One document matched: draft-ietf-rtcweb-security-arch-09.xml
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<rfc category="std" docName="draft-ietf-rtcweb-security-arch-09"
ipr="pre5378Trust200902">
<front>
<title abbrev="WebRTC Sec. Arch.">WebRTC Security Architecture</title>
<author fullname="Eric Rescorla" initials="E.K." surname="Rescorla">
<organization>RTFM, Inc.</organization>
<address>
<postal>
<street>2064 Edgewood Drive</street>
<city>Palo Alto</city>
<region>CA</region>
<code>94303</code>
<country>USA</country>
</postal>
<phone>+1 650 678 2350</phone>
<email>ekr@rtfm.com</email>
</address>
</author>
<date day="14" month="February" year="2014" />
<area>RAI</area>
<workgroup>RTCWEB</workgroup>
<abstract>
<t>
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 WebRTC.
</t>
</abstract>
</front>
<middle>
<section title="Introduction" anchor="sec.introduction">
<t>
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<xref
target="RFC3261"></xref> soft phones) WebRTC communications are directly
controlled by some Web server, via a JavaScript (JS) API as shown in
<xref target="fig.simple"/>.
</t>
<figure title="A simple WebRTC system" anchor="fig.simple">
<artwork><![CDATA[
+----------------+
| |
| Web Server |
| |
+----------------+
^ ^
/ \
HTTP / \ HTTP
/ \
/ \
v v
JS API JS API
+-----------+ +-----------+
| | Media | |
| Browser |<---------->| Browser |
| | | |
+-----------+ +-----------+
]]></artwork>
</figure>
<t>
A more complicated system might allow for interdomain calling, as shown
in <xref target="fig.multidomain"/>. 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.
</t>
<figure title="A multidomain WebRTC system" anchor="fig.multidomain">
<artwork><![CDATA[
+--------------+ +--------------+
| | SIP,XMPP,...| |
| Web Server |<----------->| Web Server |
| | | |
+--------------+ +--------------+
^ ^
| |
HTTP | | HTTP
| |
v v
JS API JS API
+-----------+ +-----------+
| | Media | |
| Browser |<---------------->| Browser |
| | | |
+-----------+ +-----------+
]]></artwork>
</figure>
<t>
This system presents a number of new security challenges, which are
analyzed in <xref target="I-D.ietf-rtcweb-security"/>. This document
describes a security architecture for WebRTC which addresses the threats
and requirements described in that document.
</t>
</section>
<section anchor="sec-term" title="Terminology">
<t>
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 <xref
target="RFC2119">RFC 2119</xref>.
</t>
</section>
<section title="Trust Model" anchor="sec.proposal.trusthierarchy">
<t>
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.
</t>
<t>
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.
</t>
<section title="Authenticated Entities" anchor="sec.proposal.authenticated">
<t>
There are two major classes of authenticated entities in the system:
</t>
<t>
<list style="symbols">
<t>
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).
</t>
<t>
Other users: WebRTC peers whose origin we can verify
cryptographically (optimally via DTLS-SRTP).
</t>
</list>
</t>
<t>
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.
</t>
<t>
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.
</t>
</section>
<section title="Unauthenticated Entities" anchor="sec.proposal.unauthenticated">
<t>
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.
</t>
</section>
</section>
<!-- Not layered ? -->
<section title="Overview" anchor="sec.proposal.overview">
<!-- TODO: Federated -->
<t>
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.
</t>
<t>
For the purposes of this example, we assume the topology shown in the
figures below. This topology is derived from the topology shown in <xref
target="fig.simple"/>, 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. <xref
target="sec.trust-relationships"/> provides an overview of Identity
Providers and the relevant terminology. Alice and Bob might have
relationships with different IdPs as well.
</t>
<t>
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 (<xref
target="fig.proposal.idp"/>) However, there are important settings where
that is not the case, such as federation (calls from one domain to
another; <xref target="fig.proposal-federated.idp"/>) 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.
</t>
<t>
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.
</t>
<figure title="A call with IdP-based identity" anchor="fig.proposal.idp">
<artwork><![CDATA[
+----------------+
| |
| Signaling |
| Server |
| |
+----------------+
^ ^
/ \
HTTPS / \ HTTPS
/ \
/ \
v v
JS API JS API
+-----------+ +-----------+
| | Media | |
Alice | Browser |<---------->| Browser | Bob
| | (DTLS+SRTP)| |
+-----------+ +-----------+
^ ^--+ +--^ ^
| | | |
v | | v
+-----------+ | | +-----------+
| |<--------+ | |
| IdP1 | | | IdP2 |
| | +------->| |
+-----------+ +-----------+
]]></artwork>
</figure>
<t>
<xref target="fig.proposal-federated.idp"/> shows essentially the same
calling scenario but with a call between two separate domains (i.e., a
federated case), as in <xref target="fig.multidomain"/>. 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.
