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Please use uppercase 'NOT' together with RFC 2119 keywords (if that is what you mean). Found 'MUST not' in this paragraph: o Browsers MUST not permit permanent screen or application sharing permissions to be installed as a response to a JS request for permissions. Instead, they must require some other user action such as a permissions setting or an application install experience to grant permission to a site. == Using lowercase 'not' together with uppercase 'MUST', 'SHALL', 'SHOULD', or 'RECOMMENDED' is not an accepted usage according to RFC 2119. Please use uppercase 'NOT' together with RFC 2119 keywords (if that is what you mean). Found 'MUST not' in this paragraph: o The browser MUST indicate any windows which are currently being shared in some unambiguous way. Windows which are not visible MUST not be shared even if the application is being shared. If the screen is being shared, then that MUST be indicated. == The document seems to contain a disclaimer for pre-RFC5378 work, but was first submitted on or after 10 November 2008. The disclaimer is usually necessary only for documents that revise or obsolete older RFCs, and that take significant amounts of text from those RFCs. If you can contact all authors of the source material and they are willing to grant the BCP78 rights to the IETF Trust, you can and should remove the disclaimer. Otherwise, the disclaimer is needed and you can ignore this comment. (See the Legal Provisions document at https://trustee.ietf.org/license-info for more information.) -- The document date (October 30, 2017) is 2368 days in the past. Is this intentional? 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'FIPS186' == Outdated reference: A later version (-12) exists of draft-ietf-rtcweb-security-09 ** Obsolete normative reference: RFC 2818 (Obsoleted by RFC 9110) ** Obsolete normative reference: RFC 4347 (Obsoleted by RFC 6347) ** Obsolete normative reference: RFC 4566 (Obsoleted by RFC 8866) ** Obsolete normative reference: RFC 4627 (Obsoleted by RFC 7158, RFC 7159) ** Obsolete normative reference: RFC 5245 (Obsoleted by RFC 8445, RFC 8839) ** Obsolete normative reference: RFC 5246 (Obsoleted by RFC 8446) ** Obsolete normative reference: RFC 5785 (Obsoleted by RFC 8615) == Outdated reference: A later version (-26) exists of draft-ietf-rtcweb-jsep-24 -- Obsolete informational reference (is this intentional?): RFC 2617 (Obsoleted by RFC 7235, RFC 7615, RFC 7616, RFC 7617) Summary: 7 errors (**), 0 flaws (~~), 6 warnings (==), 3 comments (--). Run idnits with the --verbose option for more detailed information about the items above. -------------------------------------------------------------------------------- 2 RTCWEB E. Rescorla 3 Internet-Draft RTFM, Inc. 4 Intended status: Standards Track October 30, 2017 5 Expires: May 3, 2018 7 WebRTC Security Architecture 8 draft-ietf-rtcweb-security-arch-13 10 Abstract 12 This document defines the security architecture for WebRTC, a 13 protocol suite intended for use with real-time applications that can 14 be deployed in browsers - "real time communication on the Web". 16 Status of This Memo 18 This Internet-Draft is submitted in full conformance with the 19 provisions of BCP 78 and BCP 79. 21 Internet-Drafts are working documents of the Internet Engineering 22 Task Force (IETF). Note that other groups may also distribute 23 working documents as Internet-Drafts. The list of current Internet- 24 Drafts is at http://datatracker.ietf.org/drafts/current/. 26 Internet-Drafts are draft documents valid for a maximum of six months 27 and may be updated, replaced, or obsoleted by other documents at any 28 time. It is inappropriate to use Internet-Drafts as reference 29 material or to cite them other than as "work in progress." 31 This Internet-Draft will expire on May 3, 2018. 33 Copyright Notice 35 Copyright (c) 2017 IETF Trust and the persons identified as the 36 document authors. All rights reserved. 38 This document is subject to BCP 78 and the IETF Trust's Legal 39 Provisions Relating to IETF Documents 40 (http://trustee.ietf.org/license-info) in effect on the date of 41 publication of this document. Please review these documents 42 carefully, as they describe your rights and restrictions with respect 43 to this document. Code Components extracted from this document must 44 include Simplified BSD License text as described in Section 4.e of 45 the Trust Legal Provisions and are provided without warranty as 46 described in the Simplified BSD License. 48 This document may contain material from IETF Documents or IETF 49 Contributions published or made publicly available before November 50 10, 2008. The person(s) controlling the copyright in some of this 51 material may not have granted the IETF Trust the right to allow 52 modifications of such material outside the IETF Standards Process. 53 Without obtaining an adequate license from the person(s) controlling 54 the copyright in such materials, this document may not be modified 55 outside the IETF Standards Process, and derivative works of it may 56 not be created outside the IETF Standards Process, except to format 57 it for publication as an RFC or to translate it into languages other 58 than English. 60 Table of Contents 62 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3 63 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 5 64 3. Trust Model . . . . . . . . . . . . . . . . . . . . . . . . . 5 65 3.1. Authenticated Entities . . . . . . . . . . . . . . . . . 5 66 3.2. Unauthenticated Entities . . . . . . . . . . . . . . . . 6 67 4. Overview . . . . . . . . . . . . . . . . . . . . . . . . . . 6 68 4.1. Initial Signaling . . . . . . . . . . . . . . . . . . . . 8 69 4.2. Media Consent Verification . . . . . . . . . . . . . . . 10 70 4.3. DTLS Handshake . . . . . . . . . . . . . . . . . . . . . 11 71 4.4. Communications and Consent Freshness . . . . . . . . . . 11 72 5. Detailed Technical Description . . . . . . . . . . . . . . . 12 73 5.1. Origin and Web Security Issues . . . . . . . . . . . . . 12 74 5.2. Device Permissions Model . . . . . . . . . . . . . . . . 12 75 5.3. Communications Consent . . . . . . . . . . . . . . . . . 14 76 5.4. IP Location Privacy . . . . . . . . . . . . . . . . . . . 14 77 5.5. Communications Security . . . . . . . . . . . . . . . . . 15 78 5.6. Web-Based Peer Authentication . . . . . . . . . . . . . . 17 79 5.6.1. Trust Relationships: IdPs, APs, and RPs . . . . . . . 18 80 5.6.2. Overview of Operation . . . . . . . . . . . . . . . . 20 81 5.6.3. Items for Standardization . . . . . . . . . . . . . . 21 82 5.6.4. Binding Identity Assertions to JSEP Offer/Answer 83 Transactions . . . . . . . . . . . . . . . . . . . . 21 84 5.6.4.1. Carrying Identity Assertions . . . . . . . . . . 22 85 5.6.4.2. a=identity Attribute . . . . . . . . . . . . . . 23 86 5.6.5. Determining the IdP URI . . . . . . . . . . . . . . . 23 87 5.6.5.1. Authenticating Party . . . . . . . . . . . . . . 24 88 5.6.5.2. Relying Party . . . . . . . . . . . . . . . . . . 25 89 5.6.6. Requesting Assertions . . . . . . . . . . . . . . . . 25 90 5.6.7. Managing User Login . . . . . . . . . . . . . . . . . 26 91 5.7. Verifying Assertions . . . . . . . . . . . . . . . . . . 26 92 5.7.1. Identity Formats . . . . . . . . . . . . . . . . . . 27 93 6. Security Considerations . . . . . . . . . . . . . . . . . . . 28 94 6.1. Communications Security . . . . . . . . . . . . . . . . . 28 95 6.2. Privacy . . . . . . . . . . . . . . . . . . . . . . . . . 29 96 6.3. Denial of Service . . . . . . . . . . . . . . . . . . . . 29 97 6.4. IdP Authentication Mechanism . . . . . . . . . . . . . . 31 98 6.4.1. PeerConnection Origin Check . . . . . . . . . . . . . 31 99 6.4.2. IdP Well-known URI . . . . . . . . . . . . . . . . . 31 100 6.4.3. Privacy of IdP-generated identities and the hosting 101 site . . . . . . . . . . . . . . . . . . . . . . . . 32 102 6.4.4. Security of Third-Party IdPs . . . . . . . . . . . . 32 103 6.4.5. Web Security Feature Interactions . . . . . . . . . . 32 104 6.4.5.1. Popup Blocking . . . . . . . . . . . . . . . . . 32 105 6.4.5.2. Third Party Cookies . . . . . . . . . . . . . . . 33 106 7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 33 107 8. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 33 108 9. Changes . . . . . . . . . . . . . . . . . . . . . . . . . . . 33 109 9.1. Changes since -10 . . . . . . . . . . . . . . . . . . . . 33 110 9.2. Changes since -06 . . . . . . . . . . . . . . . . . . . . 34 111 9.3. Changes since -05 . . . . . . . . . . . . . . . . . . . . 34 112 9.4. Changes since -03 . . . . . . . . . . . . . . . . . . . . 34 113 9.5. Changes since -03 . . . . . . . . . . . . . . . . . . . . 34 114 9.6. Changes since -02 . . . . . . . . . . . . . . . . . . . . 34 115 10. References . . . . . . . . . . . . . . . . . . . . . . . . . 35 116 10.1. Normative References . . . . . . . . . . . . . . . . . . 35 117 10.2. Informative References . . . . . . . . . . . . . . . . . 38 118 Appendix A. Example IdP Bindings to Specific Protocols . . . . . 38 119 A.1. OAuth . . . . . . . . . . . . . . . . . . . . . . . . . . 39 120 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 40 122 1. Introduction 124 The Real-Time Communications on the Web (WebRTC) working group is 125 tasked with standardizing protocols for real-time communications 126 between Web browsers. The major use cases for WebRTC technology are 127 real-time audio and/or video calls, Web conferencing, and direct data 128 transfer. Unlike most conventional real-time systems, (e.g., SIP- 129 based[RFC3261] soft phones) WebRTC communications are directly 130 controlled by some Web server, via a JavaScript (JS) API as shown in 131 Figure 1. 133 +----------------+ 134 | | 135 | Web Server | 136 | | 137 +----------------+ 138 ^ ^ 139 / \ 140 HTTP / \ HTTP 141 / \ 142 / \ 143 v v 144 JS API JS API 145 +-----------+ +-----------+ 146 | | Media | | 147 | Browser |<---------->| Browser | 148 | | | | 149 +-----------+ +-----------+ 151 Figure 1: A simple WebRTC system 153 A more complicated system might allow for interdomain calling, as 154 shown in Figure 2. The protocol to be used between the domains is 155 not standardized by WebRTC, but given the installed base and the form 156 of the WebRTC API is likely to be something SDP-based like SIP. 158 +--------------+ +--------------+ 159 | | SIP,XMPP,...| | 160 | Web Server |<----------->| Web Server | 161 | | | | 162 +--------------+ +--------------+ 163 ^ ^ 164 | | 165 HTTP | | HTTP 166 | | 167 v v 168 JS API JS API 169 +-----------+ +-----------+ 170 | | Media | | 171 | Browser |<---------------->| Browser | 172 | | | | 173 +-----------+ +-----------+ 175 Figure 2: A multidomain WebRTC system 177 This system presents a number of new security challenges, which are 178 analyzed in [I-D.ietf-rtcweb-security]. This document describes a 179 security architecture for WebRTC which addresses the threats and 180 requirements described in that document. 