draft-ietf-rtcweb-data-channel-10.txt   draft-ietf-rtcweb-data-channel-11.txt 
Network Working Group R. Jesup Network Working Group R. Jesup
Internet-Draft Mozilla Internet-Draft Mozilla
Intended status: Standards Track S. Loreto Intended status: Standards Track S. Loreto
Expires: December 11, 2014 Ericsson Expires: January 5, 2015 Ericsson
M. Tuexen M. Tuexen
Muenster Univ. of Appl. Sciences Muenster Univ. of Appl. Sciences
June 9, 2014 July 4, 2014
WebRTC Data Channels WebRTC Data Channels
draft-ietf-rtcweb-data-channel-10.txt draft-ietf-rtcweb-data-channel-11.txt
Abstract Abstract
The Real-Time Communication in WEB-browsers working group is charged The WebRTC framework specifies protocol support for direct
to provide protocol support for direct interactive rich communication interactive rich communication using audio, video, and data between
using audio, video, and data between two peers' web-browsers. This two peers' web-browsers. This document specifies the non-media data
document specifies the non-SRTP media data transport aspects of the transport aspects of the WebRTC framework. It provides an
WebRTC framework. It provides an architectural overview of how the architectural overview of how the Stream Control Transmission
Stream Control Transmission Protocol (SCTP) is used in the WebRTC Protocol (SCTP) is used in the WebRTC context as a generic transport
context as a generic transport service allowing WEB-browsers to service allowing WEB-browsers to exchange generic data from peer to
exchange generic data from peer to peer. peer.
Status of This Memo Status of This Memo
This Internet-Draft is submitted in full conformance with the This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79. provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet- working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/. Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress." material or to cite them other than as "work in progress."
This Internet-Draft will expire on December 11, 2014. This Internet-Draft will expire on January 5, 2015.
Copyright Notice Copyright Notice
Copyright (c) 2014 IETF Trust and the persons identified as the Copyright (c) 2014 IETF Trust and the persons identified as the
document authors. All rights reserved. document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of (http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents publication of this document. Please review these documents
skipping to change at page 2, line 24 skipping to change at page 2, line 24
2. Conventions . . . . . . . . . . . . . . . . . . . . . . . . . 3 2. Conventions . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Use Cases . . . . . . . . . . . . . . . . . . . . . . . . . . 3 3. Use Cases . . . . . . . . . . . . . . . . . . . . . . . . . . 3
3.1. Use Cases for Unreliable Data Channels . . . . . . . . . 3 3.1. Use Cases for Unreliable Data Channels . . . . . . . . . 3
3.2. Use Cases for Reliable Data Channels . . . . . . . . . . 4 3.2. Use Cases for Reliable Data Channels . . . . . . . . . . 4
4. Requirements . . . . . . . . . . . . . . . . . . . . . . . . 4 4. Requirements . . . . . . . . . . . . . . . . . . . . . . . . 4
5. SCTP over DTLS over UDP Considerations . . . . . . . . . . . 5 5. SCTP over DTLS over UDP Considerations . . . . . . . . . . . 5
6. The Usage of SCTP for Data Channels . . . . . . . . . . . . . 8 6. The Usage of SCTP for Data Channels . . . . . . . . . . . . . 8
6.1. SCTP Protocol Considerations . . . . . . . . . . . . . . 8 6.1. SCTP Protocol Considerations . . . . . . . . . . . . . . 8
6.2. Association Setup . . . . . . . . . . . . . . . . . . . . 9 6.2. Association Setup . . . . . . . . . . . . . . . . . . . . 9
6.3. SCTP Streams . . . . . . . . . . . . . . . . . . . . . . 9 6.3. SCTP Streams . . . . . . . . . . . . . . . . . . . . . . 9
6.4. Channel Definition . . . . . . . . . . . . . . . . . . . 9 6.4. WebRTC Data Channel Definition . . . . . . . . . . . . . 9
6.5. Opening a Channel . . . . . . . . . . . . . . . . . . . . 10 6.5. Opening a Channel . . . . . . . . . . . . . . . . . . . . 10
6.6. Transferring User Data on a Channel . . . . . . . . . . . 10 6.6. Transferring User Data on a Channel . . . . . . . . . . . 10
6.7. Closing a Channel . . . . . . . . . . . . . . . . . . . . 11 6.7. Closing a Channel . . . . . . . . . . . . . . . . . . . . 11
