RTC-Web E. Rescorla
Internet-Draft RTFM, Inc.
Intended status: Standards Track June 5, 2011
Expires: December 7, 2011
Security Considerations for RTC-Web
draft-rescorla-rtcweb-security-00
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 two major use cases for RTC-Web technology
are real-time audio and/or video calls and direct data transfer.
Unlike most conventional real-time systems (e.g., SIP-based soft
phones) RTC-Web communications are directly controlled by some Web
server, which poses new security challenges. For instance, a Web
browser might expose a JavaScript API which allows a server to place
a video call. Unrestricted access to such an API would allow any
site which a user visited to "bug" a user's computer, capturing any
activity which passed in front of their camera. This document
defines the RTC-Web threat model and defines an architecture which
provides security within that threat model.
Legal
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 5
3. The Browser Threat Model . . . . . . . . . . . . . . . . . . . 5
3.1. Access to Local Resources . . . . . . . . . . . . . . . . 6
3.2. Same Origin Policy . . . . . . . . . . . . . . . . . . . . 6
3.3. Bypassing SOP: CORS, WebSockets, and consent to
communicate . . . . . . . . . . . . . . . . . . . . . . . 7
4. Security for RTC-Web Applications . . . . . . . . . . . . . . 7
4.1. Access to Local Devices . . . . . . . . . . . . . . . . . 7
4.2. Communications Consent Verification . . . . . . . . . . . 9
4.2.1. ICE . . . . . . . . . . . . . . . . . . . . . . . . . 9
4.2.2. Masking . . . . . . . . . . . . . . . . . . . . . . . 10
4.2.3. Backward Compatibility . . . . . . . . . . . . . . . . 10
4.3. Communications Security . . . . . . . . . . . . . . . . . 10
4.3.1. Protecting Against Retrospective Compromise . . . . . 11
4.3.2. Protecting Against During-Call Attack . . . . . . . . 12
4.3.2.1. Key Continuity . . . . . . . . . . . . . . . . . . 12
4.3.2.2. Short Authentication Strings . . . . . . . . . . . 13
5. Security Considerations . . . . . . . . . . . . . . . . . . . 14
6. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 14
7. References . . . . . . . . . . . . . . . . . . . . . . . . . . 14
7.1. Normative References . . . . . . . . . . . . . . . . . . . 14
7.2. Informative References . . . . . . . . . . . . . . . . . . 14
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 16
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1. Introduction
The Real-Time Communications on the Web (RTC-Web) working group is
tasked with standardizing protocols for real-time communications
between Web browsers. The two major use cases for RTC-Web technology
are real-time audio and/or video calls and direct data transfer.
Unlike most conventional real-time systems, (e.g., SIP-based[RFC3261]
soft phones) RTC-Web communications are directly controlled by some
Web server. A simple case is shown below.
+----------------+
| |
| Web Server |
| |
+----------------+
^ ^
/ \
HTTP / \ HTTP
/ \
/ \
v v
JS API JS API
+-----------+ +-----------+
| | Media | |
| Browser |<---------->| Browser |
| | | |
+-----------+ +-----------+
Figure 1: A simple RTC-Web system
In the system shown in Figure 1, Alice and Bob both have RTC-Web
enabled browsers and they visit some Web server which operates a
calling service. Each of their browsers exposes standardized
JavaScript calling APIs which are used by the Web server to set up a
call between Alice and Bob. While this system is topologically
similar to a conventional SIP-based system (with the Web server
acting as the signaling service and browsers acting as softphones),
control has moved to the central Web server; the browser simply
provides API points that are used by the calling service. As with
any Web application, the Web server can move logic between the server
and JavaScript in the browser, but regardless of where the code is
executing, it is ultimately under control of the server.
It should be immediately apparent that this type of system poses new
security challenges beyond those of a conventional VoIP system. In
particular, it needs to contend with malicious calling services. For
example, if the calling service can cause the browser to make a call
at any time to any callee of its choice, then this facility can be
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used to bug a user's computer without their knowledge, simply by
placing a call to some recording service. More subtly, if the
exposed APIs allow the server to instruct the browser to send
arbitrary content, then they can be used to bypass firewalls or mount
denial of service attacks. Any successful system will need to be
resistant to this and other attacks.
2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"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.
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
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.[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
requests.
While CORS is designed to allow cross-origin HTTP requests,
WebSockets [I-D.ietf-hybi-thewebsocketprotocol] allows cross-origin
establishment of transparent channels. Once a WebSockets connection
has been established from a script to a site, the script can exchange
any traffic it likes without being required to frame it as a series
of HTTP request/response transactions. As with CORS, a WebSockets
transaction starts with a consent verification stage to avoid
allowing scripts to simply send arbitrary data to another origin.
