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Versions: (draft-rescorla-rtcweb-security) 00 01 02 03 04 05 06 07

RTC-Web                                                      E. Rescorla
Internet-Draft                                                RTFM, Inc.
Intended status:  Standards Track                          July 04, 2014
Expires:  January 5, 2015


                   Security Considerations for WebRTC
                     draft-ietf-rtcweb-security-07

Abstract

   The Real-Time Communications on the Web (RTCWEB) working group is
   tasked with standardizing protocols for real-time communications
   between Web browsers, generally called "WebRTC".  The major use cases
   for WebRTC technology are real-time audio and/or video calls, Web
   conferencing, and direct data transfer.  Unlike most conventional
   real-time systems (e.g., SIP-based soft phones) WebRTC communications
   are directly controlled by a Web server, which poses new security
   challenges.  For instance, a Web browser might expose a JavaScript
   API which allows a server to place a video call.  Unrestricted access
   to such an API would allow any site which a user visited to "bug" a
   user's computer, capturing any activity which passed in front of
   their camera.  This document defines the WebRTC threat model and
   analyzes the security threats of WebRTC in that model.

Status of this Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on January 5, 2015.

Copyright Notice

   Copyright (c) 2014 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal



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   Provisions Relating to IETF Documents
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   it for publication as an RFC or to translate it into languages other
   than English.































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Table of Contents

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











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1.  Introduction

   The Real-Time Communications on the Web (RTCWEB) working group is
   tasked with standardizing protocols for real-time communications
   between Web browsers, generally called "WebRTC"
   [I-D.ietf-rtcweb-overview].  The major use cases for WebTC 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  |
                     |           |            |           |
                     +-----------+            +-----------+

                     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 which operates a
   calling service.  Each of their browsers exposes standardized
   JavaScript calling APIs (implementated as browser built-ins) which
   are used by the Web server to set up a call between Alice and Bob.
   The Web server also serves as the signaling channel to transport
   control messages between the browsers.  While this system is
   topologically similar to a conventional SIP-based system (with the
   Web server acting as the signaling service and browsers acting as
   softphones), control has moved to the central Web server; the browser
   simply provides API points that are used by the calling service.  As
   with any Web application, the Web server can move logic between the
   server and JavaScript in the browser, but regardless of where the
   code is executing, it is ultimately under control of the server.

   It should be immediately apparent that this type of system poses new
   security challenges beyond those of a conventional VoIP system.  In



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   particular, it needs to contend with malicious calling services.  For
   example, if the calling service can cause the browser to make a call
   at any time to any callee of its choice, then this facility can be
   used to bug a user's computer without their knowledge, simply by
   placing a call to some recording service.  More subtly, if the
   exposed APIs allow the server to instruct the browser to send
   arbitrary content, then they can be used to bypass firewalls or mount
   denial of service attacks.  Any successful system will need to be
   resistant to this and other attacks.

   A companion document [I-D.ietf-rtcweb-security-arch] 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", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in RFC 2119 [RFC2119].


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:

      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 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



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   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 for HTTP traffic, a network attacker is also a Web attacker,
   since it can inject traffic as if it were any non-HTTPS Web site.
   Thus, 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 screw yourself", as
   extensive research 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 those requests are not allowed to be made to any
   server, but rather solely to the same ORIGIN from whence the script
   came xref target="RFC6454"/> (although CORS [CORS] and WebSockets
   [RFC6455] provide a escape hatch from this restriction, as described
   below.)  This SAME ORIGIN POLICY (SOP) prevents server A from
   mounting attacks on server B via the user's browser, which protects
   both the user (e.g., from misuse of his credentials) and the server B
   (e.g., from DoS attack).

   More generally, SOP forces scripts from each site to run in their
   own, isolated, sandboxes.  While there are techniques to allow them



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   to interact, those interactions generally must be mutually consensual
   (by each site) and are limited to certain channels.  For instance,
   multiple pages/browser panes from the same origin can read each
   other's JS variables, but pages from the different origins--or even
   iframes from different origins on the same page--cannot.

