draft-ietf-rtcweb-data-channel-12.txt   draft-ietf-rtcweb-data-channel-13.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: April 1, 2015 Ericsson Expires: July 8, 2015 Ericsson
M. Tuexen M. Tuexen
Muenster Univ. of Appl. Sciences Muenster Univ. of Appl. Sciences
September 28, 2014 January 4, 2015
WebRTC Data Channels WebRTC Data Channels
draft-ietf-rtcweb-data-channel-12.txt draft-ietf-rtcweb-data-channel-13.txt
Abstract Abstract
The WebRTC framework specifies protocol support for direct The WebRTC framework specifies protocol support for direct
interactive rich communication using audio, video, and data between interactive rich communication using audio, video, and data between
two peers' web-browsers. This document specifies the non-media data two peers' web-browsers. This document specifies the non-media data
transport aspects of the WebRTC framework. It provides an transport aspects of the WebRTC framework. It provides an
architectural overview of how the Stream Control Transmission architectural overview of how the Stream Control Transmission
Protocol (SCTP) is used in the WebRTC context as a generic transport Protocol (SCTP) is used in the WebRTC context as a generic transport
service allowing WEB-browsers to exchange generic data from peer to service allowing WEB-browsers to exchange generic data from peer to
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This Internet-Draft will expire on April 1, 2015. This Internet-Draft will expire on July 8, 2015.
Copyright Notice Copyright Notice
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described in the Simplified BSD License. described in the Simplified BSD License.
Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
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 . . . . . . . . . 4
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 . . . . . . . . . . . 6
6. The Usage of SCTP for Data Channels . . . . . . . . . . . . . 8 6. The Usage of SCTP for Data Channels . . . . . . . . . . . . . 8
6.1. SCTP Protocol Considerations . . . . . . . . . . . . . . 8 6.1. SCTP Protocol Considerations . . . . . . . . . . . . . . 8
6.2. SCTP Association Management . . . . . . . . . . . . . . . 9 6.2. SCTP Association Management . . . . . . . . . . . . . . . 9
6.3. SCTP Streams . . . . . . . . . . . . . . . . . . . . . . 9 6.3. SCTP Streams . . . . . . . . . . . . . . . . . . . . . . 9
6.4. Data Channel Definition . . . . . . . . . . . . . . . . . 9 6.4. Data Channel Definition . . . . . . . . . . . . . . . . . 10
6.5. Opening a Data Channel . . . . . . . . . . . . . . . . . 10 6.5. Opening a Data Channel . . . . . . . . . . . . . . . . . 10
6.6. Transferring User Data on a Data Channel . . . . . . . . 11 6.6. Transferring User Data on a Data Channel . . . . . . . . 11
6.7. Closing a Data Channel . . . . . . . . . . . . . . . . . 12 6.7. Closing a Data Channel . . . . . . . . . . . . . . . . . 12
7. Security Considerations . . . . . . . . . . . . . . . . . . . 12 7. Security Considerations . . . . . . . . . . . . . . . . . . . 13
8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 13 8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 13
9. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 13 9. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 14
10. References . . . . . . . . . . . . . . . . . . . . . . . . . 14 10. References . . . . . . . . . . . . . . . . . . . . . . . . . 14
10.1. Normative References . . . . . . . . . . . . . . . . . . 14 10.1. Normative References . . . . . . . . . . . . . . . . . . 14
10.2. Informative References . . . . . . . . . . . . . . . . . 15 10.2. Informative References . . . . . . . . . . . . . . . . . 15
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 16 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 16
1. Introduction 1. Introduction
In the WebRTC framework, communication between the parties consists In the WebRTC framework, communication between the parties consists
of media (for example audio and video) and non-media data. Media is of media (for example audio and video) and non-media data. Media is
sent using SRTP, and is not specified further here. Non-media data sent using SRTP, and is not specified further here. Non-media data
is handled by using SCTP [RFC4960] encapsulated in DTLS [RFC4347]. is handled by using SCTP [RFC4960] encapsulated in DTLS. DTLS 1.0 is
defined in [RFC4347] and the present latest version, DTLS 1.2, is
defined in [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 SRTP media transports, transport service operates in parallel to the SRTP media transports,
and all of them can eventually share a single transport-layer port and all of them can eventually share a single UDP port number.
number.
