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<rfc category='std'
     docName='draft-ietf-rtcweb-data-channel-13.txt'> xmlns:xi="http://www.w3.org/2001/XInclude" number="8831"
     docName="draft-ietf-rtcweb-data-channel-13" category="std"
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     updates="" xml:lang="en" symRefs="true" tocInclude="true"
     sortRefs="true" version="3">
  <!-- xml2rfc v2v3 conversion 2.44.0 -->
    <title>WebRTC Data Channels</title>
    <seriesInfo name="RFC" value="8831"/>
    <author initials='R.' surname='Jesup' fullname='Randell Jesup'> initials="R." surname="Jesup" fullname="Randell Jesup">
          <country>United States of America</country>
    <author initials='S.' surname='Loreto' fullname='Salvatore Loreto'> initials="S." surname="Loreto" fullname="Salvatore Loreto">
          <street>Hirsalantie 11</street>
    <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">
              Münster University of Applied Sciences</organization>
          <street>Stegerwaldstrasse 39</street>
          <city> Steinfurt</city>
    <date /> month="September" year="2020"/>

      <t>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
    <section title='Introduction'> numbered="true" toc="default">
      <t>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"/> encapsulated
in DTLS. DTLS 1.0 is defined in <xref target='RFC4347'/> target="RFC4347" format="default"/>, and the present
latest version, DTLS 1.2, is defined in <xref target='RFC6347'/>.</t> target="RFC6347" format="default"/>.</t>
      <figure title='Basic stack diagram'
        anchor='fig-stack'> anchor="fig-stack">
        <name>Basic Stack Diagram</name>
        <artwork align='center'> align="center" name="" type="" alt=""><![CDATA[
|   SCTP   |
|   DTLS   |
| ICE/UDP  |
      <t>The encapsulation of SCTP over DTLS
(see <xref target='I-D.ietf-tsvwg-sctp-dtls-encaps'/>) target="RFC8261" format="default"/>) over ICE/UDP
(see <xref target='RFC5245'/>) target="RFC8445" format="default"/>) 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, as specified in <xref target='RFC4960'/> target="RFC4960" format="default"/> with the partial reliability
extension (PR-SCTP) defined in <xref target='RFC3758'/> target="RFC3758" format="default"/> and the additional policies
defined in <xref target='I-D.ietf-tsvwg-sctp-prpolicies'/> target="RFC7496" format="default"/>,
provides multiple streams natively with reliable, and the relevant
partially reliable, delivery modes for user messages.
Using the reconfiguration extension defined in <xref target='RFC6525'/> target="RFC6525" format="default"/>
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"/> allows to the interleave of large messages to
avoid the monopolization and adds the support of for
prioritizing of SCTP streams.</t>
      <t>The remainder of this document is organized as follows:
Sections <xref target='sec-use-cases'/> target="sec-use-cases" format="counter"/> and <xref target='sec-req'/> target="sec-req" format="counter"/> 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"/> discusses SCTP over DTLS over UDP; and
<xref target='sec-sctp-usage'/> provides the specification of target="sec-sctp-usage" format="default"/> specifies how SCTP should be
used by the WebRTC protocol framework for transporting non-media data
between WEB-browsers.</t> web browsers.</t>
    <section title='Conventions'> numbered="true" toc="default">
       <t>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&nbsp;14
       <xref target="RFC2119"/> <xref target='RFC2119'/>.</t> target="RFC8174"/> when, and only
       when, they appear in all capitals, as shown here.</t>
    <section title='Use Cases'
         anchor='sec-use-cases'> anchor="sec-use-cases" numbered="true" toc="default">
      <name>Use Cases</name>
      <t>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="default">
        <name>Use Cases for Unreliable Data Channels'
<t><list style='format U-C %d:' counter='UseCases'>
<t>A Channels</name>
        <ol group="UseCases" spacing="normal" type="U-C %d:" indent="8">
          <li>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>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>
      <section title='Use anchor="sec-use-cases-reliable" numbered="true" toc="default">
        <name>Use Cases for Reliable Data Channels'
<t><list style='format U-C %d:' counter='UseCases'>
