Network Working Group R. Jesup
Internet-Draft Mozilla
Intended status: Standards Track S. Loreto
Expires: August 15, 2014 Ericsson
M. Tuexen
Muenster Univ. of Appl. Sciences
February 11, 2014
WebRTC Data Channels
draft-ietf-rtcweb-data-channel-07.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-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 August 15, 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
Jesup, et al. Expires August 15, 2014 [Page 1]
Internet-Draft WebRTC Data Channels February 2014
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 in the WebRTC Context . . . . . . . . . . . 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 . . . . . . . . . . . . . . . . . . . . . 11
9. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 12
10. References . . . . . . . . . . . . . . . . . . . . . . . . . 12
10.1. Normative References . . . . . . . . . . . . . . . . . . 12
10.2. Informative References . . . . . . . . . . . . . . . . . 14
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 14
1. Introduction
Non-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
Jesup, et al. Expires August 15, 2014 [Page 2]
Internet-Draft WebRTC Data Channels February 2014
The encapsulation of SCTP over DTLS (see
[I-D.ietf-tsvwg-sctp-dtls-encaps]) over ICE/UDP (see [RFC5245])
provides a NAT traversal solution together with confidentiality,
source authentication, and integrity protected transfers. This data
transport service operates in parallel to the media transports, and
all of them can eventually share a single transport-layer port
number.
SCTP as specified in [RFC4960] with the partial reliability extension
defined in [RFC3758] provides multiple streams natively with
reliable, and partially-reliable delivery modes 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.
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 arguments SCTP over
DTLS over UDP; Section 6 provides the specification of how SCTP
should be used by the WebRTC protocol framework for transporting non-
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 defined 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 media channels, or all media
channels may be inactive, and that there may also be reliable
data channels in use.
U-C 2: Providing non-critical information to a user about the reason
for a state update in a video chat or conference, such as
mute state.
Jesup, et al. Expires August 15, 2014 [Page 3]
Internet-Draft WebRTC Data Channels February 2014
3.2. Use Cases for Reliable Data Channels
U-C 3: A real-time game where critical state information needs to be
transferred, such as control information. Such a game may
have no media channels, or they may be inactive at any given
time, or may only be added due to in-game actions.
U-C 4: Non-realtime file transfers between people chatting. Note
that this may involve a large number of files to transfer
sequentially or in parallel, such as when sharing a folder of
images or a directory of files.
U-C 5: Realtime text chat 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 0 or more media streams in parallel
with the data channels in the same PeerConnection, and the
number and state (active/inactive) of these 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 media streams of the PeerConnection,
to ensure that data channels don't cause congestion
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
the relative priority of each data channel relative to each
other, and relative to the media streams. This will
interact with the congestion control algorithms.
Jesup, et al. Expires August 15, 2014 [Page 4]
Internet-Draft WebRTC Data Channels February 2014
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 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 message by providing fragmentation
and reassembly.
Jesup, et al. Expires August 15, 2014 [Page 5]
Internet-Draft WebRTC Data Channels February 2014
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 interesting features for transporting non-media data
between browsers:
o Support of multiple unidirectional streams.
o Ordered and unordered delivery of user messages.
o Reliable and partial-reliable transport of user messages.
Each SCTP user message contains a so called Payload Protocol
Identifier (PPID) that is passed to SCTP by its upper layer and sent
to its peer. This value can be used to multiplex multiple protocols
over a single SCTP association. The sender provides for each
protocol a specific PPID and the receiver can demultiplex the
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
listed above satisfies all the requirements listed in Section 4.
The layering of protocols for WebRTC is shown in the following
Figure 2.
Jesup, et al. Expires August 15, 2014 [Page 6]
Internet-Draft WebRTC Data Channels February 2014
+------+
|WEBRTC|
| 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 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 MUST be used, since DTLS does not expose any address
management to its upper layer. The ICE/UDP layer can handle IP
address changes during a session without needing to notify the DTLS
and SCTP layers, though it would be advantageous to retest Path MTU
on an IP address change.
DTLS implementations used for this stack SHOULD support controlling
fields of the IP layer like the Don't Fragment (DF)-bit in case of
IPv4 and the Differentiated Services Code Point (DSCP) field required
for supporting [I-D.ietf-rtcweb-qos]. Being able to set the (DF)-bit
in case of IPv4 is required for performing path MTU discovery. The
DTLS implementation SHOULD also support sending user messages
exceeding the Path MTU.
Jesup, et al. Expires August 15, 2014 [Page 7]
Internet-Draft WebRTC Data Channels February 2014
Incoming ICMP or ICMPv6 messages can't be processed by the SCTP
layer, since there is no way to identify the corresponding
association. Therefore SCTP MUST support performing Path MTU
discovery without relying on ICMP or ICMPv6 as specified in [RFC4821]
using probing messages specified in [RFC4820]. The initial Path MTU
at the IP layer MUST 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 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 the standard one might improve the impact on the
parallel SRTP media streams. Since SCTP does not support the
negotiation of a congestion control algorithm yet, alternate
congestion controls SHOULD either only require a different sender
side behavior using existing information carried in the association
or need also specify a negotiation of of a congestion control
algorithm.