SCTP as specified in [RFC4960] with the partial reliability extension SCTP as specified in [RFC4960] with the partial reliability extension
defined in [RFC3758] and the additional policies defined in defined in [RFC3758] and the additional policies defined in
[I-D.ietf-tsvwg-sctp-prpolicies] provides multiple streams natively [I-D.ietf-tsvwg-sctp-prpolicies] provides multiple streams natively
with reliable, and the relevant partially-reliable delivery modes for with reliable, and the relevant partially-reliable delivery modes for
user messages. Using the reconfiguration extension defined in user messages. Using the reconfiguration extension defined in
[RFC6525] allows to increase the number of streams during the [RFC6525] allows to increase the number of streams during the
lifetime of an SCTP association and to reset individual SCTP streams. lifetime of an SCTP association and to reset individual SCTP streams.
Using [I-D.ietf-tsvwg-sctp-ndata] allows to interleave large messages Using [I-D.ietf-tsvwg-sctp-ndata] allows to interleave large messages
to avoid the monopolization and adds the support of prioritizing of to avoid the monopolization and adds the support of prioritizing of
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media data between WEB-browsers. media data between WEB-browsers.
2. Conventions 2. Conventions
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in [RFC2119]. document are to be interpreted as described in [RFC2119].
3. Use Cases 3. Use Cases
This section defines use cases specific to data channels. For WebRTC This section defines use cases specific to data channels. Please
use cases see [I-D.ietf-rtcweb-use-cases-and-requirements]. note that this section is informational only.
3.1. Use Cases for Unreliable Data Channels 3.1. Use Cases for Unreliable Data Channels
U-C 1: A real-time game where position and object state information U-C 1: A real-time game where position and object state information
is sent via one or more unreliable data channels. Note that is sent via one or more unreliable data channels. Note that
at any time there may be no SRTP media channels, or all SRTP at any time there may be no SRTP media channels, or all SRTP
media channels may be inactive, and that there may also be media channels may be inactive, and that there may also be
reliable data channels in use. reliable data channels in use.
U-C 2: Providing non-critical information to a user about the reason U-C 2: Providing non-critical information to a user about the reason
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U-C 6: Renegotiation of the configuration of the PeerConnection. U-C 6: Renegotiation of the configuration of the PeerConnection.
U-C 7: Proxy browsing, where a browser uses data channels of a U-C 7: Proxy browsing, where a browser uses data channels of a
PeerConnection to send and receive HTTP/HTTPS requests and PeerConnection to send and receive HTTP/HTTPS requests and
data, for example to avoid local Internet filtering or data, for example to avoid local Internet filtering or
monitoring. monitoring.
4. Requirements 4. Requirements
This section lists the requirements for P2P data channels between two This section lists the requirements for P2P data channels between two
browsers. browsers. Please note that this section is informational only.
Req. 1: Multiple simultaneous data channels MUST be supported. Req. 1: Multiple simultaneous data channels must be supported.
Note that there may be 0 or more SRTP media streams in Note that there may be 0 or more SRTP media streams in
parallel with the data channels in the same PeerConnection, parallel with the data channels in the same PeerConnection,
and the number and state (active/inactive) of these SRTP and the number and state (active/inactive) of these SRTP
media streams may change at any time. media streams may change at any time.
Req. 2: Both reliable and unreliable data channels MUST be Req. 2: Both reliable and unreliable data channels must be
supported. supported.
