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Application-Layer Protocol Negotiation (ALPN) for WebRTC

The information below is for an old version of the document that is already published as an RFC.
Document Type
This is an older version of an Internet-Draft that was ultimately published as RFC 8833.
Author Martin Thomson
Last updated 2021-01-18 (Latest revision 2016-05-05)
Replaces draft-thomson-rtcweb-alpn
RFC stream Internet Engineering Task Force (IETF)
Intended RFC status Proposed Standard
Additional resources Mailing list discussion
Stream WG state Submitted to IESG for Publication
Document shepherd Sean Turner
Shepherd write-up Show Last changed 2016-02-25
IESG IESG state Became RFC 8833 (Proposed Standard)
Action Holders
Consensus boilerplate Yes
Telechat date (None)
Responsible AD Alissa Cooper
Send notices to (None)
IANA IANA review state Version Changed - Review Needed
IANA action state RFC-Ed-Ack
RTCWEB                                                        M. Thomson
Internet-Draft                                                   Mozilla
Intended status: Standards Track                             May 5, 2016
Expires: November 6, 2016

Application Layer Protocol Negotiation for Web Real-Time Communications


   This document specifies two Application Layer Protocol Negotiation
   (ALPN) labels for use with Web Real-Time Communications (WebRTC).
   The "webrtc" label identifies regular WebRTC communications: a DTLS
   session that is used establish keys for Secure Real-time Transport
   Protocol (SRTP) or to establish data channels using SCTP over DTLS.
   The "c-webrtc" label describes the same protocol, but the peers also
   agree to maintain the confidentiality of the media by not sharing it
   with other 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
   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 November 6, 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
   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

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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
   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 identification 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",
   "OPTIONAL" in this document are to be interpreted as described 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 Secure Real-
      time Transport Protocol (SRTP) - known as DTLS-SRTP - as described

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      in [RFC5764].  The DTLS record layer is used for WebRTC data
      channels [I-D.ietf-rtcweb-data-channel].

   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.  The confidentiality
      protections ensure that media is protected from other
      applications, but the confidentiality protections do not extend to
      messages on data channels.

   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 identifier.  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 prefer either.

   An endpoint that requires media confidentiality might negotiate a
   session with a peer that does not support this specification.
   Endpoint MUST abort a session if it requires confidentiality but does
   not successfully negotiate "c-webrtc".  A peer that is willing to
   accept "webrtc" SHOULD assume that a peer that does not support this
   specification has negotiated "webrtc" unless signaling provides other
   information; however, a peer MUST NOT assume that "c-webrtc" has been
   negotiated unless explicitly negotiated.

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

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   application.  This allows an application to manage signaling for a
   session, without having access to the media that is exchanged in the

   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.
   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.

   The promise to apply 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

   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.

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   This is not a digital rights management mechanism.  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
   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


      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)


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      The "c-webrtc" label identifies WebRTC communications with a
      promise to protect media confidentiality:

      Protocol:  Confidential WebRTC Media and Data

      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-13 (work in
              progress), January 2015.

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

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119,
              DOI 10.17487/RFC2119, 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,
              DOI 10.17487/RFC5764, May 2010,

   [RFC6347]  Rescorla, E. and N. Modadugu, "Datagram Transport Layer
              Security Version 1.2", RFC 6347, DOI 10.17487/RFC6347,
              January 2012, <>.

   [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, <>.

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, <>.

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

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

   [RFC4960]  Stewart, R., Ed., "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,

6.3.  URIs


Author's Address

   Martin Thomson
   331 E Evelyn Street
   Mountain View, CA  94041


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