draft-ietf-rtcweb-ip-handling-12.txt   rfc8828.txt 
Network Working Group J. Uberti Internet Engineering Task Force (IETF) J. Uberti
Internet-Draft Google Request for Comments: 8828 Google
Intended status: Standards Track July 2, 2019 Category: Standards Track G. Shieh
Expires: January 3, 2020 ISSN: 2070-1721 January 2021
WebRTC IP Address Handling Requirements WebRTC IP Address Handling Requirements
draft-ietf-rtcweb-ip-handling-12
Abstract Abstract
This document provides information and requirements for how IP This document provides information and requirements for how IP
addresses should be handled by WebRTC implementations. addresses should be handled by Web Real-Time Communication (WebRTC)
implementations.
Status of This Memo Status of This Memo
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provisions of BCP 78 and BCP 79.
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Internet Standards is available in Section 2 of RFC 7841.
This Internet-Draft will expire on January 3, 2020. Information about the current status of this document, any errata,
and how to provide feedback on it may be obtained at
https://www.rfc-editor.org/info/rfc8828.
Copyright Notice Copyright Notice
Copyright (c) 2019 IETF Trust and the persons identified as the Copyright (c) 2021 IETF Trust and the persons identified as the
document authors. All rights reserved. document authors. All rights reserved.
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Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2 1. Introduction
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 2 2. Terminology
3. Problem Statement . . . . . . . . . . . . . . . . . . . . . . 2 3. Problem Statement
4. Goals . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4 4. Goals
5. Detailed Design . . . . . . . . . . . . . . . . . . . . . . . 5 5. Detailed Design
5.1. Principles . . . . . . . . . . . . . . . . . . . . . . . 5 5.1. Principles
5.2. Modes and Recommendations . . . . . . . . . . . . . . . . 5 5.2. Modes and Recommendations
6. Implementation Guidance . . . . . . . . . . . . . . . . . . . 7 6. Implementation Guidance
6.1. Ensuring Normal Routing . . . . . . . . . . . . . . . . . 7 6.1. Ensuring Normal Routing
6.2. Determining Associated Local Addresses . . . . . . . . . 7 6.2. Determining Associated Local Addresses
7. Application Guidance . . . . . . . . . . . . . . . . . . . . 8 7. Application Guidance
8. Security Considerations . . . . . . . . . . . . . . . . . . . 8 8. Security Considerations
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 8 9. IANA Considerations
10. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 8 10. References
11. References . . . . . . . . . . . . . . . . . . . . . . . . . 8 10.1. Normative References
11.1. Normative References . . . . . . . . . . . . . . . . . . 8 10.2. Informative References
11.2. Informative References . . . . . . . . . . . . . . . . . 9 Acknowledgements
Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 10 Authors' Addresses
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 12
1. Introduction 1. Introduction
One of WebRTC's key features is its support of peer-to-peer One of WebRTC's key features is its support of peer-to-peer
connections. However, when establishing such a connection, which connections. However, when establishing such a connection, which
involves connection attempts from various IP addresses, WebRTC may involves connection attempts from various IP addresses, WebRTC may
allow a web application to learn additional information about the allow a web application to learn additional information about the
user compared to an application that only uses the Hypertext Transfer user compared to an application that only uses the Hypertext Transfer
Protocol (HTTP) [RFC7230]. This may be problematic in certain cases. Protocol (HTTP) [RFC7230]. This may be problematic in certain cases.
This document summarizes the concerns, and makes recommendations on This document summarizes the concerns and makes recommendations on
how WebRTC implementations should best handle the tradeoff between how WebRTC implementations should best handle the trade-off between
privacy and media performance. privacy and media performance.
2. Terminology 2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
"OPTIONAL" in this document are to be interpreted as described in BCP "OPTIONAL" in this document are to be interpreted as described in BCP
14 [RFC2119][RFC8174] when, and only when, they appear in all 14 [RFC2119] [RFC8174] when, and only when, they appear in all
capitals, as shown here. capitals, as shown here.
