One document matched: draft-ietf-rtcweb-security-00.xml
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<rfc category="std" docName="draft-ietf-rtcweb-security-00"
ipr="pre5378Trust200902">
<front>
<title abbrev="RTC-Web Security">Security Considerations for RTC-Web</title>
<author fullname="Eric Rescorla" initials="E.K." surname="Rescorla">
<organization>RTFM, Inc.</organization>
<address>
<postal>
<street>2064 Edgewood Drive</street>
<city>Palo Alto</city>
<region>CA</region>
<code>94303</code>
<country>USA</country>
</postal>
<phone>+1 650 678 2350</phone>
<email>ekr@rtfm.com</email>
</address>
</author>
<date day="21" month="September" year="2011" />
<area>RAI</area>
<workgroup>RTC-Web</workgroup>
<abstract>
<t>
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.
</t>
</abstract>
<note title="Legal">
<t>THIS DOCUMENT AND THE INFORMATION CONTAINED THEREIN ARE PROVIDED ON
AN “AS IS” BASIS AND THE CONTRIBUTOR, THE ORGANIZATION
HE/SHE REPRESENTS OR IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY, THE
IETF TRUST, AND THE INTERNET ENGINEERING TASK FORCE, DISCLAIM ALL
WARRANTIES, EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO ANY
WARRANTY THAT THE USE OF THE INFORMATION THEREIN WILL NOT INFRINGE ANY
RIGHTS OR ANY IMPLIED WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A
PARTICULAR PURPOSE.</t>
</note>
</front>
<middle>
<section title="Introduction" anchor="sec.introduction">
<t>
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<xref target="RFC3261"></xref> soft phones) RTC-Web communications are directly controlled
by some Web server. A simple case is shown below.
</t>
<figure title="A simple RTC-Web system" anchor="fig.simple">
<artwork><![CDATA[
+----------------+
| |
| Web Server |
| |
+----------------+
^ ^
/ \
HTTP / \ HTTP
/ \
/ \
v v
JS API JS API
+-----------+ +-----------+
| | Media | |
| Browser |<---------->| Browser |
| | | |
+-----------+ +-----------+
]]></artwork>
</figure>
<t>
In the system shown in <xref target="fig.simple"/>, 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.
</t>
<t>
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 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.
</t>
</section>
<section anchor="sec-term" title="Terminology">
<t>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 <xref
target="RFC2119">RFC 2119</xref>.</t>
</section>
<section title="The Browser Threat Model" anchor="sec.web-security">
<t>
The security requirements for RTC-Web follow directly from the
requirement that the browser's job is to protect the user.
Huang et al. <xref target="huang-w2sp"/> summarize the core browser security guarantee as:
</t>
<t>
<list style="hanging">
<t>
Users can safely visit arbitrary web sites and execute scripts provided by those sites.
</t>
</list>
</t>
<t></t>
<t>
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 model.
</t>
<t>
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.
</t>
<t>
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 <xref target="RFC3552"/> "Internet Threat Model". In general, it
is desirable to build a system which is secure against both kinds of
attackers, but realistically many sites do not run HTTPS <xref target="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.
</t>
<section title="Access to Local Resources" anchor="sec.resources">
<t>
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.]
</t>
<t>
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).
</t>
</section>
<section title="Same Origin Policy" anchor="sec.same-origin">
<t>
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="I-D.abarth-origin"/>
(although CORS <xref target="CORS"/> and WebSockets
<xref target="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).
</t>
<t>
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.
</t>
<!-- TODO: Picture -->
</section>
<section title="Bypassing SOP: CORS, WebSockets, and consent to communicate" anchor="sec.cors-etc">
<t>
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 <xref target="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.
</t>
<t>
While CORS is designed to allow cross-origin HTTP requests, WebSockets <xref target="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.
</t>
<t>
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. <xref target="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.
</t>
</section>
</section>
<section title="Security for RTC-Web Applications" anchor="sec.rtc-web">
<section title="Access to Local Devices" anchor="sec.rtc-dev-access">
<t>
As discussed in <xref target="sec.introduction"/>, 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 of user consent.
</t>
<t>
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 <xref target="sec.web-security"/>).
The user's decision will of course be based on his opinion of Site X.
However, as discussed below, this is a complicated concept.
</t>
<section title="Calling Scenarios and User Expectations">
<t>
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.
</t>
<section title="Dedicated Calling Services">
<t>
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.
</t>
<t>
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
<xref target="sec.during-attack"/>.
</t>
</section>
<section title="Calling the Site You're On">
<t>
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 <xref target="sec.resources"/>,
great care must be taken in the design of this interface
to avoid the users just clicking through.
