RTCWEB E. Rescorla
Internet-Draft RTFM, Inc.
Intended status: Standards Track March 12, 2012
Expires: September 13, 2012
RTCWEB Security Architecture
draft-ietf-rtcweb-security-arch-01
Abstract
The Real-Time Communications on the Web (RTCWEB) working group is
tasked with standardizing protocols for real-time communications
between Web browsers. The major use cases for RTCWEB 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) RTCWEB 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. [I-D.ietf-
rtcweb-security] defines the RTCWEB threat model. This document
defines an architecture which provides security within that threat
model.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4
3. Trust Model . . . . . . . . . . . . . . . . . . . . . . . . . 4
3.1. Authenticated Entities . . . . . . . . . . . . . . . . . . 5
3.2. Unauthenticated Entities . . . . . . . . . . . . . . . . . 5
4. Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . 6
4.1. Initial Signaling . . . . . . . . . . . . . . . . . . . . 7
4.2. Media Consent Verification . . . . . . . . . . . . . . . . 9
4.3. DTLS Handshake . . . . . . . . . . . . . . . . . . . . . . 10
4.4. Communications and Consent Freshness . . . . . . . . . . . 10
5. Detailed Technical Description . . . . . . . . . . . . . . . . 10
5.1. Origin and Web Security Issues . . . . . . . . . . . . . . 10
5.2. Device Permissions Model . . . . . . . . . . . . . . . . . 11
5.3. Communications Consent . . . . . . . . . . . . . . . . . . 12
5.4. IP Location Privacy . . . . . . . . . . . . . . . . . . . 13
5.5. Communications Security . . . . . . . . . . . . . . . . . 13
5.6. Web-Based Peer Authentication . . . . . . . . . . . . . . 15
6. Security Considerations . . . . . . . . . . . . . . . . . . . 16
6.1. Communications Security . . . . . . . . . . . . . . . . . 16
6.2. Privacy . . . . . . . . . . . . . . . . . . . . . . . . . 16
6.3. Denial of Service . . . . . . . . . . . . . . . . . . . . 17
7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 18
8. References . . . . . . . . . . . . . . . . . . . . . . . . . . 18
8.1. Normative References . . . . . . . . . . . . . . . . . . . 18
8.2. Informative References . . . . . . . . . . . . . . . . . . 19
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 19
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1. Introduction
The Real-Time Communications on the Web (RTCWEB) working group is
tasked with standardizing protocols for real-time communications
between Web browsers. The major use cases for RTCWEB 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) RTCWEB communications are directly
controlled by some Web server, as shown in Figure 1.
+----------------+
| |
| Web Server |
| |
+----------------+
^ ^
/ \
HTTP / \ HTTP
/ \
/ \
v v
JS API JS API
+-----------+ +-----------+
| | Media | |
| Browser |<---------->| Browser |
| | | |
+-----------+ +-----------+
Figure 1: A simple RTCWEB system
This system presents a number of new security challenges, which are
analyzed in [I-D.ietf-rtcweb-security]. This document describes a
security architecture for RTCWEB which addresses the threats and
requirements described in that document.
2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [RFC2119].
3. Trust Model
The basic assumption of this architecture 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
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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.
3.1. 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: RTCWEB 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 (perhaps to
arrange for ransom payment), it's safe to temporarily give him access
to the camera and microphone for the purpose of the call, but he
doesn't want Dr. Evil to be able to access his camera and microphone
other than during the call. The point here is that we must first
identify other elements before we can determine whether and how much
to trust them.
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.
3.2. 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
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it means that we must assume that they will behave maliciously and
design a system which is secure even if they do so.
4. 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
4.1. 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
[I-D.jennings-rtcweb-signaling] or separate media description and
transport messages as defined in [I-D.ietf-rtcweb-jsep] 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 communication to Alice's
public key [RFC5763]
[Note that it is currently unclear where JSEP will eventually put
this information, in the SDP or in the transport info.] 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
[I-D.rescorla-rtcweb-generic-idp] PeerConnection can be agnostic to
them), but for now it's easiest to think of as a BrowserID assertion.
The assertion may bind other information to the identity besides the
fingerprint, but at minimum it needs to bind the fingerprint.
This message is sent to the signaling server, e.g., by XMLHttpRequest
[XmlHttpRequest] or by WebSockets [RFC6455] 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
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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 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.
4.2. Media Consent Verification
As described in ([I-D.ietf-rtcweb-security]; Section 4.2) This
proposal specifies that media consent verification 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.
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4.3. 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
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 exported [RFC5705] 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.
4.4. 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.
