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Versions: (draft-jesup-rtcweb-data) 00 01 02 04 05 06 07 08 09 10 11 12 13 Draft is active
In: MissingRef
Network Working Group                                           R. Jesup
Internet-Draft                                                   Mozilla
Intended status: Standards Track                               S. Loreto
Expires: July 8, 2015                                           Ericsson
                                                               M. Tuexen
                                        Muenster Univ. of Appl. Sciences
                                                         January 4, 2015


                          WebRTC Data Channels
                 draft-ietf-rtcweb-data-channel-13.txt

Abstract

   The WebRTC framework specifies 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 July 8, 2015.

Copyright Notice

   Copyright (c) 2015 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  . . . . . . . . .   4
     3.2.  Use Cases for Reliable Data Channels  . . . . . . . . . .   4
   4.  Requirements  . . . . . . . . . . . . . . . . . . . . . . . .   4
   5.  SCTP over DTLS over UDP Considerations  . . . . . . . . . . .   6
   6.  The Usage of SCTP for Data Channels . . . . . . . . . . . . .   8
     6.1.  SCTP Protocol Considerations  . . . . . . . . . . . . . .   8
     6.2.  SCTP Association Management . . . . . . . . . . . . . . .   9
     6.3.  SCTP Streams  . . . . . . . . . . . . . . . . . . . . . .   9
     6.4.  Data Channel Definition . . . . . . . . . . . . . . . . .  10
     6.5.  Opening a Data Channel  . . . . . . . . . . . . . . . . .  10
     6.6.  Transferring User Data on a Data Channel  . . . . . . . .  11
     6.7.  Closing a Data Channel  . . . . . . . . . . . . . . . . .  12
   7.  Security Considerations . . . . . . . . . . . . . . . . . . .  13
   8.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  13
   9.  Acknowledgments . . . . . . . . . . . . . . . . . . . . . . .  14
   10. References  . . . . . . . . . . . . . . . . . . . . . . . . .  14
     10.1.  Normative References . . . . . . . . . . . . . . . . . .  14
     10.2.  Informative References . . . . . . . . . . . . . . . . .  15
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  16

1.  Introduction

   In the WebRTC framework, communication between the parties consists
   of media (for example audio and video) and non-media data.  Media is
   sent using SRTP, and is not specified further here.  Non-media data
   is handled by using SCTP [RFC4960] encapsulated in DTLS.  DTLS 1.0 is
   defined in [RFC4347] and the present latest version, DTLS 1.2, is
   defined in [RFC6347].











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                               +----------+
                               |   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 SRTP media transports,
   and all of them can eventually share a single UDP port number.

   SCTP as specified in [RFC4960] with the partial reliability extension
   defined in [RFC3758] and the additional policies defined in
   [I-D.ietf-tsvwg-sctp-prpolicies] provides multiple streams natively
   with reliable, and the relevant partially-reliable delivery modes for
   user messages.  Using the reconfiguration extension defined in
   [RFC6525] allows to increase the number of streams during the
   lifetime of an SCTP association and to reset individual SCTP streams.
   Using [I-D.ietf-tsvwg-sctp-ndata] allows to interleave large messages
   to avoid the monopolization and adds the support of prioritizing of
   SCTP streams.

   The remainder of this document is organized as follows: Section 3 and
   Section 4 provide use cases and requirements for both unreliable and
   reliable peer to peer data channels; Section 5 discusses SCTP over
   DTLS over UDP; Section 6 provides the specification of how SCTP
   should be used by the WebRTC protocol framework for transporting non-
   media data between WEB-browsers.

2.  Conventions

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in [RFC2119].

3.  Use Cases

   This section defines use cases specific to data channels.  Please
   note that this section is informational only.






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3.1.  Use Cases for Unreliable Data Channels

   U-C 1:  A real-time game where position and object state information
           is sent via one or more unreliable data channels.  Note that
           at any time there may be no SRTP media channels, or all SRTP
           media channels may be inactive, and that there may also be
           reliable data channels in use.

   U-C 2:  Providing non-critical information to a user about the reason
           for a state update in a video chat or conference, such as
           mute state.

3.2.  Use Cases for Reliable Data Channels

   U-C 3:  A real-time game where critical state information needs to be
           transferred, such as control information.  Such a game may
           have no SRTP media channels, or they may be inactive at any
           given time, or may only be added due to in-game actions.

   U-C 4:  Non-realtime file transfers between people chatting.  Note
           that this may involve a large number of files to transfer
           sequentially or in parallel, such as when sharing a folder of
           images or a directory of files.

