Security Considerations for RTC-Web
draft-ietf-rtcweb-security-01
The information below is for an old version of the document.
| Document | Type |
This is an older version of an Internet-Draft that was ultimately published as RFC 8826.
|
|
|---|---|---|---|
| Author | Eric Rescorla | ||
| Last updated | 2011-10-30 (Latest revision 2011-09-22) | ||
| Replaces | draft-rescorla-rtcweb-security | ||
| RFC stream | Internet Engineering Task Force (IETF) | ||
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| Stream | WG state | WG Document | |
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draft-ietf-rtcweb-security-01
RTC-Web E. Rescorla
Internet-Draft RTFM, Inc.
Intended status: Standards Track October 30, 2011
Expires: May 2, 2012
Security Considerations for RTC-Web
draft-ietf-rtcweb-security-01
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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time. It is inappropriate to use Internet-Drafts as reference
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This Internet-Draft will expire on May 2, 2012.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 5
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 6
3. The Browser Threat Model . . . . . . . . . . . . . . . . . . . 6
3.1. Access to Local Resources . . . . . . . . . . . . . . . . 7
3.2. Same Origin Policy . . . . . . . . . . . . . . . . . . . . 7
3.3. Bypassing SOP: CORS, WebSockets, and consent to
communicate . . . . . . . . . . . . . . . . . . . . . . . 8
4. Security for RTC-Web Applications . . . . . . . . . . . . . . 8
4.1. Access to Local Devices . . . . . . . . . . . . . . . . . 8
4.1.1. Calling Scenarios and User Expectations . . . . . . . 9
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 . . . . . . . . . . . 13
4.2.1. ICE . . . . . . . . . . . . . . . . . . . . . . . . . 13
4.2.2. Masking . . . . . . . . . . . . . . . . . . . . . . . 14
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. Recommendations . . . . . . . . . . . . . . . . . 19
5. Security Considerations . . . . . . . . . . . . . . . . . . . 19
6. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 19
7. References . . . . . . . . . . . . . . . . . . . . . . . . . . 19
7.1. Normative References . . . . . . . . . . . . . . . . . . . 19
7.2. Informative References . . . . . . . . . . . . . . . . . . 20
Appendix A. A Proposed Security Architecture [No Consensus on
This] . . . . . . . . . . . . . . . . . . . . . . . . 22
A.1. Trust Hierarchy . . . . . . . . . . . . . . . . . . . . . 22
A.1.1. Authenticated Entities . . . . . . . . . . . . . . . . 22
A.1.2. Unauthenticated Entities . . . . . . . . . . . . . . . 23
A.2. Overview . . . . . . . . . . . . . . . . . . . . . . . . . 23
A.2.1. Initial Signaling . . . . . . . . . . . . . . . . . . 24
A.2.2. Media Consent Verification . . . . . . . . . . . . . . 26
A.2.3. DTLS Handshake . . . . . . . . . . . . . . . . . . . . 26
A.2.4. Communications and Consent Freshness . . . . . . . . . 27
A.3. Detailed Technical Description . . . . . . . . . . . . . . 27
A.3.1. Origin and Web Security Issues . . . . . . . . . . . . 27
A.3.2. Device Permissions Model . . . . . . . . . . . . . . . 28
A.3.3. Communications Consent . . . . . . . . . . . . . . . . 29
A.3.4. IP Location Privacy . . . . . . . . . . . . . . . . . 29
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A.3.5. Communications Security . . . . . . . . . . . . . . . 30
A.3.6. Web-Based Peer Authentication . . . . . . . . . . . . 31
A.3.6.1. Generic Concepts . . . . . . . . . . . . . . . . . 31
A.3.6.2. BrowserID . . . . . . . . . . . . . . . . . . . . 32
A.3.6.3. OAuth . . . . . . . . . . . . . . . . . . . . . . 35
A.3.6.4. Generic Identity Support . . . . . . . . . . . . . 36
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 36
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1. Introduction
The Real-Time Communications on the Web (RTC-Web) working group is
tasked with standardizing protocols for real-time communications
between Web browsers. The major use cases for RTC-Web technology are
real-time audio and/or video calls, Web conferencing, and direct data
transfer. Unlike most conventional real-time systems, (e.g., SIP-
based[RFC3261] soft phones) RTC-Web communications are directly
controlled by some Web server. A simple case is shown below.
