Internet DRAFT - draft-shieh-rtcweb-ip-handling
draft-shieh-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.
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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.
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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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