   U-C 5:  Realtime text chat during an audio and/or video call with an
           individual or with multiple people in a conference.

   U-C 6:  Renegotiation of the configuration of the PeerConnection.

   U-C 7:  Proxy browsing, where a browser uses data channels of a
           PeerConnection to send and receive HTTP/HTTPS requests and
           data, for example to avoid local Internet filtering or
           monitoring.

4.  Requirements

   This section lists the requirements for P2P data channels between two
   browsers.  Please note that this section is informational only.

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

   Req. 2:   Both reliable and unreliable data channels must be
             supported.




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   Req. 3:   Data channels of a PeerConnection must be congestion
             controlled; either individually, as a class, or in
             conjunction with the SRTP media streams of the
             PeerConnection, to ensure that data channels don't cause
             congestion problems for these SRTP media streams, and that
             the WebRTC PeerConnection does not cause excessive problems
             when run in parallel with TCP connections.

   Req. 4:   The application should be able to provide guidance as to
             the relative priority of each data channel relative to each
             other, and relative to the SRTP media streams.  This will
             interact with the congestion control algorithms.

   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.








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5.  SCTP over DTLS over UDP Considerations

   The important features of SCTP in the WebRTC context are:

   o  Usage of a TCP-friendly congestion control.

   o  The congestion control is modifiable for integration with the SRTP
      media stream congestion control.

   o  Support of multiple unidirectional streams, each providing its own
      notion of ordered message delivery.

   o  Support of ordered and out-of-order message delivery.

   o  Supporting arbitrary large user messages by providing
      fragmentation and reassembly.

   o  Support of PMTU-discovery.

   o  Support of reliable or partially reliable message transport.

   The WebRTC Data Channel mechanism does not support SCTP multihoming.
   The SCTP layer will simply act as if it were running on a single-
   homed host, since that is the abstraction that the DTLS layer (a
   connection oriented, unreliable datagram service) exposes.

   The encapsulation of SCTP over DTLS defined in
   [I-D.ietf-tsvwg-sctp-dtls-encaps] provides confidentiality, source
   authenticated, and integrity protected transfers.  Using DTLS over
   UDP in combination with ICE enables middlebox traversal in IPv4 and
   IPv6 based networks.  SCTP as specified in [RFC4960] MUST be used in
   combination with the extension defined in [RFC3758] and provides the
   following features for transporting non-media data between browsers:

   o  Support of multiple unidirectional streams.

   o  Ordered and unordered delivery of user messages.

   o  Reliable and partial-reliable transport of user messages.

   Each SCTP user message contains a Payload Protocol Identifier (PPID)
   that is passed to SCTP by its upper layer on the sending side and
   provided to its upper layer on the receiving side.  The PPID can be
   used to multiplex/demultiplex multiple upper layers over a single
   SCTP association.  In the WebRTC context, the PPID is used to
   distinguish between UTF-8 encoded user data, binary encoded userdata
   and the Data Channel Establishment Protocol defined in




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   [I-D.ietf-rtcweb-data-protocol].  Please note that the PPID is not
   accessible via the Javascript API.

   The encapsulation of SCTP over DTLS, together with the SCTP features
   listed above satisfies all the requirements listed in Section 4.

   The layering of protocols for WebRTC is shown in the following
   Figure 2.

                                 +------+------+------+
                                 | DCEP | UTF-8|Binary|
                                 |      | data | data |
                                 +------+------+------+
                                 |        SCTP        |
                   +----------------------------------+
                   | STUN | SRTP |        DTLS        |
                   +----------------------------------+
                   |                ICE               |
                   +----------------------------------+
                   | UDP1 | UDP2 | UDP3 | ...         |
                   +----------------------------------+

                     Figure 2: WebRTC protocol layers

   This stack (especially in contrast to DTLS over SCTP [RFC6083] in
   combination with SCTP over UDP [RFC6951]) has been chosen because it

   o  supports the transmission of arbitrary large user messages.

   o  shares the DTLS connection with the SRTP media channels of the
      PeerConnection.

   o  provides privacy for the SCTP control information.

   Considering the protocol stack of Figure 2 the usage of DTLS 1.0 over
   UDP is specified in [RFC4347] and the usage of DTLS 1.2 over UDP in
   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 application space, the
   SCTP stack also needs to be a user application space stack.

   The ICE/UDP layer can handle IP address changes during a session
   without needing interaction with the DTLS and SCTP layers.  However,
   SCTP SHOULD be notified when an address changes has happened.  In
   this case SCTP SHOULD retest the Path MTU and reset the congestion



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   state to the initial state.  In case of a window based congestion
   control like the one specified in [RFC4960], this means setting the
   congestion window and slow start threshold to its initial values.

