Security Considerations for RTC-Web
draft-ietf-rtcweb-security-04
The information below is for an old version of the document.
| Document | Type | Active Internet-Draft (rtcweb WG) | |
|---|---|---|---|
| Author | Eric Rescorla | ||
| Last updated | 2013-01-22 | ||
| Replaces | draft-rescorla-rtcweb-security | ||
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draft-ietf-rtcweb-security-04
RTC-Web E. Rescorla
Internet-Draft RTFM, Inc.
Intended status: Standards Track January 22, 2013
Expires: July 26, 2013
Security Considerations for RTC-Web
draft-ietf-rtcweb-security-04
Abstract
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.
Legal
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FOR A PARTICULAR PURPOSE.
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Internet-Drafts are draft documents valid for a maximum of six months
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 July 26, 2013.
Copyright Notice
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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
4.1.1.1. Dedicated Calling Services . . . . . . . . . . . . 9
4.1.1.2. Calling the Site You're On . . . . . . . . . . . . 9
4.1.1.3. Calling to an Ad Target . . . . . . . . . . . . . 10
4.1.2. Origin-Based Security . . . . . . . . . . . . . . . . 10
4.1.3. Security Properties of the Calling Page . . . . . . . 12
4.2. Communications Consent Verification . . . . . . . . . . . 12
4.2.1. ICE . . . . . . . . . . . . . . . . . . . . . . . . . 13
4.2.2. Masking . . . . . . . . . . . . . . . . . . . . . . . 13
4.2.3. Backward Compatibility . . . . . . . . . . . . . . . . 14
4.2.4. IP Location Privacy . . . . . . . . . . . . . . . . . 15
4.3. Communications Security . . . . . . . . . . . . . . . . . 15
4.3.1. Protecting Against Retrospective Compromise . . . . . 16
4.3.2. Protecting Against During-Call Attack . . . . . . . . 17
4.3.2.1. Key Continuity . . . . . . . . . . . . . . . . . . 17
4.3.2.2. Short Authentication Strings . . . . . . . . . . . 18
4.3.2.3. Third Party Identity . . . . . . . . . . . . . . . 19
4.3.2.4. Page Access to Media . . . . . . . . . . . . . . . 19
5. Security Considerations . . . . . . . . . . . . . . . . . . . 20
6. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 20
7. References . . . . . . . . . . . . . . . . . . . . . . . . . . 20
7.1. Normative References . . . . . . . . . . . . . . . . . . . 20
7.2. Informative References . . . . . . . . . . . . . . . . . . 20
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 22
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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 (implementated as browser built-ins) 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
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at any time to any callee of its choice, then this facility can be
used to bug a user's computer without their knowledge, simply by
placing a call to some recording service. More subtly, if the
exposed APIs allow the server to instruct the browser to send
arbitrary content, then they can be used to bypass firewalls or mount
denial of service attacks. Any successful system will need to be
resistant to this and other attacks.
A companion document [I-D.ietf-rtcweb-security-arch] describes a
security architecture intended to address the issues raised in this
document.
2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [RFC2119].
3. The Browser Threat Model
The security requirements for 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
model.
In this model, then, the browser acts as a TRUSTED COMPUTING BASE
(TCB) both from the user's perspective and to some extent from the
server's. While HTML and 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.
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Conventionally, we refer to either WEB ATTACKERS, who are able to
induce you to visit their sites but do not control the network, and
NETWORK ATTACKERS, who are able to control your network. Network
attackers correspond to the [RFC3552] "Internet Threat Model". Note
that for HTTP traffic, a network attacker is also a Web attacker,
since it can inject traffic as if it were any non-HTTPS Web site.
Thus, when analyzing HTTP connections, we must assume that traffic is
going to the attacker.
3.1. Access to Local Resources
While the browser has access to local resources such as keying
material, files, the camera and the microphone, it strictly limits or
forbids web servers from accessing those same resources. For
instance, while it is possible to produce an HTML form which will
allow file upload, a script cannot do so without user consent and in
fact cannot even suggest a specific file (e.g., /etc/passwd); the
user must explicitly select the file and consent to its upload.
