draft-ietf-rtcweb-data-channel-07.txt   draft-ietf-rtcweb-data-channel-08.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: August 15, 2014 Ericsson Expires: October 11, 2014 Ericsson
M. Tuexen M. Tuexen
Muenster Univ. of Appl. Sciences Muenster Univ. of Appl. Sciences
February 11, 2014 April 9, 2014
WebRTC Data Channels WebRTC Data Channels
draft-ietf-rtcweb-data-channel-07.txt draft-ietf-rtcweb-data-channel-08.txt
Abstract Abstract
The Real-Time Communication in WEB-browsers working group is charged The Real-Time Communication in WEB-browsers working group is charged
to provide protocol support for direct interactive rich communication to provide protocol support for direct interactive rich communication
using audio, video, and data between two peers' web-browsers. This using audio, video, and data between two peers' web-browsers. This
document specifies the non-media data transport aspects of the WebRTC document specifies the non-(S)RTP media data transport aspects of the
framework. It provides an architectural overview of how the Stream WebRTC framework. It provides an architectural overview of how the
Control Transmission Protocol (SCTP) is used in the WebRTC context as Stream Control Transmission Protocol (SCTP) is used in the WebRTC
a generic transport service allowing WEB-browsers to exchange generic context as a generic transport service allowing WEB-browsers to
data from peer to peer. exchange generic data from peer to 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 August 15, 2014. This Internet-Draft will expire on October 11, 2014.
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 20 skipping to change at page 2, line 20
Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
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 in the WebRTC Context . . . . . . . . . . . 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. 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 . . . . . . . . . . . . . . . . . . . . . 11 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-media data types in the context of WebRTC are handled by using Non-(S)RTP media data types in the context of WebRTC are handled by
SCTP [RFC4960] encapsulated in DTLS [RFC6347]. using SCTP [RFC4960] encapsulated in DTLS [RFC6347].
+----------+ +----------+
| 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
[I-D.ietf-tsvwg-sctp-dtls-encaps]) over ICE/UDP (see [RFC5245]) [I-D.ietf-tsvwg-sctp-dtls-encaps]) over ICE/UDP (see [RFC5245])
provides a NAT traversal solution together with confidentiality, provides a NAT traversal solution together with confidentiality,
source authentication, and integrity protected transfers. This data source authentication, and integrity protected transfers. This data
transport service operates in parallel to the media transports, and transport service operates in parallel to the (S)RTP media
all of them can eventually share a single transport-layer port transports, and all of them can eventually share a single transport-
number. layer port number.
SCTP as specified in [RFC4960] with the partial reliability extension SCTP as specified in [RFC4960] with the partial reliability extension
defined in [RFC3758] provides multiple streams natively with defined in [RFC3758] and the additional policies defined in
reliable, and partially-reliable delivery modes for user messages. [I-D.ietf-tsvwg-sctp-prpolicies] provides multiple streams natively
Using the reconfiguration extension defined in [RFC6525] allows to with reliable, and the relevant partially-reliable delivery modes for
increase the number of streams during the lifetime of an SCTP user messages. Using the reconfiguration extension defined in
association and to reset individual SCTP streams. [RFC6525] allows to increase the number of streams during the
lifetime of an SCTP association and to reset individual SCTP streams.
Using [I-D.ietf-tsvwg-sctp-ndata] allows to interleave large messages
to avoid the monopolization and adds the support of prioritizing of
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 arguments 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
media data between WEB-browsers. non-(S)RTP 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 defined use cases specific to data channels. For This section defines use cases specific to data channels. For
general use cases see [I-D.ietf-rtcweb-use-cases-and-requirements]. general 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 media channels, or all media at any time there may be no (S)RTP media channels, or all
channels may be inactive, and that there may also be reliable (S)RTP media channels may be inactive, and that there may
data channels in use. also be 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 media channels, or they may be inactive at any given have no (S)RTP media channels, or they may be inactive at any
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-realtime 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: Realtime 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.
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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 P2P data channels between two
browsers. browsers.
Req. 1: Multiple simultaneous data channels MUST be supported. Req. 1: Multiple simultaneous data channels MUST be supported.
Note that there may 0 or more media streams in parallel Note that there may be 0 or more (S)RTP media streams in
with the data channels in the same PeerConnection, and the parallel with the data channels in the same PeerConnection,
number and state (active/inactive) of these media streams and the number and state (active/inactive) of these (S)RTP
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 media streams of the PeerConnection, conjunction with the (S)RTP media streams of the
to ensure that data channels don't cause congestion PeerConnection, to ensure that data channels don't cause
problems for these media streams, and that the WebRTC congestion problems for these (S)RTP media streams, and
PeerConnection as a whole is fair with competing traffic that the WebRTC PeerConnection as a whole is fair with
such as TCP. competing traffic such as TCP.
