draft-ietf-rtcweb-ip-handling-02.txt   draft-ietf-rtcweb-ip-handling-03.txt 
Network Working Group J. Uberti Network Working Group J. Uberti
Internet-Draft G. Shieh Internet-Draft G. Shieh
Intended status: Standards Track Google Intended status: Standards Track Google
Expires: May 4, 2017 October 31, 2016 Expires: July 18, 2017 January 14, 2017
WebRTC IP Address Handling Requirements WebRTC IP Address Handling Requirements
draft-ietf-rtcweb-ip-handling-02 draft-ietf-rtcweb-ip-handling-03
Abstract Abstract
This document provides best practices for how IP addresses should be This document provides information and requirements for how IP
handled by WebRTC applications. addresses should be handled by WebRTC applications.
Status of This Memo Status of This Memo
This Internet-Draft is submitted in full conformance with the This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79. provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute Task Force (IETF). Note that other groups may also distribute
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time. It is inappropriate to use Internet-Drafts as reference time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress." material or to cite them other than as "work in progress."
This Internet-Draft will expire on May 4, 2017. This Internet-Draft will expire on July 18, 2017.
Copyright Notice Copyright Notice
Copyright (c) 2016 IETF Trust and the persons identified as the Copyright (c) 2017 IETF Trust and the persons identified as the
document authors. All rights reserved. document authors. All rights reserved.
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1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Problem Statement . . . . . . . . . . . . . . . . . . . . . . 2 2. Problem Statement . . . . . . . . . . . . . . . . . . . . . . 2
3. Goals . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3 3. Goals . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3
4. Detailed Design . . . . . . . . . . . . . . . . . . . . . . . 4 4. Detailed Design . . . . . . . . . . . . . . . . . . . . . . . 4
5. Application Guidance . . . . . . . . . . . . . . . . . . . . 6 5. Application Guidance . . . . . . . . . . . . . . . . . . . . 6
6. Security Considerations . . . . . . . . . . . . . . . . . . . 6 6. Security Considerations . . . . . . . . . . . . . . . . . . . 6
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 6 7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 6
8. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 6 8. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 6
9. Informative References . . . . . . . . . . . . . . . . . . . 6 9. Informative References . . . . . . . . . . . . . . . . . . . 6
Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 7 Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 8
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 8 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 8
1. Introduction 1. Introduction
As a technology that supports peer-to-peer connections, WebRTC may As a technology that supports peer-to-peer connections, WebRTC may
send data over different network paths than the path used for HTTP send data over different network paths than the path used for HTTP
traffic. This may allow a web application to learn additional traffic. This may allow a web application to learn additional
information about the user, which may be problematic in certain information about the user, which may be problematic in certain
cases. This document summarizes the concerns, and makes cases. This document summarizes the concerns, and makes
recommendations on how best to handle the tradeoff between privacy recommendations on how best to handle the tradeoff between privacy
and media performance. and media performance.
2. Problem Statement 2. Problem Statement
WebRTC enables real-time peer-to-peer communications by enumerating WebRTC enables real-time peer-to-peer communications by enumerating
network interfaces and discovering the best route through the ICE network interfaces and discovering the best route through the ICE
[RFC5245] protocol. During the ICE process, the peers involved in a [RFC5245]protocol. During the ICE process, the peers involved in a
session gather and exchange all the IP addresses they can discover, 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 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 path chosen. The addresses that are gathered usually consist of an
endpoint's private physical/virtual addresses, and its public endpoint's private physical/virtual addresses, and its public
Internet addresses. Internet addresses.
These addresses are exposed upwards to the web application, so that These addresses are exposed upwards to the web application, so that
they can be communicated to the remote endpoint. This allows the they can be communicated to the remote endpoint. This allows the
application to learn more about the local network configuration than application to learn more about the local network configuration than
it would from a typical HTTP scenario, in which the web server would 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 only see a single public Internet address, i.e. the address from
which the HTTP request was sent. which the HTTP request was sent.