[Note: in many cases browsers are explicitly designed to avoid
dialogs with the semantics of "click here to screw yourself", as
extensive research shows that users are prone to consent under such
circumstances.]
Similarly, while Flash 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.[RFC6454] (although CORS [CORS] and WebSockets [RFC6455]
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
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other's JS variables, but pages from the different origins--or even
iframes from different origins on the same page--cannot.
3.3. Bypassing SOP: CORS, WebSockets, and consent to communicate
While SOP serves an important security function, it also makes it
inconvenient to write certain classes of applications. In
particular, mash-ups, in which a script from origin A uses resources
from origin B, can only be achieved via a certain amount of hackery.
The W3C Cross-Origin Resource Sharing (CORS) spec [CORS] is a
response to this demand. In CORS, when a script from origin A
executes what would otherwise be a forbidden cross-origin request,
the browser instead contacts the target server to determine whether
it is willing to allow cross-origin requests from A. If it is so
willing, the browser then allows the request. This consent
verification process is designed to safely allow cross-origin
requests.
While CORS is designed to allow cross-origin HTTP requests,
WebSockets [RFC6455] allows cross-origin establishment of transparent
channels. Once a WebSockets connection has been established from a
script to a site, the script can exchange any traffic it likes
without being required to frame it as a series of HTTP request/
response transactions. As with CORS, a WebSockets transaction starts
with a consent verification stage to avoid allowing scripts to simply
send arbitrary data to another origin.
While consent verification is conceptually simple--just do a
handshake before you start exchanging the real data--experience has
shown that designing a correct consent verification system is
difficult. In particular, Huang et al. [huang-w2sp] have shown
vulnerabilities in the existing Java and Flash consent verification
techniques and in a simplified version of the WebSockets handshake.
In particular, it is important to be wary of CROSS-PROTOCOL attacks
in which the attacking script generates traffic which is acceptable
to some non-Web protocol state machine. In order to resist this form
of attack, WebSockets incorporates a masking technique intended to
randomize the bits on the wire, thus making it more difficult to
generate traffic which resembles a given protocol.
4. Security for 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
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user. At minimum, then, it MUST NOT be possible for arbitrary sites
to initiate calls to arbitrary locations without user consent. This
immediately raises the question, however, of what should be the scope
of user consent.
In order for the user to make an intelligent decision about whether
to allow a call (and hence his camera and microphone input to be
routed somewhere), he must understand either who is requesting
access, where the media is going, or both. As detailed below, there
are two basic conceptual models:
You are sending your media to entity A because you want to talk to
Entity A (e.g., your mother).
Entity A (e.g., a calling service) asks to access the user's
devices with the assurance that it will transfer the media to
entity B (e.g., your mother)
In either case, identity is at the heart of any consent decision.
Moreover, identity is all that the browser can meaningfully enforce;
if you are calling A, A can simply forward the media to C. Similarly,
if you authorize A to place a call to B, A can call C instead. In
either case, all the browser is able to do is verify and check
authorization for whoever is controlling where the media goes. The
target of the media can of course advertise a security/privacy
policy, but this is not something that the browser can enforce. Even
so, there are a variety of different consent scenarios that motivate
different technical consent mechanisms. We discuss these mechanisms
in the sections below.
It's important to understand that consent to access local devices is
largely orthogonal to consent to transmit various kinds of data over
the network (see Section 4.2. Consent for device access is largely a
matter of protecting the user's privacy from malicious sites. By
contrast, consent to send network traffic is about preventing the
user's browser from being used to attack its local network. Thus, we
need to ensure communications consent even if the site is not able to
access the camera and microphone at all (hence WebSockets's consent
mechanism) and similarly we need to be concerned with the site
accessing the user's camera and microphone even if the data is to be
sent back to the site via conventional HTTP-based network mechanisms
such as HTTP POST.
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.