3. Problem Statement 3. Problem Statement
In order to establish a peer-to-peer connection, WebRTC In order to establish a peer-to-peer connection, WebRTC
implementations use Interactive Connectivity Establishment (ICE) implementations use Interactive Connectivity Establishment (ICE)
[RFC8445], which attempts to discover multiple IP addresses using [RFC8445]. ICE attempts to discover multiple IP addresses using
techniques such as Session Traversal Utilities for NAT (STUN) techniques such as Session Traversal Utilities for NAT (STUN)
[RFC5389] and Traversal Using Relays around NAT (TURN) [RFC5766] and
[RFC5389] and Traversal Using Relays around NAT (TURN) [RFC5766], and
then checks the connectivity of each local-address-remote-address then checks the connectivity of each local-address-remote-address
pair in order to select the best one. The addresses that are pair in order to select the best one. The addresses that are
collected usually consist of an endpoint's private physical or collected usually consist of an endpoint's private physical or
virtual addresses and its public Internet addresses. virtual addresses and its public Internet addresses.
These addresses are provided to the web application so that they can These addresses are provided to the web application so that they can
be communicated to the remote endpoint for its checks. This allows be communicated to the remote endpoint for its checks. This allows
the application to learn more about the local network configuration the application to learn more about the local network configuration
than it would from a typical HTTP scenario, in which the web server 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 would only see a single public Internet address, i.e., the address
from which the HTTP request was sent. from which the HTTP request was sent.
The information revealed falls into three categories: The additional information revealed falls into three categories:
1. If the client is multihomed, additional public IP addresses for 1. If the client is multihomed, additional public IP addresses for
the client can be learned. In particular, if the client tries to the client can be learned. In particular, if the client tries to
hide its physical location through a Virtual Private Network hide its physical location through a Virtual Private Network
(VPN), and the VPN and local OS support routing over multiple (VPN), and the VPN and local OS support routing over multiple
interfaces (a "split-tunnel" VPN), WebRTC can discover not only interfaces (a "split-tunnel" VPN), WebRTC can discover not only
the public address for the VPN, but also the ISP public address the public address for the VPN, but also the ISP public address
over which the VPN is running. over which the VPN is running.
2. If the client is behind a Network Address Translator (NAT), the 2. If the client is behind a Network Address Translator (NAT), the
client's private IP addresses, often [RFC1918] addresses, can be client's private IP addresses, often [RFC1918] addresses, can be
learned. learned.
3. If the client is behind a proxy (a client-configured "classical 3. If the client is behind a proxy (a client-configured "classical
application proxy", as defined in [RFC1919], Section 3), but application proxy", as defined in [RFC1919], Section 3), but
direct access to the Internet is permitted, WebRTC's STUN checks direct access to the Internet is permitted, WebRTC's STUN checks
will bypass the proxy and reveal the public IP address of the will bypass the proxy and reveal the public IP address of the
client. This concern also applies to the "enterprise TURN client. This concern also applies to the "enterprise TURN
server" scenario described in [RFC7478], Section 2.3.5.1, if, as server" scenario described in [RFC7478], Section 2.3.5.1 if, as
above, direct Internet access is permitted. However, when the above, direct Internet access is permitted. However, when the
term "proxy" is used in this document, it is always in reference term "proxy" is used in this document, it is always in reference
to an [RFC1919] proxy server. to an [RFC1919] proxy server.
Of these three concerns, the first is the most significant, because Of these three concerns, the first is the most significant, because
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.
The second concern is less significant but valid nonetheless. The The second concern is less significant but valid nonetheless. The
core issue is that web applications can learn about addresses that core issue is that web applications can learn about addresses that
are not exposed to the internet; typically these address are IPv4, are not exposed to the Internet; typically, these address are IPv4,
but they can also be IPv6, as in the case of NAT64 [RFC6146]. While but they can also be IPv6, as in the case of NAT64 [RFC6146]. While
disclosure of the [RFC4941] IPv6 addresses recommended by disclosure of the [RFC4941] IPv6 addresses recommended by [RFC8835]
is fairly benign due to their intentionally short lifetimes, IPv4
[I-D.ietf-rtcweb-transports] is fairly benign due to their addresses present some challenges. Although private IPv4 addresses
intentionally short lifetimes, IPv4 addresses present some often contain minimal entropy (e.g., 192.168.0.2, a fairly common
challenges. Although private IPv4 addresses often contain minimal address), in the worst case, they can contain 24 bits of entropy with
entropy (e.g., 192.168.0.2, a fairly common address), in the worst an indefinite lifetime. As such, they can be a fairly significant
case, they can contain 24 bits of entropy with an indefinite fingerprinting surface. In addition, intranet web sites can be
lifetime. As such, they can be a fairly significant fingerprinting attacked more easily when their IPv4 address range is externally
surface. In addition, intranet web sites can be attacked more easily known.
