draft-ietf-rtcweb-alpn-03.txt   draft-ietf-rtcweb-alpn-04.txt 
RTCWEB M. Thomson RTCWEB M. Thomson
Internet-Draft Mozilla Internet-Draft Mozilla
Intended status: Standards Track April 5, 2016 Intended status: Standards Track May 5, 2016
Expires: October 7, 2016 Expires: November 6, 2016
Application Layer Protocol Negotiation for Web Real-Time Communications Application Layer Protocol Negotiation for Web Real-Time Communications
(WebRTC) (WebRTC)
draft-ietf-rtcweb-alpn-03 draft-ietf-rtcweb-alpn-04
Abstract Abstract
Application Layer Protocol Negotiation (ALPN) labels are defined for This document specifies two Application Layer Protocol Negotiation
use in identifying Web Real-Time Communications (WebRTC) usages of (ALPN) labels for use with Web Real-Time Communications (WebRTC).
Datagram Transport Layer Security (DTLS). Labels are provided for The "webrtc" label identifies regular WebRTC communications: a DTLS
identifying a session that uses a combination of WebRTC compatible session that is used establish keys for Secure Real-time Transport
media and data, and for identifying a session requiring Protocol (SRTP) or to establish data channels using SCTP over DTLS.
confidentiality protection from web applications. 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 Status of This Memo
This Internet-Draft is submitted in full conformance with the This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79. provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet- working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/. Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress." material or to cite them other than as "work in progress."
This Internet-Draft will expire on October 7, 2016. This Internet-Draft will expire on November 6, 2016.
Copyright Notice Copyright Notice
Copyright (c) 2016 IETF Trust and the persons identified as the Copyright (c) 2016 IETF Trust and the persons identified as the
document authors. All rights reserved. document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of (http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents publication of this document. Please review these documents
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Web Real-Time Communications (WebRTC) [I-D.ietf-rtcweb-overview] uses Web Real-Time Communications (WebRTC) [I-D.ietf-rtcweb-overview] uses
Datagram Transport Layer Security (DTLS) [RFC6347] to secure all Datagram Transport Layer Security (DTLS) [RFC6347] to secure all
peer-to-peer communications. peer-to-peer communications.
Identifying WebRTC protocol usage with Application Layer Protocol Identifying WebRTC protocol usage with Application Layer Protocol
Negotiation (ALPN) [RFC7301] enables an endpoint to positively Negotiation (ALPN) [RFC7301] enables an endpoint to positively
identify WebRTC uses and distinguish them from other DTLS uses. identify WebRTC uses and distinguish them from other DTLS uses.
Different WebRTC uses can be advertised and behavior can be Different WebRTC uses can be advertised and behavior can be
constrained to what is appropriate to a given use. In particular, constrained to what is appropriate to a given use. In particular,
this allows for the identifications of sessions that require this allows for the identification of sessions that require
confidentiality protection from the application that manages the confidentiality protection from the application that manages the
signaling for the session. signaling for the session.
1.1. Conventions and Terminology 1.1. Conventions and Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
"OPTIONAL" in this document are to be interpreted as described in "OPTIONAL" in this document are to be interpreted as described in
[RFC2119]. [RFC2119].
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The following identifiers are defined for use in ALPN: The following identifiers are defined for use in ALPN:
webrtc: The DTLS session is used to establish keys for Secure Real- webrtc: The DTLS session is used to establish keys for Secure Real-
time Transport Protocol (SRTP) - known as DTLS-SRTP - as described time Transport Protocol (SRTP) - known as DTLS-SRTP - as described
in [RFC5764]. The DTLS record layer is used for WebRTC data in [RFC5764]. The DTLS record layer is used for WebRTC data
channels [I-D.ietf-rtcweb-data-channel]. channels [I-D.ietf-rtcweb-data-channel].
c-webrtc: The DTLS session is used for confidential WebRTC c-webrtc: The DTLS session is used for confidential WebRTC
communications, where peers agree to maintain the confidentiality communications, where peers agree to maintain the confidentiality
of the media, as described in Section 3. However, data provided of the media, as described in Section 3. The confidentiality
over data channels do not receive the same level of protections ensure that media is protected from other
confidentiality protection. applications, but the confidentiality protections do not extend to
messages on data channels.