</t>
<figure title="A federated call with IdP-based identity" anchor="fig.proposal-federated.idp">
<artwork><![CDATA[
+----------------+ 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 |
| | +------------------------>| |
+-----------+ +-----------+
]]></artwork>
</figure>
<section title="Initial Signaling">
<t>
For simplicity, assume the topology in <xref
target="fig.proposal.idp"/>. 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.
</t>
<t>
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
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.
</t>
<t>
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.
</t>
<t>
In the current W3C API, once some streams have been added, Alice's
browser + JS generates a signaling message <xref
target="I-D.ietf-rtcweb-jsep"/> containing:
</t>
<t>
<list style="symbols">
<t>
Media channel information
</t>
<t>
Interactive Connectivity Establishment (ICE) <xref
target="RFC5245"/> candidates
</t>
<t>
A fingerprint attribute binding the communication to a key pair
<xref target="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.
</t>
</list>
</t>
<t>
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 <xref target="sec.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.
</t>
<t>
This message is sent to the signaling server, e.g., by XMLHttpRequest
<xref target="XmlHttpRequest"/> or by WebSockets <xref
target="RFC6455"/>. preferably over TLS <xref target="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 element (e.g., a
button) to allow Bob to answer the call, though this is most likely
not part of the trusted UI.
</t>
<t>
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.
</t>
<t>
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 <xref target="fig.proposal-federated.idp"/>
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.
</t>
<t>
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.
</t>
</section>
<section title="Media Consent Verification">
<t>
As described in (<xref target="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.
</t>
<t>
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.
</t>
</section>
<section title="DTLS Handshake">
<t>
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 <xref target="RFC4347"/> (for the
data channel) and DTLS-SRTP <xref target="RFC5763"/> keying for SRTP
<xref target="RFC3711"/> for the media channel and SCTP over DTLS
<xref target="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 <xref
target="RFC5705"/> and used to key SRTP for the media channels.
</t>
<t>
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).
</t>
</section>
<section title="Communications and Consent Freshness">
<t>
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.
</t>
<t>
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 <xref
target="I-D.muthu-behave-consent-freshness"/>. If a keepalive fails
and no new ICE channels can be established, then the session is
terminated.
</t>
</section>
</section>
<section title="Detailed Technical Description" anchor="sec.proposal.detailed">
<section title="Origin and Web Security Issues" anchor="sec.proposal.origin">
<t>
The basic unit of permissions for WebRTC is the origin <xref
target="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 <xref
target="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.]
</t>
<t>
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.
</t>
<t>
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.
</t>
</section>
<section title="Device Permissions Model" anchor="sec.proposal.device.permissions">
<t>
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.
</t>
<t>
<list style="symbols">
<t>
Requests for one-time camera/microphone access.
</t>
<t>
Requests for permanent access.
</t>
</list>
</t>
<t>
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.
</t>
<t>
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 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.
</t>
<t>
<list style="hanging">
<t hangText="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.
</t>
</list>
</t>
<t>
<list style="hanging">
<t hangText="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.
</t>
</list>
</t>
<t>
<list style="hanging">
<t hangText="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.
</t>
</list>
</t>
<t>
<list style="hanging">
<t hangText="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.]
</t>
</list>
</t>
<t>
[[OPEN ISSUE: This section does not have WG consensus. Because
screen/application sharing presents a more significant risk than
camera and microphone access (see the discussion in <xref
target="I-D.ietf-rtcweb-security"/> S 4.1.1), we require a higher
level of user consent.
</t>
<t>
<list style="symbols">
<t>
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.
</t>
<t>
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.
</t>
<t>
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.
</t>
</list>
</t>
<t>
-- END OF OPEN ISSUE]]
</t>
<t>
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 <xref
target="sec.proposal.communications.consent"/> for a related issue).
</t>
<t>
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:
</t>
<t>
<list style="symbols">
<t>
Allow future calls to this verified user.
</t>
<t>
Allow future calls to any verified user who is in my system
address book (this only works with address book integration, of
course).
</t>
</list>
</t>
<t>
Implementations SHOULD also provide a different user interface
indication when calls are in progress to users whose identities are
directly verifiable. <xref target="sec.proposal.comsec"/> provides
more on this.
</t>
</section>
<section title="Communications Consent" anchor="sec.proposal.communications.consent">
<t>
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 <xref target="RFC5245"/>.
</t>
<t>
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
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.
</t>
<t>
While continuing consent is required, that ICE <xref
target="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 <xref
target="I-D.muthu-behave-consent-freshness"/>.