Req. 3: Data channels of a PeerConnection MUST be congestion Req. 3: Data channels of a PeerConnection must be congestion
controlled; either individually, as a class, or in controlled; either individually, as a class, or in
conjunction with the SRTP media streams of the conjunction with the SRTP media streams of the
PeerConnection, to ensure that data channels don't cause PeerConnection, to ensure that data channels don't cause
congestion problems for these SRTP media streams, and that congestion problems for these SRTP media streams, and that
the WebRTC PeerConnection does not cause excessive problems the WebRTC PeerConnection does not cause excessive problems
when run in parallel with TCP connections. when run in parallel with TCP connections.
Req. 4: The application SHOULD be able to provide guidance as to Req. 4: The application should be able to provide guidance as to
the relative priority of each data channel relative to each the relative priority of each data channel relative to each
other, and relative to the SRTP media streams. This will other, and relative to the SRTP media streams. This will
interact with the congestion control algorithms. interact with the congestion control algorithms.
Req. 5: Data channels MUST be secured; allowing for Req. 5: Data channels must be secured; allowing for
confidentiality, integrity and source authentication. See confidentiality, integrity and source authentication. See
[I-D.ietf-rtcweb-security] and [I-D.ietf-rtcweb-security] and
[I-D.ietf-rtcweb-security-arch] for detailed info. [I-D.ietf-rtcweb-security-arch] for detailed info.
Req. 6: Data channels MUST provide message fragmentation support Req. 6: Data channels must provide message fragmentation support
such that IP-layer fragmentation can be avoided no matter such that IP-layer fragmentation can be avoided no matter
how large a message the JavaScript application passes to be how large a message the JavaScript application passes to be
sent. It also MUST ensure that large data channel sent. It also must ensure that large data channel
transfers don't unduly delay traffic on other data transfers don't unduly delay traffic on other data
channels. channels.
Req. 7: The data channel transport protocol MUST NOT encode local Req. 7: The data channel transport protocol must not encode local
IP addresses inside its protocol fields; doing so reveals IP addresses inside its protocol fields; doing so reveals
potentially private information, and leads to failure if potentially private information, and leads to failure if
the 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) unbounded-length "messages" (i.e., a virtual socket stream)
at the application layer, for such things as image-file- at the application layer, for such things as image-file-
transfer; Implementations might enforce a reasonable transfer; Implementations might enforce a reasonable
message size limit. message size limit.
Req. 9: The data channel transport protocol SHOULD avoid IP Req. 9: The data channel transport protocol should avoid IP
fragmentation. It MUST support PMTU (Path MTU) discovery fragmentation. It must support PMTU (Path MTU) discovery
and MUST NOT rely on ICMP or ICMPv6 being generated or and must not rely on ICMP or ICMPv6 being generated or
being passed back, especially for PMTU discovery. being passed back, especially for PMTU discovery.
Req. 10: It MUST be possible to implement the protocol stack in the Req. 10: It must be possible to implement the protocol stack in the
user application space. user application space.
5. SCTP over DTLS over UDP Considerations 5. SCTP over DTLS over UDP Considerations
The important features of SCTP in the WebRTC context are: The important features of SCTP in the WebRTC context are:
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 the SRTP o The congestion control is modifiable for integration with the SRTP
media stream congestion control. media stream congestion control.
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o Support of multiple unidirectional streams. o Support of multiple unidirectional streams.
o Ordered and unordered delivery of user messages. o Ordered and unordered delivery of user messages.
o Reliable and partial-reliable transport of user messages. o Reliable and partial-reliable transport of user messages.
Each SCTP user message contains a Payload Protocol Identifier (PPID) Each SCTP user message contains a Payload Protocol Identifier (PPID)
that is passed to SCTP by its upper layer on the sending side and that is passed to SCTP by its upper layer on the sending side and
provided to its upper layer on the receiving side. The PPID can be provided to its upper layer on the receiving side. The PPID can be
used to multiplex/demultiplex multiple upper layers over a single used to multiplex/demultiplex multiple upper layers over a single
SCTP association. In the WebRTP context, the PPID is used to SCTP association. In the WebRTC context, the PPID is used to
distinguish between UTF-8 encoded user data, binary encoded userdata distinguish between UTF-8 encoded user data, binary encoded userdata
and the Data Channel Establishment Protocol defined in and the Data Channel Establishment Protocol defined in
[I-D.ietf-rtcweb-data-protocol]. Please note that the PPID is not [I-D.ietf-rtcweb-data-protocol]. Please note that the PPID is not
accessible via the Javascript API. accessible via the Javascript API.
