--- 1/draft-ietf-rtcweb-security-arch-05.txt 2013-01-23 02:01:37.047792785 +0100 +++ 2/draft-ietf-rtcweb-security-arch-06.txt 2013-01-23 02:01:37.127793604 +0100 @@ -1,18 +1,18 @@ RTCWEB E. Rescorla Internet-Draft RTFM, Inc. -Intended status: Standards Track October 22, 2012 -Expires: April 25, 2013 +Intended status: Standards Track January 22, 2013 +Expires: July 26, 2013 RTCWEB Security Architecture - draft-ietf-rtcweb-security-arch-05 + draft-ietf-rtcweb-security-arch-06 Abstract The Real-Time Communications on the Web (RTCWEB) working group is tasked with standardizing protocols for enabling real-time communications within user-agents using web technologies (e.g JavaScript). The major use cases for RTCWEB technology are real-time audio and/or video calls, Web conferencing, and direct data transfer. Unlike most conventional real-time systems (e.g., SIP-based soft phones) RTCWEB communications are directly controlled by some Web @@ -43,25 +43,25 @@ Internet-Drafts are working documents of the Internet Engineering Task Force (IETF). Note that other groups may also distribute working documents as Internet-Drafts. The list of current Internet- Drafts is at http://datatracker.ietf.org/drafts/current/. Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as "work in progress." - This Internet-Draft will expire on April 25, 2013. + This Internet-Draft will expire on July 26, 2013. Copyright Notice - Copyright (c) 2012 IETF Trust and the persons identified as the + Copyright (c) 2013 IETF Trust and the persons identified as the document authors. All rights reserved. This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents (http://trustee.ietf.org/license-info) in effect on the date of publication of this document. Please review these documents carefully, as they describe your rights and restrictions with respect to this document. Code Components extracted from this document must include Simplified BSD License text as described in Section 4.e of the Trust Legal Provisions and are provided without warranty as @@ -75,68 +75,70 @@ Without obtaining an adequate license from the person(s) controlling the copyright in such materials, this document may not be modified outside the IETF Standards Process, and derivative works of it may not be created outside the IETF Standards Process, except to format it for publication as an RFC or to translate it into languages other than English. Table of Contents 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 5 - 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 5 - 3. Trust Model . . . . . . . . . . . . . . . . . . . . . . . . . 5 - 3.1. Authenticated Entities . . . . . . . . . . . . . . . . . . 6 - 3.2. Unauthenticated Entities . . . . . . . . . . . . . . . . . 6 + 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 6 + 3. Trust Model . . . . . . . . . . . . . . . . . . . . . . . . . 6 + 3.1. Authenticated Entities . . . . . . . . . . . . . . . . . . 7 + 3.2. Unauthenticated Entities . . . . . . . . . . . . . . . . . 7 4. Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . 7 - 4.1. Initial Signaling . . . . . . . . . . . . . . . . . . . . 8 - 4.2. Media Consent Verification . . . . . . . . . . . . . . . . 10 - 4.3. DTLS Handshake . . . . . . . . . . . . . . . . . . . . . . 10 - 4.4. Communications and Consent Freshness . . . . . . . . . . . 11 - 5. Detailed Technical Description . . . . . . . . . . . . . . . . 11 - 5.1. Origin and Web Security Issues . . . . . . . . . . . . . . 11 - 5.2. Device Permissions Model . . . . . . . . . . . . . . . . . 12 - 5.3. Communications Consent . . . . . . . . . . . . . . . . . . 13 - 5.4. IP Location Privacy . . . . . . . . . . . . . . . . . . . 14 - 5.5. Communications Security . . . . . . . . . . . . . . . . . 15 - 5.6. Web-Based Peer Authentication . . . . . . . . . . . . . . 16 - 5.6.1. Trust Relationships: IdPs, APs, and RPs . . . . . . . 17 - 5.6.2. Overview of Operation . . . . . . . . . . . . . . . . 18 - 5.6.3. Items for Standardization . . . . . . . . . . . . . . 20 + 4.1. Initial Signaling . . . . . . . . . . . . . . . . . . . . 10 + 4.2. Media Consent Verification . . . . . . . . . . . . . . . . 12 + 4.3. DTLS Handshake . . . . . . . . . . . . . . . . . . . . . . 13 + 4.4. Communications and Consent Freshness . . . . . . . . . . . 13 + 5. Detailed Technical Description . . . . . . . . . . . . . . . . 14 + 5.1. Origin and Web Security Issues . . . . . . . . . . . . . . 14 + 5.2. Device Permissions Model . . . . . . . . . . . . . . . . . 14 + 5.3. Communications Consent . . . . . . . . . . . . . . . . . . 16 + 5.4. IP Location Privacy . . . . . . . . . . . . . . . . . . . 17 + 5.5. Communications Security . . . . . . . . . . . . . . . . . 18 + 5.6. Web-Based Peer Authentication . . . . . . . . . . . . . . 19 + 5.6.1. Trust Relationships: IdPs, APs, and RPs . . . . . . . 20 + 5.6.2. Overview of Operation . . . . . . . . . . . . . . . . 21 + 5.6.3. Items for Standardization . . . . . . . . . . . . . . 23 5.6.4. Binding Identity Assertions to JSEP Offer/Answer - Transactions . . . . . . . . . . . . . . . . . . . . . 20 - 5.6.4.1. Input to Assertion Generation Process . . . . . . 20 - 5.6.4.2. Carrying Identity Assertions . . . . . . . . . . . 21 - 5.6.5. IdP Interaction Details . . . . . . . . . . . . . . . 21 - 5.6.5.1. General Message Structure . . . . . . . . . . . . 21 - 5.6.5.2. IdP Proxy Setup . . . . . . . . . . . . . . . . . 22 - 5.7. Security Considerations . . . . . . . . . . . . . . . . . 27 - 5.7.1. Communications Security . . . . . . . . . . . . . . . 27 - 5.7.2. Privacy . . . . . . . . . . . . . . . . . . . . . . . 28 - 5.7.3. Denial of Service . . . . . . . . . . . . . . . . . . 29 - 5.7.4. IdP Authentication Mechanism . . . . . . . . . . . . . 30 - 5.7.4.1. PeerConnection Origin Check . . . . . . . . . . . 30 - 5.7.4.2. IdP Well-known URI . . . . . . . . . . . . . . . . 30 + Transactions . . . . . . . . . . . . . . . . . . . . . 23 + 5.6.4.1. Input to Assertion Generation Process . . . . . . 23 + 5.6.4.2. Carrying Identity Assertions . . . . . . . . . . . 24 + 5.6.5. IdP Interaction Details . . . . . . . . . . . . . . . 24 + 5.6.5.1. General Message Structure . . . . . . . . . . . . 24 + 5.6.5.2. IdP Proxy Setup . . . . . . . . . . . . . . . . . 25 + 5.7. Security Considerations . . . . . . . . . . . . . . . . . 30 + 5.7.1. Communications Security . . . . . . . . . . . . . . . 30 + 5.7.2. Privacy . . . . . . . . . . . . . . . . . . . . . . . 31 + 5.7.3. Denial of Service . . . . . . . . . . . . . . . . . . 32 + 5.7.4. IdP Authentication Mechanism . . . . . . . . . . . . . 33 + 5.7.4.1. PeerConnection Origin Check . . . . . . . . . . . 33 + 5.7.4.2. IdP Well-known URI . . . . . . . . . . . . . . . . 34 5.7.4.3. Privacy of IdP-generated identities and the - hosting site . . . . . . . . . . . . . . . . . . . 30 - 5.7.4.4. Security of Third-Party IdPs . . . . . . . . . . . 