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 Cross-Origin Resource Sharing (CORS) spec [CORS] is a
   response to this demand.  In CORS, when a script from origin A
   executes what would otherwise be a forbidden cross-origin request,
   the browser instead contacts the target server to determine whether
   it is willing to allow cross-origin requests from A. If it is so
   willing, the browser then allows the request.  This consent
   verification process is designed to safely allow cross-origin
   requests.

   While CORS is designed to allow cross-origin HTTP requests,
   WebSockets [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






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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:

      You are sending your media to entity A because you want to talk to
      Entity A (e.g., your mother).
      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, identity 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) and similarly we need to be concerned with the site
   accessing the user's camera and microphone even if the data is to be
   sent back to the site via conventional HTTP-based network mechanisms
   such as HTTP POST.





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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
   for a full analysis.)

   The most obvious threats are simply those of "oversharing".  I.e.,
   the user may believe they are sharing a window when in fact they are
   sharing an application, or may forget they are sharing their whole
   screen, icons, notifications, and all.  This is already an issue with
   existing screen sharing technologies and is made somewhat worse if a
   partially trusted site is responsible for asking for the resource to
   be shared rather than having the user propose it.

   A less obvious threat involves the impact of screen sharing on the
   Web security model.  A key part of the Same Origin Policy is that
   HTML or JS from site A can reference content from site B and cause
   the browser to load it, but (unless explicitly permitted) cannot see
   the result.  However, if a web application from a site is screen
   sharing the browser, then this violates that invariant, with serious
   security consequences.  For example, an attacker site might request
   screen sharing and then briefly open up a new Window to the user's
   bank or webmail account, using screen sharing to read the resulting
   displayed content.  A more sophisticated attack would be open up a
   source view window to a site and use the screen sharing result to
   view anti cross-site request forgery tokens.

   These threats suggest that screen/application sharing might need a
   higher level of user consent than access to the camera or microphone.

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 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



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   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 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 seen another use case, 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 origin.
   Specifically, a script from origin A MUST only be allowed to initiate
   communications (and hence to access camera and microphone) if the
   user has specifically authorized access for that origin.  It is of
   course technically possible to have coarser-scoped permissions, but



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   because the Web model is scoped to origin, this creates a difficult
   mismatch.

   Arguably, origin is not fine-grained enough.  Consider the situation
   where Alice visits a site and authorizes it to make a single call.
   If consent is expressed solely in terms of origin, then at any future
   visit to that site (including one induced via mash-up or ad network),
   the site can bug Alice's computer, use the computer to place bogus
   calls, etc.  While in principle Alice could grant and then revoke the
   privilege, in practice privileges accumulate; if we are concerned
   about this attack, something else is needed.  There are a number of
   potential countermeasures to this sort of issue.

   Individual Consent
      Ask the user for permission for each call.

   Callee-oriented Consent
      Only allow calls to a given user.

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

   Unfortunately, none of these approaches is satisfactory for all
   cases.  As discussed above, individual consent puts the user's
   approval in the UI flow for every call.  Not only does this quickly
   become annoying but it can train the user to simply click "OK", at
   which point the consent becomes useless.  Thus, while it may be
   necessary to have individual consent in some case, this is not a
   suitable solution for (for instance) the calling service case.  Where
   necessary, in-flow user interfaces must be carefully designed to
   avoid the risk of the user blindly clicking through.

   The other two options are designed to restrict calls to a given
   target.  Callee-oriented consent provided by the calling site not
   work well because a malicious site can claim that the user is calling
   any user of his choice.  One fix for this is to tie calls to a
   cryptographically established identity.  While not suitable for all
   cases, this approach may be useful for some.  If we consider the case
   of advertising, it's not particularly convenient to require the
   advertiser to instantiate an iframe on the hosting site just to get
   permission; a more convenient approach is to cryptographically tie
   the advertiser's certificate to the communication directly.  We're
   still tying permissions to origin here, but to the media origin
   (and-or destination) rather than to the Web origin.
   [I-D.ietf-rtcweb-security-arch] describes mechanisms which facilitate
   this sort of consent.




