RTC-Web

Internet Engineering Task Force (IETF)                       E. Rescorla
Internet-Draft
Request for Comments: 8826                                    RTFM, Inc.
Intended status:
Category: Standards Track                            July 5, 2019
Expires: January 6,                                      June 2020
ISSN: 2070-1721

                   Security Considerations for WebRTC
                     draft-ietf-rtcweb-security-12

Abstract

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

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   https://www.rfc-editor.org/info/rfc8826.

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Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   3
   2.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . .   4
   3.  The Browser Threat Model  . . . . . . . . . . . . . . . . . .   4
     3.1.  Access to Local Resources . . . . . . . . . . . . . . . .   5
     3.2.  Same-Origin Policy  . . . . . . . . . . . . . . . . . . .   5
     3.3.  Bypassing SOP: CORS, WebSockets, and consent Consent to
           communicate . . . . . . . . . . . . . . . . . . . . . . .   6 Communicate
   4.  Security for WebRTC Applications  . . . . . . . . . . . . . .   7
     4.1.  Access to Local Devices . . . . . . . . . . . . . . . . .   7
       4.1.1.  Threats from Screen Sharing . . . . . . . . . . . . .   8
       4.1.2.  Calling Scenarios and User Expectations . . . . . . .   8
         4.1.2.1.  Dedicated Calling Services  . . . . . . . . . . .   9
         4.1.2.2.  Calling the Site You're On  . . . . . . . . . . .   9
       4.1.3.  Origin-Based Security . . . . . . . . . . . . . . . .  10
       4.1.4.  Security Properties of the Calling Page . . . . . . .  11
     4.2.  Communications Consent Verification . . . . . . . . . . .  12
       4.2.1.  ICE . . . . . . . . . . . . . . . . . . . . . . . . .  13
       4.2.2.  Masking . . . . . . . . . . . . . . . . . . . . . . .  13
       4.2.3.  Backward Compatibility  . . . . . . . . . . . . . . .  14
       4.2.4.  IP Location Privacy . . . . . . . . . . . . . . . . .  15
     4.3.  Communications Security . . . . . . . . . . . . . . . . .  15
       4.3.1.  Protecting Against against Retrospective Compromise . . . . .  16
       4.3.2.  Protecting Against against During-Call Attack . . . . . . . .  17
         4.3.2.1.  Key Continuity  . . . . . . . . . . . . . . . . .  17
         4.3.2.2.  Short Authentication Strings  . . . . . . . . . .  18
         4.3.2.3.  Third Party  Third-Party Identity  . . . . . . . . . . . . . .  19
         4.3.2.4.  Page Access to Media  . . . . . . . . . . . . . .  19
       4.3.3.  Malicious Peers . . . . . . . . . . . . . . . . . . .  20
     4.4.  Privacy Considerations  . . . . . . . . . . . . . . . . .  20
       4.4.1.  Correlation of Anonymous Calls  . . . . . . . . . . .  21
       4.4.2.  Browser Fingerprinting  . . . . . . . . . . . . . . .  21
   5.  Security Considerations . . . . . . . . . . . . . . . . . . .  21
   6.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  21
   7.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  21
   8.  Changes Since -04 . . . . . . . . . . . . . . . . . . . . . .  21
   9.
   7.  References  . . . . . . . . . . . . . . . . . . . . . . . . .  22
     9.1.
     7.1.  Normative References  . . . . . . . . . . . . . . . . . .  22
     9.2.
     7.2.  Informative References  . . . . . . . . . . . . . . . . .  22
   Acknowledgements
   Author's Address  . . . . . . . . . . . . . . . . . . . . . . . .  25

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" [I-D.ietf-rtcweb-overview]. [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, systems (e.g., SIP-based [RFC3261] soft
   phones) 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 Simple WebRTC system System

   In the system shown in Figure 1, Alice and Bob both have WebRTC-
   enabled browsers and they visit some Web server which 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 VoIP 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
   denial of service DoS attacks.  Any successful system will need to be
   resistant to this and other attacks.

   A companion document [I-D.ietf-rtcweb-security-arch] [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: as
   follows:

      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 Computing 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 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 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 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
   (Note: In many cases 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.] 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 CORS Cross-Origin
   Resource Sharing (CORS) [CORS] and WebSockets [RFC6455] provide an
   escape hatch from this restriction, as described below.) below).  This SAME 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). 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 pages / browser panes from the same origin can read each
   other's JS variables, but pages from the different origins--or origins -- or even
   iframes from different origins on the same page--cannot. page -- cannot.

