Network Working Group R. Jesup
Internet-Draft Mozilla
Intended status: Standards Track S. Loreto
Expires: August 28, 2013 Ericsson
M. Tuexen
Muenster Univ. of Appl. Sciences
February 24, 2013
RTCWeb Data Channels
draft-ietf-rtcweb-data-channel-03.txt
Abstract
The Web Real-Time Communication (WebRTC) working group is charged to
provide protocol support for direct interactive rich communication
using audio, video, and data between two peers' web-browsers. This
document specifies the non-media data transport aspects of the WebRTC
framework. It provides an architectural overview of how the Stream
Control Transmission Protocol (SCTP) is used in the WebRTC context as
a generic transport service allowing Web Browser to exchange generic
data from peer to peer.
Status of this Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
This Internet-Draft will expire on August 28, 2013.
Copyright Notice
Copyright (c) 2013 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of
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publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must
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the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Conventions . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Use Cases . . . . . . . . . . . . . . . . . . . . . . . . . . 3
3.1. Use Cases for Unreliable Data Channels . . . . . . . . . . 4
3.2. Use Cases for Reliable Data Channels . . . . . . . . . . . 4
4. Requirements . . . . . . . . . . . . . . . . . . . . . . . . . 4
5. SCTP over DTLS over UDP Considerations . . . . . . . . . . . . 6
6. The Usage of SCTP in the RTCWeb Context . . . . . . . . . . . 9
6.1. Association Setup . . . . . . . . . . . . . . . . . . . . 9
6.2. SCTP Streams . . . . . . . . . . . . . . . . . . . . . . . 9
6.3. Channel Definition . . . . . . . . . . . . . . . . . . . . 9
6.4. Usage of Payload Protocol Identifier . . . . . . . . . . . 10
7. Security Considerations . . . . . . . . . . . . . . . . . . . 10
8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 10
9. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 10
10. References . . . . . . . . . . . . . . . . . . . . . . . . . . 10
10.1. Normative References . . . . . . . . . . . . . . . . . . . 10
10.2. Informative References . . . . . . . . . . . . . . . . . . 12
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 12
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1. Introduction
Non-media data types in the context of RTCWeb are handled by using
SCTP [RFC4960] encapsulated in DTLS [RFC6347].
+----------+
| SCTP |
+----------+
| DTLS |
+----------+
| ICE/UDP |
+----------+
Figure 1: Basic stack diagram
The encapsulation of SCTP over DTLS (see
[I-D.ietf-tsvwg-sctp-dtls-encaps]) over ICE/UDP (see [RFC5245])
provides a NAT traversal solution together with confidentiality,
source authentication, and integrity protected transfers. This data
transport service operates in parallel to the media transports, and
all of them can eventually share a single transport-layer port
number.
SCTP as specified in [RFC4960] with the partial reliability extension
defined in [RFC3758] provides multiple streams natively with
reliable, and partially-reliable delivery modes.
The remainder of this document is organized as follows: Section 4 and
Section 3 provide requirements and use cases for both unreliable and
reliable peer to peer datagram base channel; Section 5 arguments SCTP
over DTLS over UDP; Section 6 provides an specification of how SCTP
should be used by the RTCWeb protocol framework for transporting non-
media data between browsers.
2. Conventions
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in [RFC2119].
3. Use Cases
This section defined use cases specific to data channels. For
general use cases see [I-D.ietf-rtcweb-use-cases-and-requirements].
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3.1. Use Cases for Unreliable Data Channels
U-C 1 A real-time game where position and object state information
is sent via one or more unreliable data channels. Note that
at any time there may be no media channels, or all media
channels may be inactive, and that there may also be reliable
data channels in use.
U-C 2 Providing non-critical information to a user about the reason
for a state update in a video chat or conference, such as Mute
state.
3.2. Use Cases for Reliable Data Channels
U-C 3 A real-time game where critical state information needs to be
transferred, such as control information. Such a game may
have no media channels, or they may be inactive at any given
time, or may only be added due to in-game actions.
U-C 4 Non-realtime file transfers between people chatting. Note
that this may involve a large number of files to transfer
sequentially or in parallel, such as when sharing a folder of
images or a directory of files.
U-C 5 Realtime text chat while talking with an individual or with
multiple people in a conference.
U-C 6 Renegotiation of the set of media streams in the
PeerConnection.
