<?xml version='1.0'?>
<?rfc symrefs='yes'?>
<!DOCTYPE rfc SYSTEM 'rfc2629.dtd'>
<?rfc toc='yes' ?>
<?rfc compact='yes' ?>
<?rfc subcompact='no' ?>
<?rfc sortrefs='no' ?>
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<rfc category='std'
     ipr='trust200902'
     docName='draft-ietf-rtcweb-data-channel-13.txt'> xmlns:xi="http://www.w3.org/2001/XInclude" version="3" category="std" consensus="true" docName="draft-ietf-rtcweb-data-channel-13" indexInclude="true" ipr="trust200902" number="8831" prepTime="2021-01-16T21:17:59" scripts="Common,Latin" sortRefs="true" submissionType="IETF" symRefs="true" tocDepth="3" tocInclude="true" xml:lang="en">
  <link href="https://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-channel-13" rel="prev"/>
  <link href="https://dx.doi.org/10.17487/rfc8831" rel="alternate"/>
  <link href="urn:issn:2070-1721" rel="alternate"/>
  <front>
    <title>WebRTC Data Channels</title>
    <seriesInfo name="RFC" value="8831" stream="IETF"/>
    <author initials='R.' surname='Jesup' fullname='Randell Jesup'>
<organization>Mozilla</organization> initials="R." surname="Jesup" fullname="Randell Jesup">
      <organization showOnFrontPage="true">Mozilla</organization>
      <address>
        <postal>
<street></street>
<code></code>
<city></city>
<country>US</country>
          <street/>
          <code/>
          <city/>
          <country>United States of America</country>
        </postal>
        <email>randell-ietf@jesup.org</email>
      </address>
    </author>
    <author initials='S.' surname='Loreto' fullname='Salvatore Loreto'>
<organization>Ericsson</organization> initials="S." surname="Loreto" fullname="Salvatore Loreto">
      <organization showOnFrontPage="true">Ericsson</organization>
      <address>
        <postal>
          <street>Hirsalantie 11</street>
          <code>02420</code>
          <city>Jorvas</city>
<country>FI</country>
          <country>Finland</country>
        </postal>
        <email>salvatore.loreto@ericsson.com</email>
      </address>
    </author>
    <author initials='M.' surname='Tuexen' fullname='Michael Tuexen'> initials="M." surname="Tüxen" fullname="Michael Tüxen">
      <organization abbrev='Muenster abbrev="Münster Univ. of Appl. Sciences'>
              Muenster Sciences" showOnFrontPage="true">Münster University of Applied Sciences</organization>
      <address>
        <postal>
          <street>Stegerwaldstrasse 39</street>
          <code>48565</code>
          <city> Steinfurt</city>
<country>DE</country>
          <country>Germany</country>
        </postal>
        <email>tuexen@fh-muenster.de</email>
      </address>
    </author>
    <date />
<area>RAI</area>

<abstract>
<t>The month="01" year="2021"/>
    <abstract pn="section-abstract">
      <t indent="0" pn="section-abstract-1">The WebRTC framework specifies protocol support for direct interactive direct, interactive,
rich communication using audio, video, and data between two peers' web-browsers. web browsers.
This document specifies the non-media data transport aspects of the WebRTC
framework. It provides an architectural overview of how the Stream Control
Transmission Protocol (SCTP) is used in the WebRTC context as a generic
transport service allowing WEB-browsers that allows web browsers to exchange generic data from peer to
peer.</t>
    </abstract>
    <boilerplate>
      <section anchor="status-of-memo" numbered="false" removeInRFC="false" toc="exclude" pn="section-boilerplate.1">
        <name slugifiedName="name-status-of-this-memo">Status of This Memo</name>
        <t indent="0" pn="section-boilerplate.1-1">
            This is an Internet Standards Track document.
        </t>
        <t indent="0" pn="section-boilerplate.1-2">
            This document is a product of the Internet Engineering Task Force
            (IETF).  It represents the consensus of the IETF community.  It has
            received public review and has been approved for publication by
            the Internet Engineering Steering Group (IESG).  Further
            information on Internet Standards is available in Section 2 of
            RFC 7841.
        </t>
        <t indent="0" pn="section-boilerplate.1-3">
            Information about the current status of this document, any
            errata, and how to provide feedback on it may be obtained at
            <eref target="https://www.rfc-editor.org/info/rfc8831" brackets="none"/>.
        </t>
      </section>
      <section anchor="copyright" numbered="false" removeInRFC="false" toc="exclude" pn="section-boilerplate.2">
        <name slugifiedName="name-copyright-notice">Copyright Notice</name>
        <t indent="0" pn="section-boilerplate.2-1">
            Copyright (c) 2021 IETF Trust and the persons identified as the
            document authors. All rights reserved.
        </t>
        <t indent="0" pn="section-boilerplate.2-2">
            This document is subject to BCP 78 and the IETF Trust's Legal
            Provisions Relating to IETF Documents
            (<eref target="https://trustee.ietf.org/license-info" brackets="none"/>) in effect on the date of
            publication of this document. Please review these documents
            carefully, as they describe your rights and restrictions with
            respect to this document. Code Components extracted from this
            document must include Simplified BSD License text as described in
            Section 4.e of the Trust Legal Provisions and are provided without
            warranty as described in the Simplified BSD License.
        </t>
      </section>
    </boilerplate>
    <toc>
      <section anchor="toc" numbered="false" removeInRFC="false" toc="exclude" pn="section-toc.1">
        <name slugifiedName="name-table-of-contents">Table of Contents</name>
        <ul bare="true" empty="true" indent="2" spacing="compact" pn="section-toc.1-1">
          <li pn="section-toc.1-1.1">
            <t indent="0" keepWithNext="true" pn="section-toc.1-1.1.1"><xref derivedContent="1" format="counter" sectionFormat="of" target="section-1"/>.  <xref derivedContent="" format="title" sectionFormat="of" target="name-introduction">Introduction</xref></t>
          </li>
          <li pn="section-toc.1-1.2">
            <t indent="0" keepWithNext="true" pn="section-toc.1-1.2.1"><xref derivedContent="2" format="counter" sectionFormat="of" target="section-2"/>.  <xref derivedContent="" format="title" sectionFormat="of" target="name-conventions">Conventions</xref></t>
          </li>
          <li pn="section-toc.1-1.3">
            <t indent="0" pn="section-toc.1-1.3.1"><xref derivedContent="3" format="counter" sectionFormat="of" target="section-3"/>.  <xref derivedContent="" format="title" sectionFormat="of" target="name-use-cases">Use Cases</xref></t>
            <ul bare="true" empty="true" indent="2" spacing="compact" pn="section-toc.1-1.3.2">
              <li pn="section-toc.1-1.3.2.1">
                <t indent="0" keepWithNext="true" pn="section-toc.1-1.3.2.1.1"><xref derivedContent="3.1" format="counter" sectionFormat="of" target="section-3.1"/>.  <xref derivedContent="" format="title" sectionFormat="of" target="name-use-cases-for-unreliable-da">Use Cases for Unreliable Data Channels</xref></t>
              </li>
              <li pn="section-toc.1-1.3.2.2">
                <t indent="0" pn="section-toc.1-1.3.2.2.1"><xref derivedContent="3.2" format="counter" sectionFormat="of" target="section-3.2"/>.  <xref derivedContent="" format="title" sectionFormat="of" target="name-use-cases-for-reliable-data">Use Cases for Reliable Data Channels</xref></t>
              </li>
            </ul>
          </li>
          <li pn="section-toc.1-1.4">
            <t indent="0" pn="section-toc.1-1.4.1"><xref derivedContent="4" format="counter" sectionFormat="of" target="section-4"/>.  <xref derivedContent="" format="title" sectionFormat="of" target="name-requirements">Requirements</xref></t>
          </li>
          <li pn="section-toc.1-1.5">
            <t indent="0" pn="section-toc.1-1.5.1"><xref derivedContent="5" format="counter" sectionFormat="of" target="section-5"/>.  <xref derivedContent="" format="title" sectionFormat="of" target="name-sctp-over-dtls-over-udp-con">SCTP over DTLS over UDP Considerations</xref></t>
          </li>
          <li pn="section-toc.1-1.6">
            <t indent="0" pn="section-toc.1-1.6.1"><xref derivedContent="6" format="counter" sectionFormat="of" target="section-6"/>.  <xref derivedContent="" format="title" sectionFormat="of" target="name-the-usage-of-sctp-for-data-">The Usage of SCTP for Data Channels</xref></t>
            <ul bare="true" empty="true" indent="2" spacing="compact" pn="section-toc.1-1.6.2">
              <li pn="section-toc.1-1.6.2.1">
                <t indent="0" pn="section-toc.1-1.6.2.1.1"><xref derivedContent="6.1" format="counter" sectionFormat="of" target="section-6.1"/>.  <xref derivedContent="" format="title" sectionFormat="of" target="name-sctp-protocol-consideration">SCTP Protocol Considerations</xref></t>
              </li>
              <li pn="section-toc.1-1.6.2.2">
                <t indent="0" pn="section-toc.1-1.6.2.2.1"><xref derivedContent="6.2" format="counter" sectionFormat="of" target="section-6.2"/>.  <xref derivedContent="" format="title" sectionFormat="of" target="name-sctp-association-management">SCTP Association Management</xref></t>
              </li>
              <li pn="section-toc.1-1.6.2.3">
                <t indent="0" pn="section-toc.1-1.6.2.3.1"><xref derivedContent="6.3" format="counter" sectionFormat="of" target="section-6.3"/>.  <xref derivedContent="" format="title" sectionFormat="of" target="name-sctp-streams">SCTP Streams</xref></t>
              </li>
              <li pn="section-toc.1-1.6.2.4">
                <t indent="0" pn="section-toc.1-1.6.2.4.1"><xref derivedContent="6.4" format="counter" sectionFormat="of" target="section-6.4"/>.  <xref derivedContent="" format="title" sectionFormat="of" target="name-data-channel-definition">Data Channel Definition</xref></t>
              </li>
              <li pn="section-toc.1-1.6.2.5">
                <t indent="0" pn="section-toc.1-1.6.2.5.1"><xref derivedContent="6.5" format="counter" sectionFormat="of" target="section-6.5"/>.  <xref derivedContent="" format="title" sectionFormat="of" target="name-opening-a-data-channel">Opening a Data Channel</xref></t>
              </li>
              <li pn="section-toc.1-1.6.2.6">
                <t indent="0" pn="section-toc.1-1.6.2.6.1"><xref derivedContent="6.6" format="counter" sectionFormat="of" target="section-6.6"/>.  <xref derivedContent="" format="title" sectionFormat="of" target="name-transferring-user-data-on-a">Transferring User Data on a Data Channel</xref></t>
              </li>
              <li pn="section-toc.1-1.6.2.7">
                <t indent="0" pn="section-toc.1-1.6.2.7.1"><xref derivedContent="6.7" format="counter" sectionFormat="of" target="section-6.7"/>.  <xref derivedContent="" format="title" sectionFormat="of" target="name-closing-a-data-channel">Closing a Data Channel</xref></t>
              </li>
            </ul>
          </li>
          <li pn="section-toc.1-1.7">
            <t indent="0" pn="section-toc.1-1.7.1"><xref derivedContent="7" format="counter" sectionFormat="of" target="section-7"/>.  <xref derivedContent="" format="title" sectionFormat="of" target="name-security-considerations">Security Considerations</xref></t>
          </li>
          <li pn="section-toc.1-1.8">
            <t indent="0" pn="section-toc.1-1.8.1"><xref derivedContent="8" format="counter" sectionFormat="of" target="section-8"/>.  <xref derivedContent="" format="title" sectionFormat="of" target="name-iana-considerations">IANA Considerations</xref></t>
          </li>
          <li pn="section-toc.1-1.9">
            <t indent="0" pn="section-toc.1-1.9.1"><xref derivedContent="9" format="counter" sectionFormat="of" target="section-9"/>.  <xref derivedContent="" format="title" sectionFormat="of" target="name-references">References</xref></t>
            <ul bare="true" empty="true" indent="2" spacing="compact" pn="section-toc.1-1.9.2">
              <li pn="section-toc.1-1.9.2.1">
                <t indent="0" pn="section-toc.1-1.9.2.1.1"><xref derivedContent="9.1" format="counter" sectionFormat="of" target="section-9.1"/>.  <xref derivedContent="" format="title" sectionFormat="of" target="name-normative-references">Normative References</xref></t>
              </li>
              <li pn="section-toc.1-1.9.2.2">
                <t indent="0" pn="section-toc.1-1.9.2.2.1"><xref derivedContent="9.2" format="counter" sectionFormat="of" target="section-9.2"/>.  <xref derivedContent="" format="title" sectionFormat="of" target="name-informative-references">Informative References</xref></t>
              </li>
            </ul>
          </li>
          <li pn="section-toc.1-1.10">
            <t indent="0" pn="section-toc.1-1.10.1"><xref derivedContent="" format="none" sectionFormat="of" target="section-appendix.a"/><xref derivedContent="" format="title" sectionFormat="of" target="name-acknowledgements">Acknowledgements</xref></t>
          </li>
          <li pn="section-toc.1-1.11">
            <t indent="0" pn="section-toc.1-1.11.1"><xref derivedContent="" format="none" sectionFormat="of" target="section-appendix.b"/><xref derivedContent="" format="title" sectionFormat="of" target="name-authors-addresses">Authors' Addresses</xref></t>
          </li>
        </ul>
      </section>
    </toc>
  </front>
  <middle>
    <section title='Introduction'>
<t>In numbered="true" toc="include" removeInRFC="false" pn="section-1">
      <name slugifiedName="name-introduction">Introduction</name>
      <t indent="0" pn="section-1-1">In the WebRTC framework, communication between the parties consists of media
(for example example, audio and video) and non-media data.