when their IPv4 address range is externally known.
Private IP addresses can also act as an identifier that allows web Private IP addresses can also act as an identifier that allows web
applications running in isolated browsing contexts (e.g., normal and applications running in isolated browsing contexts (e.g., normal and
private browsing) to learn that they are running on the same device. private browsing) to learn that they are running on the same device.
This could allow the application sessions to be correlated, defeating This could allow the application sessions to be correlated, defeating
some of the privacy protections provided by isolation. It should be some of the privacy protections provided by isolation. It should be
noted that private addresses are just one potential mechanism for noted that private addresses are just one potential mechanism for
this correlation and this is an area for further study. this correlation and this is an area for further study.
The third concern is the least common, as proxy administrators can The third concern is the least common, as proxy administrators can
already control this behavior through organizational firewall policy, already control this behavior through organizational firewall policy,
and generally, forcing WebRTC traffic through a proxy server will and generally, forcing WebRTC traffic through a proxy server will
have negative effects on both the proxy and on media quality. have negative effects on both the proxy and media quality.
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 Real-Time provided similar functionality since the introduction of Real-Time
Media Flow Protocol (RTMFP) support [RFC7016] in 2008. Media Flow Protocol (RTMFP) support [RFC7016] in 2008.
4. Goals 4. Goals
WebRTC's support of secure peer-to-peer connections facilitates WebRTC's support of secure peer-to-peer connections facilitates
deployment of decentralized systems, which can have privacy benefits. deployment of decentralized systems, which can have privacy benefits.
As a result, blunt solutions that disable WebRTC or make it As a result, blunt solutions that disable WebRTC or make it
significantly harder to use are undesirable. This document takes a significantly harder to use are undesirable. This document takes a
more nuanced approach, with the following goals: more nuanced approach, with the following goals:
o Provide a framework for understanding the problem so that controls * Provide a framework for understanding the problem so that controls
might be provided to make different tradeoffs regarding might be provided to make different trade-offs regarding
performance and privacy concerns with WebRTC. performance and privacy concerns with WebRTC.
o Using that framework, define settings that enable peer-to-peer * Using that framework, define settings that enable peer-to-peer
communications, each with a different balance between performance communications, each with a different balance between performance
and privacy. and privacy.
o Finally, provide recommendations for default settings that provide * Finally, provide recommendations for default settings that provide
reasonable performance without also exposing addressing reasonable performance without also exposing addressing
information in a way that might violate user expectations. information in a way that might violate user expectations.
5. Detailed Design 5. Detailed Design
5.1. Principles 5.1. Principles
The key principles for our framework are stated below: The key principles for our framework are stated below:
1. By default, WebRTC traffic should follow typical IP routing, 1. By default, WebRTC traffic should follow typical IP routing
i.e., WebRTC should use the same interface used for HTTP traffic, (i.e., WebRTC should use the same interface used for HTTP
and only the system's 'typical' public addresses (or those of an traffic) and only the system's 'typical' public addresses (or
enterprise TURN server, if present) should be visible to the those of an enterprise TURN server, if present) should be visible
application. However, in the interest of optimal media quality, to the application. However, in the interest of optimal media
it should be possible to enable WebRTC to make use of all network quality, it should be possible to enable WebRTC to make use of
interfaces to determine the ideal route. all network interfaces to determine the ideal route.