31 - 5.7.4.5. Web Security Feature Interactions . . . . . . . . 31 - 6. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 32 - 7. Changes since -03 . . . . . . . . . . . . . . . . . . . . . . 32 - 8. Changes since -03 . . . . . . . . . . . . . . . . . . . . . . 32 - 9. Changes since -02 . . . . . . . . . . . . . . . . . . . . . . 32 - 10. References . . . . . . . . . . . . . . . . . . . . . . . . . . 32 - 10.1. Normative References . . . . . . . . . . . . . . . . . . . 32 - 10.2. Informative References . . . . . . . . . . . . . . . . . . 33 - Appendix A. Example IdP Bindings to Specific Protocols . . . . . 34 - A.1. BrowserID . . . . . . . . . . . . . . . . . . . . . . . . 34 - A.2. OAuth . . . . . . . . . . . . . . . . . . . . . . . . . . 37 - Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 38 + hosting site . . . . . . . . . . . . . . . . . . . 34 + 5.7.4.4. Security of Third-Party IdPs . . . . . . . . . . . 34 + 5.7.4.5. Web Security Feature Interactions . . . . . . . . 35 + 6. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 35 + 7. Changes . . . . . . . . . . . . . . . . . . . . . . . . . . . 35 + 7.1. Changes since -05 . . . . . . . . . . . . . . . . . . . . 36 + 7.2. Changes since -03 . . . . . . . . . . . . . . . . . . . . 36 + 7.3. Changes since -03 . . . . . . . . . . . . . . . . . . . . 36 + 7.4. Changes since -02 . . . . . . . . . . . . . . . . . . . . 36 + 8. References . . . . . . . . . . . . . . . . . . . . . . . . . . 36 + 8.1. Normative References . . . . . . . . . . . . . . . . . . . 36 + 8.2. Informative References . . . . . . . . . . . . . . . . . . 37 + Appendix A. Example IdP Bindings to Specific Protocols . . . . . 38 + A.1. BrowserID . . . . . . . . . . . . . . . . . . . . . . . . 38 + A.2. OAuth . . . . . . . . . . . . . . . . . . . . . . . . . . 41 + Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 42 1. Introduction The Real-Time Communications on the Web (RTCWEB) working group is tasked with standardizing protocols for real-time communications between Web browsers. The major use cases for RTCWEB technology are real-time audio and/or video calls, Web conferencing, and direct data transfer. Unlike most conventional real-time systems, (e.g., SIP- based[RFC3261] soft phones) RTCWEB communications are directly controlled by some Web server, as shown in Figure 1. @@ -154,20 +156,45 @@ v v JS API JS API +-----------+ +-----------+ | | Media | | | Browser |<---------->| Browser | | | | | +-----------+ +-----------+ Figure 1: A simple RTCWEB system + A more complicated system might allow for interdomain calling, as + shown in Figure 2. The protocol to be used between the domains is + not standardized by RTCWEB, but given the installed base and the form + of the RTCWEB API is likely to be something SDP-based like SIP or + XMPP. + + +--------------+ +--------------+ + | | SIP,XMPP,...| | + | Web Server |<----------->| Web Server | + | | | | + +--------------+ +--------------+ + ^ ^ + | | + HTTP | | HTTP + | | + v v + JS API JS API + +-----------+ +-----------+ + | | Media | | + | Browser |<---------------->| Browser | + | | | | + +-----------+ +-----------+ + + Figure 2: A multidomain RTCWEB system + This system presents a number of new security challenges, which are analyzed in [I-D.ietf-rtcweb-security]. This document describes a security architecture for RTCWEB which addresses the threats and requirements described in that document. 2. Terminology The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in RFC 2119 [RFC2119]. @@ -238,32 +265,46 @@ media privacy with the minimal level of trust in the calling service. Simpler versions with lower levels of security are also possible and are noted in the text where applicable. It's also important to recognize the tension between security (or performance) and privacy. The example shown here is aimed towards settings where we are more concerned about secure calling than about privacy, but as we shall see, there are settings where one might wish to make different tradeoffs--this architecture is still compatible with those settings. For the purposes of this example, we assume the topology shown in the - figure below. This topology is derived from the topology shown in + figures below. This topology is derived from the topology shown in Figure 1, but separates Alice and Bob's identities from the process of signaling. Specifically, Alice and Bob have relationships with - some Identity Provider (IdP) that supports a protocol such OpenID or - BrowserID) that can be used to attest to their identity. This - separation isn't particularly important in "closed world" cases where - Alice and Bob are users on the same social network and have - identities based on that network. However, there are important - settings where that is not the case, such as federation (calls from - one network to another) and calling on untrusted sites, such as where - two users who have a relationship via a given social network want to - call each other on another, untrusted, site, such as a poker site. + some Identity Provider (IdP) that supports a protocol such as OpenID + or BrowserID) that can be used to demonstrate their identity to other + parties. For instance, Alice might have an account with a social + network which she can then use to authenticate to other web sites + without explicitly having an account with those sites; this is a + fairly conventional pattern on the Web. Section 5.6.1 provides an + overview of Identity Providers and the relevant terminology. + + This separation of identity provision and signaling isn't + particularly important in "closed world" cases where Alice and Bob + are users on the same social network and have identities based on + that domain (Figure 3) However, there are important settings where + that is not the case, such as federation (calls from one domain to + another; Figure 4) and calling on untrusted sites, such as where two + users who have a relationship via a given social network want to call + each other on another, untrusted, site, such as a poker site. + + Note that the servers themselves are also authenticated by an + external identity service, the SSL/TLS certificate infrastructure + (not shown). As is conventional in the Web, all identities are + ultimately rooted that system. For instance, when an IdP makes an + identity assertion, the Relying Party consuming that assertion is + able to verify because it is able to connect to the IdP via HTTPS. +----------------+ | | | Signaling | | Server | | | +----------------+ ^ ^ / \ HTTPS / \ HTTPS @@ -278,37 +319,73 @@ +-----------+ +-----------+ ^ ^--+ +--^ ^ | | | | v | | v +-----------+ | | +-----------+ | |<--------+ | | | IdP | | | IdP | | | +------->| | +-----------+ +-----------+ - Figure 2: A call with IdP-based identity + Figure 3: A call with IdP-based identity + + Figure 4 shows essentially the same calling scenario but with a call + between two separate