RTCWEB E. Rescorla
Internet-Draft RTFM, Inc.
Intended status: Standards Track July 14, 2013
Expires: January 15, 2014
WebRTC Security Architecture
draft-ietf-rtcweb-security-arch-07
Abstract
The Real-Time Communications on the Web (RTCWEB) working group is
tasked with standardizing protocols for enabling real-time
communications within user-agents using web technologies (commonly
called "WebRTC"). This document defines the security architecture
for
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 5
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 6
3. Trust Model . . . . . . . . . . . . . . . . . . . . . . . . . 6
3.1. Authenticated Entities . . . . . . . . . . . . . . . . . . 7
3.2. Unauthenticated Entities . . . . . . . . . . . . . . . . . 7
4. Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . 7
4.1. Initial Signaling . . . . . . . . . . . . . . . . . . . . 10
4.2. Media Consent Verification . . . . . . . . . . . . . . . . 12
4.3. DTLS Handshake . . . . . . . . . . . . . . . . . . . . . . 13
4.4. Communications and Consent Freshness . . . . . . . . . . . 13
5. Detailed Technical Description . . . . . . . . . . . . . . . . 14
5.1. Origin and Web Security Issues . . . . . . . . . . . . . . 14
5.2. Device Permissions Model . . . . . . . . . . . . . . . . . 14
5.3. Communications Consent . . . . . . . . . . . . . . . . . . 16
5.4. IP Location Privacy . . . . . . . . . . . . . . . . . . . 17
5.5. Communications Security . . . . . . . . . . . . . . . . . 18
5.6. Web-Based Peer Authentication . . . . . . . . . . . . . . 19
5.6.1. Trust Relationships: IdPs, APs, and RPs . . . . . . . 20
5.6.2. Overview of Operation . . . . . . . . . . . . . . . . 22
5.6.3. Items for Standardization . . . . . . . . . . . . . . 23
5.6.4. Binding Identity Assertions to JSEP Offer/Answer
Transactions . . . . . . . . . . . . . . . . . . . . . 23
5.6.4.1. Input to Assertion Generation Process . . . . . . 23
5.6.4.2. Carrying Identity Assertions . . . . . . . . . . . 24
5.6.5. IdP Interaction Details . . . . . . . . . . . . . . . 25
5.6.5.1. General Message Structure . . . . . . . . . . . . 25
5.6.5.2. IdP Proxy Setup . . . . . . . . . . . . . . . . . 26
5.7. Security Considerations . . . . . . . . . . . . . . . . . 30
5.7.1. Communications Security . . . . . . . . . . . . . . . 30
5.7.2. Privacy . . . . . . . . . . . . . . . . . . . . . . . 31
5.7.3. Denial of Service . . . . . . . . . . . . . . . . . . 32
5.7.4. IdP Authentication Mechanism . . . . . . . . . . . . . 33
5.7.4.1. PeerConnection Origin Check . . . . . . . . . . . 33
5.7.4.2. IdP Well-known URI . . . . . . . . . . . . . . . . 34
5.7.4.3. Privacy of IdP-generated identities and the
hosting site . . . . . . . . . . . . . . . . . . . 34
5.7.4.4. Security of Third-Party IdPs . . . . . . . . . . . 35
5.7.4.5. Web Security Feature Interactions . . . . . . . . 35
5.8. IANA Considerations . . . . . . . . . . . . . . . . . . . 35
6. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 36
7. Changes . . . . . . . . . . . . . . . . . . . . . . . . . . . 36
7.1. Changes since -06 . . . . . . . . . . . . . . . . . . . . 36
7.2. Changes since -05 . . . . . . . . . . . . . . . . . . . . 36
7.3. Changes since -03 . . . . . . . . . . . . . . . . . . . . 36
7.4. Changes since -03 . . . . . . . . . . . . . . . . . . . . 36
7.5. Changes since -02 . . . . . . . . . . . . . . . . . . . . 37
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8. References . . . . . . . . . . . . . . . . . . . . . . . . . . 37
8.1. Normative References . . . . . . . . . . . . . . . . . . . 37
8.2. Informative References . . . . . . . . . . . . . . . . . . 38
Appendix A. Example IdP Bindings to Specific Protocols . . . . . 39
A.1. BrowserID . . . . . . . . . . . . . . . . . . . . . . . . 39
A.2. OAuth . . . . . . . . . . . . . . . . . . . . . . . . . . 42
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 43
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1. Introduction
The Real-Time Communications on the Web (WebRTC) working group is
tasked with standardizing protocols for real-time communications
between Web browsers. The major use cases for WebRTC 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) WebRTC communications are directly
controlled by some Web server, via a JavaScript (JS) API as shown in
Figure 1.
+----------------+
| |
| Web Server |
| |
+----------------+
^ ^
/ \
HTTP / \ HTTP
/ \
/ \
v v
JS API JS API
+-----------+ +-----------+
| | Media | |
| Browser |<---------->| Browser |
| | | |
+-----------+ +-----------+
Figure 1: A simple WebRTC system
A more complicated system might allow for interdomain calling, as
shown in Figure 2. The protocol to be used between the domains is
not standardized by WebRTC, but given the installed base and the form
of the WebRTC API is likely to be something SDP-based like SIP.
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+--------------+ +--------------+
| | SIP,XMPP,...| |
| Web Server |<----------->| Web Server |
| | | |
+--------------+ +--------------+
^ ^
| |
HTTP | | HTTP
| |
v v
JS API JS API
+-----------+ +-----------+
| | Media | |
| Browser |<---------------->| Browser |
| | | |
+-----------+ +-----------+
Figure 2: A multidomain WebRTC 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 WebRTC 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
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
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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.
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, but in some cases because we are on a topologically
restricted network, such as behind a firewall, and can infer
authentication from firewall behavior).
o Other users: WebRTC 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 and 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. Additionally, sometimes we need to
identify the communicating peer before we know what policies to
apply.
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
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
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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
figures 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 as OpenID
or BrowserID) that can be used to demonstrate their identity to other
parties. For instance, Alice might have an account with a social
network which she can then use to authenticate to other web sites
without explicitly having an account with those sites; this is a
fairly conventional pattern on the Web. Section 5.6.1 provides an
overview of Identity Providers and the relevant terminology. Alice
and Bob might have relationships with different IdPs as well.
