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
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |Ver| T |  TKL  |      Code     | TKL bytes ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |  Options (if any) ...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |1 1 1 1 1 1 1 1|    Payload (if any)...
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
           

Figure 4: CoAP Header with SCTP with 8-bit Length in Header

The "Version (Ver)" and "Type (T)" fields are re-introduced as compared to [I-D.ietf-core-coap-tcp-tls] to maintain compatibility with CoAP [RFC7252]. Thus the above header is only different from the CoAP header in that 2 bytes are saved due to the removal of the "Message ID".

Version (Ver) is effectively redundant due to the negotiation of the sub-protocol "coap.v1" at data channel establishment. This field MUST match the version indicated in the protocol field.

The use of the message type (T) is used as an indication to the CoAP receiver as to the characteristics of the message. The receiver MUST NOT use this information to implement a CoAP level reliability mechanism. Sections 5.2 and 5.3 detail the usage of message types.

Author's Note 1: The 4 bits required for "Version" and "Type" could be removed however this would mean that octet aligned boundaries would no longer be reserved. [I-D.savolainen-core-coap-websockets] instead indicates that the four most significant bits (i.e. those for Ver and T) be set to zero and are reserved.

Author's Note 2: A further alternative option is to use the CoAP header as defined in [RFC7252] and to indicate that for CoAP over WebRTC DC the "T" and "Message ID" are populated with "0-bits" when sending and ignored when receiving messages.

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

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

4.1. 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.

5. 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 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.

5.1. Messages and Endpoints

As per section 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.

5.2. 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.

5.3. 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 [3].

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. However to facilitate potential interworking at the receiver the sender SHALL still indicate a "Non-confirmable" message type in a CoAP request/response when required.

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

5.4. Message Correlation

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

5.5. Message Duplication

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

5.6. 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 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].

5.7. Congestion Control

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

5.8. Transmission Parameters

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

6. Request/Response Semantics

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

• 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.
• section 5.3.1/[RFC7252]: due to the use of DTLS the advice regarding token use without using TLS is invalid.
• section 5.3.2/[RFC7252]: In addition CoAP request/response matching is unique to a particular WebRTC DC (SCTP StreamID pair).
• section 5.8/[RFC7252]: It is not possible to use a 4.05 piggybacked response.

7. 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, 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.

7.1. 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:

• 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.

8. Discovery

8.1. 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.

8.2. Resource Discovery

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

9. Multicast CoAP

WebRTC DCs do not support multicast.

10. 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].

11. 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:

• Map received Empty CoAP message to SCTP level operations and discard the empty message.
• Map received ACK message to SCTP level operations and discard the ACK message.
• Separate piggy-backed messages.
• 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.

12. Security Considerations

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

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

• section 11.3/[RFC7252] on insecure UDP and multicast being used to aid an amplification attack.
• section 11.4/[RFC7252] on IP address spoofing and section 11.5/[RFC7252] on Cross-Protocol attacks.
• 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].

13. IANA Considerations

13.1. 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.

Subprotocol Identifier: coap.v1

Subprotocol Common Name: Constrained Application Protocol (CoAP)

Subprotocol Definition: [This document]

13.2. 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.

13.3. 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.

13.4. 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].

13.5. 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 draft]

Security Considerations: Security considerations for this media feature tag are discussed in Section 12.

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

CORE workgroup: core@ietf.org

CORE chairs: core-chairs@ietf.org

Intended usage: COMMON

14. 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.

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"
           

15. 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.

This template was derived from an initial version written by Pekka Savola and contributed by him to the xml2rfc project.

