draft-ietf-rtcweb-data-channel-06.txt   draft-ietf-rtcweb-data-channel-07.txt 
RTCWeb 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: April 24, 2014 Ericsson Expires: August 15, 2014 Ericsson
M. Tuexen M. Tuexen
Muenster Univ. of Appl. Sciences Muenster Univ. of Appl. Sciences
October 21, 2013 February 11, 2014
RTCWeb Data Channels WebRTC Data Channels
draft-ietf-rtcweb-data-channel-06.txt draft-ietf-rtcweb-data-channel-07.txt
Abstract Abstract
The Real-Time Communication in WEB-browsers (RTCWeb) working group is The Real-Time Communication in WEB-browsers working group is charged
charged to provide protocol support for direct interactive rich to provide protocol support for direct interactive rich communication
communication using audio, video, and data between two peers' web- using audio, video, and data between two peers' web-browsers. This
browsers. This document specifies the non-media data transport document specifies the non-media data transport aspects of the WebRTC
aspects of the RTCWeb framework. It provides an architectural framework. It provides an architectural overview of how the Stream
overview of how the Stream Control Transmission Protocol (SCTP) is Control Transmission Protocol (SCTP) is used in the WebRTC context as
used in the RTCWeb context as a generic transport service allowing a generic transport service allowing WEB-browsers to exchange generic
WEB-browsers to exchange generic data from peer to peer. data from peer to peer.
Status of This Memo Status of This Memo
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Copyright Notice Copyright Notice
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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 . . . . . . . . . . 3 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 RTCWeb Context . . . . . . . . . . . 8 6. The Usage of SCTP in the WebRTC Context . . . . . . . . . . . 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 . . . . . . . . . . . . . . . . . . . 10 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 . . . . . . . . . . . . . . . . . . . . . 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-media data types in the context of RTCWeb are handled by using Non-media data types in the context of WebRTC are handled by using
SCTP [RFC4960] encapsulated in DTLS [RFC6347]. 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 media transports, and
all of them can eventually share a single transport-layer port all of them can eventually share a single transport-layer port
number. number.
skipping to change at page 3, line 24 skipping to change at page 3, line 24
defined in [RFC3758] provides multiple streams natively with defined in [RFC3758] provides multiple streams natively with
reliable, and partially-reliable delivery modes for user messages. reliable, and partially-reliable delivery modes for user messages.
Using the reconfiguration extension defined in [RFC6525] allows to Using the reconfiguration extension defined in [RFC6525] allows to
increase the number of streams during the lifetime of an SCTP increase the number of streams during the lifetime of an SCTP
association and to reset individual SCTP streams. association and to reset individual 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 arguments 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 RTCWeb protocol framework for transporting non- should be used by the WebRTC protocol framework for transporting non-
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 defined use cases specific to data channels. For This section defined 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 at is sent via one or more unreliable data channels. Note that
any time there may be no media channels, or all media channels may at any time there may be no media channels, or all media
be inactive, and that there may also be reliable data channels in channels may be inactive, and that there may also be reliable
use. 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 mute for a state update in a video chat or conference, such as
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 have no transferred, such as control information. Such a game may
media channels, or they may be inactive at any given time, or may have no media channels, or they may be inactive at any given
only be added due to in-game actions. 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 set of media streams in the U-C 6: Renegotiation of the configuration of the PeerConnection.
PeerConnection.