6. The Usage of SCTP in the WebRTC Context
6.1. SCTP Protocol Considerations
The DTLS encapsulation of SCTP packets as described in
[I-D.ietf-tsvwg-sctp-dtls-encaps] MUST be used.
Jesup, et al. Expires August 15, 2014 [Page 8]
Internet-Draft WebRTC Data Channels February 2014
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
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.
Once support for message interleaving as currently being discussed in
[I-D.ietf-tsvwg-sctp-ndata] is available, it SHOULD be supported.
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]. Additionally,
the negotiation SHOULD include some type of congestion control
selection. 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 DTLS-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 one to another SCTP endpoint. The streams
are used to provide the notion of in-sequence delivery and for
multiplexing. Each user message is sent on a particular stream,
either order or unordered. Ordering is preserved only for 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
Jesup, et al. Expires August 15, 2014 [Page 9]
Internet-Draft WebRTC Data Channels February 2014
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. These priorities MUST NOT be
strict priorities.
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 internal or external
negotiation. The details are out of scope of this document.
A simple protocol for internal negotiation is specified in
[I-D.ietf-rtcweb-data-protocol] and MUST be supported.
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
even stream identifiers, the server odd stream identifiers) or done
in a different way. 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
inform the protocol of the options to use for sending messages. The
application MUST ensure in an application-specific manner that the
other side will also inform the protocol that the selected stream is
to be used, and the parameters 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).
Jesup, et al. Expires August 15, 2014 [Page 10]
Internet-Draft WebRTC Data Channels February 2014
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
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. Once this extension is available, it
MUST 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 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]. Resetting a stream set the
Stream Sequence Numbers (SSNs) of the stream back to 'zero' with a
corresponding notification to the application layer that the reset
has been performed. Streams are available to reuse after a reset has
been performed.
[RFC6525] also guarantees that all the messages are delivered (or
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].
8. IANA Considerations
[NOTE to RFC-Editor:
"RFCXXXX" is to be replaced by the RFC number you assign this
document.
]
Jesup, et al. Expires August 15, 2014 [Page 11]
Internet-Draft WebRTC Data Channels February 2014
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, Cullen Jennings, Eric Rescorla, Randall Stewart,
Justin Uberti, and Magnus Westerlund.
10. References
10.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3758] Stewart, R., Ramalho, M., Xie, Q., Tuexen, M., and P.
Conrad, "Stream Control Transmission Protocol (SCTP)
Partial Reliability Extension", RFC 3758, May 2004.
[RFC4820] Tuexen, M., Stewart, R., and P. Lei, "Padding Chunk and
Parameter for the Stream Control Transmission Protocol
(SCTP)", RFC 4820, March 2007.
[RFC4821] Mathis, M. and J. Heffner, "Packetization Layer Path MTU
Discovery", RFC 4821, March 2007.
[RFC4960] Stewart, R., "Stream Control Transmission Protocol", RFC
4960, September 2007.
[RFC5061] Stewart, R., Xie, Q., Tuexen, M., Maruyama, S., and M.
Kozuka, "Stream Control Transmission Protocol (SCTP)
Dynamic Address Reconfiguration", RFC 5061, September
2007.
Jesup, et al. Expires August 15, 2014 [Page 12]
Internet-Draft WebRTC Data Channels February 2014
[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
Protocol", draft-ietf-rtcweb-data-protocol-02 (work in
progress), February 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-03 (work in progress), February 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-08 (work in progress), January 2014.
[I-D.ietf-rtcweb-jsep]
Uberti, J. and C. Jennings, "Javascript Session
Establishment Protocol", draft-ietf-rtcweb-jsep-05 (work
in progress), October 2013.
[I-D.ietf-rtcweb-qos]
Dhesikan, S., Druta, D., Jones, P., and J. Polk, "DSCP and
other packet markings for RTCWeb QoS", draft-ietf-rtcweb-
qos-00 (work in progress), October 2012.
Jesup, et al. Expires August 15, 2014 [Page 13]
Internet-Draft WebRTC Data Channels 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-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
[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-13 (work in
progress), February 2014.
Authors' Addresses
Randell Jesup
Mozilla
US
Email: randell-ietf@jesup.org
Salvatore Loreto
Ericsson
Hirsalantie 11
Jorvas 02420
FI
Email: salvatore.loreto@ericsson.com
Jesup, et al. Expires August 15, 2014 [Page 14]
Internet-Draft WebRTC Data Channels February 2014
Michael Tuexen
Muenster University of Applied Sciences
Stegerwaldstrasse 39
Steinfurt 48565
DE
Email: tuexen@fh-muenster.de
Jesup, et al. Expires August 15, 2014 [Page 15]