+----------------+
| |
| Web Server |
| |
+----------------+
^ ^
/ \
HTTP / \ HTTP
/ \
/ \
v v
JS API JS API
+-----------+ +-----------+
| | Media | |
| Browser |<---------->| Browser |
| | | |
+-----------+ +-----------+
Figure 1: A simple RTC-Web system
In the system shown in Figure 1, Alice and Bob both have RTC-Web
enabled browsers and they visit some Web server which operates a
calling service. Each of their browsers exposes standardized
JavaScript calling APIs which are used by the Web server to set up a
call between Alice and Bob. While this system is topologically
similar to a conventional SIP-based system (with the Web server
acting as the signaling service and browsers acting as softphones),
control has moved to the central Web server; the browser simply
provides API points that are used by the calling service. As with
any Web application, the Web server can move logic between the server
and JavaScript in the browser, but regardless of where the code is
executing, it is ultimately under control of the server.
It should be immediately apparent that this type of system poses new
security challenges beyond those of a conventional VoIP system. In
particular, it needs to contend with malicious calling services. For
example, if the calling service can cause the browser to make a call
at any time to any callee of its choice, then this facility can be
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used to bug a user's computer without their knowledge, simply by
placing a call to some recording service. More subtly, if the
exposed APIs allow the server to instruct the browser to send
arbitrary content, then they can be used to bypass firewalls or mount
denial of service attacks. Any successful system will need to be
resistant to this and other attacks.
2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"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 locations without user consent. This
immediately raises the question, however, of what should be the scope
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of user consent.
For the rest of this discussion we assume that the user is somehow
going to grant consent to some entity (e.g., a social networking
site) to initiate a call on his behalf. This consent may be limited
to a single call or may be a general consent. In order for the user
to make an intelligent decision about whether to allow a call (and
hence his camera and microphone input to be routed somewhere), he
must understand either who is requesting 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 3). The user's
decision will of course be based on his opinion of Site X. However,
as discussed below, this is a complicated concept.
4.1.1. Calling Scenarios and User Expectations
While a large number of possible calling scenarios are possible, the
scenarios discussed in this section illustrate many of the
difficulties of identifying the relevant scope of consent.
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. In general, this is difficult
as most of the user interface is presented by the calling site.
However, communications security mechanisms can be used to give some
assurance, as described in Section 4.3.2.
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
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case, the user's expectation is that they are calling the site
they're actually visiting. However, it is unlikely that they want to
provide a general consent to such a site; just because I want some
information on a car doesn't mean that I want the car manufacturer to
be able to activate my microphone whenever they please. Thus, this
suggests the need for a second consent mechanism where I only grant
consent for the duration of a given call. As described in
Section 3.1, great care must be taken in the design of this interface
to avoid the users just clicking through. 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.
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 often learn very little about whether individual users
clicked through to the ads, or even which ads were presented.
4.1.2. Origin-Based Security
As discussed in Section 3.2, the basic unit of Web sandboxing is the
origin, and so it is natural to scope consent to origin.
Specifically, a script from origin A MUST only be allowed to initiate
communications (and hence to access camera and microphone) if the
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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, 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. Unfortunately, Callee-oriented consent does not work well
because a malicious site can claim that the user is calling any user
of his choice. One fix for this is to tie calls to a
cryptographically established identity. While not suitable for all
cases, this approach may be useful for some. If we consider the
advertising case described in Section 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.
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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.
While this is primarily a question not for IETF, it should be clear
that there is no really good answer. In general, if you cannot trust
the site which you have authorized for calling not to bug you then
your security situation is not really ideal. It is RECOMMENDED that
browsers have explicit (and obvious) indicators that they are in a
call in order to mitigate this risk.
4.1.3. Security Properties of the Calling Page
Origin-based security is intended to 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
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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.
[[ OPEN ISSUE: What recommendation should IETF make about (a)
whether RTCWeb long-term consent should be available over HTTP pages
and (b) How to handle origins where the consent is to an HTTPS URL
but the page contains active mixed content? ]]
4.2. Communications Consent Verification
As discussed in Section 3.3, allowing web applications unrestricted
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
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
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work here, so some other mechanism is needed.
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
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.
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
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consents to this particular call. This may be an issue if existing
STUN servers are operated at addresses that are not able to handle
bandwidth-based attacks. Thus, this approach does not seem
satisfactory either.
If the systems are tightly integrated (i.e., the STUN endpoint
responds with responses authenticated with ICE credentials) then this
issue does not exist. However, such a design is very close to an
ICE-Lite implementation (indeed, arguably is one). An intermediate
approach would be to have a STUN extension that indicated that one
was responding to RTC-Web checks but not computing integrity checks
based on the ICE credentials. This would allow the use of standalone
STUN servers without the risk of confusing them with legacy STUN
servers. If a non-ICE legacy solution is needed, then this is
probably the best choice.
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.
[[ OPEN ISSUE: Exactly what should be the requirements here?
Proposals include ICE all the time or ICE but with allowing one of
these non-ICE things for legacy. ]]
4.2.4. IP Location Privacy
Note that as soon as the callee sends their ICE candidates, the
callee 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
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well understood. However, we must examine this technology to the
RTC-Web context, where the threat model is somewhat different.
