RFC 8826 WebRTC Security June 2020
Rescorla Standards Track [Page]
Stream:
Internet Engineering Task Force (IETF)
RFC:
8826
Category:
Standards Track
Published:
ISSN:
2070-1721
Author:
E. Rescorla
RTFM, Inc.

RFC 8826

Security Considerations for WebRTC

Abstract

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.

Status of This Memo

This is an Internet Standards Track document.

This document is a product of the Internet Engineering Task Force (IETF). It represents the consensus of the IETF community. It has received public review and has been approved for publication by the Internet Engineering Steering Group (IESG). Further information on Internet Standards is available in Section 2 of RFC 7841.

Information about the current status of this document, any errata, and how to provide feedback on it may be obtained at https://www.rfc-editor.org/info/rfc8826.

Table of Contents

1. Introduction

The Real-Time Communications on the Web (RTCWEB) working group has standardized protocols for real-time communications between Web browsers, generally called "WebRTC" [RFC8825]. 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 [RFC3261] soft phones), WebRTC communications are directly controlled by some Web server. A simple case is shown below.

                          +----------------+
                          |                |
                          |   Web Server   |
                          |                |
                          +----------------+
                              ^        ^
                             /          \
                     HTTP   /            \   HTTP
                      or   /              \   or
               WebSockets /                \ WebSockets
                         v                  v
                      JS API              JS API
                +-----------+            +-----------+
                |           |    Media   |           |
                |  Browser  |<---------->|  Browser  |
                |           |            |           |
                +-----------+            +-----------+
                    Alice                     Bob
Figure 1: A Simple WebRTC System

In the system shown in Figure 1, Alice and Bob both have WebRTC-enabled browsers and they visit some Web server that operates a calling service. Each of their browsers exposes standardized JavaScript (JS) 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 Voice over IP (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 DoS attacks. Any successful system will need to be resistant to this and other attacks.

A companion document [RFC8827] describes a security architecture intended to address the issues raised in this document.

2. 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 [RFC2119] [RFC8174] when, and only when, they appear in all capitals, as shown here.

3. 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. [huang-w2sp] summarize the core browser security guarantee as follows:

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 Computing Base (TCB) both from the user's perspective and to some extent from the server's. While HTML and JavaScript 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". 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.

3.1. 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 [cranor-wolf] shows that users are prone to consent under such circumstances.)

Similarly, while Flash programs (SWFs) [SWF] 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).

3.2. 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 [XmlHttpRequest]) those requests are not allowed to be made to any server, but rather solely to the same ORIGIN from whence the script came [RFC6454] (although Cross-Origin Resource Sharing (CORS) [CORS] and WebSockets [RFC6455] 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 server B (e.g., from DoS attacks).

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 different origins -- or even iframes from different origins on the same page -- cannot.

3.3. 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 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 [RFC6455] 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.

4. Security for WebRTC Applications

4.1. Access to Local Devices

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 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 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 case, 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 Section 4.2). 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); 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.

4.1.1. 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 <https://lists.w3.org/Archives/Public/public-webrtc/2013Mar/0024.html> for a full analysis.)

The most obvious threats are simply those of "oversharing". That is, 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 to 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.

4.1.2. 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.

4.1.2.1. 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, 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 Section 4.3.2 and especially Section 4.3.2.3). 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.

4.1.2.2. 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 Section 3.1, 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.

4.1.3. Origin-Based Security

Now that we have described the calling scenarios, we can start to reason about the security requirements.

As discussed in Section 3.2, the basic unit of Web sandboxing is the origin, and so it is natural to scope consent to the origin. Specifically, a script from origin A MUST only be allowed to initiate communications (and hence to access the 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 the origin, this creates a difficult mismatch.

Arguably, the 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 the origin, then upon any future visit to that site (including one induced via a 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 cases, 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 the origin here, but to the media origin (and/or destination) rather than to the Web origin. [RFC8827] 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 that express this sort of policy. The problem becomes even more challenging in multi-user calling cases.

4.1.4. 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 [RFC2818] 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 [finer-grained]. 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.

4.3. 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 [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, SRTP [RFC3711], DTLS [RFC6347], and DTLS-SRTP [RFC5763]) 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 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:

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 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 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 Section 4.3.2.

4.3.1. Protecting against Retrospective Compromise

In a retrospective attack, the calling service was uncompromised during the call, but 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 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 context of the Web 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 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.

4.3.2. 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 [RFC8827].

4.3.2.1. 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. The Secure Shell (SSH) protocol [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 WebRTC context. First, much of the virtue of WebRTC (and any Web application) is that it is not bound to a particular piece of client software. Thus, it will be not only possible but routine for a user to use multiple browsers on different computers that will of course have different keying material (Securely Available Credentials (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 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.

