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WebRTC Data Channels
RFC 8831

Document Type RFC - Proposed Standard (January 2021)
Authors Randell Jesup , Salvatore Loreto , Michael Tüxen
Last updated 2021-01-18
RFC stream Internet Engineering Task Force (IETF)
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IESG Responsible AD Alissa Cooper
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RFC 8831

Internet Engineering Task Force (IETF)                          R. Jesup
Request for Comments: 8831                                       Mozilla
Category: Standards Track                                      S. Loreto
ISSN: 2070-1721                                                 Ericsson
                                                                M. Tüxen
                                         Münster Univ. of Appl. Sciences
                                                            January 2021

                          WebRTC Data Channels


   The WebRTC framework specifies protocol support for direct,
   interactive, rich communication using audio, video, and data between
   two peers' 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 that allows web browsers to exchange generic data from peer
   to peer.

Status of This Memo

   This is an Internet Standards Track document.

   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.

   Information about the current status of this document, any errata,
   and how to provide feedback on it may be obtained at

Copyright Notice

   Copyright (c) 2021 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   ( 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.

Table of Contents

   1.  Introduction
   2.  Conventions
   3.  Use Cases
     3.1.  Use Cases for Unreliable Data Channels
     3.2.  Use Cases for Reliable Data Channels
   4.  Requirements
   5.  SCTP over DTLS over UDP Considerations
   6.  The Usage of SCTP for Data Channels
     6.1.  SCTP Protocol Considerations
     6.2.  SCTP Association Management
     6.3.  SCTP Streams
     6.4.  Data Channel Definition
     6.5.  Opening a Data Channel
     6.6.  Transferring User Data on a Data Channel
     6.7.  Closing a Data Channel
   7.  Security Considerations
   8.  IANA Considerations
   9.  References
     9.1.  Normative References
     9.2.  Informative References
   Authors' Addresses

1.  Introduction

   In the WebRTC framework, communication between the parties consists
   of media (for example, audio and video) and non-media data.  Media is
   sent using the Secure Real-time Transport Protocol (SRTP) and is not
   specified further here.  Non-media data is handled by using the
   Stream Control Transmission Protocol (SCTP) [RFC4960] encapsulated in
   DTLS.  DTLS 1.0 is defined in [RFC4347]; the present latest version,
   DTLS 1.2, is defined in [RFC6347]; and an upcoming version, DTLS 1.3,
   is defined in [TLS-DTLS13].

                               |   SCTP   |
                               |   DTLS   |
                               | ICE/UDP  |

                       Figure 1: Basic Stack Diagram

   The encapsulation of SCTP over DTLS (see [RFC8261]) over ICE/UDP (see
   [RFC8445]) provides a NAT traversal solution together with
   confidentiality, source authentication, and 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.

   SCTP, as specified in [RFC4960] with the partial reliability
   extension (PR-SCTP) defined in [RFC3758] and the additional policies
   defined in [RFC7496], provides multiple streams natively with
   reliable, and the relevant partially reliable, delivery modes for
   user messages.  Using the reconfiguration extension defined in
   [RFC6525] allows an increase in the number of streams during the
   lifetime of an SCTP association and allows individual SCTP streams to
   be reset.  Using [RFC8260] allows the interleave of large messages to
   avoid monopolization and adds support for prioritizing SCTP streams.

   The remainder of this document is organized as follows: Sections 3
   and 4 provide use cases and requirements for both unreliable and
   reliable peer-to-peer data channels; Section 5 discusses SCTP over
   DTLS over UDP; and Section 6 specifies how SCTP should be used by the
   WebRTC protocol framework for transporting non-media data between web

2.  Conventions

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "OPTIONAL" in this document are to be interpreted as described in
   BCP 14 [RFC2119] [RFC8174] when, and only when, they appear in all
   capitals, as shown here.

3.  Use Cases

   This section defines use cases specific to data channels.  Please
   note that this section is informational only.