   Incoming ICMP or ICMPv6 messages can't be processed by the SCTP
   layer, since there is no way to identify the corresponding
   association.  Therefore SCTP MUST support performing Path MTU
   discovery without relying on ICMP or ICMPv6 as specified in [RFC4821]
   using probing messages specified in [RFC4820].  The initial Path MTU
   at the IP layer SHOULD NOT exceed 1200 bytes for IPv4 and 1280 for
   IPv6.

   In general, the lower layer interface of an SCTP implementation
   should be adapted to address the differences between IPv4 and IPv6
   (being connection-less) or DTLS (being connection-oriented).

   When the protocol stack of Figure 2 is used, DTLS protects the
   complete SCTP packet, so it provides confidentiality, integrity and
   source authentication of the complete SCTP packet.

   SCTP provides congestion control on a per-association base.  This
   means that all SCTP streams within a single SCTP association share
   the same congestion window.  Traffic not being sent over SCTP is not
   covered by the SCTP congestion control.  Using a congestion control
   different from than the standard one might improve the impact on the
   parallel SRTP media streams.

   SCTP uses the same port number concept as TCP and UDP do.  Therefore
   an SCTP association uses two port numbers, one at each SCTP end-
   point.

6.  The Usage of SCTP for Data Channels

6.1.  SCTP Protocol Considerations

   The DTLS encapsulation of SCTP packets as described in
   [I-D.ietf-tsvwg-sctp-dtls-encaps] MUST be used.

   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 following SCTP protocol extensions are required:

   o  The stream reconfiguration extension defined in [RFC6525] MUST be
      supported.  It is used for closing channels.





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   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 OPTIONAL.

   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.  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.

   The support for message interleaving as defined in
   [I-D.ietf-tsvwg-sctp-ndata] SHOULD be used.

6.2.  SCTP Association Management

   In the WebRTC context, the SCTP association will be set up when the
   two endpoints of the WebRTC PeerConnection agree on opening it, as
   negotiated by JSEP (typically an exchange of SDP)
   [I-D.ietf-rtcweb-jsep].  It will use the DTLS connection selected via
   ICE; typically this will be shared via BUNDLE or equivalent with DTLS
   connections used to key the SRTP media streams.

   The number of streams negotiated during SCTP association setup SHOULD
   be 65535, which is the maximum number of streams that can be
   negotiated during the association setup.

   SCTP supports two ways of terminating an SCTP association.  A
   graceful one, using a procedure which ensures that no messages are
   lost during the shutdown of the association.  The second method is a
   non-graceful one, where one side can just abort the association.

   Each SCTP end-point supervises continuously the reachability of its
   peer by monitoring the number of retransmissions of user messages and
   test messages.  In case of excessive retransmissions, the association
   is terminated in a non-graceful way.

   If an SCTP association is closed in a graceful way, all of its data
   channels are closed.  In case of a non-graceful teardown, all data
   channels are also closed, but an error indication SHOULD be provided
   if possible.

6.3.  SCTP Streams

   SCTP defines a stream as a unidirectional logical channel existing
   within an SCTP association to another SCTP endpoint.  The streams are
   used to provide the notion of in-sequence delivery and for



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   multiplexing.  Each user message is sent on a particular stream,
   either ordered or unordered.  Ordering is preserved only for ordered
   messages sent on the same stream.

6.4.  Data Channel Definition

   Data channels are defined such that their accompanying application-
   level API can closely mirror the API for WebSockets, which implies
   bidirectional streams of data and a textual field called 'label' used
   to identify the meaning of the data channel.

   The realization of a data channel is a pair of one incoming stream
   and one outgoing SCTP stream having the same SCTP stream identifier.
   How these SCTP stream identifiers are selected is protocol and
   implementation dependent.  This allows a bidirectional communication.

   Additionally, each data channel has the following properties in each
   direction:

   o  reliable or unreliable message transmission.  In case of
      unreliable transmissions, the same level of unreliability is used.
      Please note that in SCTP this is a property of an SCTP user
      message and not of an SCTP stream.

   o  in-order or out-of-order message delivery for message sent.
      Please note that in SCTP this is a property of an SCTP user
      message and not of an SCTP stream.

   o  A priority, which is a 2 byte unsigned integer.  These priorities
      MUST be interpreted as weighted-fair-queuing scheduling priorities
      per the definition of the corresponding stream scheduler
      supporting interleaving in [I-D.ietf-tsvwg-sctp-ndata].  For use
      in WebRTC, the values used SHOULD be one of 128 ("below normal"),
      256 ("normal"), 512 ("high") or 1024 ("extra high").

   o  an optional label.

   o  an optional protocol.