U-C 7 Proxy browsing, where a browser uses data channels of a
PeerConnection to send and receive HTTP/HTTPS requests and
data, for example to avoid local internet filtering or
monitoring.
4. Requirements
This section lists the requirements for P2P data channels between two
browsers.
Req. 1 Multiple simultaneous data channels MUST be supported. Note
that there may 0 or more media streams in parallel with the
data channels, and the number and state (active/inactive) of
the media streams may change at any time.
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Req. 2 Both reliable and unreliable data channels MUST be
supported.
Req. 3 Data channels MUST be congestion controlled; either
individually, as a class, or in conjunction with the media
streams, to ensure that data channels don't cause congestion
problems for the media streams, and that the RTCWeb
PeerConnection as a whole is fair with competing traffic
such as TCP.
Req. 4 The application SHOULD be able to provide guidance as to the
relative priority of each data channel relative to each
other, and relative to the media streams. [ TBD: how this is
encoded and what the impact of this is. ] This will interact
with the congestion control algorithms.
Req. 5 Data channels MUST be secured; allowing for confidentiality,
integrity and source authentication. See
[I-D.ietf-rtcweb-security] and
[I-D.ietf-rtcweb-security-arch] for detailed info.
Req. 6 Data channels MUST provide message fragmentation support
such that IP-layer fragmentation can be avoided no matter
how large a message the Javascript application passes to be
sent.
Req. 7 The data channel transport protocol MUST NOT encode local IP
addresses inside its protocol fields; doing so reveals
potentially private information, and leads to failure if the
address is depended upon.
Req. 8 The data channel transport protocol SHOULD support
unbounded-length "messages" (i.e., a virtual socket stream)
at the application layer, for such things as image-file-
transfer; Implementations might enforce a reasonable message
size limit.
Req. 9 The data channel packet format/encoding MUST be such that it
is impossible for a malicious Javascript to generate an
application message crafted such that it could be
interpreted as a native protocol over UDP - such as UPnP,
RTP, SNMP, STUN, etc.
Req. 10 The data channel transport protocol SHOULD avoid IP
fragmentation. It MUST support PMTU discovery and MUST NOT
rely on ICMP or ICMPv6 being generated or being passed back,
especially for PMTU discovery.
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Req. 11 It MUST be possible to implement the protocol stack in the
user application space.
5. SCTP over DTLS over UDP Considerations
The important features of SCTP in the RTCWeb context are:
o TCP-friendly congestion control.
o The congestion control is modifiable for integration with media
stream congestion control.
o Support for multiple channels with different characteristics.
o Support for out-of-order delivery.
o Support for large datagrams and PMTU-discovery and fragmentation.
o Reliable or partial reliability support.
o Support of multiple streams.
SCTP multihoming will not be used in RTCWeb. The SCTP layer will
simply act as if it were running on a single-homed host, since that
is the abstraction that the lower layer (a connection oriented,
unreliable datagram service) exposes.
The encapsulation of SCTP over DTLS defined in
[I-D.ietf-tsvwg-sctp-dtls-encaps] provides confidentiality, source
authenticated, and integrity protected transfers. Using DTLS over
UDP in combination with ICE enables NAT traversal in IPv4 based
networks. SCTP as specified in [RFC4960] MUST be used in combination
with the extension defined in [RFC3758] and provides the following
interesting features for transporting non-media data between
browsers:
o Support of multiple unidirectional streams.
o Ordered and unordered delivery of user messages.
o Reliable and partial-reliable transport of user messages.
Each SCTP user message contains a so called Payload Protocol
Identifier (PPID) that is passed to SCTP by its upper layer and sent
to its peer. This value can be used to multiplex multiple protocols
over a single SCTP association. The sender provides for each
protocol a specific PPID and the receiver can demultiplex the
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messages based on the received PPID.
The encapsulation of SCTP over DTLS, together with the SCTP features
listed above satisfies all the requirements listed in Section 4.
The layering of protocols for WebRTC is shown in the following
Figure 2.
+------+
|RTCWEB|
| DATA |
+------+
| SCTP |
+--------------------+
| STUN | SRTP | DTLS |
+--------------------+
| ICE |
+--------------------+
| UDP1 | UDP2 | ... |
+--------------------+
Figure 2: WebRTC protocol layers
This stack (especially in contrast to DTLS over SCTP [RFC6083] in
combination with SCTP over UDP [I-D.ietf-tsvwg-sctp-udp-encaps]) has
been chosen because it
o supports the transmission of arbitrary large user messages.
o shares the DTLS connection with the media channels.
o provides privacy for the SCTP control information.
