RTC-Web                                                      E. Rescorla
Internet-Draft                                                RTFM, Inc.
Intended status:  Standards Track                     September 21, 2011
Expires:  March 24, 2012

                  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 major use cases for RTC-Web 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) 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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   and may be updated, replaced, or obsoleted by other documents at any

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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 March 24, 2012.

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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
       4.1.1.  Calling Scenarios and User Expectations  . . . . . . .  8  Dedicated Calling Services . . . . . . . . . . . .  8  Calling the Site You're On . . . . . . . . . . . .  8  Calling to an Ad Target  . . . . . . . . . . . . .  9
       4.1.2.  Origin-Based Security  . . . . . . . . . . . . . . . .  9
       4.1.3.  Security Properties of the Calling Page  . . . . . . . 11
     4.2.  Communications Consent Verification  . . . . . . . . . . . 12
       4.2.1.  ICE  . . . . . . . . . . . . . . . . . . . . . . . . . 12
       4.2.2.  Masking  . . . . . . . . . . . . . . . . . . . . . . . 12
       4.2.3.  Backward Compatibility . . . . . . . . . . . . . . . . 13
     4.3.  Communications Security  . . . . . . . . . . . . . . . . . 14
       4.3.1.  Protecting Against Retrospective Compromise  . . . . . 15
       4.3.2.  Protecting Against During-Call Attack  . . . . . . . . 15  Key Continuity . . . . . . . . . . . . . . . . . . 16  Short Authentication Strings . . . . . . . . . . . 16  Recommendations  . . . . . . . . . . . . . . . . . 17
   5.  Security Considerations  . . . . . . . . . . . . . . . . . . . 17
   6.  Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 18
   7.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 18
     7.1.  Normative References . . . . . . . . . . . . . . . . . . . 18
     7.2.  Informative References . . . . . . . . . . . . . . . . . . 18
   Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 20

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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 major use cases for RTC-Web 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) 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.

   For the rest of this discussion we assume that the user is somehow
   going to grant consent to some entity (e.g., a social networking
   site) to initiate a call on his behalf.  This consent may be limited
   to a single call or may be a general 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 request access, where the media is
   going, or both.  So, for instance, one might imagine that at the time
   access to camera and microphone is requested, the user is shown a
   dialog that says "site X has requested access to camera and
   microphone, yes or no" (though note that this type of in-flow
   interface violates one of the guidelines in Section Section 3).  The
   user's decision will of course be based on his opinion of Site X.
   However, as discussed below, this is a complicated concept.

4.1.1.  Calling Scenarios and User Expectations

   While a large number of possible calling scenarios are possible, the
   scenarios discussed in this section illustrate many of the
   difficulties of identifying the relevant scope of consent.  Dedicated Calling Services

   The first scenario we consider is a dedicated calling service.  In
   this case, the user has a relationship with a calling site and
   repeatedly makes calls on it.  It is likely that rather than having
   to give permission for each call that the user will want to give the
   calling service long-term access to the camera and microphone.  This
   is a natural fit for a long-term consent mechanism (e.g., installing
   an app store "application" to indicate permission for the calling
   service.)  A variant of the dedicated calling service is a gaming
   site (e.g., a poker site) which hosts a dedicated calling service to
   allow players to call each other.

   With any kind of service where the user may use the same service to
   talk to many different people, there is a question about whether the
   user can know who they are talking to.  In general, this is difficult
   as most of the user interface is presented by the calling site.
   However, communications security mechanisms can be used to give some
   assurance, as described in Section 4.3.2.  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

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   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.  Calling to an Ad Target

   In both of the previous cases, the user has a direct relationship
   (though perhaps a transient one) with the target of the call.
   Moreover, in both cases he is actually visiting the site of the
   person he is being asked to trust.  However, this is not always so.
   Consider the case where a user is a visiting a content site which
   hosts an advertisement with an invitation to call for more
   information.  When the user clicks the ad, they are connected with
   the advertiser or their agent.

   The relationships here are far more complicated:  the site the user
   is actually visiting has no direct relationship with the advertiser;
   they are just hosting ads from an ad network.  The user has no
   relationship with the ad network, but desires one with the
   advertiser, at least for long enough to learn about their products.
   At minimum, then, whatever consent dialog is shown needs to allow the
   user to have some idea of the organization that they are actually

   However, because the user also has some relationship with the hosting
   site, it is also arguable that the hosting site should be allowed to
   express an opinion (e.g., to be able to allow or forbid a call) since
   a bad experience with an advertiser reflect negatively on the hosting
   site [this idea was suggested by Adam Barth].  However, this
   obviously presents a privacy challenge, as sites which host
   advertisements often learn very little about whether individual users
   clicked through to the ads, or even which ads were presented.

4.1.2.  Origin-Based Security

   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

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   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
   necssary 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.  Unfortunately, Callee-oriented consent does 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
   advertising case described in Section, 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 tieing permissions to origin
   here, but to the media origin (and-or destination) rather than to the
   Web origin.

   Another case where media-level cryptographic identity makes sense is
   when a user really does not trust the calling site.  For instance, I

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   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.  However, this is also not that convenient an
   interface, since managing individual user permissions can be painful.

   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
   call in order to mitigate this risk.

4.1.3.  Security Properties of the Calling Page

   Origin-based security is intended to security 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
   popunder) 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 RTC-Web 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

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   infect the browser's notion of that origin semi-permanently.

   [[ OPEN ISSUE:  What recommendation should IETF make about (a)
   whether RTCWeb long-term consent should be available over HTTP pages
   and (b) How to handle origins where the consent is to an HTTPS URL
   but the page contains active mixed content? ]]

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.

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.  I.e., once
   communications consent has been verified, it is most likely safe to
   allow the implementation to send arbitrary UDP traffic to the chosen
   destination, provided that the STUN keepalives continue to succeed.
   However, with TCP the risk of transparent proxies becomes much more
   severe.  If TCP is to be used, then WebSockets style masking MUST be

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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 RTC-Web endpoint is allowed to send a
   limited number of RTP packets prior to receiving consent.  This
   allows a short window of attack.  In addition, some legacy endpoints
   do not support RTCP, so this is a much more expensive solution for
   such endpoints, for which it would likely be easier to implement ICE.
   For these two reasons, an RTCP-based approach does not seem to
   address the security issue satisfactorily.

   In the STUN approach, the RTC-Web 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 RTC-Web 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 RTC-Web 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.

   [TODO:  Write something about consent freshness and RTCP].

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   [[ OPEN ISSUE:  Exactly what should be the requirements here?
   Proposals include ICE all the time or ICE but with allowing one of
   these non-ICE things for legacy. ]]

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
   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

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   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.

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

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Internet-Draft              RTC-Web Security              September 2011  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
   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"

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   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.  Recommendations

   [[ OPEN ISSUE:  What are the best UI recommendations to make?
   Proposal:  take the text from [I-D.kaufman-rtcweb-security-ui]
   Section 2]]

   [[ OPEN ISSUE:  Exactly what combination of media security primitives
   should be specified and/or mandatory to implement?  In particular,
   should we allow DTLS-SRTP only, or both DTLS-SRTP and SDES.  Should
   we allow RTP for backward compatibility? ]]

5.  Security Considerations

   This entire document is about security.

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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. and A. Melnikov, "The WebSocket protocol",
              draft-ietf-hybi-thewebsocketprotocol-15 (work in
              progress), September 2011.

              Kaufman, M., "Client Security User Interface Requirements
              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.

   [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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