that was being partially received previously, but had not
completed, encountered an error and will not be completed. This can be useful
for an application, which may want to use this error as a hint to remove
previously received Message parts from memory. As another example,
if an incoming Message does not fulfill the <tt>recvChecksumLen</tt> property
(see <xref target="conn-recv-checksum"/>),
an application can use this error as a hint to inform the peer application
to adjust the <tt>msgChecksumLen</tt> property (see <xref target="msg-checksum"/>).</t>
            <t>In contrast, internal protocol reception errors (e.g., loss causing retransmissions
in TCP) are not signalled by this Event. Conditions that irrevocably lead to
the termination of the Connection are signaled using ConnectionError
(see <xref target="termination"/>).</t>
          </section>
        </section>
        <section anchor="recv-meta">
          <name>Receive Message Properties</name>
          <t>Each Message Context may contain metadata from protocols in the Protocol Stack;
which metadata is available is Protocol Stack dependent. These are exposed through additional read-only Message Properties that can be queried from the MessageContext object (see <xref target="msg-ctx"/>) passed by the receive event.
The following metadata values are supported:</t>
          <section anchor="receive-ecn">
            <name>UDP(-Lite)-specific Property: ECN</name>
            <t>When available, Message metadata carries the value of the Explicit Congestion
Notification (ECN) field. This information can be used for logging and debugging,
and for building applications that need access to information about
the transport internals for their own operation. This property is specific to UDP
and UDP-Lite because these protocols do not implement congestion control,
and hence expose this functionality to the application (see <xref target="RFC8293"/>,
following the guidance in <xref target="RFC8085"/>)</t>
          </section>
          <section anchor="receive-early">
            <name>Early Data</name>
            <t>In some cases it can be valuable to know whether data was read as part of early
data transfer (before connection establishment has finished). This is useful if
applications need to treat early data separately,
e.g., if early data has different security properties than data sent after
connection establishment. In the case of TLS 1.3, client early data can be replayed
maliciously (see <xref target="RFC8446"/>). Thus, receivers might wish to perform additional
checks for early data to ensure it is safely replayable. If TLS 1.3 is available
and the recipient Message was sent as part of early data, the corresponding metadata carries
a flag indicating as such. If early data is enabled, applications should check this metadata
field for Messages received during connection establishment and respond accordingly.</t>
          </section>
          <section anchor="receiving-final-messages">
            <name>Receiving Final Messages</name>
            <t>The Message Context can indicate whether or not this Message is
the Final Message on a Connection. For any Message that is marked as Final,
the application can assume that there will be no more Messages received on the
Connection once the Message has been completely delivered. This corresponds
to the <tt>final</tt> property that may be marked on a sent Message, see <xref target="msg-final"/>.</t>
            <t>Some transport protocols and peers do not support signaling of the <tt>final</tt> property.
Applications therefore should not rely on receiving a Message marked Final to know
that the sending endpoint is done sending on a connection.</t>
            <t>Any calls to Receive once the Final Message has been delivered will result in errors.</t>
          </section>
        </section>
      </section>
    </section>
    <section anchor="termination">
      <name>Connection Termination</name>
      <t>A Connection can be terminated i) by the Local Endpoint (i.e., the application calls the Close, CloseGroup, Abort or AbortGroup Action), ii) by the Remote Endpoint (i.e., the remote application calls the Close, CloseGroup, Abort or AbortGroup Action), or iii) because of an error (e.g., a timeout). A local call of the Close Action will
cause the Connection to either send a Closed Event or a ConnectionError Event, and a local call of
the CloseGroup Action will cause all of the Connections in the group to either send a Closed Event
or a ConnectionError Event. A local call of the Abort Action will cause the Connection to send
a ConnectionError Event, indicating local Abort as a reason, and a local call of the AbortGroup Action
will cause all of the Connections in the group to send a ConnectionError Event, indicating local Abort
as a reason.</t>
      <t>Remote Action calls map to Events similar to local calls (e.g., a remote Close causes the
Connection to either send a Closed Event or a ConnectionError Event), but, different from local Action calls,
it is not guaranteed that such Events will indeed be invoked. When an application needs to free resources
associated with a Connection, it should therefore not rely on the invocation of such Events due to
termination calls from the Remote Endpoint, but instead use the local termination Actions.</t>
      <t>Close terminates a Connection after satisfying all the requirements that were
specified regarding the delivery of Messages that the application has already
given to the Transport Services system. Upon successfully satisfying all these
requirements, the Connection will send the Closed Event. For example, if reliable delivery was requested
for a Message handed over before calling Close, the Closed Event will signify
that this Message has indeed been delivered. This Action does not affect any other Connection
in the same Connection Group.</t>
      <artwork><![CDATA[
Connection.Close()
]]></artwork>
      <t>The Closed Event informs the application that a Close Action has successfully
completed, or that the Remote Endpoint has closed the Connection.
There is no guarantee that a remote Close will be signaled.</t>
      <artwork><![CDATA[
Connection -> Closed<>
]]></artwork>
      <t>Abort terminates a Connection without delivering any remaining Messages. This action does
not affect any other Connection that is entangled with this one in a Connection Group.
When the Abort Action has finished, the Connection will send a ConnectionError Event,
indicating local Abort as a reason.</t>
      <artwork><![CDATA[
Connection.Abort()
]]></artwork>
      <t>CloseGroup gracefully terminates a Connection and any other Connections in the
same Connection Group. For example, all of the Connections in a
group might be streams of a single session for a multistreaming protocol; closing the entire
group will close the underlying session. See also <xref target="groups"/>. All Connections in the group
will send a Closed Event when the CloseGroup Action was successful.
As with Close, any Messages
remaining to be processed on a Connection will be handled prior to closing.</t>
      <artwork><![CDATA[
Connection.CloseGroup()
]]></artwork>
      <t>AbortGroup terminates a Connection and any other Connections that are
in the same Connection Group without delivering any remaining Messages.
When the AbortGroup Action has finished, all Connections in the group will
send a ConnectionError Event, indicating local Abort as a reason.</t>
      <artwork><![CDATA[
Connection.AbortGroup()
]]></artwork>
      <t>A ConnectionError informs the application that: 1) data could not be delivered to the
peer after a timeout,
or 2) the Connection has been aborted (e.g., because the peer has called Abort).
There is no guarantee that an Abort from the peer will be signaled.</t>
      <artwork><![CDATA[
Connection -> ConnectionError<reason?>
]]></artwork>
    </section>
    <section anchor="connection-state-and-ordering-of-operations-and-events">
      <name>Connection State and Ordering of Operations and Events</name>
      <t>This Transport Services API is designed to be independent of an implementation's
concurrency model.  The details of how exactly actions are handled, and how
events are dispatched, are implementation dependent.</t>
      <t>Each transition of connection state is associated with one of more events:</t>
      <ul spacing="normal">
        <li>Ready&lt;&gt; occurs when a Connection created with Initiate() or
InitiateWithSend() transitions to Established state.</li>
        <li>ConnectionReceived&lt;&gt; occurs when a Connection created with Listen()
transitions to Established state.</li>
        <li>RendezvousDone&lt;&gt; occurs when a Connection created with Rendezvous()
transitions to Established state.</li>
        <li>Closed&lt;&gt; occurs when a Connection transitions to Closed state without error.</li>
        <li>EstablishmentError&lt;&gt; occurs when a Connection created with Initiate() transitions
from Establishing state to Closed state due to an error.</li>
        <li>ConnectionError&lt;&gt; occurs when a Connection transitions to Closed state due to
an error in all other circumstances.</li>
      </ul>
      <t>The following diagram shows the possible states of a Connection and the
events that occur upon a transition from one state to another.</t>
      <figure anchor="fig-connstates">
        <name>Connection State Diagram</name>
        <artwork><![CDATA[
              (*)                               (**)
Establishing -----> Established -----> Closing ------> Closed
     |                                                   ^
     |                                                   |
     +---------------------------------------------------+
                  EstablishmentError<>

(*) Ready<>, ConnectionReceived<>, RendezvousDone<>
(**) Closed<>, ConnectionError<>

]]></artwork>
      </figure>
      <t>The Transport Services API  provides the following guarantees about the ordering of
 operations:</t>
      <ul spacing="normal">
        <li>Sent&lt;&gt; events will occur on a Connection in the order in which the Messages
were sent (i.e., delivered to the kernel or to the network interface,
depending on implementation).</li>
        <li>Received&lt;&gt; will never occur on a Connection before it is Established; i.e.
before a Ready&lt;&gt; event on that Connection, or a ConnectionReceived&lt;&gt; or
RendezvousDone&lt;&gt; containing that Connection.</li>
        <li>No events will occur on a Connection after it is Closed; i.e., after a
Closed&lt;&gt; event, an EstablishmentError&lt;&gt; or ConnectionError&lt;&gt; will not occur on that connection. To
ensure this ordering, Closed&lt;&gt; will not occur on a Connection while other
events on the Connection are still locally outstanding (i.e., known to the
Transport Services API and waiting to be dealt with by the application).</li>
      </ul>
    </section>
    <section anchor="iana-considerations">
      <name>IANA Considerations</name>
      <t>RFC-EDITOR: Please remove this section before publication.</t>
      <t>This document has no Actions for IANA.