3.1.  Use Cases for Unreliable Data Channels

   U-C 1:  A real-time game where position and object state information
           are sent via one or more unreliable data channels.  Note that
           at any time, there may not be any SRTP media channels or all
           SRTP media channels may be inactive, and there may also be
           reliable data channels in use.

   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.

3.2.  Use Cases for Reliable Data Channels

   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 or may only be added due to in-game actions.

   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.

   U-C 5:  Real-time text chat during an audio and/or video call with an
           individual or with multiple people in a conference.

   U-C 6:  Renegotiation of the configuration of the PeerConnection.

   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, to avoid local Internet filtering or

4.  Requirements

   This section lists the requirements for Peer-to-Peer (P2P) data
   channels between two browsers.  Please note that this section is
   informational only.

   Req. 1:   Multiple simultaneous data channels must be supported.
             Note that there may be 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.

   Req. 2:   Both reliable and unreliable data channels must be

   Req. 3:   Data channels of a PeerConnection must be congestion
             controlled either individually, as a class, or in
             conjunction with the SRTP media streams of the
             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.

   Req. 4:   The application should be able to provide guidance as to
             the relative priority of each data channel relative to each
             other and relative to the SRTP media streams.  This will
             interact with the congestion control algorithms.

   Req. 5:   Data channels must be secured, which allows for
             confidentiality, integrity, and source authentication.  See
             [RFC8826] and [RFC8827] for detailed information.

   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

   Req. 7:   The data channel transport protocol must not encode local
             IP addresses inside its protocol fields; doing so reveals
             potentially private information and leads to failure if the
             address is depended upon.

   Req. 8:   The data channel transport protocol should support
             unbounded-length "messages" (i.e., a virtual socket stream)
             at the application layer for such things as image-file-
             transfer; implementations might enforce a reasonable
             message size limit.

   Req. 9:   The data channel transport protocol should avoid IP
             fragmentation.  It must support Path MTU (PMTU) discovery
             and must not rely on ICMP or ICMPv6 being generated or
             being passed back, especially for PMTU discovery.

   Req. 10:  It must be possible to implement the protocol stack in the
             user application space.

5.  SCTP over DTLS over UDP Considerations

   The important features of SCTP in the WebRTC context are the

   *  Usage of TCP-friendly congestion control.

   *  modifiable congestion control for integration with the SRTP media
      stream congestion control.

   *  Support of multiple unidirectional streams, each providing its own
      notion of ordered message delivery.

   *  Support of ordered and out-of-order message delivery.

   *  Support of arbitrarily large user messages by providing
      fragmentation and reassembly.

   *  Support of PMTU discovery.

   *  Support of reliable or partially reliable message transport.

   The WebRTC 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, unreliable datagram service) exposes.

   The encapsulation of SCTP over DTLS defined in [RFC8261] provides
   confidentiality, source authentication, and integrity-protected
   transfers.  Using DTLS over UDP in combination with Interactive
   Connectivity Establishment (ICE) [RFC8445] enables middlebox
   traversal in IPv4- and IPv6-based networks.  SCTP as specified in
   [RFC4960] MUST be used in combination with the extension defined in
   [RFC3758] and provides the following features for transporting non-
   media data between browsers:

   *  Support of multiple unidirectional streams.

   *  Ordered and unordered delivery of user messages.

   *  Reliable and partially reliable transport of user messages.

   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 user
   data, and the Data Channel Establishment Protocol (DCEP) defined in
   [RFC8832].  Please note that the PPID is not accessible via the
   JavaScript API.

   The encapsulation of SCTP over DTLS, together with the SCTP features
   listed above, satisfies all the requirements listed in Section 4.

   The layering of protocols for WebRTC is shown in Figure 2.

                                 | DCEP | UTF-8|Binary|
                                 |      | Data | Data |
                                 |        SCTP        |
                   | STUN | SRTP |        DTLS        |
                   |                ICE               |
                   | UDP1 | UDP2 | UDP3 | ...         |

                      Figure 2: WebRTC Protocol Layers

   This stack (especially in contrast to DTLS over SCTP [RFC6083] and in
   combination with SCTP over UDP [RFC6951]) has been chosen for the
   following reasons:

   *  supports the transmission of arbitrarily large user messages;

   *  shares the DTLS connection with the SRTP media channels of the
      PeerConnection; and

   *  provides privacy for the SCTP control information.