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4.1.1.1. Dedicated Calling Services
The first scenario we consider is a dedicated calling service. In
this case, the user has a relationship with a calling site and
repeatedly makes calls on it. It is likely that rather than having
to give permission for each call that the user will want to give the
calling service long-term access to the camera and microphone. This
is a natural fit for a long-term consent mechanism (e.g., installing
an app store "application" to indicate permission for the calling
service.) A variant of the dedicated calling service is a gaming
site (e.g., a poker site) which hosts a dedicated calling service to
allow players to call each other.
With any kind of service where the user may use the same service to
talk to many different people, there is a question about whether the
user can know who they are talking to. If I grant permission to
calling service A to make calls on my behalf, then I am implicitly
granting it permission to bug my computer whenever it wants. This
suggests another consent model in which a site is authorized to make
calls but only to certain target entities (identified via media-plane
cryptographic mechanisms as described in Section 4.3.2 and especially
Section 4.3.2.3.) Note that the question of consent here is related
to but distinct from the question of peer identity: I might be
willing to allow a calling site to in general initiate calls on my
behalf but still have some calls via that site where I can be sure
that the site is not listening in.
4.1.1.2. Calling the Site You're On
Another simple scenario is calling the site you're actually visiting.
The paradigmatic case here is the "click here to talk to a
representative" windows that appear on many shopping sites. In this
case, the user's expectation is that they are calling the site
they're actually visiting. However, it is unlikely that they want to
provide a general consent to such a site; just because I want some
information on a car doesn't mean that I want the car manufacturer to
be able to activate my microphone whenever they please. Thus, this
suggests the need for a second consent mechanism where I only grant
consent for the duration of a given call. As described in
Section 3.1, great care must be taken in the design of this interface
to avoid the users just clicking through. Note also that the user
interface chrome must clearly display elements showing that the call
is continuing in order to avoid attacks where the calling site just
leaves it up indefinitely but shows a Web UI that implies otherwise.
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4.1.1.3. 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
calling.
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 in IFRAMEs 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
Now that we have seen another use case, we can start to reason about
the security requirements.
As discussed in Section 3.2, the basic unit of Web sandboxing is the
origin, and so it is natural to scope consent to origin.
Specifically, a script from origin A MUST only be allowed to initiate
communications (and hence to access camera and microphone) if the
user has specifically authorized access for that origin. It is of
course technically possible to have coarser-scoped permissions, but
because the Web model is scoped to origin, this creates a difficult
mismatch.
Arguably, origin is not fine-grained enough. Consider the situation
where Alice visits a site and authorizes it to make a single call.
If consent is expressed solely in terms of origin, then at any future
visit to that site (including one induced via mash-up or ad network),
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the site can bug Alice's computer, use the computer to place bogus
calls, etc. While in principle Alice could grant and then revoke the
privilege, in practice privileges accumulate; if we are concerned
about this attack, something else is needed. There are a number of
potential countermeasures to this sort of issue.
Individual Consent
Ask the user for permission for each call.
Callee-oriented Consent
Only allow calls to a given user.
Cryptographic Consent
Only allow calls to a given set of peer keying material or to a
cryptographically established identity.
Unfortunately, none of these approaches is satisfactory for all
cases. As discussed above, individual consent puts the user's
approval in the UI flow for every call. Not only does this quickly
become annoying but it can train the user to simply click "OK", at
which point the consent becomes useless. Thus, while it may be
necessary to have individual consent in some case, this is not a
suitable solution for (for instance) the calling service case. Where
necessary, in-flow user interfaces must be carefully designed to
avoid the risk of the user blindly clicking through.
The other two options are designed to restrict calls to a given
target. Callee-oriented consent provided by the calling site not
work well because a malicious site can claim that the user is calling
any user of his choice. One fix for this is to tie calls to a
cryptographically established identity. While not suitable for all
cases, this approach may be useful for some. If we consider the
advertising case described in Section 4.1.1.3, it's not particularly
convenient to require the advertiser to instantiate an iframe on the
hosting site just to get permission; a more convenient approach is to
cryptographically tie the advertiser's certificate to the
communication directly. We're still tying permissions to origin
here, but to the media origin (and-or destination) rather than to the
Web origin. [I-D.ietf-rtcweb-security-arch] and
[I-D.rescorla-rtcweb-generic-idp] describe mechanisms which
facilitate this sort of consent.