Later versions of this document may create IANA registries for generic transport property names and transport property namespaces (see <xref target="property-names"/>).</t>
    </section>
    <section anchor="privacy-security">
      <name>Privacy and Security Considerations</name>
      <t>This document describes a generic API for interacting with a Transport Services system.
Part of this API includes configuration details for transport security protocols, as discussed
in <xref target="security-parameters"/>. It does not recommend use (or disuse) of specific
algorithms or protocols. Any API-compatible transport security protocol ought to work in a Transport Services system.
Security considerations for these protocols are discussed in the respective specifications.</t>
      <t>The described API is used to exchange information between an application and the Transport Services system. While
it is not necessarily expected that both systems are implemented by the same authority, it is expected
that the Transport Services system implementation is either provided as a library that is selected by the application
from a trusted party, or that it is part of the operating system that the application also relies on for
other tasks.</t>
      <t>In either case, the Transport Services API is an internal interface that is used to change information locally between two systems.
However, as the Transport Services system is responsible for network communication, it is in the position to
potentially share any information provided by the application with the network or another communication peer.
Most of the information provided over the Transport Services API are useful to configure and select protocols and paths
and are not necessarily privacy sensitive. Still, some information could be privacy sensitive because
it might reveal usage characteristics and habits of the user of an application.</t>
      <t>Of course any communication over a network reveals usage characteristics, because all
packets, as well as their timing and size, are part of the network-visible wire image <xref target="RFC8546"/>. However,
the selection of a protocol and its configuration also impacts which information is visible, potentially in
clear text, and which other entities can access it. In most cases, information provided for protocol and path selection
should not directly translate to information that can be observed by network devices on the path.
However, there might be specific configuration
information that is intended for path exposure, e.g., a DiffServ codepoint setting, that is either provided
directly by the application or indirectly configured for a traffic profile.</t>
      <t>Applications should be aware that communication attempts can lead to more than one connection establishment.
This is the case, for example, when the Transport Services system also executes name resolution, when support mechanisms such as
TURN or ICE are used to establish connectivity, if protocols or paths are raised, or if a path fails and
fallback or re-establishment is supported in the Transport Services system.</t>
      <t>Applications should also take care to not assume that all data received using the Transport Services API is always
complete or well-formed. Specifically, messages that are received partially <xref target="receive-partial"/> could be a source
of truncation attacks if applications do not distinguish between partial messages and complete messages.</t>
      <t>The Transport Services API explicitly does not require the application to resolve names, though there is
a tradeoff between early and late binding of addresses to names. Early binding
allows the API implementation to reduce connection setup latency, at the cost
of potentially limited scope for alternate path discovery during Connection
establishment, as well as potential additional information leakage about
application interest when used with a resolution method (such as DNS without
TLS) which does not protect query confidentiality.</t>
      <t>These communication activities are not different from what is used today. However,
the goal of a Transport Services system is to support
such mechanisms as a generic service within the transport layer. This enables applications to more dynamically
benefit from innovations and new protocols in the transport, although it reduces transparency of the
underlying communication actions to the application itself. The Transport Services API is designed such that protocol and path selection
can be limited to a small and controlled set if required by the application for functional or security purposes. Further,
a Transport Services system should provide an interface to poll information about which protocol and path is currently in use as
well as provide logging about the communication events of each connection.</t>
    </section>
    <section anchor="acknowledgements">
      <name>Acknowledgements</name>
      <t>This work has received funding from the European Union's Horizon 2020 research and
innovation programme under grant agreements No. 644334 (NEAT) and No. 688421 (MAMI).</t>
      <t>This work has been supported by Leibniz Prize project funds of DFG - German
Research Foundation: Gottfried Wilhelm Leibniz-Preis 2011 (FKZ FE 570/4-1).</t>
      <t>This work has been supported by the UK Engineering and Physical Sciences
Research Council under grant EP/R04144X/1.</t>
      <t>This work has been supported by the Research Council of Norway under its "Toppforsk"
programme through the "OCARINA" project.</t>
      <t>Thanks to Stuart Cheshire, Josh Graessley, David Schinazi, and Eric Kinnear for
their implementation and design efforts, including Happy Eyeballs, that heavily
influenced this work. Thanks to Laurent Chuat and Jason Lee for initial work on
the Post Sockets interface, from which this work has evolved. Thanks to
Maximilian Franke for asking good questions based on implementation experience
and for contributing text, e.g., on multicast.</t>
    </section>
  </middle>
  <back>
    <references>
      <name>References</name>
      <references>
        <name>Normative References</name>
        <reference anchor="I-D.ietf-taps-arch">
          <front>
            <title>An Architecture for Transport Services</title>
            <author fullname="Tommy Pauly">
              <organization>Apple Inc.</organization>
            </author>
            <author fullname="Brian Trammell">
              <organization>Google Switzerland GmbH</organization>
            </author>
            <author fullname="Anna Brunstrom">
              <organization>Karlstad University</organization>
            </author>
            <author fullname="Godred Fairhurst">
              <organization>University of Aberdeen</organization>
            </author>
            <author fullname="Colin Perkins">
              <organization>University of Glasgow</organization>
            </author>
            <date day="27" month="June" year="2022"/>
            <abstract>
              <t>   This document describes an architecture for exposing transport
   protocol features to applications for network communication, a
   Transport Services system.  The Transport Services Application
   Programming Interface (API) is based on an asynchronous, event-driven
   interaction pattern.  This API uses messages for representing data
   transfer to applications, and describes how implementations can use
   multiple IP addresses, multiple protocols, and multiple paths, and
   provide multiple application streams.  This document further defines
   common terminology and concepts to be used in definitions of a
   Transport Service API and a Transport Services implementation.

              </t>
            </abstract>
          </front>
          <seriesInfo name="Internet-Draft" value="draft-ietf-taps-arch-13"/>
        </reference>
        <reference anchor="RFC2119">
          <front>
            <title>Key words for use in RFCs to Indicate Requirement Levels</title>
            <author fullname="S. Bradner" initials="S." surname="Bradner">
              <organization/>
            </author>
            <date month="March" year="1997"/>
            <abstract>
              <t>In many standards track documents several words are used to signify the requirements in the specification.  These words are often capitalized. This document defines these words as they should be interpreted in IETF documents.  This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements.</t>
            </abstract>
          </front>
          <seriesInfo name="BCP" value="14"/>
          <seriesInfo name="RFC" value="2119"/>
          <seriesInfo name="DOI" value="10.17487/RFC2119"/>
        </reference>
        <reference anchor="RFC8174">
          <front>
            <title>Ambiguity of Uppercase vs Lowercase in RFC 2119 Key Words</title>
            <author fullname="B. Leiba" initials="B." surname="Leiba">
              <organization/>
            </author>
            <date month="May" year="2017"/>
            <abstract>
              <t>RFC 2119 specifies common key words that may be used in protocol  specifications.  This document aims to reduce the ambiguity by clarifying that only UPPERCASE usage of the key words have the  defined special meanings.</t>
            </abstract>
          </front>
          <seriesInfo name="BCP" value="14"/>
          <seriesInfo name="RFC" value="8174"/>
          <seriesInfo name="DOI" value="10.17487/RFC8174"/>
        </reference>
        <reference anchor="RFC2914">
          <front>
            <title>Congestion Control Principles</title>
            <author fullname="S. Floyd" initials="S." surname="Floyd">
              <organization/>
            </author>
            <date month="September" year="2000"/>
            <abstract>
              <t>The goal of this document is to explain the need for congestion control in the Internet, and to discuss what constitutes correct congestion control.  This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements.</t>
            </abstract>
          </front>
          <seriesInfo name="BCP" value="41"/>
          <seriesInfo name="RFC" value="2914"/>
          <seriesInfo name="DOI" value="10.17487/RFC2914"/>
        </reference>
        <reference anchor="RFC8084">
          <front>
            <title>Network Transport Circuit Breakers</title>
            <author fullname="G. Fairhurst" initials="G." surname="Fairhurst">
              <organization/>
            </author>
            <date month="March" year="2017"/>
            <abstract>
              <t>This document explains what is meant by the term "network transport                          Circuit Breaker".  It describes the need for Circuit Breakers (CBs) for network tunnels and applications when using non-congestion- controlled traffic and explains where CBs are, and are not, needed. It also defines requirements for building a CB and the expected outcomes of using a CB within the Internet.</t>
            </abstract>
          </front>
          <seriesInfo name="BCP" value="208"/>
          <seriesInfo name="RFC" value="8084"/>
          <seriesInfo name="DOI" value="10.17487/RFC8084"/>
        </reference>
        <reference anchor="RFC8981">
          <front>
            <title>Temporary Address Extensions for Stateless Address Autoconfiguration in IPv6</title>
