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Run idnits with the --verbose option for more detailed information about the items above. -------------------------------------------------------------------------------- 2 RTCWEB M. Thomson 3 Internet-Draft Mozilla 4 Intended status: Standards Track February 28, 2015 5 Expires: September 1, 2015 7 Application Layer Protocol Negotiation for Web Real-Time Communications 8 (WebRTC) 9 draft-ietf-rtcweb-alpn-01 11 Abstract 13 Application Layer Protocol Negotiation (ALPN) labels are defined for 14 use in identifying Web Real-Time Communications (WebRTC) usages of 15 Datagram Transport Layer Security (DTLS). Labels are provided for 16 identifying a session that uses a combination of WebRTC compatible 17 media and data, and for identifying a session requiring 18 confidentiality protection. 20 Status of This Memo 22 This Internet-Draft is submitted in full conformance with the 23 provisions of BCP 78 and BCP 79. 25 Internet-Drafts are working documents of the Internet Engineering 26 Task Force (IETF). Note that other groups may also distribute 27 working documents as Internet-Drafts. The list of current Internet- 28 Drafts is at http://datatracker.ietf.org/drafts/current/. 30 Internet-Drafts are draft documents valid for a maximum of six months 31 and may be updated, replaced, or obsoleted by other documents at any 32 time. It is inappropriate to use Internet-Drafts as reference 33 material or to cite them other than as "work in progress." 35 This Internet-Draft will expire on September 1, 2015. 37 Copyright Notice 39 Copyright (c) 2015 IETF Trust and the persons identified as the 40 document authors. All rights reserved. 42 This document is subject to BCP 78 and the IETF Trust's Legal 43 Provisions Relating to IETF Documents 44 (http://trustee.ietf.org/license-info) in effect on the date of 45 publication of this document. Please review these documents 46 carefully, as they describe your rights and restrictions with respect 47 to this document. Code Components extracted from this document must 48 include Simplified BSD License text as described in Section 4.e of 49 the Trust Legal Provisions and are provided without warranty as 50 described in the Simplified BSD License. 52 Table of Contents 54 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2 55 1.1. Conventions and Terminology . . . . . . . . . . . . . . . 2 56 2. ALPN Labels for WebRTC . . . . . . . . . . . . . . . . . . . 2 57 3. Media Confidentiality . . . . . . . . . . . . . . . . . . . . 3 58 4. Security Considerations . . . . . . . . . . . . . . . . . . . 4 59 5. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 5 60 6. References . . . . . . . . . . . . . . . . . . . . . . . . . 6 61 6.1. Normative References . . . . . . . . . . . . . . . . . . 6 62 6.2. Informative References . . . . . . . . . . . . . . . . . 6 63 6.3. URIs . . . . . . . . . . . . . . . . . . . . . . . . . . 7 64 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 7 66 1. Introduction 68 Web Real-Time Communications (WebRTC) [I-D.ietf-rtcweb-overview] uses 69 Datagram Transport Layer Security (DTLS) [RFC6347] to secure all 70 peer-to-peer communications. 72 Identifying WebRTC protocol usage with Application Layer Protocol 73 Negotiation (ALPN) [RFC7301] enables an endpoint to positively 74 identify WebRTC uses and distinguish them from other DTLS uses. 76 Different WebRTC uses can be advertised and behavior can be 77 constrained to what is appropriate to a given use. In particular, 78 this allows for the identifications of sessions that require 79 confidentiality protection. 81 1.1. Conventions and Terminology 83 At times, this document falls back on shorthands for establishing 84 interoperability requirements on implementations: the capitalized 85 words "MUST", "SHOULD" and "MAY". These terms are defined in 86 [RFC2119]. 88 2. ALPN Labels for WebRTC 90 The following identifiers are defined for use in ALPN: 92 webrtc: The DTLS session is used to establish keys for a Secure 93 Real-time Transport Protocol (SRTP) - known as DTLS-SRTP - as 94 described in [RFC5764]. The DTLS record layer is used for WebRTC 95 data channels [I-D.ietf-rtcweb-data-channel]. 97 c-webrtc: The DTLS session is used for confidential WebRTC 98 communications, where peers agree to maintain the confidentiality 99 of the communications, as described in Section 3. 