U-C 7: Proxy browsing, where a browser uses data channels of a U-C 7: Proxy browsing, where a browser uses data channels of a
PeerConnection to send and receive HTTP/HTTPS requests and data, PeerConnection to send and receive HTTP/HTTPS requests and
for example to avoid local internet filtering or monitoring. data, for example to avoid local Internet filtering or
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 with the Note that there may 0 or more media streams in parallel
data channels, and the number and state (active/inactive) of the with the data channels in the same PeerConnection, and the
media streams may change at any time. number and state (active/inactive) of these 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 MUST be congestion controlled; either Req. 3: Data channels of a PeerConnection MUST be congestion
individually, as a class, or in conjunction with the media controlled; either individually, as a class, or in
streams, to ensure that data channels don't cause congestion conjunction with the media streams of the PeerConnection,
problems for the media streams, and that the RTCWeb PeerConnection to ensure that data channels don't cause congestion
as a whole is fair with competing traffic such as TCP. problems for these media streams, and that the WebRTC
PeerConnection as a whole is fair with 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 other, the relative priority of each data channel relative to each
and relative to the media streams. [ TBD: how this is encoded and other, and relative to the media streams. This will
what the impact of this is. ] This will interact with the interact with the congestion control algorithms.
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-arch] for [I-D.ietf-rtcweb-security] and
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 how such that IP-layer fragmentation can be avoided no matter
large a message the JavaScript application passes to be sent. It how large a message the JavaScript application passes to be
also MUST ensure that large data channel transfers don't unduly sent. It also MUST ensure that large data channel
delay traffic on other data channels. transfers don't unduly delay traffic on other data
channels.
Req. 7: The data channel transport protocol MUST NOT encode local Req. 7: The data channel transport protocol MUST NOT encode local
IP addresses inside its protocol fields; doing so reveals IP addresses inside its protocol fields; doing so reveals
potentially private information, and leads to failure if the potentially private information, and leads to failure if
address is depended upon. the address is depended upon.
Req. 8: The data channel transport protocol SHOULD support Req. 8: The data channel transport protocol SHOULD support
unbounded-length "messages" (i.e., a virtual socket stream) at the unbounded-length "messages" (i.e., a virtual socket stream)
application layer, for such things as image-file-transfer; at the application layer, for such things as image-file-
Implementations might enforce a reasonable message size limit. transfer; Implementations might enforce a reasonable
message size limit.
Req. 9: The data channel transport protocol SHOULD avoid IP Req. 9: The data channel transport protocol SHOULD avoid IP
fragmentation. It MUST support PMTU (Path MTU) discovery and MUST fragmentation. It MUST support PMTU (Path MTU) discovery
NOT rely on ICMP or ICMPv6 being generated or being passed back, and MUST NOT rely on ICMP or ICMPv6 being generated or
especially for PMTU discovery. being passed back, especially for PMTU discovery.
Req. 10: It MUST be possible to implement the protocol stack in the Req. 10: It MUST be possible to implement the protocol stack in the
user application space. user application space.
5. SCTP over DTLS over UDP Considerations 5. SCTP over DTLS over UDP Considerations
The important features of SCTP in the RTCWeb 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 media
stream congestion control. 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 message by providing fragmentation
and reassembly. 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 RTCWeb. 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 NAT traversal in IPv4 based UDP in combination with ICE enables middlebox traversal in IPv4 and
networks. SCTP as specified in [RFC4960] MUST be used in combination IPv6 based networks. SCTP as specified in [RFC4960] MUST be used in
with the extension defined in [RFC3758] and provides the following combination with the extension defined in [RFC3758] and provides the
interesting features for transporting non-media data between following interesting features for transporting non-media data
browsers: between 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 so called Payload Protocol
Identifier (PPID) that is passed to SCTP by its upper layer and sent 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 to its peer. This value can be used to multiplex multiple protocols
over a single SCTP association. The sender provides for each over a single SCTP association. The sender provides for each
protocol a specific PPID and the receiver can demultiplex the protocol a specific PPID and the receiver can demultiplex the
messages based on the received PPID. messages based on the received PPID. The PPID is used to distinguish
UTF-8 encoded user data and binary encoded userdata. The Data
Channel Establishment Protocol defined in
[I-D.ietf-rtcweb-data-protocol] uses also a specific PPID to be
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 Figure The layering of protocols for WebRTC is shown in the following
2. Figure 2.
+------+ +------+
|RTCWEB| |WEBRTC|
| 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. o shares the DTLS connection with the media channels of the
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]. specified in [I-D.ietf-tsvwg-sctp-dtls-encaps]. Please note that the
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.
