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Versions: (draft-shieh-rtcweb-ip-handling) 00 01 02 03 04

Network Working Group                                          J. Uberti
Internet-Draft                                                  G. Shieh
Intended status: Standards Track                                  Google
Expires: September 21, 2016                               March 20, 2016


               WebRTC IP Address Handling Recommendations
                    draft-ietf-rtcweb-ip-handling-01

Abstract

   This document provides best practices for how IP addresses should be
   handled by WebRTC applications.

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
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   This Internet-Draft will expire on September 21, 2016.

Copyright Notice

   Copyright (c) 2016 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
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   described in the Simplified BSD License.






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Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   2
   2.  Problem Statement . . . . . . . . . . . . . . . . . . . . . .   2
   3.  Goals . . . . . . . . . . . . . . . . . . . . . . . . . . . .   3
   4.  Detailed Design . . . . . . . . . . . . . . . . . . . . . . .   4
   5.  Application Guidance  . . . . . . . . . . . . . . . . . . . .   6
   6.  Security Considerations . . . . . . . . . . . . . . . . . . .   6
   7.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .   6
   8.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .   6
   9.  Informative References  . . . . . . . . . . . . . . . . . . .   7
   Appendix A.  Change log . . . . . . . . . . . . . . . . . . . . .   8
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .   8

1.  Introduction

   As a technology that supports peer-to-peer connections, WebRTC may
   send data over different network paths than the path used for HTTP
   traffic.  This may allow a web application to learn additional
   information about the user, which may be problematic in certain
   cases.  This document summarizes the concerns, and makes
   recommendations on how best to handle the tradeoff between privacy
   and media performance.

2.  Problem Statement

   WebRTC enables real-time peer-to-peer communications by enumerating
   network interfaces and discovering the best route through the ICE
   [RFC5245] protocol.  During the ICE process, the peers involved in a
   session gather and exchange all the IP addresses they can discover,
   so that the connectivity of each IP pair can be checked, and the best
   path chosen.  The addresses that are gathered 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.  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 behind a NAT, the client's private IP addresses,
       typically [RFC1918] addresses, can be learned.





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   2.  If the client tries to hide its physical location through a VPN,
       and the VPN and local OS support routing over multiple
       interfaces, WebRTC will discover the public address for the VPN
       as well as the ISP public address that the VPN runs over.

   3.  If the client is behind a proxy, but direct access to the
       Internet is also supported, WebRTC's STUN [RFC5389] checks will
       bypass the proxy and reveal the public address of the client.

   Of these three concerns, #2 is the most significant concern, since
   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.

   #3 is a less common concern, as proxy administrators can control this
   behavior through local firewall policy if desired, coupled with the
   fact that forcing WebRTC traffic through a proxy will have negative
   effects on both the proxy and on media quality.  For situations where
   this is an important consideration, use of a RETURN proxy, as
   described below, can be an effective solution.

   #1 is considered to be the least 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.  [RFC4941]
   IPv6 addresses).

   Note also that these concerns predate WebRTC; Adobe Flash Player has
   provided similar functionality since the introduction of RTMFP
   [RFC7016] in 2008.

3.  Goals

   Being peer-to-peer, WebRTC represents a privacy-enabling technology,
   and therefore we want to avoid solutions that disable WebRTC or make
   it harder to use.  This means that WebRTC should be configured by
   default to only reveal the minimum amount of information needed to
   establish a performant WebRTC session, while providing options to
   reveal additional information upon user consent, or further limit
   this information if the user has specifically requested this.
   Specifically, WebRTC should:

   o  Provide a privacy-friendly default behavior which strikes the
      right balance between privacy and media performance for most users
      and use cases.

   o  For users who care more about one versus the other, provide a
      means to customize the experience.



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4.  Detailed Design

   The main ideas for the design are the following:

   1.  By default, WebRTC should follow normal IP routing rules, to the
       extent that this is easy to determine (i.e., not considering
       proxies).  This can be accomplished by binding local sockets 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, and allows only the 'typical' public addresses to
       be discovered.

   2.  By default, support for direct connections between hosts (i.e.,
       without traversing a NAT or relay server) should be maintained.
       To accomplish this, the local IPv4 and IPv6 addresses of the
       interface used for outgoing STUN traffic should still be surfaced
       as candidates, even when binding to the wildcard addresses as
       mentioned above.  The appropriate addresses here can be
       discovered by the common trick of binding sockets to the wildcard
       addresses, connect()ing those sockets to some well-known public
       IP address (one particular example being "8.8.8.8"), and then
       reading the bound local addresses via getsockname().  This
       approach requires no data exchange; it simply provides a
       mechanism for applications to retrieve the desired information
       from the kernel routing table.

