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ART RTC-Web example WebRTC is a protocol suite for use with real-time applications that can be deployed in browsers - "real time communication on the Web". This document defines the WebRTC threat model and analyzes the security threats of WebRTC in that model.

Introduction The Real-Time Communications on the Web (RTCWEB) working group has standardized protocols for real-time communications between Web browsers, generally called "WebRTC" . The major use cases for WebRTC 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) WebRTC communications are directly controlled by some Web server. A simple case is shown below.

A simple WebRTC system | Browser | | | | | +-----------+ +-----------+ Alice Bob ]]>

In the system shown in , Alice and Bob both have WebRTC-enabled browsers and they visit some Web server which operates a calling service. Each of their browsers exposes standardized JavaScript calling APIs (implemented as browser built-ins) which are used by the Web server to set up a call between Alice and Bob. The Web server also serves as the signaling channel to transport control messages between the browsers. 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. A companion document describes a security architecture intended to address the issues raised in this document.

Terminology The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in BCP 14 when, and only when, they appear in all capitals, as shown here.

The Browser Threat Model The security requirements for WebRTC follow directly from the requirement that the browser's job is to protect the user. Huang et al. summarize the core browser security guarantee as:

Users can safely visit arbitrary web sites and execute scripts provided by those sites.

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 WebRTC 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 Coomputing Base (TCB) both from the user's perspective and to some extent from the server's. While HTML and JavaScript (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 "Internet Threat Model". Note that in some cases, a network attacker is also a web attacker, since transport protocols that do not provide integrity protection allow the network to inject traffic as if they were any communications peer. TLS, and HTTPS in particular, prevent against these attacks, but when analyzing HTTP connections, we must assume that traffic is going to the attacker.

Access to Local Resources 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 bypass security checks", as extensive research shows that users are prone to consent under such circumstances.] Similarly, while Flash programs (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).

Same-Origin Policy Many other resources are accessible but isolated. For instance, while scripts are allowed to make HTTP requests via the XMLHttpRequest() API (see ) those requests are not allowed to be made to any server, but rather solely to the same ORIGIN from whence the script came (although CORS and WebSockets provide an 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 B (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.

Bypassing SOP: CORS, WebSockets, and consent to communicate 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 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 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. 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.

Security for WebRTC Applications

Access to Local Devices As discussed in , 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 locations without user consent. This immediately raises the question, however, of what should be the scope of user 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 requesting access, where the media is going, or both. As detailed below, there are two basic conceptual models:

1. You are sending your media to entity A because you want to talk to Entity A (e.g., your mother).
2. Entity A (e.g., a calling service) asks to access the user's devices with the assurance that it will transfer the media to entity B (e.g., your mother)

In either case, identity is at the heart of any consent decision. Moreover, the identity of the party the browser is connecting to is all that the browser can meaningfully enforce; if you are calling A, A can simply forward the media to C. Similarly, if you authorize A to place a call to B, A can call C instead. In either cases, all the browser is able to do is verify and check authorization for whoever is controlling where the media goes. The target of the media can of course advertise a security/privacy policy, but this is not something that the browser can enforce. Even so, there are a variety of different consent scenarios that motivate different technical consent mechanisms. We discuss these mechanisms in the sections below. It's important to understand that consent to access local devices is largely orthogonal to consent to transmit various kinds of data over the network (see ). Consent for device access is largely a matter of protecting the user's privacy from malicious sites. By contrast, consent to send network traffic is about preventing the user's browser from being used to attack its local network. Thus, we need to ensure communications consent even if the site is not able to access the camera and microphone at all (hence WebSockets's consent mechanism) and similarly we need to be concerned with the site accessing the user's camera and microphone even if the data is to be sent back to the site via conventional HTTP-based network mechanisms such as HTTP POST.

