rfc8828xml2.original.xml   rfc8828.xml 
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<?xml-stylesheet type="text/xsl" href="rfc2629.xslt" ?> rue" xml:lang="en">
<?rfc toc="yes" ?> <link href="https://datatracker.ietf.org/doc/draft-ietf-rtcweb-ip-handling-12"
<?rfc symrefs="yes" ?> rel="prev"/>
<?rfc iprnotified="no" ?> <link href="https://dx.doi.org/10.17487/rfc8828" rel="alternate"/>
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<front> <front>
<title abbrev="WebRTC IP Handling">WebRTC IP Address Handling <title abbrev="WebRTC IP Handling">WebRTC IP Address Handling Requirements</
Requirements</title> title>
<seriesInfo name="RFC" value="8828" stream="IETF"/>
<author fullname="Justin Uberti" initials="J." surname="Uberti"> <author fullname="Justin Uberti" initials="J." surname="Uberti">
<organization>Google</organization> <organization showOnFrontPage="true">Google</organization>
<address> <address>
<postal> <postal>
<street>747 6th St S</street> <street>747 6th St S</street>
<city>Kirkland</city> <city>Kirkland</city>
<region>WA</region> <region>WA</region>
<code>98033</code> <code>98033</code>
<country>USA</country> <country>United States of America</country>
</postal> </postal>
<email>justin@uberti.name</email> <email>justin@uberti.name</email>
</address> </address>
</author> </author>
<date year="2019" month="July" /> <author fullname="Guo-wei Shieh" initials="G." surname="Shieh">
<organization showOnFrontPage="true"/>
<address>
<postal>
<street>333 Elliott Ave W #500</street>
<city>Seattle</city>
<region>WA</region>
<code>98119</code>
<country>United States of America</country>
</postal>
<email>guoweis@gmail.com</email>
</address>
</author>
<date month="01" year="2021"/>
<area>RAI</area> <area>RAI</area>
<abstract pn="section-abstract">
<!-- [rfced] Please insert any keywords (beyond those that appear in <t indent="0" pn="section-abstract-1">This document provides information a
the title) for use on https://www.rfc-editor.org/search. --> nd requirements for how IP
addresses should be handled by Web Real-Time Communication (WebRTC) implem
<keyword>example</keyword> entations.</t>
<abstract>
<t>This document provides information and requirements for how IP
addresses should be handled by WebRTC implementations.</t>
</abstract> </abstract>
<boilerplate>
<section anchor="status-of-memo" numbered="false" removeInRFC="false" toc=
"exclude" pn="section-boilerplate.1">
<name slugifiedName="name-status-of-this-memo">Status of This Memo</name
>
<t indent="0" pn="section-boilerplate.1-1">
This is an Internet Standards Track document.
</t>
<t indent="0" pn="section-boilerplate.1-2">
This document is a product of the Internet Engineering Task Force
(IETF). It represents the consensus of the IETF community. It has
received public review and has been approved for publication by
the Internet Engineering Steering Group (IESG). Further
information on Internet Standards is available in Section 2 of
RFC 7841.
</t>
<t indent="0" pn="section-boilerplate.1-3">
Information about the current status of this document, any
errata, and how to provide feedback on it may be obtained at
<eref target="https://www.rfc-editor.org/info/rfc8828" brackets="non
e"/>.
</t>
</section>
<section anchor="copyright" numbered="false" removeInRFC="false" toc="excl
ude" pn="section-boilerplate.2">
<name slugifiedName="name-copyright-notice">Copyright Notice</name>
<t indent="0" pn="section-boilerplate.2-1">
Copyright (c) 2021 IETF Trust and the persons identified as the
document authors. All rights reserved.
</t>
<t indent="0" pn="section-boilerplate.2-2">
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
(<eref target="https://trustee.ietf.org/license-info" brackets="none
"/>) 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.
</t>
</section>
</boilerplate>
<toc>
<section anchor="toc" numbered="false" removeInRFC="false" toc="exclude" p
n="section-toc.1">
<name slugifiedName="name-table-of-contents">Table of Contents</name>
<ul bare="true" empty="true" indent="2" spacing="compact" pn="section-to
c.1-1">
<li pn="section-toc.1-1.1">
<t indent="0" keepWithNext="true" pn="section-toc.1-1.1.1"><xref der
ivedContent="1" format="counter" sectionFormat="of" target="section-1"/>.  <xref
derivedContent="" format="title" sectionFormat="of" target="name-introduction">
Introduction</xref></t>
</li>
<li pn="section-toc.1-1.2">
<t indent="0" keepWithNext="true" pn="section-toc.1-1.2.1"><xref der
ivedContent="2" format="counter" sectionFormat="of" target="section-2"/>.  <xref
derivedContent="" format="title" sectionFormat="of" target="name-terminology">T
erminology</xref></t>
</li>
<li pn="section-toc.1-1.3">
<t indent="0" keepWithNext="true" pn="section-toc.1-1.3.1"><xref der
ivedContent="3" format="counter" sectionFormat="of" target="section-3"/>.  <xref
derivedContent="" format="title" sectionFormat="of" target="name-problem-statem
ent">Problem Statement</xref></t>
</li>
<li pn="section-toc.1-1.4">
<t indent="0" pn="section-toc.1-1.4.1"><xref derivedContent="4" form
at="counter" sectionFormat="of" target="section-4"/>.  <xref derivedContent="" f
ormat="title" sectionFormat="of" target="name-goals">Goals</xref></t>
</li>
<li pn="section-toc.1-1.5">
<t indent="0" pn="section-toc.1-1.5.1"><xref derivedContent="5" form
at="counter" sectionFormat="of" target="section-5"/>.  <xref derivedContent="" f
ormat="title" sectionFormat="of" target="name-detailed-design">Detailed Design</
xref></t>
<ul bare="true" empty="true" indent="2" spacing="compact" pn="sectio
n-toc.1-1.5.2">
<li pn="section-toc.1-1.5.2.1">
<t indent="0" pn="section-toc.1-1.5.2.1.1"><xref derivedContent=
"5.1" format="counter" sectionFormat="of" target="section-5.1"/>.  <xref derived
Content="" format="title" sectionFormat="of" target="name-principles">Principles
</xref></t>
</li>
<li pn="section-toc.1-1.5.2.2">
<t indent="0" pn="section-toc.1-1.5.2.2.1"><xref derivedContent=
"5.2" format="counter" sectionFormat="of" target="section-5.2"/>.  <xref derived
Content="" format="title" sectionFormat="of" target="name-modes-and-recommendati
