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Versions: 00 01

CoRE Working Group                                             C. Groves
Internet-Draft                                                   W. Yang
Intended status: Informational                                    Huawei
Expires: April 20, 2017                                 October 17, 2016


A WebRTC Data Channel Transport for the Constrained Application Protocol
                                 (CoAP)
                     draft-groves-coap-webrtcdc-01

Abstract

   The WebRTC framework defines a generic transport service allowing
   WEB-browsers and other endpoints to exchange generic data from peer
   to peer utilizing a Stream Control Transmission Protocol (SCTP)
   transport.  This service is known as Web Real Time Communication
   WebRTC data channels (WebRTC DC).  The use of WebRTC DCs for the
   Constrained Application Protocol (CoAP) allows WebRTC enabled devices
   to exchange CoAP data between peers in a secure reliable manner.

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 20, 2017.

Copyright Notice

   Copyright (c) 2016 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
   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



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   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  . . . . . . . . . . . . . . . . . . . . . . . .   3
   2.  Requirements Language . . . . . . . . . . . . . . . . . . . .   5
   3.  Constrained Application Protocol  . . . . . . . . . . . . . .   5
     3.1.  Message Model . . . . . . . . . . . . . . . . . . . . . .   5
     3.2.  Request Response Model  . . . . . . . . . . . . . . . . .   6
     3.3.  Intermediaries and Caching  . . . . . . . . . . . . . . .   7
     3.4.  Resource Discovery  . . . . . . . . . . . . . . . . . . .   7
     3.5.  Opening Handshake . . . . . . . . . . . . . . . . . . . .   7
     3.6.  Message Format  . . . . . . . . . . . . . . . . . . . . .   8
     3.7.  Option Format and Value . . . . . . . . . . . . . . . . .   9
   4.  Message Transmission  . . . . . . . . . . . . . . . . . . . .   9
     4.1.  Messages and Endpoints  . . . . . . . . . . . . . . . . .  10
     4.2.  Messages Transmitted Reliably . . . . . . . . . . . . . .  10
     4.3.  Messages Transmitted without Reliability  . . . . . . . .  11
     4.4.  Message Correlation . . . . . . . . . . . . . . . . . . .  11
     4.5.  Message Duplication . . . . . . . . . . . . . . . . . . .  11
     4.6.  Message Size  . . . . . . . . . . . . . . . . . . . . . .  11
     4.7.  Congestion Control  . . . . . . . . . . . . . . . . . . .  12
     4.8.  Transmission Parameters . . . . . . . . . . . . . . . . .  12
   5.  Request/Response Semantics  . . . . . . . . . . . . . . . . .  12
   6.  CoAP URI  . . . . . . . . . . . . . . . . . . . . . . . . . .  12
     6.1.  coaps+wr URI scheme . . . . . . . . . . . . . . . . . . .  13
   7.  Discovery . . . . . . . . . . . . . . . . . . . . . . . . . .  13
     7.1.  Service Discovery . . . . . . . . . . . . . . . . . . . .  13
     7.2.  Resource Discovery  . . . . . . . . . . . . . . . . . . .  14
   8.  Multicast CoAP  . . . . . . . . . . . . . . . . . . . . . . .  14
   9.  Securing CoAP . . . . . . . . . . . . . . . . . . . . . . . .  14
   10. Interworking  . . . . . . . . . . . . . . . . . . . . . . . .  14
   11. Security Considerations . . . . . . . . . . . . . . . . . . .  15
   12. IANA Considerations . . . . . . . . . . . . . . . . . . . . .  15
     12.1.  New WebRTC DC Protocol Value . . . . . . . . . . . . . .  15
     12.2.  Secure Service Name and Port Number Registration . . . .  16
     12.3.  ALPN Protocol ID . . . . . . . . . . . . . . . . . . . .  16
     12.4.  URI Schemes  . . . . . . . . . . . . . . . . . . . . . .  16
     12.5.  New SIP Media Feature Tag  . . . . . . . . . . . . . . .  16
   13. Examples  . . . . . . . . . . . . . . . . . . . . . . . . . .  17
   14. Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  19
   15. Changelog . . . . . . . . . . . . . . . . . . . . . . . . . .  19
   16. References  . . . . . . . . . . . . . . . . . . . . . . . . .  19
     16.1.  Normative References . . . . . . . . . . . . . . . . . .  19
     16.2.  Informative References . . . . . . . . . . . . . . . . .  22
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  23



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1.  Introduction

   Whilst the Constrained Application Protocol (CoAP) [RFC7252] was
   designed for Internet of Things (IoT) deployments in constrained
   network environments its ready adoption has seen the use of it in a
   multitude of different network environments.  For example
   [I-D.silverajan-core-coap-alternative-transports] provides use cases
   for alternate CoAP transports.

   [I-D.ietf-core-coap-tcp-tls] highlights a number of issues using the
   native User Datagram Transport (UDP) and envisages deployments more
   closely integrated with a Web environment.  It also proposes the use
   of the WebSocket protocol [RFC6455].  The use of CoAP over WebRTC DCs
   has not yet been discussed.

   WebRTC is a framework [I-D.ietf-rtcweb-overview] that defines real
   time protocols for browser-based applications.  It allows
   communications between peer WebRTC endpoints (e.g. browsers) without
   the need to communicate through a web server.