Another case where media-level cryptographic identity makes sense is
when a user really does not trust the calling site. For instance, I
might be worried that the calling service will attempt to bug my
computer, but I also want to be able to conveniently call my friends.
If consent is tied to particular communications endpoints, then my
risk is limited. Naturally, it is somewhat challenging to design UI
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primitives which express this sort of policy. The problem becomes
even more challenging in multi-user calling cases.
4.1.3. Security Properties of the Calling Page
Origin-based security is intended to secure against web attackers.
However, we must also consider the case of network attackers.
Consider the case where I have granted permission to a calling
service by an origin that has the HTTP scheme, e.g.,
http://calling-service.example.com. If I ever use my computer on an
unsecured network (e.g., a hotspot or if my own home wireless network
is insecure), and browse any HTTP site, then an attacker can bug my
computer. The attack proceeds like this:
1. I connect to http://anything.example.org/. Note that this site
is unaffiliated with the calling service.
2. The attacker modifies my HTTP connection to inject an IFRAME (or
a redirect) to http://calling-service.example.com
3. The attacker forges the response apparently
http://calling-service.example.com/ to inject JS to initiate a
call to himself.
Note that this attack does not depend on the media being insecure.
Because the call is to the attacker, it is also encrypted to him.
Moreover, it need not be executed immediately; the attacker can
"infect" the origin semi-permanently (e.g., with a web worker or a
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
infect the browser's notion of that origin semi-permanently.
4.2. Communications Consent Verification
As discussed in Section 3.3, allowing web applications unrestricted
network access via the browser introduces the risk of using the
browser as an attack platform against machines which would not
otherwise be accessible to the malicious site, for instance because
they are topologically restricted (e.g., behind a firewall or NAT).
In order to prevent this form of attack as well as cross-protocol
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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 the site from forging
responses). In order to achieve this objective with ICE, the STUN
transaction IDs must be generated by the browser and MUST NOT be made
available to the initiating script, even via a diagnostic interface.
Verifying receiver consent also requires verifying the receiver wants
to receive traffic from a particular sender, and at this time; for
example a malicious site may simply attempt ICE to known servers that
are using ICE for other sessions. ICE provides this verification as
well, by using the STUN credentials as a form of per-session shared
secret. Those credentials are known to the Web application, but
would need to also be known and used by the STUN-receiving element to
be useful.
There also needs to be some mechanism for the browser to verify that
the target of the traffic continues to wish to receive it.
Obviously, some ICE-based mechanism will work here, but it has been
observed that because ICE keepalives are indications, they will not
work here, so some other mechanism is needed.
[[ OPEN ISSUE: Do we need some way of verifying the expected traffic
rate, not just consent to receive traffic at all.]]
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.
In particular, this is true for the data channel if DTLS is used
because DTLS (with the anti-chosen plaintext mechanisms required by
TLS 1.1) does not allow the attacker to generate predictable
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ciphertext. 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. [Note: current thinking in the
RTCWEB WG is not to support TCP and to support SCTP over DTLS, thus
removing the need for masking.]
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
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STUN servers without the risk of confusing them with legacy STUN
servers. If a non-ICE legacy solution is needed, then this is
probably the best choice.
Once initial consent is verified, we also need to verify continuing
consent, in order to avoid attacks where two people briefly share an
IP (e.g., behind a NAT in an Internet cafe) and the attacker arranges
for a large, unstoppable, traffic flow to the network and then
leaves. The appropriate technologies here are fairly similar to
those for initial consent, though are perhaps weaker since the
threats is less severe.
