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In: MissingRef
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



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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  . . . . . . . . . .   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





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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 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





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   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.







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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 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.




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   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.

















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                                        +------+
                                        |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



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   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:



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   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.









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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.




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   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].





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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).  [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.





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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.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]



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              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




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   Michael Tuexen
   Muenster University of Applied Sciences
   Stegerwaldstrasse 39
   Steinfurt  48565
   DE

   Email: tuexen@fh-muenster.de












































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