3.3.  Bypassing SOP: CORS, WebSockets, and consent Consent to communicate 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) 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 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 simple -- just do a
   handshake before you start exchanging the real data--experience 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 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 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) 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, 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) and similarly
   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 http://lists.w3.org/Archives/Public/public-
   webrtc/2013Mar/0024.html <https://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.,  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 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 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 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 that 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.)
   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.) 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 at 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 case, 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 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
   (and/or destination) rather than to the Web origin.
   [I-D.ietf-rtcweb-security-arch]  [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 which 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 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. <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/. <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 <http://calling-service.example.com>.

   3.  The attacker forges the response from http://calling-
       service.example.com/ <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 Web worker or a
   popped-up window that is hidden under the main window.) 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 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.

4.2.  Communications Consent Verification

   As discussed in Section 3.3, allowing web 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, site -- for instance 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 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.

4.2.1.  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) [RFC8445]
   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 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 Session Traversal Utilities for
   NAT (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 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 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.  [RFC7675]
   describes the mechanism for providing consent freshness.

4.2.2.  Masking

   Once consent is verified, there still is some concern about
   misinterpretation attacks as described by Huang et al.[huang-w2sp]. al. [huang-w2sp].
   Where TCP is used used, the risk is substantial due to the potential
   presence of transparent proxies and therefore proxies; therefore, if TCP is to be used,
   then WebSockets style 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. (e.g.,
   SCTP over DTLS, UDP over DTLS, etc.).

   Note that in principle an attacker could exert some control over SRTP
   Secure Real-time Transport Protocol (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 Traversal
   Using Relays around NAT (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 application-layer protocol was in use.

4.2.3.  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:

   o

   *  STUN checks without using ICE (i.e., the non-RTC-web endpoint sets
      up a STUN responder.)

   o responder).

   *  Use of RTCP the RTP Control Protocol (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) 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.

4.2.4.  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 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]
   <https://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 [I-D.ietf-rtcweb-ip-handling] [RFC8828] for additional information.

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] [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"
   "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. 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. 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") attack").

   During-call attack by calling service. 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 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 [RFC4568]), then retrospective attack is trivial.  This form
   of attack is particularly serious in the Web 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) system),
   and this mechanism SHOULD provide perfect forward secrecy Perfect Forward Secrecy (PFS).  The
   signaling channel/calling 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.

4.3.2.  Protecting Against 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
   [I-D.ietf-rtcweb-security-arch]. [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.  SSH  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). 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 which
   that will of course have different keying material (SACRED (Securely
   Available Credentials (SACRED) [RFC3760]
   notwithstanding.) 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". 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" "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 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
   [farus-conversion]
   [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 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) 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  Third-Party Identity

   The conventional approach to providing communications identity has of
   course been to have some third party 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]), [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 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 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.  I.e.,  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 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 (DRM, (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) 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 CNAMEs
   (if generated according to [RFC6222] rather than the privacy 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
   [I-D.ietf-rtcweb-ip-handling]. [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.  Acknowledgements

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

7.  IANA Considerations

   There are

   This document has no IANA considerations.

8.  Changes Since -04

   o  Replaced RTCWEB and RTC-Web with WebRTC, except when referring to
      the IETF WG

   o  Removed discussion of the IFRAMEd advertisement case, since we
      decided not to treat it specially.

   o  Added a privacy section considerations section.

   o  Significant edits to the SAS section to reflect Alan Johnston's
      comments.

   o  Added some discussion if IP location privacy and Tor.

   o  Updated the "communications consent" section to reflrect draft-
      ietf.

   o  Added a section about "malicious peers".

   o  Added a section describing screen sharing threats.

   o  Assorted editorial changes.

9. actions.

7.  References

9.1.

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, March 1997,
              <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,
              May 2017, <https://www.rfc-editor.org/info/rfc8174>.

9.2.

7.2.  Informative References

   [abarth-rtcweb]
              Barth, A., "Prompting the user is security failure", RTC-
              Web Workshop, September 2010. 2010, <http://rtc-
              web.alvestrand.com/home/papers/barth-security-
              prompt.pdf?attredirects=0>.

   [CORS]     van Kesteren, A., "Cross-Origin Resource Sharing", January
              2014.

   [cranor-wolf]
              Sunshine, J., Egelman, S., Almuhimedi, H., Atri, N., and
              L. cranor, Cranor, "Crying Wolf: An Empirical Study of SSL Warning
              Effectiveness", Proceedings of the 18th USENIX Security
              Symposium, 2009, August 2009.

   [farus-conversion] 2009,
              <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", January 2008.

