| RTC-Web | E.K. Rescorla |
| Internet-Draft | RTFM, Inc. |
| Intended status: Standards Track | September 22, 2011 |
| Expires: March 25, 2012 |
Security Considerations for RTC-Web
draft-ietf-rtcweb-security-00
The Real-Time Communications on the Web (RTC-Web) working group is tasked with standardizing protocols for real-time communications between Web browsers. The major use cases for RTC-Web 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) RTC-Web 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. This document defines the RTC-Web threat model and defines an architecture which provides security within that threat model.
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The Real-Time Communications on the Web (RTC-Web) working group is tasked with standardizing protocols for real-time communications between Web browsers. The major use cases for RTC-Web 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) RTC-Web communications are directly controlled by some Web server. A simple case is shown below.
+----------------+
| |
| Web Server |
| |
+----------------+
^ ^
/ \
HTTP / \ HTTP
/ \
/ \
v v
JS API JS API
+-----------+ +-----------+
| | Media | |
| Browser |<---------->| Browser |
| | | |
+-----------+ +-----------+
In the system shown in Figure 1, Alice and Bob both have RTC-Web enabled browsers and they visit some Web server which operates a calling service. Each of their browsers exposes standardized JavaScript calling APIs which are used by the Web server to set up a call between Alice and Bob. While this system is topologically similar to a conventional SIP-based system (with the Web server acting as the signaling service and browsers acting as softphones), control has moved to the central Web server; the browser simply provides API points that are used by the calling service. As with any Web application, the Web server can move logic between the server and JavaScript in the browser, but regardless of where the code is executing, it is ultimately under control of the server.
It should be immediately apparent that this type of system poses new security challenges beyond those of a conventional VoIP system. In particular, it needs to contend with malicious calling services. For example, if the calling service can cause the browser to make a call at any time to any callee of its choice, then this facility can be used to bug a user's computer without their knowledge, simply by placing a call to some recording service. More subtly, if the exposed APIs allow the server to instruct the browser to send arbitrary content, then they can be used to bypass firewalls or mount denial of service attacks. Any successful system will need to be resistant to this and other attacks.
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].
The security requirements for RTC-Web follow directly from the requirement that the browser's job is to protect the user. Huang et al. [huang-w2sp] summarize the core browser security guarantee as:
It is important to realize that this includes sites hosting arbitrary malicious scripts. The motivation for this requirement is simple: it is trivial for attackers to divert users to sites of their choice. For instance, an attacker can purchase display advertisements which direct the user (either automatically or via user clicking) to their site, at which point the browser will execute the attacker's scripts. Thus, it is important that it be safe to view arbitrarily malicious pages. Of course, browsers inevitably have bugs which cause them to fall short of this goal, but any new RTC-Web functionality must be designed with the intent to meet this standard. The remainder of this section provides more background on the existing Web security model.
In this model, then, the browser acts as a TRUSTED COMPUTING BASE (TCB) both from the user's perspective and to some extent from the server's. While HTML and JS provided by the server can cause the browser to execute a variety of actions, those scripts operate in a sandbox that isolates them both from the user's computer and from each other, as detailed below.
Conventionally, we refer to either WEB ATTACKERS, who are able to induce you to visit their sites but do not control the network, and NETWORK ATTACKERS, who are able to control your network. Network attackers correspond to the [RFC3552] "Internet Threat Model". In general, it is desirable to build a system which is secure against both kinds of attackers, but realistically many sites do not run HTTPS [RFC2818] and so our ability to defend against network attackers is necessarily somewhat limited. Most of the rest of this section is devoted to web attackers, with the assumption that protection against network attackers is provided by running HTTPS.
While the browser has access to local resources such as keying material, files, the camera and the microphone, it strictly limits or forbids web servers from accessing those same resources. For instance, while it is possible to produce an HTML form which will allow file upload, a script cannot do so without user consent and in fact cannot even suggest a specific file (e.g., /etc/passwd); the user must explicitly select the file and consent to its upload. [Note: in many cases browsers are explicitly designed to avoid dialogs with the semantics of "click here to screw yourself", as extensive research shows that users are prone to consent under such circumstances.]
