draft-ietf-rtcweb-data-channel-13.txt   rfc8831.txt 
Network Working Group R. Jesup Internet Engineering Task Force (IETF) R. Jesup
Internet-Draft Mozilla Request for Comments: 8831 Mozilla
Intended status: Standards Track S. Loreto Category: Standards Track S. Loreto
Expires: July 8, 2015 Ericsson ISSN: 2070-1721 Ericsson
M. Tuexen M. Tüxen
Muenster Univ. of Appl. Sciences Münster Univ. of Appl. Sciences
January 4, 2015 January 2021
WebRTC Data Channels WebRTC Data Channels
draft-ietf-rtcweb-data-channel-13.txt
Abstract Abstract
The WebRTC framework specifies protocol support for direct The WebRTC framework specifies protocol support for direct,
interactive rich communication using audio, video, and data between interactive, rich communication using audio, video, and data between
two peers' web-browsers. This document specifies the non-media data two peers' web browsers. This document specifies the non-media data
transport aspects of the WebRTC framework. It provides an transport aspects of the WebRTC framework. It provides an
architectural overview of how the Stream Control Transmission architectural overview of how the Stream Control Transmission
Protocol (SCTP) is used in the WebRTC context as a generic transport Protocol (SCTP) is used in the WebRTC context as a generic transport
service allowing WEB-browsers to exchange generic data from peer to service that allows web browsers to exchange generic data from peer
peer. to peer.
Status of This Memo Status of This Memo
This Internet-Draft is submitted in full conformance with the This is an Internet Standards Track document.
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
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working documents as Internet-Drafts. The list of current Internet-
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Internet-Drafts are draft documents valid for a maximum of six months This document is a product of the Internet Engineering Task Force
and may be updated, replaced, or obsoleted by other documents at any (IETF). It represents the consensus of the IETF community. It has
time. It is inappropriate to use Internet-Drafts as reference received public review and has been approved for publication by the
material or to cite them other than as "work in progress." Internet Engineering Steering Group (IESG). Further information on
Internet Standards is available in Section 2 of RFC 7841.
This Internet-Draft will expire on July 8, 2015. 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/rfc8831.
Copyright Notice Copyright Notice
Copyright (c) 2015 IETF Trust and the persons identified as the Copyright (c) 2021 IETF Trust and the persons identified as the
document authors. All rights reserved. document authors. All rights reserved.
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described in the Simplified BSD License. described in the Simplified BSD License.
Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2 1. Introduction
2. Conventions . . . . . . . . . . . . . . . . . . . . . . . . . 3 2. Conventions
3. Use Cases . . . . . . . . . . . . . . . . . . . . . . . . . . 3 3. Use Cases
3.1. Use Cases for Unreliable Data Channels . . . . . . . . . 4 3.1. Use Cases for Unreliable Data Channels
3.2. Use Cases for Reliable Data Channels . . . . . . . . . . 4 3.2. Use Cases for Reliable Data Channels
4. Requirements . . . . . . . . . . . . . . . . . . . . . . . . 4 4. Requirements
5. SCTP over DTLS over UDP Considerations . . . . . . . . . . . 6 5. SCTP over DTLS over UDP Considerations
6. The Usage of SCTP for Data Channels . . . . . . . . . . . . . 8 6. The Usage of SCTP for Data Channels
6.1. SCTP Protocol Considerations . . . . . . . . . . . . . . 8 6.1. SCTP Protocol Considerations
6.2. SCTP Association Management . . . . . . . . . . . . . . . 9 6.2. SCTP Association Management
6.3. SCTP Streams . . . . . . . . . . . . . . . . . . . . . . 9 6.3. SCTP Streams
6.4. Data Channel Definition . . . . . . . . . . . . . . . . . 10 6.4. Data Channel Definition
6.5. Opening a Data Channel . . . . . . . . . . . . . . . . . 10 6.5. Opening a Data Channel
6.6. Transferring User Data on a Data Channel . . . . . . . . 11 6.6. Transferring User Data on a Data Channel
6.7. Closing a Data Channel . . . . . . . . . . . . . . . . . 12 6.7. Closing a Data Channel
7. Security Considerations . . . . . . . . . . . . . . . . . . . 13 7. Security Considerations
8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 13 8. IANA Considerations
9. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 14 9. References
10. References . . . . . . . . . . . . . . . . . . . . . . . . . 14 9.1. Normative References
10.1. Normative References . . . . . . . . . . . . . . . . . . 14 9.2. Informative References
10.2. Informative References . . . . . . . . . . . . . . . . . 15 Acknowledgements
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 16 Authors' Addresses
1. Introduction 1. Introduction
In the WebRTC framework, communication between the parties consists In the WebRTC framework, communication between the parties consists
of media (for example audio and video) and non-media data. Media is of media (for example, audio and video) and non-media data. Media is
sent using SRTP, and is not specified further here. Non-media data sent using the Secure Real-time Transport Protocol (SRTP) and is not
is handled by using SCTP [RFC4960] encapsulated in DTLS. DTLS 1.0 is specified further here. Non-media data is handled by using the
defined in [RFC4347] and the present latest version, DTLS 1.2, is Stream Control Transmission Protocol (SCTP) [RFC4960] encapsulated in
defined in [RFC6347]. 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 | | SCTP |
+----------+ +----------+
| DTLS | | DTLS |
+----------+ +----------+
| ICE/UDP | | ICE/UDP |
+----------+ +----------+
Figure 1: Basic stack diagram Figure 1: Basic Stack Diagram
The encapsulation of SCTP over DTLS (see The encapsulation of SCTP over DTLS (see [RFC8261]) over ICE/UDP (see
[I-D.ietf-tsvwg-sctp-dtls-encaps]) over ICE/UDP (see [RFC5245]) [RFC8445]) provides a NAT traversal solution together with
provides a NAT traversal solution together with confidentiality, confidentiality, source authentication, and integrity-protected
source authentication, and integrity protected transfers. This data transfers. This data transport service operates in parallel to the
transport service operates in parallel to the SRTP media transports, SRTP media transports, and all of them can eventually share a single
and all of them can eventually share a single UDP port number. UDP port number.