   In addition to protocols for the realtime transport of audio and
   video, the transport of generic peer-to-peer non-media data has been
   defined using WebRTC DCs.  The non-media data is transported using
   the Stream Control Transmission Protocol (SCTP) [RFC4960]
   encapsulated in the Datagram Transport Layer Security (DTLS)
   [RFC6347].  It allows both reliable and partially reliable transport
   and provides confidentiality, source authenticated and integrity
   protected transfers.  The use of Interactive Connectivity
   Establishment (ICE) [RFC5245] allows network address translator (NAT)
   traversal.  The SCTP/DTLS association may be shared with existing
   audio and video streams enabling multiplexing of several data streams
   over a single port further facilitating NAT traversal.

   Use cases for WebRTC DCs (section 3.1/[I-D.ietf-rtcweb-data-channel]
   envisage scenarios where the real-time gaming experience is enhanced
   by additional object state information.  Additional scenarios are
   considered where information such as heart rate sensor or oxygen
   saturation sensors could augment audio and video in remote medicine
   scenarios.  The transport of such sensor information is what CoAP has
   been designed for.

   This is illustrated in Figure 1 showing the WebRTC Trapeziod with
   added sensor/CoAP information.  The left hand side WebRTC endpoint
   acts as a CoAP to CoAP proxy.







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                   +-----------+             +-----------+
                   |   Web     |             |   Web     |
                   |           |  Signaling  |           |
                   |           |-------------|           |
                   |  Server   |   path      |  Server   |
                   |           |             |           |
                   +-----------+             +-----------+
                        /                           \
                       /                             \ Application-
                      /                               \ defined over
                     /                                 \ HTTP/Websockets
                    /  Application-defined over         \
                   /   HTTP/Websockets                   \
                  /                                       \
            +-----------+                           +-----------+
            |JS/HTML/CSS|                           |JS/HTML/CSS|
            +-----------+                           +-----------+
            +-----------+                           +-----------+
   SensorA  |           |                           |           |
   CoAP/UDP |           |                           |           |
     +------+  Browser  | ------------------------- |  Browser  +
            |           |          Media path       |           |
            |           |       (CoAP/WebRTC DC)    |           |
            +-----------+                           +-----------+

                    Figure 1: CoAP and WebRTC Trapeziod

   By utilizing the WebRTC DC (SCTP over DTLS over ICE/UDP (or ICE/TCP))
   transport for CoAP a number of important features are inherited
   including: congestion control, order and unordered messages delivery,
   large message transmission by providing segmentation and reassembly
   and multiple unidirectional streams.  A more detailed analysis of the
   benefits of WebRTC DCs can be found in section
   5/[I-D.ietf-rtcweb-data-channel].  [I-D.ietf-tsvwg-sctp-dtls-encaps]
   describes the usage of SCTP over DTLS.

   WebRTC defines in-band and out-of-band methods for establishing a
   data channel and indicating its characteristics.  The Data Channel
   Establishment Protocol (DCEP) [I-D.ietf-rtcweb-data-protocol]
   provides an in band means of establishing individual data channels.
   [I-D.ietf-mmusic-data-channel-sdpneg] uses the Session Description
   Protocol (SDP) [RFC4566] to provide an out-of-band means to establish
   data channels.

   By defining the use of CoAP over WebRTC DC it negates the need for
   the WebRTC endpoint to interwork between any CoAP messages received
   from local devices to a proprietary WebRTC DC format when signalling
   a remote WebRTC endpoint.



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   The SCTP Payload Protocol Identifier (PPID) allows the identification
   of whether a UTF-8 or Binary encoding is being used and thus
   facilitates the use of text or binary CoAP protocol serializations.

2.  Requirements Language

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

3.  Constrained Application Protocol

   This section describes the use of CoAP over WebRTC DC as a delta to
   the information contained in section 2/[RFC7252].

   Figure 2 shows the CoAP abstract layering as applied to the WebRTC
   framework.

                         +---------------------------+
                          |        Application        |
                          +------+------+-------------+ \
                          | DCEP | Requests/Responses | |
                          |      +--------------------| | CoAP
                          |      | Messages           | |
                          +------+--------------------+ /
                          |        SCTP               |
            +-----------------------------------------+
            | STUN | SRTP |        DTLS               |
            +-----------------------------------------+
            |                ICE                      |
            +-----------------------------------------+
            | UDP1 | UDP2 | UDP3 | ...                |
            +-----------------------------------------+

              Figure 2: WebRTC protocol layers including CoAP

   WebRTC DC mandates the use of SCTP over DTLS.  Whilst the above
   diagram indicates the use of ICE over UDP the use of TCP is also
   possible in fall back scenarios.

3.1.  Message Model

   WebRTC DC allows application protocol messages to be exchanged by
   peers.  WebRTC supports both a reliable and partially reliable
   methods of transmitting user messages.





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   CoAP [RFC7252] supports four message types "Confirmable, Non-
   Confirmable, Acknowledge and Reset".  As SCTP provides the
   reliability mechanism the CoAP message types are not needed for CoAP
   over WebRTC DC.

   WebRTC DC does not support multicast usage.