Similarly, while Flash SWFs can access the camera and microphone, they explicitly require that the user consent to that access. In addition, some resources simply cannot be accessed from the browser at all. For instance, there is no real way to run specific executables directly from a script (though the user can of course be induced to download executable files and run them).
Many other resources are accessible but isolated. For instance, while scripts are allowed to make HTTP requests via the XMLHttpRequest() API those requests are not allowed to be made to any server, but rather solely to the same ORIGIN from whence the script came.[I-D.abarth-origin] (although CORS [CORS] and WebSockets [I-D.ietf-hybi-thewebsocketprotocol] provides a escape hatch from this restriction, as described below. This SAME ORIGIN POLICY (SOP) prevents server A from mounting attacks on server B via the user's browser, which protects both the user (e.g., from misuse of his credentials) and the server (e.g., from DoS attack).
More generally, SOP forces scripts from each site to run in their own, isolated, sandboxes. While there are techniques to allow them to interact, those interactions generally must be mutually consensual (by each site) and are limited to certain channels. For instance, multiple pages/browser panes from the same origin can read each other's JS variables, but pages from the different origins--or even iframes from different origins on the same page--cannot.
While SOP serves an important security function, it also makes it inconvenient to write certain classes of applications. In particular, mash-ups, in which a script from origin A uses resources from origin B, can only be achieved via a certain amount of hackery. The W3C Cross-Origin Resource Sharing (CORS) spec [CORS] is a response to this demand. In CORS, when a script from origin A executes what would otherwise be a forbidden cross-origin request, the browser instead contacts the target server to determine whether it is willing to allow cross-origin requests from A. If it is so willing, the browser then allows the request. This consent verification process is designed to safely allow cross-origin requests.
While CORS is designed to allow cross-origin HTTP requests, WebSockets [I-D.ietf-hybi-thewebsocketprotocol] allows cross-origin establishment of transparent channels. Once a WebSockets connection has been established from a script to a site, the script can exchange any traffic it likes without being required to frame it as a series of HTTP request/response transactions. As with CORS, a WebSockets transaction starts with a consent verification stage to avoid allowing scripts to simply send arbitrary data to another origin.
While consent verification is conceptually simple--just do a handshake before you start exchanging the real data--experience has shown that designing a correct consent verification system is difficult. In particular, Huang et al. [huang-w2sp] have shown vulnerabilities in the existing Java and Flash consent verification techniques and in a simplified version of the WebSockets handshake. In particular, it is important to be wary of CROSS-PROTOCOL attacks in which the attacking script generates traffic which is acceptable to some non-Web protocol state machine. In order to resist this form of attack, WebSockets incorporates a masking technique intended to randomize the bits on the wire, thus making it more difficult to generate traffic which resembles a given protocol.
As discussed in Section 1, allowing arbitrary sites to initiate calls violates the core Web security guarantee; without some access restrictions on local devices, any malicious site could simply bug a user. At minimum, then, it MUST NOT be possible for arbitrary sites to initiate calls to arbitrary location without user consent. This immediately raises the question, however, of what should be the scope of user consent.
For the rest of this discussion we assume that the user is somehow going to grant consent to some entity (e.g., a social networking site) to initiate a call on his behalf. This consent may be limited to a single call or may be a general consent. In order for the user to make an intelligent decision about whether to allow a call (and hence his camera and microphone input to be routed somewhere), he must understand either who is request access, where the media is going, or both. So, for instance, one might imagine that at the time access to camera and microphone is requested, the user is shown a dialog that says "site X has requested access to camera and microphone, yes or no" (though note that this type of in-flow interface violates one of the guidelines in Section Section 3). The user's decision will of course be based on his opinion of Site X. However, as discussed below, this is a complicated concept.
While a large number of possible calling scenarios are possible, the scenarios discussed in this section illustrate many of the difficulties of identifying the relevant scope of consent.
