WebRTC IP Address Handling
Requirements
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RAI
This document provides information and requirements for how IP
addresses should be handled by Web Real-Time Communication (WebRTC) implementations.
Introduction
One of WebRTC's key features is its support of peer-to-peer
connections. However, when establishing such a connection, which involves
connection attempts from various IP addresses, WebRTC may allow a web
application to learn additional information about the user compared to an
application that only uses the Hypertext Transfer Protocol (HTTP)
. This may be problematic in
certain cases. This
document summarizes the concerns and makes recommendations on how WebRTC
implementations should best handle the trade-off between privacy and media
performance.
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 when, and only when, they appear in all capitals, as
shown here.
Problem Statement
In order to establish a peer-to-peer connection, WebRTC
implementations use Interactive Connectivity Establishment (ICE)
. ICE attempts to discover multiple IP
addresses using techniques such as Session Traversal Utilities for NAT
(STUN)
and Traversal Using Relays
around NAT (TURN)
and then checks the
connectivity of each
local-address-remote-address pair in order to select the best one. The
addresses that are collected usually consist of an endpoint's private
physical or virtual addresses and its public Internet addresses.
These addresses are provided to the web application so that
they can be communicated to the remote endpoint for its checks. This
allows the application to learn more about the local network
configuration than it would from a typical HTTP scenario, in which the
web server would only see a single public Internet address, i.e., the
address from which the HTTP request was sent.
The additional information revealed falls into three categories:
- If the client is multihomed, additional public IP addresses for the
client can be learned. In particular, if the client tries to hide its
physical location through a Virtual Private Network (VPN), and the VPN
and local OS support routing over multiple interfaces (a "split-tunnel"
VPN), WebRTC can discover not only the public address for the VPN, but
also the ISP public address over which the VPN is running.
- If the client is behind a Network Address Translator (NAT), the
client's private IP addresses, often
addresses, can be learned.
- If the client is behind a proxy (a client-configured "classical
application proxy", as defined in
), but direct access to the
Internet is permitted, WebRTC's STUN checks will bypass the proxy and
reveal the public IP address of the client. This concern also applies
to the "enterprise TURN server" scenario described in
if, as above, direct
Internet access is permitted. However, when the term "proxy" is used in
this document, it is always in reference to an
proxy server.
Of these three concerns, the first is the most significant, because for some
users, the purpose of using a VPN is for anonymity. However, different
VPN users will have different needs, and some VPN users (e.g., corporate
VPN users) may in fact prefer WebRTC to send media traffic directly --
i.e., not through the VPN.
The second concern is less significant but valid nonetheless. The core
issue is that web applications can learn about addresses that are not
exposed to the Internet; typically, these address are IPv4, but they can
also be IPv6, as in the case of NAT64 .
While disclosure of the IPv6 addresses
recommended by is fairly
benign due to their intentionally short lifetimes, IPv4 addresses present
some challenges. Although private IPv4 addresses often contain minimal
entropy (e.g., 192.168.0.2, a fairly common address), in the worst case,
they can contain 24 bits of entropy with an indefinite lifetime. As such,
they can be a fairly significant fingerprinting surface. In addition,
intranet web sites can be attacked more easily when their IPv4 address
range is externally known.
Private IP addresses can also act as an identifier that allows
web applications running in isolated browsing contexts (e.g., normal and
private browsing) to learn that they are running on the same device. This
could allow the application sessions to be correlated, defeating some of
the privacy protections provided by isolation. It should be noted that
private addresses are just one potential mechanism for this correlation
and this is an area for further study.
The third concern is the least common, as proxy administrators can already
control this behavior through organizational firewall policy, and
generally, forcing WebRTC traffic through a proxy server will have
negative effects on both the proxy and media quality.
Note also that these concerns predate WebRTC; Adobe Flash Player has
provided similar functionality since the introduction of Real-Time
Media Flow Protocol (RTMFP) support
in 2008.
Goals
WebRTC's support of secure peer-to-peer connections facilitates
deployment of decentralized systems, which can have privacy benefits. As
a result, blunt solutions that disable WebRTC or make it significantly
harder to use are undesirable. This document takes a more nuanced
approach, with the following goals:
- Provide a framework for understanding the problem so that controls
might be provided to make different trade-offs regarding performance and
privacy concerns with WebRTC.
- Using that framework, define settings that enable peer-to-peer
communications, each with a different balance between performance and
privacy.
- Finally, provide recommendations for default settings that provide
reasonable performance without also exposing addressing information in
a way that might violate user expectations.
Detailed Design
Principles
The key principles for our framework are stated below:
- By default, WebRTC traffic should follow typical IP routing (i.e.,
WebRTC should use the same interface used for HTTP traffic) and only
the system's 'typical' public addresses (or those of an enterprise
TURN server, if present) should be visible to the application.
However, in the interest of optimal media quality, it should be
possible to enable WebRTC to make use of all network interfaces to
determine the ideal route.
- By default, WebRTC should be able to negotiate direct peer-to-peer
connections between endpoints (i.e., without traversing a NAT or
relay server) when such connections are possible. This ensures that
applications that need true peer-to-peer routing for bandwidth or
latency reasons can operate successfully.
- It should be possible to configure WebRTC to not disclose private
local IP addresses, to avoid the issues associated with web
applications learning such addresses. This document does not require
this to be the default state, as there is no currently defined
mechanism that can satisfy this requirement as well as the
aforementioned requirement to allow direct peer-to-peer
connections.
