RTCWeb Working Group R. Jesup
Internet-Draft Mozilla
Intended status: Standards Track S. Loreto
Expires: April 24, 2014 Ericsson
M. Tuexen
Muenster Univ. of Appl. Sciences
October 21, 2013
RTCWeb Data Channels
draft-ietf-rtcweb-data-channel-06.txt
Abstract
The Real-Time Communication in WEB-browsers (RTCWeb) 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 RTCWeb framework. It provides an architectural
overview of how the Stream Control Transmission Protocol (SCTP) is
used in the RTCWeb 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 April 24, 2014.
Copyright Notice
Copyright (c) 2013 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 April 24, 2014 [Page 1]
Internet-Draft RTCWeb Data Channels October 2013
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 . . . . . . . . . . 3
4. Requirements . . . . . . . . . . . . . . . . . . . . . . . . 4
5. SCTP over DTLS over UDP Considerations . . . . . . . . . . . 5
6. The Usage of SCTP in the RTCWeb Context . . . . . . . . . . . 8
6.1. SCTP Protocol Considerations . . . . . . . . . . . . . . 8
6.2. Association Setup . . . . . . . . . . . . . . . . . . . . 9
6.3. SCTP Streams . . . . . . . . . . . . . . . . . . . . . . 9
6.4. Channel Definition . . . . . . . . . . . . . . . . . . . 10
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-media data types in the context of RTCWeb are handled by using
SCTP [RFC4960] encapsulated in DTLS [RFC6347].
+----------+
| SCTP |
+----------+
| DTLS |
+----------+
| ICE/UDP |
+----------+
Figure 1: Basic stack diagram
Jesup, et al. Expires April 24, 2014 [Page 2]
Internet-Draft RTCWeb Data Channels October 2013
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 RTCWeb 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.
3.2. Use Cases for Reliable Data Channels
Jesup, et al. Expires April 24, 2014 [Page 3]
Internet-Draft RTCWeb Data Channels October 2013
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 set of media streams in 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, and the number and state (active/inactive) of the
media streams may change at any time.
Req. 2: Both reliable and unreliable data channels MUST be
supported.
Req. 3: Data channels MUST be congestion controlled; either
individually, as a class, or in conjunction with the media
streams, to ensure that data channels don't cause congestion
problems for the media streams, and that the RTCWeb 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. [ TBD: how this is encoded and
what the impact of this is. ] This will interact with the
congestion control algorithms.
Jesup, et al. Expires April 24, 2014 [Page 4]
Internet-Draft RTCWeb Data Channels October 2013
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 RTCWeb 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.
o Support of PMTU-discovery.
Jesup, et al. Expires April 24, 2014 [Page 5]
Internet-Draft RTCWeb Data Channels October 2013
o Support of reliable or partially reliable message transport.
SCTP multihoming will not be used in RTCWeb. 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 NAT traversal in IPv4 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 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 April 24, 2014 [Page 6]
Internet-Draft RTCWeb Data Channels October 2013
+------+
|RTCWEB|
| 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.
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].
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.
Incoming ICMP or ICMPv6 messages can't be processed by the SCTP
layer, since there is no way to identify the corresponding
Jesup, et al. Expires April 24, 2014 [Page 7]
Internet-Draft RTCWeb Data Channels October 2013
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. Taking an overhead of 20 bytes for IPv4, 40 bytes for IPv6, 8
bytes for UDP, 13 + X for DTLS and 28 bytes for SCTP into account,
this results in an SCTP payload of 1131 - X when IPv4 is used and
1192 - X bytes when IPv6 is used.
In general, the lower layer interface of an SCTP implementation
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 protocol stack 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, alternate congestion
controls SHOULD only require a different sender side behavior using
existing information carried in the association.
6. The Usage of SCTP in the RTCWeb 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.
The following SCTP protocol extensions are required:
Jesup, et al. Expires April 24, 2014 [Page 8]
Internet-Draft RTCWeb Data Channels October 2013
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] MUST NOT be
used.
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.tuexen-tsvwg-sctp-prpolicies] MUST be supported.
Once support for message interleaving as currently being discussed in
[I-D.stewart-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 SDP;
typically this will be shared via BUNDLE or equivalent with DTLS
connections used to key the DTLS-SRTP media streams.