The first scenario we consider is a dedicated calling service. In this case, the user has a relationship with a calling site and repeatedly makes calls on it. It is likely that rather than having to give permission for each call that the user will want to give the calling service long-term access to the camera and microphone. This is a natural fit for a long-term consent mechanism (e.g., installing an app store "application" to indicate permission for the calling service.) A variant of the dedicated calling service is a gaming site (e.g., a poker site) which hosts a dedicated calling service to allow players to call each other.
With any kind of service where the user may use the same service to talk to many different people, there is a question about whether the user can know who they are talking to. In general, this is difficult as most of the user interface is presented by the calling site. However, communications security mechanisms can be used to give some assurance, as described in Section 4.3.2.
Another simple scenario is calling the site you're actually visiting. The paradigmatic case here is the "click here to talk to a representative" windows that appear on many shopping sites. In this case, the user's expectation is that they are calling the site they're actually visiting. However, it is unlikely that they want to provide a general consent to such a site; just because I want some information on a car doesn't mean that I want the car manufacturer to be able to activate my microphone whenever they please. Thus, this suggests the need for a second consent mechanism where I only grant consent for the duration of a given call. As described in Section 3.1, great care must be taken in the design of this interface to avoid the users just clicking through.
In both of the previous cases, the user has a direct relationship (though perhaps a transient one) with the target of the call. Moreover, in both cases he is actually visiting the site of the person he is being asked to trust. However, this is not always so. Consider the case where a user is a visiting a content site which hosts an advertisement with an invitation to call for more information. When the user clicks the ad, they are connected with the advertiser or their agent.
The relationships here are far more complicated: the site the user is actually visiting has no direct relationship with the advertiser; they are just hosting ads from an ad network. The user has no relationship with the ad network, but desires one with the advertiser, at least for long enough to learn about their products. At minimum, then, whatever consent dialog is shown needs to allow the user to have some idea of the organization that they are actually calling.
However, because the user also has some relationship with the hosting site, it is also arguable that the hosting site should be allowed to express an opinion (e.g., to be able to allow or forbid a call) since a bad experience with an advertiser reflect negatively on the hosting site [this idea was suggested by Adam Barth]. However, this obviously presents a privacy challenge, as sites which host advertisements often learn very little about whether individual users clicked through to the ads, or even which ads were presented.
As discussed in Section 3.2, the basic unit of Web sandboxing is the origin, and so it is natural to scope consent to origin. Specifically, a script from origin A MUST only be allowed to initiate communications (and hence to access camera and microphone) if the user has specifically authorized access for that origin. It is of course technically possible to have coarser-scoped permissions, but because the Web model is scoped to origin, this creates a difficult mismatch.
Arguably, origin is not fine-grained enough. Consider the situation where Alice visits a site and authorizes it to make a single call. If consent is expressed solely in terms of origin, then at any future visit to that site (including one induced via mash-up or ad network), the site can bug Alice's computer, use the computer to place bogus calls, etc. While in principle Alice could grant and then revoke the privilege, in practice privileges accumulate; if we are concerned about this attack, something else is needed. There are a number of potential countermeasures to this sort of issue.
Unfortunately, none of these approaches is satisfactory for all cases. As discussed above, individual consent puts the user's approval in the UI flow for every call. Not only does this quickly become annoying but it can train the user to simply click "OK", at which point the consent becomes useless. Thus, while it may be necssary to have individual consent in some case, this is not a suitable solution for (for instance) the calling service case. Where necessary, in-flow user interfaces must be carefully designed to avoid the risk of the user blindly clicking through.
The other two options are designed to restrict calls to a given target. Unfortunately, Callee-oriented consent does not work well because a malicious site can claim that the user is calling any user of his choice. One fix for this is to tie calls to a cryptographically established identity. While not suitable for all cases, this approach may be useful for some. If we consider the advertising case described in Section 4.1.1.3, it's not particularly convenient to require the advertiser to instantiate an iframe on the hosting site just to get permission; a more convenient approach is to cryptographically tie the advertiser's certificate to the communication directly. We're still tieing permissions to origin here, but to the media origin (and-or destination) rather than to the Web origin.
Another case where media-level cryptographic identity makes sense is when a user really does not trust the calling site. For instance, I might be worried that the calling service will attempt to bug my computer, but I also want to be able to conveniently call my friends. If consent is tied to particular communications endpoints, then my risk is limited. However, this is also not that convenient an interface, since managing individual user permissions can be painful.