</t>
</section>
<section title="IP Location Privacy" anchor="sec.proposal.ip.location.privacy">
<t>
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.
</t>
<t>
<list style="hanging">
<t hangText="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.
</t>
</list>
</t>
<t>
<list style="hanging">
<t hangText="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.
</t>
</list>
</t>
<t>
<list style="hanging">
<t hangText="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 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
</t>
</list>
</t>
<t>
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.
</t>
</section>
<section title="Communications Security" anchor="sec.proposal.comsec">
<t>
Implementations MUST implement SRTP <xref target="RFC3711"/>.
Implementations MUST implement DTLS <xref target="RFC4347"/> and
DTLS-SRTP <xref target="RFC5763"/><xref target="RFC5764"/> for SRTP
keying. Implementations MUST implement <xref
target="I-D.ietf-tsvwg-sctp-dtls-encaps"/>.
</t>
<t>
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. WebRTC implements MUST NOT offer SDES or
select it if offered.
</t>
<t>
All data channels MUST be secured via DTLS.
</t>
<t>
[[OPEN ISSUE: Are these the right cipher suites?]]
All implementations MUST implement the following two cipher suites:
TLS_DHE_RSA_WITH_AES_128_GCM_SHA256 and TLS_ECDHE_RSA_WITH_AES_128_GCM_SHA256
and the DTLS-SRTP protection profile SRTP_AES128_CM_HMAC_SHA1_80.
Implementations SHOULD favor cipher suites which support PFS over
non-PFS cipher suites.
</t>
<!-- OPEN ISSUE: DTLS-SRTP key origin scoping? -->
<t>
<list style="hanging">
<t hangText="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.)
</t>
</list>
</t>
<t>
<list style="hanging">
<t hangText="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.
</t>
</list>
</t>
<t>
<list style="hanging">
<t hangText="UI Requirements: ">
A user-oriented client MUST provide an "inspector" interface which
allows the user to determine the security characteristics of the
media.
</t>
<t>
The following properties SHOULD be displayed "up-front" in the
browser chrome, i.e., without requiring the user to ask for them:
</t>
<t>
<list style="symbols">
<t>
A client MUST provide a user interface through which a user
may determine the security characteristics for
currently-displayed audio and video stream(s)
</t>
<t>
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.
</t>
<t>
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.
</t>
<t>
If the far endpoint was directly verified, either via a
third-party verifiable X.509 certificate or via a Web IdP
mechanism (see <xref target="sec.generic.idp"/>) 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.
</t>
</list>
</t>
<t>
</t>
<t>
The following properties are more likely to require some
"drill-down" from the user:
</t>
<t>
<list style="symbols">
<t>
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.
</t>
<t>
The "security characteristics" MUST indicate whether PFS is
provided.
</t>
<t>
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.
</t>
</list>
</t>
</list>
</t>
</section>
<section title="Web-Based Peer Authentication" anchor="sec.generic.idp">
<t>
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.)
</t>
<t>
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.
</t>
<t>
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.
</t>
<t>
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.
</t>
<section title="Trust Relationships: IdPs, APs, and RPs" anchor="sec.trust-relationships">
<t>
Any federated identity protocol has three major participants:
</t>
<t>
<list style="hanging">
<t hangText="Authenticating Party (AP):">
The entity which is trying to establish its identity.
</t>
<t>
</t>
<t hangText="Identity Provider (IdP):">
The entity which is vouching for the AP's identity.
</t>
<t>
</t>
<t hangText="Relying Party (RP):">
The entity which is trying to verify the AP's identity.
</t>
</list>
</t>
<t>
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 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.
</t>
<t>
At a high level there are two kinds of IdPs:
</t>
<t>
<list style="hanging">
<t hangText="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).
</t>
<t>
</t>
<t hangText="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.
</t>
</list>
</t>
<t>
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.
</t>
<t>
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.
</t>
</section>
<section title="Overview of Operation" anchor="sec.overview">
<t>
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.
</t>
<t>
Note that there are two logically separate functions here:
</t>
<t>
<list style="symbols">
<t>
Identity assertion generation.
</t>
<t>
Identity assertion verification.
</t>
</list>
</t>
<t>
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.
</t>
<figure>
<artwork><![CDATA[
+------------------------------------+
| https://calling-site.example.com |
| |
| |
| |
| Calling JS Code |
| ^ |
| | API Calls |
| v |
| PeerConnection |
| ^ |
| | postMessage() |
| v |
| +-------------------------+ | +---------------+
| | https://idp.example.org | | | |
| | |<--------->| Identity |
| | IdP JS | | | Provider |
| | | | | |
| +-------------------------+ | +---------------+
| |
+------------------------------------+
]]></artwork>
</figure>
<t>
When the PeerConnection object wants to interact with the IdP, the
sequence of events is as follows:
</t>
<t>
<list style="numbers">
<t>
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
security context.