- By default, WebRTC traffic should not be sent through proxy
servers, due to the media-quality problems associated with sending
WebRTC traffic over TCP, which is almost always used when
communicating with such proxies, as well as proxy performance issues
that may result from proxying WebRTC's long-lived, high-bandwidth
connections. However, it should be possible to force WebRTC to send
its traffic through a configured proxy if desired.
Modes and Recommendations
Based on these ideas, we define four specific modes of WebRTC
behavior, reflecting different media quality/privacy trade-offs:
- Mode 1 - Enumerate all addresses:
- WebRTC MUST use all network interfaces to
attempt communication with STUN servers, TURN servers, or peers. This
will converge on the best media path and is ideal when media
performance is the highest priority, but it discloses the most
information.
- Mode 2 - Default route + associated local addresses:
- WebRTC
MUST follow the
kernel routing table rules, which will typically cause media packets
to take the same route as the application's HTTP traffic. If an
enterprise TURN server is present, the preferred route MUST be
through this TURN server. Once an interface has been chosen, the
private IPv4 and IPv6 addresses associated with this interface MUST
be discovered and provided to the application as host candidates.
This ensures that direct connections can still be established in this
mode.
- Mode 3 - Default route only:
- This is the same as Mode 2, except that
the associated private addresses MUST NOT be provided; the only IP
addresses gathered are those discovered via mechanisms like STUN and
TURN (on the default route). This may cause traffic to hairpin
through a NAT, fall back to an application TURN server, or fail
altogether, with resulting quality implications.
- Mode 4 - Force proxy:
- This is the same as Mode 3, but when the
application's HTTP traffic is sent through a proxy, WebRTC media
traffic MUST also be proxied. If the proxy does not support UDP (as
is the case for all HTTP and most SOCKS
proxies), or the WebRTC implementation does
not support UDP proxying, the use of UDP will be disabled, and TCP
will be used to send and receive media through the proxy. Use of TCP
will result in reduced media quality, in addition to any performance
considerations associated with sending all WebRTC media through the
proxy server.
Mode 1 MUST NOT be used unless user consent has been provided. The
details of this consent are left to the implementation; one potential
mechanism is to tie this consent to getUserMedia (device permissions)
consent, described in .
Alternatively, implementations can provide a specific
mechanism to obtain user consent.
In cases where user consent has not been obtained, Mode 2 SHOULD be
used.
These defaults provide a reasonable trade-off that permits trusted
WebRTC applications to achieve optimal network performance but gives
applications without consent (e.g., 1-way streaming or data-channel
applications) only the minimum information needed to achieve direct
connections, as defined in Mode 2. However, implementations MAY choose
stricter modes if desired, e.g., if a user indicates they want all
WebRTC traffic to follow the default route.
Future documents may define additional modes and/or update the
recommended default modes.
Note that the suggested defaults can still be used even for
organizations that want all external WebRTC traffic to traverse a proxy
or enterprise TURN server, simply by setting an organizational firewall
policy that allows WebRTC traffic to only leave through the proxy or
TURN server. This provides a way to ensure the proxy or TURN server is
used for any external traffic but still allows direct connections
(and, in the proxy case, avoids the performance issues associated with
forcing media through said proxy) for intra-organization traffic.
Implementation Guidance
This section provides guidance to WebRTC implementations on how to
implement the policies described above.
Ensuring Normal Routing
When trying to follow typical IP routing, as required by Modes 2
and 3, the simplest approach is
to bind() the sockets used for peer-to-peer connections to the wildcard
addresses (0.0.0.0 for IPv4, :: for IPv6), which allows the OS to route
WebRTC traffic the same way as it would HTTP traffic. STUN and TURN
will work as usual, and host candidates can still be determined as
mentioned below.
Determining Associated Local Addresses
When binding to a wildcard address, some extra work is needed to
determine the associated local address required by Mode 2, which we
define as the source
address that would be used for any packets sent to the web application
host (assuming that UDP and TCP get the same routing treatment). Use of
the web-application host as a destination ensures the right source
address is selected, regardless of where the application resides (e.g.,
on an intranet).
First, the appropriate remote IPv4/IPv6 address is obtained by
resolving the host component of the web application URI
. If the client is behind a proxy and cannot
resolve these IPs via DNS, the address of the proxy can be used
instead. Or, if the web application was loaded from a file:// URI
rather than over the network, the
implementation can fall back to a well-known DNS name or IP
address.
Once a suitable remote IP has been determined, the implementation
can create a UDP socket, bind() it to the appropriate wildcard address,
and then connect() to the remote IP. Generally, this results in
the socket being assigned a local address based on the kernel routing
table, without sending any packets over the network.
Finally, the socket can be queried using getsockname() or the
equivalent to determine the appropriate local address.
Application Guidance
The recommendations mentioned in this document may cause certain
WebRTC applications to malfunction. In order to be robust in all
scenarios, the following guidelines are provided for applications:
- Applications SHOULD deploy a TURN server with support for both UDP
and TCP connections to the server. This ensures that connectivity can
still be established, even when Mode 3 or 4 is in use, assuming the
TURN server can be reached.
- Applications SHOULD detect when they don't have access to the full
set of ICE candidates by checking for the presence of host candidates.
If no host candidates are present, Mode 3 or 4 is in use; this
knowledge can be useful for diagnostic purposes.
Security Considerations
This document describes several potential privacy and security concerns
associated with WebRTC peer-to-peer connections and provides mechanisms
and recommendations for WebRTC implementations to address these concerns.
IANA Considerations
This document has no IANA actions.
References
Normative References
Informative References
WebRTC Security Architecture
Transports for WebRTC
Acknowledgements
Several people provided input into this document, including , , , , ,
, , and
.