domains (i.e., a federated case). As mentioned + above, the domains communicate by some unspecified protocol and + providing separate signaling and identity allows for calls to be + authenticated regardless of the details of the inter-domain protocol. + + +----------------+ Unspecified +----------------+ + | | protocol | | + | Signaling |<----------------->| Signaling | + | Server | (SIP, XMPP, ...) | Server | + | | | | + +----------------+ +----------------+ + ^ ^ + | | + HTTPS | | HTTPS + | | + | | + v v + JS API JS API + +-----------+ +-----------+ + | | Media | | + Alice | Browser |<--------------------------->| Browser | Bob + | | DTLS-SRTP | | + +-----------+ +-----------+ + ^ ^--+ +--^ ^ + | | | | + v | | v + +-----------+ | | +-----------+ + | |<-------------------------+ | | + | IdP | | | IdP | + | | +------------------------>| | + +-----------+ +-----------+ + + Figure 4: A federated call with IdP-based identity 4.1. Initial Signaling - Alice and Bob are both users of a common calling service; they both - have approved the calling service to make calls (we defer the - discussion of device access permissions till later). They are both - connected to the calling service via HTTPS and so know the origin - with some level of confidence. They also have accounts with some - identity provider. This sort of identity service is becoming - increasingly common in the Web environment in technologies such - (BrowserID, Federated Google Login, Facebook Connect, OAuth, OpenID, - WebFinger), and is often provided as a side effect service of your - ordinary accounts with some service. In this example, we show Alice - and Bob using a separate identity service, though they may actually - be using the same identity service as calling service or have no - identity service at all. + For simplicity, assume the topology in Figure 3. Alice and Bob are + both users of a common calling service; they both have approved the + calling service to make calls (we defer the discussion of device + access permissions till later). They are both connected to the + calling service via HTTPS and so know the origin with some level of + confidence. They also have accounts with some identity provider. + This sort of identity service is becoming increasingly common in the + Web environment in technologies such (BrowserID, Federated Google + Login, Facebook Connect, OAuth, OpenID, WebFinger), and is often + provided as a side effect service of a user's ordinary accounts with + some service. In this example, we show Alice and Bob using a + separate identity service, though the identity service may be the + same entity as the calling service or there may be no identity + service at all. Alice is logged onto the calling service and decides to call Bob. She can see from the calling service that he is online and the calling service presents a JS UI in the form of a button next to Bob's name which says "Call". Alice clicks the button, which initiates a JS callback that instantiates a PeerConnection object. This does not require a security check: JS from any origin is allowed to get this far. Once the PeerConnection is created, the calling service JS needs to @@ -317,182 +394,209 @@ to a video input. At this point the first security check is required: untrusted origins are not allowed to access the camera and microphone. In this case, because Alice is a long-term user of the calling service, she has made a permissions grant (i.e., a setting in the browser) to allow the calling service to access her camera and microphone any time it wants. The browser checks this setting when the camera and microphone requests are made and thus allows them. In the current W3C API, once some streams have been added, Alice's browser + JS generates a signaling message [I-D.ietf-rtcweb-jsep] - contianing: + containing: o Media channel information o ICE candidates - o A fingerprint attribute binding the communication to Alice's - public key [RFC5763] + o A fingerprint attribute binding the communication to a key pair + [RFC5763]. Note that this key may simply be ephemerally generated + for this call or specific to this domain, and Alice may have a + large number of such keys. Prior to sending out the signaling message, the PeerConnection code contacts the identity service and obtains an assertion binding Alice's identity to her fingerprint. The exact details depend on the identity service (though as discussed in Section 5.6 PeerConnection can be agnostic to them), but for now it's easiest to think of as a BrowserID assertion. The assertion may bind other information to the identity besides the fingerprint, but at minimum it needs to bind the fingerprint. This message is sent to the signaling server, e.g., by XMLHttpRequest [XmlHttpRequest] or by WebSockets [RFC6455] The signaling server processes the message from Alice's browser, determines that this is a call to Bob and sends a signaling message to Bob's browser (again, the format is currently undefined). The JS on Bob's browser processes it, and alerts Bob to the incoming call and to Alice's identity. In this case, Alice has provided an identity assertion and so Bob's browser contacts Alice's identity provider (again, this is done in a generic way so the browser has no specific knowledge of the IdP) to verify the assertion. This allows the browser to display a - trusted element indicating that a call is coming in from Alice. If - Alice is in Bob's address book, then this interface might also - include her real name, a picture, etc. The calling site will also - provide some user interface element (e.g., a button) to allow Bob to - answer the call, though this is most likely not part of the trusted - UI. + trusted element in the browser chrome indicating that a call is + coming in from Alice. If Alice is in Bob's address book, then this + interface might also include her real name, a picture, etc. The + calling site will also provide some user interface element (e.g., a + button) to allow Bob to answer the call, though this is most likely + not part of the trusted UI. If Bob agrees [I am ignoring early media for now], a PeerConnection is instantiated with the message from Alice's side. Then, a similar process occurs as on Alice's browser: Bob's browser verifies that the calling service is approved, the media streams are created, and a return signaling message containing media information, ICE candidates, and a fingerprint is sent back to Alice via the signaling service. If Bob has a relationship with an IdP, the message will also come with an identity assertion. At this point, Alice and Bob each know that the other party wants to have a secure call with them. Based purely on the interface provided by the signaling server, they know that the signaling server claims - that the call is from Alice to Bob. Because the far end sent an - identity assertion along with their message, they know that this is - verifiable