This separation of identity provision and signaling 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 domain (Figure 3) However, there are important settings where
that is not the case, such as federation (calls from one domain to
another; Figure 4) 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.
Note that the servers themselves are also authenticated by an
external identity service, the SSL/TLS certificate infrastructure
(not shown). As is conventional in the Web, all identities are
ultimately rooted in that system. For instance, when an IdP makes an
identity assertion, the Relying Party consuming that assertion is
able to verify because it is able to connect to the IdP via HTTPS.
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+----------------+
| |
| Signaling |
| Server |
| |
+----------------+
^ ^
/ \
HTTPS / \ HTTPS
/ \
/ \
v v
JS API JS API
+-----------+ +-----------+
| | Media | |
Alice | Browser |<---------->| Browser | Bob
| | (DTLS+SRTP)| |
+-----------+ +-----------+
^ ^--+ +--^ ^
| | | |
v | | v
+-----------+ | | +-----------+
| |<--------+ | |
| IdP1 | | | IdP2 |
| | +------->| |
+-----------+ +-----------+
Figure 3: A call with IdP-based identity
Figure 4 shows essentially the same calling scenario but with a call
between two separate domains (i.e., a federated case), as in
Figure 2. As mentioned above, the domains communicate by some
unspecified protocol and providing separate signaling and identity
allows for calls to be authenticated regardless of the details of the
inter-domain protocol.
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+----------------+ Unspecified +----------------+
| | protocol | |
| Signaling |<----------------->| Signaling |
| Server | (SIP, XMPP, ...) | Server |
| | | |
+----------------+ +----------------+
^ ^
| |
HTTPS | | HTTPS
| |
| |
v v
JS API JS API
+-----------+ +-----------+
| | Media | |
Alice | Browser |<--------------------------->| Browser | Bob
| | DTLS+SRTP | |
+-----------+ +-----------+
^ ^--+ +--^ ^
| | | |
v | | v
+-----------+ | | +-----------+
| |<-------------------------+ | |
| IdP1 | | | IdP2 |
| | +------------------------>| |
+-----------+ +-----------+
Figure 4: A federated call with IdP-based identity
4.1. Initial Signaling
For simplicity, assume the topology in Figure 3. 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 a user's ordinary accounts with
some service. In this example, we show Alice and Bob using a
separate identity service, though the identity service may be the
same entity as the calling service or there may be 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
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service presents a JS UI in the form of a button next to Bob's name
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 a
MediaStream with two MediaStreamTracks, 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, so the browser prompts Alice for
permission.
In the current W3C API, once some streams have been added, Alice's
browser + JS generates a signaling message [I-D.ietf-rtcweb-jsep]
containing:
o Media channel information
o Interactive Connectivity Establishment (ICE) [RFC5245] candidates
o A fingerprint attribute binding the communication to a key pair
[RFC5763]. Note that this key may simply be ephemerally generated
for this call or specific to this domain, and Alice may have a
large number of such keys.
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 Section 5.6 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]. preferably over TLS
[RFC5246]. 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 verify the assertion. This allows
the browser to display a trusted element in the browser chrome
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
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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 a PeerConnection is instantiated with the message from
Alice's side. Then, a similar process occurs as on Alice's browser:
Bob's browser prompts him for device permission, 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. This level of security is
provided merely by having the fingerprint in the message and having
that message received securely from the signaling server. Because
the far end sent an identity assertion along with their message, they
know that this is verifiable from the IdP as well. Note that if the
call is federated, as shown in Figure 4 then Alice is able to verify
Bob's identity in a way that is not mediated by either her signaling
server or Bob's. Rather, she verifies it directly with Bob's IdP.
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. I.e., they simply have whatever information their
calling site claims about the caller/calllee's identity. 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) media
consent verification is provided 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 between Alice and
Bob but not attached to the signaling service to spoof these checks
because they do not have the ICE credentials. Bob has the same
security guarantees with respect to Alice.
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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 or
channels. This is performed via DTLS [RFC4347] (for the data
channel) and DTLS-SRTP [RFC5763] keying for SRTP [RFC3711] for the
media channel and SCTP over DTLS [I-D.ietf-tsvwg-sctp-dtls-encaps]
for data channels. 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 mounting a man-
in-the-middle attack on their traffic. 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
(i.e., that the signaling service cannot mount a passive attack on
the communications).
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 so that Alice does not continue to send
large traffic volumes to entities which went abruptly offline. 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. As described in
Section 5.3, these keepalives MUST be based on the consent freshness
mechanism specified in [I-D.muthu-behave-consent-freshness]. If a
keepalive fails and no new ICE channels can be established, then the
session is terminated.
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5. Detailed Technical Description
5.1. Origin and Web Security Issues
The basic unit of permissions for WebRTC 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].
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. That is, when JavaScript is loaded from an
HTTP origin onto an HTTPS page, an error is displayed and the HTTP
content is not executed unless the user overrides the error. Any
browser which enforces such a policy will also not permit access to
WebRTC functionality from mixed content pages (because they never
display mixed content). Browsers which allow active mixed content
MUST nevertheless disable WebRTC functionality in mixed content
settings.
Note that it is possible for a page which was not mixed content to
become mixed content during the duration of the call. The major risk
here is that the newly arrived insecure JS might redirect media to a
location controlled by the attacker. Implementations MUST either
choose to terminate the call or display a warning at that point.
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 for HTTPS
origins.
o Requests for one-time camera/microphone access.
o Requests for permanent access.
Because HTTP origins cannot be securely established against network
attackers, implementations MUST NOT allow the setting of permanent
access permissions for HTTP origins. Implementations MAY also opt to
refuse all permissions grants for HTTP origins, but it is RECOMMENDED
that currently they support one-time camera/microphone access.
In addition, they SHOULD support requests for access that promise
that media from this grant will be sent to a single communicating
peer (obviously there could be other requests for other peers).