   Referring to the protocol stack shown in Figure 2:

   *  the usage of DTLS 1.0 over UDP is specified in [RFC4347];

   *  the usage of DTLS 1.2 over UDP in specified in [RFC6347];

   *  the usage of DTLS 1.3 over UDP is specified in an upcoming
      document [TLS-DTLS13]; and

   *  the usage of SCTP on top of DTLS is specified in [RFC8261].

   Please note that the demultiplexing Session Traversal Utilities for
   NAT (STUN) [RFC5389] vs. SRTP vs. DTLS is done as described in
   Section 5.1.2 of [RFC5764], and SCTP is the only payload of DTLS.

   Since DTLS is typically implemented in user application space, the
   SCTP stack also needs to be a user application space stack.

   The ICE/UDP layer can handle IP address changes during a session
   without needing interaction with the DTLS and SCTP layers.  However,
   SCTP SHOULD be notified when an address change has happened.  In this
   case, SCTP SHOULD retest the Path MTU and reset the congestion state
   to the initial state.  In the case of window-based congestion control
   like the one specified in [RFC4960], this means setting the
   congestion window and slow-start threshold to its initial values.

   Incoming ICMP or ICMPv6 messages can't be processed by the SCTP
   layer, since there is no way to identify the corresponding
   association.  Therefore, SCTP MUST support performing Path MTU
   discovery without relying on ICMP or ICMPv6 as specified in [RFC4821]
   by using probing messages specified in [RFC4820].  The initial Path
   MTU at the IP layer SHOULD NOT exceed 1200 bytes for IPv4 and 1280
   bytes for IPv6.

   In general, the lower-layer interface of an SCTP implementation
   should be adapted to address the differences between IPv4 and IPv6
   (being connectionless) or DTLS (being connection oriented).

   When the protocol stack shown in Figure 2 is used, DTLS protects the
   complete SCTP packet, so it provides confidentiality, integrity, and
   source authentication of the complete SCTP packet.

   SCTP provides congestion control on a per-association 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 the standard one might improve the impact on the
   parallel SRTP media streams.

   SCTP uses the same port number concept as TCP and UDP.  Therefore, an
   SCTP association uses two port numbers, one at each SCTP endpoint.

6.  The Usage of SCTP for Data Channels

6.1.  SCTP Protocol Considerations

   The DTLS encapsulation of SCTP packets as described in [RFC8261] MUST
   be used.

   This SCTP stack and its upper layer MUST support the usage of
   multiple SCTP streams.  A user message can be sent ordered or
   unordered and with partial or full reliability.

   The following SCTP protocol extensions are required:

   *  The stream reconfiguration extension defined in [RFC6525] MUST be
      supported.  It is used for closing channels.

   *  The dynamic address reconfiguration extension defined in [RFC5061]
      MUST be used to signal the support of the stream reset extension
      defined in [RFC6525].  Other features of [RFC5061] are OPTIONAL.

   *  The partial reliability extension defined in [RFC3758] MUST be
      supported.  In addition to the timed reliability PR-SCTP policy
      defined in [RFC3758], the limited retransmission policy defined in
      [RFC7496] MUST be supported.  Limiting the number of
      retransmissions to zero, combined with unordered delivery,
      provides a UDP-like service where each user message is sent
      exactly once and delivered in the order received.

   The support for message interleaving as defined in [RFC8260] SHOULD
   be used.

6.2.  SCTP Association Management

   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 the JavaScript Session Establishment Protocol (JSEP),
   which is typically an exchange of the Session Description Protocol
   (SDP) [RFC8829].  It will use the DTLS connection selected via ICE,
   and typically this will be shared via BUNDLE or equivalent with DTLS
   connections used to key the SRTP media streams.

