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Versions: (draft-shieh-rtcweb-ip-handling) 00 01 02 03 04 05 06 07 08 09

Network Working Group                                          J. Uberti
Internet-Draft                                                    Google
Intended status: Standards Track                                G. Shieh
Expires: December 15, 2018                                      Facebook
                                                           June 13, 2018


                WebRTC IP Address Handling Requirements
                    draft-ietf-rtcweb-ip-handling-09

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 December 15, 2018.

Copyright Notice

   Copyright (c) 2018 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.





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Table of Contents

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

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", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in [RFC2119].

3.  Problem Statement

   In order to establish a peer-to-peer connection, WebRTC
   implementations use Interactive Connectivity Establishment (ICE)
   [RFC5245], which 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



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   pair in order to select the best one.  The addresses that are
   collected usually consist of an endpoint's private physical/virtual
   addresses and its public Internet addresses.

   These addresses are exposed upwards 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 will 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, #1 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.

   #2 is considered to be a less significant concern, given that the
   local address values often contain minimal information (e.g.,
   192.168.0.2), or have built-in privacy protection (e.g., the
   [RFC4941] IPv6 addresses recommended by
   [I-D.ietf-rtcweb-transports]).





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   #3 is the least common concern, 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 RTMFP
   [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, we want to avoid blunt solutions that disable WebRTC or
   make it significantly harder to use.  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.

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), by providing a minimal set of local IP
       addresses to the application for use in the ICE process.  This
       ensures that applications that need true peer-to-peer routing for



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       bandwidth or latency reasons can operate successfully.  However,
       it should be possible to suppress these addresses (with the
       resultant impact on direct connections) if desired.

   3.  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.

   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.  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



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            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 only be used when 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 consent.
   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.

   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, 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.






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6.2.  Determining Host Candidates

   When binding to a wildcard address, some extra work is needed to
   determine a suitable host candidate, 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).
   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
   tell it to 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 host candidate.

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 is entirely devoted to security considerations.





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9.  IANA Considerations

   This document requires no actions from IANA.

10.  Acknowledgements

   Several people provided input into this document, including Bernard
   Aboba, Harald Alvestrand, 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>.

11.2.  Informative References

   [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>.

   [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>.

   [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>.

   [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>.



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   [RFC5245]  Rosenberg, J., "Interactive Connectivity Establishment
              (ICE): A Protocol for Network Address Translator (NAT)
              Traversal for Offer/Answer Protocols", RFC 5245,
              DOI 10.17487/RFC5245, April 2010,
              <https://www.rfc-editor.org/info/rfc5245>.

   [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>.

   [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>.

   [RFC8089]  Kerwin, M., "The "file" URI Scheme", RFC 8089,
              DOI 10.17487/RFC8089, February 2017,
              <https://www.rfc-editor.org/info/rfc8089>.

Appendix A.  Change log

   Changes in draft -09:

   o  Fixed confusing text regarding enterprise TURN servers.

   Changes in draft -08:

   o  Discuss how enterprise TURN servers should be handled.

   Changes in draft -07:

   o  Clarify consent guidance.



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   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:

   o  Recommendations -> Requirements

   o  Updated text regarding consent.

   Changes in draft -01:





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   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.

Authors' Addresses

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

   Email: justin@uberti.name


   Guo-wei Shieh
   Facebook
   1101 Dexter Ave
   Seattle, WA  98109
   USA

   Email: guoweis@facebook.com























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