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Versions: (draft-thomson-rtcweb-alpn) 00 01 02 03 04 Draft is active
In: MissingRef
RTCWEB                                                        M. Thomson
Internet-Draft                                                   Mozilla
Intended status: Standards Track                           April 5, 2016
Expires: October 7, 2016


Application Layer Protocol Negotiation for Web Real-Time Communications
                                (WebRTC)
                       draft-ietf-rtcweb-alpn-03

Abstract

   Application Layer Protocol Negotiation (ALPN) labels are defined for
   use in identifying Web Real-Time Communications (WebRTC) usages of
   Datagram Transport Layer Security (DTLS).  Labels are provided for
   identifying a session that uses a combination of WebRTC compatible
   media and data, and for identifying a session requiring
   confidentiality protection from web applications.

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
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   Internet-Drafts are draft documents valid for a maximum of six months
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   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 October 7, 2016.

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
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   publication of this document.  Please review these documents
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   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of



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   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   2
     1.1.  Conventions and Terminology . . . . . . . . . . . . . . .   2
   2.  ALPN Labels for WebRTC  . . . . . . . . . . . . . . . . . . .   2
   3.  Media Confidentiality . . . . . . . . . . . . . . . . . . . .   3
   4.  Security Considerations . . . . . . . . . . . . . . . . . . .   4
   5.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .   5
   6.  References  . . . . . . . . . . . . . . . . . . . . . . . . .   6
     6.1.  Normative References  . . . . . . . . . . . . . . . . . .   6
     6.2.  Informative References  . . . . . . . . . . . . . . . . .   6
     6.3.  URIs  . . . . . . . . . . . . . . . . . . . . . . . . . .   7
   Author's Address  . . . . . . . . . . . . . . . . . . . . . . . .   7

1.  Introduction

   Web Real-Time Communications (WebRTC) [I-D.ietf-rtcweb-overview] uses
   Datagram Transport Layer Security (DTLS) [RFC6347] to secure all
   peer-to-peer communications.

   Identifying WebRTC protocol usage with Application Layer Protocol
   Negotiation (ALPN) [RFC7301] enables an endpoint to positively
   identify WebRTC uses and distinguish them from other DTLS uses.

   Different WebRTC uses can be advertised and behavior can be
   constrained to what is appropriate to a given use.  In particular,
   this allows for the identifications of sessions that require
   confidentiality protection from the application that manages the
   signaling for the session.

1.1.  Conventions and Terminology

   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].

2.  ALPN Labels for WebRTC

   The following identifiers are defined for use in ALPN:

   webrtc:  The DTLS session is used to establish keys for Secure Real-
      time Transport Protocol (SRTP) - known as DTLS-SRTP - as described
      in [RFC5764].  The DTLS record layer is used for WebRTC data
      channels [I-D.ietf-rtcweb-data-channel].



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   c-webrtc:  The DTLS session is used for confidential WebRTC
      communications, where peers agree to maintain the confidentiality
      of the media, as described in Section 3.  However, data provided
      over data channels do not receive the same level of
      confidentiality protection.

   Both identifiers describe the same basic protocol: a DTLS session
   that is used to provide keys for an SRTP session in combination with
   WebRTC data channels.  Either SRTP or data channels could be absent.
   The data channels send Stream Control Transmission Protocol (SCTP)
   [RFC4960] over the DTLS record layer, which can be multiplexed with
   SRTP on the same UDP flow.  WebRTC requires the use of Interactive
   Communication Establishment (ICE) [RFC5245] to establish the UDP
   flow, but this is not covered by the identifier.

   A more thorough definition of what WebRTC communications entail is
   included in [I-D.ietf-rtcweb-transports].

   There is no functional difference between the identifiers except that
   an endpoint negotiating "c-webrtc" makes a promise to preserve the
   confidentiality of the media it receives.

   A peer that is not aware of whether it needs to request
   confidentiality can use either form.  A peer in the client role MUST
   offer both identifiers if it is not aware of a need for
   confidentiality.  A peer in the server role SHOULD select "webrtc" if
   it does not need confidentiality protection.

3.  Media Confidentiality

   Private communications in WebRTC depend on separating control (i.e.,
   signaling) capabilities and access to media
   [I-D.ietf-rtcweb-security-arch].  In this way, an application can
   establish a session that is end-to-end confidential, where the ends
   in question are user agents (or browsers) and not the signaling
   application.  This allows an application to manage signaling for a
   session, without having access to the media that is exchanged in the
   session.

   Without some form of indication that is securely bound to the
   session, a WebRTC endpoint is unable to properly distinguish between
   a session that requires this confidentiality protection and one that
   does not.  The ALPN identifier provides that signal.

   A browser is required to enforce this confidentiality protection
   using isolation controls similar to those used in content cross-
   origin protections (see Section 5.3 [1] of [HTML5]).  These
   protections ensure that media is protected from applications.