</t>
<t>
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.
</t>
<t>
Once the user is logged in, the IdP proxy notifies the browser
that it is ready.
</t>
<t>
The browser and the IdP proxy communicate via a standardized
series of messages delivered via a <xref
target="WebMessaging">MessageChannel</xref>. For instance, the
browser might request the IdP proxy to sign or verify a given
identity assertion.
</t>
</list>
</t>
<t>
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.
</t>
</section>
<section title="Items for Standardization" anchor="sec.standardized">
<t>
In order to make this work, we must standardize the following items:
</t>
<t>
<list style="symbols">
<t>
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. <xref
target="sec.jsep-binding"/>
</t>
<t>
The interface to the IdP. <xref target="sec.protocol-details"/>
specifies a specific protocol mechanism which allows the use of
any identity protocol without requiring specific further
protocol support in the browser
</t>
<t>
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.
</t>
</list>
</t>
<t>
The first two items are defined in this document. The final one is
defined in the companion W3C WebRTC API specification <xref
target="webrtc-api"/>.
</t>
</section>
<section title="Binding Identity Assertions to JSEP Offer/Answer Transactions" anchor="sec.jsep-binding">
<section title="Input to Assertion Generation Process">
<t>
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 as a JavaScript object (also known as a
dictionary or hash) with a single <spanx
style="verb">fingerprint</spanx> key, as shown below:
</t>
<figure>
<artwork><![CDATA[
{
"fingerprint": {
"algorithm": "sha-256",
"digest": "4A:AD:B9:B1:3F:...:E5:7C:AB"
}
}
]]>
</artwork>
</figure>
<t>
This object is encoded in a <xref target="RFC4627">JSON</xref>
string for passing to the IdP.
</t>
<t>
The <spanx style="verb">algorithm</spanx> and <spanx
style="verb">digest</spanx> values correspond directly to the
algorithm and digest values in the a=fingerprint line of the SDP.
<xref target="RFC4572"/>.
</t>
<t>
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.
</t>
</section>
<section title="Carrying Identity Assertions">
<t>
Once an IdP has generated an assertion, it is attached to the SDP
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 <xref
target="RFC4848">base-64 encoded</xref> identity assertion. For
example:
</t>
<figure>
<artwork><![CDATA[
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
...
]]></artwork>
</figure>
<t>
Each identity attribute should be paired (and attests to) with an
<spanx style="verb">a=fingerprint</spanx> 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 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.
</t>
<t>
TODO: write up paragraph on the consequences of multiple
a=fingerprint attributes.
</t>
</section>
</section>
<section title="IdP Interaction Details" anchor="sec.protocol-details">
<section title="General Message Structure">
<t>
Messages between the PeerConnection object and the IdP proxy are
JavaScript objects, shown in examples using JSON <xref
target="RFC4627"/>. For instance, the PeerConnection would
request a signature with the following "SIGN" message:
</t>
<figure>
<artwork><![CDATA[
{
"type": "SIGN",
"id": "1",
"origin": "https://calling-site.example.com",
"message": "012345678abcdefghijkl"
}
]]></artwork>
</figure>
<t>
All messages MUST contain a <spanx style="verb">type</spanx> field
which indicates the general meaning of the message.
</t>
<t>
All requests from the PeerConnection object MUST contain an <spanx
style="verb">id</spanx> field which MUST be unique within the
scope of the interaction between the browser and the IdP
instance. Responses from the IdP proxy MUST contain the same
<spanx style="verb">id</spanx> in response, which allows the
PeerConnection to correlate requests and responses, in case there
are multiple requests/responses outstanding to the same proxy.
</t>
<t>
All requests from the PeerConnection object MUST contain an <spanx
style="verb">origin</spanx> 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.
</t>
<t>
Any message-specific data is carried in a <spanx
style="verb">message</spanx> field. Depending on the message
type, this may either be a string or any JavaScript object that
can be conveyed in a message channel. This includes any object
that is able to be serialized to JSON.
</t>
</section>
<section title="Errors">
<t>
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:
</t>
<figure title="Example error" anchor="fig.example-error">
<artwork><![CDATA[
{
"type": "ERROR",
"id": "1",
"error": "Signature verification failed"
}
]]></artwork>
</figure>
</section>
<section title="IdP Proxy Setup" anchor="sec.iframe-setup">
<t>
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:
</t>
<t>
<list style="symbols">
<t>
The JS runs in the IdP's security context with the base page
retrieved from the URL specified in <xref
target="sec.idp-uri"/>.