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   Another case where media-level cryptographic identity makes sense is
   when a user really does not trust the calling site.  For instance, I
   might be worried that the calling service will attempt to bug my
   computer, but I also want to be able to conveniently call my friends.
   If consent is tied to particular communications endpoints, then my
   risk is limited.  Naturally, it is somewhat challenging to design UI
   primitives which express this sort of policy.  The problem becomes
   even more challenging in multi-user calling cases.

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 apparently
       http://calling-service.example.com/ to inject JS to initiate a
       call to himself.

   Note that this attack does not depend on the media being insecure.
   Because the call is to the attacker, it is also encrypted to him.
   Moreover, it need not be executed immediately; the attacker can
   "infect" the origin semi-permanently (e.g., with a web worker or a
   popped-up window that is hidden under the main window.) and thus be
   able to bug me long after I have left the infected network.  This
   risk is created by allowing calls at all from a page fetched over
   HTTP.

   Even if calls are only possible from HTTPS sites, if the site embeds
   active content (e.g., JavaScript) that is fetched over HTTP or from
   an untrusted site, because that JavaScript is executed in the
   security context of the page [finer-grained].  Thus, it is also
   dangerous to allow WebRTC functionality from HTTPS origins that embed
   mixed content.  Note:  this issue is not restricted to PAGES which
   contain mixed content.  If a page from a given origin ever loads
   mixed content then it is possible for a network attacker to infect
   the browser's notion of that origin semi-permanently.





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4.2.  Communications Consent Verification

   As discussed in Section 3.3, allowing web applications unrestricted
   network access via the browser introduces the risk of using the
   browser as an attack platform against machines which would not
   otherwise be accessible to the malicious site, for instance because
   they are topologically restricted (e.g., behind a firewall or NAT).
   In order to prevent this form of attack as well as cross-protocol
   attacks it is important to require that the target of traffic
   explicitly consent to receiving the traffic in question.  Until that
   consent has been verified for a given endpoint, traffic other than
   the consent handshake MUST NOT be sent to that endpoint.

   Note that consent verification is not sufficient to prevent overuse
   of network resources.  Because WebRTC allows for a Web site to create
   data flows between two browser instances without user consent, it is
   possible for a malicious site to chew up a signficant 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.  ICE [RFC5245] includes a handshake designed to verify that
   the receiving element wishes to receive traffic from the sender.  It
   is important to remember here that the site initiating ICE is
   presumed malicious; in order for the handshake to be secure the
   receiving element MUST demonstrate receipt/knowledge of some value
   not available to the site (thus preventing the site from forging
   responses).  In order to achieve this objective with ICE, the STUN
   transaction IDs must be generated by the browser and MUST NOT be made
   available to the initiating script, even via a diagnostic interface.
   Verifying receiver consent also requires verifying the receiver wants
   to receive traffic from a particular sender, and at this time; for
   example a malicious site may simply attempt ICE to known servers that
   are using ICE for other sessions.  ICE provides this verification as
   well, by using the STUN credentials as a form of per-session shared
   secret.  Those credentials are known to the Web application, but
   would need to also be known and used by the STUN-receiving element to
   be useful.

   There also needs to be some mechanism for the browser to verify that



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   the target of the traffic continues to wish to receive it.  Because
   ICE keepalives are indications, they will not work here.
   [I-D.ietf-rtcweb-stun-consent-freshness] 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].
   Where TCP is used the risk is substantial due to the potential
   presence of transparent proxies and therefore if TCP is to be used,
   then WebSockets style masking MUST be employed.