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   Applications are not able to read or modify the contents of a
   protected flow of media.  Media that is produced from a session using
   the "c-webrtc" identifier MUST only be displayed to users.

   These confidentiality protections do not apply to data that is sent
   using data channels.  Confidential data depends on having both data
   sources and consumers that are exclusively browser- or user-based.
   No mechanisms currently exist to take advantage of data
   confidentiality, though some use cases suggest that this could be
   useful, for example, confidential peer-to-peer file transfer.
   Alternative labels might be provided in future to support these use
   cases.

   Generally speaking, ensuring confidentiality depends on
   authenticating the communications peer.  This mechanism explicitly
   does not define a specific authentication method; a WebRTC endpoint
   that accepts a session with this ALPN identifier MUST respect
   confidentiality no matter what identity is attributed to a peer.

   RTP middleboxes and entities that forward media or data cannot
   promise to maintain confidentiality.  Any entity that forwards
   content, or records content for later access by entities other than
   the authenticated peer, MUST NOT offer or accept a session with the
   "c-webrtc" identifier.

4.  Security Considerations

   Confidential communications depends on more than just an agreement
   from browsers.

   Information is not confidential if it is displayed to those other
   than to whom it is intended.  Peer authentication
   [I-D.ietf-rtcweb-security-arch] is necessary to ensure that data is
   only sent to the intended peer.

   This is not a digital rights management mechanism.  Even with an
   authenticated peer, a user is not prevented from using other
   mechanisms to record or forward media.  This means that (for example)
   screen recording devices, tape recorders, portable cameras, or a
   cunning arrangement of mirrors could variously be used to record or
   redistribute media once delivered.  Similarly, if media is visible or
   audible (or otherwise accessible) to others in the vicinity, there
   are no technical measures that protect the confidentiality of that
   media.

   The only guarantee provided by this mechanism and the browser that
   implements it is that the media was delivered to the user that was
   authenticated.  Individual users will still need to make a judgment



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   about how their peer intends to respect the confidentiality of any
   information provided.

   On a shared computing platform like a browser, other entities with
   access to that platform (i.e., web applications), might be able to
   access information that would compromise the confidentiality of
   communications.  Implementations MAY choose to limit concurrent
   access to input devices during confidential communications sessions.

   For instance, another application that is able to access a microphone
   might be able to sample confidential audio that is playing through
   speakers.  This is true even if acoustic echo cancellation, which
   attempts to prevent this from happening, is used.  Similarly, an
   application with access to a video camera might be able to use
   reflections to obtain all or part of a confidential video stream.

5.  IANA Considerations

   The following two entries are added to the "Application Layer
   Protocol Negotiation (ALPN) Protocol IDs" registry established by
   [RFC7301]:

   webrtc:

      The "webrtc" label identifies mixed media and data communications
      using SRTP and data channels:

      Protocol:  WebRTC Media and Data

      Identification Sequence:  0x77 0x65 0x62 0x72 0x74 0x63 ("webrtc")

      Specification:  This document (RFCXXXX)

   c-webrtc:

      The "c-webrtc" label identifies confidential WebRTC
      communications:

      Protocol:  Confidential WebRTC Media and Data

      Identification Sequence:  0x63 0x2d 0x77 0x65 0x62 0x72 0x74 0x63
         ("c-webrtc")

      Specification:  This document (RFCXXXX)







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6.  References

6.1.  Normative References

   [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.

   [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>.

   [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,
              DOI 10.17487/RFC5764, May 2010,
              <http://www.rfc-editor.org/info/rfc5764>.

   [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>.

   [RFC7301]  Friedl, S., Popov, A., Langley, A., and E. Stephan,
              "Transport Layer Security (TLS) Application-Layer Protocol
              Negotiation Extension", RFC 7301, DOI 10.17487/RFC7301,
              July 2014, <http://www.rfc-editor.org/info/rfc7301>.

6.2.  Informative References

   [HTML5]    Berjon, R., Leithead, T., Doyle Navara, E., O'Connor, E.,
              and S. Pfeiffer, "HTML 5", CR CR-html5-20121217, August
              2010, <http://www.w3.org/TR/2012/CR-html5-20121217/>.

   [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-arch]
              Rescorla, E., "WebRTC Security Architecture", draft-ietf-
              rtcweb-security-arch-11 (work in progress), March 2015.

   [I-D.ietf-rtcweb-transports]
              Alvestrand, H., "Transports for WebRTC", draft-ietf-
              rtcweb-transports-12 (work in progress), March 2016.




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   [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>.

6.3.  URIs

   [1] http://www.w3.org/TR/2012/CR-html5-20121217/browsers.html#origin

Author's Address

   Martin Thomson
   Mozilla
   331 E Evelyn Street
   Mountain View, CA  94041
   US

   Email: martin.thomson@gmail.com




























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