7. Security Considerations . . . . . . . . . . . . . . . . . . . 11 7. Security Considerations . . . . . . . . . . . . . . . . . . . 11
8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 12 8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 11
9. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 12 9. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 12
10. References . . . . . . . . . . . . . . . . . . . . . . . . . 12 10. References . . . . . . . . . . . . . . . . . . . . . . . . . 12
10.1. Normative References . . . . . . . . . . . . . . . . . . 12 10.1. Normative References . . . . . . . . . . . . . . . . . . 12
10.2. Informative References . . . . . . . . . . . . . . . . . 14 10.2. Informative References . . . . . . . . . . . . . . . . . 14
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 14 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 14
1. Introduction 1. Introduction
Non-SRTP media data types in the context of WebRTC are handled by In the WebRTC framework, communication between the parties consists
using SCTP [RFC4960] encapsulated in DTLS [RFC6347]. 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
is handled by using SCTP [RFC4960] encapsulated in DTLS [RFC4347].
+----------+ +----------+
| SCTP | | SCTP |
+----------+ +----------+
| DTLS | | DTLS |
+----------+ +----------+
| ICE/UDP | | ICE/UDP |
+----------+ +----------+
Figure 1: Basic stack diagram Figure 1: Basic stack diagram
skipping to change at page 3, line 29 skipping to change at page 3, line 29
lifetime of an SCTP association and to reset individual SCTP streams. lifetime of an SCTP association and to reset individual SCTP streams.
Using [I-D.ietf-tsvwg-sctp-ndata] allows to interleave large messages Using [I-D.ietf-tsvwg-sctp-ndata] allows to interleave large messages
to avoid the monopolization and adds the support of prioritizing of to avoid the monopolization and adds the support of prioritizing of
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: Section 3 and
Section 4 provide use cases and requirements for both unreliable and Section 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; Section 6 provides the specification of how SCTP
should be used by the WebRTC protocol framework for transporting non- should be used by the WebRTC protocol framework for transporting non-
SRTP media data between WEB-browsers. media data between WEB-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", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in [RFC2119]. document are to be interpreted as described in [RFC2119].
3. Use Cases 3. Use Cases
This section defines use cases specific to data channels. For This section defines use cases specific to data channels. For WebRTC
general use cases see [I-D.ietf-rtcweb-use-cases-and-requirements]. use cases see [I-D.ietf-rtcweb-use-cases-and-requirements].
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 is 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 be no SRTP media channels, or all SRTP
media channels may be inactive, and that there may also be media channels may be inactive, and that 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
skipping to change at page 4, line 46 skipping to change at page 4, line 46
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, to ensure that data channels don't cause
congestion problems for these SRTP media streams, and that congestion problems for these SRTP media streams, and that
the WebRTC PeerConnection as a whole is fair with competing the WebRTC PeerConnection does not cause excessive problems
traffic such as TCP. 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; allowing for
confidentiality, integrity and source authentication. See confidentiality, integrity and source authentication. See
[I-D.ietf-rtcweb-security] and [I-D.ietf-rtcweb-security] and
[I-D.ietf-rtcweb-security-arch] for detailed info. [I-D.ietf-rtcweb-security-arch] for detailed info.
skipping to change at page 6, line 9 skipping to change at page 6, line 9
o Support of ordered and out-of-order message delivery. o Support of ordered and out-of-order message delivery.
o Supporting arbitrary large user messages by providing o Supporting arbitrary large user messages by providing
fragmentation and reassembly. fragmentation and reassembly.
o Support of PMTU-discovery. o Support of PMTU-discovery.
o Support of reliable or partially reliable message transport. o Support of reliable or partially reliable message transport.