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
specified in [I-D.ietf-tsvwg-sctp-dtls-encaps].
Since DTLS is typically implemented in user-land, the SCTP stack also
needs to be a user-land stack.
When using DTLS as the lower layer, only single homed SCTP
associations MUST be used, since DTLS does not expose any address
management to its upper layer. The ICE/UDP layer can handle IP
address changes during a session without needing to notify the DTLS
and SCTP layers, though it would be advantageous to retest path MTU
on an IP address change.
DTLS implementations used for this stack SHOULD support controlling
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fields of the IP layer like the Don't fragment (DF)-bit in case of
IPv4 and the Differentiated Services Code Point (DSCP) field required
for supporting [I-D.ietf-rtcweb-qos]. Being able to set the (DF)-bit
in case of IPv4 is required for performing path MTU discovery. The
DTLS implementation SHOULD also support sending user messages
exceeding the path MTU.
Incoming ICMP or ICMPv6 messages can't be processed by the SCTP
layer, since there is no way to identify the corresponding
association. Therefore SCTP MUST support performing Path MTU
discovery without relying on ICMP or ICMPv6 as specified in [RFC4821]
using probing messages specified in [RFC4820]. The initial Path MTU
MUST NOT exceed 1280 [ *** need justification ***] bytes until
measured otherwise.
In general, the lower layer interface of an SCTP implementation
SHOULD be adapted to address the differences between IPv4 or IPv6
(being connection-less) or DTLS (being connection-oriented).
When protocol stack of Figure 2 is used, DTLS protects the complete
SCTP packet, so it provides confidentiality, integrity and source
authentication of the complete SCTP packet.
This protocol stack 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
means that all SCTP streams within a single SCTP association share
the same congestion window. Traffic not being sent over SCTP is not
covered by the SCTP congestion control. Due to the typical parallel
SRTP media streams, a delay-sensitive congestion control algorithm
MUST be supported and the congestion control MAY be coordinated
between the data channels and the media streams to avoid a data
channel transfer ending up with most or all the channel bandwidth.
Since SCTP does not support the negotiation of a congestion control
algorithm, the algorithm either MUST be negotiated before
establishment of the SCTP association or MUST NOT require any
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negotiation because it only requires sender side behavior using
existing information carried in the association.
6. The Usage of SCTP in the RTCWeb Context
6.1. Association Setup
The SCTP association will be set up when the two endpoints of the
WebRTC PeerConnection agree on opening it, as negotiated by JSEP
(typically an exchange of SDP) [I-D.ietf-rtcweb-jsep]. Additionally,
the negotiation SHOULD include some type of congestion control
selection. It will use the DTLS connection selected via SDP;
typically this will be shared via BUNDLE with DTLS connections used
to key the DTLS-SRTP media streams.
The application SHOULD indicate the initial number of streams
required when opening the association, and if no value is supplied,
the implementation SHOULD provide an appropriate default. If more
simultaneous streams are needed, [RFC6525] allows adding additional
(but not removing) streams to an existing association. Note there
can be up to 65536 SCTP streams per SCTP association in each
direction.
6.2. SCTP Streams
SCTP defines a stream as an unidirectional logical channel existing
within an SCTP association one to another SCTP endpoint. The streams
are used to provide the notion of in-sequence delivery and for
multiplexing. Each user message is sent on a particular stream,
either order or unordered. Ordering is preserved only for all
ordered messages sent on the same stream.
6.3. Channel Definition
The W3C has consensus on defining the application API for WebRTC
dataChannels to be bidirectional. They also consider the notions of
in-sequence, out-of-sequence, reliable and un-reliable as properties
of Channels. One strong wish is for the application-level API to be
close to the API for WebSockets, which implies bidirectional streams
of data and waiting for onopen to fire before sending, a textual
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.
The simple protocol specified in [I-D.jesup-rtcweb-data-protocol]
MUST be used to set up and manage the bidirectional data channels.
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Note that there's no requirement for the SCTP streams used to create
a bidirectional channel have the same number in each direction. How
stream values are selected is protocol and implementation dependent.
Closing of a Data Channel MUST be signaled by resetting the
corresponding streams [RFC6525]. Resetting a stream set the Stream
Sequence Numbers (SSNs) of the stream back to 'zero' with a
corresponding notification to the application layer that the reset
has been performed. Streams are available to reuse after a reset has
been performed.