5. Detailed Technical Description
5.1. Origin and Web Security Issues
The basic unit of permissions for RTCWEB is the origin [RFC6454].
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 [RFC2818].
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Thus, clients MUST treat HTTP and HTTPS origins as different
permissions domains. [Note: this follows directly from the origin
security model and is stated here merely for clarity.]
Many web browsers currently forbid by default any active mixed
content on HTTPS pages. I.e., when JS is loaded from an HTTP origin
onto an HTTPS page, an error is displayed and the content is not
executed unless the user overrides the error. Any browser which
enforces such a policy will also not permit access to RTCWEB
functionality from mixed content pages. It is RECOMMENDED that
browsers which allow active mixed content nevertheless disable RTCWEB
functionality in mixed content settings. [[ OPEN ISSUE: Should this
be a 2119 MUST? It's not clear what set of conditions would make
this OK, other than that browser manufacturers have traditionally
been permissive here here.]] Note that it is possible for a page
which was not mixed content to become mixed content during the
duration of the call. Implementations MAY choose to terminate the
call or display a warning at that point, but it is also permissible
to ignore this condition. This is a deliberate implementation
complexity versus security tradeoff.
5.2. 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 it SHOULD display the
appropriate UI for those permissions and MUST clearly indicate
what permissions are being requested.
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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. [Note: this may not be necessary in
systems that are non-windows-based but that have good
notifications support, such as phones.]
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 Section 5.3 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. Section 5.5 provides more on this.
5.3. Communications Consent
Browser client implementations of RTCWEB 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
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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. Instead,
the consent freshness mechanism [I-D.muthu-behave-consent-freshness]
MUST be used.
5.4. IP Location Privacy
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 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 and also from learning whether the
user is online.
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. The
API MUST provide a mechanism for the calling application to
reconfigure an existing call to add non-TURN candidates. Taken
together, these requirements allow ICE negotiation to start
immediately on incoming call notification, thus reducing post-dial
delay, but also to avoid disclosing the user's IP address until
they have decided to answer.
5.5. Communications Security
Implementations MUST implement DTLS [RFC4347] and DTLS-SRTP
[RFC5763][RFC5764]. 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.
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[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]
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 (see Section 5.6) 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.
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* 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.
5.6. Web-Based Peer Authentication
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
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. Thus, the browser MUST
securely generate the input to the IdP assertion process and MUST
securely display the results of the verification process to the user
in a way which cannot be imitated by the calling site.
In order to make this work, we must standardize the following items:
o The precise information from the signaling message that must be
cryptographically bound to the user's identity. At minimum this
MUST be the fingerprint, but we may choose to add other
information as the signaling protocol firms up. This will be
defined in a future version of this document.
o The interface to the IdP. [I-D.rescorla-rtcweb-generic-idp]
specifies a specific protocol mechanism which allows the use of
any identity protocol without requiring specific further protocol
support in the browser.
o The JavaScript interfaces which the calling application can use to
specify the IdP to use to generate assertions and to discover what
assertions were received. These interfaces should be defined in
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the W3C document.
6. Security Considerations
Much of the security analysis of this problem is contained in
[I-D.ietf-rtcweb-security] or in the discussion of the particular
issues above. In order to avoid repetition, this section focuses on
(a) residual threats that are not addressed by this document and (b)
threats produced by failure/misbehavior of one of the components in
the system.
6.1. Communications Security
While this document favors DTLS-SRTP, it permits a variety of
communications security mechanisms and thus the level of
communications security actually provided varies considerably. Any
pair of implementations which have multiple security mechanisms in
common are subject to being downgraded to the weakest of those common
mechanisms by any attacker who can modify the signaling traffic. If
communications are over HTTP, this means any on-path attacker. If
communications are over HTTPS, this means the signaling server.
Implementations which wish to avoid downgrade attack should only
offer the strongest available mechanism, which is DTLS/DTLS-SRTP.
Note that the implication of this choice will be that interop to non-
DTLS-SRTP devices will need to happen through gateways.
Even if only DTLS/DTLS-SRTP are used, the signaling server can
potentially mount a man-in-the-middle attack unless implementations
have some mechanism for independently verifying keys. The UI
requirements in Section 5.5 are designed to provide such a mechanism
for motivated/security conscious users, but are not suitable for
general use. The identity service mechanisms in Section 5.6 are more
suitable for general use. Note, however, that a malicious signaling
service can strip off any such identity assertions, though it cannot
forge new ones.
6.2. Privacy
The requirements in this document are intended to allow:
o Users to participate in calls without revealing their location.
o Potential callees to avoid revealing their location and even
presence status prior to agreeing to answer a call.