SCTP multihoming will not be used in WebRTC. The SCTP layer will The WebRTC Data Channel mechanism does not support SCTP multihoming.
simply act as if it were running on a single-homed host, since that The SCTP layer will simply act as if it were running on a single-
is the abstraction that the lower layer (a connection oriented, homed host, since that is the abstraction that the DTLS layer (a
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
[I-D.ietf-tsvwg-sctp-dtls-encaps] provides confidentiality, source [I-D.ietf-tsvwg-sctp-dtls-encaps] provides confidentiality, source
authenticated, and integrity protected transfers. Using DTLS over authenticated, and integrity protected transfers. Using DTLS over
UDP in combination with ICE enables middlebox traversal in IPv4 and UDP in combination with ICE enables middlebox traversal in IPv4 and
IPv6 based networks. SCTP as specified in [RFC4960] MUST be used in IPv6 based networks. SCTP as specified in [RFC4960] MUST be used in
combination with the extension defined in [RFC3758] and provides the combination with the extension defined in [RFC3758] and provides the
following features for transporting non-SRTP media data between following features for transporting non-media data between browsers:
browsers:
o Support of multiple unidirectional streams. o Support of multiple unidirectional streams.
o Ordered and unordered delivery of user messages. o Ordered and unordered delivery of user messages.
o Reliable and partial-reliable transport of user messages. o Reliable and partial-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
skipping to change at page 7, line 31 skipping to change at page 7, line 31
combination with SCTP over UDP [RFC6951]) has been chosen because it combination with SCTP over UDP [RFC6951]) has been chosen because it
o supports the transmission of arbitrary large user messages. o supports the transmission of arbitrary large user messages.
o shares the DTLS connection with the SRTP media channels of the o shares the DTLS connection with the SRTP media channels of the
PeerConnection. PeerConnection.
o provides privacy for the SCTP control information. o provides privacy for the SCTP control information.
Considering the protocol stack of Figure 2 the usage of DTLS over UDP Considering the protocol stack of Figure 2 the usage of DTLS over UDP
is specified in [RFC6347], while the usage of SCTP on top of DTLS is is specified in [RFC4347], while the usage of SCTP on top of DTLS is
specified in [I-D.ietf-tsvwg-sctp-dtls-encaps]. Please note that the specified in [I-D.ietf-tsvwg-sctp-dtls-encaps]. Please note that the
demultiplexing STUN vs. SRTP vs. DTLS is done as described in demultiplexing STUN vs. SRTP vs. DTLS is done as described in
Section 5.1.2 of [RFC5764] and SCTP is the only payload of DTLS. Section 5.1.2 of [RFC5764] and SCTP is the only payload of DTLS.
Since DTLS is typically implemented in user-land, the SCTP stack also Since DTLS is typically implemented in user application space, the
needs to be a user-land stack. SCTP stack also needs to be a user application space stack.
When using DTLS as the lower layer, only single homed SCTP The ICE/UDP layer can handle IP address changes during a session
associations are supported, since DTLS does not expose any address without needing interaction with the DTLS and SCTP layers. However,
management to its upper layer. The ICE/UDP layer can handle IP SCTP SHOULD be notified when an address changes has happened. In
address changes during a session without needing interaction with the this case SCTP SHOULD retest the Path MTU and reset the congestion
DTLS and SCTP layers. However, SCTP SHOULD be notified when an state to the initial state. In case of a window based congestion
address changes has happened. In this case SCTP SHOULD retest the control like the one specified in [RFC4960], this means setting the
Path MTU and reset the congestion state to the initial state. In congestion window and slow start threshold to its initial values.
case of a 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 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 using probing messages specified in [RFC4820]. The initial Path MTU
at the IP layer SHOULD NOT exceed 1200 bytes for IPv4 and 1280 for at the IP layer SHOULD NOT exceed 1200 bytes for IPv4 and 1280 for
IPv6. 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 connection-less) or DTLS (being connection-oriented).
When the protocol stack of Figure 2 is used, DTLS protects the When the protocol stack of 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.