[RFC6525] also guarantees that all the messages are delivered (or
expired) before resetting the stream.
6.4. Usage of Payload Protocol Identifier
The SCTP Payload Protocol Identifiers (PPIDs) can be used to signal
the interpretation of the "Payload data", like the protocol specified
in [I-D.jesup-rtcweb-data-protocol] uses them to identify a
Javascript string, a Javascript array or a Javascript blob.
7. Security Considerations
This document does not add any additional considerations to the ones
given in [I-D.ietf-rtcweb-security] and
[I-D.ietf-rtcweb-security-arch].
8. IANA Considerations
This document does not require any actions by the IANA.
9. Acknowledgments
Many thanks for comments, ideas, and text from Harald Alvestrand,
Adam Bergkvist, Cullen Jennings, Eric Rescorla, Randall Stewart,
Justin Uberti, and Magnus Westerlund.
10. References
10.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
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[RFC3758] Stewart, R., Ramalho, M., Xie, Q., Tuexen, M., and P.
Conrad, "Stream Control Transmission Protocol (SCTP)
Partial Reliability Extension", RFC 3758, May 2004.
[RFC4820] Tuexen, M., Stewart, R., and P. Lei, "Padding Chunk and
Parameter for the Stream Control Transmission Protocol
(SCTP)", RFC 4820, March 2007.
[RFC4821] Mathis, M. and J. Heffner, "Packetization Layer Path MTU
Discovery", RFC 4821, March 2007.
[RFC4960] Stewart, R., "Stream Control Transmission Protocol",
RFC 4960, September 2007.
[RFC5245] Rosenberg, J., "Interactive Connectivity Establishment
(ICE): A Protocol for Network Address Translator (NAT)
Traversal for Offer/Answer Protocols", RFC 5245,
April 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
Transmission Protocol (SCTP) Stream Reconfiguration",
RFC 6525, February 2012.
[I-D.jesup-rtcweb-data-protocol]
Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel
Protocol", draft-jesup-rtcweb-data-protocol-03 (work in
progress), September 2012.
[I-D.ietf-tsvwg-sctp-dtls-encaps]
Jesup, R., Loreto, S., Stewart, R., and M. Tuexen, "DTLS
Encapsulation of SCTP Packets for RTCWEB",
draft-ietf-tsvwg-sctp-dtls-encaps-00 (work in progress),
February 2013.
[I-D.ietf-rtcweb-security]
Rescorla, E., "Security Considerations for RTC-Web",
draft-ietf-rtcweb-security-04 (work in progress),
January 2013.
[I-D.ietf-rtcweb-security-arch]
Rescorla, E., "RTCWEB Security Architecture",
draft-ietf-rtcweb-security-arch-06 (work in progress),
January 2013.
[I-D.ietf-rtcweb-jsep]
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Uberti, J. and C. Jennings, "Javascript Session
Establishment Protocol", draft-ietf-rtcweb-jsep-02 (work
in progress), October 2012.
[I-D.ietf-rtcweb-qos]
Dhesikan, S., Druta, D., Jones, P., and J. Polk, "DSCP and
other packet markings for RTCWeb QoS",
draft-ietf-rtcweb-qos-00 (work in progress), October 2012.
10.2. Informative References
[RFC6083] Tuexen, M., Seggelmann, R., and E. Rescorla, "Datagram
Transport Layer Security (DTLS) for Stream Control
Transmission Protocol (SCTP)", RFC 6083, January 2011.
[I-D.ietf-rtcweb-use-cases-and-requirements]
Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-
Time Communication Use-cases and Requirements",
draft-ietf-rtcweb-use-cases-and-requirements-10 (work in
progress), December 2012.
[I-D.ietf-tsvwg-sctp-udp-encaps]
Tuexen, M. and R. Stewart, "UDP Encapsulation of SCTP
Packets for End-Host to End-Host Communication",
draft-ietf-tsvwg-sctp-udp-encaps-11 (work in progress),
February 2013.
Authors' Addresses
Randell Jesup
Mozilla
US
Email: randell-ietf@jesup.org
Salvatore Loreto
Ericsson
Hirsalantie 11
Jorvas 02420
FI
Email: salvatore.loreto@ericsson.com
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Michael Tuexen
Muenster University of Applied Sciences
Stegerwaldstrasse 39
Steinfurt 48565
DE
Email: tuexen@fh-muenster.de
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