Application-Layer Protocol Negotiation (ALPN) for WebRTC
RFC 8833
| Document | Type | RFC - Proposed Standard (January 2021; No errata) | |
|---|---|---|---|
| Author | Martin Thomson | ||
| Last updated | 2021-01-18 | ||
| Replaces | draft-thomson-rtcweb-alpn | ||
| Stream | Internet Engineering Task Force (IETF) | ||
| Formats | plain text html xml pdf htmlized bibtex | ||
| Reviews | |||
| Stream | WG state | Submitted to IESG for Publication | |
| Document shepherd | Sean Turner | ||
| Shepherd write-up | Show (last changed 2016-02-25) | ||
| IESG | IESG state | RFC 8833 (Proposed Standard) | |
| Action Holders |
(None)
|
||
| Consensus Boilerplate | Yes | ||
| Telechat date | |||
| Responsible AD | Alissa Cooper | ||
| Send notices to | (None) | ||
| IANA | IANA review state | Version Changed - Review Needed | |
| IANA action state | RFC-Ed-Ack | ||
Internet Engineering Task Force (IETF) M. Thomson
Request for Comments: 8833 Mozilla
Category: Standards Track January 2021
ISSN: 2070-1721
Application-Layer Protocol Negotiation (ALPN) for WebRTC
Abstract
This document specifies two Application-Layer Protocol Negotiation
(ALPN) labels for use with Web Real-Time Communication (WebRTC). The
"webrtc" label identifies regular WebRTC: a DTLS session that is used
to establish keys for the Secure Real-time Transport Protocol (SRTP)
or to establish data channels using 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 is an Internet Standards Track document.
This document is a product of the Internet Engineering Task Force
(IETF). It represents the consensus of the IETF community. It has
received public review and has been approved for 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 this document, any errata,
and how to provide feedback on it may be obtained at
https://www.rfc-editor.org/info/rfc8833.
Copyright Notice
Copyright (c) 2021 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
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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
1.1. Conventions
2. ALPN Labels for WebRTC
3. Media Confidentiality
4. Security Considerations
5. IANA Considerations
6. References
6.1. Normative References
6.2. Informative References
Author's Address
1. Introduction
Web Real-Time Communication (WebRTC) [RFC8825] 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
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 BCP
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-time Transport Protocol (SRTP) -- known as DTLS-SRTP -- as
described in [RFC5764]. The DTLS record layer is used for WebRTC
data channels [RFC8831].
c-webrtc: The DTLS session is used for confidential WebRTC, 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
Interactive Connectivity Establishment (ICE) [RFC8445] to establish
UDP flow, but this is not covered by the identifier.
A more thorough definition of what WebRTC entails is included in
[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. An
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 [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 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 the "Origin" section of [HTML5]). These
protections ensure that media is protected from 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
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 the future to support these
use cases.
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 depend on more than just an agreement
from browsers.
Information is not confidential if it is displayed to others than for
whom it is intended. Peer authentication [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 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 have been added to the "TLS Application-
Layer Protocol Negotiation (ALPN) Protocol IDs" registry established
by [RFC7301]:
webrtc:
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: RFC 8833 (this document)
c-webrtc:
The "c-webrtc" label identifies WebRTC with a promise to protect
media confidentiality:
Protocol: Confidential WebRTC Media and Data
Identification Sequence: 0x63 0x2d 0x77 0x65 0x62 0x72 0x74 0x63
("c-webrtc")
Specification: RFC 8833 (this document)
6. References
6.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997,
<https://www.rfc-editor.org/info/rfc2119>.
[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,
<https://www.rfc-editor.org/info/rfc5764>.
[RFC6347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer
Security Version 1.2", RFC 6347, DOI 10.17487/RFC6347,
January 2012, <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, <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, January 2021,
<https://www.rfc-editor.org/info/rfc8827>.
[RFC8831] Jesup, R., Loreto, S., and M. Tüxen, "WebRTC Data
Channels", RFC 8831, DOI 10.17487/RFC8831, January 2021,
<https://www.rfc-editor.org/info/rfc8831>.
6.2. Informative References
[HTML5] WHATWG, "HTML - Living Standard", Section 7.5, January
2021, <https://html.spec.whatwg.org/#origin>.
[RFC4960] Stewart, R., Ed., "Stream Control Transmission Protocol",
RFC 4960, DOI 10.17487/RFC4960, September 2007,
<https://www.rfc-editor.org/info/rfc4960>.
[RFC8445] Keranen, A., Holmberg, C., and J. Rosenberg, "Interactive
Connectivity Establishment (ICE): A Protocol for Network
Address Translator (NAT) Traversal", RFC 8445,
DOI 10.17487/RFC8445, July 2018,
<https://www.rfc-editor.org/info/rfc8445>.
[RFC8825] Alvestrand, H., "Overview: Real-Time Protocols for
Browser-Based Applications", RFC 8825,
DOI 10.17487/RFC8825, January 2021,
<https://www.rfc-editor.org/info/rfc8825>.
[RFC8835] Alvestrand, H., "Transports for WebRTC", RFC 8835,
DOI 10.17487/RFC8835, January 2021,
<https://www.rfc-editor.org/info/rfc8835>.
Author's Address
Martin Thomson
Mozilla
Email: martin.thomson@gmail.com