While this is primarily a question not for IETF, it should be clear that there is no really good answer. In general, if you cannot trust the site which you have authorized for calling not to bug you then your security situation is not really ideal. It is RECOMMENDED that browsers have explicit (and obvious) indicators that they are in a call in order to mitigate this risk.
Origin-based security is intended to security against web attackers. However, we must also consider the case of network attackers. Consider the case where I have granted permission to a calling service by an origin that has the HTTP scheme, e.g., http://calling-service.example.com. If I ever use my computer on an unsecured network (e.g., a hotspot or if my own home wireless network is insecure), and browse any HTTP site, then an attacker can bug my computer. The attack proceeds like this:
Note that this attack does not depend on the media being insecure. Because the call is to the attacker, it is also encrypted to him. Moreover, it need not be executed immediately; the attacker can "infect" the origin semi-permanently (e.g., with a web worker or a popunder) and thus be able to bug me long after I have left the infected network. This risk is created by allowing calls at all from a page fetched over HTTP.
Even if calls are only possible from HTTPS sites, if the site embeds active content (e.g., JavaScript) that is fetched over HTTP or from an untrusted site, because that JavaScript is executed in the security context of the page [finer-grained]. Thus, it is also dangerous to allow RTC-Web functionality from HTTPS origins that embed mixed content. Note: this issue is not restricted to PAGES which contain mixed content. If a page from a given origin ever loads mixed content then it is possible for a network attacker to infect the browser's notion of that origin semi-permanently.
[[ OPEN ISSUE: What recommendation should IETF make about (a) whether RTCWeb long-term consent should be available over HTTP pages and (b) How to handle origins where the consent is to an HTTPS URL but the page contains active mixed content? ]]
As discussed in Section 3.3, allowing web applications unrestricted access to the via the browser network introduces the risk of using the browser as an attack platform against machines which would not otherwise be accessible to the malicious site, for instance because they are topologically restricted (e.g., behind a firewall or NAT). In order to prevent this form of attack as well as cross-protocol attacks it is important to require that the target of traffic explicitly consent to receiving the traffic in question. Until that consent has been verified for a given endpoint, traffic other than the consent handshake MUST NOT be sent to that endpoint.
Verifying receiver consent requires some sort of explicit handshake, but conveniently we already need one in order to do NAT hole-punching. ICE [RFC5245] includes a handshake designed to verify that the receiving element wishes to receive traffic from the sender. It is important to remember here that the site initiating ICE is presumed malicious; in order for the handshake to be secure the receiving element MUST demonstrate receipt/knowledge of some value not available to the site (thus preventing it from forging responses). In order to achieve this objective with ICE, the STUN transaction IDs must be generated by the browser and MUST NOT be made available to the initiating script, even via a diagnostic interface.
Once consent is verified, there still is some concern about misinterpretation attacks as described by Huang et al.[huang-w2sp]. As long as communication is limited to UDP, then this risk is probably limited, thus masking is not required for UDP. I.e., once communications consent has been verified, it is most likely safe to allow the implementation to send arbitrary UDP traffic to the chosen destination, provided that the STUN keepalives continue to succeed. However, with TCP the risk of transparent proxies becomes much more severe. If TCP is to be used, then WebSockets style masking MUST be employed.
A requirement to use ICE limits compatibility with legacy non-ICE clients. It seems unsafe to completely remove the requirement for some check. All proposed checks have the common feature that the browser sends some message to the candidate traffic recipient and refuses to send other traffic until that message has been replied to. The message/reply pair must be generated in such a way that an attacker who controls the Web application cannot forge them, generally by having the message contain some secret value that must be incorporated (e.g., echoed, hashed into, etc.). Non-ICE candidates for this role (in cases where the legacy endpoint has a public address) include:
In the RTCP approach, the RTC-Web endpoint is allowed to send a limited number of RTP packets prior to receiving consent. This allows a short window of attack. In addition, some legacy endpoints do not support RTCP, so this is a much more expensive solution for such endpoints, for which it would likely be easier to implement ICE. For these two reasons, an RTCP-based approach does not seem to address the security issue satisfactorily.
