Network Working Group R. Jesup
Internet-Draft Mozilla
Intended status: Standards Track S. Loreto
Expires: December 11, 2014 Ericsson
M. Tuexen
Muenster Univ. of Appl. Sciences
June 9, 2014
WebRTC Data Channels
draft-ietf-rtcweb-data-channel-10.txt
Abstract
The Real-Time Communication in WEB-browsers working group is charged
to provide protocol support for direct interactive rich communication
using audio, video, and data between two peers' web-browsers. This
document specifies the non-SRTP media data transport aspects of the
WebRTC framework. It provides an architectural overview of how the
Stream Control Transmission Protocol (SCTP) is used in the WebRTC
context as a generic transport service allowing WEB-browsers to
exchange generic data from peer to peer.
Status of This Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
This Internet-Draft will expire on December 11, 2014.
Copyright Notice
Copyright (c) 2014 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of
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publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Conventions . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Use Cases . . . . . . . . . . . . . . . . . . . . . . . . . . 3
3.1. Use Cases for Unreliable Data Channels . . . . . . . . . 3
3.2. Use Cases for Reliable Data Channels . . . . . . . . . . 4
4. Requirements . . . . . . . . . . . . . . . . . . . . . . . . 4
5. SCTP over DTLS over UDP Considerations . . . . . . . . . . . 5
6. The Usage of SCTP for Data Channels . . . . . . . . . . . . . 8
6.1. SCTP Protocol Considerations . . . . . . . . . . . . . . 8
6.2. Association Setup . . . . . . . . . . . . . . . . . . . . 9
6.3. SCTP Streams . . . . . . . . . . . . . . . . . . . . . . 9
6.4. Channel Definition . . . . . . . . . . . . . . . . . . . 9
6.5. Opening a Channel . . . . . . . . . . . . . . . . . . . . 10
6.6. Transferring User Data on a Channel . . . . . . . . . . . 10
6.7. Closing a Channel . . . . . . . . . . . . . . . . . . . . 11
7. Security Considerations . . . . . . . . . . . . . . . . . . . 11
8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 12
9. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 12
10. References . . . . . . . . . . . . . . . . . . . . . . . . . 12
10.1. Normative References . . . . . . . . . . . . . . . . . . 12
10.2. Informative References . . . . . . . . . . . . . . . . . 14
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 14
1. Introduction
Non-SRTP media data types in the context of WebRTC are handled by
using SCTP [RFC4960] encapsulated in DTLS [RFC6347].
+----------+
| SCTP |
+----------+
| DTLS |
+----------+
| ICE/UDP |
+----------+
Figure 1: Basic stack diagram
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The encapsulation of SCTP over DTLS (see
[I-D.ietf-tsvwg-sctp-dtls-encaps]) over ICE/UDP (see [RFC5245])
provides a NAT traversal solution together with confidentiality,
source authentication, and integrity protected transfers. This data
transport service operates in parallel to the SRTP media transports,
and all of them can eventually share a single transport-layer port
number.
SCTP as specified in [RFC4960] with the partial reliability extension
defined in [RFC3758] and the additional policies defined in
[I-D.ietf-tsvwg-sctp-prpolicies] provides multiple streams natively
with reliable, and the relevant partially-reliable delivery modes for
user messages. Using the reconfiguration extension defined in
[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
Section 4 provide use cases and requirements for both unreliable and
reliable peer to peer data channels; Section 5 discusses SCTP over
DTLS over UDP; Section 6 provides the specification of how SCTP
should be used by the WebRTC protocol framework for transporting non-
SRTP media data between WEB-browsers.
2. Conventions
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in [RFC2119].
3. Use Cases
This section defines use cases specific to data channels. For
general use cases see [I-D.ietf-rtcweb-use-cases-and-requirements].
3.1. Use Cases for Unreliable Data Channels
U-C 1: A real-time game where position and object state information
is sent via one or more unreliable data channels. Note that
at any time there may be no SRTP media channels, or all SRTP
media channels may be inactive, and that there may also be
reliable data channels in use.
U-C 2: Providing non-critical information to a user about the reason
for a state update in a video chat or conference, such as
mute state.
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3.2. Use Cases for Reliable Data Channels
U-C 3: A real-time game where critical state information needs to be
transferred, such as control information. Such a game may
have no SRTP media channels, or they may be inactive at any
given time, or may only be added due to in-game actions.