ons">Modes and Recommendations</xref></t>
</li>
</ul>
</li>
<li pn="section-toc.1-1.6">
<t indent="0" pn="section-toc.1-1.6.1"><xref derivedContent="6" form
at="counter" sectionFormat="of" target="section-6"/>.  <xref derivedContent="" f
ormat="title" sectionFormat="of" target="name-implementation-guidance">Implement
ation Guidance</xref></t>
<ul bare="true" empty="true" indent="2" spacing="compact" pn="sectio
n-toc.1-1.6.2">
<li pn="section-toc.1-1.6.2.1">
<t indent="0" pn="section-toc.1-1.6.2.1.1"><xref derivedContent=
"6.1" format="counter" sectionFormat="of" target="section-6.1"/>.  <xref derived
Content="" format="title" sectionFormat="of" target="name-ensuring-normal-routin
g">Ensuring Normal Routing</xref></t>
</li>
<li pn="section-toc.1-1.6.2.2">
<t indent="0" pn="section-toc.1-1.6.2.2.1"><xref derivedContent=
"6.2" format="counter" sectionFormat="of" target="section-6.2"/>.  <xref derived
Content="" format="title" sectionFormat="of" target="name-determining-associated
-loca">Determining Associated Local Addresses</xref></t>
</li>
</ul>
</li>
<li pn="section-toc.1-1.7">
<t indent="0" pn="section-toc.1-1.7.1"><xref derivedContent="7" form
at="counter" sectionFormat="of" target="section-7"/>.  <xref derivedContent="" f
ormat="title" sectionFormat="of" target="name-application-guidance">Application
Guidance</xref></t>
</li>
<li pn="section-toc.1-1.8">
<t indent="0" pn="section-toc.1-1.8.1"><xref derivedContent="8" form
at="counter" sectionFormat="of" target="section-8"/>.  <xref derivedContent="" f
ormat="title" sectionFormat="of" target="name-security-considerations">Security
Considerations</xref></t>
</li>
<li pn="section-toc.1-1.9">
<t indent="0" pn="section-toc.1-1.9.1"><xref derivedContent="9" form
at="counter" sectionFormat="of" target="section-9"/>.  <xref derivedContent="" f
ormat="title" sectionFormat="of" target="name-iana-considerations">IANA Consider
ations</xref></t>
</li>
<li pn="section-toc.1-1.10">
<t indent="0" pn="section-toc.1-1.10.1"><xref derivedContent="10" fo
rmat="counter" sectionFormat="of" target="section-10"/>. <xref derivedContent=""
format="title" sectionFormat="of" target="name-references">References</xref></t
>
<ul bare="true" empty="true" indent="2" spacing="compact" pn="sectio
n-toc.1-1.10.2">
<li pn="section-toc.1-1.10.2.1">
<t indent="0" pn="section-toc.1-1.10.2.1.1"><xref derivedContent
="10.1" format="counter" sectionFormat="of" target="section-10.1"/>.  <xref deri
vedContent="" format="title" sectionFormat="of" target="name-normative-reference
s">Normative References</xref></t>
</li>
<li pn="section-toc.1-1.10.2.2">
<t indent="0" pn="section-toc.1-1.10.2.2.1"><xref derivedContent
="10.2" format="counter" sectionFormat="of" target="section-10.2"/>.  <xref deri
vedContent="" format="title" sectionFormat="of" target="name-informative-referen
ces">Informative References</xref></t>
</li>
</ul>
</li>
<li pn="section-toc.1-1.11">
<t indent="0" pn="section-toc.1-1.11.1"><xref derivedContent="" form
at="none" sectionFormat="of" target="section-appendix.a"/><xref derivedContent="
" format="title" sectionFormat="of" target="name-acknowledgements">Acknowledgeme
nts</xref></t>
</li>
<li pn="section-toc.1-1.12">
<t indent="0" pn="section-toc.1-1.12.1"><xref derivedContent="" form
at="none" sectionFormat="of" target="section-appendix.b"/><xref derivedContent="
" format="title" sectionFormat="of" target="name-authors-addresses">Authors' Add
resses</xref></t>
</li>
</ul>
</section>
</toc>
</front> </front>
<middle> <middle>
<section title="Introduction"> <section numbered="true" toc="include" removeInRFC="false" pn="section-1">
<name slugifiedName="name-introduction">Introduction</name>
<t>One of WebRTC's key features is its support of peer-to-peer <t indent="0" pn="section-1-1">One of WebRTC's key features is its support
of peer-to-peer
connections. However, when establishing such a connection, which involves connections. However, when establishing such a connection, which involves
connection attempts from various IP addresses, WebRTC may allow a web connection attempts from various IP addresses, WebRTC may allow a web
application to learn additional information about the user compared to an application to learn additional information about the user compared to an
application that only uses the Hypertext Transfer Protocol (HTTP) application that only uses the Hypertext Transfer Protocol (HTTP)
<xref target="RFC7230" />. This may be problematic in certain cases. This <xref target="RFC7230" format="default" sectionFormat="of" derivedContent
document summarizes the concerns, and makes recommendations on how WebRTC ="RFC7230"/>. This may be problematic in
implementations should best handle the tradeoff between privacy and media certain cases. This
performance.</t> document summarizes the concerns and makes recommendations on how WebRTC
implementations should best handle the trade-off between privacy and medi
a
performance.</t>
</section> </section>
<section title="Terminology"> <section numbered="true" toc="include" removeInRFC="false" pn="section-2">
<name slugifiedName="name-terminology">Terminology</name>
<t>The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", <t indent="0" pn="section-2-1">
"SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and The key words "<bcp14>MUST</bcp14>", "<bcp14>MUST NOT</bcp14>",
"OPTIONAL" in this document are to be interpreted as described in BCP 14 "<bcp14>REQUIRED</bcp14>", "<bcp14>SHALL</bcp14>", "<bcp14>SHALL NOT</bcp14>
<xref target="RFC2119"></xref><xref target="RFC8174"></xref> ",
when, and only when, they appear in all capitals, as shown here.</t> "<bcp14>SHOULD</bcp14>", "<bcp14>SHOULD NOT</bcp14>",
"<bcp14>RECOMMENDED</bcp14>", "<bcp14>NOT RECOMMENDED</bcp14>",
"<bcp14>MAY</bcp14>", and "<bcp14>OPTIONAL</bcp14>" in this document are to
be
interpreted as described in BCP 14 <xref target="RFC2119" format="default" s
ectionFormat="of" derivedContent="RFC2119"/> <xref target="RFC8174" format="defa
ult" sectionFormat="of" derivedContent="RFC8174"/> when, and only when, they app
ear in all capitals, as
shown here.