16. References

16.1. Normative References

[I-D.ietf-mmusic-data-channel-sdpneg]	Drage, K., Makaraju, M., Stoetzer-Bradler, J., Ejzak, R. and J. Marcon, "SDP-based Data Channel Negotiation", Internet-Draft draft-ietf-mmusic-data-channel-sdpneg-08, February 2016.
[I-D.ietf-rtcweb-alpn]	Thomson, M., "Application Layer Protocol Negotiation for Web Real-Time Communications (WebRTC)", Internet-Draft draft-ietf-rtcweb-alpn-04, May 2016.
[I-D.ietf-rtcweb-data-channel]	Jesup, R., Loreto, S. and M. Tuexen, "WebRTC Data Channels", Internet-Draft draft-ietf-rtcweb-data-channel-13, January 2015.
[I-D.ietf-rtcweb-data-protocol]	Jesup, R., Loreto, S. and M. Tuexen, "WebRTC Data Channel Establishment Protocol", Internet-Draft draft-ietf-rtcweb-data-protocol-09, January 2015.
[I-D.ietf-rtcweb-jsep]	Uberti, J., Jennings, C. and E. Rescorla, "Javascript Session Establishment Protocol", Internet-Draft draft-ietf-rtcweb-jsep-14, March 2016.
[I-D.ietf-rtcweb-overview]	Alvestrand, H., "Overview: Real Time Protocols for Browser-based Applications", Internet-Draft draft-ietf-rtcweb-overview-15, January 2016.
[I-D.ietf-rtcweb-security]	Rescorla, E., "Security Considerations for WebRTC", Internet-Draft draft-ietf-rtcweb-security-08, February 2015.
[I-D.ietf-rtcweb-security-arch]	Rescorla, E., "WebRTC Security Architecture", Internet-Draft draft-ietf-rtcweb-security-arch-12, June 2016.
[I-D.ietf-tsvwg-sctp-dtls-encaps]	Tuexen, M., Stewart, R., Jesup, R. and S. Loreto, "DTLS Encapsulation of SCTP Packets", Internet-Draft draft-ietf-tsvwg-sctp-dtls-encaps-09, 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", Internet-Draft draft-ietf-tsvwg-sctp-ndata-06, 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.
[RFC2506]	Holtman, K., Mutz, A. and T. Hardie, "Media Feature Tag Registration Procedure", BCP 31, RFC 2506, DOI 10.17487/RFC2506, March 1999.
[RFC3264]	Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with Session Description Protocol (SDP)", RFC 3264, DOI 10.17487/RFC3264, June 2002.
[RFC3311]	Rosenberg, J., "The Session Initiation Protocol (SIP) UPDATE Method", RFC 3311, DOI 10.17487/RFC3311, October 2002.
[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.
[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.
[RFC4566]	Handley, M., Jacobson, V. and C. Perkins, "SDP: Session Description Protocol", RFC 4566, DOI 10.17487/RFC4566, July 2006.
[RFC4960]	Stewart, R., "Stream Control Transmission Protocol", RFC 4960, DOI 10.17487/RFC4960, September 2007.
[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.
[RFC5630]	Audet, F., "The Use of the SIPS URI Scheme in the Session Initiation Protocol (SIP)", RFC 5630, DOI 10.17487/RFC5630, October 2009.
[RFC6347]	Rescorla, E. and N. Modadugu, "Datagram Transport Layer Security Version 1.2", RFC 6347, DOI 10.17487/RFC6347, January 2012.
[RFC6525]	Stewart, R., Tuexen, M. and P. Lei, "Stream Control Transmission Protocol (SCTP) Stream Reconfiguration", RFC 6525, DOI 10.17487/RFC6525, February 2012.
[RFC6690]	Shelby, Z., "Constrained RESTful Environments (CoRE) Link Format", RFC 6690, DOI 10.17487/RFC6690, August 2012.
[RFC7228]	Bormann, C., Ersue, M. and A. Keranen, "Terminology for Constrained-Node Networks", RFC 7228, DOI 10.17487/RFC7228, May 2014.
[RFC7252]	Shelby, Z., Hartke, K. and C. Bormann, "The Constrained Application Protocol (CoAP)", RFC 7252, DOI 10.17487/RFC7252, June 2014.
[RFC7595]	Thaler, D., Hansen, T. and T. Hardie, "Guidelines and Registration Procedures for URI Schemes", BCP 35, RFC 7595, DOI 10.17487/RFC7595, June 2015.
[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.

16.2. Informative References

[I-D.bormann-core-coap-sig]	Bormann, C., "CoAP Signaling Messages", Internet-Draft draft-bormann-core-coap-sig-02, June 2016.
[I-D.ietf-core-coap-tcp-tls]	Bormann, C., Lemay, S., Technologies, Z. and H. Tschofenig, "A TCP and TLS Transport for the Constrained Application Protocol (CoAP)", Internet-Draft draft-ietf-core-coap-tcp-tls-02, April 2016.
[I-D.savolainen-core-coap-websockets]	Savolainen, T., Hartke, K. and B. Silverajan, "CoAP over WebSockets", Internet-Draft draft-savolainen-core-coap-websockets-07, June 2016.
[I-D.silverajan-core-coap-alternative-transports]	Silverajan, B. and T. Savolainen, "CoAP Communication with Alternative Transports", Internet-Draft draft-silverajan-core-coap-alternative-transports-09, December 2015.
[RFC6455]	Fette, I. and A. Melnikov, "The WebSocket Protocol", RFC 6455, DOI 10.17487/RFC6455, December 2011.

Appendix A. Additional Stuff

This becomes an Appendix.

Authors' Addresses