rfc8826
Internet Engineering Task Force (IETF) E. Rescorla
Request for Comments: 8826 Mozilla
Category: Standards Track January 2021
ISSN: 2070-1721
Security Considerations for WebRTC
Abstract
WebRTC is a protocol suite for use with real-time applications that
can be deployed in browsers -- "real-time communication on the Web".
This document defines the WebRTC threat model and analyzes the
security threats of WebRTC in that model.
Status of This Memo
This is an Internet Standards Track document.
This document is a product of the Internet Engineering Task Force
(IETF). It represents the consensus of the IETF community. It has
received public review and has been approved for publication by the
Internet Engineering Steering Group (IESG). Further information on
Internet Standards is available in Section 2 of RFC 7841.
Information about the current status of this document, any errata,
and how to provide feedback on it may be obtained at
https://www.rfc-editor.org/info/rfc8826.
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Table of Contents
1. Introduction
2. Terminology
3. The Browser Threat Model
3.1. Access to Local Resources
3.2. Same-Origin Policy
3.3. Bypassing SOP: CORS, WebSockets, and Consent to Communicate
4. Security for WebRTC Applications
4.1. Access to Local Devices
4.1.1. Threats from Screen Sharing
4.1.2. Calling Scenarios and User Expectations
4.1.2.1. Dedicated Calling Services
4.1.2.2. Calling the Site You're On
4.1.3. Origin-Based Security
4.1.4. Security Properties of the Calling Page
4.2. Communications Consent Verification
4.2.1. ICE
4.2.2. Masking
4.2.3. Backward Compatibility
4.2.4. IP Location Privacy
4.3. Communications Security
4.3.1. Protecting Against Retrospective Compromise
4.3.2. Protecting Against During-Call Attack
4.3.2.1. Key Continuity
4.3.2.2. Short Authentication Strings
4.3.2.3. Third-Party Identity
4.3.2.4. Page Access to Media
4.3.3. Malicious Peers
4.4. Privacy Considerations
4.4.1. Correlation of Anonymous Calls
4.4.2. Browser Fingerprinting
5. Security Considerations
6. IANA Considerations
7. References
7.1. Normative References
7.2. Informative References
Acknowledgements
Author's Address
1. Introduction
The Real-Time Communications on the Web (RTCWEB) Working Group has
standardized protocols for real-time communications between Web
browsers, generally called "WebRTC" [RFC8825]. 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. A simple
case is shown below.
+----------------+
| |
| Web Server |
| |
+----------------+
^ ^
/ \
HTTPS / \ HTTPS
or / \ or
WebSockets / \ WebSockets
v v
JS API JS API
+-----------+ +-----------+
| | Media | |
| Browser |<---------->| Browser |
| | | |
+-----------+ +-----------+
Alice Bob
Figure 1: A Simple WebRTC System
In the system shown in Figure 1, Alice and Bob both have WebRTC-
enabled browsers and they visit some Web server which operates a
calling service. Each of their browsers exposes standardized
JavaScript (JS) calling APIs (implemented as browser built-ins) which
are used by the Web server to set up a call between Alice and Bob.
The Web server also serves as the signaling channel to transport
control messages between the browsers. While this system is
topologically similar to a conventional SIP-based system (with the
Web server acting as the signaling service and browsers acting as
softphones), control has moved to the central Web server; the browser
simply provides API points that are used by the calling service. As
with any Web application, the Web server can move logic between the
server and JavaScript in the browser, but regardless of where the
code is executing, it is ultimately under control of the server.
It should be immediately apparent that this type of system poses new
security challenges beyond those of a conventional Voice over IP
(VoIP) system. In particular, it needs to contend with malicious
calling services. For example, if the calling service can cause the
browser to make a call at any time to any callee of its choice, then
this facility can be used to bug a user's computer without their
knowledge, simply by placing a call to some recording service. More
subtly, if the exposed APIs allow the server to instruct the browser
to send arbitrary content, then they can be used to bypass firewalls
or mount denial-of-service (DoS) attacks. Any successful system will
need to be resistant to this and other attacks.
A companion document [RFC8827] describes a security architecture
intended to address the issues raised in this document.
2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
"OPTIONAL" in this document are to be interpreted as described in
BCP 14 [RFC2119] [RFC8174] when, and only when, they appear in all
capitals, as shown here.
3. The Browser Threat Model
The security requirements for WebRTC follow directly from the
requirement that the browser's job is to protect the user. Huang et
al. [huang-w2sp] summarize the core browser security guarantee as
follows:
Users can safely visit arbitrary web sites and execute scripts
provided by those sites.
It is important to realize that this includes sites hosting arbitrary
malicious scripts. The motivation for this requirement is simple: it
is trivial for attackers to divert users to sites of their choice.
For instance, an attacker can purchase display advertisements which
direct the user (either automatically or via user clicking) to their
site, at which point the browser will execute the attacker's scripts.
Thus, it is important that it be safe to view arbitrarily malicious
pages. Of course, browsers inevitably have bugs which cause them to
fall short of this goal, but any new WebRTC functionality must be
designed with the intent to meet this standard. The remainder of
this section provides more background on the existing Web security
model.
In this model, then, the browser acts as a Trusted Computing Base
(TCB) both from the user's perspective and to some extent from the
server's. While HTML and JavaScript provided by the server can cause
the browser to execute a variety of actions, those scripts operate in
a sandbox that isolates them both from the user's computer and from
each other, as detailed below.
Conventionally, we refer to either Web attackers, who are able to
induce you to visit their sites but do not control the network, or
network attackers, who are able to control your network. Network
attackers correspond to the [RFC3552] "Internet Threat Model". Note
that in some cases, a network attacker is also a Web attacker, since
transport protocols that do not provide integrity protection allow
the network to inject traffic as if they were any communications
peer. TLS, and HTTPS in particular, prevent against these attacks,
but when analyzing HTTP connections, we must assume that traffic is
going to the attacker.