Threats from Screen Sharing In addition to camera and microphone access, there has been demand for screen and/or application sharing functionality. Unfortunately, the security implications of this functionality are much harder for users to intuitively analyze than for camera and microphone access. (See http://lists.w3.org/Archives/Public/public-webrtc/2013Mar/0024.html for a full analysis.) The most obvious threats are simply those of "oversharing". I.e., the user may believe they are sharing a window when in fact they are sharing an application, or may forget they are sharing their whole screen, icons, notifications, and all. This is already an issue with existing screen sharing technologies and is made somewhat worse if a partially trusted site is responsible for asking for the resource to be shared rather than having the user propose it. A less obvious threat involves the impact of screen sharing on the Web security model. A key part of the Same-Origin Policy is that HTML or JS from site A can reference content from site B and cause the browser to load it, but (unless explicitly permitted) cannot see the result. However, if a web application from a site is screen sharing the browser, then this violates that invariant, with serious security consequences. For example, an attacker site might request screen sharing and then briefly open up a new Window to the user's bank or webmail account, using screen sharing to read the resulting displayed content. A more sophisticated attack would be open up a source view window to a site and use the screen sharing result to view anti cross-site request forgery tokens. These threats suggest that screen/application sharing might need a higher level of user consent than access to the camera or microphone.

Calling Scenarios and User Expectations 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.

Dedicated Calling Services 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. If I grant permission to calling service A to make calls on my behalf, then I am implicitly granting it permission to bug my computer whenever it wants. This suggests another consent model in which a site is authorized to make calls but only to certain target entities (identified via media-plane cryptographic mechanisms as described in and especially .) Note that the question of consent here is related to but distinct from the question of peer identity: I might be willing to allow a calling site to in general initiate calls on my behalf but still have some calls via that site where I can be sure that the site is not listening in.

Calling the Site You're On 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 , great care must be taken in the design of this interface to avoid the users just clicking through. Note also that the user interface chrome, which is the representation through which the user interacts with the user agent itself, must clearly display elements showing that the call is continuing in order to avoid attacks where the calling site just leaves it up indefinitely but shows a Web UI that implies otherwise.

Origin-Based Security Now that we have described the calling scenarios, we can start to reason about the security requirements. As discussed in , 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.

Individual Consent

Ask the user for permission for each call.

Callee-oriented Consent

Only allow calls to a given user.

Cryptographic Consent

Only allow calls to a given set of peer keying material or to a cryptographically established identity.

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 necessary 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. Callee-oriented consent provided by the calling site would 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 case of advertising, 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 tying permissions to origin here, but to the media origin (and-or destination) rather than to the Web origin. describes mechanisms which facilitate this sort of consent. 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. Naturally, it is somewhat challenging to design UI primitives which express this sort of policy. The problem becomes even more challenging in multi-user calling cases.

Security Properties of the Calling Page Origin-based security is intended to secure 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:

1. I connect to http://anything.example.org/. Note that this site is unaffiliated with the calling service.
2. The attacker modifies my HTTP connection to inject an IFRAME (or a redirect) to http://calling-service.example.com
3. The attacker forges the response from http://calling-service.example.com/ to inject JS to initiate a call to himself.

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 popped-up window that is hidden under the main window.) 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 those sites include active content (e.g., JavaScript) from an untrusted site, that JavaScript is executed in the security context of the page . This could lead to compromise of a call even if the parent page is safe. Note: this issue is not restricted to PAGES which contain untrusted content. If any page from a given origin ever loads JavaScript from an attacker, then it is possible for that attacker to infect the browser's notion of that origin semi-permanently.

Communications Consent Verification As discussed in , allowing web applications unrestricted network access via the browser 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. Note that consent verification is not sufficient to prevent overuse of network resources. Because WebRTC allows for a Web site to create data flows between two browser instances without user consent, it is possible for a malicious site to chew up a significant amount of a user's bandwidth without incurring significant costs to himself by setting up such a channel to another user. However, as a practical matter there are a large number of Web sites which can act as data sources, so an attacker can at least use downlink bandwidth with existing Web APIs. However, this potential DoS vector reinforces the need for adequate congestion control for WebRTC protocols to ensure that they play fair with other demands on the user's bandwidth.

ICE Verifying receiver consent requires some sort of explicit handshake, but conveniently we already need one in order to do NAT hole-punching. Interactive Connectivity Establishment (ICE) 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 the site 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. Verifying receiver consent also requires verifying the receiver wants to receive traffic from a particular sender, and at this time; for example a malicious site may simply attempt ICE to known servers that are using ICE for other sessions. ICE provides this verification as well, by using the STUN credentials as a form of per-session shared secret. Those credentials are known to the Web application, but would need to also be known and used by the STUN-receiving element to be useful. There also needs to be some mechanism for the browser to verify that the target of the traffic continues to wish to receive it. Because ICE keepalives are indications, they will not work here. describes the mechanism for providing consent freshness.