182 2. Terminology 184 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 185 "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this 186 document are to be interpreted as described in RFC 2119 [RFC2119]. 188 3. Trust Model 190 The basic assumption of this architecture is that network resources 191 exist in a hierarchy of trust, rooted in the browser, which serves as 192 the user's TRUSTED COMPUTING BASE (TCB). Any security property which 193 the user wishes to have enforced must be ultimately guaranteed by the 194 browser (or transitively by some property the browser verifies). 195 Conversely, if the browser is compromised, then no security 196 guarantees are possible. Note that there are cases (e.g., Internet 197 kiosks) where the user can't really trust the browser that much. In 198 these cases, the level of security provided is limited by how much 199 they trust the browser. 201 Optimally, we would not rely on trust in any entities other than the 202 browser. However, this is unfortunately not possible if we wish to 203 have a functional system. Other network elements fall into two 204 categories: those which can be authenticated by the browser and thus 205 can be granted permissions to access sensitive resources, and those 206 which cannot be authenticated and thus are untrusted. 208 3.1. Authenticated Entities 210 There are two major classes of authenticated entities in the system: 212 o Calling services: Web sites whose origin we can verify (optimally 213 via HTTPS, but in some cases because we are on a topologically 214 restricted network, such as behind a firewall, and can infer 215 authentication from firewall behavior). 217 o Other users: WebRTC peers whose origin we can verify 218 cryptographically (optimally via DTLS-SRTP). 220 Note that merely being authenticated does not make these entities 221 trusted. For instance, just because we can verify that 222 https://www.evil.org/ is owned by Dr. Evil does not mean that we can 223 trust Dr. Evil to access our camera and microphone. However, it 224 gives the user an opportunity to determine whether he wishes to trust 225 Dr. Evil or not; after all, if he desires to contact Dr. Evil 226 (perhaps to arrange for ransom payment), it's safe to temporarily 227 give him access to the camera and microphone for the purpose of the 228 call, but he doesn't want Dr. Evil to be able to access his camera 229 and microphone other than during the call. The point here is that we 230 must first identify other elements before we can determine whether 231 and how much to trust them. Additionally, sometimes we need to 232 identify the communicating peer before we know what policies to 233 apply. 235 3.2. Unauthenticated Entities 237 Other than the above entities, we are not generally able to identify 238 other network elements, thus we cannot trust them. This does not 239 mean that it is not possible to have any interaction with them, but 240 it means that we must assume that they will behave maliciously and 241 design a system which is secure even if they do so. 243 4. Overview 245 This section describes a typical WebRTC session and shows how the 246 various security elements interact and what guarantees are provided 247 to the user. The example in this section is a "best case" scenario 248 in which we provide the maximal amount of user authentication and 249 media privacy with the minimal level of trust in the calling service. 250 Simpler versions with lower levels of security are also possible and 251 are noted in the text where applicable. It's also important to 252 recognize the tension between security (or performance) and privacy. 253 The example shown here is aimed towards settings where we are more 254 concerned about secure calling than about privacy, but as we shall 255 see, there are settings where one might wish to make different 256 tradeoffs--this architecture is still compatible with those settings. 258 For the purposes of this example, we assume the topology shown in the 259 figures below. This topology is derived from the topology shown in 260 Figure 1, but separates Alice and Bob's identities from the process 261 of signaling. Specifically, Alice and Bob have relationships with 262 some Identity Provider (IdP) that supports a protocol (such as OpenID 263 Connect) that can be used to demonstrate their identity to other 264 parties. For instance, Alice might have an account with a social 265 network which she can then use to authenticate to other web sites 266 without explicitly having an account with those sites; this is a 267 fairly conventional pattern on the Web. Section 5.6.1 provides an 268 overview of Identity Providers and the relevant terminology. Alice 269 and Bob might have relationships with different IdPs as well. 271 This separation of identity provision and signaling isn't 272 particularly important in "closed world" cases where Alice and Bob 273 are users on the same social network and have identities based on 274 that domain (Figure 3) However, there are important settings where 275 that is not the case, such as federation (calls from one domain to 276 another; Figure 4) and calling on untrusted sites, such as where two 277 users who have a relationship via a given social network want to call 278 each other on another, untrusted, site, such as a poker site. 280 Note that the servers themselves are also authenticated by an 281 external identity service, the SSL/TLS certificate infrastructure 282 (not shown). As is conventional in the Web, all identities are 283 ultimately rooted in that system. For instance, when an IdP makes an 284 identity assertion, the Relying Party consuming that assertion is 285 able to verify because it is able to connect to the IdP via HTTPS. 287 +----------------+ 288 | | 289 | Signaling | 290 | Server | 291 | | 292 +----------------+ 293 ^ ^ 294 / \ 295 HTTPS / \ HTTPS 296 / \ 297 / \ 298 v v 299 JS API JS API 300 +-----------+ +-----------+ 301 | | Media | | 302 Alice | Browser |<---------->| Browser | Bob 303 | | (DTLS+SRTP)| | 304 +-----------+ +-----------+ 305 ^ ^--+ +--^ ^ 306 | | | | 307 v | | v 308 +-----------+ | | +-----------+ 309 | |<--------+ | | 310 | IdP1 | | | IdP2 | 311 | | +------->| | 312 +-----------+ +-----------+ 314 Figure 3: A call with IdP-based identity 316 Figure 4 shows essentially the same calling scenario but with a call 317 between two separate domains (i.e., a federated case), as in 318 Figure 2. As mentioned above, the domains communicate by some 319 unspecified protocol and providing separate signaling and identity 320 allows for calls to be authenticated regardless of the details of the 321 inter-domain protocol. 323 +----------------+ Unspecified +----------------+ 324 | | protocol | | 325 | Signaling |<----------------->| Signaling | 326 | Server | (SIP, XMPP, ...) | Server | 327 | | | | 328 +----------------+ +----------------+ 329 ^ ^ 330 | | 331 HTTPS | | HTTPS 332 | | 333 | | 334 v v 335 JS API JS API 336 +-----------+ +-----------+ 337 | | Media | | 338 Alice | Browser |<--------------------------->| Browser | Bob 339 | | DTLS+SRTP | | 340 +-----------+ +-----------+ 341 ^ ^--+ +--^ ^ 342 | | | | 343 v | | v 344 +-----------+ | | +-----------+ 345 | |<-------------------------+ | | 346 | IdP1 | | | IdP2 | 347 | | +------------------------>| | 348 +-----------+ +-----------+ 350 Figure 4: A federated call with IdP-based identity 352 4.1. Initial Signaling 354 For simplicity, assume the topology in Figure 3. Alice and Bob are 355 both users of a common calling service; they both have approved the 356 calling service to make calls (we defer the discussion of device 357 access permissions till later). They are both connected to the 358 calling service via HTTPS and so know the origin with some level of 359 confidence. They also have accounts with some identity provider. 360 This sort of identity service is becoming increasingly common in the 361 Web environment (with technologies such as Federated Google Login, 362 Facebook Connect, OAuth, OpenID, WebFinger), and is often provided as 363 a side effect service of a user's ordinary accounts with some 364 service. In this example, we show Alice and Bob using a separate 365 identity service, though the identity service may be the same entity 366 as the calling service or there may be no identity service at all. 368 Alice is logged onto the calling service and decides to call Bob. 369 She can see from the calling service that he is online and the 370 calling service presents a JS UI in the form of a button next to 371 Bob's name which says "Call". Alice clicks the button, which 372 initiates a JS callback that instantiates a PeerConnection object. 373 This does not require a security check: JS from any origin is allowed 374 to get this far. 376 Once the PeerConnection is created, the calling service JS needs to 377 set up some media. Because this is an audio/video call, it creates a 378 MediaStream with two MediaStreamTracks, one connected to an audio 379 input and one connected to a video input. At this point the first 380 security check is required: untrusted origins are not allowed to 381 access the camera and microphone, so the browser prompts Alice for 382 permission. 384 In the current W3C API, once some streams have been added, Alice's 385 browser + JS generates a signaling message [I-D.ietf-rtcweb-jsep] 386 containing: 388 o Media channel information 390 o Interactive Connectivity Establishment (ICE) [RFC5245] candidates 392 o A fingerprint attribute binding the communication to a key pair 393 [RFC5763]. Note that this key may simply be ephemerally generated 394 for this call or specific to this domain, and Alice may have a 395 large number of such keys. 397 Prior to sending out the signaling message, the PeerConnection code 398 contacts the identity service and obtains an assertion binding 399 Alice's identity to her fingerprint. The exact details depend on the 400 identity service (though as discussed in Section 5.6 PeerConnection 401 can be agnostic to them), but for now it's easiest to think of as an 402 OAuth token. The assertion may bind other information to the 403 identity besides the fingerprint, but at minimum it needs to bind the 404 fingerprint. 406 This message is sent to the signaling server, e.g., by XMLHttpRequest 407 [XmlHttpRequest] or by WebSockets [RFC6455]. preferably over TLS 408 [RFC5246]. The signaling server processes the message from Alice's 409 browser, determines that this is a call to Bob and sends a signaling 410 message to Bob's browser (again, the format is currently undefined). 