3.1. Access to Local Resources
While the browser has access to local resources such as keying
material, files, the camera, and the microphone, it strictly limits
or forbids Web servers from accessing those same resources. For
instance, while it is possible to produce an HTML form which will
allow file upload, a script cannot do so without user consent and in
fact cannot even suggest a specific file (e.g., /etc/passwd); the
user must explicitly select the file and consent to its upload.
(Note: In many cases, browsers are explicitly designed to avoid
dialogs with the semantics of "click here to bypass security checks",
as extensive research [cranor-wolf] shows that users are prone to
consent under such circumstances.)
Similarly, while Flash programs (SWFs) [SWF] can access the camera
and microphone, they explicitly require that the user consent to that
access. In addition, some resources simply cannot be accessed from
the browser at all. For instance, there is no real way to run
specific executables directly from a script (though the user can of
course be induced to download executable files and run them).
3.2. Same-Origin Policy
Many other resources are accessible but isolated. For instance,
while scripts are allowed to make HTTP requests via the fetch() API
(see [fetch]) when requests are made to a server other than from the
same *origin* from whence the script came [RFC6454] they are not able
to read the responses. Cross-Origin Resource Sharing (CORS) [fetch]
and WebSockets [RFC6455] provide an escape hatch from this
restriction, as described below. This same-origin policy (SOP)
prevents server A from mounting attacks on server B via the user's
browser, which protects both the user (e.g., from misuse of their
credentials) and server B (e.g., from DoS attacks).
More generally, SOP forces scripts from each site to run in their
own, isolated, sandboxes. While there are techniques to allow them
to interact, those interactions generally must be mutually consensual
(by each site) and are limited to certain channels. For instance,
multiple pages/browser panes from the same origin can read each
other's JS variables, but pages from different origins -- or even
IFRAMEs from different origins on the same page -- cannot.
3.3. Bypassing SOP: CORS, WebSockets, and Consent to Communicate
While SOP serves an important security function, it also makes it
inconvenient to write certain classes of applications. In
particular, mash-ups, in which a script from origin A uses resources
from origin B, can only be achieved via a certain amount of hackery.
The W3C CORS spec [fetch] is a response to this demand. In CORS,
when a script from origin A executes a potentially unsafe cross-
origin request, the browser instead contacts the target server to
determine whether it is willing to allow cross-origin requests from
A. If it is so willing, the browser then allows the request. This
consent verification process is designed to safely allow cross-origin
requests.
While CORS is designed to allow cross-origin HTTP requests,
WebSockets [RFC6455] allows cross-origin establishment of transparent
channels. Once a WebSockets connection has been established from a
script to a site, the script can exchange any traffic it likes
without being required to frame it as a series of HTTP request/
response transactions. As with CORS, a WebSockets transaction starts
with a consent verification stage to avoid allowing scripts to simply
send arbitrary data to another origin.
While consent verification is conceptually simple -- just do a
handshake before you start exchanging the real data -- experience has
shown that designing a correct consent verification system is
difficult. In particular, Huang et al. [huang-w2sp] have shown
vulnerabilities in the existing Java and Flash consent verification
techniques and in a simplified version of the WebSockets handshake.
It is important to be wary of CROSS-PROTOCOL attacks in which the
attacking script generates traffic which is acceptable to some non-
Web protocol state machine. In order to resist this form of attack,
WebSockets incorporates a masking technique intended to randomize the
bits on the wire, thus making it more difficult to generate traffic
which resembles a given protocol.
4. Security for WebRTC Applications
4.1. Access to Local Devices
As discussed in Section 1, allowing arbitrary sites to initiate calls
violates the core Web security guarantee; without some access
restrictions on local devices, any malicious site could simply bug a
user. At minimum, then, it MUST NOT be possible for arbitrary sites
to initiate calls to arbitrary locations without user consent. This
immediately raises the question, however, of what should be the scope
of user consent.
In order for the user to make an intelligent decision about whether
to allow a call (and hence their camera and microphone input to be
routed somewhere), they must understand either who is requesting
access, where the media is going, or both. As detailed below, there
are two basic conceptual models:
1. You are sending your media to entity A because you want to talk
to entity A (e.g., your mother).
2. Entity A (e.g., a calling service) asks to access the user's
devices with the assurance that it will transfer the media to
entity B (e.g., your mother).
In either case, identity is at the heart of any consent decision.
Moreover, the identity of the party the browser is connecting to is
all that the browser can meaningfully enforce; if you are calling A,
A can simply forward the media to C. Similarly, if you authorize A
to place a call to B, A can call C instead. In either case, all the
browser is able to do is verify and check authorization for whoever
is controlling where the media goes. The target of the media can of
course advertise a security/privacy policy, but this is not something
that the browser can enforce. Even so, there are a variety of
different consent scenarios that motivate different technical consent
mechanisms. We discuss these mechanisms in the sections below.
It's important to understand that consent to access local devices is
largely orthogonal to consent to transmit various kinds of data over
the network (see Section 4.2). Consent for device access is largely
a matter of protecting the user's privacy from malicious sites. By
contrast, consent to send network traffic is about preventing the
user's browser from being used to attack its local network. Thus, we
need to ensure communications consent even if the site is not able to
access the camera and microphone at all (hence WebSockets's consent
mechanism) and similarly, we need to be concerned with the site
accessing the user's camera and microphone even if the data is to be
sent back to the site via conventional HTTP-based network mechanisms
such as HTTP POST.