E.g., "Call customerservice@ford.com". The semantics of this request
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are that the media stream from the camera and microphone will only be
routed through a connection which has been cryptographically verified
(through the IdP mechanism or an X.509 certificate in the DTLS-SRTP
handshake) as being associated with the stated identity. Note that
it is unlikely that browsers would have an X.509 certificate, but
servers might. Browsers servicing such requests SHOULD clearly
indicate that identity to the user when asking for permission. The
idea behind this type of permissions is that a user might have a
fairly narrow list of peers he is willing to communicate with, e.g.,
"my mother" rather than "anyone on Facebook". Narrow permissions
grants allow the browser to do that enforcement.
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 browser client to know what sort of
user interface experience to provide to the user, including what
permissions to request from the user and hence what to enforce
later. For instance, 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.
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
device access, 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 when the indication is hidden. [Note:
this may not be necessary in systems that are non-windows-based
but that have good notifications support, such as phones.]
[[OPEN ISSUE: This section does not have WG consensus. Because
screen/application sharing presents a more significant risk than
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camera and microphone access (see the discussion in
[I-D.ietf-rtcweb-security] S 4.1.1), we require a higher level of
user consent.
o Browsers MUST not permit permanent screen or application sharing
permissions to be installed as a response to a JS request for
permissions. Instead, they must require some other user action
such as a permissions setting or an application install experience
to grant permission to a site.
o Browsers MUST provide a separate dialog request for screen/
application sharing permissions even if the media request is made
at the same time as camera and microphone.
o The browser MUST indicate any windows which are currently being
shared in some unambiguous way. Windows which are not visible
MUST not be shared even if the application is being shared. If
the screen is being shared, then that MUST be indicated.
-- END OF OPEN ISSUE]]
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 communicating peers. 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 WebRTC MUST implement ICE. Server
gateway implementations which operate only at public IP addresses
MUST implement either full ICE or ICE-Lite [RFC5245].
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 during the
lifetime of the transaction (i.e., during the period when the ICE
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stack would accept a new response for that transaction). The JS MUST
NOT be permitted to control the local ufrag and password, though it
of course knows it.
While continuing consent is required, that ICE [RFC5245]; Section 10
keepalives STUN Binding Indications are one-way and therefore not
sufficient. The current WG consensus is to use ICE Binding Requests
for continuing consent freshness. ICE already requires that
implementations respond to such requests, so this approach is
maximally compatible. A separate document will profile the ICE
timers to be used; see [I-D.muthu-behave-consent-freshness].
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.
This has negative privacy consequences in some circumstances. The
API requirements in this section are intended to mitigate this issue.
Note that these requirements are NOT intended to protect the user's
IP address from a malicious site. In general, the site will learn at
least a user's server reflexive address from any HTTP transaction.
Rather, these requirements are intended to allow a site to cooperate
with the user to hide the user's IP address from the other side of
the call. Hiding the user's IP address from the server requires some
sort of explicit privacy preserving mechanism on the client (e.g.,
Torbutton [https://www.torproject.org/torbutton/]) and is out of
scope for this specification.
API Requirement: The API MUST provide a mechanism to allow the JS 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 JS to indicate that only TURN candidates are to be
used. This prevents the peer from learning one's IP address at
all. This mechanism MUST also permit suppression of the related
address field, since that leaks local addresses.
API Requirement: The API MUST provide a mechanism for the calling
application to reconfigure an existing call to add non-TURN
candidates. Taken together, this and the previous requirement
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
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answer. They also allow users to completely hide their IP address
for the duration of the call. Finally, they allow a mechanism for
the user to optimize performance by reconfiguring to allow non-
turn candidates during an active call if the user decides they no
longer need to hide their IP address
Note that some enterprises may operate proxies and/or NATs designed
to hide internal IP addresses from the outside world. WebRTC
provides no explicit mechanism to allow this function. Either such
enterprises need to proxy the HTTP/HTTPS and modify the SDP and/or
the JS, or there needs to be browser support to set the "TURN-only"
policy regardless of the site's preferences.
5.5. Communications Security
Implementations MUST implement SRTP [RFC3711]. Implementations MUST
implement DTLS [RFC4347] and DTLS-SRTP [RFC5763][RFC5764] for SRTP
keying. Implementations MUST implement
[I-D.ietf-tsvwg-sctp-dtls-encaps].
All media channels MUST be secured via SRTP. Media traffic MUST NOT
be sent over plain (unencrypted) RTP. 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, DTLS-SRTP
MUST be selected.
All data channels MUST be secured via DTLS.
[OPEN ISSUE: What should the settings be here? MUST?]
Implementations MAY support SDES 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. Unless the user specifically configures an
external key pair, different key pairs MUST be used for each
origin. (This avoids creating a super-cookie.)
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.
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UI Requirements: A user-oriented client MUST provide an
"inspector" interface which allows the user to determine the
security characteristics of the media.
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 the cryptographic keys were delivered out-of-band (from
a server) or were generated as a result of a pairwise
negotiation.
* If the far endpoint was directly verified, either via a third-
party verifiable X.509 certificate or via a Web IdP mechanism
(see Section 5.6) the "security characteristics" MUST include
the verified information. X.509 identities and Web IdP
identities have similar semantics and should be displayed in a
similar way.
The following properties are more likely to require some "drill-
down" from the user:
* The "security characteristics" MUST indicate the cryptographic
algorithms in use (For example: "AES-CBC" or "Null Cipher".)
However, if Null ciphers are used, that MUST be presented to
the user at the top-level UI.
* The "security characteristics" MUST indicate whether PFS is
provided.
* 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,
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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.
The mechanisms defined in this document do not require the browser to
implement any particular identity protocol or to support any
particular IdP. Instead, this document provides a generic interface
which any IdP can implement. Thus, new IdPs and protocols can be
introduced without change to either the browser or the calling
service. This avoids the need to make a commitment to any particular
identity protocol, although browsers may opt to directly implement
some identity protocols in order to provide superior performance or
UI properties.
5.6.1. Trust Relationships: IdPs, APs, and RPs
Any federated identity protocol has three major participants:
Authenticating Party (AP): The entity which is trying to establish
its identity.
Identity Provider (IdP): The entity which is vouching for the AP's
identity.