101 Both identifiers describe the same basic protocol: a DTLS session 102 that is used to provide keys for an SRTP session in combination with 103 WebRTC data channels. Either SRTP or data channels MAY be absent. 104 The data channels send Stream Control Transmission Protocol (SCTP) 105 [RFC4960] over the DTLS record layer, which can be multiplexed with 106 SRTP on the same UDP flow. WebRTC requires the use of Interactive 107 Communication Establishment (ICE) [RFC5245] to establish the UDP 108 flow, but this is not covered by the identifier. 110 A more thorough definition of what WebRTC communications entail is 111 included in [I-D.ietf-rtcweb-transports]. 113 There is no functional difference between the identifiers except that 114 an endpoint negotiating "c-webrtc" makes a promise to preserve the 115 confidentiality of the data it receives. 117 A peer that is not aware of whether it needs to request 118 confidentiality can use either form. A peer in the client role MUST 119 offer both identifiers if it is not aware of a need for 120 confidentiality. A peer in the server role SHOULD select "webrtc" if 121 it does not prefer either. 123 3. Media Confidentiality 125 Private communications in WebRTC depend on separating control (i.e., 126 signaling) capabilities and access to media 127 [I-D.ietf-rtcweb-security-arch]. In this way, an application can 128 establish a session that is end-to-end confidential, where the ends 129 in question are user agents (or browsers) and not the signaling 130 application. 132 A browser is required to enforce this control using isolation 133 controls similar to those used in cross-origin protections. These 134 protections ensure that media is protected from applications. 135 Applications are not able to read or modify the contents of a 136 protected flow of media. Media that is produced from a session using 137 the "c-webrtc" identifier MUST only be displayed to users. 139 Without some form of indication that is securely bound to the 140 session, a WebRTC endpoint is unable to properly distinguish between 141 session that requires confidentiality protection and one that does 142 not. 144 A browser is required to enforce confidentiality using isolation 145 controls similar to those used in content cross-origin protections 146 (see Section 5.3 [1] of [HTML5]). These protections ensure that 147 media is protected from applications. Applications are not able to 148 read or modify the contents of a protected flow of media. Media that 149 is produced from a session using the "c-webrtc" identifier MUST only 150 be displayed to users. 152 These confidentiality protections do not apply to data that is sent 153 using data channels. Confidential data depends on having both data 154 sources and consumers that are exclusively browser- or user-based. 155 No mechanisms currently exist to take advantage of data 156 confidentiality, though some use cases suggest that this could be 157 useful, for example, confidential peer-to-peer file transfer. 158 Alternative labels might be provided in future to support these use 159 cases. 161 Generally speaking, ensuring confidentiality depends on 162 authenticating the communications peer. This mechanism explicitly 163 does not define a specific authentication method; a WebRTC endpoint 164 that accepts a session with this ALPN identifier MUST respect 165 confidentiality no matter what identity is attributed to a peer. 167 RTP middleboxes and entities that forward media or data cannot 168 promise to maintain confidentiality. Any entity that forwards 169 content, or records content for later access by entities other than 170 the authenticated peer, SHOULD NOT offer or accept a session with the 171 "c-webrtc" identifier. 173 4. Security Considerations 175 Confidential communications depends on more than just an agreement 176 from browsers. 