While consent verification is conceptually simple--just do a
handshake before you start exchanging the real data--experience has
shown that designing a correct consent verification system is
difficult. In particular, Huang et al. [huang-w2sp] have shown
vulnerabilities in the existing Java and Flash consent verification
techniques and in a simplified version of the WebSockets handshake.
In particular, it is important to be wary of CROSS-PROTOCOL attacks
in which the attacking script generates traffic which is acceptable
to some non-Web protocol state machine. In order to resist this form
of attack, WebSockets incorporates a masking technique intended to
randomize the bits on the wire, thus making it more difficult to
generate traffic which resembles a given protocol.
4. Security for RTC-Web Applications
4.1. Access to Local Devices
As discussed in Section 1, allowing arbitrary sites to initiate calls
violates the core Web security guarantee; without some access
restrictions on local devices, any malicious site could simply bug a
user. At minimum, then, it MUST NOT be possible for arbitrary sites
to initiate calls to arbitrary location without user consent. This
immediately raises the question, however, of what should be the scope
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of user consent.
As discussed in Section 3.2, the basic unit of Web sandboxing is the
origin, and so it is natural to scope consent to origin.
Specifically, a script from origin A MUST only be allowed to initiate
communications (and hence to access camera and microphone) if the
user has specifically authorized access for that origin. It is of
course technically possible to have coarser-scoped permissions, but
because the Web model is scoped to origin, this creates a difficult
mismatch.
Arguably, origin is not fine-grained enough. Consider the situation
where Alice visits a site and authorizes it to make a single call.
If consent is expressed solely in terms of origin, then at any future
visit to that site (including one induced via mash-up or ad network),
the site can bug Alice's computer. While in principle Alice could
grant and then revoke the privilege, in practice privileges
accumulate; if we are concerned about this attack, something else is
needed. There are a number of potential countermeasures to this sort
of issue.
Individual Consent
Ask the user for permission for each call.
Callee-oriented Consent
Only allow calls to a given user.
Cryptographic Consent
Only allow calls to a given set of peer keying material.
Unfortunately, none of these approaches is really satisfactory.
Individual consent puts the user's approval in the UI flow for every
call. Not only does this quickly become annoying but it rapidly
trains the user to simply click "OK", at which point the consent
becomes useless.
The other two options are designed to restrict calls to a given
target. Unfortunately, Callee-oriented consent does not work because
the malicious site can claim that the user is calling any user of his
choice. The fix for this is to tie calls to a specific set of
cryptographic keying material, but that breaks any portability for
the callee's client, and is thus problematic. (Section 4.3.2.1)
While this is primarily a question not for IETF, it should be clear
that there is no really good answer. In general, if you cannot trust
the site which you have authorized for calling not to bug you then
your security situation is not really ideal. It is RECOMMENDED that
browsers have explicit (and obvious) indicators that they are in a
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call in order to mitigate this risk.
The above recommendations provide security against web attackers.
However, if a legitimate site is fetched over HTTP rather than HTTPS,
a network attacker can inject code to initiate calls as if it were
that origin, thus bypassing origin restrictions. Note that this form
of attack is also possible if a site embeds active content (e.g.,
JavaScript) that is fetched over HTTP or from an untrusted site,
because that JavaScript is executed in the security context of the
page [finer-grained]. Therefore, it is RECOMMENDED that sites which
embed RTC-Web functionality serve that functionality only over HTTPS
and that browsers disallow execution of calling functionality in
origins which contain mixed content. Note: this issue is not
restricted to PAGES which contain mixed content. If a page from a
given origin ever loads mixed content then it is possible for a
network attacker to infect the browser's notion of that origin semi-
permanently.
4.2. Communications Consent Verification
As discussed in Section 3.3, allowing web applications unrestricted
access to the via the browser network introduces the risk of using
the browser as an attack platform against machines which would not
otherwise be accessible to the malicious site, for instance because
they are topologically restricted (e.g., behind a firewall or NAT).
In order to prevent this form of attack as well as cross-protocol
attacks it is important to require that the target of traffic
explicitly consent to receiving the traffic in question. Until that
consent has been verified for a given endpoint, traffic other than
the consent handshake MUST NOT be sent to that endpoint.
4.2.1. ICE
Verifying receiver consent requires some sort of explicit handshake,
but conveniently we already need one in order to do NAT hole-
punching. ICE [RFC5245] includes a handshake designed to verify that
the receiving element wishes to receive traffic from the sender. It
is important to remember here that the site initiating ICE is
presumed malicious; in order for the handshake to be secure the
receiving element MUST demonstrate receipt/knowledge of some value
not available to the site (thus preventing it from forging
responses). In order to achieve this objective with ICE, the STUN
transaction IDs must be generated by the browser and MUST NOT be made
available to the initiating script, even via a diagnostic interface.
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4.2.2. Masking
Once consent is verified, there still is some concern about
misinterpretation attacks as described by Huang et al.[huang-w2sp].
