RTCWEB E. Rescorla Internet-Draft RTFM, Inc. Intended status: Standards Track January 22, 2012 Expires: July 25, 2012 RTCWEB Security Architecture draft-ietf-rtcweb-security-arch-00 Abstract 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 soft phones) RTCWEB communications are directly controlled by some Web server, which poses new security challenges. For instance, a Web browser might expose a JavaScript API which allows a server to place a video call. Unrestricted access to such an API would allow any site which a user visited to "bug" a user's computer, capturing any activity which passed in front of their camera. [I-D.ietf- rtcweb-security] defines the RTCWEB threat model. This document defines an architecture which provides security within that threat model. Legal THIS DOCUMENT AND THE INFORMATION CONTAINED THEREIN ARE PROVIDED ON AN "AS IS" BASIS AND THE CONTRIBUTOR, THE ORGANIZATION HE/SHE REPRESENTS OR IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY, THE IETF TRUST, AND THE INTERNET ENGINEERING TASK FORCE, DISCLAIM ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION THEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE. Status of this Memo This Internet-Draft is submitted in full conformance with the provisions of BCP 78 and BCP 79. 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 Rescorla Expires July 25, 2012 [Page 1] Internet-Draft RTCWEB Sec. Arch. January 2012 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 July 25, 2012. Copyright Notice Copyright (c) 2012 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 described in the Simplified BSD License. This document may contain material from IETF Documents or IETF Contributions published or made publicly available before November 10, 2008. The person(s) controlling the copyright in some of this material may not have granted the IETF Trust the right to allow modifications of such material outside the IETF Standards Process. 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. Rescorla Expires July 25, 2012 [Page 2] Internet-Draft RTCWEB Sec. Arch. January 2012 Table of Contents 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4 3. Trust Model . . . . . . . . . . . . . . . . . . . . . . . . . 4 3.1. Authenticated Entities . . . . . . . . . . . . . . . . . . 5 3.2. Unauthenticated Entities . . . . . . . . . . . . . . . . . 5 4. Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . 6 4.1. Initial Signaling . . . . . . . . . . . . . . . . . . . . 7 4.2. Media Consent Verification . . . . . . . . . . . . . . . . 9 4.3. DTLS Handshake . . . . . . . . . . . . . . . . . . . . . . 10 4.4. Communications and Consent Freshness . . . . . . . . . . . 10 5. Detailed Technical Description . . . . . . . . . . . . . . . . 10 5.1. Origin and Web Security Issues . . . . . . . . . . . . . . 10 5.2. Device Permissions Model . . . . . . . . . . . . . . . . . 11 5.3. Communications Consent . . . . . . . . . . . . . . . . . . 12 5.4. IP Location Privacy . . . . . . . . . . . . . . . . . . . 13 5.5. Communications Security . . . . . . . . . . . . . . . . . 13 5.6. Web-Based Peer Authentication . . . . . . . . . . . . . . 15 6. Security Considerations . . . . . . . . . . . . . . . . . . . 16 6.1. Communications Security . . . . . . . . . . . . . . . . . 16 6.2. Privacy . . . . . . . . . . . . . . . . . . . . . . . . . 16 6.3. Denial of Service . . . . . . . . . . . . . . . . . . . . 17 7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 17 8. References . . . . . . . . . . . . . . . . . . . . . . . . . . 18 8.1. Normative References . . . . . . . . . . . . . . . . . . . 18 8.2. Informative References . . . . . . . . . . . . . . . . . . 18 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 19 Rescorla Expires July 25, 2012 [Page 3] Internet-Draft RTCWEB Sec. Arch. January 2012 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. +----------------+ | | | Web Server | | | +----------------+ ^ ^ / \ HTTP / \ HTTP / \ / \ v v JS API JS API +-----------+ +-----------+ | | Media | | | Browser |<---------->| Browser | | | | | +-----------+ +-----------+ Figure 1: A simple 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]. 3. Trust Model The basic assumption of this architecture is that network resources exist in a hierarchy of trust, rooted in the browser, which serves as the user's TRUSTED COMPUTING BASE (TCB). Any security property which Rescorla Expires July 25, 2012 [Page 4] Internet-Draft RTCWEB Sec. Arch. January 2012 the user wishes to have enforced must be ultimately guaranteed by the browser (or transitively by some property the browser verifies). Conversely, if the browser is compromised, then no security guarantees are possible. Note that there are cases (e.g., Internet kiosks) where the user can't really trust the browser that much. In these cases, the level of security provided is limited by how much they trust the browser. Optimally, we would not rely on trust in any entities other than the browser. However, this is unfortunately not possible if we wish to have a functional system. Other network elements fall into two categories: those which can be authenticated by the browser and thus are partly trusted--though to the minimum extent necessary--and those which cannot be authenticated and thus are untrusted. This is a natural extension of the end-to-end principle. 