Masking Once consent is verified, there still is some concern about misinterpretation attacks as described by Huang et al.. Where TCP is used the risk is substantial due to the potential presence of transparent proxies and therefore if TCP is to be used, then WebSockets style masking MUST be employed. Since DTLS (with the anti-chosen plaintext mechanisms required by TLS 1.1) does not allow the attacker to generate predictable ciphertext, there is no need for masking of protocols running over DTLS (e.g. SCTP over DTLS, UDP over DTLS, etc.). Note that in principle an attacker could exert some control over SRTP packets by using a combination of the WebAudio API and extremely tight timing control. The primary risk here seems to be carriage of SRTP over TURN TCP. However, as SRTP packets have an extremely characteristic packet header it seems unlikely that any but the most aggressive intermediaries would be confused into thinking that another application layer protocol was in use.

Backward Compatibility 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:

• STUN checks without using ICE (i.e., the non-RTC-web endpoint sets up a STUN responder.)
• Use of RTCP as an implicit reachability check.

In the RTCP approach, the WebRTC 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 WebRTC 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 WebRTC 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 WebRTC 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. Once initial consent is verified, we also need to verify continuing consent, in order to avoid attacks where two people briefly share an IP (e.g., behind a NAT in an Internet cafe) and the attacker arranges for a large, unstoppable, traffic flow to the network and then leaves. The appropriate technologies here are fairly similar to those for initial consent, though are perhaps weaker since the threats are less severe.

IP Location Privacy Note that as soon as the callee sends their ICE candidates, the caller learns the callee's IP addresses. The callee's server reflexive address reveals a lot of information about the callee's location. In order to avoid tracking, implementations may wish to suppress the start of ICE negotiation until the callee has answered. In addition, either side may wish to hide their location from the other side entirely by forcing all traffic through a TURN server. In ordinary operation, the site learns the browser's IP address, though it may be hidden via mechanisms like Tor [http://www.torproject.org] or a VPN. However, because sites can cause the browser to provide IP addresses, this provides a mechanism for sites to learn about the user's network environment even if the user is behind a VPN that masks their IP address. Implementations may wish to provide settings which suppress all non-VPN candidates if the user is on certain kinds of VPN, especially privacy-oriented systems such as Tor. See for additional information.

Communications Security 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 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, SRTP , DTLS and DTLS-SRTP ) is well understood. However, we must examine this technology in the WebRTC 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. . 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:

Retrospective compromise of calling service.

The calling service is non-malicious during a call but subsequently is compromised and wishes to attack an older call (often called a "passive attack")

During-call attack by calling service.

The calling service is compromised during the call it wishes to attack (often called an "active attack").

Providing security against the former type of attack is practical using the techniques discussed in . However, it is extremely difficult to prevent a trusted but malicious calling service from actively attacking a user's calls, either by mounting a Man-in-the-Middle (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 .

Protecting Against Retrospective Compromise 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 ), 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 WebRTC (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, if end-to-end keying is in used, 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 During-Call Attack 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. Protecting against this form of attack requires positive authentication of the remote endpoint such as explicit out-of-band key verification (e.g., by a fingerprint) or a third-party identity service as described in .

Key Continuity 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 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 WebRTC context. First, much of the virtue of WebRTC (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 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 , 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 confusable with 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 use the trusted UI to provide the name and (b) the user to not be fooled by similar appearing names.

Short Authentication Strings ZRTP uses a "short authentication string" (SAS) which is derived from the key agreement protocol. This SAS is designed to be compared by the users (e.g., read aloud over the voice channel or transmitted via an out of band 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 and humans 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. Additionally, it is unclear that users will actually use an SAS. 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 is well-known that when faced with optional security mechanisms, many users simply ignore them . Once users 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 . 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.

Third Party Identity The conventional approach to providing communications identity has of course been to have some third party identity system (e.g., PKI) to authenticate the endpoints. Such mechanisms have proven to be too cumbersome for use by typical users (and nearly too cumbersome for administrators). However, a new generation of Web-based identity providers (BrowserID, Federated Google Login, Facebook Connect, OAuth , OpenID , WebFinger ), has recently been developed and use Web technologies to provide lightweight (from the user's perspective) third-party authenticated transactions. It is possible to use systems of this type to authenticate WebRTC calls, linking them to existing user notions of identity (e.g., Facebook adjacencies). Specifically, the third-party identity system is used to bind the user's identity to cryptographic keying material which is then used to authenticate the calling endpoints. Calls which are authenticated in this fashion are naturally resistant even to active MITM attack by the calling site. Note that there is one special case in which PKI-style certificates do provide a practical solution: calls from end-users to large sites. For instance, if you are making a call to Amazon.com, then Amazon can easily get a certificate to authenticate their media traffic, just as they get one to authenticate their Web traffic. This does not provide additional security value in cases in which the calling site and the media peer are one in the same, but might be useful in cases in which third parties (e.g., ad networks or retailers) arrange for calls but do not participate in them.