In general, it is important to understand that unlike a conventional
SIP proxy, the calling service (i.e., the Web server) controls not
only the channel between the communicating endpoints but also the
application running on the user's browser. While in principle it is
possible for the browser to cut the calling service out of the loop
and directly present trusted information (and perhaps get consent),
practice in modern browsers is to avoid this whenever possible. "In-
flow" modal dialogs which require the user to consent to specific
actions are particularly disfavored as human factors research
indicates that unless they are made extremely invasive, users simply
agree to them without actually consciously giving consent.
[abarth-rtcweb]. Thus, nearly all the UI will necessarily be
rendered by the browser but under control of the calling service.
This likely includes the peer's identity information, which, after
all, is only meaningful in the context of some calling service.
This limitation does not mean that preventing attack by the calling
service is completely hopeless. However, we need to distinguish
between two classes of attack:
Retrospective compromise of calling service.
The calling service is is non-malicious during a call but
subsequently is compromised and wishes to attack an older call.
During-call attack by calling service.
The calling service is compromised during the call it wishes to
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.
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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.
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
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for a user to use multiple browsers on different computers which will
of course have different keying material (SACRED [RFC3760]
notwithstanding.) Thus, users will frequently be alerted to key
mismatches which are in fact completely legitimate, with the result
that they are trained to simply click through them. As it is known
that users routinely will click through far more dire warnings
[cranor-wolf], it seems extremely unlikely that any key continuity
mechanism will be effective rather than simply annoying.
Moreover, it is trivial to bypass even this kind of mechanism.
Recall that unlike the case of SSH, the browser never directly gets
the peer's identity from the user. Rather, it is provided by the
calling service. Even enabling a mechanism of this type would
require an API to allow the calling service to tell the browser "this
is a call to user X". All the calling service needs to do to avoid
triggering a key continuity warning is to tell the browser that "this
is a call to user Y" where Y is close to X. Even if the user actually
checks the other side's name (which all available evidence indicates
is unlikely), this would require (a) the browser to trusted UI to
provide the name and (b) the user to not be fooled by similar
appearing names.
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
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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.
4.3.2.3. Recommendations
[[ OPEN ISSUE: What are the best UI recommendations to make?
Proposal: take the text from [I-D.kaufman-rtcweb-security-ui]
Section 2]]
[[ OPEN ISSUE: Exactly what combination of media security primitives
should be specified and/or mandatory to implement? In particular,
should we allow DTLS-SRTP only, or both DTLS-SRTP and SDES. Should
we allow RTP for backward compatibility? ]]
5. Security Considerations
This entire document is about security.
6. Acknowledgements
Bernard Aboba, Harald Alvestrand, Cullen Jennings, Hadriel Kaplan (S
4.2.1), Matthew Kaufman, 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.
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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. and A. Melnikov, "The WebSocket protocol",
draft-ietf-hybi-thewebsocketprotocol-17 (work in
progress), September 2011.
[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.
[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.
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[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
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.
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Appendix A. A Proposed Security Architecture [No Consensus on This]
This section contains a proposed security architecture, based on the
considerations discussed in the main body of this memo. This section
is currently the opinion of the author and does not have consensus
though some (many?) elements of this proposal do seem to have general
consensus.
A.1. Trust Hierarchy
The basic assumption of this proposal is that network resources exist
in a hierarchy of trust, rooted in the browser, which serves as the
user's TRUSTED COMPUTING BASE (TCB). Any security property which the
user wishes to have enforced must be ultimately guaranteed by the
browser (or transitively by some property the browser verifies).
Conversely, if the browser is compromised, then no security
guarantees are possible. Note that there are cases (e.g., Internet
kiosks) where the user can't really trust the browser that much. In
these cases, the level of security provided is limited by how much
they trust the browser.
Optimally, we would not rely on trust in any entities other than the
browser. However, this is unfortunately not possible if we wish to
have a functional system. Other network elements fall into two
categories: those which can be authenticated by the browser and thus
are partly trusted--though to the minimum extent necessary--and those
which cannot be authenticated and thus are untrusted. This is a
natural extension of the end-to-end principle.
A.1.1. Authenticated Entities
There are two major classes of authenticated entities in the system:
o Calling services: Web sites whose origin we can verify (optimally
via HTTPS).
o Other users: RTC-Web peers whose origin we can verify
cryptographically (optimally via DTLS-SRTP).
Note that merely being authenticated does not make these entities
trusted. For instance, just because we can verify that
https://www.evil.org/ is owned by Dr. Evil does not mean that we can
trust Dr. Evil to access our camera an microphone. However, it gives
the user an opportunity to determine whether he wishes to trust Dr.
Evil or not; after all, if he desires to contact Dr. Evil, it's safe
to temporarily give him access to the camera and microphone for the
purpose of the call. The point here is that we must first identify
other elements before we can determine whether to trust them.