The information revealed falls into three categories: The information revealed falls into three categories:
1. If the client is behind a NAT, the client's private IP addresses, 1. If the client is behind a NAT, the client's private IP addresses,
typically [RFC1918] addresses, can be learned. typically [RFC1918]addresses, can be learned.
2. If the client tries to hide its physical location through a VPN, 2. If the client tries to hide its physical location through a VPN,
and the VPN and local OS support routing over multiple and the VPN and local OS support routing over multiple interfaces
interfaces, WebRTC will discover the public address for the VPN (i.e., a "split-tunnel" VPN), WebRTC will discover the public
as well as the ISP public address that the VPN runs over. address for the VPN as well as the ISP public address that the
VPN runs over.
3. If the client is behind a proxy, but direct access to the 3. If the client is behind a proxy (a client-configured "classical
Internet is also supported, WebRTC's STUN [RFC5389] checks will application proxy", as defined in [RFC1919], Section 3), but
bypass the proxy and reveal the public address of the client. 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 Of these three concerns, #2 is the most significant concern, since
for some users, the purpose of using a VPN is for anonymity. for some users, the purpose of using a VPN is for anonymity.
However, different VPN users will have different needs, and some VPN However, different VPN users will have different needs, and some VPN
users (e.g. corporate VPN users) may in fact prefer WebRTC to send users (e.g. corporate VPN users) may in fact prefer WebRTC to send
media traffic directly, i.e. not through the VPN. media traffic directly, i.e., not through the VPN.
#3 is a less common concern, as proxy administrators can control this #3 is a less common concern, as proxy administrators can control this
behavior through local firewall policy if desired, coupled with the behavior through organization firewall policy if desired, coupled
fact that forcing WebRTC traffic through a proxy will have negative with the fact that forcing WebRTC traffic through a proxy will have
effects on both the proxy and on media quality. For situations where negative effects on both the proxy and on media quality. For
this is an important consideration, use of a RETURN proxy, as situations where this is an important consideration, use of a RETURN
described below, can be an effective solution. proxy, as described below, can be an effective solution.
#1 is considered to be the least significant concern, given that the #1 is considered to be the least significant concern, given that the
local address values often contain minimal information (e.g. local address values often contain minimal information (e.g.
192.168.0.2), or have built-in privacy protection (e.g. [RFC4941] 192.168.0.2), or have built-in privacy protection (e.g.
IPv6 addresses). [RFC4941]IPv6 addresses).
Note also that these concerns predate WebRTC; Adobe Flash Player has Note also that these concerns predate WebRTC; Adobe Flash Player has
provided similar functionality since the introduction of RTMFP provided similar functionality since the introduction of RTMFP
[RFC7016] in 2008. [RFC7016]in 2008.
3. Goals 3. Goals
Being peer-to-peer, WebRTC represents a privacy-enabling technology, Being peer-to-peer, WebRTC represents a privacy-enabling technology,
and therefore we want to avoid solutions that disable WebRTC or make and therefore we want to avoid solutions that disable WebRTC or make
it harder to use. This means that WebRTC should be configured by it harder to use. This means that WebRTC should be configured by
default to only reveal the minimum amount of information needed to default to only reveal the minimum amount of information needed to
establish a performant WebRTC session, while providing options to establish a performant WebRTC session, while providing options to
reveal additional information upon user consent, or further limit reveal additional information upon user consent, or further limit
this information if the user has specifically requested this. this information if the user has specifically requested this.
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o Provide a privacy-friendly default behavior which strikes the o Provide a privacy-friendly default behavior which strikes the
right balance between privacy and media performance for most users right balance between privacy and media performance for most users
and use cases. and use cases.
o For users who care more about one versus the other, provide a o For users who care more about one versus the other, provide a
means to customize the experience. means to customize the experience.