   The number of streams negotiated during SCTP association setup SHOULD
   be 65535, which is the maximum number of streams that can be
   negotiated during the association setup.

   SCTP supports two ways of terminating an SCTP association.  The first
   method is a graceful one, where a procedure that ensures no messages
   are lost during the shutdown of the association is used.  The second
   method is a non-graceful one, where one side can just abort the

   Each SCTP 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.

   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 be provided
   if possible.

6.3.  SCTP Streams

   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.

6.4.  Data Channel Definition

   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.

   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.

   Additionally, each data channel has the following properties in each

   *  reliable or unreliable message transmission: In case of unreliable
      transmissions, the same level of unreliability is used.  Note
      that, in SCTP, this is a property of an SCTP user message and not
      of an SCTP stream.

   *  in-order or out-of-order message delivery for message sent: Note
      that, in SCTP, this is a property of an SCTP user message and not
      of an SCTP stream.

   *  a priority, which is a 2-byte unsigned integer: These priorities
      MUST be interpreted as weighted-fair-queuing scheduling priorities
      per the definition of the corresponding stream scheduler
      supporting interleaving in [RFC8260].  For use in WebRTC, the
      values used SHOULD be one of 128 ("below normal"), 256 ("normal"),
      512 ("high"), or 1024 ("extra high").

   *  an optional label.

   *  an optional protocol.

   Note that for a data channel being negotiated with the protocol
   specified in [RFC8832], all of the above properties are the same in
   both directions.

6.5.  Opening a Data Channel

   Data channels can be opened by using negotiation within the SCTP
   association (called in-band negotiation) or out-of-band negotiation.
   Out-of-band negotiation is defined as any method 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 endpoints.

   A simple protocol for in-band negotiation is specified in [RFC8832].

   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 reuse a stream that is part of
   an existing data channel, the addition MUST fail.  In addition to
   choosing a stream, the application SHOULD also determine the options
   to be used for sending messages.  The application MUST ensure in an
   application-specific manner that the application at the peer will
   also know the selected stream to be used, as well as the options for
   sending data from that side.

6.6.  Transferring User Data on a Data Channel

   All data sent on a data channel in both directions MUST be sent over
   the underlying stream using the reliability defined when the data
   channel was opened, unless the options are changed or per-message
   options are specified by a higher level.

   The message orientation of SCTP is used to preserve the message
   boundaries of user messages.  Therefore, senders MUST NOT put more
   than one application message into an SCTP user message.  Unless the
   deprecated PPID-based fragmentation and reassembly is used, the
   sender MUST include exactly one application message in each SCTP user

   The SCTP Payload Protocol Identifiers (PPIDs) are used to signal the
   interpretation of the "payload data".  The following PPIDs MUST be
   used (see Section 8):

   WebRTC String:  to identify a non-empty JavaScript string encoded in

   WebRTC String Empty:  to identify an empty JavaScript string encoded
      in UTF-8.

   WebRTC Binary:  to identify non-empty JavaScript binary data
      (ArrayBuffer, ArrayBufferView, or Blob).

   WebRTC Binary Empty:  to identify empty JavaScript binary data
      (ArrayBuffer, ArrayBufferView, or Blob).

   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, 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 ignore the SCTP
   user message and process it as an empty message.

   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.

   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 close the
   corresponding data channel.  This implies in particular that
   extensions using additional PPIDs can't be used without prior

   The SCTP base protocol specified in [RFC4960] does not support the
   interleaving of user messages.  Therefore, sending a large user
   message can monopolize the SCTP association.  To overcome this
   limitation, [RFC8260] defines an extension to support message
   interleaving, which SHOULD be used.  As long as message interleaving
   is not supported, the sender SHOULD limit the maximum message size to
   16 KB to avoid monopolization.

   It is recommended that the message size be kept within certain size
   bounds, as applications will not be able to support arbitrarily large
   single messages.  This limit has to be negotiated, for example, by
   using [RFC8841].