4.3.2.2. Short Authentication Strings

ZRTP [RFC6189] 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 [farrus-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 the 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 [whitten-johnny].

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 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.

4.3.2.3. 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 [RFC6749], OpenID [OpenID], WebFinger [RFC7033]) 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 and 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.

4.3.2.4. 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. That is, users must be able to verify that the person called has engaged a secure media mode (see Section 4.3.3). 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 Section 3.

4.3.3. 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 (Digital Rights Management (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.

4.4. Privacy Considerations

4.4.1. Correlation of Anonymous Calls

While persistent endpoint identifiers can be a useful security feature (see Section 4.3.2.1), 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 [RFC6222] rather than the privacy-preserving mode of [RFC7022]). 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 [RFC8828].

4.4.2. 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 [Fingerprinting].

5. Security Considerations

This entire document is about security.

6. IANA Considerations

This document has no IANA actions.

7. References

7.1. Normative References

[RFC2119]
Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, DOI 10.17487/RFC2119, , <https://www.rfc-editor.org/info/rfc2119>.
[RFC8174]
Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC 2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174, , <https://www.rfc-editor.org/info/rfc8174>.

7.2. Informative References

[abarth-rtcweb]
Barth, A., "Prompting the user is security failure", RTC-Web Workshop, , <http://rtc-web.alvestrand.com/home/papers/barth-security-prompt.pdf?attredirects=0>.
[CORS]
van Kesteren, A., "Cross-Origin Resource Sharing", .
[cranor-wolf]
Sunshine, J., Egelman, S., Almuhimedi, H., Atri, N., and L. Cranor, "Crying Wolf: An Empirical Study of SSL Warning Effectiveness", Proceedings of the 18th USENIX Security Symposium, , <https://www.usenix.org/legacy/event/sec09/tech/full_papers/sunshine.pdf>.
[farrus-conversion]
Farrus, M., Erro, D., and J. Hernando, "Speaker Recognition Robustness to Voice Conversion", .
[finer-grained]
Jackson, C. and A. Barth, "Beware of Finer-Grained Origins", Web 2.0 Security and Privacy (W2SP 2008), .
[Fingerprinting]
"Fingerprinting Guidance for Web Specification Authors (Draft)", , <https://www.w3.org/TR/fingerprinting-guidance/#acknowledgement/>.
[huang-w2sp]
Huang, L-S., Chen, E.Y., Barth, A., Rescorla, E., and C. Jackson, "Talking to Yourself for Fun and Profit", Web 2.0 Security and Privacy (W2SP 2011), .
[kain-conversion]
Kain, A. and M. Macon, "Design and Evaluation of a Voice Conversion Algorithm based on Spectral Envelope Mapping and Residual Prediction", Proceedings of the 2001 IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP), DOI 10.1109/ICASSP.2001.941039, , <https://doi.org/10.1109/ICASSP.2001.941039>.
[OpenID]
Sakimura, N., Bradley, J., Jones, M., de Medeiros, B., and C. Mortimore, "OpenID Connect Core 1.0", , <https://openid.net/specs/openid-connect-core-1_0.html>.
[RFC2818]
Rescorla, E., "HTTP Over TLS", RFC 2818, DOI 10.17487/RFC2818, , <https://www.rfc-editor.org/info/rfc2818>.
[RFC3261]
Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A., Peterson, J., Sparks, R., Handley, M., and E. Schooler, "SIP: Session Initiation Protocol", RFC 3261, DOI 10.17487/RFC3261, , <https://www.rfc-editor.org/info/rfc3261>.
[RFC3552]
Rescorla, E. and B. Korver, "Guidelines for Writing RFC Text on Security Considerations", BCP 72, RFC 3552, DOI 10.17487/RFC3552, , <https://www.rfc-editor.org/info/rfc3552>.