</t>
</section>
<section title="Calling to an Ad Target" anchor="sec.ad-target">
<t>
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.
</t>
<t>
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 calling.
</t>
<t>
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.
</t>
</section>
</section>
<section title="Origin-Based Security">
<t>
As discussed in <xref target="sec.same-origin"/>, 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
mismatch.
</t>
<t>
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.
</t>
<t><list style="hanging">
<t hangText="Individual Consent"></t><t>Ask the user for permission for each call.</t>
<t></t>
<t hangText="Callee-oriented Consent"></t><t>Only allow calls to a given user.</t>
<t></t>
<t hangText="Cryptographic Consent"></t><t>Only allow calls to a given set of peer keying material or
to a cryptographically established identity.</t>
</list>
</t>
<t>
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.
</t>
<t>
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 <xref target="sec.ad-target"/>, 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.
</t>
<t>
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. However, this is also not
that convenient an interface, since managing individual user permissions can
be painful.
</t>
<t>
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.
</t>
</section>
<section title="Security Properties of the Calling Page">
<t>
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:
</t>
<t>
<list style="numbers">
<t>I connect to http://anything.example.org/. Note that this site is unaffiliated
with the calling service.</t>
<t>The attacker modifies my HTTP connection to inject an IFRAME (or a redirect)
to http://calling-service.example.com</t>
<t>The attacker forges the response apparently http://calling-service.example.com/ to
inject JS to initiate a call to himself.</t>
</list>
</t>
<t>
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.
</t>
<t>
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 <xref target="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 infect the browser's notion of that
origin semi-permanently.
</t>
<t>
[[ 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? ]]
</t>
</section>
</section>
<section title="Communications Consent Verification" anchor="sec.rtc-comm-consent">
<t>
As discussed in <xref target="sec.cors-etc"/>, 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.
</t>
<section title="ICE" anchor="sec.ice">
<t>
Verifying receiver consent requires some sort of explicit handshake, but conveniently
we already need one in order to do NAT hole-punching. ICE <xref target="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.
</t>
</section>
<section title="Masking" anchor="sec.masking">
<t>
Once consent is verified, there still is some concern about misinterpretation
attacks as described by Huang et al.<xref target="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 employed.
<!-- [TODO: DNS injection?].-->
</t>
</section>
<section title="Backward Compatibility">
<t>
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:
</t>
<t>
<list style="symbols">
<t>STUN checks without using ICE (i.e., the non-RTC-web endpoint sets up a STUN responder.)</t>
<t>Use or RTCP as an implicit reachability check.</t>
</list>
</t>
<t>
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.
</t>
<t>
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.
</t>
<t>
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.
</t>
<t>
[TODO: Write something about consent freshness and RTCP].
</t>
<t>
[[ 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. ]]
</t>
</section>
</section>
<section title="Communications Security" anchor="sec.rtc-comsec">
<t>
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 <xref target="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 <xref target="RFC4347"/> and
DTLS-SRTP <xref target="RFC5763"/>) is well understood. However, we must
examine this technology to the RTC-Web context, where the threat
model is somewhat different.
</t>
<t>
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. <xref target="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.
</t>
<t>
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:
</t>
<t><list style="hanging">
<t hangText="Retrospective compromise of calling service."></t><t>The calling service is
is non-malicious during a call but subsequently is compromised and wishes to
attack an older call.</t>
<t></t>
<t hangText="During-call attack by calling service."></t><t>The calling service is compromised
during the call it wishes to attack.</t>
</list>
</t>
<t>
Providing security against the former type of attack is practical using the
techniques discussed in <xref target="sec.retrospective-compromise"/>.
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 <xref target="sec.during-attack"/>.
</t>
<section title="Protecting Against Retrospective Compromise" anchor="sec.retrospective-compromise">
<t>
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.
</t>
<t>
If the calling service has access to the traffic keying material
(as in SDES <xref target="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 mechanism.
</t>
<t>
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.
</t>
</section>
<section title="Protecting Against During-Call Attack" anchor="sec.during-attack">
<t>
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 <xref target="abarth-rtcweb"/>.
</t>
<section title="Key Continuity" anchor="sec.key-continuity">
<t>
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 <xref target="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).
</t>
<t>
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 <xref target="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
<xref target="cranor-wolf"/>, it seems extremely unlikely that
any key continuity mechanism will be effective rather than
simply annoying.
</t>
<t>
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.
</t>
</section>
<section title="Short Authentication Strings" anchor="sec.sas">
<t>
ZRTP <xref target="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.
