RTC-Web                                                      E. Rescorla
Internet-Draft                                                RTFM, Inc.
Intended status:  Standards Track                           June 5, 2011
Expires:  December 7, 2011

                  Security Considerations for RTC-Web


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



Status of this Memo

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   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 December 7, 2011.

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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 RTC-Web Applications  . . . . . . . . . . . . . .  7
     4.1.  Access to Local Devices  . . . . . . . . . . . . . . . . .  7
     4.2.  Communications Consent Verification  . . . . . . . . . . .  9
       4.2.1.  ICE  . . . . . . . . . . . . . . . . . . . . . . . . .  9
       4.2.2.  Masking  . . . . . . . . . . . . . . . . . . . . . . . 10
       4.2.3.  Backward Compatibility . . . . . . . . . . . . . . . . 10
     4.3.  Communications Security  . . . . . . . . . . . . . . . . . 10
       4.3.1.  Protecting Against Retrospective Compromise  . . . . . 11
       4.3.2.  Protecting Against During-Call Attack  . . . . . . . . 12  Key Continuity . . . . . . . . . . . . . . . . . . 12  Short Authentication Strings . . . . . . . . . . . 13
   5.  Security Considerations  . . . . . . . . . . . . . . . . . . . 14
   6.  Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 14
   7.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 14
     7.1.  Normative References . . . . . . . . . . . . . . . . . . . 14
     7.2.  Informative References . . . . . . . . . . . . . . . . . . 14
   Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 16

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1.  Introduction

   The Real-Time Communications on the Web (RTC-Web) working group is
   tasked with standardizing protocols for real-time communications
   between Web browsers.  The two major use cases for RTC-Web technology
   are real-time audio and/or video calls and direct data transfer.
   Unlike most conventional real-time systems, (e.g., SIP-based[RFC3261]
   soft phones) RTC-Web communications are directly controlled by some
   Web server.  A simple case is shown below.

                               |                |
                               |   Web Server   |
                               |                |
                                   ^        ^
                                  /          \
                          HTTP   /            \   HTTP
                                /              \
                               /                \
                              v                  v
                           JS API              JS API
                     +-----------+            +-----------+
                     |           |    Media   |           |
                     |  Browser  |<---------->|  Browser  |
                     |           |            |           |
                     +-----------+            +-----------+

                     Figure 1: A simple RTC-Web system

   In the system shown in Figure 1, Alice and Bob both have RTC-Web
   enabled browsers and they visit some Web server which operates a
   calling service.  Each of their browsers exposes standardized
   JavaScript calling APIs which are used by the Web server to set up a
   call between Alice and Bob. While this system is topologically
   similar to a conventional SIP-based system (with the Web server
   acting as the signaling service and browsers acting as softphones),
   control has moved to the central Web server; the browser simply
   provides API points that are used by the calling service.  As with
   any Web application, the Web server can move logic between the server
   and JavaScript in the browser, but regardless of where the code is
   executing, it is ultimately under control of the server.

   It should be immediately apparent that this type of system poses new
   security challenges beyond those of a conventional VoIP system.  In
   particular, it needs to contend with malicious calling services.  For
   example, if the calling service can cause the browser to make a call
   at any time to any callee of its choice, then this facility can be

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   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.

2.  Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   document are to be interpreted as described in RFC 2119 [RFC2119].

3.  The Browser Threat Model

   The security requirements for RTC-Web follow directly from the
   requirement that the browser's job is to protect the user.  Huang et
   al. [huang-w2sp] summarize the core browser security guarantee as:

      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 RTC-Web functionality must be
   designed with the intent to meet this standard.  The remainder of
   this section provides more background on the existing Web security

   In this model, then, the browser acts as a TRUSTED COMPUTING BASE
   (TCB) both from the user's perspective and to some extent from the
   server's.  While HTML and JS provided by the server can cause the
   browser to execute a variety of actions, those scripts operate in a
   sandbox that isolates them both from the user's computer and from
   each other, as detailed below.

   Conventionally, we refer to either WEB ATTACKERS, who are able to
   induce you to visit their sites but do not control the network, and
   NETWORK ATTACKERS, who are able to control your network.  Network
   attackers correspond to the [RFC3552] "Internet Threat Model".  In

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   general, it is desirable to build a system which is secure against
   both kinds of attackers, but realistically many sites do not run
   HTTPS [RFC2818] and so our ability to defend against network
   attackers is necessarily somewhat limited.  Most of the rest of this
   section is devoted to web attackers, with the assumption that
   protection against network attackers is provided by running HTTPS.

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

   Similarly, while Flash SWFs can access the camera and microphone,
   they explicitly require that the user consent to that access.  In
   addition, some resources simply cannot be accessed from the browser
   at all.  For instance, there is no real way to run specific
   executables directly from a script (though the user can of course be
   induced to download executable files and run them).