411 The JS on Bob's browser processes it, and alerts Bob to the incoming 412 call and to Alice's identity. In this case, Alice has provided an 413 identity assertion and so Bob's browser contacts Alice's identity 414 provider (again, this is done in a generic way so the browser has no 415 specific knowledge of the IdP) to verify the assertion. This allows 416 the browser to display a trusted element in the browser chrome 417 indicating that a call is coming in from Alice. If Alice is in Bob's 418 address book, then this interface might also include her real name, a 419 picture, etc. The calling site will also provide some user interface 420 element (e.g., a button) to allow Bob to answer the call, though this 421 is most likely not part of the trusted UI. 423 If Bob agrees a PeerConnection is instantiated with the message from 424 Alice's side. Then, a similar process occurs as on Alice's browser: 425 Bob's browser prompts him for device permission, the media streams 426 are created, and a return signaling message containing media 427 information, ICE candidates, and a fingerprint is sent back to Alice 428 via the signaling service. If Bob has a relationship with an IdP, 429 the message will also come with an identity assertion. 431 At this point, Alice and Bob each know that the other party wants to 432 have a secure call with them. Based purely on the interface provided 433 by the signaling server, they know that the signaling server claims 434 that the call is from Alice to Bob. This level of security is 435 provided merely by having the fingerprint in the message and having 436 that message received securely from the signaling server. Because 437 the far end sent an identity assertion along with their message, they 438 know that this is verifiable from the IdP as well. Note that if the 439 call is federated, as shown in Figure 4 then Alice is able to verify 440 Bob's identity in a way that is not mediated by either her signaling 441 server or Bob's. Rather, she verifies it directly with Bob's IdP. 443 Of course, the call works perfectly well if either Alice or Bob 444 doesn't have a relationship with an IdP; they just get a lower level 445 of assurance. I.e., they simply have whatever information their 446 calling site claims about the caller/calllee's identity. Moreover, 447 Alice might wish to make an anonymous call through an anonymous 448 calling site, in which case she would of course just not provide any 449 identity assertion and the calling site would mask her identity from 450 Bob. 452 4.2. Media Consent Verification 454 As described in ([I-D.ietf-rtcweb-security]; Section 4.2) media 455 consent verification is provided via ICE. Thus, Alice and Bob 456 perform ICE checks with each other. At the completion of these 457 checks, they are ready to send non-ICE data. 459 At this point, Alice knows that (a) Bob (assuming he is verified via 460 his IdP) or someone else who the signaling service is claiming is Bob 461 is willing to exchange traffic with her and (b) that either Bob is at 462 the IP address which she has verified via ICE or there is an attacker 463 who is on-path to that IP address detouring the traffic. Note that 464 it is not possible for an attacker who is on-path between Alice and 465 Bob but not attached to the signaling service to spoof these checks 466 because they do not have the ICE credentials. Bob has the same 467 security guarantees with respect to Alice. 469 4.3. DTLS Handshake 471 Once the ICE checks have completed [more specifically, once some ICE 472 checks have completed], Alice and Bob can set up a secure channel or 473 channels. This is performed via DTLS [RFC4347] and DTLS-SRTP 474 [RFC5763] keying for SRTP [RFC3711] for the media channel and SCTP 475 over DTLS [I-D.ietf-tsvwg-sctp-dtls-encaps] for data channels. 476 Specifically, Alice and Bob perform a DTLS handshake on every channel 477 which has been established by ICE. The total number of channels 478 depends on the amount of muxing; in the most likely case we are using 479 both RTP/RTCP mux and muxing multiple media streams on the same 480 channel, in which case there is only one DTLS handshake. Once the 481 DTLS handshake has completed, the keys are exported [RFC5705] and 482 used to key SRTP for the media channels. 484 At this point, Alice and Bob know that they share a set of secure 485 data and/or media channels with keys which are not known to any 486 third-party attacker. If Alice and Bob authenticated via their IdPs, 487 then they also know that the signaling service is not mounting a man- 488 in-the-middle attack on their traffic. Even if they do not use an 489 IdP, as long as they have minimal trust in the signaling service not 490 to perform a man-in-the-middle attack, they know that their 491 communications are secure against the signaling service as well 492 (i.e., that the signaling service cannot mount a passive attack on 493 the communications). 495 4.4. Communications and Consent Freshness 497 From a security perspective, everything from here on in is a little 498 anticlimactic: Alice and Bob exchange data protected by the keys 499 negotiated by DTLS. Because of the security guarantees discussed in 500 the previous sections, they know that the communications are 501 encrypted and authenticated. 503 The one remaining security property we need to establish is "consent 504 freshness", i.e., allowing Alice to verify that Bob is still prepared 505 to receive her communications so that Alice does not continue to send 506 large traffic volumes to entities which went abruptly offline. ICE 507 specifies periodic STUN keepalives but only if media is not flowing. 508 Because the consent issue is more difficult here, we require WebRTC 509 implementations to periodically send keepalives. As described in 510 Section 5.3, these keepalives MUST be based on the consent freshness 511 mechanism specified in [I-D.muthu-behave-consent-freshness]. If a 512 keepalive fails and no new ICE channels can be established, then the 513 session is terminated. 515 5. Detailed Technical Description 517 5.1. Origin and Web Security Issues 519 The basic unit of permissions for WebRTC is the origin [RFC6454]. 520 Because the security of the origin depends on being able to 521 authenticate content from that origin, the origin can only be 522 securely established if data is transferred over HTTPS [RFC2818]. 523 Thus, clients MUST treat HTTP and HTTPS origins as different 524 permissions domains. [Note: this follows directly from the origin 525 security model and is stated here merely for clarity.] 527 Many web browsers currently forbid by default any active mixed 528 content on HTTPS pages. That is, when JavaScript is loaded from an 529 HTTP origin onto an HTTPS page, an error is displayed and the HTTP 530 content is not executed unless the user overrides the error. Any 531 browser which enforces such a policy will also not permit access to 532 WebRTC functionality from mixed content pages (because they never 533 display mixed content). Browsers which allow active mixed content 534 MUST nevertheless disable WebRTC functionality in mixed content 535 settings. 537 Note that it is possible for a page which was not mixed content to 538 become mixed content during the duration of the call. The major risk 539 here is that the newly arrived insecure JS might redirect media to a 540 location controlled by the attacker. Implementations MUST either 541 choose to terminate the call or display a warning at that point. 543 5.2. Device Permissions Model 545 Implementations MUST obtain explicit user consent prior to providing 546 access to the camera and/or microphone. Implementations MUST at 547 minimum support the following two permissions models for HTTPS 548 origins. 550 o Requests for one-time camera/microphone access. 552 o Requests for permanent access. 554 Because HTTP origins cannot be securely established against network 555 attackers, implementations MUST NOT allow the setting of permanent 556 access permissions for HTTP origins. Implementations MUST refuse all 557 permissions grants for HTTP origins. 559 In addition, they SHOULD support requests for access that promise 560 that media from this grant will be sent to a single communicating 561 peer (obviously there could be other requests for other peers). 562 E.g., "Call customerservice@ford.com". The semantics of this request 563 are that the media stream from the camera and microphone will only be 564 routed through a connection which has been cryptographically verified 565 (through the IdP mechanism or an X.509 certificate in the DTLS-SRTP 566 handshake) as being associated with the stated identity. Note that 567 it is unlikely that browsers would have an X.509 certificate, but 568 servers might. Browsers servicing such requests SHOULD clearly 569 indicate that identity to the user when asking for permission. The 570 idea behind this type of permissions is that a user might have a 571 fairly narrow list of peers he is willing to communicate with, e.g., 572 "my mother" rather than "anyone on Facebook". Narrow permissions 573 grants allow the browser to do that enforcement. 575 API Requirement: The API MUST provide a mechanism for the requesting 576 JS to relinquish the ability to see or modify the media (e.g., via 577 MediaStream.record()). Combined with secure authentication of the 578 communicating peer, this allows a user to be sure that the calling 579 site is not accessing or modifying their conversion. 581 UI Requirement: The UI MUST clearly indicate when the user's camera 582 and microphone are in use. This indication MUST NOT be 583 suppressable by the JS and MUST clearly indicate how to terminate 584 device access, and provide a UI means to immediately stop camera/ 585 microphone input without the JS being able to prevent it. 587 UI Requirement: If the UI indication of camera/microphone use are 588 displayed in the browser such that minimizing the browser window 589 would hide the indication, or the JS creating an overlapping 590 window would hide the indication, then the browser SHOULD stop 591 camera and microphone input when the indication is hidden. [Note: 592 this may not be necessary in systems that are non-windows-based 593 but that have good notifications support, such as phones.] 595 o Browsers MUST not permit permanent screen or application sharing 596 permissions to be installed as a response to a JS request for 597 permissions. Instead, they must require some other user action 598 such as a permissions setting or an application install experience 599 to grant permission to a site. 601 o Browsers MUST provide a separate dialog request for screen/ 602 application sharing permissions even if the media request is made 603 at the same time as camera and microphone. 605 o The browser MUST indicate any windows which are currently being 606 shared in some unambiguous way. Windows which are not visible 607 MUST not be shared even if the application is being shared. If 608 the screen is being shared, then that MUST be indicated. 