3.2.  Request Response Model

   WebRTC DCs are realized as a pair of one incoming and one outgoing
   SCTP stream (with the same identifier) allowing bi-directional
   communication.  Each channel has properties (see section
   6.4/[I-D.ietf-rtcweb-data-channel] as discussed below:

   o  reliable or unreliable message transmission: WebRTC DCs support
      the per message indication whether user messages are reliable or
      partially reliable.  Partial reliability indicates that message
      retransmission is limited to a certain number of retransmissions
      or lifetime.  This loosely parallels to the CoAP usage of
      Confirmable (CON) or Non-confirmable (NON) messages.

   o  in-order or out-of-order message delivery: WebRTC DCs support the
      per message indication whether user messages are delivered in or
      out of order.  CoAP has been designed for unreliable transports
      and therefore assumes that messages may arrive out-of-order.  CoAP
      implements a lightweight reliability mechanism to deal with this
      issue.

   o  priority: WebRTC DCs allows a priority to specified for stream
      scheduling.  The usage of this is application specific.  Usage of
      CoAP has no impact on this parameter.  It's up to the application
      using CoAP to set this indication.

   o  an optional label: This is an application/implementation specific
      label.  Uniqueness is not guaranteed.  Usage of CoAP has no impact
      on this parameter.

   o  an optional protocol: This is used to indicate the application
      protocol in use.  A value is required to identify the usage of
      CoAP.

   As discussed above WebRTC DC supports an unreliable / un-ordered
   delivery of messages.  Implementations utilizing these data channel
   characteristics may use CoAP messages and request/response model
   largely unchanged.  In this case the CoAP reliability mechanisms
   would be used.  However as WebRTC DC's usage of SCTP is reliable or
   partially reliable there is some redundancy between the functionality
   that WebRTC DCs and CoAP provides.



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   The redundancies are identified and discussed in section
   2/[I-D.ietf-core-coap-tcp-tls].  Namely:

   1.  There is no need to carry acknowledgement semantics at a CoAP
       level.

   2.  There is no need for duplicate delivery detection.  This is part
       of the SCTP layer.

3.3.  Intermediaries and Caching

   As CoAP over WebRTC DC is peer to peer no intermediares or caching is
   expected.

3.4.  Resource Discovery

   The usage of CoAP over WebRTC DC has no foreseeable impacts on
   resource discovery.

3.5.  Opening Handshake

   Prior to the establishment of a CoAP over WebRTC DC the
   characteristics of the SCTP association and data channel may be
   negotiated by signalling.  See Section 4 for further details.  For
   example when using SDP [I-D.ietf-mmusic-sctp-sdp] the use of the "SDP
   max-message-size" attribute indicates the maximum received SCTP
   message size.

   Further characteristics (such as those described in Section 3.2) are
   negotiated at the establishment of the WebRTC DC.

   On establishment of the CoAP over WebRTC DC the client and server MAY
   send a CoAP Capability and Settings message (CSM see
   Section 4.3/[I-D.ietf-core-coap-tcp-tls]) as its first message on the
   connection to establish CoAP specific capabilities.  Any capabilities
   signalled SHALL not contradict previously negotiated chracteristics.
   Consideration for the individual options are below:

   o  Server-Name Setting: CoAP over WebRTC DC clients MAY use the
      server-name setting option.  The initial value is derived based on
      the signalling method used to establish the WebRTC peer to peer
      communications.  WebRTC does not mandate a signalling method.  For
      example if Websockets is used then the value may be taken from the
      HTTP host header field.

   o  Max-message size Capability: The CoAP Max-Message-Size shall not
      exceed the SCTP message size.




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   o  Block-wise Transfer Capability: CoAP over WebRTC DC client and
      server MAY support the use of BERT
      (Section 5/[I-D.ietf-core-coap-tcp-tls]).  See Section 4.6 for
      message size considerations.

   o  Ping and Pong Messages: Ping and Pong messages MAY be sent by CoAP
      over WebRTC DC clients and servers.  However its use as a basic
      keepalive is not required as WebRTC defines a method to determine
      liveness (see Section 4.1).

   o  Release Messages: CoAP over WebRTC DC clients and servers may
      support the CoAP Release message.  On receipt of a release message
      the CoAP over WebRTC DC SHALL be closed as per Section 4.

   o  Abort Messages: CoAP over WebRTC DC clients and servers may
      support the CoAP Abort message.  Senders SHALL then close the CoAP
      over WebRTC DC as per Section 4.

3.6.  Message Format

   As discussed in [I-D.ietf-core-coap-tcp-tls] the use of a reliable
   underlying transport allows the use of a modified CoAP header format.
   The modified format removes the "Type (T)" and "Message ID" fields
   and introduces a "length" as illustrated below in Figure 3.

       0                   1                   2                   3
       0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |Len=13 |  TKL  | Length (8-bit)|      Code     | TKL bytes ...
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |  Options (if any) ...
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |1 1 1 1 1 1 1 1|    Payload (if any) ...
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

        Figure 3: CoAP Header with TCP with 8-bit Length in Header

   CoAP over WebRTC DC implementations shall also use the message format
   in Figure 3 with the following consideration:

   o  The length field was added for message delimitation to keep
      messages separate in TCP.  WebRTC DC uses the message orientation
      of SCTP to preserve message boundaries thus the use of single
      application message per SCTP user message is mandated by the
      WebRTC framework.  The length field shall be set to 0.

   CoAP [RFC7252] supports the use of different content-formats.  WebRTC
   DC defines the use of PPIDs per SCTP user message as follows:



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

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

   Depending on the content-format (see section 12.3/[RFC7252]) an
   appropriate PPID to the encoding type SHOULD be used to minimise the
   need for translating between encodings.  For example content type of
   "text/plain" would result in the use of PPID "WebRTC String".