4. Detailed Design 4. Detailed Design
The main ideas for the design are the following: The key principles for the design are listed below:
1. By default, WebRTC should follow normal IP routing rules, to the 1. By default, WebRTC should follow normal IP routing rules, to the
extent that this is easy to determine (i.e., not considering extent that this is easy to determine (i.e., not considering
proxies). This can be accomplished by binding local sockets to proxies). This can be accomplished by binding local sockets to
the wildcard addresses (0.0.0.0 for IPv4, :: for IPv6), which 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 allows the OS to route WebRTC traffic the same way as it would
HTTP traffic, and allows only the 'typical' public addresses to HTTP traffic, and allows only the 'typical' public addresses to
be discovered. be discovered.
2. By default, support for direct connections between hosts (i.e., 2. By default, support for direct connections between hosts (i.e.,
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as candidates, even when binding to the wildcard addresses as as candidates, even when binding to the wildcard addresses as
mentioned above. The appropriate addresses here can be mentioned above. The appropriate addresses here can be
discovered by the common trick of binding sockets to the wildcard discovered by the common trick of binding sockets to the wildcard
addresses, connect()ing those sockets to some well-known public addresses, connect()ing those sockets to some well-known public
IP address (one particular example being "8.8.8.8"), and then IP address (one particular example being "8.8.8.8"), and then
reading the bound local addresses via getsockname(). This reading the bound local addresses via getsockname(). This
approach requires no data exchange; it simply provides a approach requires no data exchange; it simply provides a
mechanism for applications to retrieve the desired information mechanism for applications to retrieve the desired information
from the kernel routing table. from the kernel routing table.
3. Gathering all possible candidates SHOULD only be performed when 3. Determining whether a web proxy is in use is a complex process,
some form of 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 key this off getUserMedia consent. The getUserMedia
suggestion takes into account that the user has provided some
consent to the application already; that when doing so the user
typically wants to engage in a conversational session, which
benefits most from an optimal network path, and lastly, the fact
that the underlying issue is complex and difficult to explain,
making explicit consent for enumeration troublesome.
4. Determining whether a web proxy is in use is a complex process,
as the answer can depend on the exact site or address being as the answer can depend on the exact site or address being
contacted. Furthermore, web proxies that support UDP are not contacted. Furthermore, web proxies that support UDP are not
widely deployed today. As a result, when WebRTC is made to go widely deployed today. As a result, when WebRTC is made to go
through a proxy, it typically must use TCP, either ICE-TCP through a proxy, it typically needs to use TCP, either ICE-TCP
[RFC6544] or TURN-over-TCP [RFC5766]. Naturally, this has [RFC6544]or TURN-over-TCP [RFC5766]. Naturally, this has
attendant costs on media quality and also proxy performance. attendant costs on media quality as well as proxy performance,
and should be avoided where possible.
5. RETURN [I-D.ietf-rtcweb-return] is a new proposal for explicit 4. RETURN [I-D.ietf-rtcweb-return]is a proposal for explicit
proxying of WebRTC media traffic. When RETURN proxies are proxying of WebRTC media traffic. When RETURN proxies are
deployed, media and STUN checks will go through the proxy, but deployed, media and STUN checks will go through the proxy, but
without the performance issues associated with sending through a without the performance issues associated with sending through a
typical web proxy. typical web proxy.
Based on these ideas, we define four modes of WebRTC behavior, Based on these ideas, we define four specific modes of WebRTC
reflecting different privacy/media tradeoffs: behavior, reflecting different media quality/privacy tradeoffs:
Mode 1: Enumerate all addresses: WebRTC will bind to all interfaces Mode 1: Enumerate all addresses: WebRTC MUST bind to all interfaces
individually and use them all to attempt communication with individually and use them all to attempt communication with
STUN servers, TURN servers, or peers. This will converge on STUN servers, TURN servers, or peers. This will converge on
the best media path, and is ideal when media performance is the best media path, and is ideal when media performance is
the highest priority, but it discloses the most information. 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.
