draft-ietf-rtcweb-alpn-00.txt   draft-ietf-rtcweb-alpn-01.txt 
RTCWEB M. Thomson RTCWEB M. Thomson
Internet-Draft Mozilla Internet-Draft Mozilla
Intended status: Standards Track July 23, 2014 Intended status: Standards Track February 28, 2015
Expires: January 24, 2015 Expires: September 1, 2015
Application Layer Protocol Negotiation for Web Real-Time Communications Application Layer Protocol Negotiation for Web Real-Time Communications
(WebRTC) (WebRTC)
draft-ietf-rtcweb-alpn-00 draft-ietf-rtcweb-alpn-01
Abstract Abstract
Application Layer Protocol Negotiation (ALPN) labels are defined for Application Layer Protocol Negotiation (ALPN) labels are defined for
use in identifying Web Real-Time Communications (WebRTC) usages of use in identifying Web Real-Time Communications (WebRTC) usages of
Datagram Transport Layer Security (DTLS). Labels are provided for Datagram Transport Layer Security (DTLS). Labels are provided for
identifying a session that uses a combination of WebRTC compatible identifying a session that uses a combination of WebRTC compatible
media and data, and for identifying a session requiring media and data, and for identifying a session requiring
confidentiality protection. confidentiality protection.
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Internet-Drafts are working documents of the Internet Engineering Internet-Drafts are working documents of the Internet Engineering
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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 January 24, 2015. This Internet-Draft will expire on September 1, 2015.
Copyright Notice Copyright Notice
Copyright (c) 2014 IETF Trust and the persons identified as the Copyright (c) 2015 IETF Trust and the persons identified as the
document authors. All rights reserved. document authors. All rights reserved.
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the Trust Legal Provisions and are provided without warranty as the Trust Legal Provisions and are provided without warranty as
skipping to change at page 3, line 9 skipping to change at page 3, line 9
webrtc: The DTLS session is used to establish keys for a Secure webrtc: The DTLS session is used to establish keys for a Secure
Real-time Transport Protocol (SRTP) - known as DTLS-SRTP - as Real-time Transport Protocol (SRTP) - known as DTLS-SRTP - as
described in [RFC5764]. The DTLS record layer is used for WebRTC described in [RFC5764]. The DTLS record layer is used for WebRTC
data channels [I-D.ietf-rtcweb-data-channel]. data 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 communications, as described in Section 3. 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 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 MAY be absent. WebRTC data channels. Either SRTP or data channels MAY 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 with There is no functional difference between the identifiers except that
respect to the promise that an endpoint makes with respect to the an endpoint negotiating "c-webrtc" makes a promise to preserve the
confidentiality of session content. An endpoint negotiating confidentiality of the data it receives.
"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 A peer that is not aware of whether it needs to request
acting in the client role MUST NOT offer both identifiers. A peer in confidentiality can use either form. A peer in the client role MUST
the server role that receives a ClientHello containing both labels offer both identifiers if it is not aware of a need for
MUST reject the session, though it MAY accept the confidential option confidentiality. A peer in the server role SHOULD select "webrtc" if
and protect content accordingly. it does not prefer either.
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. application.
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not. not.
A browser is required to enforce confidentiality using isolation A browser is required to enforce confidentiality using isolation
controls similar to those used in content cross-origin protections controls similar to those used in content cross-origin protections
(see Section 5.3 [1] of [HTML5]). These protections ensure that (see Section 5.3 [1] of [HTML5]). These protections ensure that
media is protected from applications. Applications are not able to media is protected from applications. Applications are not able to
read or modify the contents of a protected flow of media. Media that 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 is produced from a session using the "c-webrtc" identifier MUST only
be displayed to users. be displayed to users.
Confidentiality protections of this sort are not expected to be These confidentiality protections do not apply to data that is sent
possible for data that is sent using data channels. Thus, it is using data channels. Confidential data depends on having both data
expected that data channels will not be employed for sessions that sources and consumers that are exclusively browser- or user-based.
negotiate confidentiality. In the browser context, confidential data No mechanisms currently exist to take advantage of data
depends on having both data sources and consumers that are confidentiality, though some use cases suggest that this could be
exclusively browser- or user-based. No mechanisms currently exist to useful, for example, confidential peer-to-peer file transfer.
take advantage of data confidentiality, though some use cases suggest Alternative labels might be provided in future to support these use
that this could be useful, for example, confidential peer-to-peer cases.
file transfer.
Generally speaking, ensuring confidentiality depends on Generally speaking, ensuring confidentiality depends on
authenticating the communications peer. This mechanism explicitly authenticating the communications peer. This mechanism explicitly
does not define a specific authentication method; a WebRTC endpoint does not define a specific authentication method; a WebRTC endpoint
that accepts a session with this ALPN identifier MUST respect that accepts a session with this ALPN identifier MUST respect
confidentiality no matter what identity is 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, SHOULD 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
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("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-09 (work in Channels", draft-ietf-rtcweb-data-channel-11 (work in
progress), May 2014. progress), July 2014.
[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, March 1997. Requirement Levels", BCP 14, RFC 2119, March 1997.
[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, May 2010. Real-time Transport Protocol (SRTP)", RFC 5764, May 2010.
[RFC6347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer [RFC6347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer
Security Version 1.2", RFC 6347, January 2012. Security Version 1.2", RFC 6347, January 2012.
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"Transport Layer Security (TLS) Application-Layer Protocol "Transport Layer Security (TLS) Application-Layer Protocol
Negotiation Extension", RFC 7301, July 2014. Negotiation Extension", RFC 7301, July 2014.
6.2. Informative References 6.2. Informative References
[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 Brower- Alvestrand, H., "Overview: Real Time Protocols for
based Applications", draft-ietf-rtcweb-overview-09 (work Browser-based Applications", draft-ietf-rtcweb-overview-11
in progress), February 2014. (work in progress), August 2014.
[I-D.ietf-rtcweb-security-arch] [I-D.ietf-rtcweb-security-arch]
Rescorla, E., "WebRTC Security Architecture", draft-ietf- Rescorla, E., "WebRTC Security Architecture", draft-ietf-
rtcweb-security-arch-09 (work in progress), February 2014. rtcweb-security-arch-10 (work in progress), July 2014.
[I-D.ietf-rtcweb-transports] [I-D.ietf-rtcweb-transports]
Alvestrand, H., "Transports for RTCWEB", draft-ietf- Alvestrand, H., "Transports for WebRTC", draft-ietf-
rtcweb-transports-04 (work in progress), April 2014. rtcweb-transports-06 (work in progress), August 2014.
[RFC4960] Stewart, R., "Stream Control Transmission Protocol", RFC [RFC4960] Stewart, R., "Stream Control Transmission Protocol", RFC
4960, September 2007. 4960, September 2007.
[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)
Traversal for Offer/Answer Protocols", RFC 5245, April Traversal for Offer/Answer Protocols", RFC 5245, April
2010. 2010.
6.3. URIs 6.3. URIs
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