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 partial
reliability extension MUST support policies to limit
o the transmission and retransmission by time.
o the number of retransmissions.
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.
SCTP provides congestion control on a per-association base. This SCTP provides congestion control on a per-association base. 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 the SCTP congestion control. Using a congestion control
different from than the standard one might improve the impact on the different from than the standard one might improve the impact on the
parallel SRTP media streams. parallel SRTP media streams.
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
[I-D.ietf-tsvwg-sctp-dtls-encaps] MUST be used. [I-D.ietf-tsvwg-sctp-dtls-encaps] 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 following SCTP protocol extensions are required:
o The stream reset extension defined in [RFC6525] MUST be supported. o The stream reconfiguration extension defined in [RFC6525] MUST be
It is used for closing channels. supported. It is used for closing channels.
o The dynamic address reconfiguration extension defined in [RFC5061] o 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 not REQUIRED defined in [RFC6525], other features of [RFC5061] are not REQUIRED
to be implemented. to be implemented.
o The partial reliability extension defined in [RFC3758] MUST be o 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. [I-D.ietf-tsvwg-sctp-prpolicies] 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 The support for message interleaving as defined in
[I-D.ietf-tsvwg-sctp-ndata] SHOULD be used. [I-D.ietf-tsvwg-sctp-ndata] SHOULD be used.
6.2. Association Setup 6.2. Association Setup
The SCTP association will be set up when the two endpoints of the In the WebRTC context, the SCTP association will be set up when the
WebRTC PeerConnection agree on opening it, as negotiated by JSEP two endpoints of the WebRTC PeerConnection agree on opening it, as
(typically an exchange of SDP) [I-D.ietf-rtcweb-jsep]. It will use negotiated by JSEP (typically an exchange of SDP)
the DTLS connection selected via ICE; typically this will be shared [I-D.ietf-rtcweb-jsep]. It will use the DTLS connection selected via
via BUNDLE or equivalent with DTLS connections used to key the SRTP ICE; typically this will be shared via BUNDLE or equivalent with DTLS
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 negotiated be 65535, which is the maximum number of streams that can be
during the association setup. negotiated during the association setup.
6.3. SCTP Streams 6.3. SCTP Streams
SCTP defines a stream as a unidirectional logical channel existing SCTP defines a stream as a unidirectional logical channel existing
within an SCTP association to another SCTP endpoint. The streams are within an SCTP association to another SCTP endpoint. The streams are
used to provide the notion of in-sequence delivery and for used to provide the notion of in-sequence delivery and for
multiplexing. Each user message is sent on a particular stream, multiplexing. Each user message is sent on a particular stream,
either ordered or unordered. Ordering is preserved only for ordered either ordered or unordered. Ordering is preserved only for ordered
messages sent on the same stream. messages sent on the same stream.
6.4. Channel Definition 6.4. WebRTC Data Channel Definition
The W3C has consensus on defining the application API for WebRTC The WebRTC Data Channels are bidirectional. They also consider the
DataChannels to be bidirectional. They also consider the notions of notions of in-sequence, out-of-sequence, reliable and unreliable as
in-sequence, out-of-sequence, reliable and unreliable as properties properties of channels. One strong wish is for the application-level
of Channels. One strong wish is for the application-level API to be API to be close to the API for WebSockets, which implies
close to the API for WebSockets, which implies bidirectional streams bidirectional streams of data and waiting for onopen to fire before
of data and waiting for onopen to fire before sending, a textual sending, a textual label used to identify the meaning of the streams.
label used to identify the meaning of the stream, among other things.
The realization of a bidirectional data channel is a pair of one
incoming stream and one outgoing SCTP stream having the same stream
SCTP identifier.
How stream values are selected is protocol and implementation
dependent.