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.[I-D.abarth-origin] (although CORS [CORS] and WebSockets
   [I-D.ietf-hybi-thewebsocketprotocol] provides a escape hatch from
   this restriction, as described below.  This SAME ORIGIN POLICY (SOP)
   prevents server A from mounting attacks on server B via the user's
   browser, which protects both the user (e.g., from misuse of his
   credentials) and the server (e.g., from DoS attack).

   More generally, SOP forces scripts from each site to run in their
   own, isolated, sandboxes.  While there are techniques to allow them
   to interact, those interactions generally must be mutually consensual
   (by each site) and are limited to certain channels.  For instance,
   multiple pages/browser panes from the same origin can read each
   other's JS variables, but pages from the different origins--or even
   iframes from different origins on the same page--cannot.

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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

   While CORS is designed to allow cross-origin HTTP requests,
   WebSockets [I-D.ietf-hybi-thewebsocketprotocol] allows cross-origin
   establishment of transparent channels.  Once a WebSockets connection
   has been established from a script to a site, the script can exchange
   any traffic it likes without being required to frame it as a series
   of HTTP request/response transactions.  As with CORS, a WebSockets
   transaction starts with a consent verification stage to avoid
   allowing scripts to simply send arbitrary data to another origin.

   While consent verification is conceptually simple--just do a
   handshake before you start exchanging the real data--experience has
   shown that designing a correct consent verification system is
   difficult.  In particular, Huang et al. [huang-w2sp] have shown
   vulnerabilities in the existing Java and Flash consent verification
   techniques and in a simplified version of the WebSockets handshake.
   In particular, it is important to be wary of CROSS-PROTOCOL attacks
   in which the attacking script generates traffic which is acceptable
   to some non-Web protocol state machine.  In order to resist this form
   of attack, WebSockets incorporates a masking technique intended to
   randomize the bits on the wire, thus making it more difficult to
   generate traffic which resembles a given protocol.

4.  Security for RTC-Web 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 location without user consent.  This
   immediately raises the question, however, of what should be the scope

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   of user consent.

   As discussed in Section 3.2, the basic unit of Web sandboxing is the
   origin, and so it is natural to scope consent to origin.
   Specifically, a script from origin A MUST only be allowed to initiate
   communications (and hence to access camera and microphone) if the
   user has specifically authorized access for that origin.  It is of
   course technically possible to have coarser-scoped permissions, but
   because the Web model is scoped to origin, this creates a difficult

   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.  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.

   Unfortunately, none of these approaches is really satisfactory.
   Individual consent puts the user's approval in the UI flow for every
   call.  Not only does this quickly become annoying but it rapidly
   trains the user to simply click "OK", at which point the consent
   becomes useless.

   The other two options are designed to restrict calls to a given
   target.  Unfortunately, Callee-oriented consent does not work because
   the malicious site can claim that the user is calling any user of his
   choice.  The fix for this is to tie calls to a specific set of
   cryptographic keying material, but that breaks any portability for
   the callee's client, and is thus problematic.  (Section

   While this is primarily a question not for IETF, it should be clear
   that there is no really good answer.  In general, if you cannot trust
   the site which you have authorized for calling not to bug you then
   your security situation is not really ideal.  It is RECOMMENDED that
   browsers have explicit (and obvious) indicators that they are in a

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   call in order to mitigate this risk.

   The above recommendations provide security against web attackers.
   However, if a legitimate site is fetched over HTTP rather than HTTPS,
   a network attacker can inject code to initiate calls as if it were
   that origin, thus bypassing origin restrictions.  Note that this form
   of attack is also possible if a 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].  Therefore, it is RECOMMENDED that sites which
   embed RTC-Web functionality serve that functionality only over HTTPS
   and that browsers disallow execution of calling functionality in
   origins which contain 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-

4.2.  Communications Consent Verification

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

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 it from forging
   responses).  In order to achieve this objective with ICE, the STUN
   transaction IDs must be generated by the browser and MUST NOT be made
   available to the initiating script, even via a diagnostic interface.

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4.2.2.  Masking

   Once consent is verified, there still is some concern about
   misinterpretation attacks as described by Huang et al.[huang-w2sp].
   As long as communication is limited to UDP, then this risk is
   probably limited, thus masking is not required for UDP.  However,
   with TCP the risk of transparent proxies becomes much more severe.
   If TCP is to be used, then WebSockets style masking MUST be employed.

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, but it might be possible to merely require a one-sided
   check where the legacy client was a STUN responder.  It's unclear
   whether that is in fact simpler than doing ICE-Lite.

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

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

   This limitation does not mean that preventing attack by the calling
   service is completely hopeless.  However, we need to distinguish

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   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.