</t>
</section> </section>
<section title="Problem Statement"> <section numbered="true" toc="include" removeInRFC="false" pn="section-3">
<name slugifiedName="name-problem-statement">Problem Statement</name>
<t>In order to establish a peer-to-peer connection, WebRTC <t indent="0" pn="section-3-1">In order to establish a peer-to-peer connec
tion, WebRTC
implementations use Interactive Connectivity Establishment (ICE) implementations use Interactive Connectivity Establishment (ICE)
<xref target="RFC8445" />, which attempts to discover multiple IP <xref target="RFC8445" format="default" sectionFormat="of" derivedContent= "RFC8445"/>. ICE attempts to discover multiple IP
addresses using techniques such as Session Traversal Utilities for NAT addresses using techniques such as Session Traversal Utilities for NAT
(STUN) (STUN)
<xref target="RFC5389" /> and Traversal Using Relays around NAT (TURN) <xref target="RFC5389" format="default" sectionFormat="of" derivedContent=
<xref target="RFC5766" />, and then checks the connectivity of each "RFC5389"/> and Traversal Using Relays
around NAT (TURN)
<xref target="RFC5766" format="default" sectionFormat="of" derivedContent=
"RFC5766"/> and then checks the
connectivity of each
local-address-remote-address pair in order to select the best one. The local-address-remote-address pair in order to select the best one. The
addresses that are collected usually consist of an endpoint's private addresses that are collected usually consist of an endpoint's private
physical or virtual addresses and its public Internet addresses.</t> physical or virtual addresses and its public Internet addresses.</t>
<t indent="0" pn="section-3-2">These addresses are provided to the web app
<t>These addresses are provided to the web application so that lication so that
they can be communicated to the remote endpoint for its checks. This they can be communicated to the remote endpoint for its checks. This
allows the application to learn more about the local network allows the application to learn more about the local network
configuration than it would from a typical HTTP scenario, in which the 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 web server would only see a single public Internet address, i.e., the
address from which the HTTP request was sent.</t> address from which the HTTP request was sent.</t>
<t indent="0" pn="section-3-3">The additional information revealed falls i
<t>The information revealed falls into three categories: nto three categories:
<list style="numbers"> </t>
<ol spacing="normal" type="1" indent="adaptive" start="1" pn="section-3-4"
<t>If the client is multihomed, additional public IP addresses for the >
<li pn="section-3-4.1" derivedCounter="1.">If the client is multihomed,
additional public IP addresses for the
client can be learned. In particular, if the client tries to hide its client can be learned. In particular, if the client tries to hide its
physical location through a Virtual Private Network (VPN), and the VPN physical location through a Virtual Private Network (VPN), and the VPN
and local OS support routing over multiple interfaces (a "split-tunnel" and local OS support routing over multiple interfaces (a "split-tunnel"
VPN), WebRTC can discover not only the public address for the VPN, but VPN), WebRTC can discover not only the public address for the VPN, but
also the ISP public address over which the VPN is running.</t> also the ISP public address over which the VPN is running.</li>
<li pn="section-3-4.2" derivedCounter="2.">If the client is behind a Net
<t>If the client is behind a Network Address Translator (NAT), the work Address Translator (NAT), the
client's private IP addresses, often client's private IP addresses, often
<xref target="RFC1918" /> addresses, can be learned.</t> <xref target="RFC1918" format="default" sectionFormat="of" derivedConten
t="RFC1918"/> addresses, can be learned.</li>
<t>If the client is behind a proxy (a client-configured "classical <li pn="section-3-4.3" derivedCounter="3.">If the client is behind a pro
xy (a client-configured "classical
application proxy", as defined in application proxy", as defined in
<xref target="RFC1919" />, Section 3), but direct access to the <xref target="RFC1919" format="default" sectionFormat="comma" section="3 " derivedLink="https://rfc-editor.org/rfc/rfc1919#section-3" derivedContent="RFC 1919"/>), but direct access to the
Internet is permitted, WebRTC's STUN checks will bypass the proxy and Internet is permitted, WebRTC's STUN checks will bypass the proxy and
reveal the public IP address of the client. This concern also applies reveal the public IP address of the client. This concern also applies
to the "enterprise TURN server" scenario described in to the "enterprise TURN server" scenario described in
<xref target="RFC7478" />, Section 2.3.5.1, if, as above, direct <xref target="RFC7478" format="default" sectionFormat="comma" section="2 .3.5.1" derivedLink="https://rfc-editor.org/rfc/rfc7478#section-2.3.5.1" derived Content="RFC7478"/> if, as above, direct
Internet access is permitted. However, when the term "proxy" is used in Internet access is permitted. However, when the term "proxy" is used in
this document, it is always in reference to an this document, it is always in reference to an
<xref target="RFC1919" /> proxy server.</t> <xref target="RFC1919" format="default" sectionFormat="of" derivedConten
</list></t> t="RFC1919"/> proxy server.</li>
</ol>
<t>Of these three concerns, the first is the most significant, because for <t indent="0" pn="section-3-5">Of these three concerns, the first is the m
some ost significant, because for some
users, the purpose of using a VPN is for anonymity. However, different 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 will have different needs, and some VPN users (e.g., corporate
VPN users) may in fact prefer WebRTC to send media traffic directly, VPN users) may in fact prefer WebRTC to send media traffic directly --
i.e., not through the VPN.</t> i.e., not through the VPN.</t>
<t indent="0" pn="section-3-6">The second concern is less significant but
<t>The second concern is less significant but valid nonetheless. The core valid nonetheless. The core
issue is that web applications can learn about addresses that are not issue is that web applications can learn about addresses that are not
exposed to the internet; typically these address are IPv4, but they can exposed to the Internet; typically, these address are IPv4, but they can
also be IPv6, as in the case of NAT64 <xref target="RFC6146" />. also be IPv6, as in the case of NAT64 <xref target="RFC6146" format="defau
While disclosure of the <xref target="RFC4941" /> IPv6 addresses lt" sectionFormat="of" derivedContent="RFC6146"/>.