   3.  When used with audio and video devices, WebRTC requires explicit
       user permission to access those devices.  We propose that this
       permission grant be expanded to include consent to allow WebRTC
       to access all IP addresses associated with the user agent, for
       the purpose of finding the absolute best route for media traffic.
       Combining these permission grants, rather than having the user
       grant permission individually, is a considered balance; this
       balance takes into account that the user has placed enough trust
       into the application to allow it to access their devices, that
       when doing so the user typically wants to engage in a
       conversational session, which benefits most from an optimal
       network path, and lastly, the fact that the underlying issue is
       complex, and difficult to explain meaningfully to the user.

   4.  Determining whether a web proxy is in use is a complex process,
       as the answer can depend on the exact site or address being
       contacted.  Furthermore, web proxies that support UDP are not
       widely deployed today.  As a result, when WebRTC is made to go
       through a proxy, it typically must use TCP, either ICE-TCP
       [RFC6544] or TURN-over-TCP [RFC5766].  Naturally, this has
       attendant costs on media quality and also proxy performance.




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   5.  RETURN [I-D.ietf-rtcweb-return] is a new proposal for explicit
       proxying of WebRTC media traffic.  When RETURN proxies are
       deployed, media and STUN checks will go through the proxy, but
       without the performance issues associated with sending through a
       typical web proxy.

   Based on these ideas, we define four modes of WebRTC behavior,
   reflecting different privacy/media tradeoffs:

   Mode 1:  Enumerate all addresses: WebRTC will bind to all interfaces
            individually and use them all 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.
            As such, this should only be performed when the user has
            explicitly given consent for local media access, as
            indicated in design idea #3 above.

   Mode 2:  Default route + the single associated local address: By
            binding solely to the wildcard address, media packets will
            follow the kernel routing table rules, which will typically
            result in the same route as the application's HTTP traffic.
            In addition, the associated private address will be
            discovered through getsockname, as mentioned above.  This
            ensures that direct connections can still be established
            even when local media access is not granted, e.g., for data
            channel applications.

   Mode 3:  Default route only: This is the the same as Mode 2, except
            that the associated private address is not provided, which
            may cause traffic to hairpin through a NAT, fall back to the
            application TURN server, or fail altogether, with resulting
            quality implications.

   Mode 4:  Force proxy: This forces all WebRTC media traffic through a
            proxy, if one is configured.  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 quality, in addition to any
            performance considerations associated with sending all
            WebRTC media through the proxy server.

   We recommend Mode 1 as the default behavior only if cam/mic
   permission has been granted, or Mode 2 if this is not the case.





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   Users who prefer Mode 3 or 4 should be able to select a preference or
   install an extension to force their browser to operate in the
   specified mode.

   Note that when a RETURN proxy is configured for the interface
   associated with the default route, Mode 2 and 3 will cause any
   external media traffic to go through the RETURN proxy.  This provides
   a way to ensure the proxy is used for external traffic, but without
   the performance issues of forcing all media through said proxy.

5.  Application Guidance

   The recommendations mentioned in this document may cause certain
   WebRTC applications to malfunction.  In order to be robust in all
   scenarios, applications should follow the following guidelines:

   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 can 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.

   o  Future versions of browsers may present an indicator to signify
      that the page is using WebRTC to set up a peer-to-peer connection.
      Applications should be careful to only use WebRTC in a fashion
      that is consistent with user expectations.

6.  Security Considerations

   This document is entirely devoted to security considerations.

7.  IANA Considerations

   This document requires no actions from IANA.

8.  Acknowledgements

   Several people provided input into this document, including Harald
   Alvestrand, Ted Hardie, Matthew Kaufmann, Eric Rescorla, and Adam
   Roach.







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9.  Informative References

   [I-D.ietf-rtcweb-return]
              Schwartz, B. and J. Uberti, "Recursively Encapsulated TURN
              (RETURN) for Connectivity and Privacy in WebRTC", draft-
              ietf-rtcweb-return-01 (work in progress), January 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,
              <http://www.rfc-editor.org/info/rfc1918>.

   [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,
              <http://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,
              <http://www.rfc-editor.org/info/rfc4941>.

   [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,
              <http://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,
              <http://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,
              <http://www.rfc-editor.org/info/rfc5766>.

   [RFC6544]  Rosenberg, J., Keranen, A., Lowekamp, B., and A. Roach,
              "TCP Candidates with Interactive Connectivity
              Establishment (ICE)", RFC 6544, DOI 10.17487/RFC6544,
              March 2012, <http://www.rfc-editor.org/info/rfc6544>.

   [RFC7016]  Thornburgh, M., "Adobe's Secure Real-Time Media Flow
              Protocol", RFC 7016, DOI 10.17487/RFC7016, November 2013,
              <http://www.rfc-editor.org/info/rfc7016>.




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Appendix A.  Change log

   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.

Authors' Addresses

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

   Email: justin@uberti.name


   Guo-wei Shieh
   Google
   747 6th St S
   Kirkland, WA  98033
   USA

   Email: guoweis@google.com



















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