Network Working Group                                          J. Uberti
Internet-Draft                                                  G. Shieh                                                    Google
Intended status: Standards Track                                  Google                                G. Shieh
Expires: January 4, August 15, 2018                                        Facebook
                                                       February 11, 2018                                    July 3, 2017

                WebRTC IP Address Handling Requirements


   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

   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 4, August 15, 2018.

Copyright Notice

   Copyright (c) 2017 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
   ( 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
   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.  Application  Implementation Guidance . . . . . . . . . . . . . . . . . . . .   6
   7.  Application Guidance  . . . . . . . . . . . . . . . . . . . .   7
   8.  Security Considerations . . . . . . . . . . . . . . . . . . .   6
   8.   7
   9.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .   6
   9.   7
   10. Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .   7
   11. References  . . . . . . . . . . . . . . . . . . . . . . . . .   7
     11.1.  Normative References . . . . . . . . . . . . . . . . . .   7
     11.2.  Informative References . . . . . . . . . . . . . . . . .   7
   Appendix A.  Change log . . . . . . . . . . . . . . . . . . . . .   8   9
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .   9  10

1.  Introduction

   One of WebRTC's key features is its support of peer-to-peer
   connections.  However, when establishing such a connection, which
   involves connectivity tests using 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",
   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 gathers and exchanges all the attempts to discover multiple IP addresses it can
   discover, using
   techniques like such as Session Traversal Utilities for NAT (STUN)
   [RFC5389] and Traversal Using Relays around NAT (TURN) [RFC5766], in order to check and
   then checks the connectivity of each local-address-
   remote-address local-address-remote-address
   pair and 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. 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. 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 Network Address Translator (NAT), the
       client's private multihomed, additional public IP addresses, typically [RFC1918] addresses, addresses for
       the client can be learned.

   2.  If  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 (i.e., a (a "split-tunnel" VPN), WebRTC will discover not only
       the public address for the VPN as well as VPN, but also the ISP public address that
       over which the VPN runs over. is running.

   2.  If the client is behind a Network Address Translator (NAT), the
       client's private IP addresses, often [RFC1918] addresses, can be

   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 also supported, WebRTC's STUN
       checks will bypass the proxy and reveal the public IP address of
       the client.

   Of these three concerns, #2 #1 is the most significant concern, since 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.
   (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.,, or have built-in privacy protection (e.g., the
   [RFC4941] IPv6 addresses recommended by

   #3 is the least common concern, as proxy administrators can already
   control this behavior through organization organizational firewall policy if desired, coupled
   with the fact that policy, and
   generally, forcing WebRTC traffic through a proxy server 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., or have built-in privacy protection (e.g.  [RFC4941]
   IPv6 addresses).

   Note also

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

4.  Goals

   Being peer-to-peer, WebRTC represents

   WebRTC's support of secure peer-to-peer connections facilitates
   deployment of decentralized systems, which can have privacy benefits.
   As a privacy-enabling technology,
   and therefore result, we want to avoid blunt solutions that disable WebRTC or
   make it significantly harder to use.  This means that WebRTC should be configured by
   default to only reveal the minimum amount of information needed to
   establish document takes a performant WebRTC session, while providing options to
   reveal additional information upon user consent, or further limit
   this information if more
   nuanced approach, with the user has specifically requested this.
   Specifically, WebRTC should: following goals:

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

   o  For users who care more about one versus the other,  Finally, provide recommendations for default settings that provide
      reasonable performance without also exposing addressing
      information in a
      means to customize the experience. way that might violate user expectations.

5.  Detailed Design

5.1.  Principles

   The key principles for the design our framework are listed stated below:

   1.  By default, WebRTC traffic should follow normal typical 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 ( for IPv4, :: for IPv6), which
       allows the OS to route routing,
       i.e., WebRTC traffic should use the same way as it would interface used for HTTP traffic,
       and allows only the system's 'typical' public addresses should be
       visible to the application.  However, in the interest of optimal
       media quality, it should be discovered. possible to enable WebRTC to make use
       of all network interfaces to determine the ideal route.

   2.  By default, support for WebRTC should be able to negotiate direct peer-to-
       peer connections between hosts endpoints (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 server), by the common trick providing a minimal set of binding sockets to the wildcard
       addresses, connect()ing those sockets to some well-known public
       IP address, 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.  Determining whether a web proxy is in IP
       addresses to the application for use is a complex process,
       as in the answer ICE process.  This
       ensures that applications that need true peer-to-peer routing for
       bandwidth or latency reasons can depend on operate successfully.  However,
       it should be possible to suppress these addresses (with the exact site or address being
       contacted.  Furthermore, web proxies that support UDP are not
       widely deployed today.  As a result, when
       resultant impact on direct connections) if desired.

   3.  By default, WebRTC is made to go traffic should not be sent through a proxy, it typically needs proxy
       servers, due to use TCP, either ICE-TCP

       [RFC6544] or TURN-over-TCP [RFC5766].  Naturally, this has
       attendant costs on the media quality problems associated with
       sending WebRTC traffic over TCP, which is almost always used when
       communicating with proxies, as well as proxy performance,
       and performance issues
       that may result from proxying WebRTC's long-lived, high-bandwidth
       connections.  However, it should be avoided where possible.