Relying Party (RP): The entity which is trying to verify the AP's
identity.
The AP and the IdP have an account relationship of some kind: the AP
registers with the IdP and is able to subsequently authenticate
directly to the IdP (e.g., with a password). This means that the
browser must somehow know which IdP(s) the user has an account
relationship with. This can either be something that the user
configures into the browser or that is configured at the calling site
and then provided to the PeerConnection by the Web application at the
calling site. The use case for having this information configured
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into the browser is that the user may "log into" the browser to bind
it to some identity. This is becoming common in new browsers.
However, it should also be possible for the IdP information to simply
be provided by the calling application.
At a high level there are two kinds of IdPs:
Authoritative: IdPs which have verifiable control of some section
of the identity space. For instance, in the realm of e-mail, the
operator of "example.com" has complete control of the namespace
ending in "@example.com". Thus, "alice@example.com" is whoever
the operator says it is. Examples of systems with authoritative
identity providers include DNSSEC, RFC 4474, and Facebook Connect
(Facebook identities only make sense within the context of the
Facebook system).
Third-Party: IdPs which don't have control of their section of the
identity space but instead verify user's identities via some
unspecified mechanism and then attest to it. Because the IdP
doesn't actually control the namespace, RPs need to trust that the
IdP is correctly verifying AP identities, and there can
potentially be multiple IdPs attesting to the same section of the
identity space. Probably the best-known example of a third-party
identity provider is SSL certificates, where there are a large
number of CAs all of whom can attest to any domain name.
If an AP is authenticating via an authoritative IdP, then the RP does
not need to explicitly configure trust in the IdP at all. The
identity mechanism can directly verify that the IdP indeed made the
relevant identity assertion (a function provided by the mechanisms in
this document), and any assertion it makes about an identity for
which it is authoritative is directly verifiable. Note that this
does not mean that the IdP might not lie, but that is a
trustworthiness judgement that the user can make at the time he looks
at the identity.
By contrast, if an AP is authenticating via a third-party IdP, the RP
needs to explicitly trust that IdP (hence the need for an explicit
trust anchor list in PKI-based SSL/TLS clients). The list of
trustable IdPs needs to be configured directly into the browser,
either by the user or potentially by the browser manufacturer. This
is a significant advantage of authoritative IdPs and implies that if
third-party IdPs are to be supported, the potential number needs to
be fairly small.
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5.6.2. Overview of Operation
In order to provide security without trusting the calling site, the
PeerConnection component of the browser must interact directly with
the IdP. The details of the mechanism are described in the W3C API
specification, but the general idea is that the PeerConnection
component downloads JS from a specific location on the IdP dictated
by the IdP domain name. That JS (the "IdP proxy") runs in an
isolated security context within the browser and the PeerConnection
talks to it via a secure message passing channel.
Note that there are two logically separate functions here:
o Identity assertion generation.
o Identity assertion verification.
The same IdP JS "endpoint" is used for both functions but of course a
given IdP might behave differently and load new JS to perform one
function or the other.
+------------------------------------+
| https://calling-site.example.com |
| |
| |
| |
| Calling JS Code |
| ^ |
| | API Calls |
| v |
| PeerConnection |
| ^ |
| | postMessage() |
| v |
| +-------------------------+ | +---------------+
| | https://idp.example.org | | | |
| | |<--------->| Identity |
| | IdP JS | | | Provider |
| | | | | |
| +-------------------------+ | +---------------+
| |
+------------------------------------+
When the PeerConnection object wants to interact with the IdP, the
sequence of events is as follows:
1. The browser (the PeerConnection component) instantiates an IdP
proxy with its source at the IdP. This allows the IdP to load
whatever JS is necessary into the proxy, which runs in the IdP's
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security context.
2. If the user is not already logged in, the IdP does whatever is
required to log them in, such as soliciting a username and
password.
3. Once the user is logged in, the IdP proxy notifies the browser
that it is ready.
4. The browser and the IdP proxy communicate via a standardized
series of messages delivered via postMessage. For instance, the
browser might request the IdP proxy to sign or verify a given
identity assertion.
This approach allows us to decouple the browser from any particular
identity provider; the browser need only know how to load the IdP's
JavaScript--which is deterministic from the IdP's identity--and the
generic protocol for requesting and verifying assertions. The IdP
provides whatever logic is necessary to bridge the generic protocol
to the IdP's specific requirements. Thus, a single browser can
support any number of identity protocols, including being forward
compatible with IdPs which did not exist at the time the browser was
written.
5.6.3. Items for Standardization
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 and a mechanism for
carrying assertions in JSEP messages. Section 5.6.4
o The interface to the IdP. Section 5.6.5 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.
The first two items are defined in this document. The final one is
defined in the companion W3C WebRTC API specification [webrtc-api].
5.6.4. Binding Identity Assertions to JSEP Offer/Answer Transactions
5.6.4.1. Input to Assertion Generation Process
As discussed above, an identity assertion binds the user's identity
(as asserted by the IdP) to the JSEP offer/exchange transaction and
specifically to the media. In order to achieve this, the
PeerConnection must provide the DTLS-SRTP fingerprint to be bound to
the identity. This is provided in a JSON structure for
extensibility, as shown below:
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{
"fingerprint" :
{
"algorithm":"SHA-1",
"digest":"4A:AD:B9:B1:3F:...:E5:7C:AB"
}
}
The "algorithm" and digest values correspond directly to the
algorithm and digest values in the a=fingerprint line of the SDP.
[RFC4572].
Note: this structure does not need to be interpreted by the IdP or
the IdP proxy. It is consumed solely by the RP's browser. The IdP
merely treats it as an opaque value to be attested to. Thus, new
parameters can be added to the assertion without modifying the IdP.
