Internet Engineering Task Force (IETF) M. Thomson
Request for Comments: 8833 Mozilla
Category: Standards Track May 5, 2016
Expires: November 6, 2016
Application Layer 2020
Application-Layer Protocol Negotiation (ALPN) for Web Real-Time Communications
This document specifies two Application Layer 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 to establish keys for the Secure Real-time
Transport Protocol (SRTP) or to establish data channels using SCTP the
Stream Control Transmission Protocol (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 an Internet Standards Track document.
This document is a product of the Internet Engineering Task Force
(IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list It represents the consensus of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid the IETF community. It has
received public review and has been approved for a maximum publication by the
Internet Engineering Steering Group (IESG). Further information on
Internet Standards is available in Section 2 of RFC 7841.
Information about the current status of six months this document, any errata,
and how to provide feedback on it may be updated, replaced, or obsoleted by other documents obtained 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 (c) 2016 2020 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
(https://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
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
Web Real-Time Communications (WebRTC) [I-D.ietf-rtcweb-overview] [RFC8825] uses Datagram
Transport Layer Security (DTLS) [RFC6347] to secure all peer-to-peer
Identifying WebRTC protocol usage with Application Layer 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",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
"OPTIONAL" in this document are to be interpreted as described in
14 [RFC2119] [RFC8174] when, and only when, they appear in all
capitals, as shown here.
2. ALPN Labels for WebRTC
The following identifiers are defined for use in ALPN:
webrtc: The DTLS session is used to establish keys for the Secure Real-
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]. [RFC8831].
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 the 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
Communication Connectivity Establishment (ICE) [RFC5245] [RFC8445] 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]. [RFC8835].
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]. [RFC8827]. 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, 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  of [HTML5]). These protections
ensure that media is protected from applications.
Applications applications, which 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- browser
or user-based. 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 the future to support these
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
4. Security Considerations
Confidential communications depends depend on more than just an agreement
Information is not confidential if it is displayed to those other others than to for
whom it is intended. Peer authentication
[I-D.ietf-rtcweb-security-arch] [RFC8827] is necessary to
ensure that data is only sent to the intended peer.
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 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
On a shared computing platform like a browser, other entities with
access to that platform (i.e., web applications), 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 have been added to the "Application "TLS Application-
Layer Protocol Negotiation (ALPN) Protocol IDs" registry established
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) RFC 8833 (this document)
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) RFC 8833 (this document)
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, <http://www.rfc-editor.org/info/rfc6347>. <https://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>. <https://www.rfc-editor.org/info/rfc7301>.
[RFC8174] Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC
2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174,
May 2017, <https://www.rfc-editor.org/info/rfc8174>.
[RFC8827] Rescorla, E., "WebRTC Security Architecture", RFC 8827,
DOI 10.17487/RFC8827, May 2020,
[RFC8831] Jesup, R., Loreto, S., and M. Tüxen, "WebRTC Data
Channels", RFC 8831, DOI 10.17487/RFC8831, May 2020,
6.2. Informative References
[HTML5] Berjon, R., Leithead, T., Doyle Navara, E., O'Connor, E.,
and S. Pfeiffer, "HTML 5", CR W3C Candidate Recommendation
CR-html5-20121217, August 2010,
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,
[RFC8445] Keranen, A., Holmberg, C., and J. Rosenberg, J., "Interactive
Connectivity Establishment (ICE): A Protocol for Network
Address Translator (NAT)
Traversal Traversal", RFC 8445,
DOI 10.17487/RFC8445, July 2018,
[RFC8825] Alvestrand, H., "Overview: Real-Time Protocols for Offer/Answer Protocols",
Browser-Based Applications", RFC 5245, 8825,
DOI 10.17487/RFC5245, April 2010,
 http://www.w3.org/TR/2012/CR-html5-20121217/browsers.html#origin 10.17487/RFC8825, May 2020,
[RFC8835] Alvestrand, H., "Transports for WebRTC", RFC 8835,
DOI 10.17487/RFC8835, May 2020,
331 E Evelyn Street
Mountain View, CA 94041
United States of America