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 MUST be used, 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 to notify the DTLS
and SCTP layers, though it would be advantageous to retest Path MTU and SCTP layers, though it would be advantageous to retest Path MTU
on an IP address change. on an IP address change.
skipping to change at page 8, line 8 skipping to change at page 8, line 11
in case of IPv4 is required for performing path MTU discovery. The in case of IPv4 is required for performing path MTU discovery. The
DTLS implementation SHOULD also support sending user messages DTLS implementation SHOULD also support sending user messages
exceeding the Path MTU. 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 MUST NOT exceed 1200 bytes for IPv4 and 1280 for
IPv6. Taking an overhead of 20 bytes for IPv4, 40 bytes for IPv6, 8 IPv6.
bytes for UDP, 13 + X for DTLS and 28 bytes for SCTP into account,
this results in an SCTP payload of 1131 - X when IPv4 is used and
1192 - X bytes when IPv6 is used.
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 protocol stack of Figure 2 is used, DTLS protects the complete
SCTP packet, so it provides confidentiality, integrity and source SCTP packet, so it provides confidentiality, integrity and source
authentication of the complete SCTP packet. authentication of the complete SCTP packet.
This protocol stack MUST support the usage of multiple SCTP streams. This SCTP stack and its upper layer MUST support the usage of
A user message can be sent ordered or unordered and with partial or multiple SCTP streams. A user message can be sent ordered or
full reliability. The partial reliability extension MUST support unordered and with partial or full reliability. The partial
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 the standard one might improve the impact on the
parallel SRTP media streams. Since SCTP does not support the parallel SRTP media streams. Since SCTP does not support the
negotiation of a congestion control algorithm, alternate congestion negotiation of a congestion control algorithm yet, alternate
controls SHOULD only require a different sender side behavior using congestion controls SHOULD either only require a different sender
existing information carried in the association. 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 RTCWeb Context 6. The Usage of SCTP in the WebRTC Context
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.
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] MUST NOT be defined in [RFC6525], other features of [RFC5061] are not REQUIRED
used. 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.tuexen-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 Once support for message interleaving as currently being discussed in
[I-D.stewart-tsvwg-sctp-ndata] is available, it SHOULD be supported. [I-D.ietf-tsvwg-sctp-ndata] is available, it SHOULD be supported.
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]. Additionally,
the negotiation SHOULD include some type of congestion control the negotiation SHOULD include some type of congestion control
selection. It will use the DTLS connection selected via SDP; selection. It will use the DTLS connection selected via ICE;
typically this will be shared via BUNDLE or equivalent with DTLS typically this will be shared via BUNDLE or equivalent with DTLS
connections used to key the DTLS-SRTP media streams. connections used to key the DTLS-SRTP media streams.
The application SHOULD indicate the initial number of streams The number of streams negotiated during SCTP association setup SHOULD
required when opening the association, and if no value is supplied, be 65535, which is the maximum number of streams that can negotiated
the implementation SHOULD provide an appropriate default. If more during the association setup.
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.
6.3. 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 one to another SCTP endpoint. The streams
are used to provide the notion of in-sequence delivery and for are 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 order or unordered. Ordering is preserved only for ordered
messages sent on the same stream. messages sent on the same stream.
skipping to change at page 10, line 18 skipping to change at page 10, line 10
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
label used to identify the meaning of the stream, among other things. label used to identify the meaning of the stream, among other things.
Each data channel also has a priority. These priorities MUST NOT be Each data channel also has a priority. These priorities MUST NOT be
strict priorities. strict priorities.
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. incoming stream and one outgoing SCTP stream having the same stream
SCTP identifier.
Note that there's no requirement for the SCTP streams used to create How stream values are selected is protocol and implementation
a bidirectional channel have the same number in each direction. How dependent.