5.6.4.2. Carrying Identity Assertions
Once an IdP has generated an assertion, it is attached to the JSEP
message. This is done by adding a new a-line to the SDP, of the form
a=identity. The sole contents of this value are a base-64-encoded
version of the identity assertion. For example:
v=0
o=- 1181923068 1181923196 IN IP4 ua1.example.com
s=example1
c=IN IP4 ua1.example.com
a=setup:actpass
a=fingerprint: SHA-1 \
4A:AD:B9:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB
a=identity: \
ImlkcCI6eyJkb21haW4iOiAiZXhhbXBsZS5vcmciLCAicHJvdG9jb2wiOiAiYm9n \
dXMifSwiYXNzZXJ0aW9uIjpcIntcImlkZW50aXR5XCI6XCJib2JAZXhhbXBsZS5v \
cmdcIixcImNvbnRlbnRzXCI6XCJhYmNkZWZnaGlqa2xtbm9wcXJzdHV2d3l6XCIs \
XCJzaWduYXR1cmVcIjpcIjAxMDIwMzA0MDUwNlwifSJ9Cg==
t=0 0
m=audio 6056 RTP/SAVP 0
a=sendrecv
Each identity attribute should be paired (and attests to) with an
a=fingerprint attribute and therefore can exist either at the session
or media level. Multiple identity attributes may appear at either
level, though it is RECOMMENDED that implementations not do this,
because it becomes very unclear what security claim that they are
making and the UI guidelines above become unclear. Browsers MAY
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choose refuse to display any identity indicators in the face of
multiple identity attributes with different identities but SHOULD
process multiple attributes with the same identity as described
above.
5.6.5. IdP Interaction Details
5.6.5.1. General Message Structure
Messages between the PeerConnection object and the IdP proxy are
formatted using JSON [RFC4627]. For instance, the PeerConnection
would request a signature with the following "SIGN" message:
{
"type":"SIGN",
"id": "1",
"origin":"https://calling-site.example.com",
"message":"012345678abcdefghijkl"
}
All messages MUST contain a "type" field which indicates the general
meaning of the message.
All requests from the PeerConnection object MUST contain an "id"
field which MUST be unique for that PeerConnection object. Any
responses from the IdP proxy MUST contain the same id in response,
which allows the PeerConnection to correlate requests and responses,
in case there are multiple requests/responses outstanding to the same
proxy.
All requests from the PeerConnection object MUST contain an "origin"
field containing the origin of the JS which initiated the PC (i.e.,
the URL of the calling site). This origin value can be used by the
IdP to make access control decisions. For instance, an IdP might
only issue identity assertions for certain calling services in the
same way that some IdPs require that relying Web sites have an API
key before learning user identity.
Any message-specific data is carried in a "message" field. Depending
on the message type, this may either be a string or a richer JSON
object.
5.6.5.1.1. Errors
If an error occurs, the IdP sends a message of type "ERROR". The
message MAY have an "error" field containing freeform text data which
containing additional information about what happened. For instance:
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{
"id":"1",
"type":"ERROR",
"error":"Signature verification failed"
}
Figure 5: Example error
5.6.5.2. IdP Proxy Setup
In order to perform an identity transaction, the PeerConnection must
first create an IdP proxy. While the details of this are specified
in the W3C API document, from the perspective of this specification,
however, the relevant facts are:
o The JS runs in the IdP's security context with the base page
retrieved from the URL specified in Section 5.6.5.2.1
o The usual browser sandbox isolation mechanisms MUST be enforced
with respect to the IdP proxy.
o JS running in the IdP proxy MUST be able to send and receive
messages to the PeerConnection and the PC and IdP proxy are able
to verify the source and destination of these messages.
Initially the IdP proxy is in an unready state; the IdP JS must be
loaded and there may be several round trips to the IdP server, for
instance to log the user in. When the IdP proxy is ready to receive
commands, it delivers a "ready" message. As this message is
unsolicited, it simply contains:
{ "type":"READY" }
Once the PeerConnection object receives the ready message, it can
send commands to the IdP proxy.
5.6.5.2.1. Determining the IdP URI
In order to ensure that the IdP is under control of the domain owner
rather than someone who merely has an account on the domain owner's
server (e.g., in shared hosting scenarios), the IdP JavaScript is
hosted at a deterministic location based on the IdP's domain name.
Each IdP proxy instance is associated with two values:
domain name: The IdP's domain name
protocol: The specific IdP protocol which the IdP is using. This is
a completely IdP-specific string, but allows an IdP to implement
two protocols in parallel. This value may be the empty string.
Each IdP MUST serve its initial entry page (i.e., the one loaded by
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the IdP proxy) from the well-known URI specified in "/.well-known/
idp-proxy/<protocol>" on the IdP's web site. This URI MUST be loaded
via HTTPS [RFC2818]. For example, for the IdP "identity.example.com"
and the protocol "example", the URL would be:
https://example.com/.well-known/idp-proxy/example
5.6.5.2.1.1. Authenticating Party
How an AP determines the appropriate IdP domain is out of scope of
this specification. In general, however, the AP has some actual
account relationship with the IdP, as this identity is what the IdP
is attesting to. Thus, the AP somehow supplies the IdP information
to the browser. Some potential mechanisms include:
o Provided by the user directly.
o Selected from some set of IdPs known to the calling site. E.g., a
button that shows "Authenticate via Facebook Connect"
5.6.5.2.1.2. Relying Party
Unlike the AP, the RP need not have any particular relationship with
the IdP. Rather, it needs to be able to process whatever assertion
is provided by the AP. As the assertion contains the IdP's identity,
the URI can be constructed directly from the assertion, and thus the
RP can directly verify the technical validity of the assertion with
no user interaction. Authoritative assertions need only be
verifiable. Third-party assertions also MUST be verified against
local policy, as described in Section 5.6.5.2.3.1.
5.6.5.2.2. Requesting Assertions
In order to request an assertion, the PeerConnection sends a "SIGN"
message. Aside from the mandatory fields, this message has a
"message" field containing a string. The contents of this string are
defined above, but are opaque from the perspective of the IdP.
A successful response to a "SIGN" message contains a message field
which is a JS dictionary consisting of two fields:
idp: A dictionary containing the domain name of the provider and the
protocol string
assertion: An opaque field containing the assertion itself. This is
only interpretable by the idp or its proxy.
Figure 6 shows an example transaction, with the message "abcde..."
(remember, the messages are opaque at this layer) being signed and
bound to identity "ekr@example.org". In this case, the message has
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presumably been digitally signed/MACed in some way that the IdP can
later verify it, but this is an implementation detail and out of
scope of this document. Line breaks are inserted solely for
readability.