from the IdP as well. Of course, the call works perfectly - well if either Alice or Bob doesn't have a relationship with an IdP; - they just get a lower level of assurance. Moreover, Alice might wish - to make an anonymous call through an anonymous calling site, in which - case she would of course just not provide any identity assertion and - the calling site would mask her identity from Bob. + that the call is from Alice to Bob. This level of security is + provided merely by having the fingerprint in the message and having + that message received securely from the signaling server. Because + the far end sent an identity assertion along with their message, they + know that this is verifiable from the IdP as well. Note that if the + call is federated, as shown in Figure 4 then Alice is able to verify + Bob's identity in a way that is not mediated by either her signaling + server or Bob's. Rather, she verifies it directly with Bob's IdP. + + Of course, the call works perfectly well if either Alice or Bob + doesn't have a relationship with an IdP; they just get a lower level + of assurance. I.e., they simply have whatever information their + calling site claims about the caller/calllee's identity. Moreover, + Alice might wish to make an anonymous call through an anonymous + calling site, in which case she would of course just not provide any + identity assertion and the calling site would mask her identity from + Bob. 4.2. Media Consent Verification - As described in ([I-D.ietf-rtcweb-security]; Section 4.2) This - proposal specifies that media consent verification be performed via - ICE. Thus, Alice and Bob perform ICE checks with each other. At the - completion of these checks, they are ready to send non-ICE data. + As described in ([I-D.ietf-rtcweb-security]; Section 4.2) media + consent verification is provided via ICE. Thus, Alice and Bob + perform ICE checks with each other. At the completion of these + checks, they are ready to send non-ICE data. At this point, Alice knows that (a) Bob (assuming he is verified via his IdP) or someone else who the signaling service is claiming is Bob is willing to exchange traffic with her and (b) that either Bob is at the IP address which she has verified via ICE or there is an attacker who is on-path to that IP address detouring the traffic. Note that - it is not possible for an attacker who is on-path but not attached to - the signaling service to spoof these checks because they do not have - the ICE credentials. Bob's security guarantees with respect to Alice - are the converse of this. + it is not possible for an attacker who is on-path between Alice and + Bob but not attached to the signaling service to spoof these checks + because they do not have the ICE credentials. Bob's has the same + security guarantees with respect to Alice. 4.3. DTLS Handshake Once the ICE checks have completed [more specifically, once some ICE checks have completed], Alice and Bob can set up a secure channel. This is performed via DTLS [RFC4347] (for the data channel) and DTLS- SRTP [RFC5763] for the media channel. Specifically, Alice and Bob perform a DTLS handshake on every channel which has been established by ICE. The total number of channels depends on the amount of muxing; in the most likely case we are using both RTP/RTCP mux and muxing multiple media streams on the same channel, in which case there is only one DTLS handshake. Once the DTLS handshake has completed, the keys are exported [RFC5705] and used to key SRTP for the media channels. At this point, Alice and Bob know that they share a set of secure data and/or media channels with keys which are not known to any third-party attacker. If Alice and Bob authenticated via their IdPs, - then they also know that the signaling service is not attacking them. - Even if they do not use an IdP, as long as they have minimal trust in - the signaling service not to perform a man-in-the-middle attack, they - know that their communications are secure against the signaling - service as well. + then they also know that the signaling service is not mounting a man- + in-the-middle attack on theor traffic. Even if they do not use an + IdP, as long as they have minimal trust in the signaling service not + to perform a man-in-the-middle attack, they know that their + communications are secure against the signaling service as well + (i.e., that the signaling service cannot mount a passive attack on + the communications). 4.4. Communications and Consent Freshness From a security perspective, everything from here on in is a little anticlimactic: Alice and Bob exchange data protected by the keys negotiated by DTLS. Because of the security guarantees discussed in the previous sections, they know that the communications are encrypted and authenticated. The one remaining security property we need to establish is "consent freshness", i.e., allowing Alice to verify that Bob is still prepared - to receive her communications. ICE specifies periodic STUN - keepalizes but only if media is not flowing. Because the consent - issue is more difficult here, we require RTCWeb implementations to - periodically send keepalives. As described in Section 5.3, these - keepalives MUST be based on the consent freshness mechanism specified - in [I-D.muthu-behave-consent-freshness]. If a keepalive fails and no - new ICE channels can be established, then the session is terminated. + to receive her communications so that Alice does not continue to send + large traffic volumes to entities which went abruptly offline. ICE + specifies periodic STUN keepalizes but only if media is not flowing. + Because the consent issue is more difficult here, we require RTCWeb + implementations to periodically send keepalives. As described in + Section 5.3, these keepalives MUST be based on the consent freshness + mechanism specified in [I-D.muthu-behave-consent-freshness]. If a + keepalive fails and no new ICE channels can be established, then the + session is terminated. 