            <author fullname="F. Gont" initials="F." surname="Gont">
              <organization/>
            </author>
            <author fullname="S. Krishnan" initials="S." surname="Krishnan">
              <organization/>
            </author>
            <author fullname="T. Narten" initials="T." surname="Narten">
              <organization/>
            </author>
            <author fullname="R. Draves" initials="R." surname="Draves">
              <organization/>
            </author>
            <date month="February" year="2021"/>
            <abstract>
              <t>This document describes an extension to IPv6 Stateless Address Autoconfiguration that causes hosts to generate temporary addresses with randomized interface identifiers for each prefix advertised with autoconfiguration enabled. Changing addresses over time limits the window of time during which eavesdroppers and other information collectors may trivially perform address-based network-activity correlation when the same address is employed for multiple transactions by the same host. Additionally, it reduces the window of exposure of a host as being accessible via an address that becomes revealed as a result of active communication. This document obsoletes RFC 4941.</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="8981"/>
          <seriesInfo name="DOI" value="10.17487/RFC8981"/>
        </reference>
        <reference anchor="RFC8085">
          <front>
            <title>UDP Usage Guidelines</title>
            <author fullname="L. Eggert" initials="L." surname="Eggert">
              <organization/>
            </author>
            <author fullname="G. Fairhurst" initials="G." surname="Fairhurst">
              <organization/>
            </author>
            <author fullname="G. Shepherd" initials="G." surname="Shepherd">
              <organization/>
            </author>
            <date month="March" year="2017"/>
            <abstract>
              <t>The User Datagram Protocol (UDP) provides a minimal message-passing transport that has no inherent congestion control mechanisms.  This document provides guidelines on the use of UDP for the designers of applications, tunnels, and other protocols that use UDP.  Congestion control guidelines are a primary focus, but the document also provides guidance on other topics, including message sizes, reliability, checksums, middlebox traversal, the use of Explicit Congestion Notification (ECN), Differentiated Services Code Points (DSCPs), and ports.</t>
              <t>Because congestion control is critical to the stable operation of the Internet, applications and other protocols that choose to use UDP as an Internet transport must employ mechanisms to prevent congestion collapse and to establish some degree of fairness with concurrent traffic.  They may also need to implement additional mechanisms, depending on how they use UDP.</t>
              <t>Some guidance is also applicable to the design of other protocols (e.g., protocols layered directly on IP or via IP-based tunnels), especially when these protocols do not themselves provide congestion control.</t>
              <t>This document obsoletes RFC 5405 and adds guidelines for multicast UDP usage.</t>
            </abstract>
          </front>
          <seriesInfo name="BCP" value="145"/>
          <seriesInfo name="RFC" value="8085"/>
          <seriesInfo name="DOI" value="10.17487/RFC8085"/>
        </reference>
        <reference anchor="RFC8446">
          <front>
            <title>The Transport Layer Security (TLS) Protocol Version 1.3</title>
            <author fullname="E. Rescorla" initials="E." surname="Rescorla">
              <organization/>
            </author>
            <date month="August" year="2018"/>
            <abstract>
              <t>This document specifies version 1.3 of the Transport Layer Security (TLS) protocol.  TLS allows client/server applications to communicate over the Internet in a way that is designed to prevent eavesdropping, tampering, and message forgery.</t>
              <t>This document updates RFCs 5705 and 6066, and obsoletes RFCs 5077, 5246, and 6961.  This document also specifies new requirements for TLS 1.2 implementations.</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="8446"/>
          <seriesInfo name="DOI" value="10.17487/RFC8446"/>
        </reference>
        <reference anchor="RFC8303">
          <front>
            <title>On the Usage of Transport Features Provided by IETF Transport Protocols</title>
            <author fullname="M. Welzl" initials="M." surname="Welzl">
              <organization/>
            </author>
            <author fullname="M. Tuexen" initials="M." surname="Tuexen">
              <organization/>
            </author>
            <author fullname="N. Khademi" initials="N." surname="Khademi">
              <organization/>
            </author>
            <date month="February" year="2018"/>
            <abstract>
              <t>This document describes how the transport protocols Transmission Control Protocol (TCP), MultiPath TCP (MPTCP), Stream Control Transmission Protocol (SCTP), User Datagram Protocol (UDP), and Lightweight User Datagram Protocol (UDP-Lite) expose services to applications and how an application can configure and use the features that make up these services.  It also discusses the service provided by the Low Extra Delay Background Transport (LEDBAT) congestion control mechanism.  The description results in a set of transport abstractions that can be exported in a transport services (TAPS) API.</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="8303"/>
          <seriesInfo name="DOI" value="10.17487/RFC8303"/>
        </reference>
      </references>
      <references>
        <name>Informative References</name>
        <reference anchor="I-D.ietf-taps-impl">
          <front>
            <title>Implementing Interfaces to Transport Services</title>
            <author fullname="Anna Brunstrom">
              <organization>Karlstad University</organization>
            </author>
            <author fullname="Tommy Pauly">
              <organization>Apple Inc.</organization>
            </author>
            <author fullname="Theresa Enghardt">
              <organization>Netflix</organization>
            </author>
            <author fullname="Philipp S. Tiesel">
              <organization>SAP SE</organization>
            </author>
            <author fullname="Michael Welzl">
              <organization>University of Oslo</organization>
            </author>
            <date day="7" month="March" year="2022"/>
            <abstract>
              <t>   The Transport Services system enables applications to use transport
   protocols flexibly for network communication and defines a protocol-
   independent Transport Services Application Programming Interface
   (API) that is based on an asynchronous, event-driven interaction
   pattern.  This document serves as a guide to implementation on how to
   build such a system.

              </t>
            </abstract>
          </front>
          <seriesInfo name="Internet-Draft" value="draft-ietf-taps-impl-12"/>
        </reference>
        <reference anchor="RFC7556">
          <front>
            <title>Multiple Provisioning Domain Architecture</title>
            <author fullname="D. Anipko" initials="D." role="editor" surname="Anipko">
              <organization/>
            </author>
            <date month="June" year="2015"/>
            <abstract>
              <t>This document is a product of the work of the Multiple Interfaces Architecture Design team.  It outlines a solution framework for some of the issues experienced by nodes that can be attached to multiple networks simultaneously.  The framework defines the concept of a Provisioning Domain (PvD), which is a consistent set of network configuration information.  PvD-aware nodes learn PvD-specific information from the networks they are attached to and/or other sources.  PvDs are used to enable separation and configuration consistency in the presence of multiple concurrent connections.</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="7556"/>
          <seriesInfo name="DOI" value="10.17487/RFC7556"/>
        </reference>
        <reference anchor="TCP-COUPLING">
          <front>
            <title>ctrlTCP: Reducing Latency through Coupled, Heterogeneous Multi-Flow TCP Congestion Control</title>
            <author initials="S." surname="Islam" fullname="Safiqul Islam">
              <organization/>
            </author>
            <author initials="M." surname="Welzl" fullname="Michael Welzl">
              <organization/>
            </author>
            <author initials="K." surname="Hiorth" fullname="Kristian Hiorth">
              <organization/>
            </author>
            <author initials="D." surname="Hayes" fullname="David Hayes">
              <organization/>
            </author>
            <author initials="G." surname="Armitage" fullname="Grenville Armitage">
              <organization/>
            </author>
            <author initials="S." surname="Gjessing" fullname="Stein Gjessing">
              <organization/>
            </author>
            <date year="2018"/>
          </front>
          <seriesInfo name="IEEE INFOCOM Global Internet Symposium (GI) workshop (GI 2018)" value=""/>
        </reference>
        <reference anchor="RFC8095">
          <front>
            <title>Services Provided by IETF Transport Protocols and Congestion Control Mechanisms</title>
            <author fullname="G. Fairhurst" initials="G." role="editor" surname="Fairhurst">
              <organization/>
            </author>
            <author fullname="B. Trammell" initials="B." role="editor" surname="Trammell">
              <organization/>
            </author>
            <author fullname="M. Kuehlewind" initials="M." role="editor" surname="Kuehlewind">
              <organization/>
            </author>
            <date month="March" year="2017"/>
            <abstract>
              <t>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.</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="8095"/>
          <seriesInfo name="DOI" value="10.17487/RFC8095"/>
        </reference>
        <reference anchor="RFC8923">
          <front>
            <title>A Minimal Set of Transport Services for End Systems</title>
            <author fullname="M. Welzl" initials="M." surname="Welzl">
              <organization/>
            </author>
            <author fullname="S. Gjessing" initials="S." surname="Gjessing">
              <organization/>
            </author>
            <date month="October" year="2020"/>
            <abstract>
              <t>This document recommends a minimal set of Transport Services offered by end systems and gives guidance on choosing among the available mechanisms and protocols. It is based on the set of transport features in RFC 8303.</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="8923"/>
          <seriesInfo name="DOI" value="10.17487/RFC8923"/>
        </reference>
        <reference anchor="RFC8922">
          <front>
            <title>A Survey of the Interaction between Security Protocols and Transport Services</title>