   [finer-grained]
              Barth, A. and C.
              Jackson, C. and A. Barth, "Beware of Finer-Grained
              Origins",  W2SP, 2008, Web 2.0 Security and Privacy (W2SP 2008), July
              2008.

   [Fingerprinting]
              "Fingerprinting Guidance for Web Specification Authors
              (Draft)", November 2013. 2013, <https://www.w3.org/TR/
              fingerprinting-guidance/#acknowledgement/>.

   [huang-w2sp]
              Huang, L-S., Chen, E., E.Y., Barth, A., Rescorla, E., and C.
              Jackson, "Talking to Yourself for Fun and Profit",  W2SP,
              2011, Web 2.0
              Security and Privacy (W2SP 2011), May 2011.

   [I-D.ietf-rtcweb-ip-handling]
              Uberti, J., "WebRTC IP Address Handling Requirements",
              draft-ietf-rtcweb-ip-handling-12 (work in progress), July
              2019.

   [I-D.ietf-rtcweb-overview]
              Alvestrand, H., "Overview: Real Time Protocols for
              Browser-based Applications", draft-ietf-rtcweb-overview-19
              (work in progress), November 2017.

   [I-D.ietf-rtcweb-security-arch]
              Rescorla, E., "WebRTC Security Architecture", draft-ietf-
              rtcweb-security-arch-18 (work in progress), February 2019.

   [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 ICASSP, the 2001 IEEE
              International Conference on Acoustics, Speech, and Signal
              Processing (ICASSP), DOI 10.1109/ICASSP.2001.941039, May
              2001, May 2001. <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", November 2014. 2014,
              <https://openid.net/specs/openid-connect-core-1_0.html>.

   [RFC2818]  Rescorla, E., "HTTP Over TLS", RFC 2818,
              DOI 10.17487/RFC2818, May 2000,
              <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, June 2002,
              <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, July 2003,
              <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, March 2004,
              <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, April 2004,
              <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,
              January 2006, <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, July 2006,
              <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, April 2009,
              <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, May
              2010, <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, April 2011,
              <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, April 2011,
              <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,
              January 2012, <https://www.rfc-editor.org/info/rfc6347>.

   [RFC6454]  Barth, A., "The Web Origin Concept", RFC 6454,
              DOI 10.17487/RFC6454, December 2011,
              <https://www.rfc-editor.org/info/rfc6454>.

   [RFC6455]  Fette, I. and A. Melnikov, "The WebSocket Protocol",
              RFC 6455, DOI 10.17487/RFC6455, December 2011,
              <https://www.rfc-editor.org/info/rfc6455>.

   [RFC6749]  Hardt, D., Ed., "The OAuth 2.0 Authorization Framework",
              RFC 6749, DOI 10.17487/RFC6749, October 2012,
              <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,
              September 2013, <https://www.rfc-editor.org/info/rfc7022>.

   [RFC7033]  Jones, P., Salgueiro, G., Jones, M., and J. Smarr,
              "WebFinger", RFC 7033, DOI 10.17487/RFC7033, September
              2013, <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,
              October 2015, <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, July 2018,
              <https://www.rfc-editor.org/info/rfc8445>.

   [RFC8825]  Alvestrand, H., "Overview: Real-Time Protocols for
              Browser-Based Applications", RFC 8825,
              DOI 10.17487/RFC8825, June 2020,
              <https://www.rfc-editor.org/info/rfc8825>.

   [RFC8827]  Rescorla, E., "WebRTC Security Architecture", RFC 8827,
              DOI 10.17487/RFC8827, June 2020,
              <https://www.rfc-editor.org/info/rfc8827>.

   [RFC8828]  Uberti, J., "WebRTC IP Address Handling Requirements",
              RFC 8828, DOI 10.17487/RFC8828, June 2020,
              <https://www.rfc-editor.org/info/rfc8828>.

   [SWF]      "SWF File Format Specification Version 19", April 2013. 2013,
              <http://www.adobe.com/content/dam/Adobe/en/devnet/swf/pdf/
              swf_file_format_spec_v10.pdf>.

   [whitten-johnny]
              Whitten, A. and J. J.D. Tygar, "Why Johnny Can't Encrypt: A
              Usability Evaluation of PGP 5.0", Proceedings of the 8th
              USENIX Security Symposium, 1999, August 1999. 1999,
              <https://www.usenix.org/legacy/publications/library/
              proceedings/sec99/whitten.html>.

   [XmlHttpRequest]
              van Kesteren, A., "XMLHttpRequesti "XMLHttpRequest Level 2", January 2012. 2012,
              <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
   USA
   United States of America

   Phone: +1 650 678 2350
   Email: ekr@rtfm.com