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 internal or external Data channels can be opened by using internal or external
negotiation. The details are out of scope of this document. negotiation. The details are out of scope of this document.
A simple protocol for internal negotiation is specified in A simple protocol for internal negotiation is specified in
[I-D.ietf-rtcweb-data-protocol] and MUST be supported. [I-D.ietf-rtcweb-data-protocol] and MUST be supported.
When one side wants to open a channel using external negotiation, it When one side wants to open a channel using external negotiation, it
picks a Stream. This can be based on the DTLS role (the client picks picks a stream. This can be based on the DTLS role (the client picks
even stream identifiers, the server odd stream identifiers) or done even stream identifiers, the server odd stream identifiers) or done
in a different way. However, the application is responsible for in a different way. However, the application is responsible for
avoiding collisions with existing Streams. If it attempts to re-use avoiding collisions with existing streams. If it attempts to re-use
a Stream which is part of an existing Channel, the addition SHOULD a stream which is part of an existing Channel, the addition SHOULD
fail. In addition to choosing a Stream, the application SHOULD also fail. In addition to choosing a stream, the application SHOULD also
inform the protocol of the options to use for sending messages. The inform the protocol of the options to use for sending messages. The
application MUST ensure in an application-specific manner that the application MUST ensure in an application-specific manner that the
other side will also inform the protocol that the selected Stream is other side will also inform the protocol that the selected stream is
to be used, and the parameters for sending data from that side. to be used, and the parameters 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 the PPID "DOMString Last" MUST be used, for JavaScript binary string encoded in UTF-8 the PPID "WebRTC String" MUST be used, for
data (ArrayBuffer or Blob) the PPID "Binary Data Last" MUST be used JavaScript binary data (ArrayBuffer or Blob) the PPID "WebRTC Binary"
(see Section 8). MUST be used (see Section 8).
The usage of the PPIDs "WebRTC String Partial" and "WebRTC Binary
Partial" is deprecated. They were used for a PPID based
fragmentation and reassembly of user messages belonging to reliable
and ordered data channels.
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.stewart-tsvwg-sctp-ndata] defines an extension to limitation, [I-D.ietf-tsvwg-sctp-ndata] defines an extension to
support message interleaving. Once such an extension is available, support message interleaving. Once this extension is available, it
it SHOULD be used. MUST be used. As long as message interleaving is not supported, the
sender SHOULD limit the maximum message size to 16 KB to avoid
As long as message interleaving is not supported, the sending monopolization.
application SHOULD fragment large user messages for reliable and
ordered data channels. For sending large JavaScript strings, it uses
the PPID "DOMString Partial" for all but the last fragments and the
PPID "DOMString Last" for the last one. For JavaScript binary data
the PPIDs "Binary Data Partial" and "Binary Data Last" are used. The
reassembly based on the PPID MUST be supported. For data channel
which are not reliable and ordered, the sender MAY limit the maximum
message size to avoid monopolization.
It is recommended that message size be kept within certain size It is recommended that message size be kept within certain size
bounds (TBD) as applications will not be able to support arbitrarily- bounds as applications will not be able to support arbitrarily-large
large single messages. single messages. This limit has to be negotiated, for example by
using [I-D.ietf-mmusic-sctp-sdp].
The sender MAY disable the Nagle algorithm to minimize the latency. The sender SHOULD disable the Nagle algorithm to minimize the
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]. Resetting a stream set 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.
skipping to change at page 12, line 13 skipping to change at page 12, line 4
[I-D.ietf-rtcweb-security-arch]. [I-D.ietf-rtcweb-security-arch].