U-C 4: Non-realtime file transfers between people chatting. Note
that this may involve a large number of files to transfer
sequentially or in parallel, such as when sharing a folder of
images or a directory of files.
U-C 5: Realtime text chat during an audio and/or video call with an
individual or with multiple people in a conference.
U-C 6: Renegotiation of the configuration of the PeerConnection.
U-C 7: Proxy browsing, where a browser uses data channels of a
PeerConnection to send and receive HTTP/HTTPS requests and
data, for example to avoid local Internet filtering or
monitoring.
4. Requirements
This section lists the requirements for P2P data channels between two
browsers.
Req. 1: Multiple simultaneous data channels MUST be supported.
Note that there may be 0 or more SRTP media streams in
parallel with the data channels in the same PeerConnection,
and the number and state (active/inactive) of these SRTP
media streams may change at any time.
Req. 2: Both reliable and unreliable data channels MUST be
supported.
Req. 3: Data channels of a PeerConnection MUST be congestion
controlled; either individually, as a class, or in
conjunction with the SRTP media streams of the
PeerConnection, to ensure that data channels don't cause
congestion problems for these SRTP 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
the relative priority of each data channel relative to each
other, and relative to the SRTP media streams. This will
interact with the congestion control algorithms.
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Req. 5: Data channels MUST be secured; allowing for
confidentiality, integrity and source authentication. See
[I-D.ietf-rtcweb-security] and
[I-D.ietf-rtcweb-security-arch] for detailed info.
Req. 6: Data channels MUST provide message fragmentation support
such that IP-layer fragmentation can be avoided no matter
how large a message the JavaScript application passes to be
sent. It also MUST ensure that large data channel
transfers don't unduly delay traffic on other data
channels.
Req. 7: The data channel transport protocol MUST NOT encode local
IP addresses inside its protocol fields; doing so reveals
potentially private information, and leads to failure if
the address is depended upon.
Req. 8: The data channel transport protocol SHOULD support
unbounded-length "messages" (i.e., a virtual socket stream)
at the application layer, for such things as image-file-
transfer; Implementations might enforce a reasonable
message size limit.
Req. 9: The data channel transport protocol SHOULD avoid IP
fragmentation. It MUST support PMTU (Path MTU) discovery
and MUST NOT rely on ICMP or ICMPv6 being generated or
being passed back, especially for PMTU discovery.
Req. 10: It MUST be possible to implement the protocol stack in the
user application space.
5. SCTP over DTLS over UDP Considerations
The important features of SCTP in the WebRTC context are:
o Usage of a TCP-friendly congestion control.
o The congestion control is modifiable for integration with the SRTP
media stream congestion control.
o Support of multiple unidirectional streams, each providing its own
notion of ordered message delivery.
o Support of ordered and out-of-order message delivery.
o Supporting arbitrary large user messages by providing
fragmentation and reassembly.
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o Support of PMTU-discovery.
o Support of reliable or partially reliable message transport.
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
is the abstraction that the lower layer (a connection oriented,
unreliable datagram service) exposes.
The encapsulation of SCTP over DTLS defined in
[I-D.ietf-tsvwg-sctp-dtls-encaps] provides confidentiality, source
authenticated, and integrity protected transfers. Using DTLS over
UDP in combination with ICE enables middlebox traversal in IPv4 and
IPv6 based networks. SCTP as specified in [RFC4960] MUST be used in
combination with the extension defined in [RFC3758] and provides the
following features for transporting non-SRTP media data between
browsers:
o Support of multiple unidirectional streams.
o Ordered and unordered delivery of user messages.
o Reliable and partial-reliable transport of user messages.
Each SCTP user message contains a Payload Protocol Identifier (PPID)
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
used to multiplex/demultiplex multiple upper layers over a single
SCTP association. In the WebRTP context, the PPID is used to
distinguish between UTF-8 encoded user data, binary encoded userdata
and the Data Channel Establishment Protocol defined in
[I-D.ietf-rtcweb-data-protocol]. Please note that the PPID is not
accessible via the Javascript API.
The encapsulation of SCTP over DTLS, together with the SCTP features
listed above satisfies all the requirements listed in Section 4.
The layering of protocols for WebRTC is shown in the following
Figure 2.