   Author's note: Specific mappings for each content-format could be
   provided however given that the formats may change in the future it
   may be sufficient to offer broad guidance instead.

3.7.  Option Format and Value

   There are no impacts to option formats or values due to the use of
   CoAP over WebRTC DCs.

   Author's note: Given that the host is determined by the usage of
   WebRTC are the Uri-Host and Uri-Port relevant?  It would seem that
   this may be valuable to establish a resource tree independent of
   WebRTC.

4.  Message Transmission

   In order to use a WebRTC DC, a SCTP over DTLS over ICE/UDP (or ICE/
   TCP) association must be established.  A DTLS connection is
   established followed by an SCTP association.  The out-of-band
   establishment method through the use of SDP-based Data Channel
   Negotiation [I-D.ietf-mmusic-data-channel-sdpneg] allows the
   negotiation of SCTP over DTLS over ICE/UDP as well as the negotiation
   and establishment of the characteristics of an individual WebRTC DC.

   The in-band establishment method through the use of the Data Channel
   Establishment Protocol (DCEP) [I-D.ietf-rtcweb-data-protocol] only
   allows for the establishment of a WebRTC DC once the SCTP over DTLS
   is established.  It relies on DATA_CHANNEL_OPEN and DATA_CHANNEL_ACK
   messages on the relevant SCTP stream to negotiate the properties of
   the channel.  A separate SCTP PPID (50) indicates that the SCTP user
   message is a WebRTC DCEP message to allow de-multiplexing by the
   endpoint.

   WebRTC DCs are realized as a pair of one incoming and one outgoing
   SCTP stream (with the same identifier).  Requests are sent on an
   outgoing SCTP stream and received on the peer incoming stream.  The
   SCTP stream identifier is bound to the WebRTC DC instance at the



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   establishment of the data channel.  The establishment protocol
   provides rules for determining the SCTP stream IDs.

   WebRTC DC closure (Stream Reset) is supported through the use of the
   SCTP stream reconfiguration extension defined in [RFC6525].  The SCTP
   Stream Reconfiguration reset has the effect of setting the numbering
   sequence of the SCTP stream back to zero.  This is separate function
   to the CoAP "Reset" message.  There is no mapping between the SCTP
   Stream Reset and the CoAP "Reset" message.

4.1.  Messages and Endpoints

   As per section 2.5/[I-D.ietf-core-coap-tcp-tls] requests can be sent
   from both the connecting host and the endpoint that accepted the
   connection.  Who initiated the SCTP/DTLS connection has no bearing on
   the meaning of the CoAP terms client and server.

   WebRTC DC mandates the use of DTLS thus the endpoint is identified
   depending on the security mode.

   WebRTC DCs allows the indication of whether a SCTP user message is
   empty through the use of PPIDs (WebRTC String Empty and WebRTC Binary
   Empty).  CoAP defines the use of empty messages.  However from the
   perspective of SCTP these CoAP messages would still contain header
   information thus PPIDs for empty data MUST not be used.

   CoAP uses an Empty Confirmable message to provoke a Reset message to
   check the liveness of an endpoint (so called "CoAP" ping).  In WebRTC
   liveness and the ability to send data is determined through the usage
   of Session Traversal Utilities for NAT (STUN) Usage for Consent
   Freshness [RFC7675].  Therefore endpoints utilising CoAP over WebRTC
   DC MUST not use CoAP "reset" messages.

   CoAP also uses Empty messages to acknowledge a request.  This is not
   required due to the SCTP level acknowledgement.  Therefore Empty
   messages MUST not be used with CoAP over WebRTC.

4.2.  Messages Transmitted Reliably

   For CoAP messages marked as confirmable the sender SHALL use a
   reliable SCTP user message.

   A CoAP endpoint MUST use the ordered delivery SCTP service, as
   described in [RFC4960], for the CoAP protocol.

   CoAP receivers MUST NOT generate CoAP "ACK" or "reset" messages.
   SCTP level acknowledgement mechanisms are used.




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4.3.  Messages Transmitted without Reliability

   WebRTC DC makes use of the SCTP Partial Reliability (SCTP-PR)
   Extension [RFC3758].  This extension allows a user to indicate on a
   per message basis how persistent the transport service should be in
   attempting to send the message to the receiver.  One of the benefits
   of using this extension identified by [RFC3758] is:

   1.  Some application layer protocols may benefit from being able to
       use a single SCTP association to carry both reliable content, -
       such as text pages, billing and accounting information, setup
       signaling - and unreliable content, e.g., state that is highly
       sensitive to timeliness, where generating a new packet is more
       advantageous than transmitting an old one.

   This benefit is also one of the reasons the CoAP "Non-Confirmable"
   message was introduced.  However the SCTP-PR and the CoAP "Non-
   Confirmable" message mechanisms differs in their approach.  The SCTP-
   PR mechanism focuses on sender side behaviour (e.g. when to abandon
   retransmission).  The CoAP "Non-Confirmable" message focuses on
   receiver side behaviour (e.g. must not send a CoAP ACK).  Even with
   the use of SCTP-PR an SCTP receiver will send an SCTP level ACK for a
   successfully received SCTP CHUNK.  The CoAP "Non-Confirmable" message
   has no effect on the SCTP level function.

   Therefore the use of a CoAP "Non-Confirmable" message type is
   redundant as the CoAP receiver will never send a CoAP ACK message in
   response.