   During-call attack by calling service.
      The calling service is compromised during the call it wishes to

   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
   RTC-Web (this is a good idea for any communications security system)
   and this mechanism SHOULD provide perfect forward secrecy (PFS).  The
   signaling channel/calling service can be used to authenticate this

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

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4.3.2.  Protecting Against During-Call Attack

   Protecting against attacks during a call is a more difficult
   proposition.  Even if the calling service cannot directly access
   keying material (as recommended in the previous section), it can
   simply mount a man-in-the-middle attack on the connection, telling
   Alice that she is calling Bob and Bob that he is calling Alice, while
   in fact the calling service is acting as a calling bridge and
   capturing all the traffic.  While in theory it is possible to
   construct techniques which protect against this form of attack, in
   practice these techniques all require far too much user intervention
   to be practical, given the user interface constraints described in
   [abarth-rtcweb].  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 RTC-Web context.  First, much of the virtue of RTC-Web
   (and any Web application) is that it is not bound to particular piece
   of client software.  Thus, it will be not only possible but routine
   for a user to use multiple browsers on different computers which will
   of course have different keying material (SACRED [RFC3760]
   notwithstanding.)  Thus, users will frequently be alerted to key
   mismatches which are in fact completely legitimate, with the result
   that they are trained to simply click through them.  As it is known
   that users routinely will click through far more dire warnings
   [cranor-wolf], it seems extremely unlikely that any key continuity
   mechanism will be effective rather than simply annoying.

   Moreover, it is trivial to bypass even this kind of mechanism.
   Recall that unlike the case of SSH, the browser never directly gets
   the peer's identity from the user.  Rather, it is provided by the
   calling service.  Even enabling a mechanism of this type would
   require an API to allow the calling service to tell the browser "this
   is a call to user X".  All the calling service needs to do to avoid
   triggering a key continuity warning is to tell the browser that "this
   is a call to user Y" where Y is close to X. Even if the user actually
   checks the other side's name (which all available evidence indicates
   is unlikely), this would require (a) the browser to trusted UI to

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   provide the name and (b) the user to not be fooled by similar
   appearing names.  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
   read over the voice channel and if confirmed by both sides precludes
   MITM attack.  The intention is that the SAS is used once and then key
   continuity (though a different mechanism from that discussed above)
   is used thereafter.

   Unfortunately, the SAS does not offer a practical solution to the
   problem of a compromised calling service.  "Voice conversion"
   systems, which modify voice from one speaker to make it sound like
   another, are an active area of research.  These systems are already
   good enough to fool both automatic recognition systems
   [farus-conversion] and humans [kain-conversion] in many cases, and
   are of course likely to improve in future, especially in an
   environment where the user just wants to get on with the phone call.
   Thus, even if SAS is effective today, it is likely not to be so for
   much longer.  Moreover, it is possible for an attacker who controls
   the browser to allow the SAS to succeed and then simulate call
   failure and reconnect, trusting that the user will not notice that
   the "no SAS" indicator has been set (which seems likely).

   Even were SAS secure if used, it seems exceedingly unlikely that
   users will actually use it.  As discussed above, the browser UI
   constraints preclude requiring the SAS exchange prior to completing
   the call and so it must be voluntary; at most the browser will
   provide some UI indicator that the SAS has not yet been checked.
   However, it it is well-known that when faced with optional mechanisms
   such as fingerprints, users simply do not check them [whitten-johnny]
   Thus, it is highly unlikely that users will ever perform the SAS

   Once uses have checked the SAS once, key continuity is required to
   avoid them needing to check it on every call.  However, this is
   problematic for reasons indicated in Section  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.

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5.  Security Considerations

   This entire document is about security.

6.  Acknowledgements

7.  References

7.1.  Normative References

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

7.2.  Informative References

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

              Barth, A., "The Web Origin Concept",
              draft-abarth-origin-09 (work in progress), November 2010.

              Fette, I., "The WebSocket protocol",
              draft-ietf-hybi-thewebsocketprotocol-07 (work in
              progress), April 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.

   [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.

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   [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,
              April 2011.

              Barth, A., "Prompting the user is security failure",  RTC-
              Web Workshop.

              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.

              Farrus, M., Erro, D., and J. Hernando, "Speaker
              Recognition Robustness to Voice Conversion".

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

              Huang, L-S., Chen, E., Barth, A., Rescorla, E., and C.
              Jackson, "Talking to Yourself for Fun and Profit",  W2SP,

              Kain, A. and M. Macon, "Design and Evaluation of a Voice
              Conversion Algorithm based on Spectral Envelope Mapping

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              and Residual Prediction",  Proceedings of ICASSP, May

              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.

Author's Address

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

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

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