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Versions: 00 draft-ietf-rtcweb-ip-handling

Network Working Group                                           G. Shieh
Internet-Draft                                                 J. Uberti
Intended status: Standards Track                                  Google
Expires: April 21, 2016                                 October 19, 2015


               WebRTC IP Address Handling Recommendations
                   draft-shieh-rtcweb-ip-handling-00

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
   Task Force (IETF).  Note that other groups may also distribute
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://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 April 21, 2016.

Copyright Notice

   Copyright (c) 2015 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
   (http://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.  Problem Statement . . . . . . . . . . . . . . . . . . . . . .   2
   3.  Goals . . . . . . . . . . . . . . . . . . . . . . . . . . . .   3
   4.  Detailed Design . . . . . . . . . . . . . . . . . . . . . . .   4
   5.  Application Guidance  . . . . . . . . . . . . . . . . . . . .   5
   6.  Security Considerations . . . . . . . . . . . . . . . . . . .   6
   7.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .   6
   8.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .   6
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .   6

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
   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 RFC 1918 addresses, can be learned.

   (2)  If the client tries to hide its physical location through a VPN,
        and the VPN and local OS supports routing over multiple
        interfaces, WebRTC will discover the public address associated



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        with both the VPN as well as the ISP public address over 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 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.

   #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.  RFC 4941
   IPv6 addresses).

   Note also that these concerns predate WebRTC; Adobe Flash Player has
   provided similar functionality since the introduction of RTMFP 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 means
      to customize the experience.







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4.  Detailed Design

   The main ideas for the design are the following:

   o  By default, WebRTC should follow the route for HTTP traffic, when
      this is easy to determine (i.e. not considering proxies).  This is
      accomplished by binding local sockets to the "any" addresses
      (0.0.0.0 for IPv4, :: for IPv6), which allows the OS to route
      WebRTC traffic the same way as normal HTTP traffic, and allows
      only the 'typical' public addresses to be discovered.

   o  By default, support for host-host connections should be
      maintained.  Even when binding to "any" addresses, the local IPv4
      and IPv6 addresses of the interface used for outgoing STUN traffic
      should still be surfaced as candidates; this is necessary for
      certain peer-to-peer data channel apps to function correctly.  The
      appropriate addresses here can be discovered by binding sockets to
      the "any" addresses, connect()ing those sockets to a public
      destination (e.g. "8.8.8.8"), and then reading the bound local
      addresses via getsockname().

   o  WebRTC incorporates an explicit permission grant for access to
      local audio and video, which are typically much more sensitive
      than the aforementioned IP address information.  If the user has
      consented to media access, this should also allow WebRTC to gather
      all possible candidates and determine the absolute best route for
      media traffic.

   o  Determining whether a proxy is in use is a complex process, as the
      answer can depend on the exact site or address being contacted.
      Furthermore, proxies that support UDP are not widely deployed
      today.  Therefore, the only way to ensure that WebRTC traffic
      traverses a proxy is to force WebRTC to use ICE-TCP or TURN-over-
      TCP, and always try to make the TCP connection through the proxy,
      if one exists.  Naturally, this will have attendant costs on media
      quality and also proxy performance.

   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 ping STUN 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.




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   Mode 2  Default route + the single associated local address: By
           binding solely to the "any" address, media packets will flow
           through the same route as normal HTTP traffic.  In addition,
           the associated private address is 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 NAT or fall back to the
           application TURN server, with resulting quality implications.

   Mode 4  Force TCP and proxy: This is disables any use of UDP and
           forces use of TCP to connect to the TURN server or peer.  If
           a proxy server is configured, the TCP traffic will be sent
           through the proxy, with resulting quality implications.

   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.

   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.  For example, Chrome users can install the WebRTC
   Network Limiter extension for this configuration.

5.  Application Guidance

   The recommendations mentioned in this document may cause breakage to
   certain WebRTC applications.  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 Mode 4 are in use.

   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.






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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, and Eric Rescorla.

Authors' Addresses

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

   Email: guoweis@google.com


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

   Email: justin@uberti.name




















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