Media is sent using SRTP, the Secure Real-time Transport Protocol (SRTP)
and is not specified further here.
Non-media data is handled by using SCTP the Stream Control Transmission Protocol (SCTP) <xref target='RFC4960'/> target="RFC4960" format="default" sectionFormat="of" derivedContent="RFC4960"/> encapsulated
in DTLS. DTLS 1.0 is defined in <xref target='RFC4347'/> and target="RFC4347" format="default" sectionFormat="of" derivedContent="RFC4347"/>; the present
latest version, DTLS 1.2, is defined in <xref target='RFC6347'/>.</t> target="RFC6347" format="default" sectionFormat="of" derivedContent="RFC6347"/>; and
an upcoming version, DTLS 1.3, is defined in <xref target="I-D.ietf-tls-dtls13" format="default" sectionFormat="of" derivedContent="TLS-DTLS13"/>.</t>
      <figure title='Basic stack diagram'
        anchor='fig-stack'> anchor="fig-stack" align="left" suppress-title="false" pn="figure-1">
        <name slugifiedName="name-basic-stack-diagram">Basic Stack Diagram</name>
        <artwork align='center'> align="center" name="" type="" alt="" pn="section-1-2.1">
+----------+
|   SCTP   |
+----------+
|   DTLS   |
+----------+
| ICE/UDP  |
+----------+
</artwork>
+----------+</artwork>
      </figure>
<t>The
      <t indent="0" pn="section-1-3">The encapsulation of SCTP over DTLS
(see <xref target='I-D.ietf-tsvwg-sctp-dtls-encaps'/>) target="RFC8261" format="default" sectionFormat="of" derivedContent="RFC8261"/>) over ICE/UDP
(see <xref target='RFC5245'/>) target="RFC8445" format="default" sectionFormat="of" derivedContent="RFC8445"/>) provides a NAT traversal
solution together with confidentiality, source authentication, and
integrity protected
integrity-protected transfers.
This data transport service operates in parallel to the SRTP media transports,
and all of them can eventually share a single UDP port number.</t>
<t>SCTP
      <t indent="0" pn="section-1-4">SCTP, as specified in <xref target='RFC4960'/> target="RFC4960" format="default" sectionFormat="of" derivedContent="RFC4960"/> with the partial reliability
extension (PR-SCTP) defined in <xref target='RFC3758'/> target="RFC3758" format="default" sectionFormat="of" derivedContent="RFC3758"/> and the additional policies
defined in <xref target='I-D.ietf-tsvwg-sctp-prpolicies'/> target="RFC7496" format="default" sectionFormat="of" derivedContent="RFC7496"/>,
provides multiple streams natively with reliable, and the relevant
partially-reliable
partially reliable, delivery modes for user messages.
Using the reconfiguration extension defined in <xref target='RFC6525'/> target="RFC6525" format="default" sectionFormat="of" derivedContent="RFC6525"/>
allows to an increase in the number of streams during the lifetime of an SCTP
association and to reset allows individual SCTP streams. streams to be reset.
Using <xref target='I-D.ietf-tsvwg-sctp-ndata'/> target="RFC8260" format="default" sectionFormat="of" derivedContent="RFC8260"/> allows to the interleave of large messages to
avoid the monopolization and adds the support of for
prioritizing of SCTP streams.</t>
<t>The
      <t indent="0" pn="section-1-5">The remainder of this document is organized as follows:
Sections <xref target='sec-use-cases'/> target="sec-use-cases" format="counter" sectionFormat="of" derivedContent="3"/> and <xref target='sec-req'/> target="sec-req" format="counter" sectionFormat="of" derivedContent="4"/> provide use cases
and requirements for both unreliable and reliable peer to peer peer-to-peer data channels;
<xref target='sec-p-a-2'/> target="sec-p-a-2" format="default" sectionFormat="of" derivedContent="Section 5"/> discusses SCTP over DTLS over UDP; and
<xref target='sec-sctp-usage'/> provides the specification of target="sec-sctp-usage" format="default" sectionFormat="of" derivedContent="Section 6"/> specifies how SCTP should be
used by the WebRTC protocol framework for transporting non-media data
between WEB-browsers.</t> web browsers.</t>
    </section>
    <section title='Conventions'>
<t>The numbered="true" toc="include" removeInRFC="false" pn="section-2">
      <name slugifiedName="name-conventions">Conventions</name>
      <t indent="0" pn="section-2-1">The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", "<bcp14>MUST</bcp14>", "<bcp14>MUST NOT</bcp14>",
       "<bcp14>REQUIRED</bcp14>", "<bcp14>SHALL</bcp14>",
       "<bcp14>SHALL NOT</bcp14>", "<bcp14>SHOULD</bcp14>",
       "<bcp14>SHOULD NOT</bcp14>",
       "<bcp14>RECOMMENDED</bcp14>", "<bcp14>NOT RECOMMENDED</bcp14>",
       "<bcp14>MAY</bcp14>", and "OPTIONAL" "<bcp14>OPTIONAL</bcp14>" in this document
       are to be interpreted as described in BCP 14
       <xref target='RFC2119'/>.</t> target="RFC2119" format="default" sectionFormat="of" derivedContent="RFC2119"/> <xref target="RFC8174" format="default" sectionFormat="of" derivedContent="RFC8174"/> when, and only
       when, they appear in all capitals, as shown here.</t>
    </section>
    <section title='Use Cases'
         anchor='sec-use-cases'>
<t>This anchor="sec-use-cases" numbered="true" toc="include" removeInRFC="false" pn="section-3">
      <name slugifiedName="name-use-cases">Use Cases</name>
      <t indent="0" pn="section-3-1">This section defines use cases specific to data channels.
Please note that this section is informational only.</t>
      <section title='Use anchor="sec-use-cases-unreliable" numbered="true" toc="include" removeInRFC="false" pn="section-3.1">
        <name slugifiedName="name-use-cases-for-unreliable-da">Use Cases for Unreliable Data Channels'
         anchor='sec-use-cases-unreliable'>
<t><list style='format U-C %d:' counter='UseCases'>
<t>A Channels</name>
        <ol group="UseCases" spacing="normal" type="U-C %d:" indent="8" start="1" pn="section-3.1-1">
          <li pn="section-3.1-1.1" derivedCounter="U-C 1:">A real-time game where position and object state information is are
sent via one or more unreliable data channels.
Note that at any time time, there may not be no any SRTP media channels, channels or all SRTP media
channels may be inactive, and that there may also be reliable data channels
in use.</t>
<t>Providing use.</li>
          <li pn="section-3.1-1.2" derivedCounter="U-C 2:">Providing non-critical information to a user about the reason for a state
update in a video chat or conference, such as mute state.</t>
</list></t> state.</li>
        </ol>
      </section>
      <section title='Use anchor="sec-use-cases-reliable" numbered="true" toc="include" removeInRFC="false" pn="section-3.2">
        <name slugifiedName="name-use-cases-for-reliable-data">Use Cases for Reliable Data Channels'
         anchor='sec-use-cases-reliable'>
<t><list style='format U-C %d:' counter='UseCases'>
<t> Channels</name>
        <ol group="UseCases" spacing="normal" type="U-C %d:" indent="8" start="3" pn="section-3.2-1">
          <li pn="section-3.2-1.1" derivedCounter="U-C 3:"> A real-time game where critical state information needs to be
transferred, such as control information.
Such a game may have no SRTP media channels, or they may be inactive at any
given time, time or may only be added due to in-game actions.</t>
<t>Non-realtime actions.</li>
          <li pn="section-3.2-1.2" derivedCounter="U-C 4:">Non-real-time file transfers between people chatting.
Note that this may involve a large number of files to transfer sequentially
or in parallel, such as when sharing a folder of images or a directory of files.</t>
<t>Realtime files.</li>
          <li pn="section-3.2-1.3" derivedCounter="U-C 5:">Real-time text chat during an audio and/or video call with an individual
or with multiple people in a conference.</t>
<t>Renegotiation conference.</li>
          <li pn="section-3.2-1.4" derivedCounter="U-C 6:">Renegotiation of the configuration of the PeerConnection.</t>
<t>Proxy PeerConnection.</li>
          <li pn="section-3.2-1.5" derivedCounter="U-C 7:">Proxy browsing, where a browser uses data channels of a PeerConnection
to send and receive HTTP/HTTPS requests and data, for example example, to avoid local
Internet filtering or monitoring.</t>
</list></t> monitoring.</li>
        </ol>
      </section>
    </section>
    <section title='Requirements'
         anchor='sec-req'>
<t>This anchor="sec-req" numbered="true" toc="include" removeInRFC="false" pn="section-4">
      <name slugifiedName="name-requirements">Requirements</name>
      <t indent="0" pn="section-4-1">This section lists the requirements for P2P Peer-to-Peer (P2P) data channels between
two browsers.
Please note that this section is informational only.</t>
<t><list style='format Req. %d:'>
<t>Multiple
      <ol spacing="normal" type="Req. %d:" indent="10" start="1" pn="section-4-2">
        <li pn="section-4-2.1" derivedCounter="Req. 1:">Multiple simultaneous data channels must be supported.