recommended by <xref target="WEBRTC-TRANSPORTS" /> is fairly While disclosure of the <xref target="RFC4941" format="default" sectionFor
mat="of" derivedContent="RFC4941"/> IPv6 addresses
recommended by <xref target="RFC8835" format="default" sectionFormat="of"
derivedContent="RFC8835"/> is fairly
benign due to their intentionally short lifetimes, IPv4 addresses present benign due to their intentionally short lifetimes, IPv4 addresses present
some challenges. Although private IPv4 addresses often contain minimal some challenges. Although private IPv4 addresses often contain minimal
entropy (e.g., 192.168.0.2, a fairly common address), in the worst case, entropy (e.g., 192.168.0.2, a fairly common address), in the worst case,
they can contain 24 bits of entropy with an indefinite lifetime. As such, they can contain 24 bits of entropy with an indefinite lifetime. As such,
they can be a fairly significant fingerprinting surface. In addition, they can be a fairly significant fingerprinting surface. In addition,
intranet web sites can be attacked more easily when their IPv4 address intranet web sites can be attacked more easily when their IPv4 address
range is externally known.</t> range is externally known.</t>
<t indent="0" pn="section-3-7">Private IP addresses can also act as an ide
<t>Private IP addresses can also act as an identifier that allows ntifier that allows
web applications running in isolated browsing contexts (e.g., normal and web applications running in isolated browsing contexts (e.g., normal and
private browsing) to learn that they are running on the same device. This private browsing) to learn that they are running on the same device. This
could allow the application sessions to be correlated, defeating some of could allow the application sessions to be correlated, defeating some of
the privacy protections provided by isolation. It should be noted that the privacy protections provided by isolation. It should be noted that
private addresses are just one potential mechanism for this correlation private addresses are just one potential mechanism for this correlation
and this is an area for further study.</t> and this is an area for further study.</t>
<t indent="0" pn="section-3-8">The third concern is the least common, as p
<t>The third concern is the least common, as proxy administrators can alre roxy administrators can already
ady
control this behavior through organizational firewall policy, and control this behavior through organizational firewall policy, and
generally, forcing WebRTC traffic through a proxy server will have generally, forcing WebRTC traffic through a proxy server will have
negative effects on both the proxy and on media quality.</t> negative effects on both the proxy and media quality.</t>
<t indent="0" pn="section-3-9">Note also that these concerns predate WebRT
<t>Note also that these concerns predate WebRTC; Adobe Flash Player has C; 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 Media Flow Protocol (RTMFP) support
<xref target="RFC7016" /> in 2008.</t> <xref target="RFC7016" format="default" sectionFormat="of" derivedContent= "RFC7016"/> in 2008.</t>
</section> </section>
<section title="Goals"> <section numbered="true" toc="include" removeInRFC="false" pn="section-4">
<name slugifiedName="name-goals">Goals</name>
<t>WebRTC's support of secure peer-to-peer connections facilitates <t indent="0" pn="section-4-1">WebRTC's support of secure peer-to-peer con
nections facilitates
deployment of decentralized systems, which can have privacy benefits. As deployment of decentralized systems, which can have privacy benefits. As
a result, blunt solutions that disable WebRTC or make it significantly a result, blunt solutions that disable WebRTC or make it significantly
harder to use are undesirable. This document takes a more nuanced harder to use are undesirable. This document takes a more nuanced
approach, with the following goals: approach, with the following goals:
<list style="symbols"> </t>
<ul spacing="normal" bare="false" empty="false" indent="3" pn="section-4-2
<t>Provide a framework for understanding the problem so that controls ">
might be provided to make different tradeoffs regarding performance and <li pn="section-4-2.1">Provide a framework for understanding the problem
privacy concerns with WebRTC.</t> so that controls
might be provided to make different trade-offs regarding performance and
<t>Using that framework, define settings that enable peer-to-peer privacy concerns with WebRTC.</li>
<li pn="section-4-2.2">Using that framework, define settings that enable
peer-to-peer
communications, each with a different balance between performance and communications, each with a different balance between performance and
privacy.</t> privacy.</li>
<li pn="section-4-2.3">Finally, provide recommendations for default sett
<t>Finally, provide recommendations for default settings that provide ings that provide
reasonable performance without also exposing addressing information in reasonable performance without also exposing addressing information in
a way that might violate user expectations.</t> a way that might violate user expectations.</li>
</list></t> </ul>
</section> </section>
<section title="Detailed Design"> <section numbered="true" toc="include" removeInRFC="false" pn="section-5">
<section title="Principles"> <name slugifiedName="name-detailed-design">Detailed Design</name>
<section numbered="true" toc="include" removeInRFC="false" pn="section-5.1
<t>The key principles for our framework are stated below: ">
<list style="numbers"> <name slugifiedName="name-principles">Principles</name>
<t indent="0" pn="section-5.1-1">The key principles for our framework ar
<t>By default, WebRTC traffic should follow typical IP routing, i.e., e stated below:
WebRTC should use the same interface used for HTTP traffic, and only </t>
<ol spacing="normal" type="1" indent="adaptive" start="1" pn="section-5.