</t>
<t>
The usual browser sandbox isolation mechanisms MUST be
enforced with respect to the IdP proxy. The IdP cannot be
provided with escalated privileges.
</t>
<t>
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.
</t>
<t>
The IdP proxy is unable to interact with the user. This
includes the creation of popup windows and dialogs.
</t>
</list>
</t>
<t>
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 to
load and prepare necessary resources.
</t>
<t>
When the IdP proxy is ready to receive commands, it delivers a
"READY" message. As this message is unsolicited, it contains only
the <spanx style="verb">type</spanx>:
</t>
<figure>
<artwork><![CDATA[
{ "type":"READY" }
]]></artwork>
</figure>
<t>
Once the PeerConnection object receives the ready message, it can
send commands to the IdP proxy.
</t>
<section title="Determining the IdP URI" anchor="sec.idp-uri">
<t>
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:
</t>
<t>
<list style="hanging">
<t hangText="domain name:">
The IdP's domain name
</t>
<t hangText="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.
</t>
</list>
</t>
<t>
Each IdP MUST serve its initial entry page (i.e., the one loaded
by the IdP proxy) from a <xref target="RFC5785">well-known
URI</xref>. The well-known URI for an IdP proxy is formed from
the following URI components:
<list style="numbers">
<t>
The scheme, "https:". An IdP MUST be loaded using <xref
target="RFC2818">HTTPS</xref>.
</t>
<t>
The authority, which is the IdP domain name. The authority
MAY contain a non-default port number. Any port number is
removed when determining if an asserted identity matches the
name of the IdP. The authority MUST NOT include a userinfo
sub-component.
</t>
<t>
The path, starting with "/.well-known/idp-proxy/" and
appended with the IdP protocol. Note that the separator
characters '/' (%2F) and '\' (%5C) MUST NOT be permitted in
the protocol field, lest an attacker be able to direct
requests outside of the controlled "/.well-known/" prefix.
Query and fragment values MAY be used by including '?' or
'#' characters.
</t>
</list>
For example, for the IdP "identity.example.com" and the protocol
"example", the URL would be:
</t>
<figure>
<artwork><![CDATA[
https://example.com/.well-known/idp-proxy/example
]]></artwork>
</figure>
<!-- [[TODO: suggested by mt]]
<t>
The IdP proxy MUST be prevented from redirecting or navigating
the IdP frame to another site.
</t>
-->
<section title="Authenticating Party">
<t>
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:
<list style="symbols">
<t>
Provided by the user directly.
</t>
<t>
Selected from some set of IdPs known to the calling site.
E.g., a button that shows "Authenticate via Facebook
Connect"
</t>
</list>
</t>
</section>
<section title="Relying Party">
<t>
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 <xref
target="sec.id-format"/>.
</t>
</section>
</section>
<section title="Requesting Assertions" anchor="sec.request-assert">
<t>
In order to request an assertion, the PeerConnection sends a
"SIGN" message. Aside from the mandatory fields, this message
has a <spanx style="verb">message</spanx> field containing a
string. The string contains a JSON-encoded object containing
certificate fingerprints but are treated as opaque from the
perspective of the IdP.
</t>
<t>
A successful response to a "SIGN" message contains a <spanx
style="verb">message</spanx> field which is a JavaScript
dictionary consisting of two fields:
</t>
<t>
<list style="hanging">
<t hangText="idp:">
A dictionary containing the domain name of the provider and
the protocol string.
</t>
<t hangText="assertion:">
An opaque value containing the assertion itself. This is
only interpretable by the IdP or its proxy.
</t>
</list>
</t>
<t>
<xref target="fig.assert-request"/> 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 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.
</t>
<figure title="Example assertion request" anchor="fig.assert-request">
<artwork><![CDATA[
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\",
\"signature\":\"010203040506\"}"
}
}
]]></artwork>
</figure>
<t>
The <spanx style="verb">message</spanx> structure is serialized
into JSON, <xref target="RFC4848">base64-encoded</xref>, and
placed in an <spanx style="verb">a=identity</spanx> attribute.
</t>
</section>
<section title="Verifying Assertions" anchor="sec.verify-assert">
<t>
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 <spanx style="verb">message</spanx> field.
</t>
<t>
The IdP proxy verifies the assertion. Depending on the identity
protocol, the proxy might contact the IdP server or other
servers. 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.
</t>
<t>
Regardless of the mechanism, if verification succeeds, a
successful response from the IdP proxy MUST contain a message
field consisting of a object with the following fields:
<list style="hanging">
<t hangText="identity:">
The identity of the AP from the IdP's perspective. Details
of this are provided in <xref target="sec.id-format"/>.