Each data channel also has a priority, which is an 2 byte unsigned Each data channel also has a priority, which is an 2 byte unsigned
integer value. These priorities MUST be interpreted as weighted- integer value. These priorities MUST be interpreted as weighted-
fair-queuing scheduling priorities per the definition of the fair-queuing scheduling priorities per the definition of the
corresponding stream scheduler supporting interleaving in corresponding stream scheduler supporting interleaving in
[I-D.ietf-tsvwg-sctp-ndata]. For use in WebRTC, the values used [I-D.ietf-tsvwg-sctp-ndata]. For use in WebRTC, the values used
SHOULD be one of 128 ("below normal"), 256 ("normal"), 512 ("high") SHOULD be one of 128 ("below normal"), 256 ("normal"), 512 ("high")
or 1024 ("extra high"). or 1024 ("extra high").
The realization of a bidirectional Data Channel is a pair of one
incoming stream and one outgoing SCTP stream having the same stream
SCTP identifier.
How stream values are selected is protocol and implementation
dependent.
6.5. Opening a Channel 6.5. Opening a 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 which 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. The details are out of scope of this document.
A simple protocol for in-band negotiation is specified in A simple protocol for in-band negotiation is specified in
[I-D.ietf-rtcweb-data-protocol]. [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, the server odd stream identifiers). However, the
application is responsible for avoiding collisions with existing application is responsible for avoiding collisions with existing
streams. If it attempts to re-use a stream which is part of an streams. If it attempts to re-use a stream which is part of an
existing Channel, the addition SHOULD fail. In addition to choosing existing data channel, the addition SHOULD fail. In addition to
a stream, the application SHOULD also determine the options to use choosing a stream, the application SHOULD also determine the options
for sending messages. The application MUST ensure in an application- to use for sending messages. The application MUST ensure in an
specific manner that the application at the peer will also know the application-specific manner that the application at the peer will
selected stream to be used, and the options for sending data from also know the selected stream to be used, and the options for sending
that side. data from that side.
6.6. Transferring User Data on a Channel 6.6. Transferring User Data on a Channel
All data sent on a Channel in both directions MUST be sent over the All data sent on a data channel in both directions MUST be sent over
underlying stream using the reliability defined when the Channel was the underlying stream using the reliability defined when the data
opened unless the options are changed, or per-message options are channel was opened unless the options are changed, or per-message
specified by a higher level. options are specified by a higher level.
No more than one message should be put into an SCTP user message. No more than one message should be put into an SCTP user 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". For identifying a JavaScript interpretation of the "Payload data". For identifying a JavaScript
string encoded in UTF-8 the PPID "WebRTC String" MUST be used, for string encoded in UTF-8 the PPID "WebRTC String" MUST be used, for
JavaScript binary data (ArrayBuffer or Blob) the PPID "WebRTC Binary" JavaScript binary data (ArrayBuffer or Blob) the PPID "WebRTC Binary"
MUST be used (see Section 8). MUST be used (see Section 8).
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 is If a message with an unsupported PPID is received or some error is
detected by the receiver (for example, illegal ordering), the detected by the receiver (for example, illegal ordering), the
receiver SHOULD close the corresponding channel. receiver SHOULD close the corresponding data channel.
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, [I-D.ietf-tsvwg-sctp-ndata] defines an extension to
support message interleaving, which SHOULD be used. As long as support message interleaving, which SHOULD be used. As long as
message interleaving is not supported, the sender SHOULD limit the message interleaving is not supported, the sender SHOULD limit the
maximum message size to 16 KB to avoid monopolization. maximum message size to 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 [I-D.ietf-mmusic-sctp-sdp].
The sender SHOULD disable the Nagle algorithm to minimize the The sender SHOULD disable the Nagle algorithm to minimize the
latency. latency.
6.7. Closing a Channel 6.7. Closing a 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 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
reset, it also resets its corresponding outgoing stream. Once this reset, it also resets its corresponding outgoing stream. Once this
is completed, the channel is closed. Resetting a stream sets the is completed, the data channel is closed. Resetting a stream sets
Stream Sequence Numbers (SSNs) of the stream back to 'zero' with a the Stream Sequence Numbers (SSNs) of the stream back to 'zero' with
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 to reuse after a reset has has been performed. Streams are available for reuse after a reset
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 resetting the stream. 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 [I-D.ietf-rtcweb-security] and
[I-D.ietf-rtcweb-security-arch]. [I-D.ietf-rtcweb-security-arch].
