Internet Engineering Task Force (IETF) J. Uberti
Request for Comments: 8828
Category: Standards Track G. Shieh
ISSN: 2070-1721 January 2021
WebRTC IP Address Handling Requirements
This document provides information and requirements for how IP
addresses should be handled by Web Real-Time Communication (WebRTC)
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/rfc8828
Copyright (c) 2021 IETF Trust and the persons identified as the
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) in effect on the date of
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Table of Contents 1.
Problem Statement 4.
Detailed Design 5.1.
Modes and Recommendations 6.
Implementation Guidance 6.1.
Ensuring Normal Routing 6.2.
Determining Associated Local Addresses 7.
Application Guidance 8.
Security Considerations 9.
IANA Considerations 10.
Normative References 10.2.
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) [RFC7230
]. 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.
The key words "MUST
", "MUST NOT
", "SHALL NOT
", "SHOULD NOT
", "NOT RECOMMENDED
" in this document are to be interpreted as described in BCP
] when, and only when, they appear in all
capitals, as shown here.
3. 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) [RFC5766
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: 1.
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. 2.
If the client is behind a Network Address Translator (NAT), the
client's private IP addresses, often [RFC1918
] addresses, can be
If the client is behind a proxy (a client-configured "classical
application proxy", as defined in [RFC1919
], Section 3
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 [RFC7478
], Section 126.96.36.199 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 [RFC1919
] 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 [RFC6146
disclosure of the [RFC4941
] IPv6 addresses recommended by [RFC8835
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
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 [RFC7016
] in 2008.
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
* Finally, provide recommendations for default settings that provide
reasonable performance without also exposing addressing
information in a way that might violate user expectations.
5. Detailed Design
The key principles for our framework are stated below: 1.
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. 2.
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. 3.
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
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.
5.2. 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:
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:
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
proxied. If the proxy does not support UDP (as is the case for
all HTTP and most SOCKS [RFC1928
] 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 [RFC8827
], Section 6.2
implementations can provide a specific mechanism to obtain user
In cases where user consent has not been obtained, Mode 2 SHOULD
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-
6. Implementation Guidance
This section provides guidance to WebRTC implementations on how to
implement the policies described above.
6.1. 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.
6.2. 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 [RFC3986
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 [RFC8089
] 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.
7. 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.
8. 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
9. IANA Considerations
This document has no IANA actions.
10.1. Normative References
] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119
, March 1997,
] Berners-Lee, T., Fielding, R., and L. Masinter, "Uniform
Resource Identifier (URI): Generic Syntax", STD 66, RFC 3986
, DOI 10.17487/RFC3986
, January 2005,
] Rosenberg, J., Mahy, R., Matthews, P., and D. Wing,
"Session Traversal Utilities for NAT (STUN)", RFC 5389
, October 2008,
] Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using
Relays around NAT (TURN): Relay Extensions to Session
Traversal Utilities for NAT (STUN)", RFC 5766
, April 2010,
] Kerwin, M., "The "file" URI Scheme", RFC 8089
, February 2017,
] 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
] Keranen, A., Holmberg, C., and J. Rosenberg, "Interactive
Connectivity Establishment (ICE): A Protocol for Network
Address Translator (NAT) Traversal", RFC 8445
, July 2018,
10.2. Informative References
] Rekhter, Y., Moskowitz, B., Karrenberg, D., de Groot, G.
J., and E. Lear, "Address Allocation for Private
Internets", BCP 5, RFC 1918
, DOI 10.17487/RFC1918
February 1996, <https://www.rfc-editor.org/info/rfc1918
] Chatel, M., "Classical versus Transparent IP Proxies", RFC 1919
, DOI 10.17487/RFC1919
, March 1996,
] Leech, M., Ganis, M., Lee, Y., Kuris, R., Koblas, D., and
L. Jones, "SOCKS Protocol Version 5", RFC 1928
, March 1996,
] Narten, T., Draves, R., and S. Krishnan, "Privacy
Extensions for Stateless Address Autoconfiguration in
IPv6", RFC 4941
, DOI 10.17487/RFC4941
, September 2007,
] Bagnulo, M., Matthews, P., and I. van Beijnum, "Stateful
NAT64: Network Address and Protocol Translation from IPv6
Clients to IPv4 Servers", RFC 6146
, DOI 10.17487/RFC6146
April 2011, <https://www.rfc-editor.org/info/rfc6146
] Thornburgh, M., "Adobe's Secure Real-Time Media Flow
Protocol", RFC 7016
, DOI 10.17487/RFC7016
, November 2013,
] Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer
Protocol (HTTP/1.1): Message Syntax and Routing", RFC 7230
, DOI 10.17487/RFC7230
, June 2014,
] Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-
Time Communication Use Cases and Requirements", RFC 7478
, March 2015,
] Rescorla, E., "WebRTC Security Architecture", RFC 8827
, January 2021,
] Alvestrand, H., "Transports for WebRTC", RFC 8835
, January 2021,
Several people provided input into this document, including Bernard
Aboba, Harald Alvestrand, Youenn Fablet, Ted Hardie, Matthew
Kaufmann, Eric Rescorla, Adam Roach, and Martin Thomson.
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