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It's also worth noting that there are settings where authentication
is non-cryptographic, such as other machines behind a firewall.
Naturally, the level of trust one can have in identities verified in
this way depends on how strong the topology enforcement is.
A.1.2. Unauthenticated Entities
Other than the above entities, we are not generally able to identify
other network elements, thus we cannot trust them. This does not
mean that it is not possible to have any interaction with them, but
it means that we must assume that they will behave maliciously and
design a system which is secure even if they do so.
A.2. Overview
This section describes a typical RTCWeb session and shows how the
various security elements interact and what guarantees are provided
to the user. The example in this section is a "best case" scenario
in which we provide the maximal amount of user authentication and
media privacy with the minimal level of trust in the calling service.
Simpler versions with lower levels of security are also possible and
are noted in the text where applicable. It's also important to
recognize the tension between security (or performance) and privacy.
The example shown here is aimed towards settings where we are more
concerned about secure calling than about privacy, but as we shall
see, there are settings where one might wish to make different
tradeoffs--this architecture is still compatible with those settings.
For the purposes of this example, we assume the topology shown in the
figure below. This topology is derived from the topology shown in
Figure 1, but separates Alice and Bob's identities from the process
of signaling. Specifically, Alice and Bob have relationships with
some Identity Provider (IDP) that supports a protocol such OpenID or
BrowserID) that can be used to attest to their identity. This
separation isn't particularly important in "closed world" cases where
Alice and Bob are users on the same social network and have
identities based on that network. However, there are important
settings where that is not the case, such as federation (calls from
one network to another) and calling on untrusted sites, such as where
two users who have a relationship via a given social network want to
call each other on another, untrusted, site, such as a poker site.
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+----------------+
| |
| Signaling |
| Server |
| |
+----------------+
^ ^
/ \
HTTPS / \ HTTPS
/ \
/ \
v v
JS API JS API
+-----------+ +-----------+
| | Media | |
Alice | Browser |<---------->| Browser | Bob
| | (DTLS-SRTP)| |
+-----------+ +-----------+
^ ^--+ +--^ ^
| | | |
v | | v
+-----------+ | | +-----------+
| |<--------+ | |
| IDP | | | IDP |
| | +------->| |
+-----------+ +-----------+
Figure 2: A call with IDP-based identity
A.2.1. Initial Signaling
Alice and Bob are both users of a common calling service; they both
have approved the calling service to make calls (we defer the
discussion of device access permissions till later). They are both
connected to the calling service via HTTPS and so know the origin
with some level of confidence. They also have accounts with some
identity provider. This sort of identity service is becoming
increasingly common in the Web environment in technologies such
(BrowserID, Federated Google Login, Facebook Connect, OAuth, OpenID,
WebFinger), and is often provided as a side effect service of your
ordinary accounts with some service. In this example, we show Alice
and Bob using a separate identity service, though they may actually
be using the same identity service as calling service or have no
identity service at all.
Alice is logged onto the calling service and decides to call Bob. She
can see from the calling service that he is online and the calling
service presents a JS UI in the form of a button next to Bob's name
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which says "Call". Alice clicks the button, which initiates a JS
callback that instantiates a PeerConnection object. This does not
require a security check: JS from any origin is allowed to get this
far.
Once the PeerConnection is created, the calling service JS needs to
set up some media. Because this is an audio/video call, it creates
two MediaStreams, one connected to an audio input and one connected
to a video input. At this point the first security check is
required: untrusted origins are not allowed to access the camera and
microphone. In this case, because Alice is a long-term user of the
calling service, she has made a permissions grant (i.e., a setting in
the browser) to allow the calling service to access her camera and
microphone any time it wants. The browser checks this setting when
the camera and microphone requests are made and thus allows them.
In the current W3C API, once some streams have been added, Alice's
browser + JS generates a signaling message The format of this data is
currently undefined. It may be a complete message as defined by ROAP
[REF] or may be assembled piecemeal by the JS. In either case, it
will contain:
o Media channel information
o ICE candidates
o A fingerprint attribute binding the message to Alice's public key
[RFC5763]
Prior to sending out the signaling message, the PeerConnection code
contacts the identity service and obtains an assertion binding
Alice's identity to her fingerprint. The exact details depend on the
identity service (though as discussed in Appendix A.3.6.4 I believe
PeerConnection can be agnostic to them), but for now it's easiest to
think of as a BrowserID assertion.