</t>
<t hangText="contents:">
The original unmodified string provided by the AP in the
original SIGN request.
</t>
</list>
</t>
<t>
<xref target="fig.verify-request"/> shows an example
transaction. Line breaks are inserted solely for readability.
</t>
<figure title="Example verification request" anchor="fig.verify-request">
<artwork>
<![CDATA[
PeerConnection -> IdP Proxy:
{
"type": "VERIFY",
"id": 2,
"origin": "https://calling-service.example.com/",
"message": "{\"identity\":\"bob@example.org\",
\"contents\":\"abcdefghijklmnopqrstuvwyz\",
\"signature\":\"010203040506\"}"
}
IdP Proxy -> PeerConnection:
{
"type": "SUCCESS",
"id": 2,
"message": {
"identity": "bob@example.org",
"contents": "abcdefghijklmnopqrstuvwyz"
}
}
]]></artwork>
</figure>
<section title="Identity Formats" anchor="sec.id-format">
<t>
Identities passed from the IdP proxy to the PeerConnection are
passed in the <spanx style="verb">identity</spanx> field. This
field MUST consist of a string representing the user's
identity. This string is in the form "<user>@<domain>",
where <spanx style="verb">user</spanx> consists of any
character except '@', and domain is an <xref
target="RFC5890">internationalized domain name</xref>.
</t>
<t>
The PeerConnection API MUST check this string as follows:
<list style="numbers">
<t>
If the domain portion 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. Comparison of
domain names is done using the label equivalence rule
defined in Section 2.3.2.4 of <xref target="RFC5890"/>.
</t>
<t>
If the domain portion of the string is not equal to the
domain name of the IdP proxy, then the PeerConnection
object MUST reject the assertion unless:
<list style="numbers">
<t>
the IdP domain is trusted as an acceptable third-party
IdP; and
</t>
<t>
local policy is configured to trust this IdP domain
for the RHS of the identity string.
</t>
</list>
</t>
</list>
</t>
<t>
Sites which have identities that do not fit into the RFC822
style (for instance, identifiers that are simple numeric
values, or values that contain '@' characters) SHOULD convert
them to this form by escaping illegal characters and appending
their IdP domain (e.g., user%40133@identity.example.com), thus
ensuring that they are authoritative for the identity.
</t>
</section>
</section>
</section>
</section>
</section>
<section title="Security Considerations" anchor="sec.sec-cons">
<t>
Much of the security analysis of this problem is contained in <xref
target="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.
</t>
<section title="Communications Security">
<t>
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.
</t>
<t>
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 <xref target="sec.proposal.comsec"/> are designed to
provide such a mechanism for motivated/security conscious users, but
are not suitable for general use. The identity service mechanisms
in <xref target="sec.generic.idp"/> 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.
</t>
<t>
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. <xref target="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.
</t>
</section>
<section title="Privacy">
<t>
The requirements in this document are intended to allow:
</t>
<t>
<list style="symbols">
<t>
Users to participate in calls without revealing their location.
</t>
<t>
Potential callees to avoid revealing their location and even
presence status prior to agreeing to answer a call.
</t>
</list>
</t>
<t>
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.
</t>
<t>
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.
</t>
<t>
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 <xref target="I-D.ietf-avtcore-6222bis"/>.
</t>
</section>
<section title="Denial of Service">
<t>
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.
</t>
<t>
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).
</t>
<t>
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.
</t>
<t>
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).
</t>
<t>
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.
</t>
</section>
<section title="IdP Authentication Mechanism">
<t>
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.
</t>
<section title="PeerConnection Origin Check" anchor="sec.pc-origin">
<t>
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
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.
</t>
<t>
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.
</t>
<t>
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.
</t>
</section>
<section title="IdP Well-known URI" anchor="sec.sec-idp-uri">
<t>
As described in <xref target="sec.idp-uri"/> 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.
</t>
</section>
<section title="Privacy of IdP-generated identities and the hosting site">
<t>
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 correct.
</t>
</section>
<section title="Security of Third-Party IdPs" anchor="sec.sec-third-party">
<t>
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.
</t>
</section>
<section title="Web Security Feature Interactions">
<t>
A number of optional Web security features have the potential to
cause issues for this mechanism, as discussed below.
</t>
<section title="Popup Blocking" anchor="sec.popup-blocking">
<t>
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).
</t>
</section>
<section title="Third Party Cookies" anchor="sec.3rd-party-cookies">
<t>
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.
</t>
</section>
</section>
</section>
</section>
<section title="IANA Considerations" anchor="sec.iana-cons">
<t>
[TODO: IANA registration for Identity header. Or should this be in
MMUSIC?]