Note that there may be 0 zero or more SRTP media streams in parallel with the data
channels in the same PeerConnection, and the number and state (active/inactive)
of these SRTP media streams may change at any time.</t>
<t>Both time.</li>
        <li pn="section-4-2.2" derivedCounter="Req. 2:">Both reliable and unreliable data channels must be supported.</t>
<t>Data supported.</li>
        <li pn="section-4-2.3" derivedCounter="Req. 3:">Data channels of a PeerConnection must be congestion controlled; controlled
either individually, as a class, or in conjunction with the SRTP media streams
of the PeerConnection, to ensure PeerConnection.  This ensures that data channels don't cause congestion
problems for these SRTP media streams, and that the WebRTC PeerConnection does
not cause excessive problems when run in parallel with TCP connections.</t>
<t>The connections.</li>
        <li pn="section-4-2.4" derivedCounter="Req. 4:">The application should be able to provide guidance as to the
relative priority of each data channel relative to each other, other
and relative to the SRTP media streams.
This will interact with the congestion control algorithms.</t>
<t>Data algorithms.</li>
        <li pn="section-4-2.5" derivedCounter="Req. 5:">Data channels must be secured; allowing secured, which allows for confidentiality,
integrity
integrity, and source authentication.
See <xref target='I-D.ietf-rtcweb-security'/> target="RFC8826" format="default" sectionFormat="of" derivedContent="RFC8826"/> and
<xref target='I-D.ietf-rtcweb-security-arch'/> target="RFC8827" format="default" sectionFormat="of" derivedContent="RFC8827"/> for detailed info.</t>
<!--<t>Consent and NAT traversal mechanism: These are handled by the
PeerConnection's ICE <xref target='RFC5245'/> connectivity checks and
optional TURN servers.</t>-->
<t>Data information.</li>
        <li pn="section-4-2.6" derivedCounter="Req. 6:">Data channels must provide message fragmentation support such that
IP-layer fragmentation can be avoided no matter how large a message the
JavaScript application passes to be sent. It also must ensure that large
data channel transfers don't unduly delay traffic on other data
channels.</t>
<t>The
channels.</li>
        <li pn="section-4-2.7" derivedCounter="Req. 7:">The data channel transport protocol must not encode local IP addresses
inside its protocol fields; doing so reveals potentially private information, information
and leads to failure if the address is depended upon.</t>
<t>The upon.</li>
        <li pn="section-4-2.8" derivedCounter="Req. 8:">The data channel transport protocol should support unbounded-length "messages"
(i.e., a virtual socket stream) at the application layer, layer for such things as
image-file-transfer; Implementations implementations might enforce a reasonable message size
limit.</t>
<t>The
limit.</li>
        <li pn="section-4-2.9" derivedCounter="Req. 9:">The data channel transport protocol should avoid IP fragmentation. It
must support PMTU (Path MTU) Path MTU (PMTU) discovery and must not rely on ICMP or ICMPv6
being generated or being passed back, especially for PMTU discovery.</t>
<t>It discovery.</li>
        <li pn="section-4-2.10" derivedCounter="Req. 10:">It must be possible to implement the protocol stack in the user application space.</t>
</list></t> space.</li>
      </ol>
    </section>
    <section title='SCTP anchor="sec-p-a-2" numbered="true" toc="include" removeInRFC="false" pn="section-5">
      <name slugifiedName="name-sctp-over-dtls-over-udp-con">SCTP over DTLS over UDP Considerations'
         anchor='sec-p-a-2'>
<t>The Considerations</name>
      <t indent="0" pn="section-5-1">The important features of SCTP in the WebRTC context are:
<list style='symbols'>
<t>Usage are the following:
</t>
      <ul spacing="normal" bare="false" empty="false" indent="3" pn="section-5-2">
        <li pn="section-5-2.1">Usage of a TCP-friendly congestion control.</t>
<t>The control.</li>
        <li pn="section-5-2.2">modifiable congestion control is modifiable for integration with the
   SRTP media stream congestion control.</t>
<t>Support control.</li>
        <li pn="section-5-2.3">Support of multiple unidirectional streams, each providing its own
   notion of ordered message delivery.</t>
<t>Support delivery.</li>
        <li pn="section-5-2.4">Support of ordered and out-of-order message delivery.</t>
<t>Supporting arbitrary delivery.</li>
        <li pn="section-5-2.5">Support of arbitrarily large user messages by providing fragmentation
   and reassembly.</t>
<t>Support reassembly.</li>
        <li pn="section-5-2.6">Support of PMTU-discovery.</t>
<t>Support PMTU discovery.</li>
        <li pn="section-5-2.7">Support of reliable or partially reliable message transport.</t>
</list></t>
<t>The transport.</li>
      </ul>
      <t indent="0" pn="section-5-3">The WebRTC Data Channel data channel mechanism does not support SCTP multihoming.
The SCTP layer will simply act as if it were running on a single-homed host,
since that is the abstraction that the DTLS layer (a connection oriented, connection-oriented,
unreliable datagram service) exposes.</t>
<t>The
      <t indent="0" pn="section-5-4">The encapsulation of SCTP over DTLS defined in
<xref target='I-D.ietf-tsvwg-sctp-dtls-encaps'/> target="RFC8261" format="default" sectionFormat="of" derivedContent="RFC8261"/> provides confidentiality,
source authenticated, authentication, and integrity protected integrity-protected transfers.
Using DTLS over UDP in combination with ICE Interactive Connectivity Establishment
(ICE) <xref target="RFC8445" format="default" sectionFormat="of" derivedContent="RFC8445"/> enables middlebox traversal
in IPv4 IPv4- and IPv6 based IPv6-based networks.
SCTP as specified in <xref target='RFC4960'/> MUST target="RFC4960" format="default" sectionFormat="of" derivedContent="RFC4960"/> <bcp14>MUST</bcp14> be used in
combination with the extension defined in <xref target='RFC3758'/> target="RFC3758" format="default" sectionFormat="of" derivedContent="RFC3758"/> and
provides the following features for transporting non-media data between
browsers:
<list style='symbols'>
<t>Support
</t>
      <ul spacing="normal" bare="false" empty="false" indent="3" pn="section-5-5">
        <li pn="section-5-5.1">Support of multiple unidirectional streams.</t>
<t>Ordered streams.</li>
        <li pn="section-5-5.2">Ordered and unordered delivery of user messages.</t>
<t>Reliable messages.</li>
        <li pn="section-5-5.3">Reliable and partial-reliable partially reliable transport of user messages.</t>
</list></t>
<t>Each messages.</li>
      </ul>
      <t indent="0" pn="section-5-6">Each SCTP user message contains a Payload Protocol Identifier (PPID)
that is passed to SCTP by its upper layer on the sending side and
provided to its upper layer on the receiving side.
The PPID can be used to multiplex/demultiplex multiple upper layers over
a single SCTP association.
In the WebRTC context, the PPID is used to distinguish between
UTF-8 encoded user data,
binary encoded userdata
binary-encoded user data, and
the Data Channel Establishment Protocol (DCEP) defined in
<xref target='I-D.ietf-rtcweb-data-protocol'/>. target="RFC8832" format="default" sectionFormat="of" derivedContent="RFC8832"/>.
Please note that the PPID is not accessible via the Javascript JavaScript API.</t>
<!--
<t>Moreover SCTP provides the possibility to transport different "protocols"
over multiple streams and associations using the PPID
(Payload Protocol Identifier).
An application can set a different PPID with each send call.
This allows the receiving application to look at this information
(as well as the stream id/seq) on receiving to know what type of protocol
the data payload has.</t>
-->
<t>The
      <t indent="0" pn="section-5-7">The encapsulation of SCTP over DTLS, together with the SCTP features listed
above
above, satisfies all the requirements listed in <xref target='sec-req'/>.</t>
<!--
<t>There are SCTP implementations for most Operating Systems in wide use:</t>
<t>
<list style='empty'>
<t>Linux (mainline kernel 2.6.36)</t>
<t>FreeBSD (release kernel 8.2)</t>
<t>Mac OS X</t>
<t>Windows (SctpDrv4)</t>
<t>Solaris (OpenSolaris 2009.06)</t>
<t>and a user-land SCTP implementation (based on the FreeBSD implementation).</t>
</list>
</t>

<section title='User Space versus Kernel Implementation'
         anchor='sec-sctp-1'>
<t>Even though kernel implementations of SCTP are already available for most
platforms (see <xref target='sec-p-a-2'/> ), there are compelling reasons for
using an SCTP stack that works well in user land.</t>
<t>The main reason is deployability.</t>
<t>Web browsers supporting WebRTC are expected to run on a wide range of old
and new operating systems. They support operating systems 10 years old or more,
they run on mobile and desktop operating systems,
and they are highly portable to new operating systems.
This is achieved by having a fairly narrow portability layer to minimize
what needs to be supported on old operating systems and ported to new ones.
This creates a need to implement as much functionality as possible
inside the application instead of relying on the operating system.</t>
<t>As a user-land implementation of SCTP is available, this meets
requirement 12.</t>
</section>
-->
<t>The target="sec-req" format="default" sectionFormat="of" derivedContent="Section 4"/>.</t>
      <t indent="0" pn="section-5-8">The layering of protocols for WebRTC is shown in the following <xref target='fig-sctp-layering'/>.</t> target="fig-sctp-layering" format="default" sectionFormat="of" derivedContent="Figure 2"/>.</t>
      <figure title='WebRTC protocol layers'
        anchor='fig-sctp-layering'> anchor="fig-sctp-layering" align="left" suppress-title="false" pn="figure-2">
        <name slugifiedName="name-webrtc-protocol-layers">WebRTC Protocol Layers</name>
        <artwork align='center'> align="center" name="" type="" alt="" pn="section-5-9.1">
              +------+------+------+
              | DCEP | UTF-8|Binary|
              |      | data Data | data Data |
              +------+------+------+
              |        SCTP        |
+----------------------------------+
| STUN | SRTP |        DTLS        |
+----------------------------------+
|                ICE               |
+----------------------------------+
| UDP1 | UDP2 | UDP3 | ...         |
+----------------------------------+
</artwork>
+----------------------------------+</artwork>
      </figure>
<t>This
      <t indent="0" pn="section-5-10">This stack (especially in contrast to DTLS over SCTP <xref target='RFC6083'/> target="RFC6083" format="default" sectionFormat="of" derivedContent="RFC6083"/> and
in combination with SCTP over UDP <xref target='RFC6951'/>) target="RFC6951" format="default" sectionFormat="of" derivedContent="RFC6951"/>)
has been chosen because it
<list style='symbols'>
<t>supports for the following reasons:
</t>
      <ul spacing="normal" bare="false" empty="false" indent="3" pn="section-5-11">
        <li pn="section-5-11.1">supports the transmission of arbitrary arbitrarily large user messages.</t>
<t>shares messages;</li>
        <li pn="section-5-11.2">shares the DTLS connection with the SRTP media channels of the PeerConnection.</t>
<t>provides
PeerConnection; and</li>
        <li pn="section-5-11.3">provides privacy for the SCTP control information.</t>
</list></t>
<t>Considering information.</li>
      </ul>
      <t indent="0" pn="section-5-12">Referring to the protocol stack of shown in <xref target='fig-sctp-layering'/> the target="fig-sctp-layering" format="default" sectionFormat="of" derivedContent="Figure 2"/>:</t>
      <ul spacing="normal" bare="false" empty="false" indent="3" pn="section-5-13">
        <li pn="section-5-13.1">the usage of DTLS 1.0 over UDP is specified in <xref target='RFC4347'/> and
the target="RFC4347" format="default" sectionFormat="of" derivedContent="RFC4347"/>;</li>
        <li pn="section-5-13.2">the usage of DTLS 1.2 over UDP in specified in <xref target='RFC6347'/>,
while the target="RFC6347" format="default" sectionFormat="of" derivedContent="RFC6347"/>;</li>
        <li pn="section-5-13.3">the usage of DTLS 1.3 over UDP is specified in an upcoming document
<xref target="I-D.ietf-tls-dtls13" format="default" sectionFormat="of" derivedContent="TLS-DTLS13"/>; and</li>
        <li pn="section-5-13.4">the usage of SCTP on top of DTLS is specified in
<xref target='I-D.ietf-tsvwg-sctp-dtls-encaps'/>.