SCTP as specified in [RFC4960] with the partial reliability extension SCTP, as specified in [RFC4960] with the partial reliability
defined in [RFC3758] and the additional policies defined in extension (PR-SCTP) defined in [RFC3758] and the additional policies
[I-D.ietf-tsvwg-sctp-prpolicies] provides multiple streams natively defined in [RFC7496], provides multiple streams natively with
with reliable, and the relevant partially-reliable delivery modes for reliable, and the relevant partially reliable, delivery modes for
user messages. Using the reconfiguration extension defined in user messages. Using the reconfiguration extension defined in
[RFC6525] allows to increase the number of streams during the [RFC6525] allows an increase in the number of streams during the
lifetime of an SCTP association and to reset individual SCTP streams. lifetime of an SCTP association and allows individual SCTP streams to
Using [I-D.ietf-tsvwg-sctp-ndata] allows to interleave large messages be reset. Using [RFC8260] allows the interleave of large messages to
to avoid the monopolization and adds the support of prioritizing of avoid monopolization and adds support for prioritizing SCTP streams.
SCTP streams.
The remainder of this document is organized as follows: Section 3 and The remainder of this document is organized as follows: Sections 3
Section 4 provide use cases and requirements for both unreliable and and 4 provide use cases and requirements for both unreliable and
reliable peer to peer data channels; Section 5 discusses SCTP over reliable peer-to-peer data channels; Section 5 discusses SCTP over
DTLS over UDP; Section 6 provides the specification of how SCTP DTLS over UDP; and Section 6 specifies how SCTP should be used by the
should be used by the WebRTC protocol framework for transporting non- WebRTC protocol framework for transporting non-media data between web
media data between WEB-browsers. browsers.
2. Conventions 2. Conventions
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
document are to be interpreted as described in [RFC2119]. "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 3. Use Cases
This section defines use cases specific to data channels. Please This section defines use cases specific to data channels. Please
note that this section is informational only. note that this section is informational only.
3.1. Use Cases for Unreliable Data Channels 3.1. Use Cases for Unreliable Data Channels
U-C 1: A real-time game where position and object state information U-C 1: A real-time game where position and object state information
is sent via one or more unreliable data channels. Note that are sent via one or more unreliable data channels. Note that
at any time there may be no SRTP media channels, or all SRTP at any time, there may not be any SRTP media channels or all
media channels may be inactive, and that there may also be SRTP media channels may be inactive, and there may also be
reliable data channels in use. reliable data channels in use.
U-C 2: Providing non-critical information to a user about the reason 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 for a state update in a video chat or conference, such as
mute state. mute state.
3.2. Use Cases for Reliable Data Channels 3.2. Use Cases for Reliable Data Channels
U-C 3: A real-time game where critical state information needs to be U-C 3: A real-time game where critical state information needs to be
transferred, such as control information. Such a game may transferred, such as control information. Such a game may
have no SRTP media channels, or they may be inactive at any have no SRTP media channels, or they may be inactive at any
given time, or may only be added due to in-game actions. given time or may only be added due to in-game actions.
U-C 4: Non-realtime file transfers between people chatting. Note U-C 4: Non-real-time file transfers between people chatting. Note
that this may involve a large number of files to transfer that this may involve a large number of files to transfer
sequentially or in parallel, such as when sharing a folder of sequentially or in parallel, such as when sharing a folder of
images or a directory of files. images or a directory of files.
U-C 5: Realtime text chat during an audio and/or video call with an U-C 5: Real-time text chat during an audio and/or video call with an
individual or with multiple people in a conference. individual or with multiple people in a conference.
U-C 6: Renegotiation of the configuration of the PeerConnection. U-C 6: Renegotiation of the configuration of the PeerConnection.
U-C 7: Proxy browsing, where a browser uses data channels of a U-C 7: Proxy browsing, where a browser uses data channels of a
PeerConnection to send and receive HTTP/HTTPS requests and PeerConnection to send and receive HTTP/HTTPS requests and
data, for example to avoid local Internet filtering or data, for example, to avoid local Internet filtering or
monitoring. monitoring.
4. Requirements 4. Requirements
This section lists the requirements for P2P data channels between two This section lists the requirements for Peer-to-Peer (P2P) data
browsers. Please note that this section is informational only. channels between two browsers. Please note that this section is
informational only.
Req. 1: Multiple simultaneous data channels must be supported. Req. 1: Multiple simultaneous data channels must be supported.
Note that there may be 0 or more SRTP media streams in Note that there may be zero or more SRTP media streams in
parallel with the data channels in the same PeerConnection, parallel with the data channels in the same PeerConnection,
and the number and state (active/inactive) of these SRTP and the number and state (active/inactive) of these SRTP
media streams may change at any time. media streams may change at any time.
Req. 2: Both reliable and unreliable data channels must be Req. 2: Both reliable and unreliable data channels must be
supported. supported.
Req. 3: Data channels of a PeerConnection must be congestion Req. 3: Data channels of a PeerConnection must be congestion
controlled; either individually, as a class, or in controlled either individually, as a class, or in
conjunction with the SRTP media streams of the conjunction with the SRTP media streams of the
PeerConnection, to ensure that data channels don't cause PeerConnection. This ensures that data channels don't
congestion problems for these SRTP media streams, and that cause congestion problems for these SRTP media streams, and
the WebRTC PeerConnection does not cause excessive problems that the WebRTC PeerConnection does not cause excessive
when run in parallel with TCP connections. problems when run in parallel with TCP connections.
Req. 4: The application should be able to provide guidance as to Req. 4: The application should be able to provide guidance as to
the relative priority of each data channel relative to each the relative priority of each data channel relative to each
other, and relative to the SRTP media streams. This will other and relative to the SRTP media streams. This will
interact with the congestion control algorithms. interact with the congestion control algorithms.
Req. 5: Data channels must be secured; allowing for Req. 5: Data channels must be secured, which allows for
confidentiality, integrity and source authentication. See confidentiality, integrity, and source authentication. See
[I-D.ietf-rtcweb-security] and [RFC8826] and [RFC8827] for detailed information.
[I-D.ietf-rtcweb-security-arch] for detailed info.
Req. 6: Data channels must provide message fragmentation support Req. 6: Data channels must provide message fragmentation support
such that IP-layer fragmentation can be avoided no matter such that IP-layer fragmentation can be avoided no matter
how large a message the JavaScript application passes to be how large a message the JavaScript application passes to be
sent. It also must ensure that large data channel sent. It also must ensure that large data channel
transfers don't unduly delay traffic on other data transfers don't unduly delay traffic on other data
channels. channels.