This message is sent to the signaling server, e.g., by XMLHttpRequest
[REF] or by WebSockets [I-D.ietf-hybi-thewebsocketprotocol]. The
signaling server processes the message from Alice's browser,
determines that this is a call to Bob and sends a signaling message
to Bob's browser (again, the format is currently undefined). The JS
on Bob's browser processes it, and alerts Bob to the incoming call
and to Alice's identity. In this case, Alice has provided an
identity assertion and so Bob's browser contacts Alice's identity
provider (again, this is done in a generic way so the browser has no
specific knowledge of the IDP) to verity the assertion. This allows
the browser to display a trusted element indicating that a call is
coming in from Alice. If Alice is in Bob's address book, then this
interface might also include her real name, a picture, etc. The
calling site will also provide some user interface element (e.g., a
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button) to allow Bob to answer the call, though this is most likely
not part of the trusted UI.
If Bob agrees [I am ignoring early media for now], a PeerConnection
is instantiated with the message from Alice's side. Then, a similar
process occurs as on Alice's browser: Bob's browser verifies that
the calling service is approved, the media streams are created, and a
return signaling message containing media information, ICE
candidates, and a fingerprint is sent back to Alice via the signaling
service. If Bob has a relationship with an IDP, the message will
also come with an identity assertion.
At this point, Alice and Bob each know that the other party wants to
have a secure call with them. Based purely on the interface provided
by the signaling server, they know that the signaling server claims
that the call is from Alice to Bob. Because the far end sent an
identity assertion along with their message, they know that this is
verifiable from the IDP as well. Of course, the call works perfectly
well if either Alice or Bob doesn't have a relationship with an IDP;
they just get a lower level of assurance. Moreover, Alice might wish
to make an anonymous call through an anonymous calling site, in which
case she would of course just not provide any identity assertion and
the calling site would mask her identity from Bob.
A.2.2. Media Consent Verification
As described in Section 4.2. This proposal specifies that that be
performed via ICE. Thus, Alice and Bob perform ICE checks with each
other. At the completion of these checks, they are ready to send
non-ICE data.
At this point, Alice knows that (a) Bob (assuming he is verified via
his IDP) or someone else who the signaling service is claiming is Bob
is willing to exchange traffic with her and (b) that either Bob is at
the IP address which she has verified via ICE or there is an attacker
who is on-path to that IP address detouring the traffic. Note that
it is not possible for an attacker who is on-path but not attached to
the signaling service to spoof these checks because they do not have
the ICE credentials. Bob's security guarantees with respect to Alice
are the converse of this.
A.2.3. DTLS Handshake
Once the ICE checks have completed [more specifically, once some ICE
checks have completed], Alice and Bob can set up a secure channel.
This is performed via DTLS [RFC4347] (for the data channel) and DTLS-
SRTP [RFC5763] for the media channel. Specifically, Alice and Bob
perform a DTLS handshake on every channel which has been established
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by ICE. The total number of channels depends on the amount of
muxing; in the most likely case we are using both RTP/RTCP mux and
muxing multiple media streams on the same channel, in which case
there is only one DTLS handshake. Once the DTLS handshake has
completed, the keys are extracted and used to key SRTP for the media
channels.
At this point, Alice and Bob know that they share a set of secure
data and/or media channels with keys which are not known to any
third-party attacker. If Alice and Bob authenticated via their IDPs,
then they also know that the signaling service is not attacking them.
Even if they do not use an IDP, as long as they have minimal trust in
the signaling service not to perform a man-in-the-middle attack, they
know that their communications are secure against the signaling
service as well.
A.2.4. Communications and Consent Freshness
From a security perspective, everything from here on in is a little
anticlimactic: Alice and Bob exchange data protected by the keys
negotiated by DTLS. Because of the security guarantees discussed in
the previous sections, they know that the communications are
encrypted and authenticated.
The one remaining security property we need to establish is "consent
freshness", i.e., allowing Alice to verify that Bob is still prepared
to receive her communications. ICE specifies periodic STUN
keepalizes but only if media is not flowing. Because the consent
issue is more difficult here, we require RTCWeb implementations to
periodically send keepalives. If a keepalive fails and no new ICE
channels can be established, then the session is terminated.
A.3. Detailed Technical Description
A.3.1. Origin and Web Security Issues
The basic unit of permissions for RTC-Web is the origin
[I-D.abarth-origin]. Because the security of the origin depends on
being able to authenticate content from that origin, the origin can
only be securely established if data is transferred over HTTPS.
Thus, clients MUST treat HTTP and HTTPS origins as different
permissions domains and SHOULD NOT permit access to any RTC-Web
functionality from scripts fetched over non-secure (HTTP) origins.
If an HTTPS origin contains mixed active content (regardless of
whether it is present on the specific page attempting to access RTC-
Web functionality), any access MUST be treated as if it came from the
HTTP origin. For instance, if a https://www.example.com/example.html
loads https://www.example.com/example.js and
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http://www.example.org/jquery.js, any attempt by example.js to access
RTCWeb functionality MUST be treated as if it came from
http://www.example.com/. Note that many browsers already track mixed
content and either forbid it by default or display a warning.
