



Network Working Group                                          J. Uberti
Internet-Draft                                                    Google
Intended status: Standards Track                            July 2, 2019
Expires: January 3, 2020


                WebRTC IP Address Handling Requirements
                    draft-ietf-rtcweb-ip-handling-12

Abstract

   This document provides information and requirements for how IP
   addresses should be handled by WebRTC implementations.

Status of This Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at https://datatracker.ietf.org/drafts/current/.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on January 3, 2020.

Copyright Notice

   Copyright (c) 2019 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
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   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.






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Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   2
   2.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . .   2
   3.  Problem Statement . . . . . . . . . . . . . . . . . . . . . .   2
   4.  Goals . . . . . . . . . . . . . . . . . . . . . . . . . . . .   4
   5.  Detailed Design . . . . . . . . . . . . . . . . . . . . . . .   5
     5.1.  Principles  . . . . . . . . . . . . . . . . . . . . . . .   5
     5.2.  Modes and Recommendations . . . . . . . . . . . . . . . .   5
   6.  Implementation Guidance . . . . . . . . . . . . . . . . . . .   7
     6.1.  Ensuring Normal Routing . . . . . . . . . . . . . . . . .   7
     6.2.  Determining Associated Local Addresses  . . . . . . . . .   7
   7.  Application Guidance  . . . . . . . . . . . . . . . . . . . .   8
   8.  Security Considerations . . . . . . . . . . . . . . . . . . .   8
   9.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .   8
   10. Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .   8
   11. References  . . . . . . . . . . . . . . . . . . . . . . . . .   8
     11.1.  Normative References . . . . . . . . . . . . . . . . . .   8
     11.2.  Informative References . . . . . . . . . . . . . . . . .   9
   Appendix A.  Change log . . . . . . . . . . . . . . . . . . . . .  10
   Author's Address  . . . . . . . . . . . . . . . . . . . . . . . .  12

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 tradeoff 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], which attempts to discover multiple IP addresses using
   techniques such as Session Traversal Utilities for NAT (STUN)



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   [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 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



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   [I-D.ietf-rtcweb-transports] 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 on 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:

   o  Provide a framework for understanding the problem so that controls
      might be provided to make different tradeoffs regarding
      performance and privacy concerns with WebRTC.

   o  Using that framework, define settings that enable peer-to-peer
      communications, each with a different balance between performance
      and privacy.

   o  Finally, provide recommendations for default settings that provide
      reasonable performance without also exposing addressing
      information in a way that might violate user expectations.






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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 tradeoffs:

   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.





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   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 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 [I-D.ietf-rtcweb-security-arch], 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 tradeoff 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.




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   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



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   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:

   o  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 are in use,
      assuming the TURN server can be reached.

   o  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 above
      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 requires no actions from IANA.

10.  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.

11.  References

11.1.  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>.




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   [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>.

11.2.  Informative References

   [I-D.ietf-rtcweb-security-arch]
              Rescorla, E., "WebRTC Security Architecture", draft-ietf-
              rtcweb-security-arch-18 (work in progress), February 2019.

   [I-D.ietf-rtcweb-transports]
              Alvestrand, H., "Transports for WebRTC", draft-ietf-
              rtcweb-transports-17 (work in progress), October 2016.

   [RFC1918]  Rekhter, Y., Moskowitz, B., Karrenberg, D., de Groot, G.,
              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>.



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   [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>.

Appendix A.  Change log

   Changes in draft -12:

   o  Editorial updates from IETF LC review.

   Changes in draft -11:

   o  Editorial updates from AD review.

   Changes in draft -10:

   o  Incorporate feedback from IETF 102 on the problem space.

   o  Note that future versions of the document may define new modes.

   Changes in draft -09:

   o  Fixed confusing text regarding enterprise TURN servers.



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   Changes in draft -08:

   o  Discuss how enterprise TURN servers should be handled.

   Changes in draft -07:

   o  Clarify consent guidance.

   Changes in draft -06:

   o  Clarify recommendations.

   o  Split implementation guidance into two sections.

   Changes in draft -05:

   o  Separated framework definition from implementation techniques.

   o  Removed RETURN references.

   o  Use origin when determining local IPs, rather than a well-known
      IP.

   Changes in draft -04:

   o  Rewording and cleanup in abstract, intro, and problem statement.

   o  Added 2119 boilerplate.

   o  Fixed weird reference spacing.

   o  Expanded acronyms on first use.

   o  Removed 8.8.8.8 mention.

   o  Removed mention of future browser considerations.

   Changes in draft -03:

   o  Clarified when to use which modes.

   o  Added 2119 qualifiers to make normative statements.

   o  Defined 'proxy'.

   o  Mentioned split tunnels in problem statement.

   Changes in draft -02:



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   o  Recommendations -> Requirements

   o  Updated text regarding consent.

   Changes in draft -01:

   o  Incorporated feedback from Adam Roach; changes to discussion of
      cam/mic permission, as well as use of proxies, and various
      editorial changes.

   o  Added several more references.

   Changes in draft -00:

   o  Published as WG draft.

Author's Address

   Justin Uberti
   Google
   747 6th St S
   Kirkland, WA  98033
   USA

   Email: justin@uberti.name


























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