In the STUN approach, the RTC-Web endpoint is able to verify that the recipient is running some kind of STUN endpoint but unless the STUN responder is integrated with the ICE username/password establishment system, the RTC-Web endpoint cannot verify that the recipient consents to this particular call. This may be an issue if existing STUN servers are operated at addresses that are not able to handle bandwidth-based attacks. Thus, this approach does not seem satisfactory either.
If the systems are tightly integrated (i.e., the STUN endpoint responds with responses authenticated with ICE credentials) then this issue does not exist. However, such a design is very close to an ICE-Lite implementation (indeed, arguably is one). An intermediate approach would be to have a STUN extension that indicated that one was responding to RTC-Web checks but not computing integrity checks based on the ICE credentials. This would allow the use of standalone STUN servers without the risk of confusing them with legacy STUN servers. If a non-ICE legacy solution is needed, then this is probably the best choice.
[TODO: Write something about consent freshness and RTCP].
[[ OPEN ISSUE: Exactly what should be the requirements here? Proposals include ICE all the time or ICE but with allowing one of these non-ICE things for legacy. ]]
Finally, we consider a problem familiar from the SIP world: communications security. For obvious reasons, it MUST be possible for the communicating parties to establish a channel which is secure against both message recovery and message modification. (See [RFC5479] for more details.) This service must be provided for both data and voice/video. Ideally the same security mechanisms would be used for both types of content. Technology for providing this service (for instance, DTLS [RFC4347] and DTLS-SRTP [RFC5763]) is well understood. However, we must examine this technology to the RTC-Web context, where the threat model is somewhat different.
In general, it is important to understand that unlike a conventional SIP proxy, the calling service (i.e., the Web server) controls not only the channel between the communicating endpoints but also the application running on the user's browser. While in principle it is possible for the browser to cut the calling service out of the loop and directly present trusted information (and perhaps get consent), practice in modern browsers is to avoid this whenever possible. "In-flow" modal dialogs which require the user to consent to specific actions are particularly disfavored as human factors research indicates that unless they are made extremely invasive, users simply agree to them without actually consciously giving consent. [abarth-rtcweb]. Thus, nearly all the UI will necessarily be rendered by the browser but under control of the calling service. This likely includes the peer's identity information, which, after all, is only meaningful in the context of some calling service.
This limitation does not mean that preventing attack by the calling service is completely hopeless. However, we need to distinguish between two classes of attack:
Providing security against the former type of attack is practical using the techniques discussed in Section 4.3.1. However, it is extremely difficult to prevent a trusted but malicious calling service from actively attacking a user's calls, either by mounting a MITM attack or by diverting them entirely. (Note that this attack applies equally to a network attacker if communications to the calling service are not secured.) We discuss some potential approaches and why they are likely to be impractical in Section 4.3.2.
In a retrospective attack, the calling service was uncompromised during the call, but that an attacker subsequently wants to recover the content of the call. We assume that the attacker has access to the protected media stream as well as having full control of the calling service.
If the calling service has access to the traffic keying material (as in SDES [RFC4568]), then retrospective attack is trivial. This form of attack is particularly serious in the Web context because it is standard practice in Web services to run extensive logging and monitoring. Thus, it is highly likely that if the traffic key is part of any HTTP request it will be logged somewhere and thus subject to subsequent compromise. It is this consideration that makes an automatic, public key-based key exchange mechanism imperative for RTC-Web (this is a good idea for any communications security system) and this mechanism SHOULD provide perfect forward secrecy (PFS). The signaling channel/calling service can be used to authenticate this mechanism.
In addition, the system MUST NOT provide any APIs to extract either long-term keying material or to directly access any stored traffic keys. Otherwise, an attacker who subsequently compromised the calling service might be able to use those APIs to recover the traffic keys and thus compromise the traffic.