4.1.1. Threats from Screen Sharing
In addition to camera and microphone access, there has been demand
for screen and/or application sharing functionality. Unfortunately,
the security implications of this functionality are much harder for
users to intuitively analyze than for camera and microphone access.
(See <https://lists.w3.org/Archives/Public/public-
webrtc/2013Mar/0024.html> for a full analysis.)
The most obvious threats are simply those of "oversharing". I.e.,
the user may believe they are sharing a window when in fact they are
sharing an application, or may forget they are sharing their whole
screen, icons, notifications, and all. This is already an issue with
existing screen sharing technologies and is made somewhat worse if a
partially trusted site is responsible for asking for the resource to
be shared rather than having the user propose it.
A less obvious threat involves the impact of screen sharing on the
Web security model. A key part of the Same-Origin Policy is that
HTML or JS from site A can reference content from site B and cause
the browser to load it, but (unless explicitly permitted) cannot see
the result. However, if a Web application from a site is screen
sharing the browser, then this violates that invariant, with serious
security consequences. For example, an attacker site might request
screen sharing and then briefly open up a new window to the user's
bank or webmail account, using screen sharing to read the resulting
displayed content. A more sophisticated attack would be to open up a
source view window to a site and use the screen sharing result to
view anti-cross-site request forgery tokens.
These threats suggest that screen/application sharing might need a
higher level of user consent than access to the camera or microphone.
4.1.2. Calling Scenarios and User Expectations
While a large number of possible calling scenarios are possible, the
scenarios discussed in this section illustrate many of the
difficulties of identifying the relevant scope of consent.
4.1.2.1. Dedicated Calling Services
The first scenario we consider is a dedicated calling service. In
this case, the user has a relationship with a calling site and
repeatedly makes calls on it. It is likely that rather than having
to give permission for each call, the user will want to give the
calling service long-term access to the camera and microphone. This
is a natural fit for a long-term consent mechanism (e.g., installing
an app store "application" to indicate permission for the calling
service). A variant of the dedicated calling service is a gaming
site (e.g., a poker site) which hosts a dedicated calling service to
allow players to call each other.
With any kind of service where the user may use the same service to
talk to many different people, there is a question about whether the
user can know who they are talking to. If I grant permission to
calling service A to make calls on my behalf, then I am implicitly
granting it permission to bug my computer whenever it wants. This
suggests another consent model in which a site is authorized to make
calls but only to certain target entities (identified via media-plane
cryptographic mechanisms as described in Section 4.3.2 and especially
Section 4.3.2.3). Note that the question of consent here is related
to but distinct from the question of peer identity: I might be
willing to allow a calling site to in general initiate calls on my
behalf but still have some calls via that site where I can be sure
that the site is not listening in.
4.1.2.2. Calling the Site You're On
Another simple scenario is calling the site you're actually visiting.
The paradigmatic case here is the "click here to talk to a
representative" windows that appear on many shopping sites. In this
case, the user's expectation is that they are calling the site
they're actually visiting. However, it is unlikely that they want to
provide a general consent to such a site; just because I want some
information on a car doesn't mean that I want the car manufacturer to
be able to activate my microphone whenever they please. Thus, this
suggests the need for a second consent mechanism where I only grant
consent for the duration of a given call. As described in
Section 3.1, great care must be taken in the design of this interface
to avoid the users just clicking through. Note also that the user
interface chrome, which is the representation through which the user
interacts with the user agent itself, must clearly display elements
showing that the call is continuing in order to avoid attacks where
the calling site just leaves it up indefinitely but shows a Web UI
that implies otherwise.
4.1.3. Origin-Based Security
Now that we have described the calling scenarios, we can start to
reason about the security requirements.
As discussed in Section 3.2, the basic unit of Web sandboxing is the
origin, and so it is natural to scope consent to the origin.
Specifically, a script from origin A MUST only be allowed to initiate
communications (and hence to access the camera and microphone) if the
user has specifically authorized access for that origin. It is of
course technically possible to have coarser-scoped permissions, but
because the Web model is scoped to the origin, this creates a
difficult mismatch.
Arguably, the origin is not fine-grained enough. Consider the
situation where Alice visits a site and authorizes it to make a
single call. If consent is expressed solely in terms of the origin,
then on any future visit to that site (including one induced via a
mash-up or ad network), the site can bug Alice's computer, use the
computer to place bogus calls, etc. While in principle Alice could
grant and then revoke the privilege, in practice privileges
accumulate; if we are concerned about this attack, something else is
needed. There are a number of potential countermeasures to this sort
of issue.
Individual Consent
Ask the user for permission for each call.
Callee-oriented Consent
Only allow calls to a given user.
Cryptographic Consent
Only allow calls to a given set of peer keying material or to a
cryptographically established identity.
Unfortunately, none of these approaches is satisfactory for all
cases. As discussed above, individual consent puts the user's
approval in the UI flow for every call. Not only does this quickly
become annoying but it can train the user to simply click "OK", at
which point the consent becomes useless. Thus, while it may be
necessary to have individual consent in some cases, this is not a
suitable solution for (for instance) the calling service case. Where
necessary, in-flow user interfaces must be carefully designed to
avoid the risk of the user blindly clicking through.
The other two options are designed to restrict calls to a given
target. Callee-oriented consent provided by the calling site would
not work well because a malicious site can claim that the user is
calling any user of their choice. One fix for this is to tie calls
to a cryptographically established identity. While not suitable for
all cases, this approach may be useful for some. If we consider the
case of advertising, it's not particularly convenient to require the
advertiser to instantiate an IFRAME on the hosting site just to get
permission; a more convenient approach is to cryptographically tie
the advertiser's certificate to the communication directly. We're
still tying permissions to the origin here, but to the media origin
(and/or destination) rather than to the Web origin. [RFC8827]
describes mechanisms which facilitate this sort of consent.