2. By default, WebRTC should be able to negotiate direct peer-to- 2. By default, WebRTC should be able to negotiate direct peer-to-
peer connections between endpoints (i.e., without traversing a peer connections between endpoints (i.e., without traversing a
NAT or relay server) when such connections are possible. This NAT or relay server) when such connections are possible. This
ensures that applications that need true peer-to-peer routing for ensures that applications that need true peer-to-peer routing for
bandwidth or latency reasons can operate successfully. bandwidth or latency reasons can operate successfully.
3. It should be possible to configure WebRTC to not disclose private 3. It should be possible to configure WebRTC to not disclose private
local IP addresses, to avoid the issues associated with web local IP addresses, to avoid the issues associated with web
applications learning such addresses. This document does not applications learning such addresses. This document does not
require this to be the default state, as there is no currently require this to be the default state, as there is no currently
defined mechanism that can satisfy this requirement as well as defined mechanism that can satisfy this requirement as well as
the aforementioned requirement to allow direct peer-to-peer the aforementioned requirement to allow direct peer-to-peer
connections. connections.
4. By default, WebRTC traffic should not be sent through proxy 4. By default, WebRTC traffic should not be sent through proxy
servers, due to the media quality problems associated with servers, due to the media-quality problems associated with
sending WebRTC traffic over TCP, which is almost always used when sending WebRTC traffic over TCP, which is almost always used when
communicating with such proxies, as well as proxy performance communicating with such proxies, as well as proxy performance
issues that may result from proxying WebRTC's long-lived, high- issues that may result from proxying WebRTC's long-lived, high-
bandwidth connections. However, it should be possible to force bandwidth connections. However, it should be possible to force
WebRTC to send its traffic through a configured proxy if desired. WebRTC to send its traffic through a configured proxy if desired.
5.2. Modes and Recommendations 5.2. Modes and Recommendations
Based on these ideas, we define four specific modes of WebRTC Based on these ideas, we define four specific modes of WebRTC
behavior, reflecting different media quality/privacy tradeoffs: behavior, reflecting different media quality/privacy trade-offs:
Mode 1: Enumerate all addresses: WebRTC MUST use all network Mode 1 - Enumerate all addresses:
interfaces to attempt communication with STUN servers, TURN WebRTC MUST use all network interfaces to attempt communication
servers, or peers. This will converge on the best media with STUN servers, TURN servers, or peers. This will converge on
path, and is ideal when media performance is the highest the best media path and is ideal when media performance is the
priority, but it discloses the most information. highest priority, but it discloses the most information.
Mode 2: Default route + associated local addresses: WebRTC MUST Mode 2 - Default route + associated local addresses:
follow the kernel routing table rules, which will typically WebRTC MUST follow the kernel routing table rules, which will
cause media packets to take the same route as the typically cause media packets to take the same route as the
application's HTTP traffic. If an enterprise TURN server is application's HTTP traffic. If an enterprise TURN server is
present, the preferred route MUST be through this TURN present, the preferred route MUST be through this TURN server.
server. Once an interface has been chosen, the private IPv4 Once an interface has been chosen, the private IPv4 and IPv6
and IPv6 addresses associated with this interface MUST be addresses associated with this interface MUST be discovered and
discovered and provided to the application as host provided to the application as host candidates. This ensures that
candidates. This ensures that direct connections can still direct connections can still be established in this mode.
be established in this mode.
Mode 3: Default route only: This is the the same as Mode 2, except Mode 3 - Default route only:
that the associated private addresses MUST NOT be provided; This is the same as Mode 2, except that the associated private
the only IP addresses gathered are those discovered via addresses MUST NOT be provided; the only IP addresses gathered are
mechanisms like STUN and TURN (on the default route). This those discovered via mechanisms like STUN and TURN (on the default
may cause traffic to hairpin through a NAT, fall back to an route). This may cause traffic to hairpin through a NAT, fall
application TURN server, or fail altogether, with resulting back to an application TURN server, or fail altogether, with
quality implications. resulting quality implications.