Page Access to Media Identifying the identity of the far media endpoint is a necessary but not sufficient condition for providing media security. In WebRTC, media flows are rendered into HTML5 MediaStreams which can be manipulated by the calling site. Obviously, if the site can modify or view the media, then the user is not getting the level of assurance they would expect from being able to authenticate their peer. In many cases, this is acceptable because the user values site-based special effects over complete security from the site. However, there are also cases where users wish to know that the site cannot interfere. In order to facilitate that, it will be necessary to provide features whereby the site can verifiably give up access to the media streams. This verification must be possible both from the local side and the remote side. I.e., users must be able to verify that the person called has engaged a secure media mode (see ). In order to achieve this it will be necessary to cryptographically bind an indication of the local media access policy into the cryptographic authentication procedures detailed in the previous sections. It should be noted that the use of this secure media mode is left to the discretion of the site. When such a mode is engaged, the browser will need to provide indicia to the user that the associated media has been authenticated as coming from the identified user. This allows WebRTC services that wish to claim end-to-end security to do so in a way that can be easily verified by the user. This model requires that the remote party's browser be included in the TCB, as described in .

Malicious Peers One class of attack that we do not generally try to prevent is malicious peers. For instance, no matter what confidentiality measures you employ the person you are talking to might record the call and publish it on the Internet. Similarly, we do not attempt to prevent them from using voice or video processing technology from hiding or changing their appearance. While technologies (DRM, etc.) do exist to attempt to address these issues, they are generally not compatible with open systems and WebRTC does not address them. Similarly, we make no attempt to prevent prank calling or other unwanted calls. In general, this is in the scope of the calling site, though because WebRTC does offer some forms of strong authentication, that may be useful as part of a defense against such attacks.

Privacy Considerations

Correlation of Anonymous Calls While persistent endpoint identifiers can be a useful security feature (see ) they can also represent a privacy threat in settings where the user wishes to be anonymous. WebRTC provides a number of possible persistent identifiers such as DTLS certificates (if they are reused between connections) and RTCP CNAMES (if generated according to rather than the privacy preserving mode of ). In order to prevent this type of correlation, browsers need to provide mechanisms to reset these identifiers (e.g., with the same lifetime as cookies). Moreover, the API should provide mechanisms to allow sites intended for anonymous calling to force the minting of fresh identifiers. In addition, IP addresses can be a source of call linkage .

Browser Fingerprinting Any new set of API features adds a risk of browser fingerprinting, and WebRTC is no exception. Specifically, sites can use the presence or absence of specific devices as a browser fingerprint. In general, the API needs to be balanced between functionality and the incremental fingerprint risk. See .

Security Considerations This entire document is about security.

Acknowledgements Bernard Aboba, Harald Alvestrand, Dan Druta, Cullen Jennings, Alan Johnston, Hadriel Kaplan (S 4.2.1), Matthew Kaufman, Martin Thomson, Magnus Westerlund.

IANA Considerations There are no IANA considerations.

Changes Since -04

• Replaced RTCWEB and RTC-Web with WebRTC, except when referring to the IETF WG
• Removed discussion of the IFRAMEd advertisement case, since we decided not to treat it specially.
• Added a privacy section considerations section.
• Significant edits to the SAS section to reflect Alan Johnston's comments.
• Added some discussion if IP location privacy and Tor.
• Updated the "communications consent" section to reflrect draft-ietf.
• Added a section about "malicious peers".
• Added a section describing screen sharing threats.
• Assorted editorial changes.