   Since DTLS (with the anti-chosen plaintext mechanisms required by TLS
   1.1) does not allow the attacker to generate predictable ciphertext,
   there is no need for masking of protocols running over DTLS (e.g.
   SCTP over DTLS, UDP over DTLS, etc.).

   Note that in principle an attacker could exert some control over SRTP
   packets by using a combination of the WebAudio API and extremely
   tight timing control.  The primary risk here seems to be carriage of
   SRTP over TURN TCP.  However, as SRTP packets have an extremely
   characteristic packet header it seems unlikely that any but the most
   aggressive intermediaries would be confused into thinking that
   another application layer protocol was in use.

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  Use or 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



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   such endpoints, for which it would likely be easier to implement ICE.
   For these two reasons, an RTCP-based approach does not seem to
   address the security issue satisfactorily.

   In the STUN approach, the WebRTC endpoint is able to verify that the
   recipient is running some kind of STUN endpoint but unless the STUN
   responder is integrated with the ICE username/password establishment
   system, the WebRTC endpoint cannot verify that the recipient consents
   to this particular call.  This may be an issue if existing STUN
   servers are operated at addresses that are not able to handle
   bandwidth-based attacks.  Thus, this approach does not seem
   satisfactory either.

   If the systems are tightly integrated (i.e., the STUN endpoint
   responds with responses authenticated with ICE credentials) then this
   issue does not exist.  However, such a design is very close to an
   ICE-Lite implementation (indeed, arguably is one).  An intermediate
   approach would be to have a STUN extension that indicated that one
   was responding to WebRTC checks but not computing integrity checks
   based on the ICE credentials.  This would allow the use of standalone
   STUN servers without the risk of confusing them with legacy STUN
   servers.  If a non-ICE legacy solution is needed, then this is
   probably the best choice.

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

   In ordinary operation, the site learns the browser's IP address,
   though it may be hidden via mechanisms like Tor
   [http://www.torproject.org] or a VPN.  However, because sites can
   cause the browser to provide IP addresses, this provides a mechanism
   for sites to learn about the user's network environment even if the
   user is behind a VPN that masks their IP address.  Implementations



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   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.

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 [RFC4347] and DTLS-SRTP
   [RFC5763]) is well understood.  However, we must examine this
   technology to the WebRTC context, where the threat model is somewhat
   different.

   In general, it is important to understand that unlike a conventional
   SIP proxy, the calling service (i.e., the Web server) controls not
   only the channel between the communicating endpoints but also the
   application running on the user's browser.  While in principle it is
   possible for the browser to cut the calling service out of the loop
   and directly present trusted information (and perhaps get consent),
   practice in modern browsers is to avoid this whenever possible.  "In-
   flow" modal dialogs which require the user to consent to specific
   actions are particularly disfavored as human factors research
   indicates that unless they are made extremely invasive, users simply
   agree to them without actually consciously giving consent.
   [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 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.







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      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
   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 that an attacker subsequently wants to recover
   the content of the call.  We assume that the attacker has access to
   the protected media stream as well as having full control of the
   calling service.

   If the calling service has access to the traffic keying material (as
   in SDES [RFC4568]), then retrospective attack is trivial.  This form
   of attack is particularly serious in the Web context because it is
   standard practice in Web services to run extensive logging and
   monitoring.  Thus, it is highly likely that if the traffic key is
   part of any HTTP request it will be logged somewhere and thus subject
   to subsequent compromise.  It is this consideration that makes an
   automatic, public key-based key exchange mechanism imperative for
   WebRTC (this is a good idea for any communications security system)
   and this mechanism SHOULD provide perfect forward secrecy (PFS).  The
   signaling channel/calling service can be used to authenticate this
   mechanism.

   In addition, if end-to-end keying is in used, the system MUST NOT
   provide any APIs to extract either long-term keying material or to
   directly access any stored traffic keys.  Otherwise, an attacker who
   subsequently compromised the calling service might be able to use
   those APIs to recover the traffic keys and thus compromise the
   traffic.