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+------+------+------+
| DCEP | UTF-8|Binary|
| | data | data |
+------+------+------+
| SCTP |
+----------------------------------+
| STUN | SRTP | DTLS |
+----------------------------------+
| ICE |
+----------------------------------+
| UDP1 | UDP2 | ... |
+----------------------------------+
Figure 2: WebRTC protocol layers
This stack (especially in contrast to DTLS over SCTP [RFC6083] in
combination with SCTP over UDP [RFC6951]) has been chosen because it
o supports the transmission of arbitrary large user messages.
o shares the DTLS connection with the SRTP media channels of the
PeerConnection.
o provides privacy for the SCTP control information.
Considering the protocol stack of Figure 2 the usage of DTLS over UDP
is specified in [RFC6347], while the usage of SCTP on top of DTLS is
specified in [I-D.ietf-tsvwg-sctp-dtls-encaps]. 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
needs to be a user-land stack.
When using DTLS as the lower layer, only single homed SCTP
associations are supported, since DTLS does not expose any address
management to its upper layer. The ICE/UDP layer can handle IP
address changes during a session without needing interaction with the
DTLS and SCTP layers. However, SCTP SHOULD be notified when an
address changes has happened. In this case SCTP SHOULD retest the
Path MTU and reset the congestion state to the initial state. In
case of a window based congestion control like the one specified in
[RFC4960], this means setting the congestion window and slow start
threshold to its initial values.
Incoming ICMP or ICMPv6 messages can't be processed by the SCTP
layer, since there is no way to identify the corresponding
association. Therefore SCTP MUST support performing Path MTU
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discovery without relying on ICMP or ICMPv6 as specified in [RFC4821]
using probing messages specified in [RFC4820]. The initial Path MTU
at the IP layer SHOULD NOT exceed 1200 bytes for IPv4 and 1280 for
IPv6.
In general, the lower layer interface of an SCTP implementation
SHOULD be adapted to address the differences between IPv4 and IPv6
(being connection-less) or DTLS (being connection-oriented).
When the protocol stack of Figure 2 is used, DTLS protects the
complete SCTP packet, so it provides confidentiality, integrity and
source authentication of the complete SCTP packet.
This SCTP stack and its upper layer MUST support the usage of
multiple SCTP streams. A user message can be sent ordered or
unordered and with partial or full reliability. The partial
reliability extension MUST support policies to limit
o the transmission and retransmission by time.
o the number of retransmissions.
Limiting the number of retransmissions to zero combined with
unordered delivery provides a UDP-like service where each user
message is sent exactly once and delivered in the order received.
SCTP provides congestion control on a per-association base. This
means that all SCTP streams within a single SCTP association share
the same congestion window. Traffic not being sent over SCTP is not
covered by the SCTP congestion control. Using a congestion control
different from than the standard one might improve the impact on the
parallel SRTP media streams.
6. The Usage of SCTP for Data Channels
6.1. SCTP Protocol Considerations
The DTLS encapsulation of SCTP packets as described in
[I-D.ietf-tsvwg-sctp-dtls-encaps] MUST be used.
The following SCTP protocol extensions are required:
o The stream reset extension defined in [RFC6525] MUST be supported.
It is used for closing channels.
o The dynamic address reconfiguration extension defined in [RFC5061]
MUST be used to signal the support of the stream reset extension
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defined in [RFC6525], other features of [RFC5061] are not REQUIRED
to be implemented.
o The partial reliability extension defined in [RFC3758] MUST be
supported. In addition to the timed reliability PR-SCTP policy
defined in [RFC3758], the limited retransmission policy defined in
[I-D.ietf-tsvwg-sctp-prpolicies] MUST be supported.
The support for message interleaving as defined in
[I-D.ietf-tsvwg-sctp-ndata] SHOULD be used.
6.2. Association Setup
The SCTP association will be set up when the two endpoints of the
WebRTC PeerConnection agree on opening it, as negotiated by JSEP
(typically an exchange of SDP) [I-D.ietf-rtcweb-jsep]. It will use
the DTLS connection selected via ICE; typically this will be shared
via BUNDLE or equivalent with DTLS connections used to key the SRTP
media streams.
The number of streams negotiated during SCTP association setup SHOULD
be 65535, which is the maximum number of streams that can negotiated
during the association setup.