PeerConnection -> IdP proxy:
{
"type":"SIGN",
"id":1,
"origin":"https://calling-service.example.com/",
"message":"abcdefghijklmnopqrstuvwyz"
}
IdPProxy -> PeerConnection:
{
"type":"SUCCESS",
"id":1,
"message": {
"idp":{
"domain": "example.org"
"protocol": "bogus"
},
"assertion":\"{\"identity\":\"bob@example.org\",
\"contents\":\"abcdefghijklmnopqrstuvwyz\",
\"request_origin\":\"rtcweb://peerconnection\",
\"signature\":\"010203040506\"}"
}
}
Figure 6: Example assertion request
The message structure is serialized, base64-encoded, and placed in an
a=identity attribute.
5.6.5.2.3. Verifying Assertions
In order to verify an assertion, an RP sends a "VERIFY" message to
the IdP proxy containing the assertion supplied by the AP in the
"message" field.
The IdP proxy verifies the assertion. Depending on the identity
protocol, this may require one or more round trips to the IdP. For
instance, an OAuth-based protocol will likely require using the IdP
as an oracle, whereas with BrowserID the IdP proxy can likely verify
the signature on the assertion without contacting the IdP, provided
that it has cached the IdP's public key.
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Regardless of the mechanism, if verification succeeds, a successful
response from the IdP proxy MUST contain a message field consisting
of a dictionary/hash with the following fields:
identity The identity of the AP from the IdP's perspective. Details
of this are provided in Section 5.6.5.2.3.1
contents The original unmodified string provided by the AP in the
original SIGN request.
request_origin The original origin of the SIGN request on the AP
side as determined by the origin of the PostMessage call. The IdP
MUST somehow arrange to propagate this information as part of the
assertion. The receiving PeerConnection MUST verify that this
value is "rtcweb://peerconnection" (which implies that
PeerConnection must arrange that its messages to the IdP proxy are
from this origin.) See Section 5.7.4.1 for the security purpose
of this field. [[ OPEN ISSUE: Can a URI person help make a better
URI.]]
Figure 7 shows an example transaction. Line breaks are inserted
solely for readability.
PeerConnection -> IdP Proxy:
{
"type":"VERIFY",
"id":2,
"origin":"https://calling-service.example.com/",
"message":\"{\"identity\":\"bob@example.org\",
\"contents\":\"abcdefghijklmnopqrstuvwyz\",
\"request_origin\":\"rtcweb://peerconnection\",
\"signature\":\"010203040506\"}"
}
IdP Proxy -> PeerConnection:
{
"type":"SUCCESS",
"id":2,
"message": {
"identity" : {
"name" : "bob@example.org",
"displayname" : "Bob"
},
"request_origin":"rtcweb://peerconnection",
"contents":"abcdefghijklmnopqrstuvwyz"
}
}
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Figure 7: Example verification request
5.6.5.2.3.1. Identity Formats
Identities passed from the IdP proxy to the PeerConnection are
structured as JSON dictionaries with one mandatory field: "name".
This field MUST consist of an RFC822-formatted string representing
the user's identity. [[ OPEN ISSUE: Would it be better to have a
typed field? ]] The PeerConnection API MUST check this string as
follows:
1. If the RHS of the string is equal to the domain name of the IdP
proxy, then the assertion is valid, as the IdP is authoritative
for this domain.
2. If the RHS of the string is not equal to the domain name of the
IdP proxy, then the PeerConnection object MUST reject the
assertion unless (a) the IdP domain is listed as an acceptable
third-party IdP and (b) local policy is configured to trust this
IdP domain for the RHS of the identity string.
Sites which have identities that do not fit into the RFC822 style
(for instance, Facebook ids are simple numeric values) SHOULD convert
them to this form by appending their IdP domain (e.g.,
12345@identity.facebook.com), thus ensuring that they are
authoritative for the identity.
The IdP proxy MAY also include a "displayname" field which contains a
more user-friendly identity assertion. Browsers SHOULD take care in
the UI to distinguish the "name" assertion which is verifiable
directly from the "displayname" which cannot be verified and thus
relies on trust in the IdP. In future, we may define other fields to
allow the IdP to provide more information to the browser. [[OPEN
ISSUE: Should this field exist? Is it confusing? ]]
5.7. 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.
5.7.1. 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
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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. Note that all of the third-party security mechanisms
available (whether X.509 certificates or a third-party IdP) rely on
the security of the third party--this is of course also true of your
connection to the Web site itself. Users who wish to assure
themselves of security against a malicious identity provider can only
do so by verifying peer credentials directly, e.g., by checking the
peer's fingerprint against a value delivered out of band.
In order to protect against malicious content JavaScript, that
JavaScript MUST NOT be allowed to have direct access to---or perform
computations with---DTLS keys. For instance, if content JS were able
to compute digital signatures, then it would be possible for content
JS to get an identity assertion for a browser's generated key and
then use that assertion plus a signature by the key to authenticate a
call protected under an ephemeral DH key controlled by the content
JS, thus violating the security guarantees otherwise provided by the
IdP mechanism. Note that it is not sufficient merely to deny the
content JS direct access to the keys, as some have suggested doing
with the WebCrypto API. [webcrypto]. The JS must also not be allowed
to perform operations that would be valid for a DTLS endpoint. By
far the safest approach is simply to deny the ability to perform any
operations that depend on secret information associated with the key.
Operations that depend on public information, such as exporting the
public key are of course safe.
5.7.2. Privacy
The requirements in this document are intended to allow:
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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.
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 WebRTC/Tor
implementations SHOULD arrange to route the media as well as the
signaling through Tor. Currently this will produce very suboptimal
performance.
Additionally, any identifier which persists across multiple calls is
potentially a problem for privacy, especially for anonymous calling
services. Such services SHOULD instruct the browser to use separate
DTLS keys for each call and also to use TURN throughout the call.
Otherwise, the other side will learn linkable information.
Additionally, browsers SHOULD implement the privacy-preserving CNAME
generation mode of [I-D.ietf-avtcore-6222bis].
