wdiff draft-ietf-rtcweb-ip-handling-04.txt draft-ietf-rtcweb-ip-handling-05.txt

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
draft-ietf-rtcweb-ip-handling-04
draft-ietf-rtcweb-ip-handling-05

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/. https://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, August 15, 2018.

This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info)
(https://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.

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
10.
11. References . . . . . . . . . . . . . . . . . . . . . . . . . 7
10.1.
11.1. Normative References . . . . . . . . . . . . . . . . . . 7
10.2.
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",
"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 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
gathered
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
learned.

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.,
192.168.0.2), or have built-in privacy protection (e.g., the
[RFC4941] IPv6 addresses recommended by
[I-D.ietf-rtcweb-transports]).

#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.
192.168.0.2), 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 (0.0.0.0 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
thwarts
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-
organization
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
(0.0.0.0 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
below.

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.

10.

11. References

10.1.

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,
<http://www.rfc-editor.org/info/rfc2119>.

10.2.
<https://www.rfc-editor.org/info/rfc2119>.

11.2. Informative References

[I-D.ietf-rtcweb-return]
Schwartz, B. and J. Uberti, "Recursively Encapsulated TURN
(RETURN)

[I-D.ietf-rtcweb-transports]
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,
<http://www.rfc-editor.org/info/rfc1918>.
<https://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>.
<https://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>.
<https://www.rfc-editor.org/info/rfc1928>.

[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,
<https://www.rfc-editor.org/info/rfc3986>.

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

[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>.
<https://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>.
<https://www.rfc-editor.org/info/rfc7230>.

[RFC8089] Kerwin, M., "The "file" URI Scheme", RFC 8089,
DOI 10.17487/RFC8089, February 2017,
<https://www.rfc-editor.org/info/rfc8089>.

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
IP.

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'.

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,
Facebook
1101 Dexter Ave
Seattle, WA 98033 98109
USA

Email: guoweis@google.com guoweis@facebook.com