The encapsulation of SCTP over DTLS, together with the SCTP features The encapsulation of SCTP over DTLS, together with the SCTP features
listed above satisfies all the requirements listed in Section 4. listed above satisfies all the requirements listed in Section 4.
The layering of protocols for WebRTC is shown in the following The layering of protocols for WebRTC is shown in the following
Figure 2. Figure 2.
+------+------+------+ +------+------+------+
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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 SRTP media channels of the o shares the DTLS connection with the SRTP media channels of the
PeerConnection. PeerConnection.
o provides privacy for the SCTP control information. o provides privacy for the SCTP control information.
Considering the protocol stack of Figure 2 the usage of DTLS over UDP Considering the protocol stack of Figure 2 the usage of DTLS 1.0 over
is specified in [RFC4347], while the usage of SCTP on top of DTLS is UDP is specified in [RFC4347] and the usage of DTLS 1.2 over UDP in
specified in [RFC6347], while the usage of SCTP on top of DTLS is
specified in [I-D.ietf-tsvwg-sctp-dtls-encaps]. Please note that the specified in [I-D.ietf-tsvwg-sctp-dtls-encaps]. Please note that the
demultiplexing STUN vs. SRTP vs. DTLS is done as described in demultiplexing STUN vs. SRTP vs. DTLS is done as described in
Section 5.1.2 of [RFC5764] and SCTP is the only payload of DTLS. Section 5.1.2 of [RFC5764] and SCTP is the only payload of DTLS.
Since DTLS is typically implemented in user application space, the Since DTLS is typically implemented in user application space, the
SCTP stack also needs to be a user application space stack. SCTP stack also needs to be a user application space stack.
The ICE/UDP layer can handle IP address changes during a session The ICE/UDP layer can handle IP address changes during a session
without needing interaction with the DTLS and SCTP layers. However, without needing interaction with the DTLS and SCTP layers. However,
SCTP SHOULD be notified when an address changes has happened. In SCTP SHOULD be notified when an address changes has happened. In
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Incoming ICMP or ICMPv6 messages can't be processed by the SCTP Incoming ICMP or ICMPv6 messages can't be processed by the SCTP
layer, since there is no way to identify the corresponding layer, since there is no way to identify the corresponding
association. Therefore SCTP MUST support performing Path MTU association. Therefore SCTP MUST support performing Path MTU
discovery without relying on ICMP or ICMPv6 as specified in [RFC4821] discovery without relying on ICMP or ICMPv6 as specified in [RFC4821]
using probing messages specified in [RFC4820]. The initial Path MTU using probing messages specified in [RFC4820]. The initial Path MTU
at the IP layer SHOULD NOT exceed 1200 bytes for IPv4 and 1280 for at the IP layer SHOULD NOT exceed 1200 bytes for IPv4 and 1280 for
IPv6. IPv6.
In general, the lower layer interface of an SCTP implementation In general, the lower layer interface of an SCTP implementation
SHOULD be adapted to address the differences between IPv4 and IPv6 should be adapted to address the differences between IPv4 and IPv6
(being connection-less) or DTLS (being connection-oriented). (being connection-less) or DTLS (being connection-oriented).
When the protocol stack of Figure 2 is used, DTLS protects the When the protocol stack of Figure 2 is used, DTLS protects the
complete SCTP packet, so it provides confidentiality, integrity and complete SCTP packet, so it provides confidentiality, integrity and
source authentication of the complete SCTP packet. source authentication of the complete SCTP packet.
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
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multiple SCTP streams. A user message can be sent ordered or multiple SCTP streams. A user message can be sent ordered or
unordered and with partial or full reliability. unordered and with partial or full reliability.