The application SHOULD indicate the initial number of streams
required when opening the association, and if no value is supplied,
the implementation SHOULD provide an appropriate default. If more
simultaneous streams are needed, [RFC6525] allows adding additional
(but not removing) streams to an existing association. Note there
can be up to 65536 SCTP streams per SCTP association in each
direction.
6.3. SCTP Streams
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.
Jesup, et al. Expires April 24, 2014 [Page 9]
Internet-Draft RTCWeb Data Channels October 2013
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. 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.
Note that there's no requirement for the SCTP streams used to create
a bidirectional channel have the same number in each direction. 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.
Jesup, et al. Expires April 24, 2014 [Page 10]
Internet-Draft RTCWeb Data Channels October 2013
The SCTP Payload Protocol Identifiers (PPIDs) are used to signal the
interpretation of the "Payload data". For identifying a JavaScript
string the PPID "DOMString Last" MUST be used, for JavaScript binary
data (ArrayBuffer or Blob) the PPID "Binary Data Last" MUST be used
(see Section 8).
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.stewart-tsvwg-sctp-ndata] defines an extension to
support message interleaving. Once such an extension is available,
it SHOULD be used.
As long as message interleaving is not supported, the sending
application SHOULD fragment large user messages for reliable and
ordered data channels. For sending large JavaScript strings, it uses
the PPID "DOMString Partial" for all but the last fragments and the
PPID "DOMString Last" for the last one. For JavaScript binary data
the PPIDs "Binary Data Partial" and "Binary Data Last" are used. The
reassembly based on the PPID MUST be supported. For data channel
which are not reliable and ordered, the sender MAY limit the maximum
message size to avoid monopolization.
It is recommended that message size be kept within certain size
bounds (TBD) as applications will not be able to support arbitrarily-
large single messages.
The sender MAY 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].
Jesup, et al. Expires April 24, 2014 [Page 11]
Internet-Draft RTCWeb Data Channels October 2013
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). [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. Therefore these four assignments should
be updated to read:
+---------------------+-----------+-----------+
| Value | SCTP PPID | Reference |
+---------------------+-----------+-----------+
| DOMString Last | 51 | [RFCXXXX] |
| Binary Data Partial | 52 | [RFCXXXX] |
| Binary Data Last | 53 | [RFCXXXX] |
| DOMString Partial | 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.
Jesup, et al. Expires April 24, 2014 [Page 12]
Internet-Draft RTCWeb Data Channels October 2013
[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.stewart-tsvwg-sctp-ndata]
Stewart, R., Tuexen, M., Loreto, S., and R. Seggelmann, "A
New Data Chunk for Stream Control Transmission Protocol",
draft-stewart-tsvwg-sctp-ndata-03 (work in progress),
October 2013.
[I-D.ietf-rtcweb-data-protocol]
Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel
Protocol", draft-ietf-rtcweb-data-protocol-00 (work in
progress), July 2013.
[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-02 (work in progress), October 2013.
[I-D.ietf-rtcweb-security]
Rescorla, E., "Security Considerations for WebRTC", draft-
ietf-rtcweb-security-05 (work in progress), July 2013.
[I-D.ietf-rtcweb-security-arch]
Rescorla, E., "WebRTC Security Architecture", draft-ietf-
rtcweb-security-arch-07 (work in progress), July 2013.
[I-D.ietf-rtcweb-jsep]
Jesup, et al. Expires April 24, 2014 [Page 13]
Internet-Draft RTCWeb Data Channels October 2013
Uberti, J. and C. Jennings, "Javascript Session
Establishment Protocol", draft-ietf-rtcweb-jsep-04 (work
in progress), September 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.
10.2. Informative References
[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-12 (work in
progress), October 2013.
[I-D.tuexen-tsvwg-sctp-prpolicies]
Tuexen, M., Seggelmann, R., Stewart, R., and S. Loreto,
"Additional Policies for the Partial Delivery Extension of
the Stream Control Transmission Protocol", draft-tuexen-
tsvwg-sctp-prpolicies-03 (work in progress), October 2013.
Authors' Addresses
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 April 24, 2014 [Page 14]
Internet-Draft RTCWeb Data Channels October 2013
Michael Tuexen
Muenster University of Applied Sciences
Stegerwaldstrasse 39
Steinfurt 48565
DE
Email: tuexen@fh-muenster.de
Jesup, et al. Expires April 24, 2014 [Page 15]