Req. 7: The data channel transport protocol must not encode local Req. 7: The data channel transport protocol must not encode local
IP addresses inside its protocol fields; doing so reveals IP addresses inside its protocol fields; doing so reveals
potentially private information, and leads to failure if potentially private information and leads to failure if the
the address is depended upon. address is depended upon.
Req. 8: The data channel transport protocol should support Req. 8: The data channel transport protocol should support
unbounded-length "messages" (i.e., a virtual socket stream) unbounded-length "messages" (i.e., a virtual socket stream)
at the application layer, for such things as image-file- at the application layer for such things as image-file-
transfer; Implementations might enforce a reasonable transfer; implementations might enforce a reasonable
message size limit. message size limit.
Req. 9: The data channel transport protocol should avoid IP Req. 9: The data channel transport protocol should avoid IP
fragmentation. It must support PMTU (Path MTU) discovery fragmentation. It must support Path MTU (PMTU) discovery
and must not rely on ICMP or ICMPv6 being generated or and must not rely on ICMP or ICMPv6 being generated or
being passed back, especially for PMTU discovery. being passed back, especially for PMTU discovery.
Req. 10: It must be possible to implement the protocol stack in the Req. 10: It must be possible to implement the protocol stack in the
user application space. user application space.
5. SCTP over DTLS over UDP Considerations 5. SCTP over DTLS over UDP Considerations
The important features of SCTP in the WebRTC context are: The important features of SCTP in the WebRTC context are the
following:
o Usage of a TCP-friendly congestion control. * Usage of TCP-friendly congestion control.
o The congestion control is modifiable for integration with the SRTP * modifiable congestion control for integration with the SRTP media
media stream congestion control. stream congestion control.
o Support of multiple unidirectional streams, each providing its own * Support of multiple unidirectional streams, each providing its own
notion of ordered message delivery. notion of ordered message delivery.
o Support of ordered and out-of-order message delivery. * Support of ordered and out-of-order message delivery.
o Supporting arbitrary large user messages by providing * Support of arbitrarily large user messages by providing
fragmentation and reassembly. fragmentation and reassembly.
o Support of PMTU-discovery. * Support of PMTU discovery.
o Support of reliable or partially reliable message transport. * Support of reliable or partially reliable message transport.
The WebRTC Data Channel mechanism does not support SCTP multihoming. The WebRTC data channel mechanism does not support SCTP multihoming.
The SCTP layer will simply act as if it were running on a single- 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 homed host, since that is the abstraction that the DTLS layer (a
connection oriented, unreliable datagram service) exposes. connection-oriented, unreliable datagram service) exposes.
The encapsulation of SCTP over DTLS defined in The encapsulation of SCTP over DTLS defined in [RFC8261] provides
[I-D.ietf-tsvwg-sctp-dtls-encaps] provides confidentiality, source confidentiality, source authentication, and integrity-protected
authenticated, and integrity protected transfers. Using DTLS over transfers. Using DTLS over UDP in combination with Interactive
UDP in combination with ICE enables middlebox traversal in IPv4 and Connectivity Establishment (ICE) [RFC8445] enables middlebox
IPv6 based networks. SCTP as specified in [RFC4960] MUST be used in traversal in IPv4- and IPv6-based networks. SCTP as specified in
combination with the extension defined in [RFC3758] and provides the [RFC4960] MUST be used in combination with the extension defined in
following features for transporting non-media data between browsers: [RFC3758] and provides the following features for transporting non-
media data between browsers:
o Support of multiple unidirectional streams. * Support of multiple unidirectional streams.
o Ordered and unordered delivery of user messages. * Ordered and unordered delivery of user messages.
o Reliable and partial-reliable transport of user messages. * Reliable and partially reliable transport of user messages.
Each SCTP user message contains a Payload Protocol Identifier (PPID) Each SCTP user message contains a Payload Protocol Identifier (PPID)
that is passed to SCTP by its upper layer on the sending side and 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 provided to its upper layer on the receiving side. The PPID can be
used to multiplex/demultiplex multiple upper layers over a single used to multiplex/demultiplex multiple upper layers over a single
SCTP association. In the WebRTC context, the PPID is used to SCTP association. In the WebRTC context, the PPID is used to
distinguish between UTF-8 encoded user data, binary encoded userdata distinguish between UTF-8 encoded user data, binary-encoded user
and the Data Channel Establishment Protocol defined in data, and the Data Channel Establishment Protocol (DCEP) defined in
[RFC8832]. Please note that the PPID is not accessible via the
[I-D.ietf-rtcweb-data-protocol]. Please note that the PPID is not JavaScript API.
accessible via the Javascript API.
The encapsulation of SCTP over DTLS, together with the SCTP features The encapsulation of SCTP over DTLS, together with the SCTP features
listed above satisfies all the requirements listed in Section 4. listed above, satisfies all the requirements listed in Section 4.
The layering of protocols for WebRTC is shown in the following The layering of protocols for WebRTC is shown in Figure 2.
Figure 2.
+------+------+------+ +------+------+------+
| DCEP | UTF-8|Binary| | DCEP | UTF-8|Binary|
| | data | data | | | Data | Data |
+------+------+------+ +------+------+------+
| SCTP | | SCTP |
+----------------------------------+ +----------------------------------+
| STUN | SRTP | DTLS | | STUN | SRTP | DTLS |
+----------------------------------+ +----------------------------------+
| ICE | | ICE |
+----------------------------------+ +----------------------------------+
| UDP1 | UDP2 | UDP3 | ... | | UDP1 | UDP2 | UDP3 | ... |
+----------------------------------+ +----------------------------------+
Figure 2: WebRTC protocol layers Figure 2: WebRTC Protocol Layers
This stack (especially in contrast to DTLS over SCTP [RFC6083] in This stack (especially in contrast to DTLS over SCTP [RFC6083] and in
combination with SCTP over UDP [RFC6951]) has been chosen because it combination with SCTP over UDP [RFC6951]) has been chosen for the
following reasons:
o supports the transmission of arbitrary large user messages. * supports the transmission of arbitrarily large user messages;
o shares the DTLS connection with the SRTP media channels of the * shares the DTLS connection with the SRTP media channels of the
PeerConnection. PeerConnection; and
o provides privacy for the SCTP control information. * provides privacy for the SCTP control information.