5. Detailed Technical Description 5.1. Origin and Web Security Issues The basic unit of permissions for RTCWEB is the origin [RFC6454]. Because the security of the origin depends on being able to authenticate content from that origin, the origin can only be securely established if data is transferred over HTTPS [RFC2818]. Thus, clients MUST treat HTTP and HTTPS origins as different permissions domains. [Note: this follows directly from the origin security model and is stated here merely for clarity.] Many web browsers currently forbid by default any active mixed - content on HTTPS pages. I.e., when JS is loaded from an HTTP origin - onto an HTTPS page, an error is displayed and the content is not - executed unless the user overrides the error. Any browser which - enforces such a policy will also not permit access to RTCWEB - functionality from mixed content pages. It is RECOMMENDED that - browsers which allow active mixed content nevertheless disable RTCWEB - functionality in mixed content settings. [[ OPEN ISSUE: Should this - be a 2119 MUST? It's not clear what set of conditions would make - this OK, other than that browser manufacturers have traditionally - been permissive here here.]] Note that it is possible for a page - which was not mixed content to become mixed content during the - duration of the call. Implementations MAY choose to terminate the - call or display a warning at that point, but it is also permissible - to ignore this condition. This is a deliberate implementation - complexity versus security tradeoff. [[ OPEN ISSUE:: Should we be - more aggressive about this?]] + content on HTTPS pages. That is, when JavaScript is loaded from an + HTTP origin onto an HTTPS page, an error is displayed and the HTTP + content is not executed unless the user overrides the error. Any + browser which enforces such a policy will also not permit access to + RTCWEB functionality from mixed content pages (because they never + display mixed content). It is RECOMMENDED that browsers which allow + active mixed content nevertheless disable RTCWEB functionality in + mixed content settings. [[ OPEN ISSUE: Should this be a 2119 MUST? + It's not clear what set of conditions would make this OK, other than + that browser manufacturers have traditionally been permissive here + here.]] Note that it is possible for a page which was not mixed + content to become mixed content during the duration of the call. + Implementations MAY choose to terminate the call or display a warning + at that point, but it is also permissible to ignore this condition. + The major risk here is that the newly arrived insecure JS might + redirect media to a location controlled by the attacker. This is a + deliberate implementation complexity versus security tradeoff. [[ + OPEN ISSUE:: Should we be more aggressive about this?]] 5.2. Device Permissions Model Implementations MUST obtain explicit user consent prior to providing access to the camera and/or microphone. Implementations MUST at minimum support the following two permissions models for HTTPS origins. o Requests for one-time camera/microphone access. o Requests for permanent access. Because HTTP origins cannot be securely established against network attackers, implementations MUST NOT allow the setting of permanent access permissions for HTTP origins. Implementations MAY also opt to refuse all permissions grants for HTTP origins, but it is RECOMMENDED that currently they support one-time camera/microphone access. - In addition, they SHOULD support requests for access to a single - communicating peer. E.g., "Call customerservice@ford.com". Browsers + In addition, they SHOULD support requests for access that promise + that media from this grant will be sent to a single communicating + peer (obviously there could be other requests for other peers). + E.g., "Call customerservice@ford.com". The semantics of this request + are that the media stream from the camera and microphone will only be + routed through a connection which has been cryptographically verified + (through the IdP mechanism or an X.509 certificate in the DTLS-SRTP + handshake) as being associated with the stated identity. Browsers servicing such requests SHOULD clearly indicate that identity to the - user when asking for permission. + user when asking for permission. The idea behind this type of + permissions is that a user might have a fairly narrow list of peers + he is willing to communicate with, e.g., "my mother" rather than + "anyone on Facebook". Narrow permissions grants allow the browser to + do that enforcement. API Requirement: The API MUST provide a mechanism for the requesting JS to indicate which of these forms of permissions it is requesting. This allows the browser client to know what sort of user interface experience to provide to the user, including what permissions to request from the user and hence what to enforce later. For instance, browsers might display a non-invasive door hanger ("some features of this site may not work..." when asking for long-term permissions) but a more invasive UI ("here is your own video") for single-call permissions. The API MAY grant weaker @@ -503,30 +607,30 @@ API Requirement: The API MUST provide a mechanism for the requesting JS to relinquish the ability to see or modify the media (e.g., via MediaStream.record()). Combined with secure authentication of the communicating peer, this allows a user to be sure that the calling site is not accessing or modifying their conversion. UI Requirement: The UI MUST clearly indicate when the user's camera and microphone are in use. This indication MUST NOT be suppressable by the JS and MUST clearly indicate how to terminate - a call, and provide a UI means to immediately stop camera/ + device access, and provide a UI means to immediately stop camera/ microphone input without the JS being able to prevent it. UI Requirement: If the UI indication of camera/microphone use are displayed in the browser such that minimizing the browser window would hide the indication, or the JS creating an overlapping window would hide the indication, then the browser SHOULD stop - camera and microphone input. [Note: this may not be necessary in - systems that are non-windows-based but that have good - notifications support, such as phones.] + camera and microphone input when the indication is hidden. [Note: + this may not be necessary in systems that are non-windows-based + but that have good notifications support, such as phones.] Clients MAY permit the formation of data channels without any direct user approval. Because sites can always tunnel data through the server, further restrictions on the data channel do not provide any additional security. (though see Section 5.3 for a related issue). Implementations which support some form of direct user authentication SHOULD also provide a policy by which a user can authorize calls only to specific counterparties. Specifically, the implementation SHOULD provide the following interfaces/controls: @@ -717,21 +821,26 @@ Relying Party (RP): The entity which is trying to verify the AP's identity. The AP and the IdP have an account relationship of some kind: the AP registers with the IdP and is able to subsequently authenticate directly to the IdP (e.g., with a password). This means that the browser must somehow know which IdP(s) the user has an account relationship with. This can either be something that the user configures into the browser or that is configured at the calling site - and then provided to the PeerConnection by the calling site. + and then provided to the PeerConnection by the Web application at the + calling site. The use case for having this information configured + into the browser is that the user may "log into" the browser to bind + it to some identity. This is becoming common in new