Please target="RFC8261" format="default" sectionFormat="of" derivedContent="RFC8261"/>.</li>
      </ul>
      <t indent="0" pn="section-5-14">Please note that the demultiplexing STUN Session Traversal Utilities for NAT (STUN)
<xref target="RFC5389" format="default" sectionFormat="of" derivedContent="RFC5389"/> vs. SRTP vs. DTLS is done
as described in Section 5.1.2 of <xref target='RFC5764'/> target="RFC5764" sectionFormat="of" section="5.1.2" format="default" derivedLink="https://rfc-editor.org/rfc/rfc5764#section-5.1.2" derivedContent="RFC5764"/>, and SCTP
is the only payload of DTLS.</t>
<t>Since
      <t indent="0" pn="section-5-15">Since DTLS is typically implemented in user application space, the SCTP
stack also needs to be a user application space stack.</t>
<t>The
      <t indent="0" pn="section-5-16">The ICE/UDP layer can handle IP address changes during a session without
needing interaction with the DTLS and SCTP layers.
However, SCTP SHOULD <bcp14>SHOULD</bcp14> be notified when an address changes change has happened.
In this case case, SCTP SHOULD <bcp14>SHOULD</bcp14> retest the Path MTU and reset the congestion
state to the initial state.
In the case of a window based window-based congestion control like the one specified in
<xref target='RFC4960'/>, target="RFC4960" format="default" sectionFormat="of" derivedContent="RFC4960"/>, this means setting the congestion window and
slow start
slow-start threshold to its initial values.</t>
<t>Incoming
      <t indent="0" pn="section-5-17">Incoming ICMP or ICMPv6 messages can't be processed by
the SCTP layer, since there is no way to identify the corresponding
association. Therefore Therefore, SCTP MUST <bcp14>MUST</bcp14> support performing Path MTU discovery
without relying on ICMP or ICMPv6 as specified in <xref target='RFC4821'/> target="RFC4821" format="default" sectionFormat="of" derivedContent="RFC4821"/>
by using probing messages specified in <xref target='RFC4820'/>. target="RFC4820" format="default" sectionFormat="of" derivedContent="RFC4820"/>.
The initial Path MTU at the IP layer SHOULD NOT <bcp14>SHOULD NOT</bcp14> exceed 1200 bytes for IPv4
and 1280 bytes for IPv6.</t>
<t>In
      <t indent="0" pn="section-5-18">In general, the lower layer lower-layer interface of an SCTP implementation should be
adapted to address the differences between IPv4 and IPv6 (being connection-less) connectionless)
or DTLS (being connection-oriented).</t>
<t>When connection oriented).</t>
      <t indent="0" pn="section-5-19">When the protocol stack of shown in <xref target='fig-sctp-layering'/> target="fig-sctp-layering" format="default" sectionFormat="of" derivedContent="Figure 2"/> is used, DTLS
protects the complete SCTP packet, so it provides confidentiality, integrity integrity, and
source authentication of the complete SCTP packet.</t>
<t>SCTP
      <t indent="0" pn="section-5-20">SCTP provides congestion control on a per-association base. basis. This means
that all SCTP streams within a single SCTP association share the same
congestion window. Traffic not being sent over SCTP is not covered by
the
SCTP congestion control.
Using a congestion control different from than the standard one might improve
the impact on the parallel SRTP media streams.</t>
<t>SCTP
      <t indent="0" pn="section-5-21">SCTP uses the same port number concept as TCP and UDP do.
Therefore UDP.
Therefore, an SCTP association uses two port numbers, one at each SCTP
end-point.</t>
endpoint.</t>
    </section>
    <section title='The anchor="sec-sctp-usage" numbered="true" toc="include" removeInRFC="false" pn="section-6">
      <name slugifiedName="name-the-usage-of-sctp-for-data-">The Usage of SCTP for Data Channels'
         anchor='sec-sctp-usage'> Channels</name>
      <section title='SCTP numbered="true" toc="include" removeInRFC="false" pn="section-6.1">
        <name slugifiedName="name-sctp-protocol-consideration">SCTP Protocol Considerations'>
<t>The Considerations</name>
        <t indent="0" pn="section-6.1-1">The DTLS encapsulation of SCTP packets as described in
<xref target='I-D.ietf-tsvwg-sctp-dtls-encaps'/> MUST target="RFC8261" format="default" sectionFormat="of" derivedContent="RFC8261"/> <bcp14>MUST</bcp14> be used.</t>
<t>This
        <t indent="0" pn="section-6.1-2">This SCTP stack and its upper layer MUST <bcp14>MUST</bcp14> support the usage of multiple
SCTP streams.
A user message can be sent ordered or unordered and with partial or full
reliability.</t>
<t>The
        <t indent="0" pn="section-6.1-3">The following SCTP protocol extensions are required:
<list style='symbols'>
<t>The
</t>
        <ul spacing="normal" bare="false" empty="false" indent="3" pn="section-6.1-4">
          <li pn="section-6.1-4.1">The stream reconfiguration extension defined in <xref target='RFC6525'/>
   MUST target="RFC6525" format="default" sectionFormat="of" derivedContent="RFC6525"/>
            <bcp14>MUST</bcp14> be supported. It is used for closing channels.</t>
<t>The channels.</li>
          <li pn="section-6.1-4.2">The dynamic address reconfiguration extension defined in
   <xref target='RFC5061'/> MUST target="RFC5061" format="default" sectionFormat="of" derivedContent="RFC5061"/> <bcp14>MUST</bcp14> be used to signal the support of the
   stream reset extension defined in <xref target='RFC6525'/>. target="RFC6525" format="default" sectionFormat="of" derivedContent="RFC6525"/>.
   Other features of <xref target='RFC5061'/> target="RFC5061" format="default" sectionFormat="of" derivedContent="RFC5061"/> are OPTIONAL.</t>
<t>The <bcp14>OPTIONAL</bcp14>.</li>
          <li pn="section-6.1-4.3">The partial reliability extension defined in <xref target='RFC3758'/> MUST target="RFC3758" format="default" sectionFormat="of" derivedContent="RFC3758"/> <bcp14>MUST</bcp14>
   be supported. In addition to the timed reliability PR-SCTP policy defined
   in <xref target='RFC3758'/>, target="RFC3758" format="default" sectionFormat="of" derivedContent="RFC3758"/>, the limited retransmission policy defined in
   <xref target='I-D.ietf-tsvwg-sctp-prpolicies'/> MUST target="RFC7496" format="default" sectionFormat="of" derivedContent="RFC7496"/> <bcp14>MUST</bcp14> be supported.
   Limiting the number of retransmissions to zero zero, combined with unordered
   delivery
   delivery, provides a UDP-like service where each user message is sent
   exactly once and delivered in the order received.</t>
</list></t>
<t>The received.</li>
        </ul>
        <t indent="0" pn="section-6.1-5">The support for message interleaving as defined in
<xref target='I-D.ietf-tsvwg-sctp-ndata'/> SHOULD target="RFC8260" format="default" sectionFormat="of" derivedContent="RFC8260"/> <bcp14>SHOULD</bcp14> be used.</t>
      </section>
      <section title='SCTP anchor="sec-sctp-management" numbered="true" toc="include" removeInRFC="false" pn="section-6.2">
        <name slugifiedName="name-sctp-association-management">SCTP Association Management'
         anchor='sec-sctp-management'>
<t>In Management</name>
        <t indent="0" pn="section-6.2-1">In the WebRTC context, the SCTP association will be set up when the
two endpoints of the WebRTC PeerConnection agree on opening it, as negotiated
by JSEP (typically the JavaScript Session Establishment Protocol (JSEP), which is typically an
exchange of SDP) the Session Description Protocol (SDP) <xref target='I-D.ietf-rtcweb-jsep'/>. target="RFC8829" format="default" sectionFormat="of" derivedContent="RFC8829"/>.
It will use the DTLS connection selected via ICE; ICE, and typically this will be
shared via BUNDLE or equivalent with DTLS connections used to key the
SRTP media streams.</t>
<!-- FIXME: Bundle Issue. -->
<t>The
        <t indent="0" pn="section-6.2-2">The number of streams negotiated during SCTP association setup SHOULD <bcp14>SHOULD</bcp14>
be 65535, which is the maximum number of streams that can be negotiated during
the association setup.</t>

<t>SCTP
        <t indent="0" pn="section-6.2-3">SCTP supports two ways of terminating an SCTP association.
A
The first method is a graceful one, using where a procedure which ensures that ensures no messages
are lost during the shutdown of the association. association is used.
The second method is a non-graceful one, where one side can just abort the
association.</t>
<t>Each
        <t indent="0" pn="section-6.2-4">Each SCTP end-point supervises endpoint continuously supervises the reachability of its peer by
monitoring the number of retransmissions of user messages and test messages.
In case of excessive retransmissions, the association is terminated in a
non-graceful way.</t>
<t>If
        <t indent="0" pn="section-6.2-5">If an SCTP association is closed in a graceful way, all of its data channels
are closed.
In case of a non-graceful teardown, all data channels are also closed,
but an error indication SHOULD <bcp14>SHOULD</bcp14> be provided if possible.</t>
      </section>
      <section title='SCTP Streams'>
<t>SCTP numbered="true" toc="include" removeInRFC="false" pn="section-6.3">
        <name slugifiedName="name-sctp-streams">SCTP Streams</name>
        <t indent="0" pn="section-6.3-1">SCTP defines a stream as a unidirectional logical channel existing within
an SCTP association to another SCTP endpoint. The streams are used to
provide the notion of in-sequence delivery and for multiplexing.
Each user message is sent on a particular stream, either ordered or unordered.
Ordering is preserved only for ordered messages sent on the same stream.</t>
      </section>
      <section title='Data numbered="true" toc="include" removeInRFC="false" pn="section-6.4">
        <name slugifiedName="name-data-channel-definition">Data Channel Definition'>
<t>Data Definition</name>
        <t indent="0" pn="section-6.4-1">Data channels are defined such that their accompanying application-level API
can closely mirror the API for WebSockets, which implies bidirectional streams
of data and a textual field called 'label' used to identify the meaning of the
data channel.</t>
<t>The
        <t indent="0" pn="section-6.4-2">The realization of a data channel is a pair of one incoming stream and
one outgoing SCTP stream having the same SCTP stream identifier.
How these SCTP stream identifiers are selected is protocol and implementation
dependent. This allows a bidirectional communication.</t>
<t>Additionally,
        <t indent="0" pn="section-6.4-3">Additionally, each data channel has the following properties in each
direction:
<list style='symbols'>
<t>reliable
</t>
        <ul spacing="normal" bare="false" empty="false" indent="3" pn="section-6.4-4">
          <li pn="section-6.4-4.1">reliable or unreliable message transmission. transmission:
In case of unreliable transmissions, the same level of unreliability is used.
Please note that
Note that, in SCTP SCTP, this is a property of an SCTP user message and not
of an SCTP stream.</t>
<t>in-order stream.</li>
          <li pn="section-6.4-4.2">in-order or out-of-order message delivery for message sent.