            <author fullname="T. Enghardt" initials="T." surname="Enghardt">
              <organization/>
            </author>
            <author fullname="T. Pauly" initials="T." surname="Pauly">
              <organization/>
            </author>
            <author fullname="C. Perkins" initials="C." surname="Perkins">
              <organization/>
            </author>
            <author fullname="K. Rose" initials="K." surname="Rose">
              <organization/>
            </author>
            <author fullname="C. Wood" initials="C." surname="Wood">
              <organization/>
            </author>
            <date month="October" year="2020"/>
            <abstract>
              <t>This document provides a survey of commonly used or notable network security protocols, with a focus on how they interact and integrate with applications and transport protocols. Its goal is to supplement efforts to define and catalog Transport Services by describing the interfaces required to add security protocols. This survey is not limited to protocols developed within the scope or context of the IETF, and those included represent a superset of features a Transport Services system may need to support.</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="8922"/>
          <seriesInfo name="DOI" value="10.17487/RFC8922"/>
        </reference>
        <reference anchor="RFC3376">
          <front>
            <title>Internet Group Management Protocol, Version 3</title>
            <author fullname="B. Cain" initials="B." surname="Cain">
              <organization/>
            </author>
            <author fullname="S. Deering" initials="S." surname="Deering">
              <organization/>
            </author>
            <author fullname="I. Kouvelas" initials="I." surname="Kouvelas">
              <organization/>
            </author>
            <author fullname="B. Fenner" initials="B." surname="Fenner">
              <organization/>
            </author>
            <author fullname="A. Thyagarajan" initials="A." surname="Thyagarajan">
              <organization/>
            </author>
            <date month="October" year="2002"/>
          </front>
          <seriesInfo name="RFC" value="3376"/>
          <seriesInfo name="DOI" value="10.17487/RFC3376"/>
        </reference>
        <reference anchor="RFC4604">
          <front>
            <title>Using Internet Group Management Protocol Version 3 (IGMPv3) and Multicast Listener Discovery Protocol Version 2 (MLDv2) for Source-Specific Multicast</title>
            <author fullname="H. Holbrook" initials="H." surname="Holbrook">
              <organization/>
            </author>
            <author fullname="B. Cain" initials="B." surname="Cain">
              <organization/>
            </author>
            <author fullname="B. Haberman" initials="B." surname="Haberman">
              <organization/>
            </author>
            <date month="August" year="2006"/>
            <abstract>
              <t>The Internet Group Management Protocol Version 3 (IGMPv3) and the Multicast Listener Discovery Protocol Version 2 (MLDv2) are protocols that allow a host to inform its neighboring routers of its desire to receive IPv4 and IPv6 multicast transmissions, respectively. Source-specific multicast (SSM) is a form of multicast in which a receiver is required to specify both the network-layer address of the source and the multicast destination address in order to receive the multicast transmission.  This document defines the notion of an "SSM-aware" router and host, and clarifies and (in some cases) modifies the behavior of IGMPv3 and MLDv2 on SSM-aware routers and hosts to accommodate source-specific multicast.  This document updates the IGMPv3 and MLDv2 specifications.  [STANDARDS-TRACK]</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="4604"/>
          <seriesInfo name="DOI" value="10.17487/RFC4604"/>
        </reference>
        <reference anchor="RFC5280">
          <front>
            <title>Internet X.509 Public Key Infrastructure Certificate and Certificate Revocation List (CRL) Profile</title>
            <author fullname="D. Cooper" initials="D." surname="Cooper">
              <organization/>
            </author>
            <author fullname="S. Santesson" initials="S." surname="Santesson">
              <organization/>
            </author>
            <author fullname="S. Farrell" initials="S." surname="Farrell">
              <organization/>
            </author>
            <author fullname="S. Boeyen" initials="S." surname="Boeyen">
              <organization/>
            </author>
            <author fullname="R. Housley" initials="R." surname="Housley">
              <organization/>
            </author>
            <author fullname="W. Polk" initials="W." surname="Polk">
              <organization/>
            </author>
            <date month="May" year="2008"/>
            <abstract>
              <t>This memo profiles the X.509 v3 certificate and X.509 v2 certificate revocation list (CRL) for use in the Internet.  An overview of this approach and model is provided as an introduction.  The X.509 v3 certificate format is described in detail, with additional information regarding the format and semantics of Internet name forms.  Standard certificate extensions are described and two Internet-specific extensions are defined.  A set of required certificate extensions is specified.  The X.509 v2 CRL format is described in detail along with standard and Internet-specific extensions.  An algorithm for X.509 certification path validation is described.  An ASN.1 module and examples are provided in the appendices.  [STANDARDS-TRACK]</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="5280"/>
          <seriesInfo name="DOI" value="10.17487/RFC5280"/>
        </reference>
        <reference anchor="RFC8445">
          <front>
            <title>Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal</title>
            <author fullname="A. Keranen" initials="A." surname="Keranen">
              <organization/>
            </author>
            <author fullname="C. Holmberg" initials="C." surname="Holmberg">
              <organization/>
            </author>
            <author fullname="J. Rosenberg" initials="J." surname="Rosenberg">
              <organization/>
            </author>
            <date month="July" year="2018"/>
            <abstract>
              <t>This document describes a protocol for Network Address Translator (NAT) traversal for UDP-based communication.  This protocol is called Interactive Connectivity Establishment (ICE).  ICE makes use of the Session Traversal Utilities for NAT (STUN) protocol and its extension, Traversal Using Relay NAT (TURN).</t>
              <t>This document obsoletes RFC 5245.</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="8445"/>
          <seriesInfo name="DOI" value="10.17487/RFC8445"/>
        </reference>
        <reference anchor="RFC8489">
          <front>
            <title>Session Traversal Utilities for NAT (STUN)</title>
            <author fullname="M. Petit-Huguenin" initials="M." surname="Petit-Huguenin">
              <organization/>
            </author>
            <author fullname="G. Salgueiro" initials="G." surname="Salgueiro">
              <organization/>
            </author>
            <author fullname="J. Rosenberg" initials="J." surname="Rosenberg">
              <organization/>
            </author>
            <author fullname="D. Wing" initials="D." surname="Wing">
              <organization/>
            </author>
            <author fullname="R. Mahy" initials="R." surname="Mahy">
              <organization/>
            </author>
            <author fullname="P. Matthews" initials="P." surname="Matthews">
              <organization/>
            </author>
            <date month="February" year="2020"/>
            <abstract>
              <t>Session Traversal Utilities for NAT (STUN) is a protocol that serves as a tool for other protocols in dealing with NAT traversal.  It can be used by an endpoint to determine the IP address and port allocated to it by a NAT.  It can also be used to check connectivity between two endpoints and as a keep-alive protocol to maintain NAT bindings. STUN works with many existing NATs and does not require any special behavior from them.</t>
              <t>STUN is not a NAT traversal solution by itself.  Rather, it is a tool to be used in the context of a NAT traversal solution.</t>
              <t>This document obsoletes RFC 5389.</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="8489"/>
          <seriesInfo name="DOI" value="10.17487/RFC8489"/>
        </reference>
        <reference anchor="RFC8656">
          <front>
            <title>Traversal Using Relays around NAT (TURN): Relay Extensions to Session Traversal Utilities for NAT (STUN)</title>
            <author fullname="T. Reddy" initials="T." role="editor" surname="Reddy">
              <organization/>
            </author>
            <author fullname="A. Johnston" initials="A." role="editor" surname="Johnston">
              <organization/>
            </author>
            <author fullname="P. Matthews" initials="P." surname="Matthews">
              <organization/>
            </author>
            <author fullname="J. Rosenberg" initials="J." surname="Rosenberg">
              <organization/>
            </author>
            <date month="February" year="2020"/>
            <abstract>
              <t>If a host is located behind a NAT, it can be impossible for that host to communicate directly with other hosts (peers) in certain situations. In these situations, it is necessary for the host to use the services of an intermediate node that acts as a communication relay. This specification defines a protocol, called "Traversal Using Relays around NAT" (TURN), that allows the host to control the operation of the relay and to exchange packets with its peers using the relay. TURN differs from other relay control protocols in that it allows a client to communicate with multiple peers using a single relay address.</t>
              <t>The TURN protocol was designed to be used as part of the Interactive Connectivity Establishment (ICE) approach to NAT traversal, though it can also be used without ICE.</t>
              <t>This document obsoletes RFCs 5766 and 6156.</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="8656"/>
          <seriesInfo name="DOI" value="10.17487/RFC8656"/>
        </reference>
        <reference anchor="RFC5245">
          <front>
            <title>Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer Protocols</title>
            <author fullname="J. Rosenberg" initials="J." surname="Rosenberg">
              <organization/>
            </author>
            <date month="April" year="2010"/>
            <abstract>
              <t>This document describes a protocol for Network Address Translator (NAT) traversal for UDP-based multimedia sessions established with the offer/answer model.  This protocol is called Interactive Connectivity Establishment (ICE).  ICE makes use of the Session Traversal Utilities for NAT (STUN) protocol and its extension, Traversal Using Relay NAT (TURN).  ICE can be used by any protocol utilizing the offer/answer model, such as the Session Initiation Protocol (SIP).  [STANDARDS-TRACK]</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="5245"/>