However, these privacy protections come at a performance cost in
terms of using TURN relays and, in the latter case, delaying ICE.
Sites SHOULD make users aware of these tradeoffs.
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Note that the protections provided here assume a non-malicious
calling service. As the calling service always knows the users
status and (absent the use of a technology like Tor) their IP
address, they can violate the users privacy at will. Users who wish
privacy against the calling sites they are using must use separate
privacy enhancing technologies such as Tor. Combined RTCWEB/Tor
implementations SHOULD arrange to route the media as well as the
signaling through Tor. [Currently this will produce very suboptimal
performance.]
6.3. Denial of Service
The consent mechanisms described in this document are intended to
mitigate denial of service attacks in which an attacker uses clients
to send large amounts of traffic to a victim without the consent of
the victim. While these mechanisms are sufficient to protect victims
who have not implemented RTCWEB at all, RTCWEB implementations need
to be more careful.
Consider the case of a call center which accepts calls via RTCWeb.
An attacker proxies the call center's front-end and arranges for
multiple clients to initiate calls to the call center. Note that
this requires user consent in many cases but because the data channel
does not need consent, he can use that directly. Since ICE will
complete, browsers can then be induced to send large amounts of data
to the victim call center if it supports the data channel at all.
Preventing this attack requires that automated RTCWEB
implemementations implement sensible flow control and have the
ability to triage out (i.e., stop responding to ICE probes on) calls
which are behaving badly, and especially to be prepared to remotely
throttle the data channel in the absence of plausible audio and video
(which the attacker cannot control).
Another related attack is for the signaling service to swap the ICE
candidates for the audio and video streams, thus forcing a browser to
send video to the sink that the other victim expects will contain
audio (perhaps it is only expecting audio!) potentially causing
overload. Muxing multiple media flows over a single transport makes
it harder to individually suppress a single flow by denying ICE
keepalives. Media-level (RTCP) mechanisms must be used in this case.
[TODO: Write up Magnus's ICE forking attack when we get some clarity
on it.]
Note that attacks based on confusing one end or the other about
consent are possible primarily even in the face of the third-party
identity mechanism as long as major parts of the signaling messages
are not signed. On the other hand, signing the entire message
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severely restricts the capabilities of the calling application, so
there are difficult tradeoffs here.
7. Acknowledgements
Bernard Aboba, Harald Alvestrand, Cullen Jennings, Hadriel Kaplan,
Matthew Kaufman, Magnus Westerland.
8. References
8.1. Normative References
[I-D.ietf-rtcweb-security]
Rescorla, E., "Security Considerations for RTC-Web",
draft-ietf-rtcweb-security-01 (work in progress),
October 2011.
[I-D.muthu-behave-consent-freshness]
Perumal, M., Wing, D., and H. Kaplan, "STUN Usage for
Consent Freshness",
draft-muthu-behave-consent-freshness-00 (work in
progress), March 2012.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC2818] Rescorla, E., "HTTP Over TLS", RFC 2818, May 2000.
[RFC4347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer
Security", RFC 4347, April 2006.
[RFC5245] Rosenberg, J., "Interactive Connectivity Establishment
(ICE): A Protocol for Network Address Translator (NAT)
Traversal for Offer/Answer Protocols", RFC 5245,
April 2010.
[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.
[RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer
Security (DTLS) Extension to Establish Keys for the Secure
Real-time Transport Protocol (SRTP)", RFC 5764, May 2010.
[RFC6454] Barth, A., "The Web Origin Concept", RFC 6454,
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December 2011.
8.2. Informative References
[I-D.ietf-rtcweb-jsep]
Uberti, J. and C. Jennings, "Javascript Session
Establishment Protocol", draft-ietf-rtcweb-jsep-00 (work
in progress), March 2012.
[I-D.jennings-rtcweb-signaling]
Jennings, C., Rosenberg, J., and R. Jesup, "RTCWeb Offer/
Answer Protocol (ROAP)",
draft-jennings-rtcweb-signaling-01 (work in progress),
October 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.
[I-D.rescorla-rtcweb-generic-idp]
Rescorla, E., "RTCWeb Generic Identity Provider
Interface", draft-rescorla-rtcweb-generic-idp-00 (work in
progress), January 2012.
[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.
[RFC5705] Rescorla, E., "Keying Material Exporters for Transport
Layer Security (TLS)", RFC 5705, March 2010.
[RFC6455] Fette, I. and A. Melnikov, "The WebSocket Protocol",
RFC 6455, December 2011.
[XmlHttpRequest]
van Kesteren, A., "XMLHttpRequest Level 2".
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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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