610 Clients MAY permit the formation of data channels without any direct 611 user approval. Because sites can always tunnel data through the 612 server, further restrictions on the data channel do not provide any 613 additional security. (though see Section 5.3 for a related issue). 615 Implementations which support some form of direct user authentication 616 SHOULD also provide a policy by which a user can authorize calls only 617 to specific communicating peers. Specifically, the implementation 618 SHOULD provide the following interfaces/controls: 620 o Allow future calls to this verified user. 622 o Allow future calls to any verified user who is in my system 623 address book (this only works with address book integration, of 624 course). 626 Implementations SHOULD also provide a different user interface 627 indication when calls are in progress to users whose identities are 628 directly verifiable. Section 5.5 provides more on this. 630 5.3. Communications Consent 632 Browser client implementations of WebRTC MUST implement ICE. Server 633 gateway implementations which operate only at public IP addresses 634 MUST implement either full ICE or ICE-Lite [RFC5245]. 636 Browser implementations MUST verify reachability via ICE prior to 637 sending any non-ICE packets to a given destination. Implementations 638 MUST NOT provide the ICE transaction ID to JavaScript during the 639 lifetime of the transaction (i.e., during the period when the ICE 640 stack would accept a new response for that transaction). The JS MUST 641 NOT be permitted to control the local ufrag and password, though it 642 of course knows it. 644 While continuing consent is required, the ICE [RFC5245]; Section 10 645 keepalives use STUN Binding Indications which are one-way and 646 therefore not sufficient. The current WG consensus is to use ICE 647 Binding Requests for continuing consent freshness. ICE already 648 requires that implementations respond to such requests, so this 649 approach is maximally compatible. A separate document will profile 650 the ICE timers to be used; see [I-D.muthu-behave-consent-freshness]. 652 5.4. IP Location Privacy 654 A side effect of the default ICE behavior is that the peer learns 655 one's IP address, which leaks large amounts of location information. 656 This has negative privacy consequences in some circumstances. The 657 API requirements in this section are intended to mitigate this issue. 659 Note that these requirements are NOT intended to protect the user's 660 IP address from a malicious site. In general, the site will learn at 661 least a user's server reflexive address from any HTTP transaction. 662 Rather, these requirements are intended to allow a site to cooperate 663 with the user to hide the user's IP address from the other side of 664 the call. Hiding the user's IP address from the server requires some 665 sort of explicit privacy preserving mechanism on the client (e.g., 666 Tor Browser [https://www.torproject.org/projects/torbrowser.html.en]) 667 and is out of scope for this specification. 669 API Requirement: The API MUST provide a mechanism to allow the JS to 670 suppress ICE negotiation (though perhaps to allow candidate 671 gathering) until the user has decided to answer the call [note: 672 determining when the call has been answered is a question for the 673 JS.] This enables a user to prevent a peer from learning their IP 674 address if they elect not to answer a call and also from learning 675 whether the user is online. 677 API Requirement: The API MUST provide a mechanism for the calling 678 application JS to indicate that only TURN candidates are to be 679 used. This prevents the peer from learning one's IP address at 680 all. This mechanism MUST also permit suppression of the related 681 address field, since that leaks local addresses. 683 API Requirement: The API MUST provide a mechanism for the calling 684 application to reconfigure an existing call to add non-TURN 685 candidates. Taken together, this and the previous requirement 686 allow ICE negotiation to start immediately on incoming call 687 notification, thus reducing post-dial delay, but also to avoid 688 disclosing the user's IP address until they have decided to 689 answer. They also allow users to completely hide their IP address 690 for the duration of the call. Finally, they allow a mechanism for 691 the user to optimize performance by reconfiguring to allow non- 692 turn candidates during an active call if the user decides they no 693 longer need to hide their IP address 695 Note that some enterprises may operate proxies and/or NATs designed 696 to hide internal IP addresses from the outside world. WebRTC 697 provides no explicit mechanism to allow this function. Either such 698 enterprises need to proxy the HTTP/HTTPS and modify the SDP and/or 699 the JS, or there needs to be browser support to set the "TURN-only" 700 policy regardless of the site's preferences. 702 5.5. Communications Security 704 Implementations MUST implement SRTP [RFC3711]. Implementations MUST 705 implement DTLS [RFC4347] and DTLS-SRTP [RFC5763][RFC5764] for SRTP 706 keying. Implementations MUST implement 707 [I-D.ietf-tsvwg-sctp-dtls-encaps]. 709 All media channels MUST be secured via SRTP and SRTCP. Media traffic 710 MUST NOT be sent over plain (unencrypted) RTP or RTCP; that is, 711 implementations MUST NOT negotiate cipher suites with NULL encryption 712 modes. DTLS-SRTP MUST be offered for every media channel. WebRTC 713 implementations MUST NOT offer SDP Security Descriptions [RFC4568] or 714 select it if offered. A SRTP MKI MUST NOT be used. 716 All data channels MUST be secured via DTLS. 718 All implementations MUST implement DTLS 1.0, with the cipher suite 719 TLS_ECDHE_ECDSA_WITH_AES_128_CBC_SHA with the the P-256 curve 720 [FIPS186]. The DTLS-SRTP protection profile 721 SRTP_AES128_CM_HMAC_SHA1_80 MUST be supported for SRTP. 722 Implementations SHOULD implement DTLS 1.2 with the 723 TLS_ECDHE_ECDSA_WITH_AES_128_GCM_SHA256 cipher suite. 724 Implementations MUST favor cipher suites which support PFS over non- 725 PFS cipher suites and SHOULD favor AEAD over non-AEAD cipher suites. 727 Implementations MUST NOT implement DTLS renegotiation and MUST reject 728 it with an appropriate alert ("no_renegotiation" for TLS 1.2) if 729 offered. 731 API Requirement: The API MUST generate a new authentication key pair 732 for every new call by default. This is intended to allow for 733 unlinkability. 735 API Requirement: The API MUST provide a means to reuse a key pair 736 for calls. This can be used to enable key continuity-based 737 authentication, and could be used to amortize key generation 738 costs. 740 API Requirement: Unless the user specifically configures an external 741 key pair, different key pairs MUST be used for each origin. (This 742 avoids creating a super-cookie.) 744 API Requirement: When DTLS-SRTP is used, the API MUST NOT permit the 745 JS to obtain the negotiated keying material. This requirement 746 preserves the end-to-end security of the media. 748 UI Requirements: A user-oriented client MUST provide an "inspector" 749 interface which allows the user to determine the security 750 characteristics of the media. 752 The following properties SHOULD be displayed "up-front" in the 753 browser chrome, i.e., without requiring the user to ask for them: 755 * A client MUST provide a user interface through which a user may 756 determine the security characteristics for currently-displayed 757 audio and video stream(s) 759 * A client MUST provide a user interface through which a user may 760 determine the security characteristics for transmissions of 761 their microphone audio and camera video. 763 * If the far endpoint was directly verified, either via a third- 764 party verifiable X.509 certificate or via a Web IdP mechanism 765 (see Section 5.6) the "security characteristics" MUST include 766 the verified information. X.509 identities and Web IdP 767 identities have similar semantics and should be displayed in a 768 similar way. 770 The following properties are more likely to require some "drill- 771 down" from the user: 773 * The "security characteristics" MUST indicate the cryptographic 774 algorithms in use (For example: "AES-CBC" or "Null Cipher".) 775 However, if Null ciphers are used, that MUST be presented to 776 the user at the top-level UI. 778 * The "security characteristics" MUST indicate whether PFS is 779 provided. 781 * The "security characteristics" MUST include some mechanism to 782 allow an out-of-band verification of the peer, such as a 783 certificate fingerprint or an SAS. 785 5.6. Web-Based Peer Authentication 787 In a number of cases, it is desirable for the endpoint (i.e., the 788 browser) to be able to directly identify the endpoint on the other 789 side without trusting the signaling service to which they are 790 connected. For instance, users may be making a call via a federated 791 system where they wish to get direct authentication of the other 792 side. Alternately, they may be making a call on a site which they 793 minimally trust (such as a poker site) but to someone who has an 794 identity on a site they do trust (such as a social network.) 796 Recently, a number of Web-based identity technologies (OAuth, 797 Facebook Connect etc.) have been developed. While the details vary, 798 what these technologies share is that they have a Web-based (i.e., 799 HTTP/HTTPS) identity provider which attests to your identity. For 800 instance, if I have an account at example.org, I could use the 801 example.org identity provider to prove to others that I was 802 alice@example.org. The development of these technologies allows us 803 to separate calling from identity provision: I could call you on 804 Poker Galaxy but identify myself as alice@example.org. 806 Whatever the underlying technology, the general principle is that the 807 party which is being authenticated is NOT the signaling site but 808 rather the user (and their browser). Similarly, the relying party is 809 the browser and not the signaling site. Thus, the browser MUST 810 generate the input to the IdP assertion process and display the 811 results of the verification process to the user in a way which cannot 812 be imitated by the calling site. 814 The mechanisms defined in this document do not require the browser to 815 implement any particular identity protocol or to support any 816 particular IdP. Instead, this document provides a generic interface 817 which any IdP can implement. Thus, new IdPs and protocols can be 818 introduced without change to either the browser or the calling 819 service. This avoids the need to make a commitment to any particular 820 identity protocol, although browsers may opt to directly implement 821 some identity protocols in order to provide superior performance or 822 UI properties. 