Mode 2: Default route + the single associated local address: By Mode 2: Default route + associated local addresses: WebRTC MUST
binding solely to the wildcard address, media packets will follow the kernel routing table rules (e.g., by binding
follow the kernel routing table rules, which will typically solely to the wildcard address), which will typically cause
result in the same route as the application's HTTP traffic. media packets to take the same route as the application's
In addition, the associated private address will be HTTP traffic. In addition, any private IPv4 and IPv6
discovered through getsockname, as mentioned above. This addresses associated with the kernel-chosen interface MUST
ensures that direct connections can still be established be discovered through getsockname, as mentioned above, and
even when local media access is not granted, e.g., for data provided to the application. This ensures that direct
channel applications. connections can still be established in this mode.
Mode 3: Default route only: This is the the same as Mode 2, except Mode 3: Default route only: This is the the same as Mode 2, except
that the associated private address is not provided, which that the associated private address MUST NOT be provided.
may cause traffic to hairpin through a NAT, fall back to the This may cause traffic to hairpin through a NAT, fall back
application TURN server, or fail altogether, with resulting to the application TURN server, or fail altogether, with
quality implications. resulting quality implications.
Mode 4: Force proxy: This forces all WebRTC media traffic through a Mode 4: Force proxy: This forces all WebRTC media traffic through a
proxy, if one is configured. If the proxy does not support proxy, if one is configured. If the proxy does not support
UDP (as is the case for all HTTP and most SOCKS [RFC1928] UDP (as is the case for all HTTP and most SOCKS
proxies), or the WebRTC implementation does not support UDP [RFC1928]proxies), or the WebRTC implementation does not
proxying, the use of UDP will be disabled, and TCP will be support UDP proxying, the use of UDP will be disabled, and
used to send and receive media through the proxy. Use of TCP will be used to send and receive media through the
TCP will result in reduced quality, in addition to any proxy. Use of TCP will result in reduced quality, in
performance considerations associated with sending all addition to any performance considerations associated with
WebRTC media through the proxy server. sending all WebRTC media through the proxy server.
We recommend Mode 1 as the default behavior only if the user has Mode 1 MUST only be used when user consent has been provided; this
provided some form of consent, as discussed above, or Mode 2 if not. 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.
Users who prefer Mode 3 or 4 should be able to select a preference or In cases where user consent has not been obtained, Mode 2 SHOULD be
install an extension to force their browser to operate in the used. This allows applications to still achieve direct connections
specified mode. 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 Note that when a RETURN proxy is configured for the interface
associated with the default route, Mode 2 and 3 will cause any associated with the default route, Mode 2 and 3 will cause any
external media traffic to go through the RETURN proxy. This provides external media traffic to go through the RETURN proxy. While the
a way to ensure the proxy is used for external traffic, but without RETURN approach gives the best performance, a similar result can be
the performance issues of forcing all media through said proxy. 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.
5. Application Guidance 5. Application Guidance
The recommendations mentioned in this document may cause certain The recommendations mentioned in this document may cause certain
WebRTC applications to malfunction. In order to be robust in all WebRTC applications to malfunction. In order to be robust in all
scenarios, applications should follow the following guidelines: scenarios, the following guidelines are provided for applications:
o Applications should deploy a TURN server with support for both UDP o Applications SHOULD deploy a TURN server with support for both UDP
and TCP connections to the server. This ensures that connectivity and TCP connections to the server. This ensures that connectivity
can still be established, even when Mode 3 or 4 are in use, can still be established, even when Mode 3 or 4 are in use,
assuming the TURN server can be reached. assuming the TURN server can be reached.
o Applications can detect when they don't have access to the full o Applications SHOULD detect when they don't have access to the full
set of ICE candidates by checking for the presence of host set of ICE candidates by checking for the presence of host
candidates. If no host candidates are present, Mode 3 or 4 above candidates. If no host candidates are present, Mode 3 or 4 above
is in use. is in use; this knowledge can be useful for diagnostic purposes.
o Future versions of browsers may present an indicator to signify 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. 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 Applications MUST only use WebRTC in a fashion that is consistent
that is consistent with user expectations. with user expectations.
6. Security Considerations 6. Security Considerations
This document is entirely devoted to security considerations. This document is entirely devoted to security considerations.