Considering the protocol stack of Figure 2 the usage of DTLS 1.0 over Referring to the protocol stack shown in Figure 2:
UDP is specified in [RFC4347] and the usage of DTLS 1.2 over UDP in
specified in [RFC6347], while the usage of SCTP on top of DTLS is * the usage of DTLS 1.0 over UDP is specified in [RFC4347];
specified in [I-D.ietf-tsvwg-sctp-dtls-encaps]. Please note that the
demultiplexing STUN vs. SRTP vs. DTLS is done as described in * the usage of DTLS 1.2 over UDP in specified in [RFC6347];
Section 5.1.2 of [RFC5764] and SCTP is the only payload of DTLS.
* 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 Since DTLS is typically implemented in user application space, the
SCTP stack also needs to be a user application space stack. SCTP stack also needs to be a user application space stack.
The ICE/UDP layer can handle IP address changes during a session The ICE/UDP layer can handle IP address changes during a session
without needing interaction with the DTLS and SCTP layers. However, without needing interaction with the DTLS and SCTP layers. However,
SCTP SHOULD be notified when an address changes has happened. In SCTP SHOULD be notified when an address change has happened. In this
this case SCTP SHOULD retest the Path MTU and reset the congestion case, SCTP SHOULD retest the Path MTU and reset the congestion state
state to the initial state. In case of a window based congestion to the initial state. In the case of window-based congestion control
control like the one specified in [RFC4960], this means setting the like the one specified in [RFC4960], this means setting the
congestion window and slow start threshold to its initial values. congestion window and slow-start threshold to its initial values.
Incoming ICMP or ICMPv6 messages can't be processed by the SCTP Incoming ICMP or ICMPv6 messages can't be processed by the SCTP
layer, since there is no way to identify the corresponding layer, since there is no way to identify the corresponding
association. Therefore SCTP MUST support performing Path MTU association. Therefore, SCTP MUST support performing Path MTU
discovery without relying on ICMP or ICMPv6 as specified in [RFC4821] discovery without relying on ICMP or ICMPv6 as specified in [RFC4821]
using probing messages specified in [RFC4820]. The initial Path MTU by using probing messages specified in [RFC4820]. The initial Path
at the IP layer SHOULD NOT exceed 1200 bytes for IPv4 and 1280 for MTU at the IP layer SHOULD NOT exceed 1200 bytes for IPv4 and 1280
IPv6. bytes for IPv6.
In general, the lower layer interface of an SCTP implementation In general, the lower-layer interface of an SCTP implementation
should be adapted to address the differences between IPv4 and IPv6 should be adapted to address the differences between IPv4 and IPv6
(being connection-less) or DTLS (being connection-oriented). (being connectionless) or DTLS (being connection oriented).
When the protocol stack of Figure 2 is used, DTLS protects the When the protocol stack shown in Figure 2 is used, DTLS protects the
complete SCTP packet, so it provides confidentiality, integrity and complete SCTP packet, so it provides confidentiality, integrity, and
source authentication of the complete SCTP packet. source authentication of the complete SCTP packet.
SCTP provides congestion control on a per-association base. This SCTP provides congestion control on a per-association basis. This
means that all SCTP streams within a single SCTP association share means that all SCTP streams within a single SCTP association share
the same congestion window. Traffic not being sent over SCTP is not the same congestion window. Traffic not being sent over SCTP is not
covered by the SCTP congestion control. Using a congestion control covered by SCTP congestion control. Using a congestion control
different from than the standard one might improve the impact on the different from the standard one might improve the impact on the
parallel SRTP media streams. parallel SRTP media streams.
SCTP uses the same port number concept as TCP and UDP do. Therefore SCTP uses the same port number concept as TCP and UDP. Therefore, an
an SCTP association uses two port numbers, one at each SCTP end- SCTP association uses two port numbers, one at each SCTP endpoint.
point.
6. The Usage of SCTP for Data Channels 6. The Usage of SCTP for Data Channels
6.1. SCTP Protocol Considerations 6.1. SCTP Protocol Considerations
The DTLS encapsulation of SCTP packets as described in The DTLS encapsulation of SCTP packets as described in [RFC8261] MUST
[I-D.ietf-tsvwg-sctp-dtls-encaps] MUST be used. be used.
This SCTP stack and its upper layer MUST support the usage of This SCTP stack and its upper layer MUST support the usage of
multiple SCTP streams. A user message can be sent ordered or multiple SCTP streams. A user message can be sent ordered or
unordered and with partial or full reliability. unordered and with partial or full reliability.
The following SCTP protocol extensions are required: The following SCTP protocol extensions are required:
o The stream reconfiguration extension defined in [RFC6525] MUST be * The stream reconfiguration extension defined in [RFC6525] MUST be
supported. It is used for closing channels. supported. It is used for closing channels.
o The dynamic address reconfiguration extension defined in [RFC5061] * The dynamic address reconfiguration extension defined in [RFC5061]
MUST be used to signal the support of the stream reset extension MUST be used to signal the support of the stream reset extension
defined in [RFC6525]. Other features of [RFC5061] are OPTIONAL. defined in [RFC6525]. Other features of [RFC5061] are OPTIONAL.
o The partial reliability extension defined in [RFC3758] MUST be * The partial reliability extension defined in [RFC3758] MUST be
supported. In addition to the timed reliability PR-SCTP policy supported. In addition to the timed reliability PR-SCTP policy
defined in [RFC3758], the limited retransmission policy defined in defined in [RFC3758], the limited retransmission policy defined in
[I-D.ietf-tsvwg-sctp-prpolicies] MUST be supported. Limiting the [RFC7496] MUST be supported. Limiting the number of
number of retransmissions to zero combined with unordered delivery retransmissions to zero, combined with unordered delivery,
provides a UDP-like service where each user message is sent provides a UDP-like service where each user message is sent
exactly once and delivered in the order received. exactly once and delivered in the order received.
The support for message interleaving as defined in The support for message interleaving as defined in [RFC8260] SHOULD
[I-D.ietf-tsvwg-sctp-ndata] SHOULD be used. be used.
