WebRTC IP Address Handling Requirements
draft-ietf-rtcweb-ip-handling-03
The information below is for an old version of the document.
| Document | Type | Active Internet-Draft (rtcweb WG) | |
|---|---|---|---|
| Authors | Justin Uberti , Guo-wei Shieh | ||
| Last updated | 2017-04-07 (Latest revision 2017-01-14) | ||
| Replaces | draft-shieh-rtcweb-ip-handling | ||
| Stream | Internet Engineering Task Force (IETF) | ||
| Formats | plain text xml htmlized pdfized bibtex | ||
| Reviews |
GENART Last Call review
(of
-11)
Ready with Nits
|
||
| Stream | WG state | WG Document | |
| Document shepherd | Sean Turner | ||
| IESG | IESG state | I-D Exists | |
| Consensus boilerplate | Yes | ||
| Telechat date | (None) | ||
| Responsible AD | (None) | ||
| Send notices to | Sean Turner <sean@sn3rd.com> |
draft-ietf-rtcweb-ip-handling-03
Network Working Group J. Uberti
Internet-Draft G. Shieh
Intended status: Standards Track Google
Expires: July 18, 2017 January 14, 2017
WebRTC IP Address Handling Requirements
draft-ietf-rtcweb-ip-handling-03
Abstract
This document provides information and requirements 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 July 18, 2017.
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
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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 . . . . . . . . . . . . . . . . . . . 6
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
(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
[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 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.
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.
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o For users who care more about one versus the other, provide a
means to customize the experience.
4. 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 (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. 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
[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:
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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
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
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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.
5. 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.
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 MUST 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, Adam Roach,
and Martin Thomson.
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.
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[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>.
[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 -03:
o Clarified when to use which modes.
o Use 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
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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