Please note that sent:
Note that, in SCTP SCTP, this is a property of an SCTP user message and not
of an SCTP stream.</t>
<t>A stream.</li>
          <li pn="section-6.4-4.3">a priority, which is a 2 byte 2-byte unsigned integer. integer:
These priorities MUST <bcp14>MUST</bcp14> be interpreted as weighted-fair-queuing scheduling
priorities per the definition of the corresponding stream scheduler
supporting interleaving in <xref target='I-D.ietf-tsvwg-sctp-ndata'/>. target="RFC8260" format="default" sectionFormat="of" derivedContent="RFC8260"/>.
For use in WebRTC, the values used SHOULD <bcp14>SHOULD</bcp14> be one of 128 ("below normal"),
256 ("normal"), 512 ("high") ("high"), or 1024 ("extra high").</t>
<t>an high").</li>
          <li pn="section-6.4-4.4">an optional label.</t>
<t>an label.</li>
          <li pn="section-6.4-4.5">an optional protocol.</t>
</list></t>
<t>Please note protocol.</li>
        </ul>
        <t indent="0" pn="section-6.4-5">Note that for a data channel being negotiated with the protocol
specified in <xref target='I-D.ietf-rtcweb-data-protocol'/> target="RFC8832" format="default" sectionFormat="of" derivedContent="RFC8832"/>, all of the above
properties are the same in both directions.</t>
      </section>
      <section title='Opening numbered="true" toc="include" removeInRFC="false" pn="section-6.5">
        <name slugifiedName="name-opening-a-data-channel">Opening a Data Channel'>
<t>Data Channel</name>
        <t indent="0" pn="section-6.5-1">Data channels can be opened by using negotiation within the SCTP association,
called association
(called in-band negotiation, negotiation) or out-of-band negotiation.
Out-of-band negotiation is defined as any method which that results in an agreement
as to the parameters of a channel and the creation thereof.
The details are out of scope of this document. Applications using data
channels need to use the negotiation methods consistently on both end-points.</t>
<t>A endpoints.</t>
        <t indent="0" pn="section-6.5-2">A simple protocol for in-band negotiation is specified in
<xref target='I-D.ietf-rtcweb-data-protocol'/>.</t>
<t>When target="RFC8832" format="default" sectionFormat="of" derivedContent="RFC8832"/>.</t>
        <t indent="0" pn="section-6.5-3">When one side wants to open a channel using out-of-band negotiation, it
picks a stream.
Unless otherwise defined or negotiated, the streams are picked based on
the DTLS role (the client picks even stream identifiers, and
the server picks odd stream identifiers).
However, the application is responsible for avoiding collisions with
existing streams.
If it attempts to re-use reuse a stream which that is part of an existing data channel,
the addition MUST <bcp14>MUST</bcp14> fail.
In addition to choosing a stream, the application SHOULD <bcp14>SHOULD</bcp14> also determine
the options to use be used for sending messages.
The application MUST <bcp14>MUST</bcp14> ensure in an application-specific manner that
the application at the peer will also know the selected stream to
be used, and as well as the options for sending data from that side.</t>
      </section>
      <section title='Transferring numbered="true" toc="include" removeInRFC="false" pn="section-6.6">
        <name slugifiedName="name-transferring-user-data-on-a">Transferring User Data on a Data Channel'>
<t>All Channel</name>
        <t indent="0" pn="section-6.6-1">All data sent on a data channel in both directions MUST <bcp14>MUST</bcp14> be sent over the
underlying stream using the reliability defined when the data channel was
opened
opened, unless the options are changed, changed or per-message options are specified
by a higher level.</t>
<t>The message-orientation
        <t indent="0" pn="section-6.6-2">The message orientation of SCTP is used to preserve the message boundaries
of user messages. Therefore, senders MUST NOT <bcp14>MUST NOT</bcp14> put more than one application
message into an SCTP user message. Unless the deprecated PPID-based
fragmentation and reassembly is used, the sender MUST <bcp14>MUST</bcp14> include exactly one
application message in each SCTP user message.</t>
<t>The
        <t indent="0" pn="section-6.6-3">The SCTP Payload Protocol Identifiers (PPIDs) are used to signal the
interpretation of the "Payload "payload data". The following PPIDs MUST <bcp14>MUST</bcp14> be used
(see <xref target='sec-IANA'/>):
<list style="hanging">
<t hangText='WebRTC String:'> target="sec-IANA" format="default" sectionFormat="of" derivedContent="Section 8"/>):
</t>
        <dl newline="false" spacing="normal" indent="3" pn="section-6.6-4">
          <dt pn="section-6.6-4.1">WebRTC String:</dt>
          <dd pn="section-6.6-4.2">
to identify a non-empty JavaScript string encoded in UTF-8.</t>
<t hangText='WebRTC UTF-8.</dd>
          <dt pn="section-6.6-4.3">WebRTC String Empty:'> Empty:</dt>
          <dd pn="section-6.6-4.4">
to identify an empty JavaScript string encoded in UTF-8.</t>
<t hangText='WebRTC Binary:'> UTF-8.</dd>
          <dt pn="section-6.6-4.5">WebRTC Binary:</dt>
          <dd pn="section-6.6-4.6">
to identify a non-empty JavaScript binary data
(ArrayBuffer, ArrayBufferView ArrayBufferView, or Blob).</t>
<t hangText='WebRTC Blob).</dd>
          <dt pn="section-6.6-4.7">WebRTC Binary Empty:'> Empty:</dt>
          <dd pn="section-6.6-4.8">
to identify an empty JavaScript binary data
(ArrayBuffer, ArrayBufferView ArrayBufferView, or Blob).</t>
</list></t>
<t>SCTP Blob).</dd>
        </dl>
        <t indent="0" pn="section-6.6-5">SCTP does not support the sending of empty user messages. Therefore, if an
empty message has to be sent, the appropriate PPID (WebRTC String Empty or
WebRTC Binary Empty) is used used, and the SCTP user message of one zero byte is
sent. When receiving an SCTP user message with one of these PPIDs, the receiver
MUST
<bcp14>MUST</bcp14> ignore the SCTP user message and process it as an empty message.</t>
<t>The
        <t indent="0" pn="section-6.6-6">The usage of the PPIDs "WebRTC String Partial" and "WebRTC Binary Partial"
is deprecated. They were used for a PPID-based fragmentation and reassembly
of user messages belonging to reliable and ordered data channels.</t>
<t>If
        <t indent="0" pn="section-6.6-7">If a message with an unsupported PPID is received or some error condition
related to the received message is detected by the receiver
(for example, illegal ordering), the receiver SHOULD <bcp14>SHOULD</bcp14> close the corresponding
data channel. This implies in particular that extensions using additional
PPIDs can't be used without prior negotiation.</t>
<t>The
        <t indent="0" pn="section-6.6-8">The SCTP base protocol specified in <xref target='RFC4960'/> target="RFC4960" format="default" sectionFormat="of" derivedContent="RFC4960"/> does not
support the interleaving of user messages. Therefore Therefore, sending a large user
message can monopolize the SCTP association.
To overcome this limitation, <xref target='I-D.ietf-tsvwg-sctp-ndata'/> target="RFC8260" format="default" sectionFormat="of" derivedContent="RFC8260"/>
defines an extension to support message interleaving, which SHOULD <bcp14>SHOULD</bcp14> be used.
As long as message interleaving is not supported, the sender
SHOULD
<bcp14>SHOULD</bcp14> limit the maximum message size to 16 KB to avoid monopolization.</t>
<t>It
        <t indent="0" pn="section-6.6-9">It is recommended that the message size be kept within certain size bounds bounds,
as applications will not be able to support arbitrarily-large arbitrarily large single
messages. This limit has to be negotiated, for example example, by using
<xref target='I-D.ietf-mmusic-sctp-sdp'/>.</t>
<t>The target="RFC8841" format="default" sectionFormat="of" derivedContent="RFC8841"/>.</t>
        <t indent="0" pn="section-6.6-10">The sender SHOULD <bcp14>SHOULD</bcp14> disable the Nagle algorithm (see <xref target='RFC1122'/>) target="RFC1122" format="default" sectionFormat="of" derivedContent="RFC1122"/>)
to minimize the latency.</t>
      </section>
      <section title='Closing numbered="true" toc="include" removeInRFC="false" pn="section-6.7">
        <name slugifiedName="name-closing-a-data-channel">Closing a Data Channel'>
<t>Closing Channel</name>
        <t indent="0" pn="section-6.7-1">Closing of a data channel MUST <bcp14>MUST</bcp14> be signaled by resetting the corresponding
outgoing streams <xref target='RFC6525'/>. target="RFC6525" format="default" sectionFormat="of" derivedContent="RFC6525"/>. This means that if one side
decides to close the data channel, it resets the corresponding outgoing stream.
When the peer sees that an incoming stream was reset, it also resets its
corresponding outgoing stream. Once this is completed, the data channel is closed.
Resetting a stream sets the Stream Sequence Numbers (SSNs) of the stream back to
'zero' with a corresponding notification to the application layer
that the reset has been performed. Streams are available for reuse after a reset
has been performed.</t>
<t><xref target='RFC6525'/>
        <t indent="0" pn="section-6.7-2"><xref target="RFC6525" format="default" sectionFormat="of" derivedContent="RFC6525"/> also guarantees that all the messages are delivered
(or abandoned) before the stream is reset.</t>
      </section>
    </section>
    <section title='Security Considerations'
         anchor='sec-security'>
<t>This anchor="sec-security" numbered="true" toc="include" removeInRFC="false" pn="section-7">
      <name slugifiedName="name-security-considerations">Security Considerations</name>
      <t indent="0" pn="section-7-1">This document does not add any additional considerations to the ones given in
<xref target='I-D.ietf-rtcweb-security'/> target="RFC8826" format="default" sectionFormat="of" derivedContent="RFC8826"/> and
<xref target='I-D.ietf-rtcweb-security-arch'/>.</t>
<t>It target="RFC8827" format="default" sectionFormat="of" derivedContent="RFC8827"/>.</t>
      <t indent="0" pn="section-7-2">It should be noted that a receiver must be prepared that the for a sender that tries
to send arbitrary arbitrarily large messages.</t>
    </section>
    <section title='IANA Considerations'
         anchor='sec-IANA'>
<t>[NOTE to RFC-Editor:
<list>
<t>"RFCXXXX" is to be replaced by the RFC number you assign this document.</t>
</list>
]</t>
<t>This anchor="sec-IANA" numbered="true" toc="include" removeInRFC="false" pn="section-8">
      <name slugifiedName="name-iana-considerations">IANA Considerations</name>
      <t indent="0" pn="section-8-1">This document uses six already registered SCTP Payload Protocol
Identifiers (PPIDs):
"DOMString Last",
"Binary Data Partial",
"Binary Data Last",
"DOMString Partial",
"WebRTC String Empty", and
"WebRTC Binary Empty".
<xref target='RFC4960'/> target="RFC4960" format="default" sectionFormat="of" derivedContent="RFC4960"/> creates the registry "SCTP Payload Protocol Identifiers" registry
from which these identifiers were assigned.
IANA is requested to update has updated the reference of these six assignments to point
to this document and change changed the names of the first four PPIDs.