Another case where media-level cryptographic identity makes sense is
when a user really does not trust the calling site. For instance, I
might be worried that the calling service will attempt to bug my
computer, but I also want to be able to conveniently call my friends.
If consent is tied to particular communications endpoints, then my
risk is limited. Naturally, it is somewhat challenging to design UI
primitives which express this sort of policy. The problem becomes
even more challenging in multi-user calling cases.
4.1.4. Security Properties of the Calling Page
Origin-based security is intended to secure against Web attackers.
However, we must also consider the case of network attackers.
Consider the case where I have granted permission to a calling
service by an origin that has the HTTP scheme, e.g., <http://calling-
service.example.com>. If I ever use my computer on an unsecured
network (e.g., a hotspot or if my own home wireless network is
insecure), and browse any HTTP site, then an attacker can bug my
computer. The attack proceeds like this:
1. I connect to <http://anything.example.org/>. Note that this site
is unaffiliated with the calling service.
2. The attacker modifies my HTTP connection to inject an IFRAME (or
a redirect) to <http://calling-service.example.com>.
3. The attacker forges the response from <http://calling-
service.example.com/> to inject JS to initiate a call to
themselves.
Note that this attack does not depend on the media being insecure.
Because the call is to the attacker, it is also encrypted to them.
Moreover, it need not be executed immediately; the attacker can
"infect" the origin semi-permanently (e.g., with a Web worker or a
popped-up window that is hidden under the main window) and thus be
able to bug me long after I have left the infected network. This
risk is created by allowing calls at all from a page fetched over
HTTP.
Even if calls are only possible from HTTPS [RFC2818] sites, if those
sites include active content (e.g., JavaScript) from an untrusted
site, that JavaScript is executed in the security context of the page
[finer-grained]. This could lead to compromise of a call even if the
parent page is safe. Note: This issue is not restricted to *pages*
which contain untrusted content. If any page from a given origin
ever loads JavaScript from an attacker, then it is possible for that
attacker to infect the browser's notion of that origin semi-
permanently.
4.2. Communications Consent Verification
As discussed in Section 3.3, allowing Web applications unrestricted
network access via the browser introduces the risk of using the
browser as an attack platform against machines which would not
otherwise be accessible to the malicious site, for instance, because
they are topologically restricted (e.g., behind a firewall or NAT).
In order to prevent this form of attack as well as cross-protocol
attacks, it is important to require that the target of traffic
explicitly consent to receiving the traffic in question. Until that
consent has been verified for a given endpoint, traffic other than
the consent handshake MUST NOT be sent to that endpoint.
Note that consent verification is not sufficient to prevent overuse
of network resources. Because WebRTC allows for a Web site to create
data flows between two browser instances without user consent, it is
possible for a malicious site to chew up a significant amount of a
user's bandwidth without incurring significant costs to themselves by
setting up such a channel to another user. However, as a practical
matter there are a large number of Web sites which can act as data
sources, so an attacker can at least use downlink bandwidth with
existing Web APIs. However, this potential DoS vector reinforces the
need for adequate congestion control for WebRTC protocols to ensure
that they play fair with other demands on the user's bandwidth.
4.2.1. ICE
Verifying receiver consent requires some sort of explicit handshake,
but conveniently we already need one in order to do NAT hole-
punching. Interactive Connectivity Establishment (ICE) [RFC8445]
includes a handshake designed to verify that the receiving element
wishes to receive traffic from the sender. It is important to
remember here that the site initiating ICE is presumed malicious; in
order for the handshake to be secure, the receiving element MUST
demonstrate receipt/knowledge of some value not available to the site
(thus preventing the site from forging responses). In order to
achieve this objective with ICE, the Session Traversal Utilities for
NAT (STUN) transaction IDs must be generated by the browser and MUST
NOT be made available to the initiating script, even via a diagnostic
interface. Verifying receiver consent also requires verifying the
receiver wants to receive traffic from a particular sender, and at
this time; for example, a malicious site may simply attempt ICE to
known servers that are using ICE for other sessions. ICE provides
this verification as well, by using the STUN credentials as a form of
per-session shared secret. Those credentials are known to the Web
application, but would need to also be known and used by the STUN-
receiving element to be useful.
There also needs to be some mechanism for the browser to verify that
the target of the traffic continues to wish to receive it. Because
ICE keepalives are indications, they will not work here. [RFC7675]
describes the mechanism for providing consent freshness.
4.2.2. Masking
Once consent is verified, there still is some concern about
misinterpretation attacks as described by Huang et al. [huang-w2sp].
This does not seem like it is of serious concern with DTLS because
the ICE handshake enforces receiver consent and there is little
evidence of passive DTLS proxies of the type studied by Huang.
However, because RTCWEB can run over TCP there is some concern that
attackers might control the ciphertext by controlling the plaintext
input to SCTP. This risk is only partially mitigated by the fact
that the SCTP stack controls the framing of the packets.
Note that in principle an attacker could exert some control over
Secure Real-time Transport Protocol (SRTP) packets by using a
combination of the WebAudio API and extremely tight timing control.
The primary risk here seems to be carriage of SRTP over Traversal
Using Relays around NAT (TURN) TCP. However, as SRTP packets have an
extremely characteristic packet header it seems unlikely that any but
the most aggressive intermediaries would be confused into thinking
that another application-layer protocol was in use.
4.2.3. Backward Compatibility
| Note: The RTCWEB WG ultimately decided to require ICE. This
| section provides context for that decision.
A requirement to use ICE limits compatibility with legacy non-ICE
clients. It seems unsafe to completely remove the requirement for
some check. All proposed checks have the common feature that the
browser sends some message to the candidate traffic recipient and
refuses to send other traffic until that message has been replied to.