browsers. + However, it should also be possible for the IdP information to simply + be provided by the calling application. At a high level there are two kinds of IdPs: Authoritative: IdPs which have verifiable control of some section of the identity space. For instance, in the realm of e-mail, the operator of "example.com" has complete control of the namespace ending in "@example.com". Thus, "alice@example.com" is whoever the operator says it is. Examples of systems with authoritative identity providers include DNSSEC, RFC 4474, and Facebook Connect (Facebook identities only make sense within the context of the @@ -741,25 +850,28 @@ identity space but instead verify user's identities via some unspecified mechanism and then attest to it. Because the IdP doesn't actually control the namespace, RPs need to trust that the IdP is correctly verifying AP identities, and there can potentially be multiple IdPs attesting to the same section of the identity space. Probably the best-known example of a third-party identity provider is SSL certificates, where there are a large number of CAs all of whom can attest to any domain name. If an AP is authenticating via an authoritative IdP, then the RP does - not need to explicitly trust the IdP at all: as long as the RP knows - how to verify that the IdP indeed made the relevant identity - assertion (a function provided by the mechanisms in this document), - then any assertion it makes about an identity for which it is - authoritative is directly verifiable. + not need to explicitly configure trust in the IdP at all. The + identity mechanism can directly verify that the IdP indeed made the + relevant identity assertion (a function provided by the mechanisms in + this document), and any assertion it makes about an identity for + which it is authoritative is directly verifiable. Note that this + does not mean that the IdP might not lie, but that is a + trustworthiness judgement that the user can make at the time he looks + at the identity. By contrast, if an AP is authenticating via a third-party IdP, the RP needs to explicitly trust that IdP (hence the need for an explicit trust anchor list in PKI-based SSL/TLS clients). The list of trustable IdPs needs to be configured directly into the browser, either by the user or potentially by the browser manufacturer. This is a significant advantage of authoritative IdPs and implies that if third-party IdPs are to be supported, the potential number needs to be fairly small. @@ -767,20 +879,29 @@ In order to provide security without trusting the calling site, the PeerConnection component of the browser must interact directly with the IdP. The details of the mechanism are described in the W3C API specification, but the general idea is that the PeerConnection component downloads JS from a specific location on the IdP dictated by the IdP domain name. That JS (the "IdP proxy") runs in an isolated security context within the browser and the PeerConnection talks to it via a secure message passing channel. + Note that there are two logically separate functions here: + + o Identity assertion generation. + o Identity assertion verification. + + The same IdP JS "endpoint" is used for both functions but of course a + given IdP might behave differently and load new JS to perform one + function or the other. + +------------------------------------+ | https://calling-site.example.com | | | | | | | | Calling JS Code | | ^ | | | API Calls | | v | | PeerConnection | @@ -862,24 +983,24 @@ The "algorithm" and digest values correspond directly to the algorithm and digest in the a=fingerprint line of the SDP. Note: this structure does not need to be interpreted by the IdP or the IdP proxy. It is consumed solely by the RP's browser. The IdP merely treats it as an opaque value to be attested to. Thus, new parameters can be added to the assertion without modifying the IdP. 5.6.4.2. Carrying Identity Assertions - Once an IdP has generated an assertion, the JSEP message. This is - done by adding a new a-line to the SDP, of the form a=identity. The - sole contents of this value are a base-64-encoded version of the - identity assertion. For example: + Once an IdP has generated an assertion, it is attached to the JSEP + message. This is done by adding a new a-line to the SDP, of the form + a=identity. The sole contents of this value are a base-64-encoded + version of the identity assertion. For example: v=0 o=- 1181923068 1181923196 IN IP4 ua1.example.com s=example1 c=IN IP4 ua1.example.com a=setup:actpass a=fingerprint: SHA-1 \ 4A:AD:B9:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB a=identity: \ ImlkcCI6eyJkb21haW4iOiAiZXhhbXBsZS5vcmciLCAicHJvdG9jb2wiOiAiYm9n \ @@ -888,23 +1009,27 @@ XCJzaWduYXR1cmVcIjpcIjAxMDIwMzA0MDUwNlwifSJ9Cg== t=0 0 m=audio 6056 RTP/AVP 0 a=sendrecv a=tcap:1 UDP/TLS/RTP/SAVP RTP/AVP a=pcfg:1 t=1 Each identity attribute should be paired (and attests to) with an a=fingerprint attribute and therefore can exist either at the session or media level. Multiple identity attributes may appear at either - level, though implementations are discouraged from doing this unless - they have a clear idea of what security claim they intend to be - making. + level, though it is RECOMMENDED that implementations not do this, + because it becomes very unclear what security claim that they are + making and the UI guidelines above become unclear. Browsers MAY + choose refuse to display any identity indicators in the face of + multiple identity attributes with different identities but SHOULD + process multiple attributes with the same identity as described + above. 5.6.5. IdP Interaction Details 5.6.5.1. General Message Structure Messages between the PeerConnection object and the IdP proxy are formatted using JSON [RFC4627]. For instance, the PeerConnection would request a signature with the following "SIGN" message: { @@ -915,21 +1040,21 @@ } All messages MUST contain a "type" field which indicates the general meaning of the message. All requests from the PeerConnection object MUST contain an "id" field which MUST be unique for that PeerConnection object. Any responses from the IdP proxy MUST contain the same id in response, which allows the PeerConnection to correlate requests and responses. - All requests from the PeerConnection object MUST contain a "origin" + All requests from the PeerConnection object MUST contain an "origin" field containing the origin of the JS which initiated the PC (i.e., the URL of the calling site). This origin value can be used by the IdP to make access control decisions. For instance, an IdP might only issue identity assertions for certain calling services in the same way that some IdPs require that relying Web sites have an API key before learning user identity. Any message-specific data is carried in a "message" field. Depending on the message type, this may either be a string or a richer JSON object. @@ -938,21 +1063,21 @@ If an error occurs, the IdP sends a message of type "ERROR". The message MAY have an "error" field containing freeform text data which containing additional information about what happened. For instance: { "type":"ERROR", "error":"Signature verification failed" } - Figure 3: Example error + Figure 5: Example error 5.6.5.2. IdP Proxy Setup In order to perform an identity transaction, the PeerConnection must first create an IdP proxy. While the details of this are specified in the W3C API document, from the perspective of this specification, however, the relevant facts are: o The JS runs in the IdP's security context with the base page retrieved from the URL specified in Section 5.6.5.2.1 @@ -971,20 +1096,24 @@ { "type":"READY" } [[ OPEN ISSUE: if the W3C half of this converges on WebIntents, then the READY message will not be necessary.]] Once the PeerConnection object receives the ready message, it can send commands to the IdP proxy. 