824 5.6.1. Trust Relationships: IdPs, APs, and RPs 826 Any federated identity protocol has three major participants: 828 Authenticating Party (AP): The entity which is trying to establish 829 its identity. 831 Identity Provider (IdP): The entity which is vouching for the AP's 832 identity. 834 Relying Party (RP): The entity which is trying to verify the AP's 835 identity. 837 The AP and the IdP have an account relationship of some kind: the AP 838 registers with the IdP and is able to subsequently authenticate 839 directly to the IdP (e.g., with a password). This means that the 840 browser must somehow know which IdP(s) the user has an account 841 relationship with. This can either be something that the user 842 configures into the browser or that is configured at the calling site 843 and then provided to the PeerConnection by the Web application at the 844 calling site. The use case for having this information configured 845 into the browser is that the user may "log into" the browser to bind 846 it to some identity. This is becoming common in new browsers. 847 However, it should also be possible for the IdP information to simply 848 be provided by the calling application. 850 At a high level there are two kinds of IdPs: 852 Authoritative: IdPs which have verifiable control of some section 853 of the identity space. For instance, in the realm of e-mail, the 854 operator of "example.com" has complete control of the namespace 855 ending in "@example.com". Thus, "alice@example.com" is whoever 856 the operator says it is. Examples of systems with authoritative 857 identity providers include DNSSEC, RFC 4474, and Facebook Connect 858 (Facebook identities only make sense within the context of the 859 Facebook system). 861 Third-Party: IdPs which don't have control of their section of the 862 identity space but instead verify user's identities via some 863 unspecified mechanism and then attest to it. Because the IdP 864 doesn't actually control the namespace, RPs need to trust that the 865 IdP is correctly verifying AP identities, and there can 866 potentially be multiple IdPs attesting to the same section of the 867 identity space. Probably the best-known example of a third-party 868 identity provider is SSL certificates, where there are a large 869 number of CAs all of whom can attest to any domain name. 871 If an AP is authenticating via an authoritative IdP, then the RP does 872 not need to explicitly configure trust in the IdP at all. The 873 identity mechanism can directly verify that the IdP indeed made the 874 relevant identity assertion (a function provided by the mechanisms in 875 this document), and any assertion it makes about an identity for 876 which it is authoritative is directly verifiable. Note that this 877 does not mean that the IdP might not lie, but that is a 878 trustworthiness judgement that the user can make at the time he looks 879 at the identity. 881 By contrast, if an AP is authenticating via a third-party IdP, the RP 882 needs to explicitly trust that IdP (hence the need for an explicit 883 trust anchor list in PKI-based SSL/TLS clients). The list of 884 trustable IdPs needs to be configured directly into the browser, 885 either by the user or potentially by the browser manufacturer. This 886 is a significant advantage of authoritative IdPs and implies that if 887 third-party IdPs are to be supported, the potential number needs to 888 be fairly small. 890 5.6.2. Overview of Operation 892 In order to provide security without trusting the calling site, the 893 PeerConnection component of the browser must interact directly with 894 the IdP. The details of the mechanism are described in the W3C API 895 specification, but the general idea is that the PeerConnection 896 component downloads JS from a specific location on the IdP dictated 897 by the IdP domain name. That JS (the "IdP proxy") runs in an 898 isolated security context within the browser and the PeerConnection 899 talks to it via a secure message passing channel. 901 Note that there are two logically separate functions here: 903 o Identity assertion generation. 905 o Identity assertion verification. 907 The same IdP JS "endpoint" is used for both functions but of course a 908 given IdP might behave differently and load new JS to perform one 909 function or the other. 911 +--------------------------------------+ 912 | Browser | 913 | | 914 | +----------------------------------+ | 915 | | https://calling-site.example.com | | 916 | | | | 917 | | Calling JS Code | | 918 | | ^ | | 919 | +---------------|------------------+ | 920 | | API Calls | 921 | v | 922 | PeerConnection | 923 | ^ | 924 | | API Calls | 925 | +-----------|-------------+ | +---------------+ 926 | | v | | | | 927 | | IdP Proxy |<-------->| Identity | 928 | | | | | Provider | 929 | | https://idp.example.org | | | | 930 | +-------------------------+ | +---------------+ 931 | | 932 +--------------------------------------+ 934 When the PeerConnection object wants to interact with the IdP, the 935 sequence of events is as follows: 937 1. The browser (the PeerConnection component) instantiates an IdP 938 proxy. This allows the IdP to load whatever JS is necessary into 939 the proxy. The resulting code runs in the IdP's security 940 context. 942 2. The IdP registers an object with the browser that conforms to the 943 API defined in [webrtc-api]. 945 3. The browser invokes methods on the object registered by the IdP 946 proxy to create or verify identity assertions. 948 This approach allows us to decouple the browser from any particular 949 identity provider; the browser need only know how to load the IdP's 950 JavaScript--the location of which is determined based on the IdP's 951 identity--and to call the generic API for requesting and verifying 952 identity assertions. The IdP provides whatever logic is necessary to 953 bridge the generic protocol to the IdP's specific requirements. 954 Thus, a single browser can support any number of identity protocols, 955 including being forward compatible with IdPs which did not exist at 956 the time the browser was written. 958 5.6.3. Items for Standardization 960 There are two parts to this work: 962 o The precise information from the signaling message that must be 963 cryptographically bound to the user's identity and a mechanism for 964 carrying assertions in JSEP messages. This is specified in 965 Section 5.6.4. 967 o The interface to the IdP, which is defined in the companion W3C 968 WebRTC API specification [webrtc-api]. 970 The WebRTC API specification also defines JavaScript interfaces that 971 the calling application can use to specify which IdP to use. That 972 API also provides access to the assertion-generation capability and 973 the status of the validation process. 975 5.6.4. Binding Identity Assertions to JSEP Offer/Answer Transactions 977 An identity assertion binds the user's identity (as asserted by the 978 IdP) to the SDP offer/exchange transaction and specifically to the 979 media. In order to achieve this, the PeerConnection must provide the 980 DTLS-SRTP fingerprint to be bound to the identity. This is provided 981 as a JavaScript object (also known as a dictionary or hash) with a 982 single "fingerprint" key, as shown below: 984 { 985 "fingerprint": [ { 986 "algorithm": "sha-256", 987 "digest": "4A:AD:B9:B1:3F:...:E5:7C:AB" 988 }, { 989 "algorithm": "sha-1", 990 "digest": "74:E9:76:C8:19:...:F4:45:6B" 991 } ] 992 } 994 The "fingerprint" value is an array of objects. Each object in the 995 array contains "algorithm" and "digest" values, which correspond 996 directly to the algorithm and digest values in the "a=fingerprint" 997 line of the SDP [RFC8122]. 999 This object is encoded in a JSON [RFC4627] string for passing to the 1000 IdP. 1002 This structure does not need to be interpreted by the IdP or the IdP 1003 proxy. It is consumed solely by the RP's browser. The IdP merely 1004 treats it as an opaque value to be attested to. Thus, new parameters 1005 can be added to the assertion without modifying the IdP. 1007 5.6.4.1. Carrying Identity Assertions 1009 Once an IdP has generated an assertion, it is attached to the SDP 1010 message. This is done by adding a new identity attribute to the SDP. 1011 The sole contents of this value are a base-64 encoded [RFC4648] 1012 identity assertion. For example: 1014 v=0 1015 o=- 1181923068 1181923196 IN IP4 ua1.example.com 1016 s=example1 1017 c=IN IP4 ua1.example.com 1018 a=fingerprint:sha-1 \ 1019 4A:AD:B9:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB 1020 a=identity:\ 1021 eyJpZHAiOnsiZG9tYWluIjoiZXhhbXBsZS5vcmciLCJwcm90b2NvbCI6ImJvZ3Vz\ 1022 In0sImFzc2VydGlvbiI6IntcImlkZW50aXR5XCI6XCJib2JAZXhhbXBsZS5vcmdc\ 1023 IixcImNvbnRlbnRzXCI6XCJhYmNkZWZnaGlqa2xtbm9wcXJzdHV2d3l6XCIsXCJz\ 1024 aWduYXR1cmVcIjpcIjAxMDIwMzA0MDUwNlwifSJ9 1025 a=... 1026 t=0 0 1027 m=audio 6056 RTP/SAVP 0 1028 a=sendrecv 1029 ... 1031 The identity attribute attests to all "a=fingerprint" attributes in 1032 the session description. It is therefore a session-level attribute. 1034 Multiple "a=fingerprint" values can be used to offer alternative 1035 certificates for a peer. The "a=identity" attribute MUST include all 1036 fingerprint values that are included in "a=fingerprint" lines. 1038 The RP browser MUST verify that the in-use certificate for a DTLS 1039 connection is in the set of fingerprints returned from the IdP when 1040 verifying an assertion. 1042 5.6.4.2. a=identity Attribute 1044 The identity attribute is session level only. It contains an 1045 identity assertion, encoded as a base-64 string [RFC4648]. 1047 The syntax of this SDP attribute is defined using Augmented BNF 1048 [RFC5234]: 1050 identity-attribute = "identity:" identity-assertion 1051 [ SP identity-extension 1052 *(";" [ SP ] identity-extension) ] 1053 identity-assertion = base64 1054 base64 = 1*(ALPHA / DIGIT / "+" / "/" / "=" ) 1055 identity-extension = extension-att-name [ "=" extension-att-value ] 1056 extension-att-name = token 1057 extension-att-value = 1*(%x01-09 / %x0b-0c / %x0e-3a / %x3c-ff) 1058 ; byte-string from [RFC4566] omitting ";" 1060 No extensions are defined for this attribute. 1062 The identity assertion is a JSON [RFC4627] encoded dictionary that 1063 contains two values. The "assertion" attribute contains an opaque 1064 string that is consumed by the IdP. The "idp" attribute is a 1065 dictionary with one or two further values that identify the IdP, as 1066 described in Section 5.6.5. 