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 media streams. This will other, and relative to the (S)RTP 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.
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
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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:
o Usage of a TCP-friendly congestion control. o Usage of a TCP-friendly congestion control.
o The congestion control is modifiable for integration with media o The congestion control is modifiable for integration with the
stream congestion control. (S)RTP media stream congestion control.
o Support of multiple unidirectional streams, each providing its own o 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. o Support of ordered and out-of-order message delivery.
o Supporting arbitrary large user message by providing fragmentation o Supporting arbitrary large user messages by providing
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 SCTP multihoming will not be used in WebRTC. The SCTP layer will
simply act as if it were running on a single-homed host, since that simply act as if it were running on a single-homed host, since that
is the abstraction that the lower layer (a connection oriented, is the abstraction that the lower layer (a connection oriented,
unreliable datagram service) exposes. 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 interesting features for transporting non-media data following features for transporting non-(S)RTP media data between
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 so called Payload Protocol Each SCTP user message contains a Payload Protocol Identifier (PPID)
Identifier (PPID) that is passed to SCTP by its upper layer and sent that is passed to SCTP by its upper layer on the sending side and
to its peer. This value can be used to multiplex multiple protocols provided to its upper layer on the receiving side. The PPID can be
over a single SCTP association. The sender provides for each used to multiplex/demultiplex multiple upper layers over a single
protocol a specific PPID and the receiver can demultiplex the SCTP association. In the WebRTP context, the PPID is used to
messages based on the received PPID. The PPID is used to distinguish distinguish between UTF-8 encoded user data, binary encoded userdata
UTF-8 encoded user data and binary encoded userdata. The Data and the Data Channel Establishment Protocol defined in
Channel Establishment Protocol defined in [I-D.ietf-rtcweb-data-protocol]. Please note that the PPID is not
[I-D.ietf-rtcweb-data-protocol] uses also a specific PPID to be accessible via the Javascript API.
distinguished from user data.
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 the following
Figure 2. Figure 2.
+------+ +------+------+------+
|WEBRTC| | DCEP | UTF-8|Binary|
| DATA | | | data | data |
+------+ +------+------+------+
| SCTP | | SCTP |
+--------------------+ +----------------------------------+
| STUN | SRTP | DTLS | | STUN | SRTP | DTLS |
+--------------------+ +----------------------------------+
| ICE | | ICE |
+--------------------+ +----------------------------------+
| UDP1 | UDP2 | ... | | UDP1 | UDP2 | ... |
+--------------------+ +----------------------------------+
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] in
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 media channels of the o shares the DTLS connection with the (S)RTP 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 [RFC6347], 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. (S)RTP 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-land, the SCTP stack also
needs to be a user-land stack. needs to be a user-land stack.
When using DTLS as the lower layer, only single homed SCTP When using DTLS as the lower layer, only single homed SCTP
associations MUST be used, since DTLS does not expose any address associations are supported, since DTLS does not expose any address
management to its upper layer. The ICE/UDP layer can handle IP management to its upper layer. The ICE/UDP layer can handle IP
address changes during a session without needing to notify the DTLS address changes during a session without needing interaction with the
and SCTP layers, though it would be advantageous to retest Path MTU DTLS and SCTP layers. However, SCTP SHOULD be notified when an
on an IP address change. address changes has happened. In this case SCTP SHOULD retest the
Path MTU and reset the congestion state to the initial state. In
DTLS implementations used for this stack SHOULD support controlling case of a window based congestion control like the one specified in
fields of the IP layer like the Don't Fragment (DF)-bit in case of [RFC4960], this means setting the congestion window and slow start
IPv4 and the Differentiated Services Code Point (DSCP) field required threshold to its initial values.
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 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 MUST 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 protocol stack of Figure 2 is used, DTLS protects the complete When the protocol stack of Figure 2 is used, DTLS protects the
SCTP packet, so it provides confidentiality, integrity and source complete SCTP packet, so it provides confidentiality, integrity and
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 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. The partial unordered and with partial or full reliability. The partial
reliability extension MUST support policies to limit reliability extension MUST support policies to limit
o the transmission and retransmission by time. o the transmission and retransmission by time.
o the number of retransmissions. o the number of retransmissions.
Limiting the number of retransmissions to zero combined with Limiting the number of retransmissions to zero combined with
unordered delivery provides a UDP-like service where each user unordered delivery provides a UDP-like service where each user
message is sent exactly once and delivered in the order received. 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 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. Since SCTP does not support the parallel (S)RTP media streams.
negotiation of a congestion control algorithm yet, alternate
congestion controls SHOULD either only require a different sender
side behavior using existing information carried in the association
or need also specify a negotiation of of a congestion control
algorithm.
6. The Usage of SCTP in the WebRTC Context 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.
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 reset extension defined in [RFC6525] MUST be supported.