Both identifiers describe the same basic protocol: a DTLS session Both identifiers describe the same basic protocol: a DTLS session
that is used to provide keys for an SRTP session in combination with that is used to provide keys for an SRTP session in combination with
WebRTC data channels. Either SRTP or data channels could be absent. WebRTC data channels. Either SRTP or data channels could be absent.
The data channels send Stream Control Transmission Protocol (SCTP) The data channels send Stream Control Transmission Protocol (SCTP)
[RFC4960] over the DTLS record layer, which can be multiplexed with [RFC4960] over the DTLS record layer, which can be multiplexed with
SRTP on the same UDP flow. WebRTC requires the use of Interactive SRTP on the same UDP flow. WebRTC requires the use of Interactive
Communication Establishment (ICE) [RFC5245] to establish the UDP Communication Establishment (ICE) [RFC5245] to establish the UDP
flow, but this is not covered by the identifier. flow, but this is not covered by the identifier.
A more thorough definition of what WebRTC communications entail is A more thorough definition of what WebRTC communications entail is
included in [I-D.ietf-rtcweb-transports]. included in [I-D.ietf-rtcweb-transports].
There is no functional difference between the identifiers except that There is no functional difference between the identifiers except that
an endpoint negotiating "c-webrtc" makes a promise to preserve the an endpoint negotiating "c-webrtc" makes a promise to preserve the
confidentiality of the media it receives. confidentiality of the media it receives.
A peer that is not aware of whether it needs to request A peer that is not aware of whether it needs to request
confidentiality can use either form. A peer in the client role MUST confidentiality can use either identifier. A peer in the client role
offer both identifiers if it is not aware of a need for MUST offer both identifiers if it is not aware of a need for
confidentiality. A peer in the server role SHOULD select "webrtc" if confidentiality. A peer in the server role SHOULD select "webrtc" if
it does not need confidentiality protection. 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 3. Media Confidentiality
Private communications in WebRTC depend on separating control (i.e., Private communications in WebRTC depend on separating control (i.e.,
signaling) capabilities and access to media signaling) capabilities and access to media
[I-D.ietf-rtcweb-security-arch]. In this way, an application can [I-D.ietf-rtcweb-security-arch]. In this way, an application can
establish a session that is end-to-end confidential, where the ends establish a session that is end-to-end confidential, where the ends
in question are user agents (or browsers) and not the signaling in question are user agents (or browsers) and not the signaling
application. This allows an application to manage signaling for a application. This allows an application to manage signaling for a
session, without having access to the media that is exchanged in the session, without having access to the media that is exchanged in the
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Without some form of indication that is securely bound to the Without some form of indication that is securely bound to the
session, a WebRTC endpoint is unable to properly distinguish between session, a WebRTC endpoint is unable to properly distinguish between
a session that requires this confidentiality protection and one that a session that requires this confidentiality protection and one that
does not. The ALPN identifier provides that signal. does not. The ALPN identifier provides that signal.
A browser is required to enforce this confidentiality protection A browser is required to enforce this confidentiality protection
using isolation controls similar to those used in content cross- using isolation controls similar to those used in content cross-
origin protections (see Section 5.3 [1] of [HTML5]). These origin protections (see Section 5.3 [1] of [HTML5]). These
protections ensure that media is protected from applications. protections ensure that media is protected from applications.
Applications are not able to read or modify the contents of a Applications are not able to read or modify the contents of a
protected flow of media. Media that is produced from a session using protected flow of media. Media that is produced from a session using
the "c-webrtc" identifier MUST only be displayed to users. the "c-webrtc" identifier MUST only be displayed to users.