4.2.4. IP Location Privacy
Note that as soon as the callee sends their ICE candidates, the
caller learns the callee's IP addresses. The callee's server
reflexive address reveals a lot of information about the callee's
location. In order to avoid tracking, implementations may wish to
suppress the start of ICE negotiation until the callee has answered.
In addition, either side may wish to hide their location entirely by
forcing all traffic through a TURN server.
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.
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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
attack.
Providing security against the former type of attack is practical
using the techniques discussed in Section 4.3.1. However, it is
extremely difficult to prevent a trusted but malicious calling
service from actively attacking a user's calls, either by mounting a
MITM attack or by diverting them entirely. (Note that this attack
applies equally to a network attacker if communications to the
calling service are not secured.) We discuss some potential
approaches and why they are likely to be impractical in
Section 4.3.2.
4.3.1. Protecting Against Retrospective Compromise
In a retrospective attack, the calling service was uncompromised
during the call, but that an attacker subsequently wants to recover
the content of the call. We assume that the attacker has access to
the protected media stream as well as having full control of the
calling service.
If the calling service has access to the traffic keying material (as
in SDES [RFC4568]), then retrospective attack is trivial. This form
of attack is particularly serious in the Web context because it is
standard practice in Web services to run extensive logging and
monitoring. Thus, it is highly likely that if the traffic key is
part of any HTTP request it will be logged somewhere and thus subject
to subsequent compromise. It is this consideration that makes an
automatic, public key-based key exchange mechanism imperative for
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.
In addition, the system MUST NOT provide any APIs to extract either
long-term keying material or to directly access any stored traffic
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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
[abarth-rtcweb].
4.3.2.1. Key Continuity
One natural approach is to use "key continuity". While a malicious
calling service can present any identity it chooses to the user, it
cannot produce a private key that maps to a given public key. Thus,
it is possible for the browser to note a given user's public key and
generate an alarm whenever that user's key changes. SSH [RFC4251]
uses a similar technique. (Note that the need to avoid explicit user
consent on every call precludes the browser requiring an immediate
manual check of the peer's key).
Unfortunately, this sort of key continuity mechanism is far less
useful in the 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
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triggering a key continuity warning is to tell the browser that "this
is a call to user Y" where Y is close to X. Even if the user actually
checks the other side's name (which all available evidence indicates
is unlikely), this would require (a) the browser to trusted UI to
provide the name and (b) the user to not be fooled by similar
appearing names.
4.3.2.2. Short Authentication Strings
ZRTP [RFC6189] uses a "short authentication string" (SAS) which is
derived from the key agreement protocol. This SAS is designed to be
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
exchange.
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 4.3.2.1. In principle
it is of course possible to render a different UI element to indicate
that calls are using an unauthenticated set of keying material
(recall that the attacker can just present a slightly different name
so that the attack shows the same UI as a call to a new device or to
someone you haven't called before) but as a practical matter, users
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simply ignore such indicators even in the rather more dire case of
mixed content warnings.
Despite these difficulties, users should be afforded an opportunity
to view an SAS or fingerprint where available, as it is the only
mechanism for the user to directly verify the peer's identity without
trusting any third party identity system (assuming, of course, that
they trust their own software).
4.3.2.3. Third Party Identity
The conventional approach to providing communications identity has of
course been to have some third party identity system (e.g., PKI) to
authenticate the endpoints. Such mechanisms have proven to be too
cumbersome for use by typical users (and nearly too cumbersome for
administrators). However, a new generation of Web-based identity
providers (BrowserID, Federated Google Login, Facebook Connect,
OAuth, OpenID, WebFinger), has recently been developed and use Web
technologies to provide lightweight (from the user's perspective)
third-party authenticated transactions. It is possible (see
[I-D.rescorla-rtcweb-generic-idp]) to use systems of this type to
authenticate RTCWEB calls, linking them to existing user notions of
identity (e.g., Facebook adjacencies). Specifically, the third-party
identity system is used to bind the user's identity to cryptographic
keying material which is then used to authenticate the calling
endpoints. Calls which are authenticated in this fashion are
naturally resistant even to active MITM attack by the calling site.