1-2">
<li pn="section-5.1-2.1" derivedCounter="1.">By default, WebRTC traffi
c should follow typical IP routing (i.e.,
WebRTC should use the same interface used for HTTP traffic) and only
the system's 'typical' public addresses (or those of an enterprise the system's 'typical' public addresses (or those of an enterprise
TURN server, if present) should be visible to the application. TURN server, if present) should be visible to the application.
However, in the interest of optimal media quality, it should be However, in the interest of optimal media quality, it should be
possible to enable WebRTC to make use of all network interfaces to possible to enable WebRTC to make use of all network interfaces to
determine the ideal route.</t> determine the ideal route.</li>
<li pn="section-5.1-2.2" derivedCounter="2.">By default, WebRTC should
<t>By default, WebRTC should be able to negotiate direct peer-to-peer be able to negotiate direct peer-to-peer
connections between endpoints (i.e., without traversing a NAT or connections between endpoints (i.e., without traversing a NAT or
relay server) when such connections are possible. This ensures that relay server) when such connections are possible. This ensures that
applications that need true peer-to-peer routing for bandwidth or applications that need true peer-to-peer routing for bandwidth or
latency reasons can operate successfully.</t> latency reasons can operate successfully.</li>
<li pn="section-5.1-2.3" derivedCounter="3.">It should be possible to
<t>It should be possible to configure WebRTC to not disclose private 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 require applications learning such addresses. This document does not require
this to be the default state, as there is no currently defined this to be the default state, as there is no currently defined
mechanism that can satisfy this requirement as well as the mechanism that can satisfy this requirement as well as the
aforementioned requirement to allow direct peer-to-peer aforementioned requirement to allow direct peer-to-peer
connections.</t> connections.</li>
<li pn="section-5.1-2.4" derivedCounter="4.">By default, WebRTC traffi
<t>By default, WebRTC traffic should not be sent through proxy c should not be sent through proxy
servers, due to the media quality problems associated with sending servers, due to the media-quality problems associated with sending
WebRTC traffic over TCP, which is almost always used when WebRTC traffic over TCP, which is almost always used when
communicating with such proxies, as well as proxy performance issues communicating with such proxies, as well as proxy performance issues
that may result from proxying WebRTC's long-lived, high-bandwidth that may result from proxying WebRTC's long-lived, high-bandwidth
connections. However, it should be possible to force WebRTC to send connections. However, it should be possible to force WebRTC to send
its traffic through a configured proxy if desired.</t> its traffic through a configured proxy if desired.</li>
</list></t> </ol>
</section> </section>
<section title="Modes and Recommendations"> <section numbered="true" toc="include" removeInRFC="false" pn="section-5.2
">
<t>Based on these ideas, we define four specific modes of WebRTC <name slugifiedName="name-modes-and-recommendations">Modes and Recommend
behavior, reflecting different media quality/privacy tradeoffs: ations</name>
<list style="format Mode %d:"> <t indent="0" pn="section-5.2-1">Based on these ideas, we define four sp
ecific modes of WebRTC
<t>Enumerate all addresses: WebRTC MUST use all network interfaces to behavior, reflecting different media quality/privacy trade-offs:
</t>
<dl newline="true" indent="3" spacing="normal" pn="section-5.2-2">
<dt pn="section-5.2-2.1">Mode 1 - Enumerate all addresses:</dt>
<dd pn="section-5.2-2.2">WebRTC <bcp14>MUST</bcp14> use all network in
terfaces to
attempt communication with STUN servers, TURN servers, or peers. This attempt communication with STUN servers, TURN servers, or peers. This
will converge on the best media path, and is ideal when media will converge on the best media path and is ideal when media
performance is the highest priority, but it discloses the most performance is the highest priority, but it discloses the most
information.</t> information.</dd>
<dt pn="section-5.2-2.3">Mode 2 - Default route + associated local add
<t>Default route + associated local addresses: WebRTC MUST follow the resses:</dt>
<dd pn="section-5.2-2.4">WebRTC
<bcp14>MUST</bcp14> follow the
kernel routing table rules, which will typically cause media packets kernel routing table rules, which will typically cause media packets
to take the same route as the application's HTTP traffic. If an to take the same route as the application's HTTP traffic. If an
enterprise TURN server is present, the preferred route MUST be enterprise TURN server is present, the preferred route <bcp14>MUST</bc p14> be
through this TURN server. Once an interface has been chosen, the through this TURN server. Once an interface has been chosen, the
private IPv4 and IPv6 addresses associated with this interface MUST private IPv4 and IPv6 addresses associated with this interface <bcp14> MUST</bcp14>
be discovered and provided to the application as host candidates. be discovered and provided to the application as host candidates.
This ensures that direct connections can still be established in this This ensures that direct connections can still be established in this
mode.</t> mode.</dd>
<dt pn="section-5.2-2.5">Mode 3 - Default route only: </dt>
<t>Default route only: This is the the same as Mode 2, except that <dd pn="section-5.2-2.6">This is the same as Mode 2, except that
the associated private addresses MUST NOT be provided; the only IP the associated private addresses <bcp14>MUST NOT</bcp14> be provided;
the only IP
addresses gathered are those discovered via mechanisms like STUN and addresses gathered are those discovered via mechanisms like STUN and
TURN (on the default route). This may cause traffic to hairpin TURN (on the default route). This may cause traffic to hairpin
through a NAT, fall back to an application TURN server, or fail through a NAT, fall back to an application TURN server, or fail
altogether, with resulting quality implications.</t> altogether, with resulting quality implications.</dd>
<dt pn="section-5.2-2.7">Mode 4 - Force proxy:</dt>
<t>Force proxy: This is the same as Mode 3, but when the <dd pn="section-5.2-2.8">This is the same as Mode 3, but when the
application's HTTP traffic is sent through a proxy, WebRTC media application's HTTP traffic is sent through a proxy, WebRTC media
traffic MUST also be proxied. If the proxy does not support UDP (as traffic <bcp14>MUST</bcp14> also be proxied. If the proxy does not sup port UDP (as
is the case for all HTTP and most SOCKS is the case for all HTTP and most SOCKS
<xref target="RFC1928" /> proxies), or the WebRTC implementation does <xref target="RFC1928" format="default" sectionFormat="of" derivedCont ent="RFC1928"/> proxies), or the WebRTC implementation does
not support UDP proxying, the use of UDP will be disabled, and TCP 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 be used to send and receive media through the proxy. Use of TCP
will result in reduced media quality, in addition to any performance will result in reduced media quality, in addition to any performance
considerations associated with sending all WebRTC media through the considerations associated with sending all WebRTC media through the
proxy server.</t> proxy server.</dd>
</list></t> </dl>
<t indent="0" pn="section-5.2-3">Mode 1 <bcp14>MUST NOT</bcp14> be used
<t>Mode 1 MUST NOT be used unless user consent has been provided. The 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 <xref target="WEBRTC-SECURITY" />, consent, described in <xref target="RFC8827" format="default" sectionFor
Section 6.2. Alternatively, implementations can provide a specific mat="comma" section="6.2" derivedLink="https://rfc-editor.org/rfc/rfc8827#sectio
n-6.2" derivedContent="RFC8827"/>.