<t> Channels</name>
        <ol group="UseCases" spacing="normal" type="U-C %d:" indent="8">
          <li> 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>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>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>Renegotiation of the configuration of the PeerConnection.</t>
<t>Proxy PeerConnection.</li>
          <li>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>
    <section title='Requirements'
         anchor='sec-req'> anchor="sec-req" numbered="true" toc="default">
      <t>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:'>
      <ol spacing="normal" type="Req. %d:" indent="10">
        <li>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>Both reliable and unreliable data channels must be supported.</t>
<t>Data supported.</li>
        <li>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>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>Data channels must be secured; allowing secured, which allows for confidentiality,
integrity, and source authentication.
See <xref target='I-D.ietf-rtcweb-security'/> target="RFC8826" format="default"/> and
<xref target='I-D.ietf-rtcweb-security-arch'/> target="RFC8827" format="default"/> for detailed info.</t> information.</li>
        <!--<t>Consent and NAT traversal mechanism: These are handled by the
PeerConnection's ICE <xref target='RFC5245'/> connectivity checks and
optional TURN servers.</t>-->
<li>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
        <li>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>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
        <li>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>It must be possible to implement the protocol stack in the user application space.</t>
</list></t> space.</li>
    <section title='SCTP anchor="sec-p-a-2" numbered="true" toc="default">
      <name>SCTP over DTLS over UDP Considerations'
         anchor='sec-p-a-2'> Considerations</name>
      <t>The important features of SCTP in the WebRTC context are:
<list style='symbols'>
<t>Usage are the following:
      <ul spacing="normal">
        <li>Usage of a TCP-friendly congestion control.</t>
<t>The control.</li>
        <li>modifiable congestion control is modifiable for integration with the
   SRTP media stream congestion control.</t>
<t>Support control.</li>
        <li>Support of multiple unidirectional streams, each providing its own
   notion of ordered message delivery.</t>
<t>Support delivery.</li>
        <li>Support of ordered and out-of-order message delivery.</t>
<t>Supporting arbitrary delivery.</li>
<li>Support of arbitrarily large user messages by providing fragmentation
   and reassembly.</t>
<t>Support reassembly.</li>
        <li>Support of PMTU-discovery.</t>
<t>Support PMTU discovery.</li>
        <li>Support of reliable or partially reliable message transport.</t>
</list></t> transport.</li>
      <t>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 encapsulation of SCTP over DTLS defined in
<xref target='I-D.ietf-tsvwg-sctp-dtls-encaps'/> target="RFC8261" format="default"/> 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"/> enables middlebox traversal
in IPv4 IPv4- and IPv6 based IPv6-based networks.
SCTP as specified in <xref target='RFC4960'/> MUST target="RFC4960" format="default"/> <bcp14>MUST</bcp14> be used in
combination with the extension defined in <xref target='RFC3758'/> target="RFC3758" format="default"/> and
provides the following features for transporting non-media data between
<list style='symbols'>
      <ul spacing="normal">
        <li>Support of multiple unidirectional streams.</t>
<t>Ordered streams.</li>
        <li>Ordered and unordered delivery of user messages.</t>
<t>Reliable messages.</li>
        <li>Reliable and partial-reliable partially reliable transport of user messages.</t>
</list></t> messages.</li>
      <t>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"/>.
Please note that the PPID is not accessible via the 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 encapsulation of SCTP over DTLS, together with the SCTP features listed
above, satisfies all the requirements listed in <xref target='sec-req'/>.</t> target="sec-req" format="default"/>.</t>
<t>There are SCTP implementations for most Operating Systems in wide use:</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>