A.3.2. Device Permissions Model
Implementations MUST obtain explicit user consent prior to providing
access to the camera and/or microphone. Implementations MUST at
minimum support the following two permissions models:
o Requests for one-time camera/microphone access.
o Requests for permanent access.
In addition, they SHOULD support requests for access to a single
communicating peer. E.g., "Call customerservice@ford.com". Browsers
servicing such requests SHOULD clearly indicate that identity to the
user when asking for permission.
API Requirement: The API MUST provide a mechanism for the requesting
JS to indicate which of these forms of permissions it is
requesting. This allows the client to know what sort of user
interface experience to provide. In particular, browsers might
display a non-invasive door hanger ("some features of this site
may not work..." when asking for long-term permissions) but a more
invasive UI ("here is your own video") for single-call
permissions. The API MAY grant weaker permissions than the JS
asked for if the user chooses to authorize only those permissions,
but if it intends to grant stronger ones SHOULD display the
appropriate UI for those permissions.
API Requirement: The API MUST provide a mechanism for the requesting
JS to relinquish the ability to see or modify the media (e.g., via
MediaStream.record()). Combined with secure authentication of the
communicating peer, this allows a user to be sure that the calling
site is not accessing or modifying their conversion.
UI Requirement: The UI MUST clearly indicate when the user's camera
and microphone are in use. This indication MUST NOT be
suppressable by the JS and MUST clearly indicate how to terminate
a call, and provide a UI means to immediately stop camera/
microphone input without the JS being able to prevent it.
UI Requirement: If the UI indication of camera/microphone use are
displayed in the browser such that minimizing the browser window
would hide the indication, or the JS creating an overlapping
window would hide the indication, then the browser SHOULD stop
camera and microphone input.
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Clients MAY permit the formation of data channels without any direct
user approval. Because sites can always tunnel data through the
server, further restrictions on the data channel do not provide any
additional security. (though see Appendix A.3.3 for a related issue).
Implementations which support some form of direct user authentication
SHOULD also provide a policy by which a user can authorize calls only
to specific counterparties. Specifically, the implementation SHOULD
provide the following interfaces/controls:
o Allow future calls to this verified user.
o Allow future calls to any verified user who is in my system
address book (this only works with address book integration, of
course).
Implementations SHOULD also provide a different user interface
indication when calls are in progress to users whose identities are
directly verifiable. Appendix A.3.5 provides more on this.
A.3.3. Communications Consent
Browser client implementations of RTC-Web MUST implement ICE. Server
gateway implementations which operate only at public IP addresses may
implement ICE-Lite.
Browser implementations MUST verify reachability via ICE prior to
sending any non-ICE packets to a given destination. Implementations
MUST NOT provide the ICE transaction ID to JavaScript. [Note: this
document takes no position on the split between ICE in JS and ICE in
the browser. The above text is written the way it is for editorial
convenience and will be modified appropriately if the WG decides on
ICE in the JS.]
Implementations MUST send keepalives no less frequently than every 30
seconds regardless of whether traffic is flowing or not. If a
keepalive fails then the implementation MUST either attempt to find a
new valid path via ICE or terminate media for that ICE component.
Note that ICE [RFC5245]; Section 10 keepalives use STUN Binding
Indications which are one-way and therefore not sufficient. We will
need to define a new mechanism for this. [OPEN ISSUE: what to do
here.]
A.3.4. IP Location Privacy
As mentioned in Section 4.2.4 above, a side effect of the default ICE
behavior is that the peer learns one's IP address, which leaks large
amounts of location information, especially for mobile devices. This
has negative privacy consequences in some circumstances. The
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following two API requirements are intended to mitigate this issue:
API Requirement: The API MUST provide a mechanism to suppress ICE
negotiation (though perhaps to allow candidate gathering) until
the user has decided to answer the call [note: determining when
the call has been answered is a question for the JS.] This
enables a user to prevent a peer from learning their IP address if
they elect not to answer a call.
API Requirement: The API MUST provide a mechanism for the calling
application to indicate that only TURN candidates are to be used.
This prevents the peer from learning one's IP address at all.
A.3.5. Communications Security
Implementations MUST implement DTLS and DTLS-SRTP. All data channels
MUST be secured via DTLS. DTLS-SRTP MUST be offered for every media
channel and MUST be the default; i.e., if an implementation receives
an offer for DTLS-SRTP and SDES and/or plain RTP, DTLS-SRTP MUST be
selected.
[OPEN ISSUE: What should the settings be here? MUST?]
Implementations MAY support SDES and RTP for media traffic for
backward compatibility purposes.
API Requirement: The API MUST provide a mechanism to indicate that a
fresh DTLS key pair is to be generated for a specific call. This
is intended to allow for unlinkability. Note that there are also
settings where it is attractive to use the same keying material
repeatedly, especially those with key continuity-based
authentication.
