[Docs] [txt|pdf|xml|html] [Tracker] [Email] [Nits]

Versions: 00 draft-ietf-rtcweb-alpn

RTCWEB                                                        M. Thomson
Internet-Draft                                                   Mozilla
Intended status: Standards Track                           April 9, 2014
Expires: October 11, 2014


Application Layer Protocol Negotiation for Web Real-Time Communications
                                (WebRTC)
                      draft-thomson-rtcweb-alpn-00

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.

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 October 11, 2014.

Copyright Notice

   Copyright (c) 2014 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
   include Simplified BSD License text as described in Section 4.e of



Thomson                 Expires October 11, 2014                [Page 1]


Internet-Draft               ALPN for WebRTC                  April 2014


   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  . . . . . . . . . . . . . . . . . . . . . . . . .   5
     6.1.  Normative References  . . . . . . . . . . . . . . . . . .   5
     6.2.  Informative References  . . . . . . . . . . . . . . . . .   6
     6.3.  URIs  . . . . . . . . . . . . . . . . . . . . . . . . . .   6
   Author's Address  . . . . . . . . . . . . . . . . . . . . . . . .   6

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) [I-D.ietf-tls-applayerprotoneg] 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.

1.1.  Conventions and Terminology

   At times, this document falls back on shorthands for establishing
   interoperability requirements on implementations: the capitalized
   words "MUST", "SHOULD" and "MAY".  These terms are defined in
   [RFC2119].

2.  ALPN Labels for WebRTC

   The following four labels are defined for use in ALPN:

   webrtc  The DTLS session is used to establish keys for a 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].



Thomson                 Expires October 11, 2014                [Page 2]


Internet-Draft               ALPN for WebRTC                  April 2014


   c-webrtc  The DTLS session is used for confidential WebRTC
      communications, where peers agree to maintain the confidentiality
      of the communications, as described in Section 3.

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

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.

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

   A browser is required to enforce confidentiality 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.  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.

   A WebRTC implementation MUST only send media over a "c-webrtc"
   session that comes from user-controlled sources.  Media from sources
   that are controlled by third parties (such as the signaling
   application) could be used to create confusion about the content and
   origin of data.  This cannot prevent confusion when rendered as a web
   page, for example, but browsers are required to provide ways to
   identify media and its provenance (see Section 5.5 of
   [I-D.ietf-rtcweb-security-arch]).

   Confidentiality protections of this sort are not expected to be
   possible for data that is sent using data channels.  Thus, it is
   expected that data channels will not be employed for sessions that
   negotiate confidentiality.  In the browser context, 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.




Thomson                 Expires October 11, 2014                [Page 3]


Internet-Draft               ALPN for WebRTC                  April 2014


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

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.  In other cases, effects might not be temporally localized:
   transmitted smells could linger for a period after communications
   cease.

   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
   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 session.

   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 being possible, is used.  Similarly, an
   application with access to a video camera might be able to use
   reflections to access confidential video.





Thomson                 Expires October 11, 2014                [Page 4]


Internet-Draft               ALPN for WebRTC                  April 2014


5.  IANA Considerations

   The following two entries are added to the "Application Layer
   Protocol Negotiation (ALPN) Protocol IDs" registry established by
   [I-D.ietf-tls-applayerprotoneg].

   The "webrtc" 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)

   The "c-webrtc" 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)

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-07 (work in
              progress), February 2014.

   [I-D.ietf-tls-applayerprotoneg]
              Friedl, S., Popov, A., Langley, A., and S. Emile,
              "Transport Layer Security (TLS) Application Layer Protocol
              Negotiation Extension", draft-ietf-tls-applayerprotoneg-05
              (work in progress), March 2014.

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.

   [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.

   [RFC6347]  Rescorla, E. and N. Modadugu, "Datagram Transport Layer
              Security Version 1.2", RFC 6347, January 2012.



Thomson                 Expires October 11, 2014                [Page 5]


Internet-Draft               ALPN for WebRTC                  April 2014


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 Brower-
              based Applications", draft-ietf-rtcweb-overview-09 (work
              in progress), February 2014.

   [I-D.ietf-rtcweb-security-arch]
              Rescorla, E., "WebRTC Security Architecture", draft-ietf-
              rtcweb-security-arch-09 (work in progress), February 2014.

   [I-D.ietf-rtcweb-transports]
              Alvestrand, H., "Transports for RTCWEB", draft-ietf-
              rtcweb-transports-03 (work in progress), March 2014.

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



















Thomson                 Expires October 11, 2014                [Page 6]


Html markup produced by rfcmarkup 1.129c, available from https://tools.ietf.org/tools/rfcmarkup/