<section title='User Space versus Kernel Implementation'
<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>
<t>The layering of protocols for WebRTC is shown in the following <xref target='fig-sctp-layering'/>.</t> target="fig-sctp-layering" format="default"/>.</t>
      <figure title='WebRTC protocol layers'
        anchor='fig-sctp-layering'> anchor="fig-sctp-layering">
        <name>WebRTC Protocol Layers</name>
        <artwork align='center'> align="center" name="" type="" alt=""><![CDATA[
              | DCEP | UTF-8|Binary|
              |      | data Data | data Data |
              |        SCTP        |
| STUN | SRTP |        DTLS        |
|                ICE               |
| UDP1 | UDP2 | UDP3 | ...         |
      <t>This stack (especially in contrast to DTLS over SCTP <xref target='RFC6083'/> target="RFC6083" format="default"/> and
in combination with SCTP over UDP <xref target='RFC6951'/>) target="RFC6951" format="default"/>)
has been chosen because it
<list style='symbols'>
<t>supports for the following reasons:
      <ul spacing="normal">
        <li>supports the transmission of arbitrary arbitrarily large user messages.</t>
<t>shares messages;</li>
        <li>shares the DTLS connection with the SRTP media channels of the PeerConnection.</t>
PeerConnection; and</li>
        <li>provides privacy for the SCTP control information.</t>
<t>Considering information.</li>
      <t>In the protocol stack of <xref target='fig-sctp-layering'/> target="fig-sctp-layering" format="default"/>, the
usage of DTLS 1.0 over UDP is specified in <xref target='RFC4347'/> target="RFC4347" format="default"/> and
the usage of DTLS 1.2 over UDP in specified in <xref target='RFC6347'/>, target="RFC6347" format="default"/>,
while the usage of SCTP on top of DTLS is specified in
<xref target='I-D.ietf-tsvwg-sctp-dtls-encaps'/>. target="RFC8261" format="default"/>.
Please note that the demultiplexing STUN Session Traversal Utilities for NAT (STUN)
<xref target="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"/>, and SCTP
is the only payload of DTLS.</t>
      <t>Since DTLS is typically implemented in user application space, the SCTP
stack also needs to be a user application space stack.</t>
      <t>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"/>, this means setting the congestion window and
slow start
slow-start threshold to its initial values.</t>
      <t>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"/>
by using probing messages specified in <xref target='RFC4820'/>. target="RFC4820" format="default"/>.
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 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> connection oriented).</t>
      <t>When the protocol stack of <xref target='fig-sctp-layering'/> target="fig-sctp-layering" format="default"/> 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 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
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 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
    <section title='The anchor="sec-sctp-usage" numbered="true" toc="default">
      <name>The Usage of SCTP for Data Channels'
         anchor='sec-sctp-usage'> Channels</name>
      <section title='SCTP numbered="true" toc="default">
        <name>SCTP Protocol Considerations'> Considerations</name>
        <t>The DTLS encapsulation of SCTP packets as described in
<xref target='I-D.ietf-tsvwg-sctp-dtls-encaps'/> MUST target="RFC8261" format="default"/> <bcp14>MUST</bcp14> be used.</t>
        <t>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
        <t>The following SCTP protocol extensions are required:
<list style='symbols'>
        <ul spacing="normal">
          <li>The stream reconfiguration extension defined in <xref target='RFC6525'/>
   MUST target="RFC6525" format="default"/>
   <bcp14>MUST</bcp14> be supported. It is used for closing channels.</t>
<t>The channels.</li>
          <li>The dynamic address reconfiguration extension defined in
   <xref target='RFC5061'/> MUST target="RFC5061" format="default"/> <bcp14>MUST</bcp14> be used to signal the support of the
   stream reset extension defined in <xref target='RFC6525'/>. target="RFC6525" format="default"/>.
   Other features of <xref target='RFC5061'/> target="RFC5061" format="default"/> are OPTIONAL.</t>
<t>The <bcp14>OPTIONAL</bcp14>.</li>
          <li>The partial reliability extension defined in <xref target='RFC3758'/> MUST target="RFC3758" format="default"/> <bcp14>MUST</bcp14>
   be supported. In addition to the timed reliability PR-SCTP policy defined
   in <xref target='RFC3758'/>, target="RFC3758" format="default"/>, the limited retransmission policy defined in
   <xref target='I-D.ietf-tsvwg-sctp-prpolicies'/> MUST target="RFC7496" format="default"/> <bcp14>MUST</bcp14> be supported.
   Limiting the number of retransmissions to zero zero, combined with unordered
   delivery, provides a UDP-like service where each user message is sent
   exactly once and delivered in the order received.</t>
</list></t> received.</li>
        <t>The support for message interleaving as defined in
<xref target='I-D.ietf-tsvwg-sctp-ndata'/> SHOULD target="RFC8260" format="default"/> <bcp14>SHOULD</bcp14> be used.</t>
      <section title='SCTP anchor="sec-sctp-management" numbered="true" toc="default">
        <name>SCTP Association Management'
         anchor='sec-sctp-management'> Management</name>
        <t>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"/>.
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 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 supports two ways of terminating an SCTP association.
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
        <t>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 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 title='SCTP Streams'> numbered="true" toc="default">
        <name>SCTP Streams</name>
        <t>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 title='Data numbered="true" toc="default">
        <name>Data Channel Definition'> Definition</name>
        <t>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 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, each data channel has the following properties in each
<list style='symbols'>
        <ul spacing="normal">
          <li>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>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>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"/>.
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>an optional label.</t>
<t>an label.</li>
          <li>an optional protocol.</t>
<t>Please note protocol.</li>
        <t>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"/>, all of the above
properties are the same in both directions.</t>
      <section title='Opening numbered="true" toc="default">