</t>
</section>
</section>
<section title="Acknowledgements">
<t>
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
<xref target="sec.proposal.comsec"/>.
</t>
</section>
<section title="Changes">
<section title="Changes since -06">
<t>
Replaced RTCWEB and RTC-Web with WebRTC, except when referring to the
IETF WG
</t>
<t>
Forbade use in mixed content as discussed in Orlando.
</t>
<t>
Added a requirement to surface NULL ciphers to the top-level.
</t>
<t>
Tried to clarify SRTP versus DTLS-SRTP.
</t>
<t>
Added a section on screen sharing permissions.
</t>
<t>
Assorted editorial work.
</t>
</section>
<section title="Changes since -05">
<t>
The following changes have been made since the -05 draft.
</t>
<t>
<list style="symbols">
<t>
Response to comments from Richard Barnes
</t>
<t>
More explanation of the IdP security properties and the federation
use case.
</t>
<t>
Editorial cleanup.
</t>
</list>
</t>
</section>
<section title="Changes since -03">
<t>
Version -04 was a version control mistake. Please ignore.
</t>
<t>
The following changes have been made since the -04 draft.
</t>
<t>
<list style="symbols">
<t>
Move origin check from IdP to RP per discussion in YVR.
</t>
<t>
Clarified treatment of X.509-level identities.
</t>
<t>
Editorial cleanup.
</t>
</list>
</t>
</section>
<section title="Changes since -03">
</section>
<section title="Changes since -02">
<t>
The following changes have been made since the -02 draft.
</t>
<t>
<list style="symbols">
<t>
Forbid persistent HTTP permissions.
</t>
<t>
Clarified the text in S 5.4 to clearly refer to requirements on
the API to provide functionality to the site.
</t>
<t>
Fold in the IETF portion of draft-rescorla-rtcweb-generic-idp
</t>
<t>
Retarget the continuing consent section to assume Binding Requests
</t>
<t>
Added some more privacy and linkage text in various places.
</t>
<t>
Editorial improvements
</t>
</list>
</t>
</section>
</section>
</middle>
<back>
<references title="Normative References">
&RFC2119;
&RFC2818;
&RFC3711;
&RFC4347;
&RFC4572;
&RFC4627;
&RFC4848;
&RFC5245;
&RFC5246;
&RFC5763;
&RFC5764;
&RFC5785;
&RFC5890;
&RFC6454;
&I-D.ietf-rtcweb-security;
&I-D.muthu-behave-consent-freshness;
&I-D.ietf-avtcore-6222bis;
&I-D.ietf-tsvwg-sctp-dtls-encaps;
<reference anchor="webcrypto">
<front>
<title>Web Cryptography API</title>
<author fullname="W3C editors"
surname="Dahl, Sleevi">
<organization>W3C</organization>
</author>
<date day="25" month="June" year="2013" />
</front>
<annotation>Available at
http://www.w3.org/TR/WebCryptoAPI/</annotation>
</reference>
<reference anchor="webrtc-api">
<front>
<title>WebRTC 1.0: Real-time Communication Between Browsers</title>
<author fullname="W3C editors"
surname="Bergkvist, Burnett, Jennings, Narayanan">
<organization>W3C</organization>
</author>
<date day="4" month="October" year="2011" />
</front>
<annotation>Available at
http://dev.w3.org/2011/webrtc/editor/webrtc.html</annotation>
</reference>
<reference anchor="WebMessaging" target="http://www.w3.org/TR/2012/CR-webmessaging-20120501/">
<front>
<title>HTML5 Web Messaging</title>
<author fullname="W3C editors" surname="Hickson">
<organization>W3C</organization>
</author>
<date day="1" month="May" year="2012"/>
</front>
</reference>
</references>
<references title="Informative References">
&RFC3261;
&RFC5705;
&RFC6455;
&I-D.ietf-rtcweb-jsep;
<reference anchor="XmlHttpRequest">
<front>
<title>XMLHttpRequest Level 2</title>
<author initials="A." surname="van Kesteren">
<organization></organization>
</author>
</front>
<format target="http://www.w3.org/TR/XMLHttpRequest/" type="TXT"/>
</reference>
</references>
<section title="Example IdP Bindings to Specific Protocols">
<t>
[[TODO: These still need some cleanup.]]
</t>
<t>
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).
</t>
<section title="BrowserID">
<t>
BrowserID [https://browserid.org/] is a technology which allows a user
with a verified email address to generate an assertion (authenticated
by their identity provider) attesting to their identity (phrased as an
email address). The way that this is used in practice is that the
relying party embeds JS in their site which talks to the BrowserID
code (either hosted on a trusted intermediary or embedded in the
browser). That code generates the assertion which is passed back to
the relying party for verification. The assertion can be verified
directly or with a Web service provided by the identity provider.