As long as communication is limited to UDP, then this risk is
probably limited, thus masking is not required for UDP. However,
with TCP the risk of transparent proxies becomes much more severe.
If TCP is to be used, then WebSockets style masking MUST be employed.
4.2.3. Backward Compatibility
A requirement to use ICE limits compatibility with legacy non-ICE
clients. It seems unsafe to completely remove the requirement for
some check, but it might be possible to merely require a one-sided
check where the legacy client was a STUN responder. It's unclear
whether that is in fact simpler than doing ICE-Lite.
4.3. Communications Security
Finally, we consider a problem familiar from the SIP world:
communications security. For obvious reasons, it MUST be possible
for the communicating parties to establish a channel which is secure
against both message recovery and message modification. (See
[RFC5479] for more details.) This service must be provided for both
data and voice/video. Ideally the same security mechanisms would be
used for both types of content. Technology for providing this
service (for instance, DTLS [RFC4347] and DTLS-SRTP [RFC5763]) is
well understood. However, we must examine this technology to the
RTC-Web context, where the threat model is somewhat different.
In general, it is important to understand that unlike a conventional
SIP proxy, the calling service (i.e., the Web server) controls not
only the channel between the communicating endpoints but also the
application running on the user's browser. While in principle it is
possible for the browser to cut the calling service out of the loop
and directly present trusted information (and perhaps get consent),
practice in modern browsers is to avoid this whenever possible. "In-
flow" modal dialogs which require the user to consent to specific
actions are particularly disfavored as human factors research
indicates that unless they are made extremely invasive, users simply
agree to them without actually consciously giving consent.
[abarth-rtcweb]. Thus, nearly all the UI will necessarily be
rendered by the browser but under control of the calling service.
This likely includes the peer's identity information, which, after
all, is only meaningful in the context of some calling service.
This limitation does not mean that preventing attack by the calling
service is completely hopeless. However, we need to distinguish
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between two classes of attack:
Retrospective compromise of calling service.
The calling service is is non-malicious during a call but
subsequently is compromised and wishes to attack an older call.
During-call attack by calling service.
The calling service is compromised during the call it wishes to
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
keys. Otherwise, an attacker who subsequently compromised the
calling service might be able to use those APIs to recover the
traffic keys and thus compromise the traffic.
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4.3.2. Protecting Against During-Call Attack
Protecting against attacks during a call is a more difficult
proposition. Even if the calling service cannot directly access
keying material (as recommended in the previous section), it can
simply mount a man-in-the-middle attack on the connection, telling
Alice that she is calling Bob and Bob that he is calling Alice, while
in fact the calling service is acting as a calling bridge and
capturing all the traffic. While in theory it is possible to
construct techniques which protect against this form of attack, in
practice these techniques all require far too much user intervention
to be practical, given the user interface constraints described in
[abarth-rtcweb].
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
triggering a key continuity warning is to tell the browser that "this
is a call to user Y" where Y is close to X. Even if the user actually
checks the other side's name (which all available evidence indicates
is unlikely), this would require (a) the browser to trusted UI to
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provide the name and (b) the user to not be fooled by similar
appearing names.
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
simply ignore such indicators even in the rather more dire case of
mixed content warnings.
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5. Security Considerations
This entire document is about security.
6. Acknowledgements
7. References
7.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
7.2. Informative References
[CORS] van Kesteren, A., "Cross-Origin Resource Sharing".
[I-D.abarth-origin]
Barth, A., "The Web Origin Concept",
draft-abarth-origin-09 (work in progress), November 2010.
[I-D.ietf-hybi-thewebsocketprotocol]
Fette, I., "The WebSocket protocol",
draft-ietf-hybi-thewebsocketprotocol-07 (work in
progress), April 2011.
[RFC2818] Rescorla, E., "HTTP Over TLS", RFC 2818, May 2000.
[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
A., Peterson, J., Sparks, R., Handley, M., and E.
Schooler, "SIP: Session Initiation Protocol", RFC 3261,
June 2002.
[RFC3552] Rescorla, E. and B. Korver, "Guidelines for Writing RFC
Text on Security Considerations", BCP 72, RFC 3552,
July 2003.
[RFC3760] Gustafson, D., Just, M., and M. Nystrom, "Securely
Available Credentials (SACRED) - Credential Server
Framework", RFC 3760, April 2004.
[RFC4251] Ylonen, T. and C. Lonvick, "The Secure Shell (SSH)
Protocol Architecture", RFC 4251, January 2006.
[RFC4347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer
Security", RFC 4347, April 2006.
Rescorla Expires December 7, 2011 [Page 14]
Internet-Draft RTC-Web Security June 2011
[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.
[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
Rescorla Expires December 7, 2011 [Page 15]
Internet-Draft RTC-Web Security June 2011
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
Rescorla Expires December 7, 2011 [Page 16]