--- 1/draft-ietf-rtcweb-alpn-00.txt 2015-02-28 16:14:34.801166861 -0800 +++ 2/draft-ietf-rtcweb-alpn-01.txt 2015-02-28 16:14:34.817167234 -0800 @@ -1,19 +1,19 @@ RTCWEB M. Thomson Internet-Draft Mozilla -Intended status: Standards Track July 23, 2014 -Expires: January 24, 2015 +Intended status: Standards Track February 28, 2015 +Expires: September 1, 2015 Application Layer Protocol Negotiation for Web Real-Time Communications (WebRTC) - draft-ietf-rtcweb-alpn-00 + draft-ietf-rtcweb-alpn-01 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. @@ -25,25 +25,25 @@ Internet-Drafts are working documents of the Internet Engineering Task Force (IETF). Note that other groups may also distribute working documents as Internet-Drafts. The list of current Internet- Drafts is at http://datatracker.ietf.org/drafts/current/. Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents 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 January 24, 2015. + This Internet-Draft will expire on September 1, 2015. 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. This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents (http://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 @@ -91,46 +91,41 @@ 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]. 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. + There is no functional difference between the identifiers except that + 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. + A peer that is not aware of whether it needs to request + confidentiality can use either form. 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. 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. @@ -147,40 +142,39 @@ 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. + These 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 future to support these use + cases. 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 + the authenticated peer, SHOULD 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 @@ -239,22 +235,22 @@ ("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. + Channels", draft-ietf-rtcweb-data-channel-11 (work in + progress), July 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. @@ -263,31 +259,31 @@ "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, . [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. + Alvestrand, H., "Overview: Real Time Protocols for + Browser-based Applications", draft-ietf-rtcweb-overview-11 + (work in progress), August 2014. [I-D.ietf-rtcweb-security-arch] 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] - Alvestrand, H., "Transports for RTCWEB", draft-ietf- - rtcweb-transports-04 (work in progress), April 2014. + Alvestrand, H., "Transports for WebRTC", draft-ietf- + rtcweb-transports-06 (work in progress), August 2014. [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