It is used for closing channels. It is used for closing channels.
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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.
Once support for message interleaving as currently being discussed in The support for message interleaving as defined in
[I-D.ietf-tsvwg-sctp-ndata] is available, it SHOULD be supported. [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 The SCTP association will be set up when the two endpoints of the
WebRTC PeerConnection agree on opening it, as negotiated by JSEP WebRTC PeerConnection agree on opening it, as negotiated by JSEP
(typically an exchange of SDP) [I-D.ietf-rtcweb-jsep]. Additionally, (typically an exchange of SDP) [I-D.ietf-rtcweb-jsep]. It will use
the negotiation SHOULD include some type of congestion control the DTLS connection selected via ICE; typically this will be shared
selection. It will use the DTLS connection selected via ICE; via BUNDLE or equivalent with DTLS connections used to key the (S)RTP
typically this will be shared via BUNDLE or equivalent with DTLS media streams.
connections used to key the DTLS-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 negotiated
during the association setup. 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 one to another SCTP endpoint. The streams within an SCTP association to another SCTP endpoint. The streams are
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 order 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. Channel Definition
The W3C has consensus on defining the application API for WebRTC The W3C has consensus on defining the application API for WebRTC
DataChannels to be bidirectional. They also consider the notions of DataChannels to be bidirectional. They also consider the notions of
in-sequence, out-of-sequence, reliable and unreliable as properties in-sequence, out-of-sequence, reliable and unreliable as properties
of Channels. One strong wish is for the application-level API to be of Channels. One strong wish is for the application-level API to be
close to the API for WebSockets, which implies bidirectional streams close to the API for WebSockets, which implies bidirectional streams
of data and waiting for onopen to fire before sending, a textual of data and waiting for onopen to fire before sending, a textual
skipping to change at page 10, line 18 skipping to change at page 10, line 10
The realization of a bidirectional Data Channel is a pair of one The realization of a bidirectional Data Channel is a pair of one
incoming stream and one outgoing SCTP stream having the same stream incoming stream and one outgoing SCTP stream having the same stream
SCTP identifier. SCTP identifier.
How stream values are selected is protocol and implementation How stream values are selected is protocol and implementation
dependent. dependent.
6.5. Opening a Channel 6.5. Opening a Channel
Data channels can be opened by using internal or external Data channels can be opened by using negotiation within the SCTP
negotiation. The details are out of scope of this document. association, called in-band negotiation, or out-of-band negotiation.
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.
The details are out of scope of this document.
A simple protocol for internal negotiation is specified in A simple protocol for in-band negotiation is specified in
[I-D.ietf-rtcweb-data-protocol] and MUST be supported. [I-D.ietf-rtcweb-data-protocol].
When one side wants to open a channel using external negotiation, it When one side wants to open a channel using out-of-band negotiation,
picks a stream. This can be based on the DTLS role (the client picks it picks a stream. Unless otherwise defined or negotiated, the
even stream identifiers, the server odd stream identifiers) or done streams are picked based on the DTLS role (the client picks even
in a different way. However, the application is responsible for stream identifiers, the server odd stream identifiers). However, the
avoiding collisions with existing streams. If it attempts to re-use application is responsible for avoiding collisions with existing
a stream which is part of an existing Channel, the addition SHOULD streams. If it attempts to re-use a stream which is part of an
fail. In addition to choosing a stream, the application SHOULD also existing Channel, the addition SHOULD fail. In addition to choosing
inform the protocol of the options to use for sending messages. The a stream, the application SHOULD also determine the options to use
application MUST ensure in an application-specific manner that the for sending messages. The application MUST ensure in an application-
other side will also inform the protocol that the selected stream is specific manner that the application at the peer will also know the
to be used, and the parameters for sending data from that side. selected stream to be used, and the options for sending 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 Channel in both directions MUST be sent over the
underlying stream using the reliability defined when the Channel was underlying stream using the reliability defined when the Channel was
opened unless the options are changed, or per-message options are opened unless the options are changed, or per-message options are
specified by a higher level. 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
detected by the receiver (for example, illegal ordering), the
receiver SHOULD close the corresponding 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. Once this extension is available, it support message interleaving, which SHOULD be used. As long as
MUST be used. As long as message interleaving is not supported, the message interleaving is not supported, the sender SHOULD limit the
sender SHOULD limit the maximum message size to 16 KB to avoid maximum message size to 16 KB to avoid monopolization.
monopolization.
It is recommended that 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]. Resetting a stream set the corresponding outgoing streams [RFC6525]. This means that if one
side decides to close the channel, it resets the corresponding
outgoing stream. When the peer sees that an incoming stream was
reset, it also resets its corresponding outgoing stream. Once this
is completed, the channel is closed. Resetting a stream sets the
Stream Sequence Numbers (SSNs) of the stream back to 'zero' with a Stream Sequence Numbers (SSNs) of the stream back to 'zero' with a
corresponding notification to the application layer that the reset 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 to reuse after a reset has
been performed. 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 resetting the stream.