6.3. SCTP Streams
SCTP defines a stream as a unidirectional logical channel existing
within an SCTP association to another SCTP endpoint. The streams are
used to provide the notion of in-sequence delivery and for
multiplexing. Each user message is sent on a particular stream,
either ordered or unordered. Ordering is preserved only for ordered
messages sent on the same stream.
6.4. Channel Definition
The W3C has consensus on defining the application API for WebRTC
DataChannels to be bidirectional. They also consider the notions of
in-sequence, out-of-sequence, reliable and unreliable as properties
of Channels. One strong wish is for the application-level API to be
close to the API for WebSockets, which implies bidirectional streams
of data and waiting for onopen to fire before sending, a textual
label used to identify the meaning of the stream, among other things.
Each data channel also has a priority, which is an 2 byte unsigned
integer value. These priorities MUST be interpreted as weighted-
fair-queuing scheduling priorities per the definition of the
corresponding stream scheduler supporting interleaving in
[I-D.ietf-tsvwg-sctp-ndata]. For use in WebRTC, the values used
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SHOULD be one of 128 ("below normal"), 256 ("normal"), 512 ("high")
or 1024 ("extra high").
The realization of a bidirectional Data Channel is a pair of one
incoming stream and one outgoing SCTP stream having the same stream
SCTP identifier.
How stream values are selected is protocol and implementation
dependent.
6.5. Opening a Channel
Data channels can be opened by using negotiation within the SCTP
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 in-band negotiation is specified in
[I-D.ietf-rtcweb-data-protocol].
When one side wants to open a channel using out-of-band negotiation,
it picks a stream. Unless otherwise defined or negotiated, the
streams are picked based on the DTLS role (the client picks even
stream identifiers, the server odd stream identifiers). However, the
application is responsible for avoiding collisions with existing
streams. If it attempts to re-use a stream which is part of an
existing Channel, the addition SHOULD fail. In addition to choosing
a stream, the application SHOULD also determine the options to use
for sending messages. The application MUST ensure in an application-
specific manner that the application at the peer will also know the
selected stream to be used, and the options for sending data from
that side.
6.6. Transferring User Data on a Channel
All data sent on a Channel in both directions MUST be sent over the
underlying stream using the reliability defined when the Channel was
opened unless the options are changed, or per-message options are
specified by a higher level.
No more than one message should be put into an SCTP user message.
The SCTP Payload Protocol Identifiers (PPIDs) are used to signal the
interpretation of the "Payload data". For identifying a JavaScript
string encoded in UTF-8 the PPID "WebRTC String" MUST be used, for
JavaScript binary data (ArrayBuffer or Blob) the PPID "WebRTC Binary"
MUST be used (see Section 8).
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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.
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
interleaving of user messages. Therefore sending a large user
message can monopolize the SCTP association. To overcome this
limitation, [I-D.ietf-tsvwg-sctp-ndata] defines an extension to
support message interleaving, which SHOULD be used. As long as
message interleaving is not supported, the sender SHOULD limit the
maximum message size to 16 KB to avoid monopolization.
It is recommended that the message size be kept within certain size
bounds as applications will not be able to support arbitrarily-large
single messages. This limit has to be negotiated, for example by
using [I-D.ietf-mmusic-sctp-sdp].
The sender SHOULD disable the Nagle algorithm to minimize the
latency.
6.7. Closing a Channel
Closing of a Data Channel MUST be signaled by resetting 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
corresponding notification to the application layer that the reset
has been performed. Streams are available to reuse after a reset has
been performed.
[RFC6525] also guarantees that all the messages are delivered (or
abandoned) before resetting the stream.
7. Security Considerations
This document does not add any additional considerations to the ones
given in [I-D.ietf-rtcweb-security] and
[I-D.ietf-rtcweb-security-arch].
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8. IANA Considerations
[NOTE to RFC-Editor:
"RFCXXXX" is to be replaced by the RFC number you assign this
document.
