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Application Layer Protocol Negotiation for Web Real-Time Communications (WebRTC)

The information below is for an old version of the document.
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This is an older version of an Internet-Draft that was ultimately published as RFC 8833.
Expired & archived
Author Martin Thomson
Last updated 2015-01-24 (Latest revision 2014-07-23)
RFC stream Internet Engineering Task Force (IETF)
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Stream WG state WG Document
Document shepherd Sean Turner
IESG IESG state Became RFC 8833 (Proposed Standard)
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RTCWEB                                                        M. Thomson
Internet-Draft                                                   Mozilla
Intended status: Standards Track                           July 23, 2014
Expires: January 24, 2015

Application Layer Protocol Negotiation for Web Real-Time Communications


   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

   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 January 24, 2015.

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
   ( 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

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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.

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

2.  ALPN Labels for WebRTC

   The following identifiers 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].

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   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].

   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 MAY 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 with
   respect to the promise that an endpoint makes with respect to the
   confidentiality of session content.  An endpoint negotiating
   "c-webrtc" makes a promise to preserve the confidentiality of the
   data it receives.

   Only one of these labels can be used for any given session.  A peer
   acting in the client role MUST NOT offer both identifiers.  A peer in
   the server role that receives a ClientHello containing both labels
   MUST reject the session, though it MAY accept the confidential option
   and protect content accordingly.

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

   A browser is required to enforce this control using isolation
   controls similar to those used in cross-origin protections.  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.

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   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

   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.

   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.

   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)

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   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

   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 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

   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

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   Identification Sequence:  0x63 0x2d 0x77 0x65 0x62 0x72 0x74 0x63

   Specification:  This document (RFCXXXX)

6.  References

6.1.  Normative References

              Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data
              Channels", draft-ietf-rtcweb-data-channel-09 (work in
              progress), May 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.

   [RFC7301]  Friedl, S., Popov, A., Langley, A., and E. Stephan,
              "Transport Layer Security (TLS) Application-Layer Protocol
              Negotiation Extension", RFC 7301, July 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, <>.

              Alvestrand, H., "Overview: Real Time Protocols for Brower-
              based Applications", draft-ietf-rtcweb-overview-09 (work
              in progress), February 2014.

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

              Alvestrand, H., "Transports for RTCWEB", draft-ietf-
              rtcweb-transports-04 (work in progress), April 2014.

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   [RFC4960]  Stewart, R., "Stream Control Transmission Protocol", RFC
              4960, September 2007.

   [RFC5245]  Rosenberg, J., "Interactive Connectivity Establishment
              (ICE): A Protocol for Network Address Translator (NAT)
              Traversal for Offer/Answer Protocols", RFC 5245, April

6.3.  URIs


Author's Address

   Martin Thomson
   331 E Evelyn Street
   Mountain View, CA  94041


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