5.7.3. 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 WebRTC at all, WebRTC 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 WebRTC implementations
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).
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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. Either media-level (RTCP) mechanisms must be used or the
implementation must deny responses entirely, thus terminating the
call.
Yet another attack, suggested by Magnus Westerlund, is for the
attacker to cross-connect offers and answers as follows. It induces
the victim to make a call and then uses its control of other users
browsers to get them to attempt a call to someone. It then
translates their offers into apparent answers to the victim, which
looks like large-scale parallel forking. The victim still responds
to ICE responses and now the browsers all try to send media to the
victim. Implementations can defend themselves from this attack by
only responding to ICE Binding Requests for a limited number of
remote ufrags (this is the reason for the requirement that the JS not
be able to control the ufrag and password).
Note that attacks based on confusing one end or the other about
consent are possible 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 severely
restricts the capabilities of the calling application, so there are
difficult tradeoffs here.
5.7.4. IdP Authentication Mechanism
This mechanism relies for its security on the IdP and on the
PeerConnection correctly enforcing the security invariants described
above. At a high level, the IdP is attesting that the user
identified in the assertion wishes to be associated with the
assertion. Thus, it must not be possible for arbitrary third parties
to get assertions tied to a user or to produce assertions that RPs
will accept.
5.7.4.1. PeerConnection Origin Check
Fundamentally, the IdP proxy is just a piece of HTML and JS loaded by
the browser, so nothing stops a Web attacker o from creating their
own IFRAME, loading the IdP proxy HTML/JS, and requesting a
signature. In order to prevent this attack, we require that all
signatures be tied to a specific origin ("rtcweb://...") which cannot
be produced by content JavaScript. Thus, while an attacker can
instantiate the IdP proxy, they cannot send messages from an
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appropriate origin and so cannot create acceptable assertions. I.e.,
the assertion request must have come from the browser. This origin
check is enforced on the relying party side, not on the
authenticating party side. The reason for this is to take the burden
of knowing which origins are valid off of the IdP, thus making this
mechanism extensible to other applications besides WebRTC. The IdP
simply needs to gather the origin information (from the posted
message) and attach it to the assertion.
Note that although this origin check is enforced on the RP side and
not at the IdP, it is absolutely imperative that it be done. The
mechanisms in this document rely on the browser enforcing access
restrictions on the DTLS keys and assertion requests which do not
come with the right origin may be from content JS rather than from
browsers, and therefore those access restrictions cannot be assumed.
Note that this check only asserts that the browser (or some other
entity with access to the user's authentication data) attests to the
request and hence to the fingerprint. It does not demonstrate that
the browser has access to the associated private key. However,
attaching one's identity to a key that the user does not control does
not appear to provide substantial leverage to an attacker, so a proof
of possession is omitted for simplicity.
5.7.4.2. IdP Well-known URI
As described in Section 5.6.5.2.1 the IdP proxy HTML/JS landing page
is located at a well-known URI based on the IdP's domain name. This
requirement prevents an attacker who can write some resources at the
IdP (e.g., on one's Facebook wall) from being able to impersonate the
IdP.
5.7.4.3. Privacy of IdP-generated identities and the hosting site
Depending on the structure of the IdP's assertions, the calling site
may learn the user's identity from the perspective of the IdP. In
many cases this is not an issue because the user is authenticating to
the site via the IdP in any case, for instance when the user has
logged in with Facebook Connect and is then authenticating their call
with a Facebook identity. However, in other case, the user may not
have already revealed their identity to the site. In general, IdPs
SHOULD either verify that the user is willing to have their identity
revealed to the site (e.g., through the usual IdP permissions dialog)
or arrange that the identity information is only available to known
RPs (e.g., social graph adjacencies) but not to the calling site.
The "origin" field of the signature request can be used to check that
the user has agreed to disclose their identity to the calling site;
because it is supplied by the PeerConnection it can be trusted to be
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correct.
5.7.4.4. Security of Third-Party IdPs
As discussed above, each third-party IdP represents a new universal
trust point and therefore the number of these IdPs needs to be quite
limited. Most IdPs, even those which issue unqualified identities
such as Facebook, can be recast as authoritative IdPs (e.g.,
123456@facebook.com). However, in such cases, the user interface
implications are not entirely desirable. One intermediate approach
is to have special (potentially user configurable) UI for large
authoritative IdPs, thus allowing the user to instantly grasp that
the call is being authenticated by Facebook, Google, etc.
5.7.4.5. Web Security Feature Interactions
A number of optional Web security features have the potential to
cause issues for this mechanism, as discussed below.
5.7.4.5.1. Popup Blocking
If the user is not already logged into the IdP, the IdP proxy may
need to pop up a top level window in order to prompt the user for
their authentication information (it is bad practice to do this in an
IFRAME inside the window because then users have no way to determine
the destination for their password). If the user's browser is
configured to prevent popups, this may fail (depending on the exact
algorithm that the popup blocker uses to suppress popups). It may be
necessary to provide a standardized mechanism to allow the IdP proxy
to request popping of a login window. Note that care must be taken
here to avoid PeerConnection becoming a general escape hatch from
popup blocking. One possibility would be to only allow popups when
the user has explicitly registered a given IdP as one of theirs (this
is only relevant at the AP side in any case).
5.7.4.5.2. Third Party Cookies
Some browsers allow users to block third party cookies (cookies
associated with origins other than the top level page) for privacy
reasons. Any IdP which uses cookies to persist logins will be broken
by third-party cookie blocking. One option is to accept this as a
limitation; another is to have the PeerConnection object disable
third-party cookie blocking for the IdP proxy.
5.8. IANA Considerations
[TODO: IANA registration for Identity header. Or should this be in
MMUSIC?]
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6. Acknowledgements
Bernard Aboba, Harald Alvestrand, Richard Barnes, Dan Druta, Cullen
Jennings, Hadriel Kaplan, Matthew Kaufman, Jim McEachern, Martin
Thomson, Magnus Westerland. Matthew Kaufman provided the UI material
in Section 5.5.
7. Changes
7.1. Changes since -06
Replaced RTCWEB and RTC-Web with WebRTC, except when referring to the
IETF WG
Forbade use in mixed content as discussed in Orlando.
