WebRTC IP Address Handling Requirements
RFC 8828


Internet Engineering Task Force (IETF) J. Uberti
Request for Comments: 8828 Google
Category: Standards Track G. Shieh
ISSN: 2070-1721 January 2021
 WebRTC IP Address Handling Requirements
Abstract
 This document provides information and requirements for how IP
 addresses should be handled by Web Real-Time Communication (WebRTC)
 implementations.
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 Notice
 Copyright (c) 2021 IETF Trust and the persons identified as the
 document authors. All rights reserved.
 This document is subject to BCP 78 and the IETF Trust's Legal
 Provisions Relating to IETF Documents
 (https://trustee.ietf.org/license-info) in effect on the date of
 publication of this document. Please review these documents
 carefully, as they describe your rights and restrictions with respect
 to this document. Code Components extracted from this document must
 include Simplified BSD License text as described in Section 4.e of
 the Trust Legal Provisions and are provided without warranty as
 described in the Simplified BSD License.
Table of Contents
 1. Introduction
 2. Terminology
 3. Problem Statement
 4. Goals
 5. Detailed Design
 5.1. Principles
 5.2. 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. References
 10.1. Normative References
 10.2. Informative References
 Acknowledgements
 Authors' Addresses
1. 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) [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.
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. Problem Statement
 In order to establish a peer-to-peer connection, WebRTC
 implementations use Interactive Connectivity Establishment (ICE)
 [RFC8445]. ICE attempts to discover multiple IP addresses using
 techniques such as Session Traversal Utilities for NAT (STUN)
 [RFC5389] and Traversal Using Relays around NAT (TURN) [RFC5766] 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:
 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
 learned.
 3. If the client is behind a proxy (a client-configured "classical
 application proxy", as defined in [RFC1919], Section 3), 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 [RFC7478], Section 2.3.5.1 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]. While
 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
 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 [RFC7016] in 2008.
4. 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.
5. Detailed Design
5.1. Principles
 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
 connections.
 4. 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:
 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 [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. 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.
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
 these concerns.
9. IANA Considerations
 This document has no IANA actions.
10. References
10.1. 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>.
 [RFC3986] Berners-Lee, T., Fielding, R., and L. Masinter, "Uniform
 Resource Identifier (URI): Generic Syntax", STD 66,
 RFC 3986, DOI 10.17487/RFC3986, January 2005,
 <https://www.rfc-editor.org/info/rfc3986>.
 [RFC5389] Rosenberg, J., Mahy, R., Matthews, P., and D. Wing,
 "Session Traversal Utilities for NAT (STUN)", RFC 5389,
 DOI 10.17487/RFC5389, October 2008,
 <https://www.rfc-editor.org/info/rfc5389>.
 [RFC5766] Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using
 Relays around NAT (TURN): Relay Extensions to Session
 Traversal Utilities for NAT (STUN)", RFC 5766,
 DOI 10.17487/RFC5766, April 2010,
 <https://www.rfc-editor.org/info/rfc5766>.
 [RFC8089] Kerwin, M., "The "file" URI Scheme", RFC 8089,
 DOI 10.17487/RFC8089, February 2017,
 <https://www.rfc-editor.org/info/rfc8089>.
 [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>.
 [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>.
10.2. Informative References
 [RFC1918] 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>.
 [RFC1919] Chatel, M., "Classical versus Transparent IP Proxies",
 RFC 1919, DOI 10.17487/RFC1919, March 1996,
 <https://www.rfc-editor.org/info/rfc1919>.
 [RFC1928] Leech, M., Ganis, M., Lee, Y., Kuris, R., Koblas, D., and
 L. Jones, "SOCKS Protocol Version 5", RFC 1928,
 DOI 10.17487/RFC1928, March 1996,
 <https://www.rfc-editor.org/info/rfc1928>.
 [RFC4941] Narten, T., Draves, R., and S. Krishnan, "Privacy
 Extensions for Stateless Address Autoconfiguration in
 IPv6", RFC 4941, DOI 10.17487/RFC4941, September 2007,
 <https://www.rfc-editor.org/info/rfc4941>.
 [RFC6146] 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>.
 [RFC7016] Thornburgh, M., "Adobe's Secure Real-Time Media Flow
 Protocol", RFC 7016, DOI 10.17487/RFC7016, November 2013,
 <https://www.rfc-editor.org/info/rfc7016>.
 [RFC7230] 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,
 <https://www.rfc-editor.org/info/rfc7230>.
 [RFC7478] Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-
 Time Communication Use Cases and Requirements", RFC 7478,
 DOI 10.17487/RFC7478, March 2015,
 <https://www.rfc-editor.org/info/rfc7478>.
 [RFC8827] Rescorla, E., "WebRTC Security Architecture", RFC 8827,
 DOI 10.17487/RFC8827, January 2021,
 <https://www.rfc-editor.org/info/rfc8827>.
 [RFC8835] Alvestrand, H., "Transports for WebRTC", RFC 8835,
 DOI 10.17487/RFC8835, January 2021,
 <https://www.rfc-editor.org/info/rfc8835>.
Acknowledgements
 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.
Authors' Addresses
 Justin Uberti
 Google
 747 6th St S
 Kirkland, WA 98033
 United States of America
 Email: justin@uberti.name
 Guo-wei Shieh
 333 Elliott Ave W #500
 Seattle, WA 98119
 United States of America
 Email: guoweis@gmail.com