          <seriesInfo name="DOI" value="10.17487/RFC5245"/>
        </reference>
        <reference anchor="RFC3261">
          <front>
            <title>SIP: Session Initiation Protocol</title>
            <author fullname="J. Rosenberg" initials="J." surname="Rosenberg">
              <organization/>
            </author>
            <author fullname="H. Schulzrinne" initials="H." surname="Schulzrinne">
              <organization/>
            </author>
            <author fullname="G. Camarillo" initials="G." surname="Camarillo">
              <organization/>
            </author>
            <author fullname="A. Johnston" initials="A." surname="Johnston">
              <organization/>
            </author>
            <author fullname="J. Peterson" initials="J." surname="Peterson">
              <organization/>
            </author>
            <author fullname="R. Sparks" initials="R." surname="Sparks">
              <organization/>
            </author>
            <author fullname="M. Handley" initials="M." surname="Handley">
              <organization/>
            </author>
            <author fullname="E. Schooler" initials="E." surname="Schooler">
              <organization/>
            </author>
            <date month="June" year="2002"/>
            <abstract>
              <t>This document describes Session Initiation Protocol (SIP), an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants.  These sessions include Internet telephone calls, multimedia distribution, and multimedia conferences.  [STANDARDS-TRACK]</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="3261"/>
          <seriesInfo name="DOI" value="10.17487/RFC3261"/>
        </reference>
        <reference anchor="RFC7478">
          <front>
            <title>Web Real-Time Communication Use Cases and Requirements</title>
            <author fullname="C. Holmberg" initials="C." surname="Holmberg">
              <organization/>
            </author>
            <author fullname="S. Hakansson" initials="S." surname="Hakansson">
              <organization/>
            </author>
            <author fullname="G. Eriksson" initials="G." surname="Eriksson">
              <organization/>
            </author>
            <date month="March" year="2015"/>
            <abstract>
              <t>This document describes web-based real-time communication use cases. Requirements on the browser functionality are derived from the use cases.</t>
              <t>This document was developed in an initial phase of the work with rather minor updates at later stages.  It has not really served as a tool in deciding features or scope for the WG's efforts so far.  It is being published to record the early conclusions of the WG.  It will not be used as a set of rigid guidelines that specifications and implementations will be held to in the future.</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="7478"/>
          <seriesInfo name="DOI" value="10.17487/RFC7478"/>
        </reference>
        <reference anchor="RFC8699">
          <front>
            <title>Coupled Congestion Control for RTP Media</title>
            <author fullname="S. Islam" initials="S." surname="Islam">
              <organization/>
            </author>
            <author fullname="M. Welzl" initials="M." surname="Welzl">
              <organization/>
            </author>
            <author fullname="S. Gjessing" initials="S." surname="Gjessing">
              <organization/>
            </author>
            <date month="January" year="2020"/>
            <abstract>
              <t>When multiple congestion-controlled Real-time Transport Protocol (RTP) sessions traverse the same network bottleneck, combining their controls can improve the total on-the-wire behavior in terms of delay, loss, and fairness. This document describes such a method for flows that have the same sender, in a way that is as flexible and simple as possible while minimizing the number of changes needed to existing RTP applications. This document also specifies how to apply the method for the Network-Assisted Dynamic Adaptation (NADA) congestion control algorithm and provides suggestions on how to apply it to other congestion control algorithms.</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="8699"/>
          <seriesInfo name="DOI" value="10.17487/RFC8699"/>
        </reference>
        <reference anchor="RFC8838">
          <front>
            <title>Trickle ICE: Incremental Provisioning of Candidates for the Interactive Connectivity Establishment (ICE) Protocol</title>
            <author fullname="E. Ivov" initials="E." surname="Ivov">
              <organization/>
            </author>
            <author fullname="J. Uberti" initials="J." surname="Uberti">
              <organization/>
            </author>
            <author fullname="P. Saint-Andre" initials="P." surname="Saint-Andre">
              <organization/>
            </author>
            <date month="January" year="2021"/>
            <abstract>
              <t>This document describes "Trickle ICE", an extension to the Interactive Connectivity Establishment (ICE) protocol that enables ICE agents to begin connectivity checks while they are still gathering candidates, by incrementally exchanging candidates over time instead of all at once. This method can considerably accelerate the process of establishing a communication session.</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="8838"/>
          <seriesInfo name="DOI" value="10.17487/RFC8838"/>
        </reference>
        <reference anchor="RFC8260">
          <front>
            <title>Stream Schedulers and User Message Interleaving for the Stream Control Transmission Protocol</title>
            <author fullname="R. Stewart" initials="R." surname="Stewart">
              <organization/>
            </author>
            <author fullname="M. Tuexen" initials="M." surname="Tuexen">
              <organization/>
            </author>
            <author fullname="S. Loreto" initials="S." surname="Loreto">
              <organization/>
            </author>
            <author fullname="R. Seggelmann" initials="R." surname="Seggelmann">
              <organization/>
            </author>
            <date month="November" year="2017"/>
            <abstract>
              <t>The Stream Control Transmission Protocol (SCTP) is a message-oriented transport protocol supporting arbitrarily large user messages.  This document adds a new chunk to SCTP for carrying payload data.  This allows a sender to interleave different user messages that would otherwise result in head-of-line blocking at the sender.  The interleaving of user messages is required for WebRTC data channels.</t>
              <t>Whenever an SCTP sender is allowed to send user data, it may choose from multiple outgoing SCTP streams.  Multiple ways for performing this selection, called stream schedulers, are defined in this document.  A stream scheduler can choose to either implement, or not implement, user message interleaving.</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="8260"/>
          <seriesInfo name="DOI" value="10.17487/RFC8260"/>
        </reference>
        <reference anchor="RFC7657">
          <front>
            <title>Differentiated Services (Diffserv) and Real-Time Communication</title>
            <author fullname="D. Black" initials="D." role="editor" surname="Black">
              <organization/>
            </author>
            <author fullname="P. Jones" initials="P." surname="Jones">
              <organization/>
            </author>
            <date month="November" year="2015"/>
            <abstract>
              <t>This memo describes the interaction between Differentiated Services (Diffserv) network quality-of-service (QoS) functionality and real- time network communication, including communication based on the Real-time Transport Protocol (RTP).  Diffserv is based on network nodes applying different forwarding treatments to packets whose IP headers are marked with different Diffserv Codepoints (DSCPs). WebRTC applications, as well as some conferencing applications, have begun using the Session Description Protocol (SDP) bundle negotiation mechanism to send multiple traffic streams with different QoS requirements using the same network 5-tuple.  The results of using multiple DSCPs to obtain different QoS treatments within a single network 5-tuple have transport protocol interactions, particularly with congestion control functionality (e.g., reordering).  In addition, DSCP markings may be changed or removed between the traffic source and destination.  This memo covers the implications of these Diffserv aspects for real-time network communication, including WebRTC.</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="7657"/>
          <seriesInfo name="DOI" value="10.17487/RFC7657"/>
        </reference>
        <reference anchor="RFC2474">
          <front>
            <title>Definition of the Differentiated Services Field (DS Field) in the IPv4 and IPv6 Headers</title>
            <author fullname="K. Nichols" initials="K." surname="Nichols">
              <organization/>
            </author>
            <author fullname="S. Blake" initials="S." surname="Blake">
              <organization/>
            </author>
            <author fullname="F. Baker" initials="F." surname="Baker">
              <organization/>
            </author>
            <author fullname="D. Black" initials="D." surname="Black">
              <organization/>
            </author>
            <date month="December" year="1998"/>
            <abstract>
              <t>This document defines the IP header field, called the DS (for differentiated services) field.  [STANDARDS-TRACK]</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="2474"/>
          <seriesInfo name="DOI" value="10.17487/RFC2474"/>
        </reference>
        <reference anchor="RFC8622">
          <front>
            <title>A Lower-Effort Per-Hop Behavior (LE PHB) for Differentiated Services</title>
            <author fullname="R. Bless" initials="R." surname="Bless">
              <organization/>
            </author>
            <date month="June" year="2019"/>
            <abstract>
              <t>This document specifies properties and characteristics of a Lower- Effort Per-Hop Behavior (LE PHB).  The primary objective of this LE PHB is to protect Best-Effort (BE) traffic (packets forwarded with the default PHB) from LE traffic in congestion situations, i.e., when resources become scarce, BE traffic has precedence over LE traffic and may preempt it.  Alternatively, packets forwarded by the LE PHB can be associated with a scavenger service class, i.e., they scavenge otherwise-unused resources only.  There are numerous uses for this PHB, e.g., for background traffic of low precedence, such as bulk data transfers with low priority in time, non-time-critical backups, larger software updates, web search engines while gathering information from web servers and so on.  This document recommends a standard Differentiated Services Code Point (DSCP) value for the LE PHB.</t>