The following SCTP protocol extensions are required: The following SCTP protocol extensions are required:
o The stream reconfiguration extension defined in [RFC6525] MUST be o The stream reconfiguration extension defined in [RFC6525] MUST be
supported. It is used for closing channels. supported. It is used for closing channels.
o The dynamic address reconfiguration extension defined in [RFC5061] 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 OPTIONAL.
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. Limiting the [I-D.ietf-tsvwg-sctp-prpolicies] MUST be supported. Limiting the
number of retransmissions to zero combined with unordered delivery number of retransmissions to zero combined with unordered delivery
provides a UDP-like service where each user message is sent provides a UDP-like service where each user message is sent
exactly once and delivered in the order received. exactly once and delivered in the order received.
The support for message interleaving as defined in The support for message interleaving as defined in
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SCTP defines a stream as a unidirectional logical channel existing SCTP defines a stream as a unidirectional logical channel existing
within an SCTP association to another SCTP endpoint. The streams are within an SCTP association to another SCTP endpoint. The streams are
used to provide the notion of in-sequence delivery and for used to provide the notion of in-sequence delivery and for
multiplexing. Each user message is sent on a particular stream, multiplexing. Each user message is sent on a particular stream,
either ordered or unordered. Ordering is preserved only for ordered either ordered or unordered. Ordering is preserved only for ordered
messages sent on the same stream. messages sent on the same stream.
6.4. Data Channel Definition 6.4. Data Channel Definition
One strong wish is for the application-level API to be close to the Data channels are defined such that their accompanying application-
API for WebSockets, which implies bidirectional streams of data and a level API can closely mirror the API for WebSockets, which implies
textual field called 'label' used to identify the meaning of the data bidirectional streams of data and a textual field called 'label' used
channel. to identify the meaning of the data channel.
The realization of a data channel is a pair of one incoming stream The realization of a data channel is a pair of one incoming stream
and one outgoing SCTP stream having the same SCTP stream identifier. and one outgoing SCTP stream having the same SCTP stream identifier.
How these SCTP stream identifiers are selected is protocol and How these SCTP stream identifiers are selected is protocol and
implementation dependent. This allows a bidirectional communication. implementation dependent. This allows a bidirectional communication.
Additionally, each data channel has the following properties in each Additionally, each data channel has the following properties in each
direction: direction:
o reliable or unreliable message transmission. In case of o reliable or unreliable message transmission. In case of
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A simple protocol for in-band negotiation is specified in A simple protocol for in-band negotiation is specified in
[I-D.ietf-rtcweb-data-protocol]. [I-D.ietf-rtcweb-data-protocol].
When one side wants to open a channel using out-of-band negotiation, When one side wants to open a channel using out-of-band negotiation,
it picks a stream. Unless otherwise defined or negotiated, the it picks a stream. Unless otherwise defined or negotiated, the
streams are picked based on the DTLS role (the client picks even streams are picked based on the DTLS role (the client picks even
stream identifiers, the server odd stream identifiers). However, the stream identifiers, the server odd stream identifiers). However, the
application is responsible for avoiding collisions with existing application is responsible for avoiding collisions with existing
streams. If it attempts to re-use a stream which is part of an streams. If it attempts to re-use a stream which is part of an
existing data channel, the addition SHOULD fail. In addition to existing data channel, the addition MUST fail. In addition to
choosing a stream, the application SHOULD also determine the options choosing a stream, the application SHOULD also determine the options
to use for sending messages. The application MUST ensure in an to use for sending messages. The application MUST ensure in an
application-specific manner that the application at the peer will application-specific manner that the application at the peer will
also know the selected stream to be used, and the options for sending also know the selected stream to be used, and the options for sending
data from that side. data from that side.
6.6. Transferring User Data on a Data Channel 6.6. Transferring User Data on a Data Channel
All data sent on a data channel in both directions MUST be sent over All data sent on a data channel in both directions MUST be sent over
the underlying stream using the reliability defined when the data the underlying stream using the reliability defined when the data
channel was opened unless the options are changed, or per-message channel was opened unless the options are changed, or per-message
options are specified by a higher level. options are specified by a higher level.