Alternatively, implementations can provide a specific
mechanism to obtain user consent.</t> mechanism to obtain user consent.</t>
<t indent="0" pn="section-5.2-4">In cases where user consent has not bee
<t>In cases where user consent has not been obtained, Mode 2 SHOULD be n obtained, Mode 2 <bcp14>SHOULD</bcp14> be
used.</t> used.</t>
<t indent="0" pn="section-5.2-5">These defaults provide a reasonable tra
<t>These defaults provide a reasonable tradeoff that permits trusted de-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 choose connections, as defined in Mode 2. However, implementations <bcp14>MAY</ bcp14> choose
stricter modes if desired, e.g., if a user indicates they want all stricter modes if desired, e.g., if a user indicates they want all
WebRTC traffic to follow the default route.</t> WebRTC traffic to follow the default route.</t>
<t indent="0" pn="section-5.2-6">Future documents may define additional
<t>Future documents may define additional modes and/or update the modes and/or update the
recommended default modes.</t> recommended default modes.</t>
<t indent="0" pn="section-5.2-7">Note that the suggested defaults can st
<t>Note that the suggested defaults can still be used even for ill be used even for
organizations that want all external WebRTC traffic to traverse a proxy organizations that want all external WebRTC traffic to traverse a proxy
or enterprise TURN server, simply by setting an organizational firewall or enterprise TURN server, simply by setting an organizational firewall
policy that allows WebRTC traffic to only leave through the proxy or policy that allows WebRTC traffic to only leave through the proxy or
TURN server. This provides a way to ensure the proxy or TURN server is TURN server. This provides a way to ensure the proxy or TURN server is
used for any external traffic, but still allows direct connections used for any external traffic but still allows direct connections
(and, in the proxy case, avoids the performance issues associated with (and, in the proxy case, avoids the performance issues associated with
forcing media through said proxy) for intra-organization traffic.</t> forcing media through said proxy) for intra-organization traffic.</t>
</section> </section>
</section> </section>
<section title="Implementation Guidance"> <section numbered="true" toc="include" removeInRFC="false" pn="section-6">
<name slugifiedName="name-implementation-guidance">Implementation Guidance
<t>This section provides guidance to WebRTC implementations on how to </name>
<t indent="0" pn="section-6-1">This section provides guidance to WebRTC im
plementations on how to
implement the policies described above.</t> implement the policies described above.</t>
<section title="Ensuring Normal Routing"> <section numbered="true" toc="include" removeInRFC="false" pn="section-6.1
">
<t>When trying to follow typical IP routing, as required by Modes 2 <name slugifiedName="name-ensuring-normal-routing">Ensuring Normal Routi
ng</name>
<t indent="0" pn="section-6.1-1">When trying to follow typical IP routin
g, as required by Modes 2
and 3, the simplest approach is and 3, the simplest approach is
to bind() the sockets used for peer-to-peer connections to the wildcard to bind() the sockets used for peer-to-peer connections to the wildcard
addresses (0.0.0.0 for IPv4, :: for IPv6), which allows the OS to route 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 WebRTC traffic the same way as it would HTTP traffic. STUN and TURN
will work as usual, and host candidates can still be determined as will work as usual, and host candidates can still be determined as
mentioned below.</t> mentioned below.</t>
</section> </section>
<section title="Determining Associated Local Addresses"> <section numbered="true" toc="include" removeInRFC="false" pn="section-6.2
">
<t>When binding to a wildcard address, some extra work is needed to <name slugifiedName="name-determining-associated-loca">Determining Assoc
iated Local Addresses</name>
<t indent="0" pn="section-6.2-1">When binding to a wildcard address, som
e 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 define as the source
address that would be used for any packets sent to the web application address that would be used for any packets sent to the web application
host (assuming that UDP and TCP get the same routing treatment). Use of host (assuming that UDP and TCP get the same routing treatment). Use of
the web application host as a destination ensures the right source the web-application host as a destination ensures the right source
address is selected, regardless of where the application resides (e.g., address is selected, regardless of where the application resides (e.g.,
on an intranet).</t> on an intranet).</t>
<t indent="0" pn="section-6.2-2">First, the appropriate remote IPv4/IPv6
<t>First, the appropriate remote IPv4/IPv6 address is obtained by address is obtained by
resolving the host component of the web application URI resolving the host component of the web application URI
<xref target="RFC3986" />. If the client is behind a proxy and cannot <xref target="RFC3986" format="default" sectionFormat="of" derivedConten t="RFC3986"/>. If the client is behind a proxy and cannot
resolve these IPs via DNS, the address of the proxy can be used resolve these IPs via DNS, the address of the proxy can be used
instead. Or, if the web application was loaded from a file:// URI instead. Or, if the web application was loaded from a file:// URI
<xref target="RFC8089" />, rather than over the network, the <xref target="RFC8089" format="default" sectionFormat="of" derivedConten t="RFC8089"/> rather than over the network, the
implementation can fall back to a well-known DNS name or IP implementation can fall back to a well-known DNS name or IP
address.</t> address.</t>
<t indent="0" pn="section-6.2-3">Once a suitable remote IP has been dete
<t>Once a suitable remote IP has been determined, the implementation rmined, the implementation
can create a UDP socket, bind() it to the appropriate wildcard address, can create a UDP socket, bind() it to the appropriate wildcard address,