The corresponding dates should be kept.</t>
<t>Therefore these remain unchanged.</t>
      <t indent="0" pn="section-8-2">The six assignments should be have been updated to read:</t>
<texttable>
<ttcol align='left'>Value</ttcol>
<ttcol align='left'>SCTP PPID</ttcol>
<ttcol align='left'>Reference</ttcol>
<ttcol align='left'>Date</ttcol>
<c>WebRTC String</c>                      <c>51</c> <c>[RFCXXXX]</c> <c>2013-09-20</c>
<c>WebRTC
      <table align="center" pn="table-1">
        <thead>
          <tr>
            <th align="left" colspan="1" rowspan="1">Value</th>
            <th align="left" colspan="1" rowspan="1">SCTP PPID</th>
            <th align="left" colspan="1" rowspan="1">Reference</th>
            <th align="left" colspan="1" rowspan="1">Date</th>
          </tr>
        </thead>
        <tbody>
          <tr>
            <td align="left" colspan="1" rowspan="1">WebRTC String</td>
            <td align="left" colspan="1" rowspan="1">51</td>
            <td align="left" colspan="1" rowspan="1">RFC 8831</td>
            <td align="left" colspan="1" rowspan="1">2013-09-20</td>
          </tr>
          <tr>
            <td align="left" colspan="1" rowspan="1">WebRTC Binary Partial (Deprecated)</c> <c>52</c> <c>[RFCXXXX]</c> <c>2013-09-20</c>
<c>WebRTC Binary</c>                      <c>53</c> <c>[RFCXXXX]</c> <c>2013-09-20</c>
<c>WebRTC (deprecated)</td>
            <td align="left" colspan="1" rowspan="1">52</td>
            <td align="left" colspan="1" rowspan="1">RFC 8831</td>
            <td align="left" colspan="1" rowspan="1">2013-09-20</td>
          </tr>
          <tr>
            <td align="left" colspan="1" rowspan="1">WebRTC Binary</td>
            <td align="left" colspan="1" rowspan="1">53</td>
            <td align="left" colspan="1" rowspan="1">RFC 8831</td>
            <td align="left" colspan="1" rowspan="1">2013-09-20</td>
          </tr>
          <tr>
            <td align="left" colspan="1" rowspan="1">WebRTC String Partial (Deprecated)</c> <c>54</c> <c>[RFCXXXX]</c> <c>2013-09-20</c>
<c>WebRTC (deprecated)</td>
            <td align="left" colspan="1" rowspan="1">54</td>
            <td align="left" colspan="1" rowspan="1">RFC 8831</td>
            <td align="left" colspan="1" rowspan="1">2013-09-20</td>
          </tr>
          <tr>
            <td align="left" colspan="1" rowspan="1">WebRTC String Empty</c>                <c>56</c> <c>[RFCXXXX]</c> <c>2014-08-22</c>
<c>WebRTC Empty</td>
            <td align="left" colspan="1" rowspan="1">56</td>
            <td align="left" colspan="1" rowspan="1">RFC 8831</td>
            <td align="left" colspan="1" rowspan="1">2014-08-22</td>
          </tr>
          <tr>
            <td align="left" colspan="1" rowspan="1">WebRTC Binary Empty</c>                <c>57</c> <c>[RFCXXXX]</c> <c>2014-08-22</c>
</texttable>
</section>

<section title='Acknowledgments'>
<t>Many thanks for comments, ideas, and text from
Harald Alvestrand,
Richard Barnes,
Adam Bergkvist,
Alissa Cooper,
Benoit Claise,
Spencer Dawkins,
Gunnar Hellstrom,
Christer Holmberg,
Cullen Jennings,
Paul Kyzivat,
Eric Rescorla,
Adam Roach,
Irene Ruengeler,
Randall Stewart,
Martin Stiemerling,
Justin Uberti,
and Magnus Westerlund.</t> Empty</td>
            <td align="left" colspan="1" rowspan="1">57</td>
            <td align="left" colspan="1" rowspan="1">RFC 8831</td>
            <td align="left" colspan="1" rowspan="1">2014-08-22</td>
          </tr>
        </tbody>
      </table>
    </section>
  </middle>
  <back>
    <displayreference target="I-D.ietf-tls-dtls13" to="TLS-DTLS13"/>
    <references title='Normative References'>
<?rfc include='reference.RFC.2119' ?>
<?rfc include='reference.RFC.3758'?>
<?rfc include='reference.RFC.4347'?>
<?rfc include='reference.RFC.4820'?>
<?rfc include='reference.RFC.4821'?>
<?rfc include='reference.RFC.4960'?>
<?rfc include='reference.RFC.5061'?>
<?rfc include='reference.RFC.5245'?>
<?rfc include='reference.RFC.6347'?>
<?rfc include='reference.RFC.6525'?>
<?rfc include='reference.I-D.ietf-tsvwg-sctp-ndata'?>
<?rfc include='reference.I-D.ietf-rtcweb-data-protocol'?>
<?rfc include='reference.I-D.ietf-tsvwg-sctp-dtls-encaps'?>
<?rfc include='reference.I-D.ietf-rtcweb-security'?>
<?rfc include='reference.I-D.ietf-rtcweb-security-arch'?>
<?rfc include='reference.I-D.ietf-rtcweb-jsep'?>
<?rfc include='reference.I-D.ietf-tsvwg-sctp-prpolicies'?>
<?rfc include='reference.I-D.ietf-mmusic-sctp-sdp'?>
</references> pn="section-9">
      <name slugifiedName="name-references">References</name>
      <references title='Informative References'>
<?rfc include='reference.RFC.1122'?>
<?rfc include='reference.RFC.5764'?>
<?rfc include='reference.RFC.6083'?>
<?rfc include='reference.RFC.6951'?>
</references> pn="section-9.1">
        <name slugifiedName="name-normative-references">Normative References</name>
        <reference anchor="RFC2119" target="https://www.rfc-editor.org/info/rfc2119" quoteTitle="true" derivedAnchor="RFC2119">
          <front>
            <title>Key words for use in RFCs to Indicate Requirement Levels</title>
            <author initials="S." surname="Bradner" fullname="S. Bradner">
              <organization showOnFrontPage="true"/>
            </author>
            <date year="1997" month="March"/>
            <abstract>
              <t indent="0">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="RFC3758" target="https://www.rfc-editor.org/info/rfc3758" quoteTitle="true" derivedAnchor="RFC3758">
          <front>
            <title>Stream Control Transmission Protocol (SCTP) Partial Reliability Extension</title>
            <author initials="R." surname="Stewart" fullname="R. Stewart">
              <organization showOnFrontPage="true"/>
            </author>
            <author initials="M." surname="Ramalho" fullname="M. Ramalho">
              <organization showOnFrontPage="true"/>
            </author>
            <author initials="Q." surname="Xie" fullname="Q. Xie">
              <organization showOnFrontPage="true"/>
            </author>
            <author initials="M." surname="Tuexen" fullname="M. Tuexen">
              <organization showOnFrontPage="true"/>
            </author>
            <author initials="P." surname="Conrad" fullname="P. Conrad">
              <organization showOnFrontPage="true"/>
            </author>
            <date year="2004" month="May"/>
            <abstract>
              <t indent="0">This memo describes an extension to the Stream Control Transmission Protocol (SCTP) that allows an SCTP endpoint to signal to its peer that it should move the cumulative ack point forward.  When both sides of an SCTP association support this extension, it can be used by an SCTP implementation to provide partially reliable data transmission service to an upper layer protocol.  This memo describes the protocol extensions, which consist of a new parameter for INIT and INIT ACK, and a new FORWARD TSN chunk type, and provides one example of a partially reliable service that can be provided to the upper layer via this mechanism.  [STANDARDS-TRACK]</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="3758"/>
          <seriesInfo name="DOI" value="10.17487/RFC3758"/>
        </reference>
        <reference anchor="RFC4820" target="https://www.rfc-editor.org/info/rfc4820" quoteTitle="true" derivedAnchor="RFC4820">
          <front>
            <title>Padding Chunk and Parameter for the Stream Control Transmission Protocol (SCTP)</title>
            <author initials="M." surname="Tuexen" fullname="M. Tuexen">
              <organization showOnFrontPage="true"/>
            </author>
            <author initials="R." surname="Stewart" fullname="R. Stewart">
              <organization showOnFrontPage="true"/>
            </author>
            <author initials="P." surname="Lei" fullname="P. Lei">
              <organization showOnFrontPage="true"/>
            </author>
            <date year="2007" month="March"/>
            <abstract>
              <t indent="0">This document defines a padding chunk and a padding parameter and describes the required receiver side procedures.  The padding chunk is used to pad a Stream Control Transmission Protocol (SCTP) packet to an arbitrary size.  The padding parameter is used to pad an SCTP INIT chunk to an arbitrary size.  [STANDARDS-TRACK]</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="4820"/>
          <seriesInfo name="DOI" value="10.17487/RFC4820"/>
        </reference>
        <reference anchor="RFC4821" target="https://www.rfc-editor.org/info/rfc4821" quoteTitle="true" derivedAnchor="RFC4821">
          <front>
            <title>Packetization Layer Path MTU Discovery</title>
            <author initials="M." surname="Mathis" fullname="M. Mathis">
              <organization showOnFrontPage="true"/>
            </author>
            <author initials="J." surname="Heffner" fullname="J. Heffner">
              <organization showOnFrontPage="true"/>
            </author>
            <date year="2007" month="March"/>
            <abstract>
              <t indent="0">This document describes a robust method for Path MTU Discovery (PMTUD) that relies on TCP or some other Packetization Layer to probe an Internet path with progressively larger packets.  This method is described as an extension to RFC 1191 and RFC 1981, which specify ICMP-based Path MTU Discovery for IP versions 4 and 6, respectively.  [STANDARDS-TRACK]</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="4821"/>
          <seriesInfo name="DOI" value="10.17487/RFC4821"/>
        </reference>
        <reference anchor="RFC4960" target="https://www.rfc-editor.org/info/rfc4960" quoteTitle="true" derivedAnchor="RFC4960">
          <front>
            <title>Stream Control Transmission Protocol</title>
            <author initials="R." surname="Stewart" fullname="R. Stewart" role="editor">
              <organization showOnFrontPage="true"/>
            </author>
            <date year="2007" month="September"/>
            <abstract>
              <t indent="0">This document obsoletes RFC 2960 and RFC 3309.  It describes the Stream Control Transmission Protocol (SCTP).  SCTP is designed to transport Public Switched Telephone Network (PSTN) signaling messages over IP networks, but is capable of broader applications.</t>
              <t indent="0">SCTP is a reliable transport protocol operating on top of a connectionless packet network such as IP.  It offers the following services to its users:</t>
              <t indent="0">--  acknowledged error-free non-duplicated transfer of user data,</t>
              <t indent="0">--  data fragmentation to conform to discovered path MTU size,</t>
              <t indent="0">--  sequenced delivery of user messages within multiple streams, with an option for order-of-arrival delivery of individual user messages,</t>
              <t indent="0">--  optional bundling of multiple user messages into a single SCTP packet, and</t>
              <t indent="0">--  network-level fault tolerance through supporting of multi-homing at either or both ends of an association.</t>
              <t indent="0"> The design of SCTP includes appropriate congestion avoidance behavior and resistance to flooding and masquerade attacks.  [STANDARDS-TRACK]</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="4960"/>
          <seriesInfo name="DOI" value="10.17487/RFC4960"/>
        </reference>
        <reference anchor="RFC5061" target="https://www.rfc-editor.org/info/rfc5061" quoteTitle="true" derivedAnchor="RFC5061">
          <front>
            <title>Stream Control Transmission Protocol (SCTP) Dynamic Address Reconfiguration</title>
            <author initials="R." surname="Stewart" fullname="R. Stewart">
              <organization showOnFrontPage="true"/>
            </author>
            <author initials="Q." surname="Xie" fullname="Q. Xie">
              <organization showOnFrontPage="true"/>
            </author>
            <author initials="M." surname="Tuexen" fullname="M. Tuexen">
              <organization showOnFrontPage="true"/>
            </author>
            <author initials="S." surname="Maruyama" fullname="S. Maruyama">
              <organization showOnFrontPage="true"/>
            </author>
            <author initials="M." surname="Kozuka" fullname="M. Kozuka">
              <organization showOnFrontPage="true"/>
            </author>
            <date year="2007" month="September"/>
            <abstract>
              <t indent="0">A local host may have multiple points of attachment to the Internet, giving it a degree of fault tolerance from hardware failures.  Stream Control Transmission Protocol (SCTP) (RFC 4960) was developed to take full advantage of such a multi-homed host to provide a fast failover and association survivability in the face of such hardware failures. This document describes an extension to SCTP that will allow an SCTP stack to dynamically add an IP address to an SCTP association, dynamically delete an IP address from an SCTP association, and to request to set the primary address the peer will use when sending to an endpoint.  [STANDARDS-TRACK]</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="5061"/>