No more than one message should be put into an SCTP user message. The message-orientation of SCTP is used to preserve the message
boundaries of user messages. Therefore, senders MUST NOT put more
than one application message into an SCTP user message. Unless the
deprecated PPID-based fragmentation and reassembly is used, the
sender MUST include exactly one application message in each 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". The following PPIDs MUST be interpretation of the "Payload data". The following PPIDs MUST be
used (see Section 8): used (see Section 8):
WebRTC String: to identify a non-empty JavaScript string encoded in WebRTC String: to identify a non-empty JavaScript string encoded in
UTF-8. UTF-8.
WebRTC String Empty: to identify an empty JavaScript string encoded WebRTC String Empty: to identify an empty JavaScript string encoded
in UTF-8. in UTF-8.
skipping to change at page 13, line 5 skipping to change at page 13, line 16
[RFC6525] also guarantees that all the messages are delivered (or [RFC6525] also guarantees that all the messages are delivered (or
abandoned) before the stream is reset. abandoned) before the stream is reset.
7. Security Considerations 7. Security Considerations
This document does not add any additional considerations to the ones This document does not add any additional considerations to the ones
given in [I-D.ietf-rtcweb-security] and given in [I-D.ietf-rtcweb-security] and
[I-D.ietf-rtcweb-security-arch]. [I-D.ietf-rtcweb-security-arch].
I should be noted that a receiver must be prepared that the sender It should be noted that a receiver must be prepared that the sender
tries to send arbitrary large messages. tries to send arbitrary large messages.
8. IANA Considerations 8. IANA Considerations
[NOTE to RFC-Editor: [NOTE to RFC-Editor:
"RFCXXXX" is to be replaced by the RFC number you assign this "RFCXXXX" is to be replaced by the RFC number you assign this
document. document.
] ]
skipping to change at page 13, line 45 skipping to change at page 14, line 8
| WebRTC Binary | 53 | [RFCXXXX] | 2013-09-20 | | WebRTC Binary | 53 | [RFCXXXX] | 2013-09-20 |
| WebRTC String Partial | 54 | [RFCXXXX] | 2013-09-20 | | WebRTC String Partial | 54 | [RFCXXXX] | 2013-09-20 |
| (Deprecated) | | | | | (Deprecated) | | | |
| WebRTC String Empty | 56 | [RFCXXXX] | 2014-08-22 | | WebRTC String Empty | 56 | [RFCXXXX] | 2014-08-22 |
| WebRTC Binary Empty | 57 | [RFCXXXX] | 2014-08-22 | | WebRTC Binary Empty | 57 | [RFCXXXX] | 2014-08-22 |
+-------------------------------+----------+-----------+------------+ +-------------------------------+----------+-----------+------------+
9. Acknowledgments 9. Acknowledgments
Many thanks for comments, ideas, and text from Harald Alvestrand, Many thanks for comments, ideas, and text from Harald Alvestrand,
Richard Barnes, Adam Bergkvist, Gunnar Hellstrom, Christer Holmberg, Richard Barnes, Adam Bergkvist, Alissa Cooper, Benoit Claise, Spencer
Cullen Jennings, Paul Kyzivat, Eric Rescorla, Irene Ruengeler, Dawkins, Gunnar Hellstrom, Christer Holmberg, Cullen Jennings, Paul
Randall Stewart, Justin Uberti, and Magnus Westerlund. Kyzivat, Eric Rescorla, Adam Roach, Irene Ruengeler, Randall Stewart,
Martin Stiemerling, 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 14, line 39 skipping to change at page 14, line 47
[RFC5061] Stewart, R., Xie, Q., Tuexen, M., Maruyama, S., and M. [RFC5061] Stewart, R., Xie, Q., Tuexen, M., Maruyama, S., and M.