1068 5.6.5. Determining the IdP URI 1070 In order to ensure that the IdP is under control of the domain owner 1071 rather than someone who merely has an account on the domain owner's 1072 server (e.g., in shared hosting scenarios), the IdP JavaScript is 1073 hosted at a deterministic location based on the IdP's domain name. 1074 Each IdP proxy instance is associated with two values: 1076 domain name: The IdP's domain name 1077 protocol: The specific IdP protocol which the IdP is using. This is 1078 a completely opaque IdP-specific string, but allows an IdP to 1079 implement two protocols in parallel. This value may be the empty 1080 string. If no value for protocol is provided, a value of 1081 "default" is used. 1083 Each IdP MUST serve its initial entry page (i.e., the one loaded by 1084 the IdP proxy) from a well-known URI [RFC5785]. The well-known URI 1085 for an IdP proxy is formed from the following URI components: 1087 1. The scheme, "https:". An IdP MUST be loaded using HTTPS 1088 [RFC2818]. 1090 2. The authority, which is the IdP domain name. The authority MAY 1091 contain a non-default port number. Any port number is removed 1092 when determining if an asserted identity matches the name of the 1093 IdP. The authority MUST NOT include a userinfo sub-component. 1095 3. The path, starting with "/.well-known/idp-proxy/" and appended 1096 with the IdP protocol. Note that the separator characters '/' 1097 (%2F) and '\' (%5C) MUST NOT be permitted in the protocol field, 1098 lest an attacker be able to direct requests outside of the 1099 controlled "/.well-known/" prefix. Query and fragment values MAY 1100 be used by including '?' or '#' characters. 1102 For example, for the IdP "identity.example.com" and the protocol 1103 "example", the URL would be: 1105 https://example.com/.well-known/idp-proxy/example 1107 The IdP MAY redirect requests to this URL, but they MUST retain the 1108 "https" scheme. This changes the effective origin of the IdP, but 1109 not the domain of the identities that the IdP is permitted to assert 1110 and validate. I.e., the IdP is still regarded as authoritative for 1111 the original domain. 1113 5.6.5.1. Authenticating Party 1115 How an AP determines the appropriate IdP domain is out of scope of 1116 this specification. In general, however, the AP has some actual 1117 account relationship with the IdP, as this identity is what the IdP 1118 is attesting to. Thus, the AP somehow supplies the IdP information 1119 to the browser. Some potential mechanisms include: 1121 o Provided by the user directly. 1123 o Selected from some set of IdPs known to the calling site. E.g., a 1124 button that shows "Authenticate via Facebook Connect" 1126 5.6.5.2. Relying Party 1128 Unlike the AP, the RP need not have any particular relationship with 1129 the IdP. Rather, it needs to be able to process whatever assertion 1130 is provided by the AP. As the assertion contains the IdP's identity, 1131 the URI can be constructed directly from the assertion, and thus the 1132 RP can directly verify the technical validity of the assertion with 1133 no user interaction. Authoritative assertions need only be 1134 verifiable. Third-party assertions also MUST be verified against 1135 local policy, as described in Section 5.7.1. 1137 5.6.6. Requesting Assertions 1139 The input to identity assertion is the JSON-encoded object described 1140 in Section 5.6.4 that contains the set of certificate fingerprints 1141 the browser intends to use. This string is treated as opaque from 1142 the perspective of the IdP. 1144 The browser also identifies the origin that the PeerConnection is run 1145 in, which allows the IdP to make decisions based on who is requesting 1146 the assertion. 1148 An application can optionally provide a user identifier hint when 1149 specifying an IdP. This value is a hint that the IdP can use to 1150 select amongst multiple identities, or to avoid providing assertions 1151 for unwanted identities. The "username" is a string that has no 1152 meaning to any entity other than the IdP, it can contain any data the 1153 IdP needs in order to correctly generate an assertion. 1155 An identity assertion that is successfully provided by the IdP 1156 consists of the following information: 1158 idp: The domain name of an IdP and the protocol string. This MAY 1159 identify a different IdP or protocol from the one that generated 1160 the assertion. 1162 assertion: An opaque value containing the assertion itself. This is 1163 only interpretable by the identified IdP or the IdP code running 1164 in the client. 1166 Figure 5 shows an example assertion formatted as JSON. In this case, 1167 the message has presumably been digitally signed/MACed in some way 1168 that the IdP can later verify it, but this is an implementation 1169 detail and out of scope of this document. Line breaks are inserted 1170 solely for readability. 1172 { 1173 "idp":{ 1174 "domain": "example.org", 1175 "protocol": "bogus" 1176 }, 1177 "assertion": "{\"identity\":\"bob@example.org\", 1178 \"contents\":\"abcdefghijklmnopqrstuvwyz\", 1179 \"signature\":\"010203040506\"}" 1180 } 1182 Figure 5: Example assertion 1184 For use in signaling, the assertion is serialized into JSON, 1185 base64-encoded [RFC4648], and used as the value of the "a=identity" 1186 attribute. 1188 5.6.7. Managing User Login 1190 In order to generate an identity assertion, the IdP needs proof of 1191 the user's identity. It is common practice to authenticate users 1192 (using passwords or multi-factor authentication), then use Cookies 1193 [RFC6265] or HTTP authentication [RFC2617] for subsequent exchanges. 1195 The IdP proxy is able to access cookies, HTTP authentication or other 1196 persistent session data because it operates in the security context 1197 of the IdP origin. Therefore, if a user is logged in, the IdP could 1198 have all the information needed to generate an assertion. 1200 An IdP proxy is unable to generate an assertion if the user is not 1201 logged in, or the IdP wants to interact with the user to acquire more 1202 information before generating the assertion. If the IdP wants to 1203 interact with the user before generating an assertion, the IdP proxy 1204 can fail to generate an assertion and instead indicate a URL where 1205 login should proceed. 1207 The application can then load the provided URL to enable the user to 1208 enter credentials. The communication between the application and the 1209 IdP is described in [webrtc-api]. 1211 5.7. Verifying Assertions 1213 The input to identity validation is the assertion string taken from a 1214 decoded a=identity attribute. 1216 The IdP proxy verifies the assertion. Depending on the identity 1217 protocol, the proxy might contact the IdP server or other servers. 1218 For instance, an OAuth-based protocol will likely require using the 1219 IdP as an oracle, whereas with a signature-based scheme might be able 1220 to verify the assertion without contacting the IdP, provided that it 1221 has cached the relevant public key. 1223 Regardless of the mechanism, if verification succeeds, a successful 1224 response from the IdP proxy consists of the following information: 1226 identity: The identity of the AP from the IdP's perspective. 1227 Details of this are provided in Section 5.7.1. 1229 contents: The original unmodified string provided by the AP as input 1230 to the assertion generation process. 1232 Figure 6 shows an example response formatted as JSON for illustrative 1233 purposes. 1235 { 1236 "identity": "bob@example.org", 1237 "contents": "{\"fingerprint\":[ ... ]}" 1238 } 1240 Figure 6: Example verification result 1242 5.7.1. Identity Formats 1244 The identity provided from the IdP to the RP browser MUST consist of 1245 a string representing the user's identity. This string is in the 1246 form "@", where "user" consists of any character except 1247 '@', and domain is an internationalized domain name [RFC5890]. 1249 The PeerConnection API MUST check this string as follows: 1251 1. If the domain portion of the string is equal to the domain name 1252 of the IdP proxy, then the assertion is valid, as the IdP is 1253 authoritative for this domain. Comparison of domain names is 1254 done using the label equivalence rule defined in Section 2.3.2.4 1255 of [RFC5890]. 1257 2. If the domain portion of the string is not equal to the domain 1258 name of the IdP proxy, then the PeerConnection object MUST reject 1259 the assertion unless: 1261 1. the IdP domain is trusted as an acceptable third-party IdP; 1262 and 1264 2. local policy is configured to trust this IdP domain for the 1265 domain portion of the identity string. 1267 Sites that have identities that do not fit into the RFC822 style (for 1268 instance, identifiers that are simple numeric values, or values that 1269 contain '@' characters) SHOULD convert them to this form by escaping 1270 illegal characters and appending their IdP domain (e.g., 1271 user%40133@identity.example.com), thus ensuring that they are 1272 authoritative for the identity. 1274 6. Security Considerations 1276 Much of the security analysis of this problem is contained in 1277 [I-D.ietf-rtcweb-security] or in the discussion of the particular 1278 issues above. In order to avoid repetition, this section focuses on 1279 (a) residual threats that are not addressed by this document and (b) 1280 threats produced by failure/misbehavior of one of the components in 1281 the system. 1283 6.1. Communications Security 1285 IF HTTPS is not used to secure communications to the signaling 1286 server, and the identity mechanism used in Section 5.6 is not used, 1287 then any on-path attacker can replace the DTLS-SRTP fingerprints in 1288 the handshake and thus substitute its own identity for that of either 1289 endpoint. 1291 Even if HTTPS is used, the signaling server can potentially mount a 1292 man-in-the-middle attack unless implementations have some mechanism 1293 for independently verifying keys. The UI requirements in Section 5.5 1294 are designed to provide such a mechanism for motivated/security 1295 conscious users, but are not suitable for general use. The identity 1296 service mechanisms in Section 5.6 are more suitable for general use. 1297 Note, however, that a malicious signaling service can strip off any 1298 such identity assertions, though it cannot forge new ones. Note that 1299 all of the third-party security mechanisms available (whether X.509 1300 certificates or a third-party IdP) rely on the security of the third 1301 party--this is of course also true of your connection to the Web site 1302 itself. Users who wish to assure themselves of security against a 1303 malicious identity provider can only do so by verifying peer 1304 credentials directly, e.g., by checking the peer's fingerprint 1305 against a value delivered out of band. 1307 In order to protect against malicious content JavaScript, that 1308 JavaScript MUST NOT be allowed to have direct access to---or perform 1309 computations with---DTLS keys. For instance, if content JS were able 1310 to compute digital signatures, then it would be possible for content 1311 JS to get an identity assertion for a browser's generated key and 1312 then use that assertion plus a signature by the key to authenticate a 1313 call protected under an ephemeral DH key controlled by the content 1314 JS, thus violating the security guarantees otherwise provided by the 1315 IdP mechanism. Note that it is not sufficient merely to deny the 1316 content JS direct access to the keys, as some have suggested doing 1317 with the WebCrypto API. [webcrypto]. The JS must also not be 1318 allowed to perform operations that would be valid for a DTLS 1319 endpoint. By far the safest approach is simply to deny the ability 1320 to perform any operations that depend on secret information 1321 associated with the key. Operations that depend on public 1322 information, such as exporting the public key are of course safe. 1324 6.2. Privacy 1326 The requirements in this document are intended to allow: 1328 o Users to participate in calls without revealing their location. 