7. Security Considerations 7. Security Considerations
skipping to change at page 12, line 24 skipping to change at page 12, line 27
+------------------------------------+-----------+-----------+ +------------------------------------+-----------+-----------+
| WebRTC String | 51 | [RFCXXXX] | | WebRTC String | 51 | [RFCXXXX] |
| WebRTC Binary Partial (Deprecated) | 52 | [RFCXXXX] | | WebRTC Binary Partial (Deprecated) | 52 | [RFCXXXX] |
| WebRTC Binary | 53 | [RFCXXXX] | | WebRTC Binary | 53 | [RFCXXXX] |
| WebRTC String Partial (Deprecated) | 54 | [RFCXXXX] | | WebRTC String Partial (Deprecated) | 54 | [RFCXXXX] |
+------------------------------------+-----------+-----------+ +------------------------------------+-----------+-----------+
9. Acknowledgments 9. Acknowledgments
Many thanks for comments, ideas, and text from Harald Alvestrand, Many thanks for comments, ideas, and text from Harald Alvestrand,
Adam Bergkvist, Cullen Jennings, Eric Rescorla, Randall Stewart, Adam Bergkvist, Christer Holmberg, Cullen Jennings, Paul Kyzivat,
Justin Uberti, and Magnus Westerlund. Eric Rescorla, Irene Ruengeler, Randall Stewart, Justin Uberti, and
Magnus Westerlund.
10. References 10. References
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)
skipping to change at page 13, line 25 skipping to change at page 13, line 30
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
Protocol", draft-ietf-rtcweb-data-protocol-02 (work in Establishment Protocol", draft-ietf-rtcweb-data-
progress), February 2014. protocol-03 (work in progress), February 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-03 (work in progress), February 2014. dtls-encaps-03 (work in progress), February 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.
[I-D.ietf-rtcweb-security-arch] [I-D.ietf-rtcweb-security-arch]
Rescorla, E., "WebRTC Security Architecture", draft-ietf- Rescorla, E., "WebRTC Security Architecture", draft-ietf-
rtcweb-security-arch-08 (work in progress), January 2014. rtcweb-security-arch-09 (work in progress), February 2014.
[I-D.ietf-rtcweb-jsep] [I-D.ietf-rtcweb-jsep]
Uberti, J. and C. Jennings, "Javascript Session Uberti, J. and C. Jennings, "Javascript Session
Establishment Protocol", draft-ietf-rtcweb-jsep-05 (work Establishment Protocol", draft-ietf-rtcweb-jsep-06 (work
in progress), October 2013. in progress), February 2014.
[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.
[I-D.ietf-tsvwg-sctp-prpolicies] [I-D.ietf-tsvwg-sctp-prpolicies]
Tuexen, M., Seggelmann, R., Stewart, R., and S. Loreto, Tuexen, M., Seggelmann, R., Stewart, R., and S. Loreto,
"Additional Policies for the Partial Reliability Extension "Additional Policies for the Partial Reliability Extension
of the Stream Control Transmission Protocol", draft-ietf- of the Stream Control Transmission Protocol", draft-ietf-
tsvwg-sctp-prpolicies-01 (work in progress), January 2014. tsvwg-sctp-prpolicies-02 (work in progress), April 2014.
[I-D.ietf-mmusic-sctp-sdp] [I-D.ietf-mmusic-sctp-sdp]
Loreto, S. and G. Camarillo, "Stream Control Transmission Loreto, S. and G. Camarillo, "Stream Control Transmission
Protocol (SCTP)-Based Media Transport in the Session Protocol (SCTP)-Based Media Transport in the Session
Description Protocol (SDP)", draft-ietf-mmusic-sctp-sdp-05 Description Protocol (SDP)", draft-ietf-mmusic-sctp-sdp-06
(work in progress), October 2013. (work in progress), February 2014.
10.2. Informative References 10.2. Informative References
[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, May 2010.
[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, January 2011.
[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, May 2013.
[I-D.ietf-rtcweb-use-cases-and-requirements] [I-D.ietf-rtcweb-use-cases-and-requirements]
Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real- Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-
Time Communication Use-cases and Requirements", draft- Time Communication Use-cases and Requirements", draft-
ietf-rtcweb-use-cases-and-requirements-13 (work in ietf-rtcweb-use-cases-and-requirements-14 (work in
progress), February 2014. progress), February 2014.
Authors' Addresses Authors' Addresses
Randell Jesup Randell Jesup
Mozilla Mozilla
US US
Email: randell-ietf@jesup.org Email: randell-ietf@jesup.org
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