These confidentiality protections do not apply to data that is sent The promise to apply confidentiality protections do not apply to data
using data channels. Confidential data depends on having both data that is sent using data channels. Confidential data depends on
sources and consumers that are exclusively browser- or user-based. having both data sources and consumers that are exclusively browser-
No mechanisms currently exist to take advantage of data or user-based. No mechanisms currently exist to take advantage of
confidentiality, though some use cases suggest that this could be data confidentiality, though some use cases suggest that this could
useful, for example, confidential peer-to-peer file transfer. be useful, for example, confidential peer-to-peer file transfer.
Alternative labels might be provided in future to support these use Alternative labels might be provided in future to support these use
cases. cases.
Generally speaking, ensuring confidentiality depends on This mechanism explicitly does not define a specific authentication
authenticating the communications peer. This mechanism explicitly method; a WebRTC endpoint that accepts a session with this ALPN
does not define a specific authentication method; a WebRTC endpoint identifier MUST respect confidentiality no matter what identity is
that accepts a session with this ALPN identifier MUST respect attributed to a peer.
confidentiality no matter what identity is attributed to a peer.
RTP middleboxes and entities that forward media or data cannot RTP middleboxes and entities that forward media or data cannot
promise to maintain confidentiality. Any entity that forwards promise to maintain confidentiality. Any entity that forwards
content, or records content for later access by entities other than content, or records content for later access by entities other than
the authenticated peer, MUST NOT offer or accept a session with the the authenticated peer, MUST NOT offer or accept a session with the
"c-webrtc" identifier. "c-webrtc" identifier.
4. Security Considerations 4. Security Considerations
Confidential communications depends on more than just an agreement Confidential communications depends on more than just an agreement
from browsers. from browsers.
Information is not confidential if it is displayed to those other Information is not confidential if it is displayed to those other
than to whom it is intended. Peer authentication than to whom it is intended. Peer authentication
[I-D.ietf-rtcweb-security-arch] is necessary to ensure that data is [I-D.ietf-rtcweb-security-arch] is necessary to ensure that data is
only sent to the intended peer. only sent to the intended peer.
This is not a digital rights management mechanism. Even with an This is not a digital rights management mechanism. A user is not
authenticated peer, a user is not prevented from using other prevented from using other mechanisms to record or forward media.
mechanisms to record or forward media. This means that (for example) This means that (for example) screen recording devices, tape
screen recording devices, tape recorders, portable cameras, or a recorders, portable cameras, or a cunning arrangement of mirrors
cunning arrangement of mirrors could variously be used to record or could variously be used to record or redistribute media once
redistribute media once delivered. Similarly, if media is visible or delivered. Similarly, if media is visible or audible (or otherwise
audible (or otherwise accessible) to others in the vicinity, there accessible) to others in the vicinity, there are no technical
are no technical measures that protect the confidentiality of that measures that protect the confidentiality of that media.
media.
The only guarantee provided by this mechanism and the browser that The only guarantee provided by this mechanism and the browser that
implements it is that the media was delivered to the user that was implements it is that the media was delivered to the user that was
authenticated. Individual users will still need to make a judgment authenticated. Individual users will still need to make a judgment
about how their peer intends to respect the confidentiality of any about how their peer intends to respect the confidentiality of any
information provided. information provided.