7. IANA Considerations 7. IANA Considerations
This document requires no actions from IANA. This document requires no actions from IANA.
8. Acknowledgements 8. Acknowledgements
Several people provided input into this document, including Harald Several people provided input into this document, including Harald
Alvestrand, Ted Hardie, Matthew Kaufmann, Eric Rescorla, and Adam Alvestrand, Ted Hardie, Matthew Kaufmann, Eric Rescorla, Adam Roach,
Roach. and Martin Thomson.
9. Informative References 9. Informative References
[I-D.ietf-rtcweb-return] [I-D.ietf-rtcweb-return]
Schwartz, B. and J. Uberti, "Recursively Encapsulated TURN Schwartz, B. and J. Uberti, "Recursively Encapsulated TURN
(RETURN) for Connectivity and Privacy in WebRTC", draft- (RETURN) for Connectivity and Privacy in WebRTC", draft-
ietf-rtcweb-return-01 (work in progress), January 2016. ietf-rtcweb-return-01 (work in progress), January 2016.
[RFC1918] Rekhter, Y., Moskowitz, B., Karrenberg, D., de Groot, G., [RFC1918] Rekhter, Y., Moskowitz, B., Karrenberg, D., de Groot, G.,
and E. Lear, "Address Allocation for Private Internets", and E. Lear, "Address Allocation for Private Internets",
BCP 5, RFC 1918, DOI 10.17487/RFC1918, February 1996, BCP 5, RFC 1918, DOI 10.17487/RFC1918, February 1996,
<http://www.rfc-editor.org/info/rfc1918>. <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 [RFC1928] Leech, M., Ganis, M., Lee, Y., Kuris, R., Koblas, D., and
L. Jones, "SOCKS Protocol Version 5", RFC 1928, L. Jones, "SOCKS Protocol Version 5", RFC 1928,
DOI 10.17487/RFC1928, March 1996, DOI 10.17487/RFC1928, March 1996,
<http://www.rfc-editor.org/info/rfc1928>. <http://www.rfc-editor.org/info/rfc1928>.
[RFC4941] Narten, T., Draves, R., and S. Krishnan, "Privacy [RFC4941] Narten, T., Draves, R., and S. Krishnan, "Privacy
Extensions for Stateless Address Autoconfiguration in Extensions for Stateless Address Autoconfiguration in
IPv6", RFC 4941, DOI 10.17487/RFC4941, September 2007, IPv6", RFC 4941, DOI 10.17487/RFC4941, September 2007,
<http://www.rfc-editor.org/info/rfc4941>. <http://www.rfc-editor.org/info/rfc4941>.
skipping to change at page 7, line 48 skipping to change at page 8, line 7
"TCP Candidates with Interactive Connectivity "TCP Candidates with Interactive Connectivity
Establishment (ICE)", RFC 6544, DOI 10.17487/RFC6544, Establishment (ICE)", RFC 6544, DOI 10.17487/RFC6544,
March 2012, <http://www.rfc-editor.org/info/rfc6544>. March 2012, <http://www.rfc-editor.org/info/rfc6544>.
[RFC7016] Thornburgh, M., "Adobe's Secure Real-Time Media Flow [RFC7016] Thornburgh, M., "Adobe's Secure Real-Time Media Flow
Protocol", RFC 7016, DOI 10.17487/RFC7016, November 2013, Protocol", RFC 7016, DOI 10.17487/RFC7016, November 2013,
<http://www.rfc-editor.org/info/rfc7016>. <http://www.rfc-editor.org/info/rfc7016>.
Appendix A. Change log 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: Changes in draft -02:
o Recommendations -> Requirements o Recommendations -> Requirements
o Updated text regarding consent. o Updated text regarding consent.
Changes in draft -01: Changes in draft -01:
o Incorporated feedback from Adam Roach; changes to discussion of o Incorporated feedback from Adam Roach; changes to discussion of
cam/mic permission, as well as use of proxies, and various cam/mic permission, as well as use of proxies, and various
 End of changes. 35 change blocks. 
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