              <t>This specification obsoletes RFC 3662 and updates the DSCP recommended in RFCs 4594 and 8325 to use the DSCP assigned in this specification.</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="8622"/>
          <seriesInfo name="DOI" value="10.17487/RFC8622"/>
        </reference>
        <reference anchor="RFC2597">
          <front>
            <title>Assured Forwarding PHB Group</title>
            <author fullname="J. Heinanen" initials="J." surname="Heinanen">
              <organization/>
            </author>
            <author fullname="F. Baker" initials="F." surname="Baker">
              <organization/>
            </author>
            <author fullname="W. Weiss" initials="W." surname="Weiss">
              <organization/>
            </author>
            <author fullname="J. Wroclawski" initials="J." surname="Wroclawski">
              <organization/>
            </author>
            <date month="June" year="1999"/>
            <abstract>
              <t>This document defines a general use Differentiated Services (DS) Per-Hop-Behavior (PHB) Group called Assured Forwarding (AF). [STANDARDS-TRACK]</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="2597"/>
          <seriesInfo name="DOI" value="10.17487/RFC2597"/>
        </reference>
        <reference anchor="RFC3246">
          <front>
            <title>An Expedited Forwarding PHB (Per-Hop Behavior)</title>
            <author fullname="B. Davie" initials="B." surname="Davie">
              <organization/>
            </author>
            <author fullname="A. Charny" initials="A." surname="Charny">
              <organization/>
            </author>
            <author fullname="J.C.R. Bennet" initials="J.C.R." surname="Bennet">
              <organization/>
            </author>
            <author fullname="K. Benson" initials="K." surname="Benson">
              <organization/>
            </author>
            <author fullname="J.Y. Le Boudec" initials="J.Y." surname="Le Boudec">
              <organization/>
            </author>
            <author fullname="W. Courtney" initials="W." surname="Courtney">
              <organization/>
            </author>
            <author fullname="S. Davari" initials="S." surname="Davari">
              <organization/>
            </author>
            <author fullname="V. Firoiu" initials="V." surname="Firoiu">
              <organization/>
            </author>
            <author fullname="D. Stiliadis" initials="D." surname="Stiliadis">
              <organization/>
            </author>
            <date month="March" year="2002"/>
            <abstract>
              <t>This document defines a PHB (per-hop behavior) called Expedited Forwarding (EF).  The PHB is a basic building block in the Differentiated Services architecture.  EF is intended to provide a building block for low delay, low jitter and low loss services by ensuring that the EF aggregate is served at a certain configured rate. This document obsoletes RFC 2598.  [STANDARDS-TRACK]</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="3246"/>
          <seriesInfo name="DOI" value="10.17487/RFC3246"/>
        </reference>
        <reference anchor="RFC5865">
          <front>
            <title>A Differentiated Services Code Point (DSCP) for Capacity-Admitted Traffic</title>
            <author fullname="F. Baker" initials="F." surname="Baker">
              <organization/>
            </author>
            <author fullname="J. Polk" initials="J." surname="Polk">
              <organization/>
            </author>
            <author fullname="M. Dolly" initials="M." surname="Dolly">
              <organization/>
            </author>
            <date month="May" year="2010"/>
            <abstract>
              <t>This document requests one Differentiated Services Code Point (DSCP) from the Internet Assigned Numbers Authority (IANA) for a class of real-time traffic.  This traffic class conforms to the Expedited Forwarding Per-Hop Behavior.  This traffic is also admitted by the network using a Call Admission Control (CAC) procedure involving authentication, authorization, and capacity admission.  This differs from a real-time traffic class that conforms to the Expedited Forwarding Per-Hop Behavior but is not subject to capacity admission or subject to very coarse capacity admission.  [STANDARDS-TRACK]</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="5865"/>
          <seriesInfo name="DOI" value="10.17487/RFC5865"/>
        </reference>
        <reference anchor="RFC4594">
          <front>
            <title>Configuration Guidelines for DiffServ Service Classes</title>
            <author fullname="J. Babiarz" initials="J." surname="Babiarz">
              <organization/>
            </author>
            <author fullname="K. Chan" initials="K." surname="Chan">
              <organization/>
            </author>
            <author fullname="F. Baker" initials="F." surname="Baker">
              <organization/>
            </author>
            <date month="August" year="2006"/>
            <abstract>
              <t>This document describes service classes configured with Diffserv and recommends how they can be used and how to construct them using Differentiated Services Code Points (DSCPs), traffic conditioners, Per-Hop Behaviors (PHBs), and Active Queue Management (AQM) mechanisms.  There is no intrinsic requirement that particular DSCPs, traffic conditioners, PHBs, and AQM be used for a certain service class, but as a policy and for interoperability it is useful to apply them consistently.  This memo provides information for the Internet community.</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="4594"/>
          <seriesInfo name="DOI" value="10.17487/RFC4594"/>
        </reference>
        <reference anchor="RFC8899">
          <front>
            <title>Packetization Layer Path MTU Discovery for Datagram Transports</title>
            <author fullname="G. Fairhurst" initials="G." surname="Fairhurst">
              <organization/>
            </author>
            <author fullname="T. Jones" initials="T." surname="Jones">
              <organization/>
            </author>
            <author fullname="M. Tüxen" initials="M." surname="Tüxen">
              <organization/>
            </author>
            <author fullname="I. Rüngeler" initials="I." surname="Rüngeler">
              <organization/>
            </author>
            <author fullname="T. Völker" initials="T." surname="Völker">
              <organization/>
            </author>
            <date month="September" year="2020"/>
            <abstract>
              <t>This document specifies Datagram Packetization Layer Path MTU Discovery (DPLPMTUD). This is a robust method for Path MTU Discovery (PMTUD) for datagram Packetization Layers (PLs). It allows a PL, or a datagram application that uses a PL, to discover whether a network path can support the current size of datagram.  This can be used to detect and reduce the message size when a sender encounters a packet black hole. It can also probe a network path to discover whether the maximum packet size can be increased.  This provides functionality for datagram transports that is equivalent to the PLPMTUD specification for TCP, specified in RFC 4821, which it updates. It also updates the UDP Usage Guidelines to refer to this method for use with UDP datagrams and updates SCTP.</t>
              <t>The document provides implementation notes for incorporating Datagram PMTUD into IETF datagram transports or applications that use datagram transports.</t>
              <t>This specification updates RFC 4960, RFC 4821, RFC 6951, RFC 8085, and RFC 8261.</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="8899"/>
          <seriesInfo name="DOI" value="10.17487/RFC8899"/>
        </reference>
        <reference anchor="RFC5482">
          <front>
            <title>TCP User Timeout Option</title>
            <author fullname="L. Eggert" initials="L." surname="Eggert">
              <organization/>
            </author>
            <author fullname="F. Gont" initials="F." surname="Gont">
              <organization/>
            </author>
            <date month="March" year="2009"/>
            <abstract>
              <t>The TCP user timeout controls how long transmitted data may remain unacknowledged before a connection is forcefully closed.  It is a local, per-connection parameter.  This document specifies a new TCP option -- the TCP User Timeout Option -- that allows one end of a TCP connection to advertise its current user timeout value.  This information provides advice to the other end of the TCP connection to adapt its user timeout accordingly.  Increasing the user timeouts on both ends of a TCP connection allows it to survive extended periods without end-to-end connectivity.  Decreasing the user timeouts allows busy servers to explicitly notify their clients that they will maintain the connection state only for a short time without connectivity.  [STANDARDS-TRACK]</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="5482"/>
          <seriesInfo name="DOI" value="10.17487/RFC5482"/>
        </reference>
        <reference anchor="RFC8229">
          <front>
            <title>TCP Encapsulation of IKE and IPsec Packets</title>
            <author fullname="T. Pauly" initials="T." surname="Pauly">
              <organization/>
            </author>
            <author fullname="S. Touati" initials="S." surname="Touati">
              <organization/>
            </author>
            <author fullname="R. Mantha" initials="R." surname="Mantha">
              <organization/>
            </author>
            <date month="August" year="2017"/>
            <abstract>
              <t>This document describes a method to transport Internet Key Exchange Protocol (IKE) and IPsec packets over a TCP connection for traversing network middleboxes that may block IKE negotiation over UDP.  This method, referred to as "TCP encapsulation", involves sending both IKE packets for Security Association establishment and Encapsulating Security Payload (ESP) packets over a TCP connection.  This method is intended to be used as a fallback option when IKE cannot be negotiated over UDP.</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="8229"/>
          <seriesInfo name="DOI" value="10.17487/RFC8229"/>
        </reference>
        <reference anchor="I-D.ietf-httpbis-priority">
          <front>
            <title>Extensible Prioritization Scheme for HTTP</title>
            <author fullname="Kazuho Oku">
              <organization>Fastly</organization>
            </author>
            <author fullname="Lucas Pardue">
              <organization>Cloudflare</organization>
            </author>
            <date day="17" month="January" year="2022"/>
            <abstract>
              <t>This document describes a scheme that allows an HTTP client to communicate its preferences for how the upstream server prioritizes responses to its requests, and also allows a server to hint to a downstream intermediary how its responses should be prioritized when they are forwarded.  This document defines the Priority header field for communicating the initial priority in an HTTP version-independent manner, as well as HTTP/2 and HTTP/3 frames for reprioritizing responses.  These share a common format structure that is designed to provide future extensibility.