5.6.5.2.1. Determining the IdP URI + In order to ensure that the IdP is under control of the domain owner + rather than someone who merely has an account on the domain owner's + server (e.g., in shared hosting scenarios), the IdP JavaScript is + hosted at a deterministic location based on the IdP's domain name. Each IdP proxy instance is associated with two values: domain name: The IdP's domain name protocol: The specific IdP protocol which the IdP is using. This is a completely IdP-specific string, but allows an IdP to implement two protocols in parallel. This value may be the empty string. Each IdP MUST serve its initial entry page (i.e., the one loaded by the IdP proxy) from the well-known URI specified in "/.well-known/ idp-proxy/" on the IdP's web site. This URI MUST be loaded @@ -1024,21 +1153,21 @@ defined above, but are opaque from the perspective of the IdP. A successful response to a "SIGN" message contains a message field which is a JS dictionary dictionary consisting of two fields: idp: A dictionary containing the domain name of the provider and the protocol string assertion: An opaque field containing the assertion itself. This is only interpretable by the idp or its proxy. - Figure 4 shows an example transaction, with the message "abcde..." + Figure 6 shows an example transaction, with the message "abcde..." (remember, the messages are opaque at this layer) being signed and bound to identity "ekr@example.org". In this case, the message has presumably been digitally signed/MACed in some way that the IdP can later verify it, but this is an implementation detail and out of scope of this document. Line breaks are inserted solely for readability. PeerConnection -> IdP proxy: { "type":"SIGN", @@ -1056,21 +1185,21 @@ "domain": "example.org" "protocol": "bogus" }, "assertion":\"{\"identity\":\"bob@example.org\", \"contents\":\"abcdefghijklmnopqrstuvwyz\", \"request_origin\":\"rtcweb://peerconnection\", \"signature\":\"010203040506\"}" } } - Figure 4: Example assertion request + Figure 6: Example assertion request 5.6.5.2.3. Verifying Assertions In order to verify an assertion, an RP sends a "VERIFY" message to the IdP proxy containing the assertion supplied by the AP in the "message" field. The IdP proxy verifies the assertion. Depending on the identity protocol, this may require one or more round trips to the IdP. For instance, an OAuth-based protocol will likely require using the IdP @@ -1088,21 +1217,21 @@ original SIGN request. request_origin The original origin of the SIGN request on the AP side as determined by the origin of the PostMessage call. The IdP MUST somehow arrange to propagate this information as part of the assertion. The receiving PeerConnection MUST verify that this value is "rtcweb://peerconnection" (which implies that PeerConnection must arrange that its messages to the IdP proxy are from this origin.) [[ OPEN ISSUE: Can a URI person help make a better URI.]] - Figure 5 shows an example transaction. Line breaks are inserted + Figure 7 shows an example transaction. Line breaks are inserted solely for readability. PeerConnection -> IdP Proxy: { "type":"VERIFY", "id":2, "origin":"https://calling-service.example.com/", "message":\"{\"identity\":\"bob@example.org\", \"contents\":\"abcdefghijklmnopqrstuvwyz\", \"request_origin\":\"rtcweb://peerconnection\", @@ -1116,21 +1245,21 @@ "message": { "identity" : { "name" : "bob@example.org", "displayname" : "Bob" }, "request_origin":"rtcweb://peerconnection", "contents":"abcdefghijklmnopqrstuvwyz" } } - Figure 5: Example verification request + Figure 7: Example verification request 5.6.5.2.3.1. Identity Formats Identities passed from the IdP proxy to the PeerConnection are structured as JSON dictionaries with one mandatory field: "name". This field MUST consist of an RFC822-formatted string representing the user's identity. [[ OPEN ISSUE: Would it be better to have a typed field? ]] The PeerConnection API MUST check this string as follows: @@ -1190,25 +1319,43 @@ suitable for general use. Note, however, that a malicious signaling service can strip off any such identity assertions, though it cannot forge new ones. Note that all of the third-party security mechanisms available (whether X.509 certificates or a third-party IdP) rely on the security of the third party--this is of course also true of your connection to the Web site itself. Users who wish to assure themselves of security against a malicious identity provider can only do so by verifying peer credentials directly, e.g., by checking the peer's fingerprint against a value delivered out of band. + In order to protect against malicious content JavaScript, that + JavaScript MUST NOT be allowed to have direct access to---or perform + computations with---DTLS keys. For instance, if content JS were able + to compute digital signatures, then it would be possible for content + JS to get an identity assertion for a browser's generated key and + then use that assertion plus a signature by the key to authenticate a + call protected under an ephemeral DH key controlled by the content + JS, thus violating the security guarantees otherwise provided by the + IdP mechanism. Note that it is not sufficient merely to deny the + content JS direct access to the keys, as some have suggested doing + with the WebCrypto API. The JS must also not be allowed to perform + operations that would be valid for a DTLS endpoint. By far the + safest approach is simply to deny the ability to perform any + operations that depend on secret information associated with the key. + Operations that depend on public information, such as exporting the + public key are of course safe. + 5.7.2. Privacy The requirements in this document are intended to allow: o Users to participate in calls without revealing their location. + o Potential callees to avoid revealing their location and even presence status prior to agreeing to answer a call. However, these privacy protections come at a performance cost in terms of using TURN relays and, in the latter case, delaying ICE. Sites SHOULD make users aware of these tradeoffs. Note that the protections provided here assume a non-malicious calling service. As the calling service always knows the users status and (absent the use of a technology like Tor) their IP @@ -1281,29 +1428,45 @@ to get assertions tied to a user or to produce assertions that RPs will accept. 