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IETF RFC 8831
Last modified on Monday, January 18th, 2021
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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
 Abstract
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
https://www.rfc-editor.org/info/RFC 8831.
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
(https://trustee.ietf.org/license-info) 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
Acknowledgements
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) [RFC 4960] encapsulated in
DTLS. DTLS 1.0 is defined in [RFC 4347]; the present latest version,
DTLS 1.2, is defined in [RFC 6347]; 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 [RFC 8261]) over ICE/UDP (see
[RFC 8445]) 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 [RFC 4960] with the partial reliability
extension (PR-SCTP) defined in [RFC 3758] and the additional policies
defined in [RFC 7496], provides multiple streams natively with
reliable, and the relevant partially reliable, delivery modes for
user messages. Using the reconfiguration extension defined in
[RFC 6525] allows an increase in the number of streams during the
lifetime of an SCTP association and allows individual SCTP streams to
be reset. Using [RFC 8260] 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
browsers.
2. Conventions
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
"OPTIONAL" in this document are to be interpreted as described in
BCP 14 [RFC 2119] [RFC 8174] 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
monitoring.
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
supported.
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
[RFC 8826] and [RFC 8827] 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
channels.
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
following:
* 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 [RFC 8261] provides
confidentiality, source authentication, and integrity-protected
transfers. Using DTLS over UDP in combination with Interactive
Connectivity Establishment (ICE) [RFC 8445] enables middlebox
traversal in IPv4- and IPv6-based networks. SCTP as specified in
[RFC 4960] MUST be used in combination with the extension defined in
[RFC 3758] 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
[RFC 8832]. 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 [RFC 6083] and in
combination with SCTP over UDP [RFC 6951]) 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 [RFC 4347];
* the usage of DTLS 1.2 over UDP in specified in [RFC 6347];
* 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 [RFC 8261].
Please note that the demultiplexing Session Traversal Utilities for
NAT (STUN) [RFC 5389] vs. SRTP vs. DTLS is done as described in
Section 5.1.2 of [RFC 5764], 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 [RFC 4960], 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 [RFC 4821]
by using probing messages specified in [RFC 4820]. 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 [RFC 8261] 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 [RFC 6525] MUST be
supported. It is used for closing channels.
* The dynamic address reconfiguration extension defined in [RFC 5061]
MUST be used to signal the support of the stream reset extension
defined in [RFC 6525]. Other features of [RFC 5061] are OPTIONAL.
* The partial reliability extension defined in [RFC 3758] MUST be
supported. In addition to the timed reliability PR-SCTP policy
defined in [RFC 3758], the limited retransmission policy defined in
[RFC 7496] 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 [RFC 8260] 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) [RFC 8829]. 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
association.
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
direction:
* 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 [RFC 8260]. 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 [RFC 8832], 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 [RFC 8832].
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
message.
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
UTF-8.
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
negotiation.
The SCTP base protocol specified in [RFC 4960] does not support the
interleaving of user messages. Therefore, sending a large user
message can monopolize the SCTP association. To overcome this
limitation, [RFC 8260] 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 [RFC 8841].
The sender SHOULD disable the Nagle algorithm (see [RFC 1122]) to
minimize the latency.
6.7. Closing a Data Channel
Closing of a data channel MUST be signaled by resetting the
corresponding outgoing streams [RFC 6525]. 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.
[RFC 6525] 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 [RFC 8826] and [RFC 8827].
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". [RFC 4960] 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
[RFC 2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC 2119, March 1997,
<https://www.rfc-editor.org/info/RFC 2119>.
[RFC 3758] Stewart, R., Ramalho, M., Xie, Q., Tuexen, M., and P.
Conrad, "Stream Control Transmission Protocol (SCTP)
Partial Reliability Extension", RFC 3758,
DOI 10.17487/RFC 3758, May 2004,
<https://www.rfc-editor.org/info/RFC 3758>.
[RFC 4820] Tuexen, M., Stewart, R., and P. Lei, "Padding Chunk and
Parameter for the Stream Control Transmission Protocol
(SCTP)", RFC 4820, DOI 10.17487/RFC 4820, March 2007,
<https://www.rfc-editor.org/info/RFC 4820>.
[RFC 4821] Mathis, M. and J. Heffner, "Packetization Layer Path MTU
Discovery", RFC 4821, DOI 10.17487/RFC 4821, March 2007,
<https://www.rfc-editor.org/info/RFC 4821>.
[RFC 4960] Stewart, R., Ed., "Stream Control Transmission Protocol",
RFC 4960, DOI 10.17487/RFC 4960, September 2007,
<https://www.rfc-editor.org/info/RFC 4960>.
[RFC 5061] Stewart, R., Xie, Q., Tuexen, M., Maruyama, S., and M.