              </t>
            </abstract>
          </front>
          <seriesInfo name="Internet-Draft" value="draft-ietf-httpbis-priority-12"/>
        </reference>
        <reference anchor="RFC8293">
          <front>
            <title>A Framework for Multicast in Network Virtualization over Layer 3</title>
            <author fullname="A. Ghanwani" initials="A." surname="Ghanwani">
              <organization/>
            </author>
            <author fullname="L. Dunbar" initials="L." surname="Dunbar">
              <organization/>
            </author>
            <author fullname="M. McBride" initials="M." surname="McBride">
              <organization/>
            </author>
            <author fullname="V. Bannai" initials="V." surname="Bannai">
              <organization/>
            </author>
            <author fullname="R. Krishnan" initials="R." surname="Krishnan">
              <organization/>
            </author>
            <date month="January" year="2018"/>
            <abstract>
              <t>This document provides a framework for supporting multicast traffic in a network that uses Network Virtualization over Layer 3 (NVO3). Both infrastructure multicast and application-specific multicast are discussed.  It describes the various mechanisms that can be used for delivering such traffic as well as the data plane and control plane considerations for each of the mechanisms.</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="8293"/>
          <seriesInfo name="DOI" value="10.17487/RFC8293"/>
        </reference>
        <reference anchor="RFC8546">
          <front>
            <title>The Wire Image of a Network Protocol</title>
            <author fullname="B. Trammell" initials="B." surname="Trammell">
              <organization/>
            </author>
            <author fullname="M. Kuehlewind" initials="M." surname="Kuehlewind">
              <organization/>
            </author>
            <date month="April" year="2019"/>
            <abstract>
              <t>This document defines the wire image, an abstraction of the information available to an on-path non-participant in a networking protocol.  This abstraction is intended to shed light on the implications that increased encryption has for network functions that use the wire image.</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="8546"/>
          <seriesInfo name="DOI" value="10.17487/RFC8546"/>
        </reference>
      </references>
    </references>
    <section anchor="implmapping">
      <name>Implementation Mapping</name>
      <t>The way the concepts from this abstract API map into concrete APIs in a
given language on a given platform largely depends on the features and norms of
the language and the platform. Actions could be implemented as functions or
method calls, for instance, and Events could be implemented via event queues,
handler functions or classes, communicating sequential processes, or other
asynchronous calling conventions.</t>
      <section anchor="types">
        <name>Types</name>
        <t>The basic types mentioned in <xref target="notation"/> typically have natural
correspondences in practical programming languages, perhaps constrained by
implementation-specific limitations. For example:</t>
        <ul spacing="normal">
          <li>An Integer can typically be represented in C by an <tt>int</tt> or <tt>long</tt>, subject
to the underlying platform's ranges for each.</li>
          <li>In C, a Tuple may be represented as a <tt>struct</tt> with one member for each of
the value types in the ordered grouping. In Python, by contrast, a Tuple may
be represented natively as a <tt>tuple</tt>, a sequence of dynamically-typed
elements.</li>
          <li>A Collection may be represented as a <tt>std::set</tt> in C++ or as a <tt>set</tt> in
Python. In C, it may be represented as an array or as a higher-level data
structure with appropriate accessors defined.</li>
        </ul>
        <t>The objects described in <xref target="notation"/> can similarly be represented in
different ways depending on which programming language is used. Objects
like Preconnections, Connections, and Listeners can be long-lived, and
benefit from using object-oriented constructs. Note that in C, these
objects may need to provide a way to release or free their underlying
memory when the application is done using them. For example, since a
Preconnection can be used to initiate multiple Connections, it is the
responsibility of the application to clean up the Preconnection memory
if necessary.</t>
      </section>
      <section anchor="events-and-errors">
        <name>Events and Errors</name>
        <t>This specification treats Events and Errors similarly. Errors, just as any
other Events, may occur asynchronously in network applications. However,
implementations of this API may report Errors synchronously,
according to the error handling idioms of the implementation
platform, where they can be immediately detected, such as by generating an
exception when attempting to initiate a connection with inconsistent
Transport Properties. An error can provide an optional reason to the
application with further details about why the error occurred.</t>
      </section>
      <section anchor="time-duration">
        <name>Time Duration</name>
        <t>Time duration types are implementation-specific.
For instance, it could be a number of seconds, number of milliseconds, or a <tt>struct timeval</tt> in C or a user-defined <tt>Duration</tt> class in C++.</t>
      </section>
    </section>
    <section anchor="convenience-functions">
      <name>Convenience Functions</name>
      <section anchor="preference-conv">
        <name>Adding Preference Properties</name>
        <t>TransportProperties will frequently need to set
Selection Properties of type <tt>Preference</tt>, therefore implementations can provide special actions
for adding each preference level i.e, <tt>TransportProperties.Set(some_property, avoid)
is equivalent to </tt>TransportProperties.Avoid(some_property)`:</t>
        <artwork><![CDATA[
TransportProperties.Require(property)
TransportProperties.Prefer(property)
TransportProperties.Ignore(property)
TransportProperties.Avoid(property)
TransportProperties.Prohibit(property)
]]></artwork>
      </section>
      <section anchor="property-profiles">
        <name>Transport Property Profiles</name>
        <t>To ease the use of the Transport Services API, implementations
can provide a mechanism to create Transport Property objects (see <xref target="selection-props"/>)
that are pre-configured with frequently used sets of properties; the following are
in common use in current applications:</t>
        <section anchor="reliable-inorder-stream">
          <name>reliable-inorder-stream</name>
          <t>This profile provides reliable, in-order transport service with
congestion control.
TCP is an example of a protocol that provides this service.
It should consist of the following properties:</t>
          <table anchor="tabrio">
            <name>reliable-inorder-stream preferences</name>
            <thead>
              <tr>
                <th align="left">Property</th>
                <th align="left">Value</th>
              </tr>
            </thead>
            <tbody>
              <tr>
                <td align="left">reliability</td>
                <td align="left">require</td>
              </tr>
              <tr>
                <td align="left">preserveOrder</td>
                <td align="left">require</td>
              </tr>
              <tr>
                <td align="left">congestionControl</td>
                <td align="left">require</td>
              </tr>
              <tr>
                <td align="left">preserveMsgBoundaries</td>
                <td align="left">ignore</td>
              </tr>
            </tbody>
          </table>
        </section>
        <section anchor="reliable-message">
          <name>reliable-message</name>
          <t>This profile provides message-preserving, reliable, in-order
transport service with congestion control.
SCTP is an example of a protocol that provides this service.
It should consist of the following properties:</t>
          <table anchor="tabrm">
            <name>reliable-message preferences</name>
            <thead>
              <tr>
                <th align="left">Property</th>
                <th align="left">Value</th>
              </tr>
            </thead>
            <tbody>
              <tr>
                <td align="left">reliability</td>
                <td align="left">require</td>
              </tr>
              <tr>
                <td align="left">preserveOrder</td>
                <td align="left">require</td>
              </tr>
              <tr>
                <td align="left">congestionControl</td>
                <td align="left">require</td>
              </tr>
              <tr>
                <td align="left">preserveMsgBoundaries</td>
                <td align="left">require</td>
              </tr>
            </tbody>
          </table>
        </section>
        <section anchor="unreliable-datagram">
          <name>unreliable-datagram</name>
          <t>This profile provides a datagram transport service without any
reliability guarantee.
An example of a protocol that provides this service is UDP.