1330 o Potential callees to avoid revealing their location and even 1331 presence status prior to agreeing to answer a call. 1333 However, these privacy protections come at a performance cost in 1334 terms of using TURN relays and, in the latter case, delaying ICE. 1335 Sites SHOULD make users aware of these tradeoffs. 1337 Note that the protections provided here assume a non-malicious 1338 calling service. As the calling service always knows the users 1339 status and (absent the use of a technology like Tor) their IP 1340 address, they can violate the users privacy at will. Users who wish 1341 privacy against the calling sites they are using must use separate 1342 privacy enhancing technologies such as Tor. Combined WebRTC/Tor 1343 implementations SHOULD arrange to route the media as well as the 1344 signaling through Tor. Currently this will produce very suboptimal 1345 performance. 1347 Additionally, any identifier which persists across multiple calls is 1348 potentially a problem for privacy, especially for anonymous calling 1349 services. Such services SHOULD instruct the browser to use separate 1350 DTLS keys for each call and also to use TURN throughout the call. 1351 Otherwise, the other side will learn linkable information. 1352 Additionally, browsers SHOULD implement the privacy-preserving CNAME 1353 generation mode of [I-D.ietf-avtcore-6222bis]. 1355 6.3. Denial of Service 1357 The consent mechanisms described in this document are intended to 1358 mitigate denial of service attacks in which an attacker uses clients 1359 to send large amounts of traffic to a victim without the consent of 1360 the victim. While these mechanisms are sufficient to protect victims 1361 who have not implemented WebRTC at all, WebRTC implementations need 1362 to be more careful. 1364 Consider the case of a call center which accepts calls via WebRTC. 1365 An attacker proxies the call center's front-end and arranges for 1366 multiple clients to initiate calls to the call center. Note that 1367 this requires user consent in many cases but because the data channel 1368 does not need consent, he can use that directly. Since ICE will 1369 complete, browsers can then be induced to send large amounts of data 1370 to the victim call center if it supports the data channel at all. 1371 Preventing this attack requires that automated WebRTC implementations 1372 implement sensible flow control and have the ability to triage out 1373 (i.e., stop responding to ICE probes on) calls which are behaving 1374 badly, and especially to be prepared to remotely throttle the data 1375 channel in the absence of plausible audio and video (which the 1376 attacker cannot control). 1378 Another related attack is for the signaling service to swap the ICE 1379 candidates for the audio and video streams, thus forcing a browser to 1380 send video to the sink that the other victim expects will contain 1381 audio (perhaps it is only expecting audio!) potentially causing 1382 overload. Muxing multiple media flows over a single transport makes 1383 it harder to individually suppress a single flow by denying ICE 1384 keepalives. Either media-level (RTCP) mechanisms must be used or the 1385 implementation must deny responses entirely, thus terminating the 1386 call. 1388 Yet another attack, suggested by Magnus Westerlund, is for the 1389 attacker to cross-connect offers and answers as follows. It induces 1390 the victim to make a call and then uses its control of other users 1391 browsers to get them to attempt a call to someone. It then 1392 translates their offers into apparent answers to the victim, which 1393 looks like large-scale parallel forking. The victim still responds 1394 to ICE responses and now the browsers all try to send media to the 1395 victim. Implementations can defend themselves from this attack by 1396 only responding to ICE Binding Requests for a limited number of 1397 remote ufrags (this is the reason for the requirement that the JS not 1398 be able to control the ufrag and password). 1400 [I-D.ietf-rtcweb-rtp-usage] Section 13 documents a number of 1401 potential RTCP-based DoS attacks and countermeasures. 1403 Note that attacks based on confusing one end or the other about 1404 consent are possible even in the face of the third-party identity 1405 mechanism as long as major parts of the signaling messages are not 1406 signed. On the other hand, signing the entire message severely 1407 restricts the capabilities of the calling application, so there are 1408 difficult tradeoffs here. 1410 6.4. IdP Authentication Mechanism 1412 This mechanism relies for its security on the IdP and on the 1413 PeerConnection correctly enforcing the security invariants described 1414 above. At a high level, the IdP is attesting that the user 1415 identified in the assertion wishes to be associated with the 1416 assertion. Thus, it must not be possible for arbitrary third parties 1417 to get assertions tied to a user or to produce assertions that RPs 1418 will accept. 1420 6.4.1. PeerConnection Origin Check 1422 Fundamentally, the IdP proxy is just a piece of HTML and JS loaded by 1423 the browser, so nothing stops a Web attacker from creating their own 1424 IFRAME, loading the IdP proxy HTML/JS, and requesting a signature. 1425 In order to prevent this attack, we require that all signatures be 1426 tied to a specific origin ("rtcweb://...") which cannot be produced 1427 by content JavaScript. Thus, while an attacker can instantiate the 1428 IdP proxy, they cannot send messages from an appropriate origin and 1429 so cannot create acceptable assertions. I.e., the assertion request 1430 must have come from the browser. This origin check is enforced on 1431 the relying party side, not on the authenticating party side. The 1432 reason for this is to take the burden of knowing which origins are 1433 valid off of the IdP, thus making this mechanism extensible to other 1434 applications besides WebRTC. The IdP simply needs to gather the 1435 origin information (from the posted message) and attach it to the 1436 assertion. 1438 Note that although this origin check is enforced on the RP side and 1439 not at the IdP, it is absolutely imperative that it be done. The 1440 mechanisms in this document rely on the browser enforcing access 1441 restrictions on the DTLS keys and assertion requests which do not 1442 come with the right origin may be from content JS rather than from 1443 browsers, and therefore those access restrictions cannot be assumed. 1445 Note that this check only asserts that the browser (or some other 1446 entity with access to the user's authentication data) attests to the 1447 request and hence to the fingerprint. It does not demonstrate that 1448 the browser has access to the associated private key. However, 1449 attaching one's identity to a key that the user does not control does 1450 not appear to provide substantial leverage to an attacker, so a proof 1451 of possession is omitted for simplicity. 1453 6.4.2. IdP Well-known URI 1455 As described in Section 5.6.5 the IdP proxy HTML/JS landing page is 1456 located at a well-known URI based on the IdP's domain name. This 1457 requirement prevents an attacker who can write some resources at the 1458 IdP (e.g., on one's Facebook wall) from being able to impersonate the 1459 IdP. 1461 6.4.3. Privacy of IdP-generated identities and the hosting site 1463 Depending on the structure of the IdP's assertions, the calling site 1464 may learn the user's identity from the perspective of the IdP. In 1465 many cases this is not an issue because the user is authenticating to 1466 the site via the IdP in any case, for instance when the user has 1467 logged in with Facebook Connect and is then authenticating their call 1468 with a Facebook identity. However, in other case, the user may not 1469 have already revealed their identity to the site. In general, IdPs 1470 SHOULD either verify that the user is willing to have their identity 1471 revealed to the site (e.g., through the usual IdP permissions dialog) 1472 or arrange that the identity information is only available to known 1473 RPs (e.g., social graph adjacencies) but not to the calling site. 1474 The "origin" field of the signature request can be used to check that 1475 the user has agreed to disclose their identity to the calling site; 1476 because it is supplied by the PeerConnection it can be trusted to be 1477 correct. 1479 6.4.4. Security of Third-Party IdPs 1481 As discussed above, each third-party IdP represents a new universal 1482 trust point and therefore the number of these IdPs needs to be quite 1483 limited. Most IdPs, even those which issue unqualified identities 1484 such as Facebook, can be recast as authoritative IdPs (e.g., 1485 123456@facebook.com). However, in such cases, the user interface 1486 implications are not entirely desirable. One intermediate approach 1487 is to have special (potentially user configurable) UI for large 1488 authoritative IdPs, thus allowing the user to instantly grasp that 1489 the call is being authenticated by Facebook, Google, etc. 1491 6.4.5. Web Security Feature Interactions 1493 A number of optional Web security features have the potential to 1494 cause issues for this mechanism, as discussed below. 1496 6.4.5.1. Popup Blocking 1498 The IdP proxy is unable to generate popup windows, dialogs or any 1499 other form of user interactions. This prevents the IdP proxy from 1500 being used to circumvent user interaction. The "LOGINNEEDED" message 1501 allows the IdP proxy to inform the calling site of a need for user 1502 login, providing the information necessary to satisfy this 1503 requirement without resorting to direct user interaction from the IdP 1504 proxy itself. 1506 6.4.5.2. Third Party Cookies 1508 Some browsers allow users to block third party cookies (cookies 1509 associated with origins other than the top level page) for privacy 1510 reasons. Any IdP which uses cookies to persist logins will be broken 1511 by third-party cookie blocking. One option is to accept this as a 1512 limitation; another is to have the PeerConnection object disable 1513 third-party cookie blocking for the IdP proxy. 