The message/reply pair must be generated in such a way that an
attacker who controls the Web application cannot forge them,
generally by having the message contain some secret value that must
be incorporated (e.g., echoed, hashed into, etc.). Non-ICE
candidates for this role (in cases where the legacy endpoint has a
public address) include:
* STUN checks without using ICE (i.e., the non-RTC-web endpoint sets
up a STUN responder).
* Use of the RTP Control Protocol (RTCP) as an implicit reachability
check.
In the RTCP approach, the WebRTC endpoint is allowed to send a
limited number of RTP packets prior to receiving consent. This
allows a short window of attack. In addition, some legacy endpoints
do not support RTCP, so this is a much more expensive solution for
such endpoints, for which it would likely be easier to implement ICE.
For these two reasons, an RTCP-based approach does not seem to
address the security issue satisfactorily.
In the STUN approach, the WebRTC endpoint is able to verify that the
recipient is running some kind of STUN endpoint but unless the STUN
responder is integrated with the ICE username/password establishment
system, the WebRTC endpoint cannot verify that the recipient consents
to this particular call. This may be an issue if existing STUN
servers are operated at addresses that are not able to handle
bandwidth-based attacks. Thus, this approach does not seem
satisfactory either.
If the systems are tightly integrated (i.e., the STUN endpoint
responds with responses authenticated with ICE credentials), then
this issue does not exist. However, such a design is very close to
an ICE-Lite implementation (indeed, arguably is one). An
intermediate approach would be to have a STUN extension that
indicated that one was responding to WebRTC checks but not computing
integrity checks based on the ICE credentials. This would allow the
use of standalone STUN servers without the risk of confusing them
with legacy STUN servers. If a non-ICE legacy solution is needed,
then this is probably the best choice.
Once initial consent is verified, we also need to verify continuing
consent, in order to avoid attacks where two people briefly share an
IP (e.g., behind a NAT in an Internet cafe) and the attacker arranges
for a large, unstoppable, traffic flow to the network and then
leaves. The appropriate technologies here are fairly similar to
those for initial consent, though are perhaps weaker since the
threats are less severe.
4.2.4. IP Location Privacy
Note that as soon as the callee sends their ICE candidates, the
caller learns the callee's IP addresses. The callee's server-
reflexive address reveals a lot of information about the callee's
location. In order to avoid tracking, implementations may wish to
suppress the start of ICE negotiation until the callee has answered.
In addition, either side may wish to hide their location from the
other side entirely by forcing all traffic through a TURN server.
In ordinary operation, the site learns the browser's IP address,
though it may be hidden via mechanisms like Tor
<https://www.torproject.org> or a VPN. However, because sites can
cause the browser to provide IP addresses, this provides a mechanism
for sites to learn about the user's network environment even if the
user is behind a VPN that masks their IP address. Implementations
may wish to provide settings which suppress all non-VPN candidates if
the user is on certain kinds of VPN, especially privacy-oriented
systems such as Tor. See [RFC8828] for additional information.
4.3. Communications Security
Finally, we consider a problem familiar from the SIP world:
communications security. For obvious reasons, it MUST be possible
for the communicating parties to establish a channel which is secure
against both message recovery and message modification. (See
[RFC5479] for more details.) This service must be provided for both
data and voice/video. Ideally the same security mechanisms would be
used for both types of content. Technology for providing this
service (for instance, SRTP [RFC3711], DTLS [RFC6347], and DTLS-SRTP
[RFC5763]) is well understood. However, we must examine this
technology in the WebRTC context, where the threat model is somewhat
different.
In general, it is important to understand that unlike a conventional
SIP proxy, the calling service (i.e., the Web server) controls not
only the channel between the communicating endpoints but also the
application running on the user's browser. While in principle it is
possible for the browser to cut the calling service out of the loop
and directly present trusted information (and perhaps get consent),
practice in modern browsers is to avoid this whenever possible.
"In-flow" modal dialogs which require the user to consent to specific
actions are particularly disfavored as human factors research
indicates that unless they are made extremely invasive, users simply
agree to them without actually consciously giving consent
[abarth-rtcweb]. Thus, nearly all the UI will necessarily be
rendered by the browser but under control of the calling service.
This likely includes the peer's identity information, which, after
all, is only meaningful in the context of some calling service.
This limitation does not mean that preventing attack by the calling
service is completely hopeless. However, we need to distinguish
between two classes of attack:
Retrospective compromise of calling service:
The calling service is non-malicious during a call but
subsequently is compromised and wishes to attack an older call
(often called a "passive attack").
During-call attack by calling service:
The calling service is compromised during the call it wishes to
attack (often called an "active attack").
Providing security against the former type of attack is practical
using the techniques discussed in Section 4.3.1. However, it is
extremely difficult to prevent a trusted but malicious calling
service from actively attacking a user's calls, either by mounting a
Man-in-the-Middle (MITM) attack or by diverting them entirely. (Note
that this attack applies equally to a network attacker if
communications to the calling service are not secured.) We discuss
some potential approaches in Section 4.3.2.
4.3.1. Protecting Against Retrospective Compromise
In a retrospective attack, the calling service was uncompromised
during the call, but an attacker subsequently wants to recover the
content of the call. We assume that the attacker has access to the
protected media stream as well as full control of the calling
service.
If the calling service has access to the traffic keying material (as
in Security Descriptions (SDES) [RFC4568]), then retrospective attack
is trivial. This form of attack is particularly serious in the Web
context because it is standard practice in Web services to run
extensive logging and monitoring. Thus, it is highly likely that if
the traffic key is part of any HTTP request it will be logged
somewhere and thus subject to subsequent compromise. It is this
consideration that makes an automatic, public key-based key exchange
mechanism imperative for WebRTC (this is a good idea for any
communications security system), and this mechanism SHOULD provide
Forward Secrecy (FS). The signaling channel/calling service can be
used to authenticate this mechanism.