        <name>Opening a Data Channel'> Channel</name>
        <t>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> endpoints.</t>
        <t>A simple protocol for in-band negotiation is specified in
<xref target='I-D.ietf-rtcweb-data-protocol'/>.</t> target="RFC8832" format="default"/>.</t>
        <t>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 title='Transferring numbered="true" toc="default">
        <name>Transferring User Data on a Data Channel'> Channel</name>
        <t>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, unless the options are changed, changed or per-message options are specified
by a higher level.</t>
        <t>The message-orientation 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 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"/>):
        <dl newline="false" spacing="normal">
          <dt>WebRTC String:</dt>
to identify a non-empty JavaScript string encoded in UTF-8.</t>
<t hangText='WebRTC UTF-8.</dd>
          <dt>WebRTC String Empty:'> Empty:</dt>
to identify an empty JavaScript string encoded in UTF-8.</t>
<t hangText='WebRTC Binary:'> UTF-8.</dd>
          <dt>WebRTC Binary:</dt>
to identify a non-empty JavaScript binary data
(ArrayBuffer, ArrayBufferView ArrayBufferView, or Blob).</t>
<t hangText='WebRTC Blob).</dd>
          <dt>WebRTC Binary Empty:'> Empty:</dt>
to identify an empty JavaScript binary data
(ArrayBuffer, ArrayBufferView ArrayBufferView, or Blob).</t>
</list></t> Blob).</dd>
        <t>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
<bcp14>MUST</bcp14> ignore the SCTP user message and process it as an empty message.</t>
        <t>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 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 SCTP base protocol specified in <xref target='RFC4960'/> target="RFC4960" format="default"/> 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"/>
defines an extension to support message interleaving, which SHOULD <bcp14>SHOULD</bcp14> be used.
As long as message interleaving is not supported, the sender
<bcp14>SHOULD</bcp14> limit the maximum message size to 16 KB to avoid monopolization.</t>
        <t>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> target="RFC8841" format="default"/>.</t>
        <t>The sender SHOULD <bcp14>SHOULD</bcp14> disable the Nagle algorithm (see <xref target='RFC1122'/>) target="RFC1122" format="default"/>)
to minimize the latency.</t>
      <section title='Closing numbered="true" toc="default">
        <name>Closing a Data Channel'> Channel</name>
        <t>Closing of a data channel MUST <bcp14>MUST</bcp14> be signaled by resetting the corresponding
outgoing streams <xref target='RFC6525'/>. target="RFC6525" format="default"/>. 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'/> target="RFC6525" format="default"/> also guarantees that all the messages are delivered
(or abandoned) before the stream is reset.</t>
    <section title='Security Considerations'
         anchor='sec-security'> anchor="sec-security" numbered="true" toc="default">
      <name>Security Considerations</name>
      <t>This document does not add any additional considerations to the ones given in
<xref target='I-D.ietf-rtcweb-security'/> target="RFC8826" format="default"/> and
<xref target='I-D.ietf-rtcweb-security-arch'/>.</t> target="RFC8827" format="default"/>.</t>
      <t>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 title='IANA Considerations'
<t>[NOTE to RFC-Editor:
<t>"RFCXXXX" is to be replaced by the RFC number you assign this document.</t>
]</t> anchor="sec-IANA" numbered="true" toc="default">
      <name>IANA Considerations</name>
      <t>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"/> 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>The six assignments should be have been updated to read:</t>
<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>
      <table align="center">
            <th align="left">Value</th>
            <th align="left">SCTP PPID</th>
            <th align="left">Reference</th>
            <th align="left">Date</th>
            <td align="left">WebRTC String</td>
            <td align="left">51</td>
            <td align="left">RFC 8831</td>
            <td align="left">2013-09-20</td>
            <td align="left">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">52</td>
            <td align="left">RFC 8831</td>
            <td align="left">2013-09-20</td>
            <td align="left">WebRTC Binary</td>
            <td align="left">53</td>
            <td align="left">RFC 8831</td>
            <td align="left">2013-09-20</td>
            <td align="left">WebRTC String Partial (Deprecated)</c> <c>54</c> <c>[RFCXXXX]</c> <c>2013-09-20</c>
<c>WebRTC (deprecated)</td>
            <td align="left">54</td>
            <td align="left">RFC 8831</td>
            <td align="left">2013-09-20</td>
            <td align="left">WebRTC String Empty</c>                <c>56</c> <c>[RFCXXXX]</c> <c>2014-08-22</c>
<c>WebRTC Empty</td>
            <td align="left">56</td>
            <td align="left">RFC 8831</td>
            <td align="left">2014-08-22</td>
            <td align="left">WebRTC Binary Empty</c>                <c>57</c> <c>[RFCXXXX]</c> <c>2014-08-22</c>
</texttable> Empty</td>
            <td align="left">57</td>
            <td align="left">RFC 8831</td>
            <td align="left">2014-08-22</td>
        <name>Normative References</name>
        <xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.2119.xml"/>
        <xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.3758.xml"/>
        <xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.4347.xml"/>
        <xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.4820.xml"/>
        <xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.4821.xml"/>
        <xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.4960.xml"/>
        <xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.5061.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.8445.xml"/>
        <xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.6347.xml"/>
        <xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.6525.xml"/>
        <xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.8174.xml"/>
        <xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.8260.xml"/>
<!--draft-ietf-rtcweb-data-protocol: 8832 -->
<reference anchor="RFC8832" target="https://www.rfc-editor.org/info/rfc8832">
<title>WebRTC Data Channel Establishment Protocol</title>
<author initials='R.' surname='Jesup' fullname='Randell Jesup'>
<author initials='S.' surname='Loreto' fullname='Salvatore Loreto'>
<author initials='M' surname='Tüxen' fullname='Michael Tüxen'>
<date month='September' year='2020'/>
<seriesInfo name="RFC" value="8832"/>
<seriesInfo name="DOI" value="10.17487/RFC8832"/>