3.1. Authenticated Entities There are two major classes of authenticated entities in the system: o Calling services: Web sites whose origin we can verify (optimally via HTTPS). o Other users: RTCWEB peers whose origin we can verify cryptographically (optimally via DTLS-SRTP). Note that merely being authenticated does not make these entities trusted. For instance, just because we can verify that https://www.evil.org/ is owned by Dr. Evil does not mean that we can trust Dr. Evil to access our camera an microphone. However, it gives the user an opportunity to determine whether he wishes to trust Dr. Evil or not; after all, if he desires to contact Dr. Evil (perhaps to arrange for ransom payment), it's safe to temporarily give him access to the camera and microphone for the purpose of the call, but he doesn't want Dr. Evil to be able to access his camera and microphone other than during the call. The point here is that we must first identify other elements before we can determine whether and how much to trust them. It's also worth noting that there are settings where authentication is non-cryptographic, such as other machines behind a firewall. Naturally, the level of trust one can have in identities verified in this way depends on how strong the topology enforcement is. 3.2. Unauthenticated Entities Other than the above entities, we are not generally able to identify other network elements, thus we cannot trust them. This does not mean that it is not possible to have any interaction with them, but Rescorla Expires July 25, 2012 [Page 5] Internet-Draft RTCWEB Sec. Arch. January 2012 it means that we must assume that they will behave maliciously and design a system which is secure even if they do so. 4. Overview This section describes a typical RTCWeb session and shows how the various security elements interact and what guarantees are provided to the user. The example in this section is a "best case" scenario in which we provide the maximal amount of user authentication and 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 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. Rescorla Expires July 25, 2012 [Page 6] Internet-Draft RTCWEB Sec. Arch. January 2012 +----------------+ | | | Signaling | | Server | | | +----------------+ ^ ^ / \ HTTPS / \ HTTPS / \ / \ v v JS API JS API +-----------+ +-----------+ | | Media | | Alice | Browser |<---------->| Browser | Bob | | (DTLS-SRTP)| | +-----------+ +-----------+ ^ ^--+ +--^ ^ | | | | v | | v +-----------+ | | +-----------+ | |<--------+ | | | IdP | | | IdP | | | +------->| | +-----------+ +-----------+ Figure 2: A 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. 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 Rescorla Expires July 25, 2012 [Page 7] Internet-Draft RTCWEB Sec. Arch. January 2012 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 set up some media. Because this is an audio/video call, it creates two MediaStreams, one connected to an audio input and one connected 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 The format of this data is currently undefined. It may be a complete message as defined by ROAP [I-D.jennings-rtcweb-signaling] or separate media description and transport messages as defined in JSEP [REF] or may be assembled piecemeal by the JS. In either case, it will contain: o Media channel information o ICE candidates o A fingerprint attribute binding the communication to Alice's public key [RFC5763] [Note that the above implies that this information should appear in JSEP's transport-level messages.] 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 [I-D.rescorla-rtcweb-generic-idp] 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 [I-D.ietf-hybi-thewebsocketprotocol]. 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 Rescorla Expires July 25, 2012 [Page 8] Internet-Draft RTCWEB Sec. Arch. January 2012 way so the browser has no specific knowledge of the IdP) to verity 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. 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. 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. 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. Rescorla Expires July 25, 2012 [Page 9] Internet-Draft RTCWEB Sec. Arch. January 2012 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 extracted 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. 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. 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 [I-D.abarth-origin]. 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 Rescorla Expires July 25, 2012 [Page 10] Internet-Draft RTCWEB Sec. Arch. January 2012 [RFC2818]. Thus, clients MUST treat HTTP and HTTPS origins as different permissions domains and SHOULD NOT permit access to any RTCWEB functionality from scripts fetched over non-secure (HTTP) origins. If an HTTPS origin contains mixed active content (regardless of whether it is present on the specific page attempting to access RTCWEB functionality), any access MUST be treated as if it came from the HTTP origin. For instance, if a https://www.example.com/example.html loads https://www.example.com/example.js and http://www.example.org/jquery.js, any attempt by example.js to access RTCWeb functionality MUST be treated as if it came from http://www.example.com/. Note that many browsers already track mixed content and either forbid it by default or display a warning. [[ OPEN ISSUE: This seems to be wrong, but I'm not sure what's right yet. ]] 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: o Requests for one-time camera/microphone access. o Requests for permanent access. In addition, they SHOULD support requests for access to a single communicating peer. E.g., "Call customerservice@ford.com". Browsers servicing such requests SHOULD clearly indicate that identity to the user when asking for permission. 