   SCTP-PR provides a complimentary function and thus CoAP senders who
   send Non-confirmable messages SHALL also use SCTP-PR for that
   message.

4.4.  Message Correlation

   Due to reliability being handled at the SCTP layers the CoAP "Message
   ID" is not required.

4.5.  Message Duplication

   The SCTP layer provides message duplication protection.  The CoAP
   application level procedure is not required.

4.6.  Message Size

   The considerations in section 4.1/[I-D.ietf-core-coap-tcp-tls]
   regarding message size limitations also apply to the use of WebRTC
   DCs.  However [I-D.ietf-rtcweb-data-channel] indicates that senders



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   SHOULD limit the maximum message size to 16KB to avoid monopolization
   of the SCTP association.  Section 5/[I-D.ietf-tsvwg-sctp-dtls-encaps]
   provides further details regarding segmentation and reassembly and
   path maximum transmission unit (MTU) discovery.

   Interleaving of large user messages is supported by an SCTP protocol
   extension defined in [I-D.ietf-tsvwg-sctp-ndata].

4.7.  Congestion Control

   SCTP provides congestion control on a per-association basis (see
   section 5/[I-D.ietf-rtcweb-data-channel].

4.8.  Transmission Parameters

   The application level parameters defined in section 4.8/[RFC7252] are
   not relevant to SCTP.

5.  Request/Response Semantics

   Request and response semantics for CoAP over WebRTC DC is as per
   section 5/[RFC7252] with the following exceptions:

   o  section 5.2/[RFC7252]: separate responses MUST be used.  Given
      that WebRTC DC provides an SCTP level acknowledgement it is not
      possible to piggy back CoAP responses.

   o  section 5.3.1/[RFC7252]: due to the use of DTLS the advice
      regarding token use without using TLS is invalid.

   o  section 5.3.2/[RFC7252]: In addition CoAP request/response
      matching is unique to a particular WebRTC DC (SCTP StreamID pair).

   o  section 5.8/[RFC7252]: It is not possible to use a 4.05
      piggybacked response.

6.  CoAP URI

   CoAP [RFC7252] defines the "coap" and "coaps" URI schemes for
   identifying CoAP resources and providing a means of locating the
   resource.  [RFC7252] defines these resources for use with CoAP over
   UDP.

   Section 8/[RFC7252] (Multicast CoAP), does not apply to the URI
   schemes defined in the present specification.

   Resources made available via the "coaps+wr" schemes have no shared
   identity with the other scheme or with the "coap" or "coaps" scheme,



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   even if their resource identifiers indicate the same authority (the
   same host listening to the same port).  The schemes constitute
   distinct namespaces and, in combination with the authority, are
   considered to be distinct origin servers.

6.1.  coaps+wr URI scheme

   coaps-wr-URI = "coaps+wr:" "//" host [ ":" port ] path-abempty
                     [ "?" query ]

   The semantics defined in section 6.3/[RFC7252], apply to this URI
   scheme, with the following changes:

   o  The port SHALL be omitted.  The underlying UDP or TCP port and
      SCTP port is negotiated prior to the establishment of the CoAP
      over WebRTC DC.

7.  Discovery

7.1.  Service Discovery

   WebRTC does not define peer discovery mechanisms.  Peers discover
   each other through the use of the ICE protocol.  ICE candidates need
   to be sent from peer to peer via signalling.  The Javascript Session
   Establishment Protocol (JSEP) [I-D.ietf-rtcweb-jsep] details the
   generic SDP media descriptions for peer endpoints to determine the
   characteristics of a session.  The actual signalling protocol between
   application servers is unspecified.  WebRTC endpoints MUST implement
   the network functions detailed by JSEP including ICE functionality.

   Whilst the inter-application server signalling protocol is
   unspecified, the Session Initiation Protocol (SIP) is able to carry
   SDP for the purposes of establishing a CoAP over WebRTC DC session.
   SIP allows the use of media feature tags to indicate user agent
   capabilities [RFC3840].  In order to indicate that a SIP user agent
   supports the use of CoAP a new "sip.coap" media feature tag is
   proposed.  A CoAP-capable endpoint SHOULD include this media feature
   tag in its REGISTER requests and OPTION responses.  It SHOULD also
   include the media feature tag in INVITE and UPDATE [RFC3311] requests
   and responses.  Presence of the media feature tag in the contact
   field of a requestor response can be used to determine that the far
   end supports CLUE.

   The exchange of SDP results in: the underlying transport address
   (e.g.  IPv4 or IPv6), the underlying transport port (e.g.  UDP port)
   the SCTP port and the SCTP StreamID used for the CoAP WebRTC DC being
   exchanged between the peer endpoints.




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7.2.  Resource Discovery

   On establishment of a CoAP WebRTC DC endpoints are able to use the
   resource discovery mechanism defined in [RFC6690] for CoAP resources.

8.  Multicast CoAP

   WebRTC DCs do not support multicast.

9.  Securing CoAP

   This document defines how to convey CoAP over WebRTC DCs.  The WebRTC
   security architecture [I-D.ietf-rtcweb-security-arch] mandates the
   use of DTLS for data channels.  The use of DTLS 1.2 is compatible
   with CoAP [RFC7252] which allows makes use of DTLS 1.2.

   The use of DTLS for WebRTC is detailed in
   [I-D.ietf-rtcweb-security-arch].