   Please note that for a data channel being negotiated with the
   protocol specified in [I-D.ietf-rtcweb-data-protocol] all of the
   above properties are the same in both directions.

6.5.  Opening a Data Channel

   Data channels can be opened by using negotiation within the SCTP
   association, called in-band negotiation, or out-of-band negotiation.
   Out-of-band negotiation is defined as any method which results in an



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   agreement as to the parameters of a channel and the creation thereof.
   The details are out of scope of this document.  Applications using
   data channels need to use the negotiation methods consistently on
   both end-points.

   A simple protocol for in-band negotiation is specified in
   [I-D.ietf-rtcweb-data-protocol].

   When one side wants to open a channel using out-of-band negotiation,
   it picks a stream.  Unless otherwise defined or negotiated, the
   streams are picked based on the DTLS role (the client picks even
   stream identifiers, the server odd stream identifiers).  However, the
   application is responsible for avoiding collisions with existing
   streams.  If it attempts to re-use a stream which is part of an
   existing data channel, the addition MUST fail.  In addition to
   choosing a stream, the application SHOULD also determine the options
   to use for sending messages.  The application MUST ensure in an
   application-specific manner that the application at the peer will
   also know the selected stream to be used, and the options for sending
   data from that side.

6.6.  Transferring User Data on a Data Channel

   All data sent on a data channel in both directions MUST be sent over
   the underlying stream using the reliability defined when the data
   channel was opened unless the options are changed, or per-message
   options are specified by a higher level.

   The message-orientation of SCTP is used to preserve the message
   boundaries of user messages.  Therefore, senders MUST NOT put more
   than one application message into an SCTP user message.  Unless the
   deprecated PPID-based fragmentation and reassembly is used, the
   sender MUST include exactly one application message in each SCTP user
   message.

   The SCTP Payload Protocol Identifiers (PPIDs) are used to signal the
   interpretation of the "Payload data".  The following PPIDs MUST be
   used (see Section 8):

   WebRTC String:  to identify a non-empty JavaScript string encoded in
      UTF-8.

   WebRTC String Empty:  to identify an empty JavaScript string encoded
      in UTF-8.

   WebRTC Binary:  to identify a non-empty JavaScript binary data
      (ArrayBuffer, ArrayBufferView or Blob).




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   WebRTC Binary Empty:  to identify an empty JavaScript binary data
      (ArrayBuffer, ArrayBufferView or Blob).

   SCTP does not support the sending of empty user messages.  Therefore,
   if an empty message has to be sent, the appropriate PPID (WebRTC
   String Empty or WebRTC Binary Empty) is used and the SCTP user
   message of one zero byte is sent.  When receiving an SCTP user
   message with one of these PPIDs, the receiver MUST ignore the SCTP
   user message and process it as an empty message.

   The usage of the PPIDs "WebRTC String Partial" and "WebRTC Binary
   Partial" is deprecated.  They were used for a PPID-based
   fragmentation and reassembly of user messages belonging to reliable
   and ordered data channels.

   If a message with an unsupported PPID is received or some error
   condition related to the received message is detected by the receiver
   (for example, illegal ordering), the receiver SHOULD close the
   corresponding data channel.  This implies in particular that
   extensions using additional PPIDs can't be used without prior
   negotiation.

   The SCTP base protocol specified in [RFC4960] does not support the
   interleaving of user messages.  Therefore sending a large user
   message can monopolize the SCTP association.  To overcome this
   limitation, [I-D.ietf-tsvwg-sctp-ndata] defines an extension to
   support message interleaving, which SHOULD be used.  As long as
   message interleaving is not supported, the sender SHOULD limit the
   maximum message size to 16 KB to avoid monopolization.

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

   The sender SHOULD disable the Nagle algorithm (see [RFC1122]) to
   minimize the latency.

6.7.  Closing a Data Channel

   Closing of a data channel MUST be signaled by resetting the
   corresponding outgoing streams [RFC6525].  This means that if one
   side decides to close the data channel, it resets the corresponding
   outgoing stream.  When the peer sees that an incoming stream was
   reset, it also resets its corresponding outgoing stream.  Once this
   is completed, the data channel is closed.  Resetting a stream sets
   the Stream Sequence Numbers (SSNs) of the stream back to 'zero' with
   a corresponding notification to the application layer that the reset



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   has been performed.  Streams are available for reuse after a reset
   has been performed.