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 client to know what sort of user interface experience to provide. In particular, 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 permissions than the JS asked for if the user chooses to authorize only those permissions, but if it intends to grant stronger ones it SHOULD display the appropriate UI for those permissions and MUST clearly indicate what permissions are being requested. 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. Rescorla Expires July 25, 2012 [Page 11] Internet-Draft RTCWEB Sec. Arch. January 2012 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/ 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.] 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: o Allow future calls to this verified user. o Allow future calls to any verified user who is in my system address book (this only works with address book integration, of course). Implementations SHOULD also provide a different user interface indication when calls are in progress to users whose identities are directly verifiable. Section 5.5 provides more on this. 5.3. Communications Consent Browser client implementations of RTCWEB MUST implement ICE. Server gateway implementations which operate only at public IP addresses may implement ICE-Lite. Browser implementations MUST verify reachability via ICE prior to sending any non-ICE packets to a given destination. Implementations MUST NOT provide the ICE transaction ID to JavaScript. [Note: this document takes no position on the split between ICE in JS and ICE in the browser. The above text is written the way it is for editorial convenience and will be modified appropriately if the WG decides on ICE in the JS.] Implementations MUST send keepalives no less frequently than every 30 Rescorla Expires July 25, 2012 [Page 12] Internet-Draft RTCWEB Sec. Arch. January 2012 seconds regardless of whether traffic is flowing or not. If a keepalive fails then the implementation MUST either attempt to find a new valid path via ICE or terminate media for that ICE component. Note that ICE [RFC5245]; Section 10 keepalives use STUN Binding Indications which are one-way and therefore not sufficient. We will need to define a new mechanism for this. [PROPOSED SOLUTION: Replace STUN Binding Indications with STUN Binding Requests and require that a failed transaction causes the results above.] 5.4. IP Location Privacy A side effect of the default ICE behavior is that the peer learns one's IP address, which leaks large amounts of location information, especially for mobile devices. This has negative privacy consequences in some circumstances. The following two API requirements are intended to mitigate this issue: API Requirement: The API MUST provide a mechanism to suppress ICE negotiation (though perhaps to allow candidate gathering) until the user has decided to answer the call [note: determining when the call has been answered is a question for the JS.] This enables a user to prevent a peer from learning their IP address if they elect not to answer a call and also from learning whether the user is online. API Requirement: The API MUST provide a mechanism for the calling application to indicate that only TURN candidates are to be used. This prevents the peer from learning one's IP address at all. The API MUST provide a mechanism for the calling application to reconfigure an existing call to add non-TURN candidates. Taken together, these requirements allow ICE negotiation to start immediately on incoming call notification, thus reducing post-dial delay, but also to avoid disclosing the user's IP address until they have decided to answer. 5.5. Communications Security Implementations MUST implement DTLS and DTLS-SRTP. All data channels MUST be secured via DTLS. DTLS-SRTP MUST be offered for every media channel and MUST be the default; i.e., if an implementation receives an offer for DTLS-SRTP and SDES and/or plain RTP, DTLS-SRTP MUST be selected. [OPEN ISSUE: What should the settings be here? MUST?] Implementations MAY support SDES and RTP for media traffic for backward compatibility purposes. Rescorla Expires July 25, 2012 [Page 13] Internet-Draft RTCWEB Sec. Arch. January 2012 API Requirement: The API MUST provide a mechanism to indicate that a fresh DTLS key pair is to be generated for a specific call. This is intended to allow for unlinkability. Note that there are also settings where it is attractive to use the same keying material repeatedly, especially those with key continuity-based authentication. API Requirement: The API MUST provide a mechanism to indicate that a fresh DTLS key pair is to be generated for a