10.  Interworking

   An WebRTC endpoint supporting CoAP may in affect act as a gateway
   between local sensor devices and a remote peer endpoint.  The local
   sensors may utilise CoAP over an alternate signalling transport such
   as UDP to the local WebRTC endpoint.  The WebRTC endpoint may then
   utilise CoAP over WebRTC to signal to the remote peer.

   A CoAP gateway when converting to and from a WebRTC transport will in
   general perform the following functions:

   o  Map received Empty CoAP message to SCTP level operations and
      discard the empty message.

   o  Map received ACK message to SCTP level operations and discard the
      ACK message.

   o  Separate piggy-backed messages.

   o  Provide a mapping between received and sent Tokens in order to
      match requests and responses.

   Other behaviour depends on the type of proxy behaviour the gateway is
   performing.  See section 5.7/[RFC7252] for more details.








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11.  Security Considerations

   Security considerations for WebRTC are discussed in
   [I-D.ietf-rtcweb-security].

   The use of CoAP over WebRTC can potentially negate the risks
   mentioned in:

   o  section 11.3/[RFC7252] on insecure UDP and multicast being used to
      aid an amplification attack.

   o  section 11.4/[RFC7252] on IP address spoofing and section
      11.5/[RFC7252] on Cross-Protocol attacks.

   o  section 11.6/[RFC7252] may also not be relevant as WebRTC
      endpoints are not expected to be severely constrained.

   Of particular relevance to the support of CoAP over WebRTC DC is
   access to local devices.  Devices generating CoAP data are
   essentially the same as cameras and microphones in that they may
   expose sensitive data about the user or the location of the device.
   Thus the guidance of section 4.1/[I-D.ietf-rtcweb-security] applies
   to devices generating CoAP data.  Whilst CoAP has been designed for
   constrained devices where there is no user interface to inform/
   request consent, it is assumed that device utilising WebRTC DC for
   CoAP is more likely at minimum a Class 2 [RFC7228] device that could
   facilitate consent.

   The CoAP media feature tag defined by this document tag may be
   present in sessions not utilising CoAP, which increases the metadata
   available about the sending device, which can help an attacker
   differentiate between multiple devices and help them identify
   otherwise anonymised users via the fingerprint of features their
   device supports.  To prevent this, SIP signalling SHOULD always be
   encrypted using TLS [RFC5630].

12.  IANA Considerations

12.1.  New WebRTC DC Protocol Value

   NOTE: This registration is exactly the same as the registration in
   [I-D.savolainen-core-coap-websockets].

   This document requests the registration of the subprotocol name
   "coap.v1" in the WebSocket Subprotocol Name Registry.

   o  Subprotocol Identifier: coap.v1




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   o  Subprotocol Common Name: Constrained Application Protocol (CoAP)

   o  Subprotocol Definition: This document

12.2.  Secure Service Name and Port Number Registration

   No need has been identified to register a new service name and port
   number for CoAP over WebRTC.  Port number allocation is dynamic.  The
   use of the SCTP over DTLS over UDP/TCP results in a layering of
   services.

12.3.  ALPN Protocol ID

   [I-D.ietf-core-coap-tcp-tls] defines a new "coap" application
   protocol negotiation protocol identity.  However as the DTLS
   connection is used to establish a WebRTC application the protocol
   identifiers defined in [I-D.ietf-rtcweb-alpn] MUST be used.  Note:
   that confidentiality protection does not extend to WebRTC DCs.

12.4.  URI Schemes

   This document registers a new URI scheme "coaps+wr" for the use of
   CoAP over WebRTC DCs.  The "coaps+wr" URI schemes can be compared to
   the "https" URI scheme.

   The IANA is requested to add this new URI schemes to the registry
   established with [RFC7595].

12.5.  New SIP Media Feature Tag

   This specification registers a new media feature tag in the SIP
   [RFC3264] tree per the procedures defined in [RFC2506] and [RFC3840].

   Media feature tag name: sip.coap

   ASN.1 Identifier: 1.3.6.1.8.4.30

   Summary of the media feature indicated by this tag: This feature tag
   indicates that the device supports the Constrained Application
   Protocol (CoAP).

   Values appropriate for use with this feature tag: Boolean.

   The feature tag is intended primarily for use in the following
   applications, protocols, services, or negotiation mechanisms: This
   feature tag is useful to indicate the support of CoAP.

   Related standards or documents: This document



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   Security Considerations: Security considerations for this media
   feature tag are discussed in Section 11.

   Name(s) and email address(es) of person(s) to contact for further
   information:

   o  CORE workgroup: core@ietf.org

   o  CORE chairs: core-chairs@ietf.org

   Intended usage: COMMON

13.  Examples

   The example SDP Offer shows a CoAP over WebRTC DC utilising out-of-
   band negotiation [I-D.ietf-mmusic-data-channel-sdpneg].  It is based
   on the example in section 7.2/[I-D.ietf-rtcweb-jsep].  Modified lines
   are indicated with ">>>" at the start of the line.  These indicators
   are NOT part of the SDP syntax.  Note: some lines have been broken
   into two lines for formatting reasons.