6.2. SCTP Association Management 6.2. SCTP Association Management
In the WebRTC context, the SCTP association will be set up when the In the WebRTC context, the SCTP association will be set up when the
two endpoints of the WebRTC PeerConnection agree on opening it, as two endpoints of the WebRTC PeerConnection agree on opening it, as
negotiated by JSEP (typically an exchange of SDP) negotiated by the JavaScript Session Establishment Protocol (JSEP),
[I-D.ietf-rtcweb-jsep]. It will use the DTLS connection selected via which is typically an exchange of the Session Description Protocol
ICE; typically this will be shared via BUNDLE or equivalent with DTLS (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. connections used to key the SRTP media streams.
The number of streams negotiated during SCTP association setup SHOULD The number of streams negotiated during SCTP association setup SHOULD
be 65535, which is the maximum number of streams that can be be 65535, which is the maximum number of streams that can be
negotiated during the association setup. negotiated during the association setup.
SCTP supports two ways of terminating an SCTP association. A SCTP supports two ways of terminating an SCTP association. The first
graceful one, using a procedure which ensures that no messages are method is a graceful one, where a procedure that ensures no messages
lost during the shutdown of the association. The second method is a are lost during the shutdown of the association is used. The second
non-graceful one, where one side can just abort the association. method is a non-graceful one, where one side can just abort the
association.
Each SCTP end-point supervises continuously the reachability of its Each SCTP endpoint continuously supervises the reachability of its
peer by monitoring the number of retransmissions of user messages and peer by monitoring the number of retransmissions of user messages and
test messages. In case of excessive retransmissions, the association test messages. In case of excessive retransmissions, the association
is terminated in a non-graceful way. is terminated in a non-graceful way.
If an SCTP association is closed in a graceful way, all of its data 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 closed. In case of a non-graceful teardown, all data
channels are also closed, but an error indication SHOULD be provided channels are also closed, but an error indication SHOULD be provided
if possible. if possible.
6.3. SCTP Streams 6.3. SCTP Streams
skipping to change at page 10, line 23 skipping to change at line 447
to identify the meaning of the data channel. to identify the meaning of the data channel.
The realization of a data channel is a pair of one incoming stream The realization of a data channel is a pair of one incoming stream
and one outgoing SCTP stream having the same SCTP stream identifier. and one outgoing SCTP stream having the same SCTP stream identifier.
How these SCTP stream identifiers are selected is protocol and How these SCTP stream identifiers are selected is protocol and
implementation dependent. This allows a bidirectional communication. implementation dependent. This allows a bidirectional communication.
Additionally, each data channel has the following properties in each Additionally, each data channel has the following properties in each
direction: direction:
o reliable or unreliable message transmission. In case of * reliable or unreliable message transmission: In case of unreliable
unreliable transmissions, the same level of unreliability is used. transmissions, the same level of unreliability is used. Note
Please note that in SCTP this is a property of an SCTP user that, in SCTP, this is a property of an SCTP user message and not
message and not of an SCTP stream. of an SCTP stream.
o in-order or out-of-order message delivery for message sent. * in-order or out-of-order message delivery for message sent: Note
Please note that in SCTP this is a property of an SCTP user that, in SCTP, this is a property of an SCTP user message and not
message and not of an SCTP stream. of an SCTP stream.
o A priority, which is a 2 byte unsigned integer. These priorities * a priority, which is a 2-byte unsigned integer: These priorities
MUST be interpreted as weighted-fair-queuing scheduling priorities MUST be interpreted as weighted-fair-queuing scheduling priorities
per the definition of the corresponding stream scheduler per the definition of the corresponding stream scheduler
supporting interleaving in [I-D.ietf-tsvwg-sctp-ndata]. For use supporting interleaving in [RFC8260]. For use in WebRTC, the
in WebRTC, the values used SHOULD be one of 128 ("below normal"), values used SHOULD be one of 128 ("below normal"), 256 ("normal"),
256 ("normal"), 512 ("high") or 1024 ("extra high"). 512 ("high"), or 1024 ("extra high").
o an optional label. * an optional label.
o an optional protocol. * an optional protocol.
Please note that for a data channel being negotiated with the Note that for a data channel being negotiated with the protocol
protocol specified in [I-D.ietf-rtcweb-data-protocol] all of the specified in [RFC8832], all of the above properties are the same in
above properties are the same in both directions. both directions.
6.5. Opening a Data Channel 6.5. Opening a Data Channel
Data channels can be opened by using negotiation within the SCTP Data channels can be opened by using negotiation within the SCTP
association, called in-band negotiation, or out-of-band negotiation. association (called in-band negotiation) or out-of-band negotiation.
Out-of-band negotiation is defined as any method which results in an 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. agreement as to the parameters of a channel and the creation thereof.
The details are out of scope of this document. Applications using The details are out of scope of this document. Applications using
data channels need to use the negotiation methods consistently on data channels need to use the negotiation methods consistently on
both end-points. both endpoints.
A simple protocol for in-band negotiation is specified in A simple protocol for in-band negotiation is specified in [RFC8832].
[I-D.ietf-rtcweb-data-protocol].
When one side wants to open a channel using out-of-band negotiation, When one side wants to open a channel using out-of-band negotiation,
it picks a stream. Unless otherwise defined or negotiated, the it picks a stream. Unless otherwise defined or negotiated, the
streams are picked based on the DTLS role (the client picks even streams are picked based on the DTLS role (the client picks even
stream identifiers, the server odd stream identifiers). However, the stream identifiers, and the server picks odd stream identifiers).
application is responsible for avoiding collisions with existing However, the application is responsible for avoiding collisions with
streams. If it attempts to re-use a stream which is part of an existing streams. If it attempts to reuse a stream that is part of
existing data channel, the addition MUST fail. In addition to an existing data channel, the addition MUST fail. In addition to
choosing a stream, the application SHOULD also determine the options choosing a stream, the application SHOULD also determine the options
to use for sending messages. The application MUST ensure in an to be used for sending messages. The application MUST ensure in an
application-specific manner that the application at the peer will application-specific manner that the application at the peer will
also know the selected stream to be used, and the options for sending also know the selected stream to be used, as well as the options for
data from that side. sending data from that side.
6.6. Transferring User Data on a Data Channel 6.6. Transferring User Data on a Data Channel
All data sent on a data channel in both directions MUST be sent over All data sent on a data channel in both directions MUST be sent over
the underlying stream using the reliability defined when the data the underlying stream using the reliability defined when the data
channel was opened unless the options are changed, or per-message channel was opened, unless the options are changed or per-message
options are specified by a higher level. options are specified by a higher level.