specific call. This is intended to allow for unlinkability. API Requirement: When DTLS-SRTP is used, the API MUST NOT permit the JS to obtain the negotiated keying material. This requirement preserves the end-to-end security of the media. UI Requirements: A user-oriented client MUST provide an "inspector" interface which allows the user to determine the security characteristics of the media. [largely derived from [I-D.kaufman-rtcweb-security-ui] The following properties SHOULD be displayed "up-front" in the browser chrome, i.e., without requiring the user to ask for them: * A client MUST provide a user interface through which a user may determine the security characteristics for currently-displayed audio and video stream(s) * A client MUST provide a user interface through which a user may determine the security characteristics for transmissions of their microphone audio and camera video. * The "security characteristics" MUST include an indication as to whether or not the transmission is cryptographically protected and whether that protection is based on a key that was delivered out-of-band (from a server) or was generated as a result of a pairwise negotiation. * If the far endpoint was directly verified (see Section 5.6) the "security characteristics" MUST include the verified information. The following properties are more likely to require some "drill- down" from the user: * If the transmission is cryptographically protected, the The algorithms in use (For example: "AES-CBC" or "Null Cipher".) * If the transmission is cryptographically protected, the "security characteristics" MUST indicate whether PFS is provided. * If the transmission is cryptographically protected via an end- to-end mechanism the "security characteristics" MUST include some mechanism to allow an out-of-band verification of the peer, such as a certificate fingerprint or an SAS. Rescorla Expires July 25, 2012 [Page 14] Internet-Draft RTCWEB Sec. Arch. January 2012 5.6. Web-Based Peer Authentication In a number of cases, it is desirable for the endpoint (i.e., the browser) to be able to directly identity the endpoint on the other side without trusting only the signaling service to which they are connected. For instance, users may be making a call via a federated system where they wish to get direct authentication of the other side. Alternately, they may be making a call on a site which they minimally trust (such as a poker site) but to someone who has an identity on a site they do trust (such as a social network.) Recently, a number of Web-based identity technologies (OAuth, BrowserID, Facebook Connect), etc. have been developed. While the details vary, what these technologies share is that they have a Web- based (i.e., HTTP/HTTPS identity provider) which attests to your identity. For instance, if I have an account at example.org, I could use the example.org identity provider to prove to others that I was alice@example.org. The development of these technologies allows us to separate calling from identity provision: I could call you on Poker Galaxy but identify myself as alice@example.org. Whatever the underlying technology, the general principle is that the party which is being authenticated is NOT the signaling site but rather the user (and their browser). Similarly, the relying party is the browser and not the signaling site. Thus, the browser MUST securely generate the input to the IdP assertion process and MUST securely display the results of the verification process to the user in a way which cannot be imitated by the calling site. In order to make this work, we must standardize the following items: o The precise information from the signaling message that must be cryptographically bound to the user's identity. At minimum this MUST be the fingerprint, but we may choose to add other information as the signaling protocol firms up. This will be defined in a future version of this document. o The interface to the IdP. [I-D.rescorla-rtcweb-generic-idp] specifies a specific protocol mechanism which allows the use of any identity protocol without requiring specific further protocol support in the browser. o The JavaScript interfaces which the calling application can use to specify the IdP to use to generate assertions and to discover what assertions were received. These interfaces should be defined in the W3C document. Rescorla Expires July 25, 2012 [Page 15] Internet-Draft RTCWEB Sec. Arch. January 2012 6. Security Considerations Much of the security analysis of this problem is contained in [I-D.ietf-rtcweb-security] or in the discussion of the particular issues above. In order to avoid repetition, this section focuses on (a) residual threats that are not addressed by this document and (b) threats produced by failure/misbehavior of one of the components in the system. 