          <seriesInfo name="DOI" value="10.17487/RFC5061"/>
        </reference>
        <reference anchor="RFC6525" target="https://www.rfc-editor.org/info/rfc6525" quoteTitle="true" derivedAnchor="RFC6525">
          <front>
            <title>Stream Control Transmission Protocol (SCTP) Stream Reconfiguration</title>
            <author initials="R." surname="Stewart" fullname="R. Stewart">
              <organization showOnFrontPage="true"/>
            </author>
            <author initials="M." surname="Tuexen" fullname="M. Tuexen">
              <organization showOnFrontPage="true"/>
            </author>
            <author initials="P." surname="Lei" fullname="P. Lei">
              <organization showOnFrontPage="true"/>
            </author>
            <date year="2012" month="February"/>
            <abstract>
              <t indent="0">Many applications that use the Stream Control Transmission Protocol (SCTP) want the ability to "reset" a stream.  The intention of resetting a stream is to set the numbering sequence of the stream back to 'zero' with a corresponding notification to the application layer that the reset has been performed.  Applications requiring this feature want it so that they can "reuse" streams for different purposes but still utilize the stream sequence number so that the application can track the message flows.  Thus, without this feature, a new use of an old stream would result in message numbers greater than expected, unless there is a protocol mechanism to "reset the streams back to zero".  This document also includes methods for resetting the transmission sequence numbers, adding additional streams, and resetting all stream sequence numbers.  [STANDARDS-TRACK]</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="6525"/>
          <seriesInfo name="DOI" value="10.17487/RFC6525"/>
        </reference>
        <reference anchor="RFC7496" target="https://www.rfc-editor.org/info/rfc7496" quoteTitle="true" derivedAnchor="RFC7496">
          <front>
            <title>Additional Policies for the Partially Reliable Stream Control Transmission Protocol Extension</title>
            <author initials="M." surname="Tuexen" fullname="M. Tuexen">
              <organization showOnFrontPage="true"/>
            </author>
            <author initials="R." surname="Seggelmann" fullname="R. Seggelmann">
              <organization showOnFrontPage="true"/>
            </author>
            <author initials="R." surname="Stewart" fullname="R. Stewart">
              <organization showOnFrontPage="true"/>
            </author>
            <author initials="S." surname="Loreto" fullname="S. Loreto">
              <organization showOnFrontPage="true"/>
            </author>
            <date year="2015" month="April"/>
            <abstract>
              <t indent="0">This document defines two additional policies for the Partially Reliable Stream Control Transmission Protocol (PR-SCTP) extension. These policies allow limitation of the number of retransmissions and prioritization of user messages for more efficient usage of the send buffer.</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="7496"/>
          <seriesInfo name="DOI" value="10.17487/RFC7496"/>
        </reference>
        <reference anchor="RFC8174" target="https://www.rfc-editor.org/info/rfc8174" quoteTitle="true" derivedAnchor="RFC8174">
          <front>
            <title>Ambiguity of Uppercase vs Lowercase in RFC 2119 Key Words</title>
            <author initials="B." surname="Leiba" fullname="B. Leiba">
              <organization showOnFrontPage="true"/>
            </author>
            <date year="2017" month="May"/>
            <abstract>
              <t indent="0">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="RFC8260" target="https://www.rfc-editor.org/info/rfc8260" quoteTitle="true" derivedAnchor="RFC8260">
          <front>
            <title>Stream Schedulers and User Message Interleaving for the Stream Control Transmission Protocol</title>
            <author initials="R." surname="Stewart" fullname="R. Stewart">
              <organization showOnFrontPage="true"/>
            </author>
            <author initials="M." surname="Tuexen" fullname="M. Tuexen">
              <organization showOnFrontPage="true"/>
            </author>
            <author initials="S." surname="Loreto" fullname="S. Loreto">
              <organization showOnFrontPage="true"/>
            </author>
            <author initials="R." surname="Seggelmann" fullname="R. Seggelmann">
              <organization showOnFrontPage="true"/>
            </author>
            <date year="2017" month="November"/>
            <abstract>
              <t indent="0">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 indent="0">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="RFC8261" target="https://www.rfc-editor.org/info/rfc8261" quoteTitle="true" derivedAnchor="RFC8261">
          <front>
            <title>Datagram Transport Layer Security (DTLS) Encapsulation of SCTP Packets</title>
            <author initials="M." surname="Tuexen" fullname="M. Tuexen">
              <organization showOnFrontPage="true"/>
            </author>
            <author initials="R." surname="Stewart" fullname="R. Stewart">
              <organization showOnFrontPage="true"/>
            </author>
            <author initials="R." surname="Jesup" fullname="R. Jesup">
              <organization showOnFrontPage="true"/>
            </author>
            <author initials="S." surname="Loreto" fullname="S. Loreto">
              <organization showOnFrontPage="true"/>
            </author>
            <date year="2017" month="November"/>
            <abstract>
              <t indent="0">The Stream Control Transmission Protocol (SCTP) is a transport protocol originally defined to run on top of the network protocols IPv4 or IPv6.  This document specifies how SCTP can be used on top of the Datagram Transport Layer Security (DTLS) protocol.  Using the encapsulation method described in this document, SCTP is unaware of the protocols being used below DTLS; hence, explicit IP addresses cannot be used in the SCTP control chunks.  As a consequence, the SCTP associations carried over DTLS can only be single-homed.</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="8261"/>
          <seriesInfo name="DOI" value="10.17487/RFC8261"/>
        </reference>
        <reference anchor="RFC8445" target="https://www.rfc-editor.org/info/rfc8445" quoteTitle="true" derivedAnchor="RFC8445">
          <front>
            <title>Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal</title>
            <author initials="A." surname="Keranen" fullname="A. Keranen">
              <organization showOnFrontPage="true"/>
            </author>
            <author initials="C." surname="Holmberg" fullname="C. Holmberg">
              <organization showOnFrontPage="true"/>
            </author>
            <author initials="J." surname="Rosenberg" fullname="J. Rosenberg">
              <organization showOnFrontPage="true"/>
            </author>
            <date year="2018" month="July"/>
            <abstract>
              <t indent="0">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 indent="0">This document obsoletes RFC 5245.</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="8445"/>
          <seriesInfo name="DOI" value="10.17487/RFC8445"/>
        </reference>
        <reference anchor="RFC8826" target="https://www.rfc-editor.org/info/rfc8826" quoteTitle="true" derivedAnchor="RFC8826">
          <front>
            <title>Security Considerations for WebRTC</title>
            <author initials="E." surname="Rescorla" fullname="Eric Rescorla">
              <organization showOnFrontPage="true"/>
            </author>
            <date month="January" year="2021"/>
          </front>
          <seriesInfo name="RFC" value="8826"/>
          <seriesInfo name="DOI" value="10.17487/RFC8826"/>
        </reference>
        <reference anchor="RFC8827" target="https://www.rfc-editor.org/info/rfc8827" quoteTitle="true" derivedAnchor="RFC8827">
          <front>
            <title>WebRTC Security Architecture</title>
            <author initials="E." surname="Rescorla" fullname="Eric Rescorla">
              <organization showOnFrontPage="true"/>
            </author>
            <date month="January" year="2021"/>
          </front>
          <seriesInfo name="RFC" value="8827"/>
          <seriesInfo name="DOI" value="10.17487/RFC8827"/>
        </reference>
        <reference anchor="RFC8829" target="https://www.rfc-editor.org/info/rfc8829" quoteTitle="true" derivedAnchor="RFC8829">
          <front>
            <title>JavaScript Session Establishment Protocol (JSEP)</title>
            <author initials="J." surname="Uberti" fullname="Justin Uberti">
              <organization showOnFrontPage="true"/>
            </author>
            <author initials="C." surname="Jennings" fullname="Cullen Jennings">
              <organization showOnFrontPage="true"/>
            </author>
            <author initials="E." surname="Rescorla" fullname="Eric Rescorla" role="editor">
              <organization showOnFrontPage="true"/>
            </author>
            <date month="January" year="2021"/>
          </front>
          <seriesInfo name="RFC" value="8829"/>
          <seriesInfo name="DOI" value="10.17487/RFC8829"/>
        </reference>
        <reference anchor="RFC8832" target="https://www.rfc-editor.org/info/rfc8832" quoteTitle="true" derivedAnchor="RFC8832">
          <front>
            <title>WebRTC Data Channel Establishment Protocol</title>
            <author initials="R." surname="Jesup" fullname="Randell Jesup">
              <organization showOnFrontPage="true"/>
            </author>
            <author initials="S." surname="Loreto" fullname="Salvatore Loreto">
              <organization showOnFrontPage="true"/>
            </author>
            <author initials="M" surname="Tüxen" fullname="Michael Tüxen">
              <organization showOnFrontPage="true"/>
            </author>
            <date month="January" year="2021"/>
          </front>
          <seriesInfo name="RFC" value="8832"/>
          <seriesInfo name="DOI" value="10.17487/RFC8832"/>
        </reference>
        <reference anchor="RFC8841" target="https://www.rfc-editor.org/info/rfc8841" quoteTitle="true" derivedAnchor="RFC8841">
          <front>
            <title>Session Description Protocol (SDP) Offer/Answer Procedures for Stream Control Transmission Protocol (SCTP) over Datagram Transport Layer Security (DTLS) Transport</title>
            <author initials="C." surname="Holmberg" fullname="Christer Holmberg">
              <organization showOnFrontPage="true"/>
            </author>
            <author initials="R." surname="Shpount" fullname="Roman Shpount">
              <organization showOnFrontPage="true"/>
            </author>
            <author initials="S." surname="Loreto" fullname="Salvatore Loreto">
              <organization showOnFrontPage="true"/>
            </author>
            <author initials="G." surname="Camarillo" fullname="Gonzalo Camarillo">
              <organization showOnFrontPage="true"/>
            </author>
            <date month="January" year="2021"/>
          </front>
          <seriesInfo name="RFC" value="8841"/>
          <seriesInfo name="DOI" value="10.17487/RFC8841"/>
        </reference>
      </references>
      <references pn="section-9.2">
        <name slugifiedName="name-informative-references">Informative References</name>
        <reference anchor="RFC1122" target="https://www.rfc-editor.org/info/rfc1122" quoteTitle="true" derivedAnchor="RFC1122">
          <front>
            <title>Requirements for Internet Hosts - Communication Layers</title>
            <author initials="R." surname="Braden" fullname="R. Braden" role="editor">
              <organization showOnFrontPage="true"/>
            </author>
            <date year="1989" month="October"/>
            <abstract>
              <t indent="0">This RFC is an official specification for the Internet community.  It incorporates by reference, amends, corrects, and supplements the primary protocol standards documents relating to hosts.  [STANDARDS-TRACK]</t>
            </abstract>
          </front>
          <seriesInfo name="STD" value="3"/>
          <seriesInfo name="RFC" value="1122"/>
          <seriesInfo name="DOI" value="10.17487/RFC1122"/>
        </reference>
        <reference anchor="RFC4347" target="https://www.rfc-editor.org/info/rfc4347" quoteTitle="true" derivedAnchor="RFC4347">
          <front>