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   v=0
      o=- 4962303333179871723 1 IN IP4 0.0.0.0
      s=-
      t=0 0
      a=group:BUNDLE a1 d1
      a=ice-options:trickle
      m=audio 9 UDP/TLS/RTP/SAVPF 96 0 8 97 98
      c=IN IP4 0.0.0.0
      a=rtcp:9 IN IP4 0.0.0.0
      a=mid:a1
      a=msid:57017fee-b6c1-4162-929c-a25110252400
             e83006c5-a0ff-4e0a-9ed9-d3e6747be7d9
      a=sendrecv
      a=rtpmap:96 opus/48000/2
      a=rtpmap:0 PCMU/8000
      a=rtpmap:8 PCMA/8000
      a=rtpmap:97 telephone-event/8000
      a=rtpmap:98 telephone-event/48000
      a=maxptime:120
      a=ice-ufrag:ATEn1v9DoTMB9J4r
      a=ice-pwd:AtSK0WpNtpUjkY4+86js7ZQl
      a=fingerprint:sha-256
                    19:E2:1C:3B:4B:9F:81:E6:B8:5C:F4:A5:A8:D8:73:04
                   :BB:05:2F:70:9F:04:A9:0E:05:E9:26:33:E8:70:88:A2
      a=setup:actpass
      a=rtcp-mux
      a=rtcp-rsize
      a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
      a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:mid
      a=ssrc:1732846380 cname:FocUG1f0fcg/yvY7

      m=application 0 UDP/DTLS/SCTP webrtc-datachannel
      c=IN IP4 0.0.0.0
      a=bundle-only
      a=mid:d1
      a=fmtp:webrtc-datachannel max-message-size=65536
      a=sctp-port 5000
      a=fingerprint:sha-256
                    19:E2:1C:3B:4B:9F:81:E6:B8:5C:F4:A5:A8:D8:73:04
                    :BB:05:2F:70:9F:04:A9:0E:05:E9:26:33:E8:70:88:A2
      a=setup:actpass
   >>>a=dcmap:0 subprotocol="coap.v1";"label="coap"

                        Figure 4: Example SDP Offer







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14.  Acknowledgements

   We would like to thank the authors of [I-D.ietf-core-coap-tcp-tls]
   and [I-D.savolainen-core-coap-websockets] for providing a framework
   for this document.  In addition we would like to thank Carsten
   Bormann for his feedback on message format.

15.  Changelog

   Changes from version 00:

   o  Updated message format to align with draft-core-coap-tcp-tls-04

   o  Updates to align with draft-core-coap-tcp-tls-04 as a result of
      the merger with websockets.  Added section on opening handshake.
      Added support of CoAP capability messages and BERT.

16.  References

16.1.  Normative References

   [I-D.ietf-mmusic-data-channel-sdpneg]
              Drage, K., Makaraju, M., Stoetzer-Bradler, J., Ejzak, R.,
              and (. (Unknown), "SDP-based Data Channel Negotiation",
              draft-ietf-mmusic-data-channel-sdpneg-10 (work in
              progress), September 2016.

   [I-D.ietf-mmusic-sctp-sdp]
              Holmberg, C., Shpount, R., Loreto, S., and G. Camarillo,
              "Session Description Protocol (SDP) Offer/Answer
              Procedures For Stream Control Transmission Protocol (SCTP)
              over Datagram Transport Layer Security (DTLS) Transport.",
              draft-ietf-mmusic-sctp-sdp-18 (work in progress), October
              2016.

   [I-D.ietf-rtcweb-alpn]
              Thomson, M., "Application Layer Protocol Negotiation for
              Web Real-Time Communications (WebRTC)", draft-ietf-rtcweb-
              alpn-04 (work in progress), May 2016.

   [I-D.ietf-rtcweb-data-channel]
              Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data
              Channels", draft-ietf-rtcweb-data-channel-13 (work in
              progress), January 2015.







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   [I-D.ietf-rtcweb-data-protocol]
              Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel
              Establishment Protocol", draft-ietf-rtcweb-data-
              protocol-09 (work in progress), January 2015.

   [I-D.ietf-rtcweb-jsep]
              Uberti, J., Jennings, C., and E. Rescorla, "Javascript
              Session Establishment Protocol", draft-ietf-rtcweb-jsep-16
              (work in progress), September 2016.

   [I-D.ietf-rtcweb-overview]
              Alvestrand, H., "Overview: Real Time Protocols for
              Browser-based Applications", draft-ietf-rtcweb-overview-15
              (work in progress), January 2016.

   [I-D.ietf-rtcweb-security]
              Rescorla, E., "Security Considerations for WebRTC", draft-
              ietf-rtcweb-security-08 (work in progress), February 2015.

   [I-D.ietf-rtcweb-security-arch]
              Rescorla, E., "WebRTC Security Architecture", draft-ietf-
              rtcweb-security-arch-12 (work in progress), June 2016.

   [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-09 (work in progress), January 2015.

   [I-D.ietf-tsvwg-sctp-ndata]
              Stewart, R., Tuexen, M., Loreto, S., and R. Seggelmann,
              "Stream Schedulers and User Message Interleaving for the
              Stream Control Transmission Protocol", draft-ietf-tsvwg-
              sctp-ndata-07 (work in progress), July 2016.

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119,
              DOI 10.17487/RFC2119, March 1997,
              <http://www.rfc-editor.org/info/rfc2119>.

   [RFC2506]  Holtman, K., Mutz, A., and T. Hardie, "Media Feature Tag
              Registration Procedure", BCP 31, RFC 2506,
              DOI 10.17487/RFC2506, March 1999,
              <http://www.rfc-editor.org/info/rfc2506>.