   The sender SHOULD disable the Nagle algorithm (see [RFC1122]) to
   minimize the latency.

6.7.  Closing a Data Channel

   Closing of a data channel MUST be signaled by resetting the
   corresponding outgoing streams [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.

   [RFC6525] also guarantees that all the messages are delivered (or
   abandoned) before the stream is reset.

7.  Security Considerations

   This document does not add any additional considerations to the ones
   given in [RFC8826] and [RFC8827].

   It should be noted that a receiver must be prepared for a sender that
   tries to send arbitrarily large messages.

8.  IANA Considerations

   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".  [RFC4960] creates the "SCTP Payload Protocol
   Identifiers" registry from which these identifiers were assigned.
   IANA has updated the reference of these six assignments to point to
   this document and changed the names of the first four PPIDs.  The
   corresponding dates remain unchanged.

   The six assignments have been updated to read:

       | Value                | SCTP PPID | Reference | Date       |
       | WebRTC String        | 51        | RFC 8831  | 2013-09-20 |
       | WebRTC Binary        | 52        | RFC 8831  | 2013-09-20 |
       | Partial (deprecated) |           |           |            |
       | WebRTC Binary        | 53        | RFC 8831  | 2013-09-20 |
       | WebRTC String        | 54        | RFC 8831  | 2013-09-20 |
       | Partial (deprecated) |           |           |            |
       | WebRTC String Empty  | 56        | RFC 8831  | 2014-08-22 |
       | WebRTC Binary Empty  | 57        | RFC 8831  | 2014-08-22 |

                                  Table 1

9.  References

9.1.  Normative References

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119,
              DOI 10.17487/RFC2119, March 1997,

   [RFC3758]  Stewart, R., Ramalho, M., Xie, Q., Tuexen, M., and P.
              Conrad, "Stream Control Transmission Protocol (SCTP)
              Partial Reliability Extension", RFC 3758,
              DOI 10.17487/RFC3758, May 2004,

   [RFC4820]  Tuexen, M., Stewart, R., and P. Lei, "Padding Chunk and
              Parameter for the Stream Control Transmission Protocol
              (SCTP)", RFC 4820, DOI 10.17487/RFC4820, March 2007,

   [RFC4821]  Mathis, M. and J. Heffner, "Packetization Layer Path MTU
              Discovery", RFC 4821, DOI 10.17487/RFC4821, March 2007,

   [RFC4960]  Stewart, R., Ed., "Stream Control Transmission Protocol",
              RFC 4960, DOI 10.17487/RFC4960, September 2007,

   [RFC5061]  Stewart, R., Xie, Q., Tuexen, M., Maruyama, S., and M.
              Kozuka, "Stream Control Transmission Protocol (SCTP)
              Dynamic Address Reconfiguration", RFC 5061,
              DOI 10.17487/RFC5061, September 2007,

   [RFC6525]  Stewart, R., Tuexen, M., and P. Lei, "Stream Control
              Transmission Protocol (SCTP) Stream Reconfiguration",
              RFC 6525, DOI 10.17487/RFC6525, February 2012,

   [RFC7496]  Tuexen, M., Seggelmann, R., Stewart, R., and S. Loreto,
              "Additional Policies for the Partially Reliable Stream
              Control Transmission Protocol Extension", RFC 7496,
              DOI 10.17487/RFC7496, April 2015,

   [RFC8174]  Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC
              2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174,
              May 2017, <>.

   [RFC8260]  Stewart, R., Tuexen, M., Loreto, S., and R. Seggelmann,
              "Stream Schedulers and User Message Interleaving for the
              Stream Control Transmission Protocol", RFC 8260,
              DOI 10.17487/RFC8260, November 2017,

   [RFC8261]  Tuexen, M., Stewart, R., Jesup, R., and S. Loreto,
              "Datagram Transport Layer Security (DTLS) Encapsulation of
              SCTP Packets", RFC 8261, DOI 10.17487/RFC8261, November
              2017, <>.