In addition, if end-to-end keying is used, the system MUST NOT
provide any APIs to either extract long-term keying material or to
directly access any stored traffic keys. Otherwise, an attacker who
subsequently compromised the calling service might be able to use
those APIs to recover the traffic keys and thus compromise the
traffic.
4.3.2. Protecting Against During-Call Attack
Protecting against attacks during a call is a more difficult
proposition. Even if the calling service cannot directly access
keying material (as recommended in the previous section), it can
simply mount a man-in-the-middle attack on the connection, telling
Alice that she is calling Bob and Bob that he is calling Alice, while
in fact the calling service is acting as a calling bridge and
capturing all the traffic. Protecting against this form of attack
requires positive authentication of the remote endpoint such as
explicit out-of-band key verification (e.g., by a fingerprint) or a
third-party identity service as described in [RFC8827].
4.3.2.1. Key Continuity
One natural approach is to use "key continuity". While a malicious
calling service can present any identity it chooses to the user, it
cannot produce a private key that maps to a given public key. Thus,
it is possible for the browser to note a given user's public key and
generate an alarm whenever that user's key changes. The Secure Shell
(SSH) protocol [RFC4251] uses a similar technique. (Note that the
need to avoid explicit user consent on every call precludes the
browser requiring an immediate manual check of the peer's key.)
Unfortunately, this sort of key continuity mechanism is far less
useful in the WebRTC context. First, much of the virtue of WebRTC
(and any Web application) is that it is not bound to a particular
piece of client software. Thus, it will be not only possible but
routine for a user to use multiple browsers on different computers
that will of course have different keying material (Securely
Available Credentials (SACRED) [RFC3760] notwithstanding). Thus,
users will frequently be alerted to key mismatches which are in fact
completely legitimate, with the result that they are trained to
simply click through them. As it is known that users routinely will
click through far more dire warnings [cranor-wolf], it seems
extremely unlikely that any key continuity mechanism will be
effective rather than simply annoying.
Moreover, it is trivial to bypass even this kind of mechanism.
Recall that unlike the case of SSH, the browser never directly gets
the peer's identity from the user. Rather, it is provided by the
calling service. Even enabling a mechanism of this type would
require an API to allow the calling service to tell the browser "this
is a call to user X." All the calling service needs to do to avoid
triggering a key continuity warning is to tell the browser that "this
is a call to user Y" where Y is confusable with X. Even if the user
actually checks the other side's name (which all available evidence
indicates is unlikely), this would require (a) the browser to use the
trusted UI to provide the name and (b) the user to not be fooled by
similar appearing names.
4.3.2.2. Short Authentication Strings
ZRTP [RFC6189] uses a "Short Authentication String" (SAS) which is
derived from the key agreement protocol. This SAS is designed to be
compared by the users (e.g., read aloud over the voice channel or
transmitted via an out-of-band channel) and if confirmed by both
sides precludes MITM attack. The intention is that the SAS is used
once and then key continuity (though with a different mechanism from
that discussed above) is used thereafter.
Unfortunately, the SAS does not offer a practical solution to the
problem of a compromised calling service. "Voice cloning" systems,
which mimic the voice of a given speaker are an active area of
research [deepfakes-ftc] and are already being used in real-world
attacks [deepfakes-fraud]. These attacks are likely to improve in
future, especially in an environment where the user just wants to get
on with the phone call. Thus, even if the SAS is effective today, it
is likely not to be so for much longer.
Additionally, it is unclear that users will actually use an SAS. As
discussed above, the browser UI constraints preclude requiring the
SAS exchange prior to completing the call and so it must be
voluntary; at most the browser will provide some UI indicator that
the SAS has not yet been checked. However, it is well known that
when faced with optional security mechanisms, many users simply
ignore them [whitten-johnny].
Once users have checked the SAS once, key continuity is required to
avoid them needing to check it on every call. However, this is
problematic for reasons indicated in Section 4.3.2.1. In principle
it is of course possible to render a different UI element to indicate
that calls are using an unauthenticated set of keying material
(recall that the attacker can just present a slightly different name
so that the attack shows the same UI as a call to a new device or to
someone you haven't called before), but as a practical matter, users
simply ignore such indicators even in the rather more dire case of
mixed content warnings.
4.3.2.3. Third-Party Identity
The conventional approach to providing communications identity has of
course been to have some third-party identity system (e.g., PKI) to
authenticate the endpoints. Such mechanisms have proven to be too
cumbersome for use by typical users (and nearly too cumbersome for
administrators). However, a new generation of Web-based identity
providers (BrowserID, Federated Google Login, Facebook Connect, OAuth
[RFC6749], OpenID [OpenID], WebFinger [RFC7033]) has been developed
and use Web technologies to provide lightweight (from the user's
perspective) third-party authenticated transactions. It is possible
to use systems of this type to authenticate WebRTC calls, linking
them to existing user notions of identity (e.g., Facebook
adjacencies). Specifically, the third-party identity system is used
to bind the user's identity to cryptographic keying material which is
then used to authenticate the calling endpoints. Calls which are
authenticated in this fashion are naturally resistant even to active
MITM attack by the calling site.
Note that there is one special case in which PKI-style certificates
do provide a practical solution: calls from end users to large sites.
For instance, if you are making a call to Amazon.com, then Amazon can
easily get a certificate to authenticate their media traffic, just as
they get one to authenticate their Web traffic. This does not
provide additional security value in cases in which the calling site
and the media peer are one and the same, but might be useful in cases
in which third parties (e.g., ad networks or retailers) arrange for
calls but do not participate in them.