API Requirement: The API MUST provide a mechanism to indicate that a
fresh DTLS key pair is to be generated for a specific call. This
is intended to allow for unlinkability.
API Requirement: When DTLS-SRTP is used, the API MUST NOT permit the
JS to obtain the negotiated keying material. This requirement
preserves the end-to-end security of the media.
UI Requirements: A user-oriented client MUST provide an
"inspector" interface which allows the user to determine the
security characteristics of the media. [largely derived from
[I-D.kaufman-rtcweb-security-ui]
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The following properties SHOULD be displayed "up-front" in the
browser chrome, i.e., without requiring the user to ask for them:
* A client MUST provide a user interface through which a user may
determine the security characteristics for currently-displayed
audio and video stream(s)
* A client MUST provide a user interface through which a user may
determine the security characteristics for transmissions of
their microphone audio and camera video.
* The "security characteristics" MUST include an indication as to
whether or not the transmission is cryptographically protected
and whether that protection is based on a key that was
delivered out-of-band (from a server) or was generated as a
result of a pairwise negotiation.
* If the far endpoint was directly verified Appendix A.3.6 the
"security characteristics" MUST include the verified
information.
The following properties are more likely to require some "drill-
down" from the user:
* If the transmission is cryptographically protected, the The
algorithms in use (For example: "AES-CBC" or "Null Cipher".)
* If the transmission is cryptographically protected, the
"security characteristics" MUST indicate whether PFS is
provided.
* If the transmission is cryptographically protected via an end-
to-end mechanism the "security characteristics" MUST include
some mechanism to allow an out-of-band verification of the
peer, such as a certificate fingerprint or an SAS.
A.3.6. Web-Based Peer Authentication
A.3.6.1. Generic Concepts
In a number of cases, it is desirable for the endpoint (i.e., the
browser) to be able to directly identity the endpoint on the other
side without trusting only the signaling service to which they are
connected. For instance, users may be making a call via a federated
system where they wish to get direct authentication of the other
side. Alternately, they may be making a call on a site which they
minimally trust (such as a poker site) but to someone who has an
identity on a site they do trust (such as a social network.)
Recently, a number of Web-based identity technologies (OAuth,
BrowserID, Facebook Connect), etc. have been developed. While the
details vary, what these technologies share is that they have a Web-
based (i.e., HTTP/HTTPS identity provider) which attests to your
identity. For instance, if I have an account at example.org, I could
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use the example.org identity provider to prove to others that I was
alice@example.org. The development of these technologies allows us
to separate calling from identity provision: I could call you on
Poker Galaxy but identify myself as alice@example.org.
Whatever the underlying technology, the general principle is that the
party which is being authenticated is NOT the signaling site but
rather the user (and their browser). Similarly, the relying party is
the browser and not the signaling site. This means that the
PeerConnection API MUST arrange to talk directly to the identity
provider in a way that cannot be impersonated by the calling site.
The following sections provide two examples of this.
A.3.6.2. BrowserID
BrowserID [https://browserid.org/] is a technology which allows a
user with a verified email address to generate an assertion
(authenticated by their identity provider) attesting to their
identity (phrased as an email address). The way that this is used in
practice is that the relying party embeds JS in their site which
talks to the BrowserID code (either hosted on a trusted intermediary
or embedded in the browser). That code generates the assertion which
is passed back to the relying party for verification. The assertion
can be verified directly or with a Web service provided by the
identity provider. It's relatively easy to extend this functionality
to authenticate RTC-Web calls, as shown below.
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+----------------------+ +----------------------+
| | | |
| Alice's Browser | | Bob's Browser |
| | OFFER ------------> | |
| Calling JS Code | | Calling JS Code |
| ^ | | ^ |
| | | | | |
| v | | v |
| PeerConnection | | PeerConnection |
| | ^ | | | ^ |
| Finger| |Signed | |Signed | | |
| print | |Finger | |Finger | |"Alice"|
| | |print | |print | | |
| v | | | v | |
| +--------------+ | | +---------------+ |
| | BrowserID | | | | BrowserID | |
| | Signer | | | | Verifier | |
| +--------------+ | | +---------------+ |
| ^ | | ^ |
+-----------|----------+ +----------|-----------+
| |
| Get certificate |
v | Check
+----------------------+ | certificate
| | |
| Identity |/-------------------------------+
| Provider |
| |
+----------------------+
The way this mechanism works is as follows. On Alice's side, Alice
goes to initiate a call.
1. The calling JS instantiates a PeerConnection and tells it that it
is interested in having it authenticated via BrowserID.