and then connect() to the remote IP. Generally, this results in and then connect() to the remote IP. Generally, this results in
the socket being assigned a local address based on the kernel routing the socket being assigned a local address based on the kernel routing
table, without sending any packets over the network.</t> table, without sending any packets over the network.</t>
<t indent="0" pn="section-6.2-4">Finally, the socket can be queried usin
<t>Finally, the socket can be queried using getsockname() or the g getsockname() or the
equivalent to determine the appropriate local address.</t> equivalent to determine the appropriate local address.</t>
</section> </section>
</section> </section>
<section title="Application Guidance"> <section numbered="true" toc="include" removeInRFC="false" pn="section-7">
<name slugifiedName="name-application-guidance">Application Guidance</name
<t>The recommendations mentioned in this document may cause certain >
<t indent="0" pn="section-7-1">The recommendations mentioned in this docum
ent 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:
<list style="symbols"> </t>
<ul spacing="normal" bare="false" empty="false" indent="3" pn="section-7-2
<t>Applications SHOULD deploy a TURN server with support for both UDP ">
<li pn="section-7-2.1">Applications <bcp14>SHOULD</bcp14> deploy a TURN
server with support for both UDP
and TCP connections to the server. This ensures that connectivity can 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 still be established, even when Mode 3 or 4 is in use, assuming the
TURN server can be reached.</t> TURN server can be reached.</li>
<li pn="section-7-2.2">Applications <bcp14>SHOULD</bcp14> detect when th
<t>Applications SHOULD detect when they don't have access to the full ey don't have access to the full
set of ICE candidates by checking for the presence of host candidates. 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 If no host candidates are present, Mode 3 or 4 is in use; this
knowledge can be useful for diagnostic purposes.</t> knowledge can be useful for diagnostic purposes.</li>
</list></t> </ul>
</section> </section>
<section title="Security Considerations"> <section numbered="true" toc="include" removeInRFC="false" pn="section-8">
<name slugifiedName="name-security-considerations">Security Considerations
<t>This document describes several potential privacy and security concerns </name>
associated with WebRTC peer-to-peer connections, and provides mechanisms <t indent="0" pn="section-8-1">This document describes several potential p
rivacy and security concerns
associated with WebRTC peer-to-peer connections and provides mechanisms
and recommendations for WebRTC implementations to address these concerns. and recommendations for WebRTC implementations to address these concerns.
</t> </t>
</section> </section>
<section title="IANA Considerations"> <section numbered="true" toc="include" removeInRFC="false" pn="section-9">
<name slugifiedName="name-iana-considerations">IANA Considerations</name>
<t>This document requires no actions from IANA.</t> <t indent="0" pn="section-9-1">This document has no IANA actions.</t>
</section>
<section title="Acknowledgements">
<t>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.</t>
</section> </section>
</middle> </middle>
<back> <back>
<references title="Normative References"> <references pn="section-10">
<?rfc include='reference.RFC.2119.xml'?> <name slugifiedName="name-references">References</name>
<?rfc include='reference.RFC.3986.xml'?> <references pn="section-10.1">
<?rfc include='reference.RFC.5389.xml'?> <name slugifiedName="name-normative-references">Normative References</na
<?rfc include='reference.RFC.5766.xml'?> me>
<?rfc include='reference.RFC.8089.xml'?> <reference anchor="RFC2119" target="https://www.rfc-editor.org/info/rfc2
<?rfc include='reference.RFC.8174.xml'?> 119" quoteTitle="true" derivedAnchor="RFC2119">
<?rfc include='reference.RFC.8445.xml'?> <front>
<title>Key words for use in RFCs to Indicate Requirement Levels</tit
le>
<author initials="S." surname="Bradner" fullname="S. Bradner">
<organization showOnFrontPage="true"/>
</author>
<date year="1997" month="March"/>
<abstract>
<t indent="0">In many standards track documents several words are
used to signify the requirements in the specification. These words are often ca
pitalized. This document defines these words as they should be interpreted in IE
TF documents. This document specifies an Internet Best Current Practices for th
e Internet Community, and requests discussion and suggestions for improvements.<
/t>
</abstract>
</front>
<seriesInfo name="BCP" value="14"/>
<seriesInfo name="RFC" value="2119"/>
<seriesInfo name="DOI" value="10.17487/RFC2119"/>
</reference>
<reference anchor="RFC3986" target="https://www.rfc-editor.org/info/rfc3
986" quoteTitle="true" derivedAnchor="RFC3986">
<front>
<title>Uniform Resource Identifier (URI): Generic Syntax</title>
<author initials="T." surname="Berners-Lee" fullname="T. Berners-Lee
">
<organization showOnFrontPage="true"/>
</author>
<author initials="R." surname="Fielding" fullname="R. Fielding">
<organization showOnFrontPage="true"/>
</author>
<author initials="L." surname="Masinter" fullname="L. Masinter">
<organization showOnFrontPage="true"/>
</author>
<date year="2005" month="January"/>
<abstract>
<t indent="0">A Uniform Resource Identifier (URI) is a compact seq
uence of characters that identifies an abstract or physical resource. This spec
ification defines the generic URI syntax and a process for resolving URI referen
ces that might be in relative form, along with guidelines and security considera
tions for the use of URIs on the Internet. The URI syntax defines a grammar tha
t is a superset of all valid URIs, allowing an implementation to parse the commo
n components of a URI reference without knowing the scheme-specific requirements
of every possible identifier. This specification does not define a generative
grammar for URIs; that task is performed by the individual specifications of eac
h URI scheme. [STANDARDS-TRACK]</t>
</abstract>
</front>
<seriesInfo name="STD" value="66"/>
<seriesInfo name="RFC" value="3986"/>