[RFC3711]
Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. Norrman, "The Secure Real-time Transport Protocol (SRTP)", RFC 3711, DOI 10.17487/RFC3711, , <https://www.rfc-editor.org/info/rfc3711>.
[RFC3760]
Gustafson, D., Just, M., and M. Nystrom, "Securely Available Credentials (SACRED) - Credential Server Framework", RFC 3760, DOI 10.17487/RFC3760, , <https://www.rfc-editor.org/info/rfc3760>.
[RFC4251]
Ylonen, T. and C. Lonvick, Ed., "The Secure Shell (SSH) Protocol Architecture", RFC 4251, DOI 10.17487/RFC4251, , <https://www.rfc-editor.org/info/rfc4251>.
[RFC4568]
Andreasen, F., Baugher, M., and D. Wing, "Session Description Protocol (SDP) Security Descriptions for Media Streams", RFC 4568, DOI 10.17487/RFC4568, , <https://www.rfc-editor.org/info/rfc4568>.
[RFC5479]
Wing, D., Ed., Fries, S., Tschofenig, H., and F. Audet, "Requirements and Analysis of Media Security Management Protocols", RFC 5479, DOI 10.17487/RFC5479, , <https://www.rfc-editor.org/info/rfc5479>.
[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, DOI 10.17487/RFC5763, , <https://www.rfc-editor.org/info/rfc5763>.
[RFC6189]
Zimmermann, P., Johnston, A., Ed., and J. Callas, "ZRTP: Media Path Key Agreement for Unicast Secure RTP", RFC 6189, DOI 10.17487/RFC6189, , <https://www.rfc-editor.org/info/rfc6189>.
[RFC6222]
Begen, A., Perkins, C., and D. Wing, "Guidelines for Choosing RTP Control Protocol (RTCP) Canonical Names (CNAMEs)", RFC 6222, DOI 10.17487/RFC6222, , <https://www.rfc-editor.org/info/rfc6222>.
[RFC6347]
Rescorla, E. and N. Modadugu, "Datagram Transport Layer Security Version 1.2", RFC 6347, DOI 10.17487/RFC6347, , <https://www.rfc-editor.org/info/rfc6347>.
[RFC6454]
Barth, A., "The Web Origin Concept", RFC 6454, DOI 10.17487/RFC6454, , <https://www.rfc-editor.org/info/rfc6454>.
[RFC6455]
Fette, I. and A. Melnikov, "The WebSocket Protocol", RFC 6455, DOI 10.17487/RFC6455, , <https://www.rfc-editor.org/info/rfc6455>.
[RFC6749]
Hardt, D., Ed., "The OAuth 2.0 Authorization Framework", RFC 6749, DOI 10.17487/RFC6749, , <https://www.rfc-editor.org/info/rfc6749>.
[RFC7022]
Begen, A., Perkins, C., Wing, D., and E. Rescorla, "Guidelines for Choosing RTP Control Protocol (RTCP) Canonical Names (CNAMEs)", RFC 7022, DOI 10.17487/RFC7022, , <https://www.rfc-editor.org/info/rfc7022>.
[RFC7033]
Jones, P., Salgueiro, G., Jones, M., and J. Smarr, "WebFinger", RFC 7033, DOI 10.17487/RFC7033, , <https://www.rfc-editor.org/info/rfc7033>.
[RFC7675]
Perumal, M., Wing, D., Ravindranath, R., Reddy, T., and M. Thomson, "Session Traversal Utilities for NAT (STUN) Usage for Consent Freshness", RFC 7675, DOI 10.17487/RFC7675, , <https://www.rfc-editor.org/info/rfc7675>.
[RFC8445]
Keranen, A., Holmberg, C., and J. Rosenberg, "Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal", RFC 8445, DOI 10.17487/RFC8445, , <https://www.rfc-editor.org/info/rfc8445>.
[RFC8825]
Alvestrand, H., "Overview: Real-Time Protocols for Browser-Based Applications", RFC 8825, DOI 10.17487/RFC8825, , <https://www.rfc-editor.org/info/rfc8825>.
[RFC8827]
Rescorla, E., "WebRTC Security Architecture", RFC 8827, DOI 10.17487/RFC8827, , <https://www.rfc-editor.org/info/rfc8827>.
[RFC8828]
Uberti, J., "WebRTC IP Address Handling Requirements", RFC 8828, DOI 10.17487/RFC8828, , <https://www.rfc-editor.org/info/rfc8828>.
[SWF]
"SWF File Format Specification Version 19", , <http://www.adobe.com/content/dam/Adobe/en/devnet/swf/pdf/swf_file_format_spec_v10.pdf>.
[whitten-johnny]
Whitten, A. and J.D. Tygar, "Why Johnny Can't Encrypt: A Usability Evaluation of PGP 5.0", Proceedings of the 8th USENIX Security Symposium, , <https://www.usenix.org/legacy/publications/library/proceedings/sec99/whitten.html>.
[XmlHttpRequest]
van Kesteren, A., "XMLHttpRequest Level 2", , <https://www.w3.org/TR/XMLHttpRequest/>.

Acknowledgements

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

Author's Address

Eric Rescorla
RTFM, Inc.
2064 Edgewood Drive
Palo Alto, CA 94303
United States of America