</t>
<t>
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 <xref target="farus-conversion"/> and
humans <xref target="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).
</t>
<t>
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 <xref target="whitten-johnny"/> Thus, it is
highly unlikely that users will ever perform the SAS exchange.
</t>
<t>
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
<xref target="sec.key-continuity"/>.
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.
</t>
</section>
<section title="Recommendations">
<t>
[[ OPEN ISSUE: What are the best UI recommendations to make?
Proposal: take the text from <xref target="I-D.kaufman-rtcweb-security-ui"/>
Section 2]]
</t>
<t>
[[ 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? ]]
</t>
</section>
</section>
</section>
</section>
<section title="Security Considerations" anchor="sec.sec_cons">
<t>This entire document is about security.</t>
</section>
<section title="Acknowledgements">
<t></t>
</section>
</middle>
<back>
<references title="Normative References">
&RFC2119;
</references>
<references title="Informative References">
&RFC3261;
&RFC3552;
&RFC2818;
&RFC5479;
&RFC5763;
&RFC4347;
&RFC4568;
&RFC4251;
&RFC3760;
&RFC6189;
&RFC5245;
&I-D.abarth-origin;
&I-D.ietf-hybi-thewebsocketprotocol;
&I-D.kaufman-rtcweb-security-ui;
<reference anchor="abarth-rtcweb">
<front>
<title>Prompting the user is security failure</title>
<author initials="A." surname="Barth">
<organization></organization>
</author>
</front>
<seriesInfo name="" value="RTC-Web Workshop"/>
<format target="http://rtc-web.alvestrand.com/home/papers/barth-security-prompt.pdf?attredirects=0" type="PDF"/>
</reference>
<reference anchor="whitten-johnny">
<front>
<title>Why Johnny Can't Encrypt: A Usability Evaluation of PGP 5.0</title>
<author initials="A." surname="Whitten">
<organization></organization>
</author>
<author initials="J.D." surname="Tygar">
<organization></organization>
</author>
</front>
<seriesInfo name="" value="Proceedings of the 8th USENIX Security Symposium, 1999"/>
</reference>
<reference anchor="cranor-wolf">
<front>
<title>Crying Wolf: An Empirical Study of SSL Warning Effectiveness</title>
<author initials="J." surname="Sunshine">
<organization></organization>
</author>
<author initials="S." surname="Egelman">
<organization></organization>
</author>
<author initials="H." surname="Almuhimedi">
<organization></organization>
</author>
<author initials="N." surname="Atri">
<organization></organization>
</author>
<author initials="L." surname="cranor">
<organization></organization>
</author>
</front>
<seriesInfo name="" value="Proceedings of the 18th USENIX Security Symposium, 2009"/>
</reference>
<reference anchor="kain-conversion">
<front>
<title>Design and Evaluation of a Voice Conversion Algorithm based on Spectral Envelope Mapping and Residual Prediction</title>
<author initials="A." surname="Kain">
<organization></organization>
</author>
<author initials="M." surname="Macon">
<organization></organization>
</author>
</front>
<seriesInfo name="" value="Proceedings of ICASSP, May 2001"/>
</reference>
<reference anchor="farus-conversion">
<front>
<title>Speaker Recognition Robustness to Voice Conversion</title>
<author initials="M." surname="Farrus">
<organization></organization>
</author>
<author initials="D." surname="Erro">
<organization></organization>
</author>
<author initials="J." surname="Hernando">
<organization></organization>
</author>
</front>
</reference>
<reference anchor="huang-w2sp">
<front>
<title>Talking to Yourself for Fun and Profit</title>
<author initials="L-S." surname="Huang">
<organization></organization>
</author>
<author initials="E.Y." surname="Chen">
<organization></organization>
</author>
<author initials="A." surname="Barth">
<organization></organization>
</author>
<author initials="E." surname="Rescorla">
<organization></organization>
</author>
<author initials="C." surname="Jackson">
<organization></organization>
</author>
</front>
<seriesInfo name="" value="W2SP, 2011"/>
</reference>
<reference anchor="finer-grained">
<front>
<title>Beware of Finer-Grained Origins</title>
<author initials="A." surname="Barth">
<organization></organization>
</author>
<author initials="C." surname="Jackson">
<organization></organization>
</author>
</front>
<seriesInfo name="" value="W2SP, 2008"/>
</reference>
<reference anchor="CORS">
<front>
<title>Cross-Origin Resource Sharing</title>
<author initials="A." surname="van Kesteren">
<organization></organization>
</author>
</front>
<format target="http://www.w3.org/TR/cors/" type="TXT"/>
</reference>
</references>
</back>
</rfc>
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