   [RFC3264]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
              with Session Description Protocol (SDP)", RFC 3264,
              DOI 10.17487/RFC3264, June 2002,
              <http://www.rfc-editor.org/info/rfc3264>.



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   [RFC3311]  Rosenberg, J., "The Session Initiation Protocol (SIP)
              UPDATE Method", RFC 3311, DOI 10.17487/RFC3311, October
              2002, <http://www.rfc-editor.org/info/rfc3311>.

   [RFC3758]  Stewart, R., Ramalho, M., Xie, Q., Tuexen, M., and P.
              Conrad, "Stream Control Transmission Protocol (SCTP)
              Partial Reliability Extension", RFC 3758,
              DOI 10.17487/RFC3758, May 2004,
              <http://www.rfc-editor.org/info/rfc3758>.

   [RFC3840]  Rosenberg, J., Schulzrinne, H., and P. Kyzivat,
              "Indicating User Agent Capabilities in the Session
              Initiation Protocol (SIP)", RFC 3840,
              DOI 10.17487/RFC3840, August 2004,
              <http://www.rfc-editor.org/info/rfc3840>.

   [RFC4566]  Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
              Description Protocol", RFC 4566, DOI 10.17487/RFC4566,
              July 2006, <http://www.rfc-editor.org/info/rfc4566>.

   [RFC4960]  Stewart, R., Ed., "Stream Control Transmission Protocol",
              RFC 4960, DOI 10.17487/RFC4960, September 2007,
              <http://www.rfc-editor.org/info/rfc4960>.

   [RFC5245]  Rosenberg, J., "Interactive Connectivity Establishment
              (ICE): A Protocol for Network Address Translator (NAT)
              Traversal for Offer/Answer Protocols", RFC 5245,
              DOI 10.17487/RFC5245, April 2010,
              <http://www.rfc-editor.org/info/rfc5245>.

   [RFC5630]  Audet, F., "The Use of the SIPS URI Scheme in the Session
              Initiation Protocol (SIP)", RFC 5630,
              DOI 10.17487/RFC5630, October 2009,
              <http://www.rfc-editor.org/info/rfc5630>.

   [RFC6347]  Rescorla, E. and N. Modadugu, "Datagram Transport Layer
              Security Version 1.2", RFC 6347, DOI 10.17487/RFC6347,
              January 2012, <http://www.rfc-editor.org/info/rfc6347>.

   [RFC6525]  Stewart, R., Tuexen, M., and P. Lei, "Stream Control
              Transmission Protocol (SCTP) Stream Reconfiguration",
              RFC 6525, DOI 10.17487/RFC6525, February 2012,
              <http://www.rfc-editor.org/info/rfc6525>.

   [RFC6690]  Shelby, Z., "Constrained RESTful Environments (CoRE) Link
              Format", RFC 6690, DOI 10.17487/RFC6690, August 2012,
              <http://www.rfc-editor.org/info/rfc6690>.




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   [RFC7228]  Bormann, C., Ersue, M., and A. Keranen, "Terminology for
              Constrained-Node Networks", RFC 7228,
              DOI 10.17487/RFC7228, May 2014,
              <http://www.rfc-editor.org/info/rfc7228>.

   [RFC7252]  Shelby, Z., Hartke, K., and C. Bormann, "The Constrained
              Application Protocol (CoAP)", RFC 7252,
              DOI 10.17487/RFC7252, June 2014,
              <http://www.rfc-editor.org/info/rfc7252>.

   [RFC7595]  Thaler, D., Ed., Hansen, T., and T. Hardie, "Guidelines
              and Registration Procedures for URI Schemes", BCP 35,
              RFC 7595, DOI 10.17487/RFC7595, June 2015,
              <http://www.rfc-editor.org/info/rfc7595>.

   [RFC7675]  Perumal, M., Wing, D., Ravindranath, R., Reddy, T., and M.
              Thomson, "Session Traversal Utilities for NAT (STUN) Usage
              for Consent Freshness", RFC 7675, DOI 10.17487/RFC7675,
              October 2015, <http://www.rfc-editor.org/info/rfc7675>.

16.2.  Informative References

   [I-D.ietf-core-coap-tcp-tls]
              Bormann, C., Lemay, S., Tschofenig, H., Hartke, K.,
              Silverajan, B., and B. Raymor, "CoAP (Constrained
              Application Protocol) over TCP, TLS, and WebSockets",
              draft-ietf-core-coap-tcp-tls-05 (work in progress),
              October 2016.

   [I-D.savolainen-core-coap-websockets]
              Savolainen, T., Hartke, K., and B. Silverajan, "CoAP over
              WebSockets", draft-savolainen-core-coap-websockets-07
              (work in progress), June 2016.

   [I-D.silverajan-core-coap-alternative-transports]
              Silverajan, B. and T. Savolainen, "CoAP Communication with
              Alternative Transports", draft-silverajan-core-coap-
              alternative-transports-09 (work in progress), December
              2015.

   [RFC6455]  Fette, I. and A. Melnikov, "The WebSocket Protocol",
              RFC 6455, DOI 10.17487/RFC6455, December 2011,
              <http://www.rfc-editor.org/info/rfc6455>.








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Authors' Addresses

   Christian Groves
   Huawei

   Email: Christian.Groves@nteczone.com


   Weiwei Yang
   Huawei

   Email: tommy@huawei.com







































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