Kozuka, "Stream Control Transmission Protocol (SCTP)
Dynamic Address Reconfiguration", RFC 5061,
DOI 10.17487/RFC 5061, September 2007,
<https://www.rfc-editor.org/info/RFC 5061>.
[RFC 6525] Stewart, R., Tuexen, M., and P. Lei, "Stream Control
Transmission Protocol (SCTP) Stream Reconfiguration",
RFC 6525, DOI 10.17487/RFC 6525, February 2012,
<https://www.rfc-editor.org/info/RFC 6525>.
[RFC 7496] 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/RFC 7496, April 2015,
<https://www.rfc-editor.org/info/RFC 7496>.
[RFC 8174] Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC
2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC 8174,
May 2017, <https://www.rfc-editor.org/info/RFC 8174>.
[RFC 8260] 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/RFC 8260, November 2017,
<https://www.rfc-editor.org/info/RFC 8260>.
[RFC 8261] Tuexen, M., Stewart, R., Jesup, R., and S. Loreto,
"Datagram Transport Layer Security (DTLS) Encapsulation of
SCTP Packets", RFC 8261, DOI 10.17487/RFC 8261, November
2017, <https://www.rfc-editor.org/info/RFC 8261>.
[RFC 8445] Keranen, A., Holmberg, C., and J. Rosenberg, "Interactive
Connectivity Establishment (ICE): A Protocol for Network
Address Translator (NAT) Traversal", RFC 8445,
DOI 10.17487/RFC 8445, July 2018,
<https://www.rfc-editor.org/info/RFC 8445>.
[RFC 8826] Rescorla, E., "Security Considerations for WebRTC",
RFC 8826, DOI 10.17487/RFC 8826, January 2021,
<https://www.rfc-editor.org/info/RFC 8826>.
[RFC 8827] Rescorla, E., "WebRTC Security Architecture", RFC 8827,
DOI 10.17487/RFC 8827, January 2021,
<https://www.rfc-editor.org/info/RFC 8827>.
[RFC 8829] Uberti, J., Jennings, C., and E. Rescorla, Ed.,
"JavaScript Session Establishment Protocol (JSEP)",
RFC 8829, DOI 10.17487/RFC 8829, January 2021,
<https://www.rfc-editor.org/info/RFC 8829>.
[RFC 8832] Jesup, R., Loreto, S., and M. Tüxen, "WebRTC Data Channel
Establishment Protocol", RFC 8832, DOI 10.17487/RFC 8832,
January 2021, <https://www.rfc-editor.org/info/RFC 8832>.
[RFC 8841] 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/RFC 8841, January 2021,
<https://www.rfc-editor.org/info/RFC 8841>.
9.2. Informative References
[RFC 1122] Braden, R., Ed., "Requirements for Internet Hosts -
Communication Layers", STD 3, RFC 1122,
DOI 10.17487/RFC 1122, October 1989,
<https://www.rfc-editor.org/info/RFC 1122>.
[RFC 4347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer
Security", RFC 4347, DOI 10.17487/RFC 4347, April 2006,
<https://www.rfc-editor.org/info/RFC 4347>.
[RFC 5389] Rosenberg, J., Mahy, R., Matthews, P., and D. Wing,
"Session Traversal Utilities for NAT (STUN)", RFC 5389,
DOI 10.17487/RFC 5389, October 2008,
<https://www.rfc-editor.org/info/RFC 5389>.
[RFC 5764] 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/RFC 5764, May 2010,
<https://www.rfc-editor.org/info/RFC 5764>.
[RFC 6083] Tuexen, M., Seggelmann, R., and E. Rescorla, "Datagram
Transport Layer Security (DTLS) for Stream Control
Transmission Protocol (SCTP)", RFC 6083,
DOI 10.17487/RFC 6083, January 2011,
<https://www.rfc-editor.org/info/RFC 6083>.
[RFC 6347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer
Security Version 1.2", RFC 6347, DOI 10.17487/RFC 6347,
January 2012, <https://www.rfc-editor.org/info/RFC 6347>.
[RFC 6951] 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/RFC 6951, May 2013,
<https://www.rfc-editor.org/info/RFC 6951>.
[TLS-DTLS13]
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,
<https://tools.ietf.org/html/draft-ietf-tls-dtls13-39>.
Acknowledgements
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
Mozilla
United States of America
Email: randell-ietf@jesup.org
Salvatore Loreto
Ericsson
Hirsalantie 11
FI-02420 Jorvas
Finland
Email: salvatore.loreto@ericsson.com
Michael Tüxen
Münster University of Applied Sciences
Stegerwaldstrasse 39
48565 Steinfurt
Germany
Email: tuexen@fh-muenster.de
RFC TOTAL SIZE: 35623 bytes
PUBLICATION DATE: Monday, January 18th, 2021
LEGAL RIGHTS: The IETF Trust (see BCP 78)
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