2. The PeerConnection instantiates the BrowserID signer in an
invisible IFRAME. The IFRAME is tagged with an origin that
indicates that it was generated by the PeerConnection (this
prevents ordinary JS from implementing it). The BrowserID signer
is provided with Alice's fingerprint. Note that the IFRAME here
does not render any UI. It is being used solely to allow the
browser to load the BrowserID signer in isolation, especially
from the calling site.
3. The BrowserID signer contacts Alice's identity provider,
authenticating as Alice (likely via a cookie).
4. The identity provider returns a short-term certificate attesting
to Alice's identity and her short-term public key.
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5. The Browser-ID code signs the fingerprint and returns the signed
assertion + certificate to the PeerConnection. [Note: there are
well-understood Web mechanisms for this that I am excluding here
for simplicity.]
6. The PeerConnection returns the signed information to the calling
JS code.
7. The signed assertion gets sent over the wire to Bob's browser
(via the signaling service) as part of the call setup.
Obviously, the format of the signed assertion varies depending on
what signaling style the WG ultimately adopts. However, for
concreteness, if something like ROAP were adopted, then the entire
message might look like:
{
"messageType":"OFFER",
"callerSessionId":"13456789ABCDEF",
"seq": 1
"sdp":"
v=0\n
o=- 2890844526 2890842807 IN IP4 192.0.2.1\n
s= \n
c=IN IP4 192.0.2.1\n
t=2873397496 2873404696\n
m=audio 49170 RTP/AVP 0\n
a=fingerprint: SHA-1 \
4A:AD:B9:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB\n",
"identity":{
"identityType":"browserid",
"assertion": {
"digest":"<hash of fingerprint and session IDs>",
"audience": "[TBD]"
"valid-until": 1308859352261,
}, // signed using user's key
"certificate": {
"email": "rescorla@gmail.com",
"public-key": "<ekrs-public-key>",
"valid-until": 1308860561861,
} // certificate is signed by gmail.com
}
}
Note that we only expect to sign the fingerprint values and the
session IDs, in order to allow the JS or calling service to modify
the rest of the SDP, while protecting the identity binding. [OPEN
ISSUE: should we sign seq too?]
[TODO: NEed to talk about Audience a bit.]
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On Bob's side, he receives the signed assertion as part of the call
setup message and a similar procedure happens to verify it.
1. The calling JS instantiates a PeerConnection and provides it the
relevant signaling information, including the signed assertion.
2. The PeerConnection instantiates a BrowserID verifier in an IFRAME
and provides it the signed assertion.
3. The BrowserID verifier contacts the identity provider to verify
the certificate and then uses the key to verify the signed
fingerprint.
4. Alice's verified identity is returned to the PeerConnection (it
already has the fingerprint).
5. At this point, Bob's browser can display a trusted UI indication
that Alice is on the other end of the call.
When Bob returns his answer, he follows the converse procedure, which
provides Alice with a signed assertion of Bob's identity and keying
material.
A.3.6.3. OAuth
While OAuth is not directly designed for user-to-user authentication,
with a little lateral thinking it can be made to serve. We use the
following mapping of OAuth concepts to RTC-Web concepts:
+----------------------+----------------------+
| OAuth | RTCWeb |
+----------------------+----------------------+
| Client | Relying party |
| Resource owner | Authenticating party |
| Authorization server | Identity service |
| Resource server | Identity service |
+----------------------+----------------------+
Table 1
The idea here is that when Alice wants to authenticate to Bob (i.e.,
for Bob to be aware that she is calling). In order to do this, she
allows Bob to see a resource on the identity provider that is bound
to the call, her identity, and her public key. Then Bob retrieves
the resource from the identity provider, thus verifying the binding
between Alice and the call.
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Alice IDP Bob
---------------------------------------------------------
Call-Id, Fingerprint ------->
<------------------- Auth Code
Auth Code ---------------------------------------------->
<----- Get Token + Auth Code
Token --------------------->
<------------- Get call-info
Call-Id, Fingerprint ------>
This is a modified version of a common OAuth flow, but omits the
redirects required to have the client point the resource owner to the
IDP, which is acting as both the resource server and the
authorization server, since Alice already has a handle to the IDP.
Above, we have referred to "Alice", but really what we mean is the
PeerConnection. Specifically, the PeerConnection will instantiate an
IFRAME with JS from the IDP and will use that IFRAME to communicate
with the IDP, authenticating with Alice's identity (e.g., cookie).
Similarly, Bob's PeerConnection instantiates an IFRAME to talk to the
IDP.
A.3.6.4. Generic Identity Support
I believe it's possible to build a generic interface between the
PeerConnection and any identity sub-module so that the PeerConnection
just gets pointed to the IDP (which the relying party either trusts
or not) and JS from the IDP provides the concrete interfaces.
However, I need to work out the details, so I'm not specifying this
yet. If it works, the previous two sections will just be examples.
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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