   [RFC6525] also guarantees that all the messages are delivered (or
   abandoned) before the stream is reset.

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].

   It should be noted that a receiver must be prepared that the sender
   tries to send arbitrary large messages.

8.  IANA Considerations

   [NOTE to RFC-Editor:

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

   ]

   This document uses six already registered SCTP Payload Protocol
   Identifiers (PPIDs): "DOMString Last", "Binary Data Partial", "Binary
   Data Last", "DOMString Partial", "WebRTC String Empty", and "WebRTC
   Binary Empty".  [RFC4960] creates the registry "SCTP Payload Protocol
   Identifiers" from which these identifiers were assigned.  IANA is
   requested to update the reference of these six assignments to point
   to this document and change the names of the first four PPIDs.  The
   corresponding dates should be kept.

   Therefore these six assignments should be updated to read:

   +-------------------------------+----------+-----------+------------+
   | Value                         | SCTP     | Reference | Date       |
   |                               | PPID     |           |            |
   +-------------------------------+----------+-----------+------------+
   | WebRTC String                 | 51       | [RFCXXXX] | 2013-09-20 |
   | WebRTC Binary Partial         | 52       | [RFCXXXX] | 2013-09-20 |
   | (Deprecated)                  |          |           |            |
   | WebRTC Binary                 | 53       | [RFCXXXX] | 2013-09-20 |
   | WebRTC String Partial         | 54       | [RFCXXXX] | 2013-09-20 |
   | (Deprecated)                  |          |           |            |
   | WebRTC String Empty           | 56       | [RFCXXXX] | 2014-08-22 |
   | WebRTC Binary Empty           | 57       | [RFCXXXX] | 2014-08-22 |
   +-------------------------------+----------+-----------+------------+



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9.  Acknowledgments

   Many thanks for comments, ideas, and text from Harald Alvestrand,
   Richard Barnes, Adam Bergkvist, Alissa Cooper, Benoit Claise, Spencer
   Dawkins, Gunnar Hellstrom, Christer Holmberg, Cullen Jennings, Paul
   Kyzivat, Eric Rescorla, Adam Roach, Irene Ruengeler, Randall Stewart,
   Martin Stiemerling, 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.

   [RFC4347]  Rescorla, E. and N. Modadugu, "Datagram Transport Layer
              Security", RFC 4347, April 2006.

   [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.

   [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.



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   [I-D.ietf-tsvwg-sctp-ndata]
              Stewart, R., Tuexen, M., Loreto, S., and R. Seggelmann,
              "Stream Schedulers and a New Data Chunk for the Stream
              Control Transmission Protocol", draft-ietf-tsvwg-sctp-
              ndata-01 (work in progress), July 2014.

   [I-D.ietf-rtcweb-data-protocol]
              Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel
              Establishment Protocol", draft-ietf-rtcweb-data-
              protocol-08 (work in progress), September 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-07 (work in progress), December 2014.

   [I-D.ietf-rtcweb-security]
              Rescorla, E., "Security Considerations for WebRTC", draft-
              ietf-rtcweb-security-07 (work in progress), July 2014.

   [I-D.ietf-rtcweb-security-arch]
              Rescorla, E., "WebRTC Security Architecture", draft-ietf-
              rtcweb-security-arch-10 (work in progress), July 2014.

   [I-D.ietf-rtcweb-jsep]
              Uberti, J., Jennings, C., and E. Rescorla, "Javascript
              Session Establishment Protocol", draft-ietf-rtcweb-jsep-08
              (work in progress), October 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-06 (work in progress), December
              2014.

   [I-D.ietf-mmusic-sctp-sdp]
              Holmberg, C., Loreto, S., and G. Camarillo, "Stream
              Control Transmission Protocol (SCTP)-Based Media Transport
              in the Session Description Protocol (SDP)", draft-ietf-
              mmusic-sctp-sdp-11 (work in progress), December 2014.

10.2.  Informative References

   [RFC1122]  Braden, R., "Requirements for Internet Hosts -
              Communication Layers", STD 3, RFC 1122, October 1989.





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   [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.

Authors' Addresses

   Randell Jesup
   Mozilla
   US

   Email: randell-ietf@jesup.org


   Salvatore Loreto
   Ericsson
   Hirsalantie 11
   Jorvas  02420
   FI

   Email: salvatore.loreto@ericsson.com


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

   Email: tuexen@fh-muenster.de














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