Network Working Group                                           R. Jesup
Internet-Draft                                                   Mozilla
Intended status: Standards Track                               S. Loreto
Expires: April 24, August 15, 2014                                        Ericsson
                                                               M. Tuexen
                                        Muenster Univ. of Appl. Sciences
                                                        October 21, 2013

                                                       February 11, 2014

                          WebRTC Data Channels


   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. web-browsers.  This
   document specifies the non-media data transport aspects of the RTCWeb WebRTC
   framework.  It provides an architectural overview of how the Stream
   Control Transmission Protocol (SCTP) is used in the RTCWeb 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
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   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on April 24, August 15, 2014.

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   Copyright (c) 2013 2014 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

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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
   4.  Requirements  . . . . . . . . . . . . . . . . . . . . . . . .   4
   5.  SCTP over DTLS over UDP Considerations  . . . . . . . . . . .   5
   6.  The Usage of SCTP in the RTCWeb WebRTC Context . . . . . . . . . . .   8
     6.1.  SCTP Protocol Considerations  . . . . . . . . . . . . . .   8
     6.2.  Association Setup . . . . . . . . . . . . . . . . . . . .   9
     6.3.  SCTP Streams  . . . . . . . . . . . . . . . . . . . . . .   9
     6.4.  Channel Definition  . . . . . . . . . . . . . . . . . . .  10   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  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 RTCWeb WebRTC are handled by using
   SCTP [RFC4960] encapsulated in DTLS [RFC6347].

   |   SCTP   |
   |   DTLS   |
   | ICE/UDP  |

                       Figure 1: Basic stack diagram

   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

   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 WebRTC protocol framework for transporting non-
   media data between WEB-browsers.

2.  Conventions

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   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

   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 configuration 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 Internet filtering or

4.  Requirements

   This section lists the requirements for P2P data channels between two

   Req. 1:   Multiple simultaneous data channels MUST be supported.
             Note that there may 0 or more media streams in parallel
             with the data channels, channels in the same PeerConnection, and the
             number and state (active/inactive) of the these media streams
             may change at any time.

   Req. 2:   Both reliable and unreliable data channels MUST be

   Req. 3:   Data channels of a PeerConnection MUST be congestion
             controlled; either individually, as a class, or in
             conjunction with the media
      streams, streams of the PeerConnection,
             to ensure that data channels don't cause congestion
             problems for the these media streams, and that the RTCWeb 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.  [ TBD: how this is encoded and
      what the impact of this is. ]  This will
             interact with the congestion control algorithms.

   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

   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; 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 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.

   o  Support of PMTU-discovery.

   o  Support of reliable or partially reliable message transport.

   SCTP multihoming will not be used in RTCWeb. 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 NAT 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.

                 | 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. channels of the

   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.

   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.  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 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, algorithm yet, alternate
   congestion controls SHOULD either only require a different sender
   side behavior using existing information carried in the association. association
   or need also specify a negotiation of of a congestion control

6.  The Usage of SCTP in the RTCWeb 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.

   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] MUST NOT are not REQUIRED
      to be
      used. 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 SDP; ICE;
   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 negotiated during SCTP association setup SHOULD
   be 65535, which is supplied, the implementation SHOULD provide an appropriate default.  If more
   simultaneous streams are needed, [RFC6525] allows adding additional
   (but not removing) maximum number of streams to an existing association.  Note there that can be up to 65536 SCTP streams per SCTP negotiated
   during the association in each
   direction. 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
   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 stream having the same number in each direction. stream
   SCTP identifier.

   How stream values are selected is protocol and implementation

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. 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. streams.  If it attempts to re-use
   a Stream stream which is part of an existing Channel, the addition SHOULD
   fail.  In addition to choosing a Stream, 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 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 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 "DOMString Last" "WebRTC String" MUST be used, for
   JavaScript binary data (ArrayBuffer or Blob) the PPID "Binary Data Last" "WebRTC Binary"
   MUST be used (see Section 8).

   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.stewart-tsvwg-sctp-ndata] [I-D.ietf-tsvwg-sctp-ndata]  defines an extension to
   support message interleaving.  Once such an this extension is available, it SHOULD
   MUST 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 SHOULD limit the maximum message size to 16 KB to avoid

   It is recommended that message size be kept within certain size
   bounds (TBD) as applications will not be able to support arbitrarily-
   large arbitrarily-large
   single messages.  This limit has to be negotiated, for example by
   using [I-D.ietf-mmusic-sctp-sdp].

   The sender MAY SHOULD disable the Nagle algorithm to minimize the

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

8.  IANA Considerations

   [NOTE to RFC-Editor:

      "RFCXXXX" is to be replaced by the RFC number you assign this

   This document uses four already registered SCTP Payload Protocol
   Identifiers (PPIDs). (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. document and change the names of the
   PPIDs.  Therefore these four assignments should be updated to read:


      | Value                              | SCTP PPID | Reference |
      | DOMString Last WebRTC String                      | 51        | [RFCXXXX] |
      | WebRTC Binary Data Partial (Deprecated) | 52        | [RFCXXXX] |
      | WebRTC Binary Data Last                      | 53        | [RFCXXXX] |
      | DOMString 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

   [RFC5245]  Rosenberg, J., "Interactive Connectivity Establishment
              (ICE): A Protocol for Network Address Translator (NAT)
              Traversal for Offer/Answer Protocols", RFC 5245, April

   [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.


              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),
              October 2013.
              February 2014.

              Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel
              Protocol", draft-ietf-rtcweb-data-protocol-00 draft-ietf-rtcweb-data-protocol-02 (work in
              progress), July 2013. February 2014.

              Tuexen, M., Stewart, R., Jesup, R., and S. Loreto, "DTLS
              Encapsulation of SCTP Packets", draft-ietf-tsvwg-sctp-
              dtls-encaps-03 (work in progress), October 2013. February 2014.

              Rescorla, E., "Security Considerations for WebRTC", draft-
              ietf-rtcweb-security-06 (work in progress), July 2013. January 2014.

              Rescorla, E., "WebRTC Security Architecture", draft-ietf-
              rtcweb-security-arch-08 (work in progress), July 2013. January 2014.

              Uberti, J. and C. Jennings, "Javascript Session
              Establishment Protocol", draft-ietf-rtcweb-jsep-04 draft-ietf-rtcweb-jsep-05 (work
              in progress), September October 2013.

              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.

              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.

              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.

              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), October 2013.

              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. February 2014.

Authors' Addresses

   Randell Jesup

   Email: randell-ietf@jesup.org

   Salvatore Loreto
   Hirsalantie 11
   Jorvas  02420

   Email: salvatore.loreto@ericsson.com
   Michael Tuexen
   Muenster University of Applied Sciences
   Stegerwaldstrasse 39
   Steinfurt  48565

   Email: tuexen@fh-muenster.de