Mode 4: Force proxy: This is the same as Mode 3, but when the Mode 4 - Force proxy:
application's HTTP traffic is sent through a proxy, WebRTC This is the same as Mode 3, but when the application's HTTP
media traffic MUST also be proxied. If the proxy does not traffic is sent through a proxy, WebRTC media traffic MUST also be
support UDP (as is the case for all HTTP and most SOCKS proxied. If the proxy does not support UDP (as is the case for
[RFC1928] proxies), or the WebRTC implementation does not all HTTP and most SOCKS [RFC1928] proxies), or the WebRTC
support UDP proxying, the use of UDP will be disabled, and implementation does not support UDP proxying, the use of UDP will
TCP will be used to send and receive media through the be disabled, and TCP will be used to send and receive media
proxy. Use of TCP will result in reduced media quality, in through the proxy. Use of TCP will result in reduced media
addition to any performance considerations associated with quality, in addition to any performance considerations associated
sending all WebRTC media through the proxy server. with sending all WebRTC media through the proxy server.
Mode 1 MUST NOT be used unless user consent has been provided. The Mode 1 MUST NOT be used unless user consent has been provided. The
details of this consent are left to the implementation; one potential details of this consent are left to the implementation; one potential
mechanism is to tie this consent to getUserMedia (device permissions) mechanism is to tie this consent to getUserMedia (device permissions)
consent, described in [I-D.ietf-rtcweb-security-arch], Section 6.2. consent, described in [RFC8827], Section 6.2. Alternatively,
Alternatively, implementations can provide a specific mechanism to implementations can provide a specific mechanism to obtain user
obtain user consent. consent.
In cases where user consent has not been obtained, Mode 2 SHOULD be In cases where user consent has not been obtained, Mode 2 SHOULD be
used. used.
These defaults provide a reasonable tradeoff that permits trusted These defaults provide a reasonable trade-off that permits trusted
WebRTC applications to achieve optimal network performance, but gives WebRTC applications to achieve optimal network performance but gives
applications without consent (e.g., 1-way streaming or data channel applications without consent (e.g., 1-way streaming or data-channel
applications) only the minimum information needed to achieve direct applications) only the minimum information needed to achieve direct
connections, as defined in Mode 2. However, implementations MAY connections, as defined in Mode 2. However, implementations MAY
choose stricter modes if desired, e.g., if a user indicates they want choose stricter modes if desired, e.g., if a user indicates they want
all WebRTC traffic to follow the default route. all WebRTC traffic to follow the default route.
Future documents may define additional modes and/or update the Future documents may define additional modes and/or update the
recommended default modes. recommended default modes.
Note that the suggested defaults can still be used even for Note that the suggested defaults can still be used even for
organizations that want all external WebRTC traffic to traverse a organizations that want all external WebRTC traffic to traverse a
proxy or enterprise TURN server, simply by setting an organizational proxy or enterprise TURN server, simply by setting an organizational
firewall policy that allows WebRTC traffic to only leave through the firewall policy that allows WebRTC traffic to only leave through the
proxy or TURN server. This provides a way to ensure the proxy or proxy or TURN server. This provides a way to ensure the proxy or
TURN server is used for any external traffic, but still allows direct TURN server is used for any external traffic but still allows direct
connections (and, in the proxy case, avoids the performance issues connections (and, in the proxy case, avoids the performance issues
associated with forcing media through said proxy) for intra- associated with forcing media through said proxy) for intra-
organization traffic. organization traffic.
6. Implementation Guidance 6. Implementation Guidance
This section provides guidance to WebRTC implementations on how to This section provides guidance to WebRTC implementations on how to
implement the policies described above. implement the policies described above.
6.1. Ensuring Normal Routing 6.1. Ensuring Normal Routing
skipping to change at page 7, line 38 skipping to change at line 312
IPv6), which allows the OS to route WebRTC traffic the same way as it 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 would HTTP traffic. STUN and TURN will work as usual, and host
candidates can still be determined as mentioned below. candidates can still be determined as mentioned below.
6.2. Determining Associated Local Addresses 6.2. Determining Associated Local Addresses
When binding to a wildcard address, some extra work is needed to When binding to a wildcard address, some extra work is needed to
determine the associated local address required by Mode 2, which we determine the associated local address required by Mode 2, which we
define as the source address that would be used for any packets sent define as the source address that would be used for any packets sent
to the web application host (assuming that UDP and TCP get the same to the web application host (assuming that UDP and TCP get the same
routing treatment). Use of the web application host as a destination routing treatment). Use of the web-application host as a destination
ensures the right source address is selected, regardless of where the ensures the right source address is selected, regardless of where the
application resides (e.g., on an intranet). application resides (e.g., on an intranet).