Protecting against attacks during a call is a more difficult proposition. Even if the calling service cannot directly access keying material (as recommended in the previous section), it can simply mount a man-in-the-middle attack on the connection, telling Alice that she is calling Bob and Bob that he is calling Alice, while in fact the calling service is acting as a calling bridge and capturing all the traffic. While in theory it is possible to construct techniques which protect against this form of attack, in practice these techniques all require far too much user intervention to be practical, given the user interface constraints described in [abarth-rtcweb].
One natural approach is to use "key continuity". While a malicious calling service can present any identity it chooses to the user, it cannot produce a private key that maps to a given public key. Thus, it is possible for the browser to note a given user's public key and generate an alarm whenever that user's key changes. SSH [RFC4251] uses a similar technique. (Note that the need to avoid explicit user consent on every call precludes the browser requiring an immediate manual check of the peer's key).
Unfortunately, this sort of key continuity mechanism is far less useful in the RTC-Web context. First, much of the virtue of RTC-Web (and any Web application) is that it is not bound to particular piece of client software. Thus, it will be not only possible but routine for a user to use multiple browsers on different computers which will of course have different keying material (SACRED [RFC3760] notwithstanding.) Thus, users will frequently be alerted to key mismatches which are in fact completely legitimate, with the result that they are trained to simply click through them. As it is known that users routinely will click through far more dire warnings [cranor-wolf], it seems extremely unlikely that any key continuity mechanism will be effective rather than simply annoying.
Moreover, it is trivial to bypass even this kind of mechanism. Recall that unlike the case of SSH, the browser never directly gets the peer's identity from the user. Rather, it is provided by the calling service. Even enabling a mechanism of this type would require an API to allow the calling service to tell the browser "this is a call to user X". All the calling service needs to do to avoid triggering a key continuity warning is to tell the browser that "this is a call to user Y" where Y is close to X. Even if the user actually checks the other side's name (which all available evidence indicates is unlikely), this would require (a) the browser to trusted UI to provide the name and (b) the user to not be fooled by similar appearing names.
ZRTP [RFC6189] uses a "short authentication string" (SAS) which is derived from the key agreement protocol. This SAS is designed to be read over the voice channel and if confirmed by both sides precludes MITM attack. The intention is that the SAS is used once and then key continuity (though a different mechanism from that discussed above) is used thereafter.
Unfortunately, the SAS does not offer a practical solution to the problem of a compromised calling service. "Voice conversion" systems, which modify voice from one speaker to make it sound like another, are an active area of research. These systems are already good enough to fool both automatic recognition systems [farus-conversion] and humans [kain-conversion] in many cases, and are of course likely to improve in future, especially in an environment where the user just wants to get on with the phone call. Thus, even if SAS is effective today, it is likely not to be so for much longer. Moreover, it is possible for an attacker who controls the browser to allow the SAS to succeed and then simulate call failure and reconnect, trusting that the user will not notice that the "no SAS" indicator has been set (which seems likely).
Even were SAS secure if used, it seems exceedingly unlikely that users will actually use it. As discussed above, the browser UI constraints preclude requiring the SAS exchange prior to completing the call and so it must be voluntary; at most the browser will provide some UI indicator that the SAS has not yet been checked. However, it it is well-known that when faced with optional mechanisms such as fingerprints, users simply do not check them [whitten-johnny] Thus, it is highly unlikely that users will ever perform the SAS exchange.
Once uses have checked the SAS once, key continuity is required to avoid them needing to check it on every call. However, this is problematic for reasons indicated in Section 4.3.2.1. In principle it is of course possible to render a different UI element to indicate that calls are using an unauthenticated set of keying material (recall that the attacker can just present a slightly different name so that the attack shows the same UI as a call to a new device or to someone you haven't called before) but as a practical matter, users simply ignore such indicators even in the rather more dire case of mixed content warnings.
[[ OPEN ISSUE: What are the best UI recommendations to make? Proposal: take the text from [I-D.kaufman-rtcweb-security-ui] Section 2]]
[[ OPEN ISSUE: Exactly what combination of media security primitives should be specified and/or mandatory to implement? In particular, should we allow DTLS-SRTP only, or both DTLS-SRTP and SDES. Should we allow RTP for backward compatibility? ]]
This entire document is about security.
| [RFC2119] | Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, March 1997. |