6.1. Communications Security While this document favors DTLS-SRTP, it permits a variety of communications security mechanisms and thus the level of communications security actually provided varies considerably. Any pair of implementations which have multiple security mechanisms in common are subject to being downgraded to the weakest of those common mechanisms by any attacker who can modify the signaling traffic. If communications are over HTTP, this means any on-path attacker. If communications are over HTTPS, this means the signaling server. Implementations which wish to avoid downgrade attack should only offer the strongest available mechanism, which is DTLS/DTLS-SRTP. Note that the implication of this choice will be that interop to non- DTLS-SRTP devices will need to happen through gateways. Even if only DTLS/DTLS-SRTP are used, the signaling server can potentially mount a man-in-the-middle attack unless implementations have some mechanism for independently verifying keys. The UI requirements in Section 5.5 are designed to provide such a mechanism for motivated/security conscious users, but are not suitable for general use. The identity service mechanisms in Section 5.6 are more suitable for general use. Note, however, that a malicious signaling service can strip off any such identity assertions, though it cannot forge new ones. 6.2. 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 Rescorla Expires July 25, 2012 [Page 16] Internet-Draft RTCWEB Sec. Arch. January 2012 status and (absent the use of a technology like Tor) their IP address, they can violate the users privacy at will. Users who wish privacy against the calling sites they are using must use separate privacy enhancing technologies such as Tor. 6.3. Denial of Service The consent mechanisms described in this document are intended to mitigate denial of service attacks in which an attacker uses clients to send large amounts of traffic to a victim without the consent of the victim. While these mechanisms are sufficient to protect victims who have not implemented RTCWEB at all, RTCWEB implementations need to be more careful. Consider the case of a call center which accepts calls via RTCWeb. An attacker proxies the call center's front-end and arranges for multiple clients to initiate calls to the call center. Note that this requires user consent in many cases but because the data channel does not need consent, he can use that directly. Since ICE will complete, browsers can then be induced to send large amounts of data to the victim call center if it supports the data channel at all. Preventing this attack requires that automated RTCWEB implemementations implement sensible flow control and have the ability to triage out (i.e., stop responding to ICE probes on) calls which are behaving badly, and especially to be prepared to remotely throttle the data channel in the absence of plausible audio and video (which the attacker cannot control). Another related attack is for the signaling service to swap the ICE candidates for the audio and video streams, thus forcing a browser to send video to the sink that the other victim expects will contain audio (perhaps it is only expecting audio!) potentially causing overload. Muxing multiple media flows over a single transport makes it harder to individually suppress a single flow by denying ICE keepalives. Media-level (RTCP) mechanisms must be used in this case. Note that attacks based on confusing one end or the other about consent are possible primarily even in the face of the third-party identity mechanism as long as major parts of the signaling messages are not signed. On the other hand, signing the entire message severely restricts the capabilities of the calling application, so there are difficult tradeoffs here. 7. Acknowledgements Bernard Aboba, Harald Alvestrand, Cullen Jennings, Hadriel Kaplan, Matthew Kaufman, Magnus Westerland. Rescorla Expires July 25, 2012 [Page 17] Internet-Draft RTCWEB Sec. Arch. January 2012 8. References 8.1. Normative References [I-D.abarth-origin] Barth, A., "The Web Origin Concept", draft-abarth-origin-09 (work in progress), November 2010. [I-D.ietf-rtcweb-security] Rescorla, E., "Security Considerations for RTC-Web", draft-ietf-rtcweb-security-01 (work in progress), October 2011. [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. 8.2. Informative References [I-D.ietf-hybi-thewebsocketprotocol] Fette, I. and A. Melnikov, "The WebSocket protocol", draft-ietf-hybi-thewebsocketprotocol-17 (work in progress), September 2011. [I-D.jennings-rtcweb-signaling] Jennings, C., Rosenberg, J., Uberti, 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 progress), June 2011. [I-D.rescorla-rtcweb-generic-idp] Rescorla, E., "RTCWeb Generic Identity Provider Interface", draft-rescorla-rtcweb-generic-idp-00 (work in progress), January 2012. [RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A., Peterson, J., Sparks, R., Handley, M., and E. Schooler, "SIP: Session Initiation Protocol", RFC 3261, June 2002. [RFC4347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer Security", RFC 4347, April 2006. Rescorla Expires July 25, 2012 [Page 18] Internet-Draft RTCWEB Sec. Arch. January 2012 [RFC5245] Rosenberg, J., "Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer Protocols", RFC 5245, April 2010. [RFC5763] Fischl, J., Tschofenig, H., and E. Rescorla, "Framework for Establishing a Secure Real-time Transport Protocol (SRTP) Security Context Using Datagram Transport Layer Security (DTLS)", RFC 5763, May 2010. [XmlHttpRequest] van Kesteren, A., "XMLHttpRequest Level 2". Author's Address Eric Rescorla RTFM, Inc. 2064 Edgewood Drive Palo Alto, CA 94303 USA Phone: +1 650 678 2350 Email: ekr@rtfm.com Rescorla Expires July 25, 2012 [Page 19]