   [RFC8445]  Keranen, A., Holmberg, C., and J. Rosenberg, "Interactive
              Connectivity Establishment (ICE): A Protocol for Network
              Address Translator (NAT) Traversal", RFC 8445,
              DOI 10.17487/RFC8445, July 2018,

   [RFC8826]  Rescorla, E., "Security Considerations for WebRTC",
              RFC 8826, DOI 10.17487/RFC8826, January 2021,

   [RFC8827]  Rescorla, E., "WebRTC Security Architecture", RFC 8827,
              DOI 10.17487/RFC8827, January 2021,

   [RFC8829]  Uberti, J., Jennings, C., and E. Rescorla, Ed.,
              "JavaScript Session Establishment Protocol (JSEP)",
              RFC 8829, DOI 10.17487/RFC8829, January 2021,

   [RFC8832]  Jesup, R., Loreto, S., and M. Tüxen, "WebRTC Data Channel
              Establishment Protocol", RFC 8832, DOI 10.17487/RFC8832,
              January 2021, <>.

   [RFC8841]  Holmberg, C., Shpount, R., Loreto, S., and G. Camarillo,
              "Session Description Protocol (SDP) Offer/Answer
              Procedures for Stream Control Transmission Protocol (SCTP)
              over Datagram Transport Layer Security (DTLS) Transport",
              RFC 8841, DOI 10.17487/RFC8841, January 2021,

9.2.  Informative References

   [RFC1122]  Braden, R., Ed., "Requirements for Internet Hosts -
              Communication Layers", STD 3, RFC 1122,
              DOI 10.17487/RFC1122, October 1989,

   [RFC4347]  Rescorla, E. and N. Modadugu, "Datagram Transport Layer
              Security", RFC 4347, DOI 10.17487/RFC4347, April 2006,

   [RFC5389]  Rosenberg, J., Mahy, R., Matthews, P., and D. Wing,
              "Session Traversal Utilities for NAT (STUN)", RFC 5389,
              DOI 10.17487/RFC5389, October 2008,

   [RFC5764]  McGrew, D. and E. Rescorla, "Datagram Transport Layer
              Security (DTLS) Extension to Establish Keys for the Secure
              Real-time Transport Protocol (SRTP)", RFC 5764,
              DOI 10.17487/RFC5764, May 2010,

   [RFC6083]  Tuexen, M., Seggelmann, R., and E. Rescorla, "Datagram
              Transport Layer Security (DTLS) for Stream Control
              Transmission Protocol (SCTP)", RFC 6083,
              DOI 10.17487/RFC6083, January 2011,

   [RFC6347]  Rescorla, E. and N. Modadugu, "Datagram Transport Layer
              Security Version 1.2", RFC 6347, DOI 10.17487/RFC6347,
              January 2012, <>.

   [RFC6951]  Tuexen, M. and R. Stewart, "UDP Encapsulation of Stream
              Control Transmission Protocol (SCTP) Packets for End-Host
              to End-Host Communication", RFC 6951,
              DOI 10.17487/RFC6951, May 2013,

              Rescorla, E., Tschofenig, H., and N. Modadugu, "The
              Datagram Transport Layer Security (DTLS) Protocol Version
              1.3", Work in Progress, Internet-Draft, draft-ietf-tls-
              dtls13-39, 2 November 2020,


   Many thanks for comments, ideas, and text from Harald Alvestrand,
   Richard Barnes, Adam Bergkvist, Alissa Cooper, Benoit Claise, Spencer
   Dawkins, Gunnar Hellström, Christer Holmberg, Cullen Jennings, Paul
   Kyzivat, Eric Rescorla, Adam Roach, Irene Rüngeler, Randall Stewart,
   Martin Stiemerling, Justin Uberti, and Magnus Westerlund.

Authors' Addresses

   Randell Jesup
   United States of America


   Salvatore Loreto
   Hirsalantie 11
   FI-02420 Jorvas


   Michael Tüxen
   Münster University of Applied Sciences
   Stegerwaldstrasse 39
   48565  Steinfurt