The message-orientation of SCTP is used to preserve the message The message orientation of SCTP is used to preserve the message
boundaries of user messages. Therefore, senders MUST NOT put more boundaries of user messages. Therefore, senders MUST NOT put more
than one application message into an SCTP user message. Unless the than one application message into an SCTP user message. Unless the
deprecated PPID-based fragmentation and reassembly is used, the deprecated PPID-based fragmentation and reassembly is used, the
sender MUST include exactly one application message in each SCTP user sender MUST include exactly one application message in each SCTP user
message. message.
The SCTP Payload Protocol Identifiers (PPIDs) are used to signal the The SCTP Payload Protocol Identifiers (PPIDs) are used to signal the
interpretation of the "Payload data". The following PPIDs MUST be interpretation of the "payload data". The following PPIDs MUST be
used (see Section 8): used (see Section 8):
WebRTC String: to identify a non-empty JavaScript string encoded in WebRTC String: to identify a non-empty JavaScript string encoded in
UTF-8. UTF-8.
WebRTC String Empty: to identify an empty JavaScript string encoded WebRTC String Empty: to identify an empty JavaScript string encoded
in UTF-8. in UTF-8.
WebRTC Binary: to identify a non-empty JavaScript binary data WebRTC Binary: to identify non-empty JavaScript binary data
(ArrayBuffer, ArrayBufferView or Blob). (ArrayBuffer, ArrayBufferView, or Blob).
WebRTC Binary Empty: to identify an empty JavaScript binary data WebRTC Binary Empty: to identify empty JavaScript binary data
(ArrayBuffer, ArrayBufferView or Blob). (ArrayBuffer, ArrayBufferView, or Blob).
SCTP does not support the sending of empty user messages. Therefore, SCTP does not support the sending of empty user messages. Therefore,
if an empty message has to be sent, the appropriate PPID (WebRTC if an empty message has to be sent, the appropriate PPID (WebRTC
String Empty or WebRTC Binary Empty) is used and the SCTP user 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 of one zero byte is sent. When receiving an SCTP user
message with one of these PPIDs, the receiver MUST ignore the SCTP message with one of these PPIDs, the receiver MUST ignore the SCTP
user message and process it as an empty message. user message and process it as an empty message.
The usage of the PPIDs "WebRTC String Partial" and "WebRTC Binary The usage of the PPIDs "WebRTC String Partial" and "WebRTC Binary
Partial" is deprecated. They were used for a PPID-based Partial" is deprecated. They were used for a PPID-based
fragmentation and reassembly of user messages belonging to reliable fragmentation and reassembly of user messages belonging to reliable
and ordered data channels. and ordered data channels.
If a message with an unsupported PPID is received or some error If a message with an unsupported PPID is received or some error
condition related to the received message is detected by the receiver condition related to the received message is detected by the receiver
(for example, illegal ordering), the receiver SHOULD close the (for example, illegal ordering), the receiver SHOULD close the
corresponding data channel. This implies in particular that corresponding data channel. This implies in particular that
extensions using additional PPIDs can't be used without prior extensions using additional PPIDs can't be used without prior
negotiation. negotiation.
The SCTP base protocol specified in [RFC4960] does not support the The SCTP base protocol specified in [RFC4960] does not support the
interleaving of user messages. Therefore sending a large user interleaving of user messages. Therefore, sending a large user
message can monopolize the SCTP association. To overcome this message can monopolize the SCTP association. To overcome this
limitation, [I-D.ietf-tsvwg-sctp-ndata] defines an extension to limitation, [RFC8260] defines an extension to support message
support message interleaving, which SHOULD be used. As long as interleaving, which SHOULD be used. As long as message interleaving
message interleaving is not supported, the sender SHOULD limit the is not supported, the sender SHOULD limit the maximum message size to
maximum message size to 16 KB to avoid monopolization. 16 KB to avoid monopolization.
It is recommended that the message size be kept within certain size It is recommended that the message size be kept within certain size
bounds as applications will not be able to support arbitrarily-large bounds, as applications will not be able to support arbitrarily large
single messages. This limit has to be negotiated, for example by single messages. This limit has to be negotiated, for example, by
using [I-D.ietf-mmusic-sctp-sdp]. using [RFC8841].
The sender SHOULD disable the Nagle algorithm (see [RFC1122]) to The sender SHOULD disable the Nagle algorithm (see [RFC1122]) to
minimize the latency. minimize the latency.
6.7. Closing a Data Channel 6.7. Closing a Data Channel
Closing of a data channel MUST be signaled by resetting the Closing of a data channel MUST be signaled by resetting the
corresponding outgoing streams [RFC6525]. This means that if one corresponding outgoing streams [RFC6525]. This means that if one
side decides to close the data channel, it resets the corresponding side decides to close the data channel, it resets the corresponding
outgoing stream. When the peer sees that an incoming stream was outgoing stream. When the peer sees that an incoming stream was
skipping to change at page 13, line 13 skipping to change at line 580
a corresponding notification to the application layer that the reset a corresponding notification to the application layer that the reset
has been performed. Streams are available for reuse after a reset has been performed. Streams are available for reuse after a reset
has been performed. has been performed.
[RFC6525] also guarantees that all the messages are delivered (or [RFC6525] also guarantees that all the messages are delivered (or
abandoned) before the stream is reset. abandoned) before the stream is reset.
7. Security Considerations 7. Security Considerations
This document does not add any additional considerations to the ones This document does not add any additional considerations to the ones
given in [I-D.ietf-rtcweb-security] and given in [RFC8826] and [RFC8827].
[I-D.ietf-rtcweb-security-arch].
It should be noted that a receiver must be prepared that the sender It should be noted that a receiver must be prepared for a sender that
tries to send arbitrary large messages. tries to send arbitrarily large messages.
8. IANA Considerations 8. IANA Considerations
[NOTE to RFC-Editor:
"RFCXXXX" is to be replaced by the RFC number you assign this
document.