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



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   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].

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 [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 particular piece
   of client software.  Thus, it will be not only possible but routine
   for a user to use multiple browsers on different computers which will
   of course have different keying material (SACRED [RFC3760]
   notwithstanding.)  Thus, users will frequently be alerted to key
   mismatches which are in fact completely legitimate, with the result
   that they are trained to simply click through them.  As it is known
   that users routinely will click through far more dire warnings
   [cranor-wolf], it seems extremely unlikely that any key continuity
   mechanism will be effective rather than simply annoying.

   Moreover, it is trivial to bypass even this kind of mechanism.
   Recall that unlike the case of SSH, the browser never directly gets
   the peer's identity from the user.  Rather, it is provided by the
   calling service.  Even enabling a mechanism of this type would
   require an API to allow the calling service to tell the browser "this
   is a call to user X".  All the calling service needs to do to avoid
   triggering a key continuity warning is to tell the browser that "this
   is a call to user Y" where Y is close to X. Even if the user actually
   checks the other side's name (which all available evidence indicates
   is unlikely), this would require (a) the browser to trusted UI to
   provide the name and (b) the user to not be fooled by similar
   appearing names.

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



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   compared by the users (e.g., read aloud over the 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
   [farus-conversion] and humans [kain-conversion] in many cases, and
   are of course likely to improve in future, especially in an
   environment where the user just wants to get on with the phone call.
   Thus, even if SAS is effective today, it is likely not to be so for
   much longer.

   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 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, OpenID, WebFinger), has recently been developed and use Web
   technologies to provide lightweight (from the user's perspective)
   third-party authenticated transactions.  It is possible to use
   systems of this type to authenticate WebRTC calls, linking them to



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   existing user notions of identity (e.g., Facebook adjacencies).
   Specifically, the third-party identity system is used to bind the
   user's identity to cryptographic keying material which is then used
   to authenticate the calling endpoints.  Calls which are authenticated
   in this fashion are naturally resistant even to active MITM attack by
   the calling site.

   Note that there is one special case in which PKI-style certificates
   do provide a practical solution:  calls from end-users to large
   sites.  For instance, if you are making a call to Amazon.com, then
   Amazon can easily get a certificate to authenticate their media
   traffic, just as they get one to authenticate their Web traffic.
   This does not provide additional security value in cases in which the
   calling site and the media peer are one in the same, but might be
   useful in cases in which third parties (e.g., ad networks or
   retailers) arrange for calls but do not participate in them.

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., I must be able to verify that
   the person I am calling has engaged a secure media mode.  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.

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, etc.)
   do exist to attempt to address these issues, they are generally not
   compatible with open systems and WebRTC does not address them.




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   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 [I-D.ietf-avtcore-6222bis]).  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.

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.


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.  Changes Since -04

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




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   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.


8.  References

8.1.  Normative References

   [I-D.ietf-rtcweb-overview]
              Alvestrand, H., "Overview: Real Time Protocols for
              Browser-based Applications", draft-ietf-rtcweb-overview-10
              (work in progress), June 2014.

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.

8.2.  Informative References

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

   [I-D.ietf-avtcore-6222bis]
              Begen, A., Perkins, C., Wing, D., and E. Rescorla,
              "Guidelines for Choosing RTP Control Protocol (RTCP)
              Canonical Names (CNAMEs)", draft-ietf-avtcore-6222bis-06
              (work in progress), July 2013.

   [I-D.ietf-rtcweb-security-arch]
              Rescorla, E., "WebRTC Security Architecture",
              draft-ietf-rtcweb-security-arch-09 (work in progress),
              February 2014.

   [I-D.ietf-rtcweb-stun-consent-freshness]
              Perumal, M., Wing, D., R, R., Reddy, T., and M. Thomson,
              "STUN Usage for Consent Freshness",
              draft-ietf-rtcweb-stun-consent-freshness-04 (work in
              progress), June 2014.