   4.  RETURN [I-D.ietf-rtcweb-return] is a proposal for explicit
       proxying of possible to force 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 to
       send its traffic through a
       typical web proxy. 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 bind to all interfaces
            individually and use them 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 (e.g., by binding
            solely to the wildcard address), rules, which will typically
            cause media packets to take the same route as the
            application's HTTP traffic.  In addition, any the private IPv4
            and IPv6 addresses associated with the kernel-chosen
            interface MUST be discovered through getsockname, as mentioned above, 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 address addressses MUST NOT be provided. 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 the an
            application TURN server, or fail altogether, with resulting
            quality implications.

   Mode 4:  Force proxy: This forces is the same as Mode 3, but all WebRTC
            media traffic is forced 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 media quality, in addition to any performance
            considerations associated with sending all WebRTC media
            through the proxy server.

   The recommended defaults are as follows:

   Mode 1 MUST only be used when user consent has been provided; this
   allows trusted WebRTC applications to achieve optimal network
   performance, but significanly limites the typical drive-by enumeration attacks. network information exposed
   to arbitrary web pages.  The details of this consent are left to the
   implementation; one potential mechanism is to tie this consent to
   getUserMedia consent.

   In cases where user consent has not been obtained, Mode 2 SHOULD be
   used.  This allows applications to still achieve direct connections
   in many cases, even without consent (e.g., streaming or data channel
   applications).  However, user agents implementations MAY choose a stricter
   default policy in certain circumstances.

   Note that when a RETURN proxy is configured these defaults can still be used even for the interface
   associated with the default route, Mode 2 and 3 will cause any organizations
   that want all external media WebRTC traffic to go through the RETURN proxy.  While the
   RETURN approach gives the best performance, traverse a similar result can be
   achieved for non-RETURN proxies via proxy, simply by
   setting an organization organizational firewall policy that only allows external WebRTC traffic
   to only leave through the proxy
   (typically, over TCP). the proxy.  This provides a way to ensure the
   proxy is used for any external traffic, but avoids the performance
   issues of Mode 4, where 4 (where all media is forced through said proxy, proxy) for intra-
   intra-organization traffic.

6.  Implementation Guidance

   This section provides guidance to WebRTC implementations on how to
   implement the policies described above.

   When trying to follow typical IP routing, the simplest approach is to
   bind the sockets used for p2p connections to the wildcard addresses
   ( 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 be determined as mentioned

   In order to discover the correct local IP addresses, implementations
   can use the common trick of binding sockets to the wildcard
   addresses, connect()ing those sockets to the IPv4/IPv6 addresses of
   the web application (obtained by resolving the host component of its
   URI [RFC3986]) and then reading the bound local addresses via
   getsockname().  This requires no data exchange; it simply provides a
   mechanism for applications to retrieve the desired information from
   the kernel routing table.

   Use of the web application IPs ensures the right local IPs are
   selected, regardless of where the application is hosted (e.g., on an
   intranet).  If the client is behind a proxy and cannot resolve the
   IPs via DNS, the IPv4/v6 addresses of the proxy can be used instead.
   If the web application was loaded from a file:// URI [RFC8089], the
   implementation can fall back to a well-known DNS name or IP 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 is entirely devoted to security considerations.


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,


11.2.  Informative References

              Schwartz, B. and J. Uberti, "Recursively Encapsulated TURN

              Alvestrand, H., "Transports for Connectivity and Privacy in WebRTC", draft-
              ietf-rtcweb-return-02 draft-ietf-
              rtcweb-transports-17 (work in progress), March 2017. 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,

   [RFC1919]  Chatel, M., "Classical versus Transparent IP Proxies",
              RFC 1919, DOI 10.17487/RFC1919, March 1996,

   [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,

   [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,

   [RFC4941]  Narten, T., Draves, R., and S. Krishnan, "Privacy
              Extensions for Stateless Address Autoconfiguration in
              IPv6", RFC 4941, DOI 10.17487/RFC4941, September 2007,

   [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,

   [RFC5389]  Rosenberg, J., Mahy, R., Matthews, P., and D. Wing,
              "Session Traversal Utilities for NAT (STUN)", RFC 5389,
              DOI 10.17487/RFC5389, October 2008,

   [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,

   [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, <>.

   [RFC7016]  Thornburgh, M., "Adobe's Secure Real-Time Media Flow
              Protocol", RFC 7016, DOI 10.17487/RFC7016, November 2013,

   [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,

   [RFC8089]  Kerwin, M., "The "file" URI Scheme", RFC 8089,
              DOI 10.17487/RFC8089, February 2017,

Appendix A.  Change log

   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

   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 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:

   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
   747 6th St S
   Kirkland, WA  98033


   Guo-wei Shieh
   747 6th St S
   1101 Dexter Ave
   Seattle, WA  98033  98109