Network Working Group J. Uberti
Internet-Draft G. Shieh
Intended status: Standards Track Google
Expires: January 4, 2018 July 3, 2017
WebRTC IP Address Handling Requirements
draft-ietf-rtcweb-ip-handling-04
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 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 January 4, 2018.
Copyright Notice
Copyright (c) 2017 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
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.
Uberti & Shieh Expires January 4, 2018 [Page 1]
Internet-Draft WebRTC IP Handling July 2017
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 2
3. Problem Statement . . . . . . . . . . . . . . . . . . . . . . 2
4. Goals . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4
5. Detailed Design . . . . . . . . . . . . . . . . . . . . . . . 4
6. Application Guidance . . . . . . . . . . . . . . . . . . . . 6
7. Security Considerations . . . . . . . . . . . . . . . . . . . 6
8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 6
9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 7
10. References . . . . . . . . . . . . . . . . . . . . . . . . . 7
10.1. Normative References . . . . . . . . . . . . . . . . . . 7
10.2. Informative References . . . . . . . . . . . . . . . . . 7
Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 8
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 9
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 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 gathers and exchanges all the IP addresses it can
discover, using techniques like Session Traversal Utilities for NAT
(STUN) [RFC5389] and Traversal Using Relays around NAT (TURN)
[RFC5766], in order to check the connectivity of each local-address-
remote-address pair and select the best one. The addresses that are
gathered usually consist of an endpoint's private physical/virtual
addresses and its public Internet addresses.
Uberti & Shieh Expires January 4, 2018 [Page 2]
Internet-Draft WebRTC IP Handling July 2017
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 Network Address Translator (NAT), the
client's private IP addresses, typically [RFC1918] addresses, can
be learned.
2. 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 "split-tunnel" VPN),
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 (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 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 organization 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.
Uberti & Shieh Expires January 4, 2018 [Page 3]
Internet-Draft WebRTC IP Handling July 2017
4. 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.
5. Detailed Design
The key principles for the design are listed below:
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, 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 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 needs to use TCP, either ICE-TCP
Uberti & Shieh Expires January 4, 2018 [Page 4]
Internet-Draft WebRTC IP Handling July 2017
[RFC6544] or TURN-over-TCP [RFC5766]. Naturally, this has
attendant costs on media quality as well as proxy performance,
and should be avoided where possible.
4. RETURN [I-D.ietf-rtcweb-return] is a 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 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 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), which will typically cause
media packets to take the same route as the application's
HTTP traffic. In addition, any 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 MUST NOT be provided.
This 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.
Mode 1 MUST only be used when user consent has been provided; this
thwarts the typical drive-by enumeration attacks. The details of
Uberti & Shieh Expires January 4, 2018 [Page 5]
Internet-Draft WebRTC IP Handling July 2017
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., data channel
applications). However, user agents MAY choose a stricter default
policy in certain circumstances.
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. While the
RETURN approach gives the best performance, a similar result can be
achieved for non-RETURN proxies via an organization firewall policy
that only allows external WebRTC traffic to leave through the proxy
(typically, over TCP). This provides a way to ensure the proxy is
used for any external traffic, but avoids the performance issues of
Mode 4, where all media is forced through said proxy, for intra-
organization traffic.
6. 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.
7. Security Considerations
This document is entirely devoted to security considerations.
8. IANA Considerations
This document requires no actions from IANA.
Uberti & Shieh Expires January 4, 2018 [Page 6]
Internet-Draft WebRTC IP Handling July 2017
9. Acknowledgements
Several people provided input into this document, including Bernard
Aboba, Harald Alvestrand, Ted Hardie, Matthew Kaufmann, Eric
Rescorla, Adam Roach, and Martin Thomson.
10. References
10.1. 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,
<http://www.rfc-editor.org/info/rfc2119>.
10.2. 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-02 (work in progress), March 2017.
[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>.
[RFC1919] Chatel, M., "Classical versus Transparent IP Proxies",
RFC 1919, DOI 10.17487/RFC1919, March 1996,
<http://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,
<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>.
Uberti & Shieh Expires January 4, 2018 [Page 7]
Internet-Draft WebRTC IP Handling July 2017
[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>.
[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,
<http://www.rfc-editor.org/info/rfc7230>.
Appendix A. Change log
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'.
Uberti & Shieh Expires January 4, 2018 [Page 8]
Internet-Draft WebRTC IP Handling July 2017
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
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
Uberti & Shieh Expires January 4, 2018 [Page 9]