Added a requirement to surface NULL ciphers to the top-level.
Tried to clarify SRTP versus DTLS-SRTP.
Added a section on screen sharing permissions.
Assorted editorial work.
7.2. Changes since -05
The following changes have been made since the -05 draft.
o Response to comments from Richard Barnes
o More explanation of the IdP security properties and the federation
use case.
o Editorial cleanup.
7.3. Changes since -03
Version -04 was a version control mistake. Please ignore.
The following changes have been made since the -04 draft.
o Move origin check from IdP to RP per discussion in YVR.
o Clarified treatment of X.509-level identities.
o Editorial cleanup.
7.4. Changes since -03
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7.5. Changes since -02
The following changes have been made since the -02 draft.
o Forbid persistent HTTP permissions.
o Clarified the text in S 5.4 to clearly refer to requirements on
the API to provide functionality to the site.
o Fold in the IETF portion of draft-rescorla-rtcweb-generic-idp
o Retarget the continuing consent section to assume Binding Requests
o Added some more privacy and linkage text in various places.
o Editorial improvements
8. References
8.1. Normative References
[I-D.ietf-avtcore-6222bis]
Begen, A., Perkins, C., Wing, D., and E. Rescorla,
"Guidelines for Choosing RTP Control Protocol (RTCP)
Canonical Names (CNAMEs)", draft-ietf-avtcore-6222bis-06
(work in progress), July 2013.
[I-D.ietf-rtcweb-security]
Rescorla, E., "Security Considerations for RTC-Web",
draft-ietf-rtcweb-security-04 (work in progress),
January 2013.
[I-D.ietf-tsvwg-sctp-dtls-encaps]
Jesup, R., Loreto, S., Stewart, R., and M. Tuexen, "DTLS
Encapsulation of SCTP Packets for RTCWEB",
draft-ietf-tsvwg-sctp-dtls-encaps-00 (work in progress),
February 2013.
[I-D.muthu-behave-consent-freshness]
Perumal, M., Wing, D., R, R., and H. Kaplan, "STUN Usage
for Consent Freshness",
draft-muthu-behave-consent-freshness-03 (work in
progress), February 2013.
[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.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004.
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[RFC4347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer
Security", RFC 4347, April 2006.
[RFC4572] Lennox, J., "Connection-Oriented Media Transport over the
Transport Layer Security (TLS) Protocol in the Session
Description Protocol (SDP)", RFC 4572, July 2006.
[RFC4627] Crockford, D., "The application/json Media Type for
JavaScript Object Notation (JSON)", RFC 4627, 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.
[RFC5246] Dierks, T. and E. Rescorla, "The Transport Layer Security
(TLS) Protocol Version 1.2", RFC 5246, August 2008.
[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,
December 2011.
[webcrypto]
Dahl, Sleevi, "Web Cryptography API", June 2013.
Available at http://www.w3.org/TR/WebCryptoAPI/
[webrtc-api]
Bergkvist, Burnett, Jennings, Narayanan, "WebRTC 1.0:
Real-time Communication Between Browsers", October 2011.
Available at
http://dev.w3.org/2011/webrtc/editor/webrtc.html
8.2. Informative References
[I-D.ietf-rtcweb-jsep]
Uberti, J. and C. Jennings, "Javascript Session
Establishment Protocol", draft-ietf-rtcweb-jsep-03 (work
in progress), February 2013.
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[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.
[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".
Appendix A. Example IdP Bindings to Specific Protocols
[[TODO: These still need some cleanup.]]
This section provides some examples of how the mechanisms described
in this document could be used with existing authentication protocols
such as BrowserID or OAuth. Note that this does not require browser-
level support for either protocol. Rather, the protocols can be fit
into the generic framework. (Though BrowserID in particular works
better with some client side support).
A.1. 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
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identity provider. It's relatively easy to extend this functionality
to authenticate WebRTC calls, as shown below.
+----------------------+ +----------------------+
| | | |
| 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 | |
| +--------------+ | | +---------------+ |
| | IdP Proxy | | | | IdP Proxy | |
| | to | | | | to | |
| | 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 (i.e., it
provides "browserid.org" as the IdP name.)
2. The PeerConnection instantiates the BrowserID signer in the IdP
proxy
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.
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.
The offer might look something like:
{
"type":"OFFER",
"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 \
a=identity [[base-64 encoding of...
4A:AD:B9:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB\n",
"identity":{
"idp":{ // Standardized
"domain":"browserid.org",
"method":"default"
},
"assertion": // Contents are browserid-specific
"\"assertion\": {
\"digest\":\"<hash of the contents from the browser>\",
\"audience\": \"[TBD]\"
\"valid-until\": 1308859352261,
},
\"certificate\": {
\"email\": \"rescorla@example.org\",
\"public-key\": \"<ekrs-public-key>\",
\"valid-until\": 1308860561861,
}" // certificate is signed by example.org
}]]"
}
Note that while the IdP here is specified as "browserid.org", the
actual certificate is signed by example.org. This is because
BrowserID is a combined authoritative/third-party system in which
browserid.org delegates the right to be authoritative (what BrowserID
calls primary) to individual domains.
On Bob's side, he receives the signed assertion as part of the call
setup message and a similar procedure happens to verify it.
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1. The calling JS instantiates a PeerConnection and provides it the
relevant signaling information, including the signed assertion.
2. The PeerConnection instantiates the IdP proxy which examines the
IdP name and brings up the BrowserID verification code.
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.2. 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 WebRTC concepts:
+----------------------+----------------------+
| OAuth | WebRTC |
+----------------------+----------------------+
| 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.
Alice IdP Bob
---------------------------------------------------------
Call-Id, Fingerprint ------->
<------------------- Auth Code
Auth Code ---------------------------------------------->
<----- Get Token + Auth Code
Token --------------------->
<------------- Get call-info
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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.
Author's Address
Eric Rescorla
RTFM, Inc.
2064 Edgewood Drive
Palo Alto, CA 94303
USA
Phone: +1 650 678 2350
Email: ekr@rtfm.com
Rescorla Expires January 15, 2014 [Page 43]