1515 7. IANA Considerations 1517 This specification defines the "identity" SDP attribute per the 1518 procedures of Section 8.2.4 of [RFC4566]. The required information 1519 for the registration is included here: 1521 Contact Name: Eric Rescorla (ekr@rftm.com) 1523 Attribute Name: identity 1525 Long Form: identity 1527 Type of Attribute: session-level 1529 Charset Considerations: This attribute is not subject to the charset 1530 attribute. 1532 Purpose: This attribute carries an identity assertion, binding an 1533 identity to the transport-level security session. 1535 Appropriate Values: See Section 5.6.4.2 of RFCXXXX [[Editor Note: 1536 This document. 1538 8. Acknowledgements 1540 Bernard Aboba, Harald Alvestrand, Richard Barnes, Dan Druta, Cullen 1541 Jennings, Hadriel Kaplan, Matthew Kaufman, Jim McEachern, Martin 1542 Thomson, Magnus Westerland. Matthew Kaufman provided the UI material 1543 in Section 5.5. 1545 9. Changes 1547 9.1. Changes since -10 1549 Update cipher suite profiles. 1551 Rework IdP interaction based on implementation experience in Firefox. 1553 9.2. Changes since -06 1555 Replaced RTCWEB and RTC-Web with WebRTC, except when referring to the 1556 IETF WG 1558 Forbade use in mixed content as discussed in Orlando. 1560 Added a requirement to surface NULL ciphers to the top-level. 1562 Tried to clarify SRTP versus DTLS-SRTP. 1564 Added a section on screen sharing permissions. 1566 Assorted editorial work. 1568 9.3. Changes since -05 1570 The following changes have been made since the -05 draft. 1572 o Response to comments from Richard Barnes 1574 o More explanation of the IdP security properties and the federation 1575 use case. 1577 o Editorial cleanup. 1579 9.4. Changes since -03 1581 Version -04 was a version control mistake. Please ignore. 1583 The following changes have been made since the -04 draft. 1585 o Move origin check from IdP to RP per discussion in YVR. 1587 o Clarified treatment of X.509-level identities. 1589 o Editorial cleanup. 1591 9.5. Changes since -03 1593 9.6. Changes since -02 1595 The following changes have been made since the -02 draft. 1597 o Forbid persistent HTTP permissions. 1599 o Clarified the text in S 5.4 to clearly refer to requirements on 1600 the API to provide functionality to the site. 1602 o Fold in the IETF portion of draft-rescorla-rtcweb-generic-idp 1604 o Retarget the continuing consent section to assume Binding Requests 1606 o Added some more privacy and linkage text in various places. 1608 o Editorial improvements 1610 10. References 1612 10.1. Normative References 1614 [FIPS186] National Institute of Standards and Technology (NIST), 1615 "Digital Signature Standard (DSS)", NIST PUB 186-4 , July 1616 2013. 1618 [I-D.ietf-avtcore-6222bis] 1619 Begen, A., Perkins, C., Wing, D., and E. Rescorla, 1620 "Guidelines for Choosing RTP Control Protocol (RTCP) 1621 Canonical Names (CNAMEs)", draft-ietf-avtcore-6222bis-06 1622 (work in progress), July 2013. 1624 [I-D.ietf-rtcweb-rtp-usage] 1625 Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time 1626 Communication (WebRTC): Media Transport and Use of RTP", 1627 draft-ietf-rtcweb-rtp-usage-26 (work in progress), March 1628 2016. 1630 [I-D.ietf-rtcweb-security] 1631 Rescorla, E., "Security Considerations for WebRTC", draft- 1632 ietf-rtcweb-security-09 (work in progress), October 2017. 1634 [I-D.ietf-tsvwg-sctp-dtls-encaps] 1635 Tuexen, M., Stewart, R., Jesup, R., and S. Loreto, "DTLS 1636 Encapsulation of SCTP Packets", draft-ietf-tsvwg-sctp- 1637 dtls-encaps-09 (work in progress), January 2015. 1639 [I-D.muthu-behave-consent-freshness] 1640 Perumal, M., Wing, D., R, R., and T. Reddy, "STUN Usage 1641 for Consent Freshness", draft-muthu-behave-consent- 1642 freshness-04 (work in progress), July 2013. 1644 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 1645 Requirement Levels", BCP 14, RFC 2119, 1646 DOI 10.17487/RFC2119, March 1997, . 1649 [RFC2818] Rescorla, E., "HTTP Over TLS", RFC 2818, 1650 DOI 10.17487/RFC2818, May 2000, . 1653 [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. 1654 Norrman, "The Secure Real-time Transport Protocol (SRTP)", 1655 RFC 3711, DOI 10.17487/RFC3711, March 2004, 1656 . 1658 [RFC4347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer 1659 Security", RFC 4347, DOI 10.17487/RFC4347, April 2006, 1660 . 1662 [RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session 1663 Description Protocol", RFC 4566, DOI 10.17487/RFC4566, 1664 July 2006, . 1666 [RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session 1667 Description Protocol (SDP) Security Descriptions for Media 1668 Streams", RFC 4568, DOI 10.17487/RFC4568, July 2006, 1669 . 1671 [RFC4627] Crockford, D., "The application/json Media Type for 1672 JavaScript Object Notation (JSON)", RFC 4627, 1673 DOI 10.17487/RFC4627, July 2006, . 1676 [RFC4648] Josefsson, S., "The Base16, Base32, and Base64 Data 1677 Encodings", RFC 4648, DOI 10.17487/RFC4648, October 2006, 1678 . 1680 [RFC5234] Crocker, D., Ed. and P. Overell, "Augmented BNF for Syntax 1681 Specifications: ABNF", STD 68, RFC 5234, 1682 DOI 10.17487/RFC5234, January 2008, . 1685 [RFC5245] Rosenberg, J., "Interactive Connectivity Establishment 1686 (ICE): A Protocol for Network Address Translator (NAT) 1687 Traversal for Offer/Answer Protocols", RFC 5245, 1688 DOI 10.17487/RFC5245, April 2010, . 1691 [RFC5246] Dierks, T. and E. Rescorla, "The Transport Layer Security 1692 (TLS) Protocol Version 1.2", RFC 5246, 1693 DOI 10.17487/RFC5246, August 2008, . 1696 [RFC5763] Fischl, J., Tschofenig, H., and E. Rescorla, "Framework 1697 for Establishing a Secure Real-time Transport Protocol 1698 (SRTP) Security Context Using Datagram Transport Layer 1699 Security (DTLS)", RFC 5763, DOI 10.17487/RFC5763, May 1700 2010, . 1702 [RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer 1703 Security (DTLS) Extension to Establish Keys for the Secure 1704 Real-time Transport Protocol (SRTP)", RFC 5764, 1705 DOI 10.17487/RFC5764, May 2010, . 1708 [RFC5785] Nottingham, M. and E. Hammer-Lahav, "Defining Well-Known 1709 Uniform Resource Identifiers (URIs)", RFC 5785, 1710 DOI 10.17487/RFC5785, April 2010, . 1713 [RFC5890] Klensin, J., "Internationalized Domain Names for 1714 Applications (IDNA): Definitions and Document Framework", 1715 RFC 5890, DOI 10.17487/RFC5890, August 2010, 1716 . 1718 [RFC6454] Barth, A., "The Web Origin Concept", RFC 6454, 1719 DOI 10.17487/RFC6454, December 2011, . 1722 [RFC8122] Lennox, J. and C. Holmberg, "Connection-Oriented Media 1723 Transport over the Transport Layer Security (TLS) Protocol 1724 in the Session Description Protocol (SDP)", RFC 8122, 1725 DOI 10.17487/RFC8122, March 2017, . 1728 [webcrypto] 1729 Dahl, Sleevi, "Web Cryptography API", June 2013. 1731 Available at http://www.w3.org/TR/WebCryptoAPI/ 1733 [webrtc-api] 1734 Bergkvist, Burnett, Jennings, Narayanan, "WebRTC 1.0: 1735 Real-time Communication Between Browsers", October 2011. 1737 Available at http://dev.w3.org/2011/webrtc/editor/ 1738 webrtc.html 1740 10.2. Informative References 1742 [I-D.ietf-rtcweb-jsep] 1743 Uberti, J., Jennings, C., and E. Rescorla, "JavaScript 1744 Session Establishment Protocol", draft-ietf-rtcweb-jsep-24 1745 (work in progress), October 2017. 1747 [RFC2617] Franks, J., Hallam-Baker, P., Hostetler, J., Lawrence, S., 1748 Leach, P., Luotonen, A., and L. Stewart, "HTTP 1749 Authentication: Basic and Digest Access Authentication", 1750 RFC 2617, DOI 10.17487/RFC2617, June 1999, 1751 . 1753 [RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, 1754 A., Peterson, J., Sparks, R., Handley, M., and E. 1755 Schooler, "SIP: Session Initiation Protocol", RFC 3261, 1756 DOI 10.17487/RFC3261, June 2002, . 1759 [RFC5705] Rescorla, E., "Keying Material Exporters for Transport 1760 Layer Security (TLS)", RFC 5705, DOI 10.17487/RFC5705, 1761 March 2010, . 1763 [RFC6265] Barth, A., "HTTP State Management Mechanism", RFC 6265, 1764 DOI 10.17487/RFC6265, April 2011, . 1767 [RFC6455] Fette, I. and A. Melnikov, "The WebSocket Protocol", 1768 RFC 6455, DOI 10.17487/RFC6455, December 2011, 1769 . 1771 [XmlHttpRequest] 1772 van Kesteren, A., "XMLHttpRequest Level 2", January 2012. 1774 Appendix A. Example IdP Bindings to Specific Protocols 1776 [[TODO: These still need some cleanup.]] 1778 This section provides some examples of how the mechanisms described 1779 in this document could be used with existing authentication protocols 1780 such as OAuth. Note that this does not require browser-level support 1781 for either protocol. Rather, the protocols can be fit into the 1782 generic framework. 1784 A.1. OAuth 1786 While OAuth is not directly designed for user-to-user authentication, 1787 with a little lateral thinking it can be made to serve. We use the 1788 following mapping of OAuth concepts to WebRTC concepts: 1790 +----------------------+----------------------+ 1791 | OAuth | WebRTC | 1792 +----------------------+----------------------+ 1793 | Client | Relying party | 1794 | Resource owner | Authenticating party | 1795 | Authorization server | Identity service | 1796 | Resource server | Identity service | 1797 +----------------------+----------------------+ 1799 Table 1 1801 The idea here is that when Alice wants to authenticate to Bob (i.e., 1802 for Bob to be aware that she is calling). In order to do this, she 1803 allows Bob to see a resource on the identity provider that is bound 1804 to the call, her identity, and her public key. Then Bob retrieves 1805 the resource from the identity provider, thus verifying the binding 1806 between Alice and the call. 1808 Alice IdP Bob 1809 --------------------------------------------------------- 1810 Call-Id, Fingerprint -------> 1811 <------------------- Auth Code 1812 Auth Code ----------------------------------------------> 1813 <----- Get Token + Auth Code 1814 Token ---------------------> 1815 <------------- Get call-info 1816 Call-Id, Fingerprint ------> 1818 This is a modified version of a common OAuth flow, but omits the 1819 redirects required to have the client point the resource owner to the 1820 IdP, which is acting as both the resource server and the 1821 authorization server, since Alice already has a handle to the IdP. 1823 Above, we have referred to "Alice", but really what we mean is the 1824 PeerConnection. Specifically, the PeerConnection will instantiate an 1825 IFRAME with JS from the IdP and will use that IFRAME to communicate 1826 with the IdP, authenticating with Alice's identity (e.g., cookie). 1827 Similarly, Bob's PeerConnection instantiates an IFRAME to talk to the 1828 IdP. 1830 Author's Address 1832 Eric Rescorla 1833 RTFM, Inc. 1834 2064 Edgewood Drive 1835 Palo Alto, CA 94303 1836 USA 1838 Phone: +1 650 678 2350 1839 Email: ekr@rtfm.com