178 Information is not confidential if it is displayed to those other 179 than to whom it is intended. Peer authentication 180 [I-D.ietf-rtcweb-security-arch] is necessary to ensure that data is 181 only sent to the intended peer. 183 This is not a digital rights management mechanism. Even with an 184 authenticated peer, a user is not prevented from using other 185 mechanisms to record or forward media. This means that (for example) 186 screen recording devices, tape recorders, portable cameras, or a 187 cunning arrangement of mirrors could variously be used to record or 188 redistribute media once delivered. Similarly, if media is visible or 189 audible (or otherwise accessible) to others in the vicinity, there 190 are no technical measures that protect the confidentiality of that 191 media. In other cases, effects might not be temporally localized: 193 transmitted smells could linger for a period after communications 194 cease. 196 The only guarantee provided by this mechanism and the browser that 197 implements it is that the media was delivered to the user that was 198 authenticated. Individual users will still need to make a judgment 199 about how their peer intends to respect the confidentiality of any 200 information provided. 202 On a shared computing platform like a browser, other entities with 203 access to that platform (i.e., web applications), might be able to 204 access information that would compromise the confidentiality of 205 communications. Implementations MAY choose to limit concurrent 206 access to input devices during confidential communications session. 208 For instance, another application that is able to access a microphone 209 might be able to sample confidential audio that is playing through 210 speakers. This is true even if acoustic echo cancellation, which 211 attempts to prevent this from happening, is used. Similarly, an 212 application with access to a video camera might be able to use 213 reflections to obtain all or part of a confidential video stream. 215 5. IANA Considerations 217 The following two entries are added to the "Application Layer 218 Protocol Negotiation (ALPN) Protocol IDs" registry established by 219 [RFC7301]. 221 The "webrtc" identifies mixed media and data communications using 222 SRTP and data channels: 224 Protocol: WebRTC Media and Data 226 Identification Sequence: 0x77 0x65 0x62 0x72 0x74 0x63 ("webrtc") 228 Specification: This document (RFCXXXX) 230 The "c-webrtc" identifies confidential WebRTC communications: 232 Protocol: Confidential WebRTC Media and Data 234 Identification Sequence: 0x63 0x2d 0x77 0x65 0x62 0x72 0x74 0x63 235 ("c-webrtc") 237 Specification: This document (RFCXXXX) 239 6. References 241 6.1. Normative References 243 [I-D.ietf-rtcweb-data-channel] 244 Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data 245 Channels", draft-ietf-rtcweb-data-channel-11 (work in 246 progress), July 2014. 248 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 249 Requirement Levels", BCP 14, RFC 2119, March 1997. 251 [RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer 252 Security (DTLS) Extension to Establish Keys for the Secure 253 Real-time Transport Protocol (SRTP)", RFC 5764, May 2010. 255 [RFC6347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer 256 Security Version 1.2", RFC 6347, January 2012. 258 [RFC7301] Friedl, S., Popov, A., Langley, A., and E. Stephan, 259 "Transport Layer Security (TLS) Application-Layer Protocol 260 Negotiation Extension", RFC 7301, July 2014. 262 6.2. Informative References 264 [HTML5] Berjon, R., Leithead, T., Doyle Navara, E., O'Connor, E., 265 and S. Pfeiffer, "HTML 5", CR CR-html5-20121217, August 266 2010, . 268 [I-D.ietf-rtcweb-overview] 269 Alvestrand, H., "Overview: Real Time Protocols for 270 Browser-based Applications", draft-ietf-rtcweb-overview-11 271 (work in progress), August 2014. 273 [I-D.ietf-rtcweb-security-arch] 274 Rescorla, E., "WebRTC Security Architecture", draft-ietf- 275 rtcweb-security-arch-10 (work in progress), July 2014. 277 [I-D.ietf-rtcweb-transports] 278 Alvestrand, H., "Transports for WebRTC", draft-ietf- 279 rtcweb-transports-06 (work in progress), August 2014. 281 [RFC4960] Stewart, R., "Stream Control Transmission Protocol", RFC 282 4960, September 2007. 284 [RFC5245] Rosenberg, J., "Interactive Connectivity Establishment 285 (ICE): A Protocol for Network Address Translator (NAT) 286 Traversal for Offer/Answer Protocols", RFC 5245, April 287 2010. 289 6.3. URIs 291 [1] http://www.w3.org/TR/2012/CR-html5-20121217/browsers.html#origin 293 Author's Address 295 Martin Thomson 296 Mozilla 297 331 E Evelyn Street 298 Mountain View, CA 94041 299 US 301 Email: martin.thomson@gmail.com