First, the appropriate remote IPv4/IPv6 address is obtained by First, the appropriate remote IPv4/IPv6 address is obtained by
resolving the host component of the web application URI [RFC3986]. resolving the host component of the web application URI [RFC3986].
If the client is behind a proxy and cannot resolve these IPs via DNS, If the client is behind a proxy and cannot resolve these IPs via DNS,
the address of the proxy can be used instead. Or, if the web the address of the proxy can be used instead. Or, if the web
application was loaded from a file:// URI [RFC8089], rather than over application was loaded from a file:// URI [RFC8089] rather than over
the network, the implementation can fall back to a well-known DNS the network, the implementation can fall back to a well-known DNS
name or IP address. name or IP address.
Once a suitable remote IP has been determined, the implementation can Once a suitable remote IP has been determined, the implementation can
create a UDP socket, bind() it to the appropriate wildcard address, create a UDP socket, bind() it to the appropriate wildcard address,
and then connect() to the remote IP. Generally, this results in the and then connect() to the remote IP. Generally, this results in the
socket being assigned a local address based on the kernel routing socket being assigned a local address based on the kernel routing
table, without sending any packets over the network. table, without sending any packets over the network.
Finally, the socket can be queried using getsockname() or the Finally, the socket can be queried using getsockname() or the
equivalent to determine the appropriate local address. equivalent to determine the appropriate local address.
7. Application Guidance 7. 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, the following guidelines are provided for applications: scenarios, the following guidelines are provided for applications:
o Applications SHOULD deploy a TURN server with support for both UDP * 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 is in use,
assuming the TURN server can be reached. assuming the TURN server can be reached.
o Applications SHOULD detect when they don't have access to the full * 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 is in
is in use; this knowledge can be useful for diagnostic purposes. use; this knowledge can be useful for diagnostic purposes.
8. Security Considerations 8. Security Considerations
This document describes several potential privacy and security This document describes several potential privacy and security
concerns associated with WebRTC peer-to-peer connections, and concerns associated with WebRTC peer-to-peer connections and provides
provides mechanisms and recommendations for WebRTC implementations to mechanisms and recommendations for WebRTC implementations to address
address these concerns. these concerns.
9. IANA Considerations 9. IANA Considerations
This document requires no actions from IANA. This document has no IANA actions.
10. Acknowledgements
Several people provided input into this document, including Bernard
Aboba, Harald Alvestrand, Youenn Fablet, Ted Hardie, Matthew
Kaufmann, Eric Rescorla, Adam Roach, and Martin Thomson.
11. References 10. References
11.1. Normative References 10.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997, DOI 10.17487/RFC2119, March 1997,
<https://www.rfc-editor.org/info/rfc2119>. <https://www.rfc-editor.org/info/rfc2119>.
[RFC3986] Berners-Lee, T., Fielding, R., and L. Masinter, "Uniform [RFC3986] Berners-Lee, T., Fielding, R., and L. Masinter, "Uniform
Resource Identifier (URI): Generic Syntax", STD 66, Resource Identifier (URI): Generic Syntax", STD 66,
RFC 3986, DOI 10.17487/RFC3986, January 2005, RFC 3986, DOI 10.17487/RFC3986, January 2005,
<https://www.rfc-editor.org/info/rfc3986>. <https://www.rfc-editor.org/info/rfc3986>.
skipping to change at page 9, line 35 skipping to change at line 399
[RFC8174] Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC [RFC8174] Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC
2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174, 2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174,
May 2017, <https://www.rfc-editor.org/info/rfc8174>. May 2017, <https://www.rfc-editor.org/info/rfc8174>.