5.7.4.1. PeerConnection Origin Check Fundamentally, the IdP proxy is just a piece of HTML and JS loaded by the browser, so nothing stops a Web attacker o from creating their own IFRAME, loading the IdP proxy HTML/JS, and requesting a signature. In order to prevent this attack, we require that all signatures be tied to a specific origin ("rtcweb://...") which cannot - be produced by a page tied to a Web attacker. Thus, while an - attacker can instantiate the IdP proxy, they cannot send messages - from an appropriate origin and so cannot create acceptable - assertions. This origin check is enforced on the relying party side, - not on the authenticating party side. The reason for this is to take - the burden of knowing which origins are valid off of the IdP, thus - making this mechanism extensible to other applications besides - RTCWEB. The IdP simply needs to gather the origin information (from - the posted message) and attach it to the assertion. + be produced by content JavaScript. Thus, while an attacker can + instantiate the IdP proxy, they cannot send messages from an + appropriate origin and so cannot create acceptable assertions. I.e., + the assertion request must have come from the browser. This origin + check is enforced on the relying party side, not on the + authenticating party side. The reason for this is to take the burden + of knowing which origins are valid off of the IdP, thus making this + mechanism extensible to other applications besides RTCWEB. The IdP + simply needs to gather the origin information (from the posted + message) and attach it to the assertion. + + Note that although this origin check is enforced on the RP side and + not at the IdP, it is absolutely imperative that it be done. The + mechanisms in this document rely on the browser enforcing access + restrictions on the DTLS keys and assertion requests which do not + come with the right origin may be from content JS rather than from + browsers, and therefore those access restrcitions cannot be assumed. + + Note that this check only asserts that the browser (or some other + entity with access to the user's authentication data) attests to the + request and hence to the fingerprint. It does not demonstrate that + the browser has access to the associated private key. However, + attaching one's identity to a key that the user does not control does + not appear to provide substantial leverage to an attacker, so a proof + of possession is omitted for simplicity. 5.7.4.2. IdP Well-known URI As described in Section 5.6.5.2.1 the IdP proxy HTML/JS landing page is located at a well-known URI based on the IdP's domain name. This requirement prevents an attacker who can write some resources at the IdP (e.g., on one's Facebook wall) from being able to impersonate the IdP. 5.7.4.3. Privacy of IdP-generated identities and the hosting site @@ -1362,61 +1525,71 @@ Some browsers allow users to block third party cookies (cookies associated with origins other than the top level page) for privacy reasons. Any IdP which uses cookies to persist logins will be broken by third-party cookie blocking. One option is to accept this as a limitation; another is to have the PeerConnection object disable third-party cookie blocking for the IdP proxy. 6. Acknowledgements - Bernard Aboba, Harald Alvestrand, Dan Druta, Cullen Jennings, Hadriel - Kaplan, Matthew Kaufman, Jim McEachern, Martin Thomson, Magnus - Westerland. + Bernard Aboba, Harald Alvestrand, Richard Barnes, Dan Druta, Cullen + Jennings, Hadriel Kaplan, Matthew Kaufman, Jim McEachern, Martin + Thomson, Magnus Westerland. -7. Changes since -03 +7. Changes +7.1. Changes since -05 + + The following changes have been made since the -05 draft. + + o Response to comments from Richard Barnes + o More explanation of the IdP security properties and the federation + use case. + o Editorial cleanup. + +7.2. Changes since -03 Version -04 was a version control mistake. Please ignore. The following changes have been made since the -04 draft. o Move origin check from IdP to RP per discussion in YVR. o Clarified treatment of X.509-level identities. o Editorial cleanup. -8. Changes since -03 +7.3. Changes since -03 -9. Changes since -02 +7.4. Changes since -02 The following changes have been made since the -02 draft. o Forbid persistent HTTP permissions. o Clarified the text in S 5.4 to clearly refer to requirements on the API to provide functionality to the site. o Fold in the IETF portion of draft-rescorla-rtcweb-generic-idp o Retarget the continuing consent section to assume Binding Requests o Editorial improvements -10. References +8. References -10.1. Normative References +8.1. Normative References [I-D.ietf-rtcweb-security] Rescorla, E., "Security Considerations for RTC-Web", draft-ietf-rtcweb-security-03 (work in progress), June 2012. [I-D.muthu-behave-consent-freshness] - Perumal, M., Wing, D., and H. Kaplan, "STUN Usage for - Consent Freshness and Session Liveness", - draft-muthu-behave-consent-freshness-01 (work in - progress), July 2012. + Perumal, M., Wing, D., R, R., and H. Kaplan, "STUN Usage + for Consent Freshness", + draft-muthu-behave-consent-freshness-02 (work in + progress), January 2013. [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, March 1997. [RFC2818] Rescorla, E., "HTTP Over TLS", RFC 2818, May 2000. [RFC4347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer Security", RFC 4347, April 2006. [RFC4627] Crockford, D., "The application/json Media Type for @@ -1432,26 +1605,26 @@ (SRTP) Security Context Using Datagram Transport Layer Security (DTLS)", RFC 5763, May 2010. [RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer Security (DTLS) Extension to Establish Keys for the Secure Real-time Transport Protocol (SRTP)", RFC 5764, May 2010. [RFC6454] Barth, A., "The Web Origin Concept", RFC 6454, December 2011. -10.2. Informative References +8.2. Informative References [I-D.ietf-rtcweb-jsep] Uberti, J. and C. Jennings, "Javascript Session - Establishment Protocol", draft-ietf-rtcweb-jsep-01 (work - in progress), June 2012. + Establishment Protocol", draft-ietf-rtcweb-jsep-02 (work + in progress), October 2012. [I-D.jennings-rtcweb-signaling] Jennings, C., Rosenberg, J., and R. Jesup, "RTCWeb Offer/ Answer Protocol (ROAP)", draft-jennings-rtcweb-signaling-01 (work in progress), October 2011. [I-D.kaufman-rtcweb-security-ui] Kaufman, M., "Client Security User Interface Requirements for RTCWEB", draft-kaufman-rtcweb-security-ui-00 (work in