   [I-D.kaufman-rtcweb-security-ui]
              Kaufman, M., "Client Security User Interface Requirements



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              for RTCWEB", draft-kaufman-rtcweb-security-ui-00 (work in
              progress), June 2011.

   [RFC2818]  Rescorla, E., "HTTP Over TLS", RFC 2818, May 2000.

   [RFC3261]  Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
              A., Peterson, J., Sparks, R., Handley, M., and E.
              Schooler, "SIP: Session Initiation Protocol", RFC 3261,
              June 2002.

   [RFC3552]  Rescorla, E. and B. Korver, "Guidelines for Writing RFC
              Text on Security Considerations", BCP 72, RFC 3552,
              July 2003.

   [RFC3711]  Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
              Norrman, "The Secure Real-time Transport Protocol (SRTP)",
              RFC 3711, March 2004.

   [RFC3760]  Gustafson, D., Just, M., and M. Nystrom, "Securely
              Available Credentials (SACRED) - Credential Server
              Framework", RFC 3760, April 2004.

   [RFC4251]  Ylonen, T. and C. Lonvick, "The Secure Shell (SSH)
              Protocol Architecture", RFC 4251, January 2006.

   [RFC4347]  Rescorla, E. and N. Modadugu, "Datagram Transport Layer
              Security", RFC 4347, April 2006.

   [RFC4568]  Andreasen, F., Baugher, M., and D. Wing, "Session
              Description Protocol (SDP) Security Descriptions for Media
              Streams", RFC 4568, July 2006.

   [RFC5245]  Rosenberg, J., "Interactive Connectivity Establishment
              (ICE): A Protocol for Network Address Translator (NAT)
              Traversal for Offer/Answer Protocols", RFC 5245,
              April 2010.

   [RFC5479]  Wing, D., Fries, S., Tschofenig, H., and F. Audet,
              "Requirements and Analysis of Media Security Management
              Protocols", RFC 5479, April 2009.

   [RFC5763]  Fischl, J., Tschofenig, H., and E. Rescorla, "Framework
              for Establishing a Secure Real-time Transport Protocol
              (SRTP) Security Context Using Datagram Transport Layer
              Security (DTLS)", RFC 5763, May 2010.

   [RFC6189]  Zimmermann, P., Johnston, A., and J. Callas, "ZRTP: Media
              Path Key Agreement for Unicast Secure RTP", RFC 6189,



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              April 2011.

   [RFC6222]  Begen, A., Perkins, C., and D. Wing, "Guidelines for
              Choosing RTP Control Protocol (RTCP) Canonical Names
              (CNAMEs)", RFC 6222, April 2011.

   [RFC6454]  Barth, A., "The Web Origin Concept", RFC 6454,
              December 2011.

   [RFC6455]  Fette, I. and A. Melnikov, "The WebSocket Protocol",
              RFC 6455, December 2011.

   [SWF]      Adobe, "SWF File Format Specification Version 19".

   [abarth-rtcweb]
              Barth, A., "Prompting the user is security failure",  RTC-
              Web Workshop.

   [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, 2009.

   [farus-conversion]
              Farrus, M., Erro, D., and J. Hernando, "Speaker
              Recognition Robustness to Voice Conversion".

   [finer-grained]
              Barth, A. and C. Jackson, "Beware of Finer-Grained
              Origins",  W2SP, 2008.

   [huang-w2sp]
              Huang, L-S., Chen, E., Barth, A., Rescorla, E., and C.
              Jackson, "Talking to Yourself for Fun and Profit",  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 ICASSP, May
              2001.

   [whitten-johnny]
              Whitten, A. and J. Tygar, "Why Johnny Can't Encrypt: A
              Usability Evaluation of PGP 5.0",  Proceedings of the 8th
              USENIX Security Symposium, 1999.




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Author's Address

   Eric Rescorla
   RTFM, Inc.
   2064 Edgewood Drive
   Palo Alto, CA  94303
   USA

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









































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