[RFC8445] Keranen, A., Holmberg, C., and J. Rosenberg, "Interactive [RFC8445] Keranen, A., Holmberg, C., and J. Rosenberg, "Interactive
Connectivity Establishment (ICE): A Protocol for Network Connectivity Establishment (ICE): A Protocol for Network
Address Translator (NAT) Traversal", RFC 8445, Address Translator (NAT) Traversal", RFC 8445,
DOI 10.17487/RFC8445, July 2018, DOI 10.17487/RFC8445, July 2018,
<https://www.rfc-editor.org/info/rfc8445>. <https://www.rfc-editor.org/info/rfc8445>.
11.2. Informative References 10.2. Informative References
[I-D.ietf-rtcweb-security-arch]
Rescorla, E., "WebRTC Security Architecture", draft-ietf-
rtcweb-security-arch-18 (work in progress), February 2019.
[I-D.ietf-rtcweb-transports]
Alvestrand, H., "Transports for WebRTC", draft-ietf-
rtcweb-transports-17 (work in progress), October 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", J., and E. Lear, "Address Allocation for Private
BCP 5, RFC 1918, DOI 10.17487/RFC1918, February 1996, Internets", BCP 5, RFC 1918, DOI 10.17487/RFC1918,
<https://www.rfc-editor.org/info/rfc1918>. February 1996, <https://www.rfc-editor.org/info/rfc1918>.
[RFC1919] Chatel, M., "Classical versus Transparent IP Proxies", [RFC1919] Chatel, M., "Classical versus Transparent IP Proxies",
RFC 1919, DOI 10.17487/RFC1919, March 1996, RFC 1919, DOI 10.17487/RFC1919, March 1996,
<https://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 [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,
<https://www.rfc-editor.org/info/rfc1928>. <https://www.rfc-editor.org/info/rfc1928>.
skipping to change at page 10, line 34 skipping to change at line 439
[RFC7230] Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer [RFC7230] Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer
Protocol (HTTP/1.1): Message Syntax and Routing", Protocol (HTTP/1.1): Message Syntax and Routing",
RFC 7230, DOI 10.17487/RFC7230, June 2014, RFC 7230, DOI 10.17487/RFC7230, June 2014,
<https://www.rfc-editor.org/info/rfc7230>. <https://www.rfc-editor.org/info/rfc7230>.
[RFC7478] Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real- [RFC7478] Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-
Time Communication Use Cases and Requirements", RFC 7478, Time Communication Use Cases and Requirements", RFC 7478,
DOI 10.17487/RFC7478, March 2015, DOI 10.17487/RFC7478, March 2015,
<https://www.rfc-editor.org/info/rfc7478>. <https://www.rfc-editor.org/info/rfc7478>.
Appendix A. Change log [RFC8827] Rescorla, E., "WebRTC Security Architecture", RFC 8827,
DOI 10.17487/RFC8827, January 2021,
Changes in draft -12: <https://www.rfc-editor.org/info/rfc8827>.
o Editorial updates from IETF LC review.
Changes in draft -11:
o Editorial updates from AD review.
Changes in draft -10:
o Incorporate feedback from IETF 102 on the problem space.
o Note that future versions of the document may define new modes.
Changes in draft -09:
o Fixed confusing text regarding enterprise TURN servers.
Changes in draft -08:
o Discuss how enterprise TURN servers should be handled.
Changes in draft -07:
o Clarify consent guidance.
Changes in draft -06:
o Clarify recommendations.
o Split implementation guidance into two sections.
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. [RFC8835] Alvestrand, H., "Transports for WebRTC", RFC 8835,
DOI 10.17487/RFC8835, January 2021,
<https://www.rfc-editor.org/info/rfc8835>.
Changes in draft -00: Acknowledgements
o Published as WG draft. Several people provided input into this document, including Bernard
Aboba, Harald Alvestrand, Youenn Fablet, Ted Hardie, Matthew
Kaufmann, Eric Rescorla, Adam Roach, and Martin Thomson.
Author's Address Authors' Addresses
Justin Uberti Justin Uberti
Google Google
747 6th St S 747 6th St S
Kirkland, WA 98033 Kirkland, WA 98033
USA United States of America
Email: justin@uberti.name Email: justin@uberti.name
Guo-wei Shieh
333 Elliott Ave W #500
Seattle, WA 98119
United States of America
Email: guoweis@gmail.com
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