I should be noted that a receiver must be prepared that the sender
tries to send arbitrary large messages.
8. IANA Considerations 8. IANA Considerations
[NOTE to RFC-Editor: [NOTE to RFC-Editor:
"RFCXXXX" is to be replaced by the RFC number you assign this "RFCXXXX" is to be replaced by the RFC number you assign this
document. document.
] ]
This document uses four already registered SCTP Payload Protocol This document uses four already registered SCTP Payload Protocol
skipping to change at page 12, line 49 skipping to change at page 12, line 42
10.1. Normative References 10.1. 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, March 1997.
[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, May 2004.
[RFC4347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer
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, March 2007.
[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, March 2007.
[RFC4960] Stewart, R., "Stream Control Transmission Protocol", RFC [RFC4960] Stewart, R., "Stream Control Transmission Protocol", RFC
4960, September 2007. 4960, September 2007.
[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, September
2007. 2007.
[RFC5245] Rosenberg, J., "Interactive Connectivity Establishment [RFC5245] Rosenberg, J., "Interactive Connectivity Establishment
(ICE): A Protocol for Network Address Translator (NAT) (ICE): A Protocol for Network Address Translator (NAT)
Traversal for Offer/Answer Protocols", RFC 5245, April Traversal for Offer/Answer Protocols", RFC 5245, April
2010. 2010.
[RFC6347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer
Security Version 1.2", RFC 6347, January 2012.
[RFC6525] Stewart, R., Tuexen, M., and P. Lei, "Stream Control [RFC6525] Stewart, R., Tuexen, M., and P. Lei, "Stream Control
Transmission Protocol (SCTP) Stream Reconfiguration", RFC Transmission Protocol (SCTP) Stream Reconfiguration", RFC
6525, February 2012. 6525, February 2012.
[I-D.ietf-tsvwg-sctp-ndata] [I-D.ietf-tsvwg-sctp-ndata]
Stewart, R., Tuexen, M., Loreto, S., and R. Seggelmann, "A Stewart, R., Tuexen, M., Loreto, S., and R. Seggelmann, "A
New Data Chunk for Stream Control Transmission Protocol", New Data Chunk for Stream Control Transmission Protocol",
draft-ietf-tsvwg-sctp-ndata-00 (work in progress), draft-ietf-tsvwg-sctp-ndata-00 (work in progress),
February 2014. February 2014.
[I-D.ietf-rtcweb-data-protocol] [I-D.ietf-rtcweb-data-protocol]
Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel
Establishment Protocol", draft-ietf-rtcweb-data- Establishment Protocol", draft-ietf-rtcweb-data-
protocol-05 (work in progress), May 2014. protocol-06 (work in progress), June 2014.
[I-D.ietf-tsvwg-sctp-dtls-encaps] [I-D.ietf-tsvwg-sctp-dtls-encaps]
Tuexen, M., Stewart, R., Jesup, R., and S. Loreto, "DTLS Tuexen, M., Stewart, R., Jesup, R., and S. Loreto, "DTLS
Encapsulation of SCTP Packets", draft-ietf-tsvwg-sctp- Encapsulation of SCTP Packets", draft-ietf-tsvwg-sctp-
dtls-encaps-04 (work in progress), May 2014. dtls-encaps-04 (work in progress), May 2014.
[I-D.ietf-rtcweb-security] [I-D.ietf-rtcweb-security]
Rescorla, E., "Security Considerations for WebRTC", draft- Rescorla, E., "Security Considerations for WebRTC", draft-
ietf-rtcweb-security-06 (work in progress), January 2014. ietf-rtcweb-security-06 (work in progress), January 2014.
 End of changes. 36 change blocks. 
102 lines changed or deleted 96 lines changed or added

This html diff was produced by rfcdiff 1.41. The latest version is available from http://tools.ietf.org/tools/rfcdiff/