]
This document uses six already registered SCTP Payload Protocol This document uses six already registered SCTP Payload Protocol
Identifiers (PPIDs): "DOMString Last", "Binary Data Partial", "Binary Identifiers (PPIDs): "DOMString Last", "Binary Data Partial", "Binary
Data Last", "DOMString Partial", "WebRTC String Empty", and "WebRTC Data Last", "DOMString Partial", "WebRTC String Empty", and "WebRTC
Binary Empty". [RFC4960] creates the registry "SCTP Payload Protocol Binary Empty". [RFC4960] creates the "SCTP Payload Protocol
Identifiers" from which these identifiers were assigned. IANA is Identifiers" registry from which these identifiers were assigned.
requested to update the reference of these six assignments to point IANA has updated the reference of these six assignments to point to
to this document and change the names of the first four PPIDs. The this document and changed the names of the first four PPIDs. The
corresponding dates should be kept. corresponding dates remain unchanged.
Therefore these six assignments should be updated to read:
+-------------------------------+----------+-----------+------------+ The six assignments have been updated to read:
| Value | SCTP | Reference | Date |
| | PPID | | |
+-------------------------------+----------+-----------+------------+
| WebRTC String | 51 | [RFCXXXX] | 2013-09-20 |
| WebRTC Binary Partial | 52 | [RFCXXXX] | 2013-09-20 |
| (Deprecated) | | | |
| WebRTC Binary | 53 | [RFCXXXX] | 2013-09-20 |
| WebRTC String Partial | 54 | [RFCXXXX] | 2013-09-20 |
| (Deprecated) | | | |
| WebRTC String Empty | 56 | [RFCXXXX] | 2014-08-22 |
| WebRTC Binary Empty | 57 | [RFCXXXX] | 2014-08-22 |
+-------------------------------+----------+-----------+------------+
9. Acknowledgments +======================+===========+===========+============+
| 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 |
+----------------------+-----------+-----------+------------+
Many thanks for comments, ideas, and text from Harald Alvestrand, Table 1
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.
10. References 9. References
10.1. Normative References 9.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997. Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997,
<https://www.rfc-editor.org/info/rfc2119>.
[RFC3758] Stewart, R., Ramalho, M., Xie, Q., Tuexen, M., and P. [RFC3758] Stewart, R., Ramalho, M., Xie, Q., Tuexen, M., and P.
Conrad, "Stream Control Transmission Protocol (SCTP) Conrad, "Stream Control Transmission Protocol (SCTP)
Partial Reliability Extension", RFC 3758, May 2004. Partial Reliability Extension", RFC 3758,
DOI 10.17487/RFC3758, May 2004,
[RFC4347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer <https://www.rfc-editor.org/info/rfc3758>.
Security", RFC 4347, April 2006.
[RFC4820] Tuexen, M., Stewart, R., and P. Lei, "Padding Chunk and [RFC4820] Tuexen, M., Stewart, R., and P. Lei, "Padding Chunk and
Parameter for the Stream Control Transmission Protocol Parameter for the Stream Control Transmission Protocol
(SCTP)", RFC 4820, March 2007. (SCTP)", RFC 4820, DOI 10.17487/RFC4820, March 2007,
<https://www.rfc-editor.org/info/rfc4820>.
[RFC4821] Mathis, M. and J. Heffner, "Packetization Layer Path MTU [RFC4821] Mathis, M. and J. Heffner, "Packetization Layer Path MTU
Discovery", RFC 4821, March 2007. Discovery", RFC 4821, DOI 10.17487/RFC4821, March 2007,
<https://www.rfc-editor.org/info/rfc4821>.
[RFC4960] Stewart, R., "Stream Control Transmission Protocol", RFC [RFC4960] Stewart, R., Ed., "Stream Control Transmission Protocol",
4960, September 2007. RFC 4960, DOI 10.17487/RFC4960, September 2007,
<https://www.rfc-editor.org/info/rfc4960>.
[RFC5061] Stewart, R., Xie, Q., Tuexen, M., Maruyama, S., and M. [RFC5061] Stewart, R., Xie, Q., Tuexen, M., Maruyama, S., and M.
Kozuka, "Stream Control Transmission Protocol (SCTP) Kozuka, "Stream Control Transmission Protocol (SCTP)
Dynamic Address Reconfiguration", RFC 5061, September Dynamic Address Reconfiguration", RFC 5061,
2007. DOI 10.17487/RFC5061, September 2007,
<https://www.rfc-editor.org/info/rfc5061>.
[RFC5245] Rosenberg, J., "Interactive Connectivity Establishment [RFC6525] Stewart, R., Tuexen, M., and P. Lei, "Stream Control
(ICE): A Protocol for Network Address Translator (NAT) Transmission Protocol (SCTP) Stream Reconfiguration",
Traversal for Offer/Answer Protocols", RFC 5245, April RFC 6525, DOI 10.17487/RFC6525, February 2012,
2010. <https://www.rfc-editor.org/info/rfc6525>.
[RFC6347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer [RFC7496] Tuexen, M., Seggelmann, R., Stewart, R., and S. Loreto,
Security Version 1.2", RFC 6347, January 2012. "Additional Policies for the Partially Reliable Stream
Control Transmission Protocol Extension", RFC 7496,
DOI 10.17487/RFC7496, April 2015,
<https://www.rfc-editor.org/info/rfc7496>.
[RFC6525] Stewart, R., Tuexen, M., and P. Lei, "Stream Control [RFC8174] Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC
Transmission Protocol (SCTP) Stream Reconfiguration", RFC 2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174,
6525, February 2012. May 2017, <https://www.rfc-editor.org/info/rfc8174>.
[I-D.ietf-tsvwg-sctp-ndata] [RFC8260] Stewart, R., Tuexen, M., Loreto, S., and R. Seggelmann,
Stewart, R., Tuexen, M., Loreto, S., and R. Seggelmann, "Stream Schedulers and User Message Interleaving for the
"Stream Schedulers and a New Data Chunk for the Stream Stream Control Transmission Protocol", RFC 8260,
Control Transmission Protocol", draft-ietf-tsvwg-sctp- DOI 10.17487/RFC8260, November 2017,
ndata-01 (work in progress), July 2014. <https://www.rfc-editor.org/info/rfc8260>.
[I-D.ietf-rtcweb-data-protocol] [RFC8261] Tuexen, M., Stewart, R., Jesup, R., and S. Loreto,
Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel "Datagram Transport Layer Security (DTLS) Encapsulation of
Establishment Protocol", draft-ietf-rtcweb-data- SCTP Packets", RFC 8261, DOI 10.17487/RFC8261, November
protocol-08 (work in progress), September 2014. 2017, <https://www.rfc-editor.org/info/rfc8261>.