Kozuka, "Stream Control Transmission Protocol (SCTP) Kozuka, "Stream Control Transmission Protocol (SCTP)
Dynamic Address Reconfiguration", RFC 5061, September Dynamic Address Reconfiguration", RFC 5061, September
2007. 2007.
[RFC5245] Rosenberg, J., "Interactive Connectivity Establishment [RFC5245] Rosenberg, J., "Interactive Connectivity Establishment
(ICE): A Protocol for Network Address Translator (NAT) (ICE): A Protocol for Network Address Translator (NAT)
Traversal for Offer/Answer Protocols", RFC 5245, April Traversal for Offer/Answer Protocols", RFC 5245, April
2010. 2010.
[RFC6347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer
Security Version 1.2", RFC 6347, January 2012.
[RFC6525] Stewart, R., Tuexen, M., and P. Lei, "Stream Control [RFC6525] Stewart, R., Tuexen, M., and P. Lei, "Stream Control
Transmission Protocol (SCTP) Stream Reconfiguration", RFC Transmission Protocol (SCTP) Stream Reconfiguration", RFC
6525, February 2012. 6525, February 2012.
[I-D.ietf-tsvwg-sctp-ndata] [I-D.ietf-tsvwg-sctp-ndata]
Stewart, R., Tuexen, M., Loreto, S., and R. Seggelmann, Stewart, R., Tuexen, M., Loreto, S., and R. Seggelmann,
"Stream Schedulers and a New Data Chunk for the Stream "Stream Schedulers and a New Data Chunk for the Stream
Control Transmission Protocol", draft-ietf-tsvwg-sctp- Control Transmission Protocol", draft-ietf-tsvwg-sctp-
ndata-01 (work in progress), July 2014. ndata-01 (work in progress), July 2014.
[I-D.ietf-rtcweb-data-protocol] [I-D.ietf-rtcweb-data-protocol]
Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel
Establishment Protocol", draft-ietf-rtcweb-data- Establishment Protocol", draft-ietf-rtcweb-data-
protocol-07 (work in progress), July 2014. protocol-08 (work in progress), September 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-05 (work in progress), July 2014. dtls-encaps-07 (work in progress), December 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-07 (work in progress), July 2014. ietf-rtcweb-security-07 (work in progress), July 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-10 (work in progress), July 2014. rtcweb-security-arch-10 (work in progress), July 2014.
[I-D.ietf-rtcweb-jsep] [I-D.ietf-rtcweb-jsep]
Uberti, J., Jennings, C., and E. Rescorla, "Javascript Uberti, J., Jennings, C., and E. Rescorla, "Javascript
Session Establishment Protocol", draft-ietf-rtcweb-jsep-07 Session Establishment Protocol", draft-ietf-rtcweb-jsep-08
(work in progress), July 2014. (work in progress), October 2014.
[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-03 (work in progress), May 2014. tsvwg-sctp-prpolicies-06 (work in progress), December
2014.
[I-D.ietf-mmusic-sctp-sdp] [I-D.ietf-mmusic-sctp-sdp]
Loreto, S. and G. Camarillo, "Stream Control Transmission Holmberg, C., Loreto, S., and G. Camarillo, "Stream
Protocol (SCTP)-Based Media Transport in the Session Control Transmission Protocol (SCTP)-Based Media Transport
Description Protocol (SDP)", draft-ietf-mmusic-sctp-sdp-07 in the Session Description Protocol (SDP)", draft-ietf-
(work in progress), July 2014. mmusic-sctp-sdp-11 (work in progress), December 2014.
10.2. Informative References 10.2. Informative References
[RFC1122] Braden, R., "Requirements for Internet Hosts - [RFC1122] Braden, R., "Requirements for Internet Hosts -
Communication Layers", STD 3, RFC 1122, October 1989. Communication Layers", STD 3, RFC 1122, October 1989.
[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]
Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-
Time Communication Use-cases and Requirements", draft-
ietf-rtcweb-use-cases-and-requirements-14 (work in
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
Salvatore Loreto Salvatore Loreto
Ericsson Ericsson
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