On a shared computing platform like a browser, other entities with On a shared computing platform like a browser, other entities with
access to that platform (i.e., web applications), might be able to access to that platform (i.e., web applications), might be able to
access information that would compromise the confidentiality of access information that would compromise the confidentiality of
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using SRTP and data channels: using SRTP and data channels:
Protocol: WebRTC Media and Data Protocol: WebRTC Media and Data
Identification Sequence: 0x77 0x65 0x62 0x72 0x74 0x63 ("webrtc") Identification Sequence: 0x77 0x65 0x62 0x72 0x74 0x63 ("webrtc")
Specification: This document (RFCXXXX) Specification: This document (RFCXXXX)
c-webrtc: c-webrtc:
The "c-webrtc" label identifies confidential WebRTC The "c-webrtc" label identifies WebRTC communications with a
communications: promise to protect media confidentiality:
Protocol: Confidential WebRTC Media and Data Protocol: Confidential WebRTC Media and Data
Identification Sequence: 0x63 0x2d 0x77 0x65 0x62 0x72 0x74 0x63 Identification Sequence: 0x63 0x2d 0x77 0x65 0x62 0x72 0x74 0x63
("c-webrtc") ("c-webrtc")
Specification: This document (RFCXXXX) Specification: This document (RFCXXXX)
6. References 6. References
6.1. Normative References 6.1. Normative References
[I-D.ietf-rtcweb-data-channel] [I-D.ietf-rtcweb-data-channel]
Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data
Channels", draft-ietf-rtcweb-data-channel-13 (work in Channels", draft-ietf-rtcweb-data-channel-13 (work in
progress), January 2015. progress), January 2015.
[I-D.ietf-rtcweb-security-arch]
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 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997, DOI 10.17487/RFC2119, March 1997,
<http://www.rfc-editor.org/info/rfc2119>. <http://www.rfc-editor.org/info/rfc2119>.
[RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer [RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer
Security (DTLS) Extension to Establish Keys for the Secure Security (DTLS) Extension to Establish Keys for the Secure
Real-time Transport Protocol (SRTP)", RFC 5764, Real-time Transport Protocol (SRTP)", RFC 5764,
DOI 10.17487/RFC5764, May 2010, DOI 10.17487/RFC5764, May 2010,
<http://www.rfc-editor.org/info/rfc5764>. <http://www.rfc-editor.org/info/rfc5764>.
skipping to change at page 6, line 45 skipping to change at page 7, line 10
[HTML5] Berjon, R., Leithead, T., Doyle Navara, E., O'Connor, E., [HTML5] Berjon, R., Leithead, T., Doyle Navara, E., O'Connor, E.,
and S. Pfeiffer, "HTML 5", CR CR-html5-20121217, August and S. Pfeiffer, "HTML 5", CR CR-html5-20121217, August
2010, <http://www.w3.org/TR/2012/CR-html5-20121217/>. 2010, <http://www.w3.org/TR/2012/CR-html5-20121217/>.
[I-D.ietf-rtcweb-overview] [I-D.ietf-rtcweb-overview]
Alvestrand, H., "Overview: Real Time Protocols for Alvestrand, H., "Overview: Real Time Protocols for
Browser-based Applications", draft-ietf-rtcweb-overview-15 Browser-based Applications", draft-ietf-rtcweb-overview-15
(work in progress), January 2016. (work in progress), January 2016.
[I-D.ietf-rtcweb-security-arch]
Rescorla, E., "WebRTC Security Architecture", draft-ietf-
rtcweb-security-arch-11 (work in progress), March 2015.
[I-D.ietf-rtcweb-transports] [I-D.ietf-rtcweb-transports]
Alvestrand, H., "Transports for WebRTC", draft-ietf- Alvestrand, H., "Transports for WebRTC", draft-ietf-
rtcweb-transports-12 (work in progress), March 2016. rtcweb-transports-12 (work in progress), March 2016.
[RFC4960] Stewart, R., Ed., "Stream Control Transmission Protocol", [RFC4960] Stewart, R., Ed., "Stream Control Transmission Protocol",
RFC 4960, DOI 10.17487/RFC4960, September 2007, RFC 4960, DOI 10.17487/RFC4960, September 2007,
<http://www.rfc-editor.org/info/rfc4960>. <http://www.rfc-editor.org/info/rfc4960>.
[RFC5245] Rosenberg, J., "Interactive Connectivity Establishment [RFC5245] Rosenberg, J., "Interactive Connectivity Establishment
(ICE): A Protocol for Network Address Translator (NAT) (ICE): A Protocol for Network Address Translator (NAT)
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