4.3.2.4. Page Access to Media
Identifying the identity of the far media endpoint is a necessary but
not sufficient condition for providing media security. In WebRTC,
media flows are rendered into HTML5 MediaStreams which can be
manipulated by the calling site. Obviously, if the site can modify
or view the media, then the user is not getting the level of
assurance they would expect from being able to authenticate their
peer. In many cases, this is acceptable because the user values
site-based special effects over complete security from the site.
However, there are also cases where users wish to know that the site
cannot interfere. In order to facilitate that, it will be necessary
to provide features whereby the site can verifiably give up access to
the media streams. This verification must be possible both from the
local side and the remote side. I.e., users must be able to verify
that the person called has engaged a secure media mode (see
Section 4.3.3). In order to achieve this, it will be necessary to
cryptographically bind an indication of the local media access policy
into the cryptographic authentication procedures detailed in the
previous sections.
It should be noted that the use of this secure media mode is left to
the discretion of the site. When such a mode is engaged, the browser
will need to provide indicia to the user that the associated media
has been authenticated as coming from the identified user. This
allows WebRTC services that wish to claim end-to-end security to do
so in a way that can be easily verified by the user. This model
requires that the remote party's browser be included in the TCB, as
described in Section 3.
4.3.3. Malicious Peers
One class of attack that we do not generally try to prevent is
malicious peers. For instance, no matter what confidentiality
measures you employ the person you are talking to might record the
call and publish it on the Internet. Similarly, we do not attempt to
prevent them from using voice or video processing technology for
hiding or changing their appearance. While technologies (Digital
Rights Management (DRM), etc.) do exist to attempt to address these
issues, they are generally not compatible with open systems and
WebRTC does not address them.
Similarly, we make no attempt to prevent prank calling or other
unwanted calls. In general, this is in the scope of the calling
site, though because WebRTC does offer some forms of strong
authentication, that may be useful as part of a defense against such
attacks.
4.4. Privacy Considerations
4.4.1. Correlation of Anonymous Calls
While persistent endpoint identifiers can be a useful security
feature (see Section 4.3.2.1), they can also represent a privacy
threat in settings where the user wishes to be anonymous. WebRTC
provides a number of possible persistent identifiers such as DTLS
certificates (if they are reused between connections) and RTCP CNAMEs
(if generated according to [RFC6222] rather than the privacy-
preserving mode of [RFC7022]). In order to prevent this type of
correlation, browsers need to provide mechanisms to reset these
identifiers (e.g., with the same lifetime as cookies). Moreover, the
API should provide mechanisms to allow sites intended for anonymous
calling to force the minting of fresh identifiers. In addition, IP
addresses can be a source of call linkage [RFC8828].
4.4.2. Browser Fingerprinting
Any new set of API features adds a risk of browser fingerprinting,
and WebRTC is no exception. Specifically, sites can use the presence
or absence of specific devices as a browser fingerprint. In general,
the API needs to be balanced between functionality and the
incremental fingerprint risk. See [Fingerprinting].
5. Security Considerations
This entire document is about security.
6. IANA Considerations
This document has no IANA actions.
7. References
7.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997,
<https://www.rfc-editor.org/info/rfc2119>.
[RFC8174] Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC
2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174,
May 2017, <https://www.rfc-editor.org/info/rfc8174>.
7.2. Informative References
[abarth-rtcweb]
Barth, A., "Prompting the user is security failure", RTC-
Web Workshop, September 2010, <http://rtc-
web.alvestrand.com/home/papers/barth-security-
prompt.pdf?attredirects=0>.
[cranor-wolf]
Sunshine, J., Egelman, S., Almuhimedi, H., Atri, N., and
L. Cranor, "Crying Wolf: An Empirical Study of SSL Warning
Effectiveness", Proceedings of the 18th USENIX Security
Symposium, August 2009,
<https://www.usenix.org/legacy/event/sec09/tech/
full_papers/sunshine.pdf>.
[deepfakes-fraud]
Statt, N., "Thieves are now using AI deepfakes to trick
companies into sending them money", September 2019,
<https://www.theverge.com/2019/9/5/20851248/deepfakes-ai-
fake-audio-phone-calls-thieves-trick-companies-stealing-
money>.
[deepfakes-ftc]
Lyons, K., "FTC says the tech behind audio deepfakes is
getting better", January 2020,
<https://www.theverge.com/2020/1/29/21080553/ftc-
deepfakes-audio-cloning-joe-rogan-phone-scams>.
[fetch] van Kesteren, A., "Fetch",
<https://fetch.spec.whatwg.org/>.
[finer-grained]
Jackson, C. and A. Barth, "Beware of Finer-Grained
Origins", Web 2.0 Security and Privacy (W2SP 2008), July
2008.
[Fingerprinting]
Doty, N., Ed., "Mitigating Browser Fingerprinting in Web
Specifications", March 2019,
<https://www.w3.org/TR/fingerprinting-guidance/>.
[huang-w2sp]
Huang, L-S., Chen, E.Y., Barth, A., Rescorla, E., and C.
Jackson, "Talking to Yourself for Fun and Profit", Web 2.0
Security and Privacy (W2SP 2011), May 2011.
[OpenID] Sakimura, N., Bradley, J., Jones, M., de Medeiros, B., and
C. Mortimore, "OpenID Connect Core 1.0", November 2014,
<https://openid.net/specs/openid-connect-core-1_0.html>.
[RFC2818] Rescorla, E., "HTTP Over TLS", RFC 2818,
DOI 10.17487/RFC2818, May 2000,
<https://www.rfc-editor.org/info/rfc2818>.
[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
A., Peterson, J., Sparks, R., Handley, M., and E.
Schooler, "SIP: Session Initiation Protocol", RFC 3261,
DOI 10.17487/RFC3261, June 2002,
<https://www.rfc-editor.org/info/rfc3261>.