        <xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.8261.xml"/>

 <!--draft-ietf-rtcweb-security: RFC 8826 -->
 <reference anchor="RFC8826" target="https://www.rfc-editor.org/info/rfc8826">
 <title>Security Considerations for WebRTC</title>
 <author initials='E.' surname='Rescorla' fullname='Eric Rescorla'>
 <date month='September' year='2020'/>
 <seriesInfo name="RFC" value="8826"/>
 <seriesInfo name="DOI" value="10.17487/RFC8826"/>

 <!--draft-ietf-rtcweb-security-arch: 8827 -->
 <reference anchor="RFC8827" target="https://www.rfc-editor.org/info/rfc8827">
 <title>WebRTC Security Architecture</title>
 <author initials='E.' surname='Rescorla' fullname='Eric Rescorla'>
 <date month='September' year='2020'/>
 <seriesInfo name="RFC" value="8827"/>
 <seriesInfo name="DOI" value="10.17487/RFC8827"/>

<!--draft-ietf-rtcweb-jsep: 8829 -->
 <reference anchor="RFC8829" target="https://www.rfc-editor.org/info/rfc8829">
 <title>JavaScript Session Establishment Protocol (JSEP)</title>
 <author initials='J.' surname='Uberti' fullname='Justin Uberti'>
 <author initials="C." surname="Jennings" fullname="Cullen Jennings">
 <author initials="E." surname="Rescorla" fullname="Eric Rescorla"
 <date month='September' year='2020'/>
 <seriesInfo name="RFC" value="8829"/>
 <seriesInfo name="DOI" value="10.17487/RFC8829"/>

        <xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.7496.xml"/>

<!-- draft-ietf-mmusic-sctp-sdp: 8841 -->
<reference anchor="RFC8841" target="https://www.rfc-editor.org/info/rfc8841">

    <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 />

    <author initials="R." surname="Shpount" fullname="Roman Shpount">
      <organization />

    <author initials="S." surname="Loreto" fullname="Salvatore Loreto">
      <organization />

    <author initials="G." surname="Camarillo" fullname="Gonzalo Camarillo">
      <organization />

    <date month="September" year="2020" />
  <seriesInfo name="RFC" value="8841" />
  <seriesInfo name="DOI" value="10.17487/RFC8841"/>


        <name>Informative References</name>
        <xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.1122.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.5389.xml"/>

        <xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.5764.xml"/>
        <xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.6083.xml"/>
        <xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.6951.xml"/>
    <section title='Acknowledgments'> numbered="false" toc="default">
      <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> <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>

<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 title='Informative References'>
<?rfc include='reference.RFC.1122'?>
<?rfc include='reference.RFC.5764'?>
<?rfc include='reference.RFC.6083'?>
<?rfc include='reference.RFC.6951'?>