[I-D.ietf-tsvwg-sctp-dtls-encaps] [RFC8445] Keranen, A., Holmberg, C., and J. Rosenberg, "Interactive
Tuexen, M., Stewart, R., Jesup, R., and S. Loreto, "DTLS Connectivity Establishment (ICE): A Protocol for Network
Encapsulation of SCTP Packets", draft-ietf-tsvwg-sctp- Address Translator (NAT) Traversal", RFC 8445,
dtls-encaps-07 (work in progress), December 2014. DOI 10.17487/RFC8445, July 2018,
<https://www.rfc-editor.org/info/rfc8445>.
[I-D.ietf-rtcweb-security] [RFC8826] Rescorla, E., "Security Considerations for WebRTC",
Rescorla, E., "Security Considerations for WebRTC", draft- RFC 8826, DOI 10.17487/RFC8826, January 2021,
ietf-rtcweb-security-07 (work in progress), July 2014. <https://www.rfc-editor.org/info/rfc8826>.
[I-D.ietf-rtcweb-security-arch] [RFC8827] Rescorla, E., "WebRTC Security Architecture", RFC 8827,
Rescorla, E., "WebRTC Security Architecture", draft-ietf- DOI 10.17487/RFC8827, January 2021,
rtcweb-security-arch-10 (work in progress), July 2014. <https://www.rfc-editor.org/info/rfc8827>.
[I-D.ietf-rtcweb-jsep] [RFC8829] Uberti, J., Jennings, C., and E. Rescorla, Ed.,
Uberti, J., Jennings, C., and E. Rescorla, "Javascript "JavaScript Session Establishment Protocol (JSEP)",
Session Establishment Protocol", draft-ietf-rtcweb-jsep-08 RFC 8829, DOI 10.17487/RFC8829, January 2021,
(work in progress), October 2014. <https://www.rfc-editor.org/info/rfc8829>.
[I-D.ietf-tsvwg-sctp-prpolicies] [RFC8832] Jesup, R., Loreto, S., and M. Tüxen, "WebRTC Data Channel
Tuexen, M., Seggelmann, R., Stewart, R., and S. Loreto, Establishment Protocol", RFC 8832, DOI 10.17487/RFC8832,
"Additional Policies for the Partial Reliability Extension January 2021, <https://www.rfc-editor.org/info/rfc8832>.
of the Stream Control Transmission Protocol", draft-ietf-
tsvwg-sctp-prpolicies-06 (work in progress), December
2014.
[I-D.ietf-mmusic-sctp-sdp] [RFC8841] Holmberg, C., Shpount, R., Loreto, S., and G. Camarillo,
Holmberg, C., Loreto, S., and G. Camarillo, "Stream "Session Description Protocol (SDP) Offer/Answer
Control Transmission Protocol (SCTP)-Based Media Transport Procedures for Stream Control Transmission Protocol (SCTP)
in the Session Description Protocol (SDP)", draft-ietf- over Datagram Transport Layer Security (DTLS) Transport",
mmusic-sctp-sdp-11 (work in progress), December 2014. RFC 8841, DOI 10.17487/RFC8841, January 2021,
<https://www.rfc-editor.org/info/rfc8841>.
10.2. Informative References 9.2. Informative References
[RFC1122] Braden, R., "Requirements for Internet Hosts - [RFC1122] Braden, R., Ed., "Requirements for Internet Hosts -
Communication Layers", STD 3, RFC 1122, October 1989. Communication Layers", STD 3, RFC 1122,
DOI 10.17487/RFC1122, October 1989,
<https://www.rfc-editor.org/info/rfc1122>.
[RFC4347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer
Security", RFC 4347, DOI 10.17487/RFC4347, April 2006,
<https://www.rfc-editor.org/info/rfc4347>.
[RFC5389] Rosenberg, J., Mahy, R., Matthews, P., and D. Wing,
"Session Traversal Utilities for NAT (STUN)", RFC 5389,
DOI 10.17487/RFC5389, October 2008,
<https://www.rfc-editor.org/info/rfc5389>.
[RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer [RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer
Security (DTLS) Extension to Establish Keys for the Secure Security (DTLS) Extension to Establish Keys for the Secure
Real-time Transport Protocol (SRTP)", RFC 5764, May 2010. Real-time Transport Protocol (SRTP)", RFC 5764,
DOI 10.17487/RFC5764, May 2010,
<https://www.rfc-editor.org/info/rfc5764>.
[RFC6083] Tuexen, M., Seggelmann, R., and E. Rescorla, "Datagram [RFC6083] Tuexen, M., Seggelmann, R., and E. Rescorla, "Datagram
Transport Layer Security (DTLS) for Stream Control Transport Layer Security (DTLS) for Stream Control
Transmission Protocol (SCTP)", RFC 6083, January 2011. Transmission Protocol (SCTP)", RFC 6083,
DOI 10.17487/RFC6083, January 2011,
<https://www.rfc-editor.org/info/rfc6083>.
[RFC6347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer
Security Version 1.2", RFC 6347, DOI 10.17487/RFC6347,
January 2012, <https://www.rfc-editor.org/info/rfc6347>.
[RFC6951] Tuexen, M. and R. Stewart, "UDP Encapsulation of Stream [RFC6951] Tuexen, M. and R. Stewart, "UDP Encapsulation of Stream
Control Transmission Protocol (SCTP) Packets for End-Host Control Transmission Protocol (SCTP) Packets for End-Host
to End-Host Communication", RFC 6951, May 2013. to End-Host Communication", RFC 6951,
DOI 10.17487/RFC6951, May 2013,
<https://www.rfc-editor.org/info/rfc6951>.
[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 Authors' Addresses
Randell Jesup Randell Jesup
Mozilla Mozilla
US United States of America
Email: randell-ietf@jesup.org Email: randell-ietf@jesup.org
Salvatore Loreto Salvatore Loreto
Ericsson Ericsson
Hirsalantie 11 Hirsalantie 11
Jorvas 02420 FI-02420 Jorvas
FI Finland
Email: salvatore.loreto@ericsson.com Email: salvatore.loreto@ericsson.com
Michael Tuexen Michael Tüxen
Muenster University of Applied Sciences Münster University of Applied Sciences
Stegerwaldstrasse 39 Stegerwaldstrasse 39
Steinfurt 48565 48565 Steinfurt
DE Germany
Email: tuexen@fh-muenster.de Email: tuexen@fh-muenster.de
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