]
This document uses four already registered SCTP Payload Protocol
Identifiers (PPIDs): "DOMString Last", "Binary Data Partial", "Binary
Data Last", and "DOMString Partial". [RFC4960] creates the registry
"SCTP Payload Protocol Identifiers" from which these identifiers were
assigned. IANA is requested to update the reference of these four
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 |
+------------------------------------+-----------+-----------+
| WebRTC String | 51 | [RFCXXXX] |
| WebRTC Binary Partial (Deprecated) | 52 | [RFCXXXX] |
| WebRTC Binary | 53 | [RFCXXXX] |
| WebRTC String Partial (Deprecated) | 54 | [RFCXXXX] |
+------------------------------------+-----------+-----------+
9. Acknowledgments
Many thanks for comments, ideas, and text from Harald Alvestrand,
Adam Bergkvist, Christer Holmberg, Cullen Jennings, Paul Kyzivat,
Eric Rescorla, Irene Ruengeler, Randall Stewart, Justin Uberti, and
Magnus Westerlund.
10. References
10.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3758] Stewart, R., Ramalho, M., Xie, Q., Tuexen, M., and P.
Conrad, "Stream Control Transmission Protocol (SCTP)
Partial Reliability Extension", RFC 3758, May 2004.
[RFC4820] Tuexen, M., Stewart, R., and P. Lei, "Padding Chunk and
Parameter for the Stream Control Transmission Protocol
(SCTP)", RFC 4820, March 2007.
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[RFC4821] Mathis, M. and J. Heffner, "Packetization Layer Path MTU
Discovery", RFC 4821, March 2007.
[RFC4960] Stewart, R., "Stream Control Transmission Protocol", RFC
4960, September 2007.
[RFC5061] Stewart, R., Xie, Q., Tuexen, M., Maruyama, S., and M.
Kozuka, "Stream Control Transmission Protocol (SCTP)
Dynamic Address Reconfiguration", RFC 5061, September
2007.
[RFC5245] Rosenberg, J., "Interactive Connectivity Establishment
(ICE): A Protocol for Network Address Translator (NAT)
Traversal for Offer/Answer Protocols", RFC 5245, April
2010.
[RFC6347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer
Security Version 1.2", RFC 6347, January 2012.
[RFC6525] Stewart, R., Tuexen, M., and P. Lei, "Stream Control
Transmission Protocol (SCTP) Stream Reconfiguration", RFC
6525, February 2012.
[I-D.ietf-tsvwg-sctp-ndata]
Stewart, R., Tuexen, M., Loreto, S., and R. Seggelmann, "A
New Data Chunk for Stream Control Transmission Protocol",
draft-ietf-tsvwg-sctp-ndata-00 (work in progress),
February 2014.
[I-D.ietf-rtcweb-data-protocol]
Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel
Establishment Protocol", draft-ietf-rtcweb-data-
protocol-05 (work in progress), May 2014.
[I-D.ietf-tsvwg-sctp-dtls-encaps]
Tuexen, M., Stewart, R., Jesup, R., and S. Loreto, "DTLS
Encapsulation of SCTP Packets", draft-ietf-tsvwg-sctp-
dtls-encaps-04 (work in progress), May 2014.
[I-D.ietf-rtcweb-security]
Rescorla, E., "Security Considerations for WebRTC", draft-
ietf-rtcweb-security-06 (work in progress), January 2014.
[I-D.ietf-rtcweb-security-arch]
Rescorla, E., "WebRTC Security Architecture", draft-ietf-
rtcweb-security-arch-09 (work in progress), February 2014.
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[I-D.ietf-rtcweb-jsep]
Uberti, J. and C. Jennings, "Javascript Session
Establishment Protocol", draft-ietf-rtcweb-jsep-06 (work
in progress), February 2014.
[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-03 (work in progress), May 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-06
(work in progress), February 2014.
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
Transport Layer Security (DTLS) for Stream Control
Transmission Protocol (SCTP)", RFC 6083, January 2011.
[RFC6951] Tuexen, M. and R. Stewart, "UDP Encapsulation of Stream
Control Transmission Protocol (SCTP) Packets for End-Host
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
Randell Jesup
Mozilla
US
Email: randell-ietf@jesup.org
Jesup, et al. Expires December 11, 2014 [Page 14]
Internet-Draft WebRTC Data Channels June 2014
Salvatore Loreto
Ericsson
Hirsalantie 11
Jorvas 02420
FI
Email: salvatore.loreto@ericsson.com
Michael Tuexen
Muenster University of Applied Sciences
Stegerwaldstrasse 39
Steinfurt 48565
DE
Email: tuexen@fh-muenster.de
Jesup, et al. Expires December 11, 2014 [Page 15]