References Normative References In many standards track documents several words are used to signify the requirements in the specification. These words are often capitalized. This document defines these words as they should be interpreted in IETF documents. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements. RFC 2119 specifies common key words that may be used in protocol specifications. This document aims to reduce the ambiguity by clarifying that only UPPERCASE usage of the key words have the defined special meanings. Informative References This document describes Session Initiation Protocol (SIP), an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants. These sessions include Internet telephone calls, multimedia distribution, and multimedia conferences. [STANDARDS-TRACK] All RFCs are required to have a Security Considerations section. Historically, such sections have been relatively weak. This document provides guidelines to RFC authors on how to write a good Security Considerations section. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements. This document describes the Secure Real-time Transport Protocol (SRTP), a profile of the Real-time Transport Protocol (RTP), which can provide confidentiality, message authentication, and replay protection to the RTP traffic and to the control traffic for RTP, the Real-time Transport Control Protocol (RTCP). [STANDARDS-TRACK] This memo describes how to use Transport Layer Security (TLS) to secure Hypertext Transfer Protocol (HTTP) connections over the Internet. This memo provides information for the Internet community. This document describes requirements for a protocol to negotiate a security context for SIP-signaled Secure RTP (SRTP) media. In addition to the natural security requirements, this negotiation protocol must interoperate well with SIP in certain ways. A number of proposals have been published and a summary of these proposals is in the appendix of this document. This memo provides information for the Internet community. This document specifies how to use the Session Initiation Protocol (SIP) to establish a Secure Real-time Transport Protocol (SRTP) security context using the Datagram Transport Layer Security (DTLS) protocol. It describes a mechanism of transporting a fingerprint attribute in the Session Description Protocol (SDP) that identifies the key that will be presented during the DTLS handshake. The key exchange travels along the media path as opposed to the signaling path. The SIP Identity mechanism can be used to protect the integrity of the fingerprint attribute from modification by intermediate proxies. [STANDARDS-TRACK] This document specifies version 1.2 of the Datagram Transport Layer Security (DTLS) protocol. The DTLS protocol provides communications privacy for datagram protocols. The protocol allows client/server applications to communicate in a way that is designed to prevent eavesdropping, tampering, or message forgery. The DTLS protocol is based on the Transport Layer Security (TLS) protocol and provides equivalent security guarantees. Datagram semantics of the underlying transport are preserved by the DTLS protocol. This document updates DTLS 1.0 to work with TLS version 1.2. [STANDARDS-TRACK] This document defines a Session Description Protocol (SDP) cryptographic attribute for unicast media streams. The attribute describes a cryptographic key and other parameters that serve to configure security for a unicast media stream in either a single message or a roundtrip exchange. The attribute can be used with a variety of SDP media transports, and this document defines how to use it for the Secure Real-time Transport Protocol (SRTP) unicast media streams. The SDP crypto attribute requires the services of a data security protocol to secure the SDP message. [STANDARDS-TRACK] The Secure Shell (SSH) Protocol is a protocol for secure remote login and other secure network services over an insecure network. This document describes the architecture of the SSH protocol, as well as the notation and terminology used in SSH protocol documents. It also discusses the SSH algorithm naming system that allows local extensions. The SSH protocol consists of three major components: The Transport Layer Protocol provides server authentication, confidentiality, and integrity with perfect forward secrecy. The User Authentication Protocol authenticates the client to the server. The Connection Protocol multiplexes the encrypted tunnel into several logical channels. Details of these protocols are described in separate documents. [STANDARDS-TRACK] As the number, and more particularly the number of different types, of devices connecting to the Internet increases, credential mobility becomes an issue for IETF standardization. This document responds to the requirements on protocols for secure exchange of credentials listed in RFC 3157, by presenting an abstract protocol framework. This memo provides information for the Internet community. This document defines ZRTP, a protocol for media path Diffie-Hellman exchange to agree on a session key and parameters for establishing unicast Secure Real-time Transport Protocol (SRTP) sessions for Voice over IP (VoIP) applications. The ZRTP protocol is media path keying because it is multiplexed on the same port as RTP and does not require support in the signaling protocol. ZRTP does not assume a Public Key Infrastructure (PKI) or require the complexity of certificates in end devices. For the media session, ZRTP provides confidentiality, protection against man-in-the-middle (MiTM) attacks, and, in cases where the signaling protocol provides end-to-end integrity protection, authentication. ZRTP can utilize a Session Description Protocol (SDP) attribute to provide discovery and authentication through the signaling channel. To provide best effort SRTP, ZRTP utilizes normal RTP/AVP (Audio-Visual Profile) profiles. ZRTP secures media sessions that include a voice media stream and can also secure media sessions that do not include voice by using an optional digital signature. This document is not an Internet Standards Track specification; it is published for informational purposes. This document describes a protocol for Network Address Translator (NAT) traversal for UDP-based communication. This protocol is called Interactive Connectivity Establishment (ICE). ICE makes use of the Session Traversal Utilities for NAT (STUN) protocol and its extension, Traversal Using Relay NAT (TURN). This document obsoletes RFC 5245. The RTP Control Protocol (RTCP) Canonical Name (CNAME) is a persistent transport-level identifier for an RTP endpoint. While the Synchronization Source (SSRC) identifier of an RTP endpoint may change if a collision is detected or when the RTP application is restarted, its RTCP CNAME is meant to stay unchanged, so that RTP endpoints can be uniquely identified and associated with their RTP media streams. For proper functionality, RTCP CNAMEs should be unique within the participants of an RTP session. However, the existing guidelines for choosing the RTCP CNAME provided in the RTP standard are insufficient to achieve this uniqueness. This memo updates those guidelines to allow endpoints to choose unique RTCP CNAMEs. [STANDARDS-TRACK] This document defines the concept of an "origin", which is often used as the scope of authority or privilege by user agents. Typically, user agents isolate content retrieved from different origins to prevent malicious web site operators from interfering with the operation of benign web sites. In addition to outlining the principles that underlie the concept of origin, this document details how to determine the origin of a URI and how to serialize an origin into a string. It also defines an HTTP header field, named "Origin", that indicates which origins are associated with an HTTP request. [STANDARDS-TRACK] The WebSocket Protocol enables two-way communication between a client running untrusted code in a controlled environment to a remote host that has opted-in to communications from that code. The security model used for this is the origin-based security model commonly used by web browsers. The protocol consists of an opening handshake followed by basic message framing, layered over TCP. The goal of this technology is to provide a mechanism for browser-based applications that need two-way communication with servers that does not rely on opening multiple HTTP connections (e.g., using XMLHttpRequest or <iframe>s and long polling). [STANDARDS-TRACK] The OAuth 2.0 authorization framework enables a third-party application to obtain limited access to an HTTP service, either on behalf of a resource owner by orchestrating an approval interaction between the resource owner and the HTTP service, or by allowing the third-party application to obtain access on its own behalf. This specification replaces and obsoletes the OAuth 1.0 protocol described in RFC 5849. [STANDARDS-TRACK] The RTP Control Protocol (RTCP) Canonical Name (CNAME) is a persistent transport-level identifier for an RTP endpoint. While the Synchronization Source (SSRC) identifier of an RTP endpoint may change if a collision is detected or when the RTP application is restarted, its RTCP CNAME is meant to stay unchanged, so that RTP endpoints can be uniquely identified and associated with their RTP media streams. For proper functionality, RTCP CNAMEs should be unique within the participants of an RTP session. However, the existing guidelines for choosing the RTCP CNAME provided in the RTP standard (RFC 3550) are insufficient to achieve this uniqueness. RFC 6222 was published to update those guidelines to allow endpoints to choose unique RTCP CNAMEs. Unfortunately, later investigations showed that some parts of the new algorithms were unnecessarily complicated and/or ineffective. This document addresses these concerns and replaces RFC 6222. This specification defines the WebFinger protocol, which can be used to discover information about people or other entities on the Internet using standard HTTP methods. WebFinger discovers information for a URI that might not be usable as a locator otherwise, such as account or email URIs. To prevent WebRTC applications, such as browsers, from launching attacks by sending traffic to unwilling victims, periodic consent to send needs to be obtained from remote endpoints. This document describes a consent mechanism using a new Session Traversal Utilities for NAT (STUN) usage. This document defines the security architecture for WebRTC, a protocol suite intended for use with real-time applications that can be deployed in browsers - "real time communication on the Web". This document provides information and requirements for how IP addresses should be handled by WebRTC implementations. This document gives an overview and context of a protocol suite intended for use with real-time applications that can be deployed in browsers - "real time communication on the Web". It intends to serve as a starting and coordination point to make sure all the parts that are needed to achieve this goal are findable, and that the parts that belong in the Internet protocol suite are fully specified and on the right publication track. This document is an Applicability Statement - it does not itself specify any protocol, but specifies which other specifications WebRTC compliant implementations are supposed to follow. This document is a work item of the RTCWEB working group.