[RFC3552] Rescorla, E. and B. Korver, "Guidelines for Writing RFC
Text on Security Considerations", BCP 72, RFC 3552,
DOI 10.17487/RFC3552, July 2003,
<https://www.rfc-editor.org/info/rfc3552>.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, DOI 10.17487/RFC3711, March 2004,
<https://www.rfc-editor.org/info/rfc3711>.
[RFC3760] Gustafson, D., Just, M., and M. Nystrom, "Securely
Available Credentials (SACRED) - Credential Server
Framework", RFC 3760, DOI 10.17487/RFC3760, April 2004,
<https://www.rfc-editor.org/info/rfc3760>.
[RFC4251] Ylonen, T. and C. Lonvick, Ed., "The Secure Shell (SSH)
Protocol Architecture", RFC 4251, DOI 10.17487/RFC4251,
January 2006, <https://www.rfc-editor.org/info/rfc4251>.
[RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session
Description Protocol (SDP) Security Descriptions for Media
Streams", RFC 4568, DOI 10.17487/RFC4568, July 2006,
<https://www.rfc-editor.org/info/rfc4568>.
[RFC5479] Wing, D., Ed., Fries, S., Tschofenig, H., and F. Audet,
"Requirements and Analysis of Media Security Management
Protocols", RFC 5479, DOI 10.17487/RFC5479, April 2009,
<https://www.rfc-editor.org/info/rfc5479>.
[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, DOI 10.17487/RFC5763, May
2010, <https://www.rfc-editor.org/info/rfc5763>.
[RFC6189] Zimmermann, P., Johnston, A., Ed., and J. Callas, "ZRTP:
Media Path Key Agreement for Unicast Secure RTP",
RFC 6189, DOI 10.17487/RFC6189, April 2011,
<https://www.rfc-editor.org/info/rfc6189>.
[RFC6222] Begen, A., Perkins, C., and D. Wing, "Guidelines for
Choosing RTP Control Protocol (RTCP) Canonical Names
(CNAMEs)", RFC 6222, DOI 10.17487/RFC6222, April 2011,
<https://www.rfc-editor.org/info/rfc6222>.
[RFC6347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer
Security Version 1.2", RFC 6347, DOI 10.17487/RFC6347,
January 2012, <https://www.rfc-editor.org/info/rfc6347>.
[RFC6454] Barth, A., "The Web Origin Concept", RFC 6454,
DOI 10.17487/RFC6454, December 2011,
<https://www.rfc-editor.org/info/rfc6454>.
[RFC6455] Fette, I. and A. Melnikov, "The WebSocket Protocol",
RFC 6455, DOI 10.17487/RFC6455, December 2011,
<https://www.rfc-editor.org/info/rfc6455>.
[RFC6749] Hardt, D., Ed., "The OAuth 2.0 Authorization Framework",
RFC 6749, DOI 10.17487/RFC6749, October 2012,
<https://www.rfc-editor.org/info/rfc6749>.
[RFC7022] Begen, A., Perkins, C., Wing, D., and E. Rescorla,
"Guidelines for Choosing RTP Control Protocol (RTCP)
Canonical Names (CNAMEs)", RFC 7022, DOI 10.17487/RFC7022,
September 2013, <https://www.rfc-editor.org/info/rfc7022>.
[RFC7033] Jones, P., Salgueiro, G., Jones, M., and J. Smarr,
"WebFinger", RFC 7033, DOI 10.17487/RFC7033, September
2013, <https://www.rfc-editor.org/info/rfc7033>.
[RFC7675] Perumal, M., Wing, D., Ravindranath, R., Reddy, T., and M.
Thomson, "Session Traversal Utilities for NAT (STUN) Usage
for Consent Freshness", RFC 7675, DOI 10.17487/RFC7675,
October 2015, <https://www.rfc-editor.org/info/rfc7675>.
[RFC8445] Keranen, A., Holmberg, C., and J. Rosenberg, "Interactive
Connectivity Establishment (ICE): A Protocol for Network
Address Translator (NAT) Traversal", RFC 8445,
DOI 10.17487/RFC8445, July 2018,
<https://www.rfc-editor.org/info/rfc8445>.
[RFC8825] Alvestrand, H., "Overview: Real-Time Protocols for
Browser-Based Applications", RFC 8825,
DOI 10.17487/RFC8825, January 2021,
<https://www.rfc-editor.org/info/rfc8825>.
[RFC8827] Rescorla, E., "WebRTC Security Architecture", RFC 8827,
DOI 10.17487/RFC8827, January 2021,
<https://www.rfc-editor.org/info/rfc8827>.
[RFC8828] Uberti, J. and G. Shieh, "WebRTC IP Address Handling
Requirements", RFC 8828, DOI 10.17487/RFC8828, January
2021, <https://www.rfc-editor.org/info/rfc8828>.
[SWF] "SWF File Format Specification Version 19", April 2013,
<https://www.adobe.com/content/dam/acom/en/devnet/pdf/swf-
file-format-spec.pdf>.
[whitten-johnny]
Whitten, A. and J.D. Tygar, "Why Johnny Can't Encrypt: A
Usability Evaluation of PGP 5.0", Proceedings of the 8th
USENIX Security Symposium, August 1999,
<https://www.usenix.org/legacy/publications/library/
proceedings/sec99/whitten.html>.
Acknowledgements
Bernard Aboba, Harald Alvestrand, Dan Druta, Cullen Jennings, Alan
Johnston, Hadriel Kaplan (Section 4.2.1), Matthew Kaufman, Martin
Thomson, Magnus Westerlund.
Author's Address
Eric Rescorla
Mozilla
Email: ekr@rtfm.com
ERRATA