It consists of the following properties:</t>
          <table anchor="tabud">
            <name>unreliable-datagram preferences</name>
            <thead>
              <tr>
                <th align="left">Property</th>
                <th align="left">Value</th>
              </tr>
            </thead>
            <tbody>
              <tr>
                <td align="left">reliability</td>
                <td align="left">avoid</td>
              </tr>
              <tr>
                <td align="left">preserveOrder</td>
                <td align="left">avoid</td>
              </tr>
              <tr>
                <td align="left">congestionControl</td>
                <td align="left">ignore</td>
              </tr>
              <tr>
                <td align="left">preserveMsgBoundaries</td>
                <td align="left">require</td>
              </tr>
              <tr>
                <td align="left">safelyReplayable</td>
                <td align="left">true</td>
              </tr>
            </tbody>
          </table>
          <t>Applications that choose this Transport Property Profile would
avoid the additional latency that could be introduced
by retransmission or reordering in a transport protocol.</t>
          <t>Applications that choose this Transport Property Profile to reduce latency
should also consider setting an appropriate Capacity Profile Property,
see <xref target="prop-cap-profile"/> and might benefit from controlling checksum
coverage, see <xref target="prop-checksum-control-send"/> and <xref target="prop-checksum-control-receive"/>.</t>
        </section>
      </section>
    </section>
    <section anchor="relationship-to-the-minimal-set-of-transport-services-for-end-systems">
      <name>Relationship to the Minimal Set of Transport Services for End Systems</name>
      <t><xref target="RFC8923"/> identifies a minimal set of transport services that end systems should offer. These services make all non-security-related transport features of TCP, MPTCP, UDP, UDP-Lite, SCTP and LEDBAT available that 1) require interaction with the application, and 2) do not get in the way of a possible implementation over TCP (or, with limitations, UDP). The following text explains how this minimal set is reflected in the present API. For brevity, it is based on the list in Section 4.1 of <xref target="RFC8923"/>, updated according to the discussion in Section 5 of <xref target="RFC8923"/>. The present API covers all elements of this section.
This list is a subset of the transport features in Appendix A of <xref target="RFC8923"/>, which refers to the primitives in "pass 2" (Section 4) of <xref target="RFC8303"/> for further details on the implementation with TCP, MPTCP, UDP, UDP-Lite, SCTP and LEDBAT.</t>
      <ul spacing="normal">
        <li>Connect:
<tt>Initiate</tt> Action (<xref target="initiate"/>).</li>
        <li>Listen:
<tt>Listen</tt> Action (<xref target="listen"/>).</li>
        <li>Specify number of attempts and/or timeout for the first establishment message:
<tt>timeout</tt> parameter of <tt>Initiate</tt> (<xref target="initiate"/>) or <tt>InitiateWithSend</tt> Action (<xref target="initiate-and-send"/>).</li>
        <li>Disable MPTCP:
<tt>multipath</tt> property (<xref target="multipath-mode"/>).</li>
        <li>Hand over a message to reliably transfer (possibly multiple times) before connection establishment:
<tt>InitiateWithSend</tt> Action (<xref target="initiate-and-send"/>).</li>
        <li>Change timeout for aborting connection (using retransmit limit or time value):
<tt>connTimeout</tt> property, using a time value (<xref target="conn-timeout"/>).</li>
        <li>Timeout event when data could not be delivered for too long:
<tt>ConnectionError</tt> Event (<xref target="termination"/>).</li>
        <li>Suggest timeout to the peer:
See "TCP-specific Properties: User Timeout Option (UTO)" (<xref target="tcp-uto"/>).</li>
        <li>Notification of ICMP error message arrival:
<tt>softErrorNotify</tt> (<xref target="prop-soft-error"/>) and <tt>SoftError</tt> Event (<xref target="soft-errors"/>).</li>
        <li>Choose a scheduler to operate between streams of an association:
<tt>connScheduler</tt> property (<xref target="conn-scheduler"/>).</li>
        <li>Configure priority or weight for a scheduler:
<tt>connPriority</tt> property (<xref target="conn-priority"/>).</li>
        <li>"Specify checksum coverage used by the sender" and "Disable checksum when sending":
<tt>msgChecksumLen</tt> property (<xref target="msg-checksum"/>) and <tt>fullChecksumSend</tt> property (<xref target="prop-checksum-control-send"/>).</li>
        <li>"Specify minimum checksum coverage required by receiver" and "Disable checksum requirement when receiving":
<tt>recvChecksumLen</tt> property (<xref target="conn-recv-checksum"/>) and <tt>fullChecksumRecv</tt> property (<xref target="prop-checksum-control-receive"/>).</li>
        <li>"Specify DF field":
<tt>noFragmentation</tt> property (<xref target="send-singular"/>).</li>
        <li>Get max. transport-message size that may be sent using a non-fragmented IP packet from the configured interface:
<tt>singularTransmissionMsgMaxLen</tt> property (<xref target="conn-max-msg-notfrag"/>).</li>
        <li>Get max. transport-message size that may be received from the configured interface:
<tt>recvMsgMaxLen</tt> property (<xref target="conn-max-msg-recv"/>).</li>
        <li>Obtain ECN field:
This is a read-only Message Property of the MessageContext object (see "UDP(-Lite)-specific Property: ECN" <xref target="receive-ecn"/>).</li>
        <li>"Specify DSCP field", "Disable Nagle algorithm", "Enable and configure a <tt>Low Extra Delay Background Transfer</tt>":
as suggested in Section 5.5 of <xref target="RFC8923"/>, these transport features are collectively offered via the <tt>connCapacityProfile</tt> property (<xref target="prop-cap-profile"/>). Per-Message control ("Request not to bundle messages") is offered via the <tt>msgCapacityProfile</tt> property (<xref target="send-profile"/>).</li>
        <li>Close after reliably delivering all remaining data, causing an event informing the application on the other side:
this is offered by the <tt>Close</tt> Action with slightly changed semantics in line with the discussion in Section 5.2 of <xref target="RFC8923"/> (<xref target="termination"/>).</li>
        <li>"Abort without delivering remaining data, causing an event informing the application on the other side" and "Abort without delivering remaining data, not causing an event informing the application on the other side":
this is offered by the <tt>Abort</tt> action without promising that this is signaled to the other side. If it is, a <tt>ConnectionError</tt> Event will be invoked at the peer (<xref target="termination"/>).</li>
        <li>"Reliably transfer data, with congestion control", "Reliably transfer a message, with congestion control" and "Unreliably transfer a message":
data is transferred via the <tt>Send</tt> action (<xref target="sending"/>). Reliability is controlled via the <tt>reliability</tt> (<xref target="prop-reliable"/>) property and the <tt>msgReliable</tt> Message Property (<xref target="msg-reliable-message"/>). Transmitting data as a message or without delimiters is controlled via Message Framers (<xref target="framing"/>). The choice of congestion control is provided via the <tt>congestionControl</tt> property (<xref target="prop-cc"/>).</li>
        <li>Configurable Message Reliability:
the <tt>msgLifetime</tt> Message Property implements a time-based way to configure message reliability (<xref target="msg-lifetime"/>).</li>
        <li>"Ordered message delivery (potentially slower than unordered)" and "Unordered message delivery (potentially faster than ordered)":
these two transport features are controlled via the Message Property <tt>msgOrdered</tt> (<xref target="msg-ordered"/>).</li>
        <li>Request not to delay the acknowledgement (SACK) of a message:
should the protocol support it, this is one of the transport features the Transport Services system can apply when an application uses the <tt>connCapacityProfile</tt> Property (<xref target="prop-cap-profile"/>) or the <tt>msgCapacityProfile</tt> Message Property (<xref target="send-profile"/>) with value <tt>Low Latency/Interactive</tt>.</li>
        <li>Receive data (with no message delimiting):
<tt>Receive</tt> Action (<xref target="receiving"/>) and <tt>Received</tt> Event (<xref target="receive-complete"/>).</li>
        <li>Receive a message:
<tt>Receive</tt> Action (<xref target="receiving"/>) and <tt>Received</tt> Event (<xref target="receive-complete"/>), using Message Framers (<xref target="framing"/>).</li>
        <li>Information about partial message arrival:
<tt>Receive</tt> Action (<xref target="receiving"/>) and <tt>ReceivedPartial</tt> Event (<xref target="receive-partial"/>).</li>
        <li>Notification of send failures:
<tt>Expired</tt> Event (<xref target="expired"/>) and <tt>SendError</tt> Event (<xref target="send-error"/>).</li>
        <li>Notification that the stack has no more user data to send:
applications can obtain this information via the <tt>Sent</tt> Event (<xref target="sent"/>).</li>
        <li>Notification to a receiver that a partial message delivery has been aborted:
<tt>ReceiveError</tt> Event (<xref target="receive-error"/>).</li>
        <li>Notification of Excessive Retransmissions (early warning below abortion threshold):
 <tt>SoftError</tt> Event (<xref target="soft-errors"/>).</li>
      </ul>
    </section>
  </back>
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