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
Identifiers (PPIDs). [RFC4960] creates the registry "SCTP Payload Identifiers (PPIDs): "DOMString Last", "Binary Data Partial", "Binary
Protocol Identifiers" from which these identifiers were assigned. Data Last", and "DOMString Partial". [RFC4960] creates the registry
IANA is requested to update the reference of these four assignments "SCTP Payload Protocol Identifiers" from which these identifiers were
to point to this document. Therefore these four assignments should assigned. IANA is requested to update the reference of these four
be updated to read: assignments to point to this document and change the names of the
PPIDs. Therefore these four assignments should be updated to read:
+---------------------+-----------+-----------+ +------------------------------------+-----------+-----------+
| Value | SCTP PPID | Reference | | Value | SCTP PPID | Reference |
+---------------------+-----------+-----------+ +------------------------------------+-----------+-----------+
| DOMString Last | 51 | [RFCXXXX] | | WebRTC String | 51 | [RFCXXXX] |
| Binary Data Partial | 52 | [RFCXXXX] | | WebRTC Binary Partial (Deprecated) | 52 | [RFCXXXX] |
| Binary Data Last | 53 | [RFCXXXX] | | WebRTC Binary | 53 | [RFCXXXX] |
| DOMString Partial | 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, Cullen Jennings, Eric Rescorla, Randall Stewart,
Justin Uberti, and Magnus Westerlund. Justin Uberti, and Magnus Westerlund.
10. References 10. References
10.1. Normative References 10.1. Normative References
skipping to change at page 13, line 28 skipping to change at page 13, line 17
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 [RFC6347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer
Security Version 1.2", RFC 6347, January 2012. 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.stewart-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-stewart-tsvwg-sctp-ndata-03 (work in progress), draft-ietf-tsvwg-sctp-ndata-00 (work in progress),
October 2013. 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-00 (work in Protocol", draft-ietf-rtcweb-data-protocol-02 (work in
progress), July 2013. 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-02 (work in progress), October 2013. 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-05 (work in progress), July 2013. 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-07 (work in progress), July 2013. rtcweb-security-arch-08 (work in progress), January 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-04 (work Establishment Protocol", draft-ietf-rtcweb-jsep-05 (work
in progress), September 2013. in progress), October 2013.
[I-D.ietf-rtcweb-qos] [I-D.ietf-rtcweb-qos]
Dhesikan, S., Druta, D., Jones, P., and J. Polk, "DSCP and Dhesikan, S., Druta, D., Jones, P., and J. Polk, "DSCP and
other packet markings for RTCWeb QoS", draft-ietf-rtcweb- other packet markings for RTCWeb QoS", draft-ietf-rtcweb-
qos-00 (work in progress), October 2012. qos-00 (work in progress), October 2012.
[I-D.ietf-tsvwg-sctp-prpolicies]
Tuexen, M., Seggelmann, R., Stewart, R., and S. Loreto,
"Additional Policies for the Partial Reliability Extension
of the Stream Control Transmission Protocol", draft-ietf-
tsvwg-sctp-prpolicies-01 (work in progress), January 2014.
[I-D.ietf-mmusic-sctp-sdp]
Loreto, S. and G. Camarillo, "Stream Control Transmission
Protocol (SCTP)-Based Media Transport in the Session
Description Protocol (SDP)", draft-ietf-mmusic-sctp-sdp-05
(work in progress), October 2013.
10.2. Informative References 10.2. Informative References
[RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer
Security (DTLS) Extension to Establish Keys for the Secure
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-12 (work in ietf-rtcweb-use-cases-and-requirements-13 (work in
progress), October 2013. progress), February 2014.
[I-D.tuexen-tsvwg-sctp-prpolicies]
Tuexen, M., Seggelmann, R., Stewart, R., and S. Loreto,
"Additional Policies for the Partial Delivery Extension of
the Stream Control Transmission Protocol", draft-tuexen-
tsvwg-sctp-prpolicies-03 (work in progress), October 2013.
Authors' Addresses Authors' Addresses
Randell Jesup Randell Jesup
Mozilla Mozilla
US US
Email: randell-ietf@jesup.org Email: randell-ietf@jesup.org
Salvatore Loreto Salvatore Loreto
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