It's relatively easy to extend this functionality to authenticate
WebRTC calls, as shown below.
</t>
<figure>
<artwork><![CDATA[
+----------------------+ +----------------------+
| | | |
| 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 |
| |
+----------------------+
]]></artwork>
</figure>
<t>
The way this mechanism works is as follows. On Alice's side, Alice
goes to initiate a call.
</t>
<t>
<list style="numbers">
<t>
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.)
</t>
<t>
The PeerConnection instantiates the BrowserID signer in the IdP
proxy
</t>
<t>
The BrowserID signer contacts Alice's identity provider,
authenticating as Alice (likely via a cookie).
</t>
<t>
The identity provider returns a short-term certificate attesting
to Alice's identity and her short-term public key.
</t>
<t>
The Browser-ID code signs the fingerprint and returns the signed
assertion + certificate to the PeerConnection.
</t>
<t>
The PeerConnection returns the signed information to the calling
JS code.
</t>
<t>
The signed assertion gets sent over the wire to Bob's browser (via
the signaling service) as part of the call setup.
</t>
</list>
</t>
<t>
The offer might look something like:
</t>
<figure>
<artwork><![CDATA[
{
"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
a=fingerprint:SHA-1 ...\n
4A:AD:B9:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB\n
a=identity [[base-64 encoding of identity assertion:
{
"idp":{ // Standardized
"domain":"browserid.org",
"method":"default"
},
// Assertion contents are browserid-specific
"assertion": "{
\"assertion\": {
\"digest\":\"<hash of the SIGN message>\",
\"audience\": \"<audience>\"
\"valid-until\": 1308859352261,
},
\"certificate\": {
\"email\": \"rescorla@example.org\",
\"public-key\": \"<ekrs-public-key>\",
\"valid-until\": 1308860561861,
\"signature\": \"<signature from example.org>\"
},
\"content\": \"<content of the SIGN message>\"
}"
}
]]\n
m=audio 49170 RTP/AVP 0\n
..."
}
]]></artwork>
</figure>
<t>
<!-- [TODO: Need to talk about Audience a bit.] -->
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.
</t>
<t>
On Bob's side, he receives the signed assertion as part of the call
setup message and a similar procedure happens to verify it.
</t>
<t>
<list style="numbers">
<t>
The calling JS instantiates a PeerConnection and provides it the
relevant signaling information, including the signed assertion.
</t>
<t>
The PeerConnection instantiates the IdP proxy which examines the
IdP name and brings up the BrowserID verification code.
</t>
<t>
The BrowserID verifier contacts the identity provider to verify
the certificate and then uses the key to verify the signed
fingerprint.
</t>
<t>
Alice's verified identity is returned to the PeerConnection (it
already has the fingerprint).
</t>
<t>
At this point, Bob's browser can display a trusted UI indication
that Alice is on the other end of the call.
</t>
</list>
</t>
<t>
When Bob returns his answer, he follows the converse procedure, which
provides Alice with a signed assertion of Bob's identity and keying
material.
</t>
</section>
<section title="OAuth">
<t>
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:
</t>
<texttable anchor="oauth-rtcweb">
<ttcol align="left">OAuth</ttcol>
<ttcol align="left">WebRTC</ttcol>
<c>Client</c><c>Relying party</c>
<c>Resource owner</c><c>Authenticating party</c>
<c>Authorization server</c><c>Identity service</c>
<c>Resource server</c><c>Identity service</c>
</texttable>
<t>
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.
</t>
<figure>
<artwork><![CDATA[
Alice IdP Bob
---------------------------------------------------------
Call-Id, Fingerprint ------->
<------------------- Auth Code
Auth Code ---------------------------------------------->
<----- Get Token + Auth Code
Token --------------------->
<------------- Get call-info
Call-Id, Fingerprint ------>
]]></artwork>
</figure>
<t>
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.
</t>
<t>
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.
</t>
</section>
</section>
</back>
<!--
On Thu, Sep 22, 2011 at 5:45 AM, Magnus Westerlund <magnus.westerlund@ericsson.com> wrote:
> Hi EKR,
>
> (As an individual)
>
> Thanks for posting the draft.
>
> I am missing a few security issues that I think should be considered.
>
> 1. The attempt to overload the links in an domain by concentrating
> traffic on the domain by choosing peer-pairs. Not that I think there is
> any real protection against this other than limit the flows to their
> "fair" share.
>
> 2. Configuring RTCP or other automatically sent traffic to high
> bit-rates. Especially under conditions where continued consent can't be
> determined.
>
> Cheers
>
> Magnus Westerlund
>
>
>
-->
<!-- TODO
-->
</rfc>
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