            <title>Datagram Transport Layer Security</title>
            <author initials="E." surname="Rescorla" fullname="E. Rescorla">
              <organization showOnFrontPage="true"/>
            </author>
            <author initials="N." surname="Modadugu" fullname="N. Modadugu">
              <organization showOnFrontPage="true"/>
            </author>
            <date year="2006" month="April"/>
            <abstract>
              <t indent="0">This document specifies Version 1.0 of the Datagram Transport Layer Security (DTLS) protocol.  The DTLS protocol provides communications privacy for datagram protocols.  The protocol allows client/server applications to communicate in a way that is designed to prevent eavesdropping, tampering, or message forgery.  The DTLS protocol is based on the Transport Layer Security (TLS) protocol and provides equivalent security guarantees.  Datagram semantics of the underlying transport are preserved by the DTLS protocol.  [STANDARDS-TRACK]</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="4347"/>
          <seriesInfo name="DOI" value="10.17487/RFC4347"/>
        </reference>
        <reference anchor="RFC5389" target="https://www.rfc-editor.org/info/rfc5389" quoteTitle="true" derivedAnchor="RFC5389">
          <front>
            <title>Session Traversal Utilities for NAT (STUN)</title>
            <author initials="J." surname="Rosenberg" fullname="J. Rosenberg">
              <organization showOnFrontPage="true"/>
            </author>
            <author initials="R." surname="Mahy" fullname="R. Mahy">
              <organization showOnFrontPage="true"/>
            </author>
            <author initials="P." surname="Matthews" fullname="P. Matthews">
              <organization showOnFrontPage="true"/>
            </author>
            <author initials="D." surname="Wing" fullname="D. Wing">
              <organization showOnFrontPage="true"/>
            </author>
            <date year="2008" month="October"/>
            <abstract>
              <t indent="0">Session Traversal Utilities for NAT (STUN) is a protocol that serves as a tool for other protocols in dealing with Network Address Translator (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 indent="0">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.  This is an important change from the previous version of this specification (RFC 3489), which presented STUN as a complete solution.</t>
              <t indent="0">This document obsoletes RFC 3489.  [STANDARDS-TRACK]</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="5389"/>
          <seriesInfo name="DOI" value="10.17487/RFC5389"/>
        </reference>
        <reference anchor="RFC5764" target="https://www.rfc-editor.org/info/rfc5764" quoteTitle="true" derivedAnchor="RFC5764">
          <front>
            <title>Datagram Transport Layer Security (DTLS) Extension to Establish Keys for the Secure Real-time Transport Protocol (SRTP)</title>
            <author initials="D." surname="McGrew" fullname="D. McGrew">
              <organization showOnFrontPage="true"/>
            </author>
            <author initials="E." surname="Rescorla" fullname="E. Rescorla">
              <organization showOnFrontPage="true"/>
            </author>
            <date year="2010" month="May"/>
            <abstract>
              <t indent="0">This document describes a Datagram Transport Layer Security (DTLS) extension to establish keys for Secure RTP (SRTP) and Secure RTP Control Protocol (SRTCP) flows.  DTLS keying happens on the media path, independent of any out-of-band signalling channel present. [STANDARDS-TRACK]</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="5764"/>
          <seriesInfo name="DOI" value="10.17487/RFC5764"/>
        </reference>
        <reference anchor="RFC6083" target="https://www.rfc-editor.org/info/rfc6083" quoteTitle="true" derivedAnchor="RFC6083">
          <front>
            <title>Datagram Transport Layer Security (DTLS) for Stream Control Transmission Protocol (SCTP)</title>
            <author initials="M." surname="Tuexen" fullname="M. Tuexen">
              <organization showOnFrontPage="true"/>
            </author>
            <author initials="R." surname="Seggelmann" fullname="R. Seggelmann">
              <organization showOnFrontPage="true"/>
            </author>
            <author initials="E." surname="Rescorla" fullname="E. Rescorla">
              <organization showOnFrontPage="true"/>
            </author>
            <date year="2011" month="January"/>
            <abstract>
              <t indent="0">This document describes the usage of the Datagram Transport Layer Security (DTLS) protocol over the Stream Control Transmission Protocol (SCTP).</t>
              <t indent="0">DTLS over SCTP provides communications privacy for applications that use SCTP as their transport protocol and allows client/server applications to communicate in a way that is designed to prevent eavesdropping and detect tampering or message forgery.</t>
              <t indent="0">Applications using DTLS over SCTP can use almost all transport features provided by SCTP and its extensions.  [STANDARDS-TRACK]</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="6083"/>
          <seriesInfo name="DOI" value="10.17487/RFC6083"/>
        </reference>
        <reference anchor="RFC6347" target="https://www.rfc-editor.org/info/rfc6347" quoteTitle="true" derivedAnchor="RFC6347">
          <front>
            <title>Datagram Transport Layer Security Version 1.2</title>
            <author initials="E." surname="Rescorla" fullname="E. Rescorla">
              <organization showOnFrontPage="true"/>
            </author>
            <author initials="N." surname="Modadugu" fullname="N. Modadugu">
              <organization showOnFrontPage="true"/>
            </author>
            <date year="2012" month="January"/>
            <abstract>
              <t indent="0">This document specifies version 1.2 of the Datagram Transport Layer Security (DTLS) protocol.  The DTLS protocol provides communications privacy for datagram protocols.  The protocol allows client/server applications to communicate in a way that is designed to prevent eavesdropping, tampering, or message forgery.  The DTLS protocol is based on the Transport Layer Security (TLS) protocol and provides equivalent security guarantees.  Datagram semantics of the underlying transport are preserved by the DTLS protocol.  This document updates DTLS 1.0 to work with TLS version 1.2.  [STANDARDS-TRACK]</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="6347"/>
          <seriesInfo name="DOI" value="10.17487/RFC6347"/>
        </reference>
        <reference anchor="RFC6951" target="https://www.rfc-editor.org/info/rfc6951" quoteTitle="true" derivedAnchor="RFC6951">
          <front>
            <title>UDP Encapsulation of Stream Control Transmission Protocol (SCTP) Packets for End-Host to End-Host Communication</title>
            <author initials="M." surname="Tuexen" fullname="M. Tuexen">
              <organization showOnFrontPage="true"/>
            </author>
            <author initials="R." surname="Stewart" fullname="R. Stewart">
              <organization showOnFrontPage="true"/>
            </author>
            <date year="2013" month="May"/>
            <abstract>
              <t indent="0">This document describes a simple method of encapsulating Stream Control Transmission Protocol (SCTP) packets into UDP packets and its limitations.  This allows the usage of SCTP in networks with legacy NATs that do not support SCTP.  It can also be used to implement SCTP on hosts without directly accessing the IP layer, for example, implementing it as part of the application without requiring special privileges.</t>
              <t indent="0">Please note that this document only describes the functionality required within an SCTP stack to add on UDP encapsulation, providing only those mechanisms for two end-hosts to communicate with each other over UDP ports.  In particular, it does not provide mechanisms to determine whether UDP encapsulation is being used by the peer, nor the mechanisms for determining which remote UDP port number can be used.  These functions are out of scope for this document.</t>
              <t indent="0">This document covers only end-hosts and not tunneling (egress or ingress) endpoints.</t>
            </abstract>
          </front>
          <seriesInfo name="RFC" value="6951"/>
          <seriesInfo name="DOI" value="10.17487/RFC6951"/>
        </reference>
        <reference anchor="I-D.ietf-tls-dtls13" quoteTitle="true" target="https://tools.ietf.org/html/draft-ietf-tls-dtls13-39" derivedAnchor="TLS-DTLS13">
          <front>
            <title>The Datagram Transport Layer Security (DTLS) Protocol Version 1.3</title>
            <author initials="E." surname="Rescorla" fullname="Eric Rescorla">
              <organization showOnFrontPage="true">RTFM, Inc.</organization>
            </author>
            <author initials="H." surname="Tschofenig" fullname="Hannes Tschofenig">
              <organization showOnFrontPage="true">Arm Limited</organization>
            </author>
            <author initials="N." surname="Modadugu" fullname="Nagendra Modadugu">
              <organization showOnFrontPage="true">Google, Inc.</organization>
            </author>
            <date month="November" day="2" year="2020"/>
            <abstract>
              <t indent="0">   This document specifies Version 1.3 of the Datagram Transport Layer
   Security (DTLS) protocol.  DTLS 1.3 allows client/server applications
   to communicate over the Internet in a way that is designed to prevent
   eavesdropping, tampering, and message forgery.

   The DTLS 1.3 protocol is intentionally based on the Transport Layer
   Security (TLS) 1.3 protocol and provides equivalent security
   guarantees with the exception of order protection/non-replayability.
   Datagram semantics of the underlying transport are preserved by the
   DTLS protocol.

              </t>
            </abstract>
          </front>
          <seriesInfo name="Internet-Draft" value="draft-ietf-tls-dtls13-39"/>
          <format type="TXT" target="https://www.ietf.org/internet-drafts/draft-ietf-tls-dtls13-39.txt"/>
          <refcontent>Work in Progress</refcontent>
        </reference>
      </references>
    </references>
    <section numbered="false" toc="include" removeInRFC="false" pn="section-appendix.a">
      <name slugifiedName="name-acknowledgements">Acknowledgements</name>
      <t indent="0" pn="section-appendix.a-1">Many thanks for comments, ideas, and text from <contact fullname="Harald Alvestrand"/>, <contact fullname="Richard Barnes"/>,
      <contact fullname="Adam Bergkvist"/>, <contact fullname="Alissa Cooper"/>,
      <contact fullname="Benoit Claise"/>, <contact fullname="Spencer       Dawkins"/>, <contact fullname="Gunnar Hellström"/>, <contact fullname="Christer Holmberg"/>, <contact fullname="Cullen Jennings"/>,
      <contact fullname="Paul Kyzivat"/>, <contact fullname="Eric Rescorla"/>,
      <contact fullname="Adam Roach"/>, <contact fullname="Irene Rüngeler"/>,
      <contact fullname="Randall Stewart"/>, <contact fullname="Martin       Stiemerling"/>, <contact fullname="Justin Uberti"/>, and <contact fullname="Magnus Westerlund"/>.</t>
    </section>
    <section anchor="authors-addresses" numbered="false" removeInRFC="false" toc="include" pn="section-appendix.b">
      <name slugifiedName="name-authors-addresses">Authors' Addresses</name>
      <author initials="R." surname="Jesup" fullname="Randell Jesup">
        <organization showOnFrontPage="true">Mozilla</organization>
        <address>
          <postal>
            <street/>
            <code/>
            <city/>
            <country>United States of America</country>
          </postal>
          <email>randell-ietf@jesup.org</email>
        </address>
      </author>
      <author initials="S." surname="Loreto" fullname="Salvatore Loreto">
        <organization showOnFrontPage="true">Ericsson</organization>
        <address>
          <postal>
            <street>Hirsalantie 11</street>
            <code>02420</code>
            <city>Jorvas</city>
            <country>Finland</country>
          </postal>
          <email>salvatore.loreto@ericsson.com</email>
        </address>
      </author>
      <author initials="M." surname="Tüxen" fullname="Michael Tüxen">
        <organization abbrev="Münster Univ. of Appl. Sciences" showOnFrontPage="true">Münster University of Applied Sciences</organization>
        <address>
          <postal>
            <street>Stegerwaldstrasse 39</street>
            <code>48565</code>
            <city> Steinfurt</city>
            <country>Germany</country>
          </postal>
          <email>tuexen@fh-muenster.de</email>
        </address>
      </author>
    </section>
  </back>
</rfc>