One document matched: draft-ietf-rtcweb-alpn-00.txt
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
(WebRTC)
draft-ietf-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
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This Internet-Draft will expire on January 24, 2015.
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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
[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 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
application.
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
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.
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
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 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].
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
("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-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, <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-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
2010.
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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