Note that there is one special case in which PKI-style certificates
do provide a practical solution: calls from end-users to large
sites. For instance, if you are making a call to Amazon.com, then
Amazon can easily get a certificate to authenticate their media
traffic, just as they get one to authenticate their Web traffic.
This does not provide additional security value in cases in which the
calling site and the media peer are one in the same, but might be
useful in cases in which third parties (e.g., ad networks or
retailers) arrange for calls but do not participate in them.
4.3.2.4. Page Access to Media
Identifying the identity of the far media endpoint is a necessary but
not sufficient condition for providing media security. In RTCWEB,
media flows are rendered into HTML5 MediaStreams which can be
manipulated by the calling site. Obviously, if the site can modify
or view the media, then the user is not getting the level of
assurance they would expect from being able to authenticate their
peer. In many cases, this is acceptable because the user values
site-based special effects over complete security from the site.
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However, there are also cases where users wish to know that the site
cannot interfere. In order to facilitate that, it will be necessary
to provide features whereby the site can verifiably give up access to
the media streams. This verification must be possible both from the
local side and the remote side. I.e., I must be able to verify that
the person I am calling has engaged a secure media mode. In order to
achieve this it will be necessary to cryptographically bind an
indication of the local media access policy into the cryptographic
authentication procedures detailed in the previous sections.
5. Security Considerations
This entire document is about security.
6. Acknowledgements
Bernard Aboba, Harald Alvestrand, Dan Druta, Cullen Jennings, Hadriel
Kaplan (S 4.2.1), Matthew Kaufman, Martin Thomson, Magnus Westerland.
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".
[I-D.ietf-rtcweb-security-arch]
Rescorla, E., "RTCWEB Security Architecture",
draft-ietf-rtcweb-security-arch-05 (work in progress),
October 2012.
[I-D.kaufman-rtcweb-security-ui]
Kaufman, M., "Client Security User Interface Requirements
for RTCWEB", draft-kaufman-rtcweb-security-ui-00 (work in
progress), June 2011.
[I-D.rescorla-rtcweb-generic-idp]
Rescorla, E., "RTCWEB Generic Identity Provider
Interface", draft-rescorla-rtcweb-generic-idp-01 (work in
progress), March 2012.
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[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.
[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.
[RFC6454] Barth, A., "The Web Origin Concept", RFC 6454,
December 2011.
[RFC6455] Fette, I. and A. Melnikov, "The WebSocket Protocol",
RFC 6455, December 2011.
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[abarth-rtcweb]
Barth, A., "Prompting the user is security failure", RTC-
Web Workshop.
[cranor-wolf]
Sunshine, J., Egelman, S., Almuhimedi, H., Atri, N., and
L. cranor, "Crying Wolf: An Empirical Study of SSL Warning
Effectiveness", Proceedings of the 18th USENIX Security
Symposium, 2009.
[farus-conversion]
Farrus, M., Erro, D., and J. Hernando, "Speaker
Recognition Robustness to Voice Conversion".
[finer-grained]
Barth, A. and C. Jackson, "Beware of Finer-Grained
Origins", W2SP, 2008.
[huang-w2sp]
Huang, L-S., Chen, E., Barth, A., Rescorla, E., and C.
Jackson, "Talking to Yourself for Fun and Profit", W2SP,
2011.
[kain-conversion]
Kain, A. and M. Macon, "Design and Evaluation of a Voice
Conversion Algorithm based on Spectral Envelope Mapping
and Residual Prediction", Proceedings of ICASSP, May
2001.
[whitten-johnny]
Whitten, A. and J. Tygar, "Why Johnny Can't Encrypt: A
Usability Evaluation of PGP 5.0", Proceedings of the 8th
USENIX Security Symposium, 1999.
Author's Address
Eric Rescorla
RTFM, Inc.
2064 Edgewood Drive
Palo Alto, CA 94303
USA
Phone: +1 650 678 2350
Email: ekr@rtfm.com
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