<seriesInfo name="DOI" value="10.17487/RFC3986"/>
</reference>
<reference anchor="RFC5389" target="https://www.rfc-editor.org/info/rfc5
389" quoteTitle="true" derivedAnchor="RFC5389">
<front>
<title>Session Traversal Utilities for NAT (STUN)</title>
<author initials="J." surname="Rosenberg" fullname="J. Rosenberg">
<organization showOnFrontPage="true"/>
</author>
<author initials="R." surname="Mahy" fullname="R. Mahy">
<organization showOnFrontPage="true"/>
</author>
<author initials="P." surname="Matthews" fullname="P. Matthews">
<organization showOnFrontPage="true"/>
</author>
<author initials="D." surname="Wing" fullname="D. Wing">
<organization showOnFrontPage="true"/>
</author>
<date year="2008" month="October"/>
<abstract>
<t indent="0">Session Traversal Utilities for NAT (STUN) is a prot
ocol that serves as a tool for other protocols in dealing with Network Address T
ranslator (NAT) traversal. It can be used by an endpoint to determine the IP ad
dress and port allocated to it by a NAT. It can also be used to check connectiv
ity between two endpoints, and as a keep-alive protocol to maintain NAT bindings
. STUN works with many existing NATs, and does not require any special behavior
from them.</t>
<t indent="0">STUN is not a NAT traversal solution by itself. Rat
her, it is a tool to be used in the context of a NAT traversal solution. This i
s an important change from the previous version of this specification (RFC 3489)
, which presented STUN as a complete solution.</t>
<t indent="0">This document obsoletes RFC 3489. [STANDARDS-TRACK]
</t>
</abstract>
</front>
<seriesInfo name="RFC" value="5389"/>
<seriesInfo name="DOI" value="10.17487/RFC5389"/>
</reference>
<reference anchor="RFC5766" target="https://www.rfc-editor.org/info/rfc5
766" quoteTitle="true" derivedAnchor="RFC5766">
<front>
<title>Traversal Using Relays around NAT (TURN): Relay Extensions to
Session Traversal Utilities for NAT (STUN)</title>
<author initials="R." surname="Mahy" fullname="R. Mahy">
<organization showOnFrontPage="true"/>
</author>
<author initials="P." surname="Matthews" fullname="P. Matthews">
<organization showOnFrontPage="true"/>
</author>
<author initials="J." surname="Rosenberg" fullname="J. Rosenberg">
<organization showOnFrontPage="true"/>
</author>
<date year="2010" month="April"/>
<abstract>
<t indent="0">If a host is located behind a NAT, then in certain s
ituations it can be impossible for that host to communicate directly with other
hosts (peers). In these situations, it is necessary for the host to use the ser
vices of an intermediate node that acts as a communication relay. This specific
ation defines a protocol, called TURN (Traversal Using Relays around NAT), that
allows the host to control the operation of the relay and to exchange packets wi
th its peers using the relay. TURN differs from some other relay control protoc
ols in that it allows a client to communicate with multiple peers using a single
relay address. [STANDARDS-TRACK]</t>
</abstract>
</front>
<seriesInfo name="RFC" value="5766"/>
<seriesInfo name="DOI" value="10.17487/RFC5766"/>
</reference>
<reference anchor="RFC8089" target="https://www.rfc-editor.org/info/rfc8
089" quoteTitle="true" derivedAnchor="RFC8089">
<front>
<title>The "file" URI Scheme</title>
<author initials="M." surname="Kerwin" fullname="M. Kerwin">
<organization showOnFrontPage="true"/>
</author>
<date year="2017" month="February"/>
<abstract>
<t indent="0">This document provides a more complete specification
of the "file" Uniform Resource Identifier (URI) scheme and replaces the very br
ief definition in Section 3.10 of RFC 1738.</t>
<t indent="0">It defines a common syntax that is intended to inter
operate across the broad spectrum of existing usages. At the same time, it note
s some other current practices around the use of file URIs.</t>
</abstract>
</front>
<seriesInfo name="RFC" value="8089"/>
<seriesInfo name="DOI" value="10.17487/RFC8089"/>
</reference>
<reference anchor="RFC8174" target="https://www.rfc-editor.org/info/rfc8
174" quoteTitle="true" derivedAnchor="RFC8174">
<front>
<title>Ambiguity of Uppercase vs Lowercase in RFC 2119 Key Words</ti
tle>
<author initials="B." surname="Leiba" fullname="B. Leiba">
<organization showOnFrontPage="true"/>
</author>
<date year="2017" month="May"/>
<abstract>
<t indent="0">RFC 2119 specifies common key words that may be used
in protocol specifications. This document aims to reduce the ambiguity by cla
rifying that only UPPERCASE usage of the key words have the defined special mea
nings.</t>
</abstract>
</front>
<seriesInfo name="BCP" value="14"/>
<seriesInfo name="RFC" value="8174"/>
<seriesInfo name="DOI" value="10.17487/RFC8174"/>
</reference>
<reference anchor="RFC8445" target="https://www.rfc-editor.org/info/rfc8
445" quoteTitle="true" derivedAnchor="RFC8445">
<front>
<title>Interactive Connectivity Establishment (ICE): A Protocol for
Network Address Translator (NAT) Traversal</title>
<author initials="A." surname="Keranen" fullname="A. Keranen">
<organization showOnFrontPage="true"/>
</author>
<author initials="C." surname="Holmberg" fullname="C. Holmberg">
<organization showOnFrontPage="true"/>
</author>
<author initials="J." surname="Rosenberg" fullname="J. Rosenberg">
<organization showOnFrontPage="true"/>
</author>
<date year="2018" month="July"/>
<abstract>
<t indent="0">This document describes a protocol for Network Addre
ss Translator (NAT) traversal for UDP-based communication. This protocol is cal
led Interactive Connectivity Establishment (ICE). ICE makes use of the Session
Traversal Utilities for NAT (STUN) protocol and its extension, Traversal Using R
elay NAT (TURN).</t>
<t indent="0">This document obsoletes RFC 5245.</t>
</abstract>
</front>
<seriesInfo name="RFC" value="8445"/>
<seriesInfo name="DOI" value="10.17487/RFC8445"/>
</reference>
</references> </references>
<references title="Informative References"> <references pn="section-10.2">
<?rfc include='reference.RFC.1918.xml'?> <name slugifiedName="name-informative-references">Informative References
<?rfc include='reference.RFC.1919.xml'?> </name>
<?rfc include='reference.RFC.1928.xml'?> <reference anchor="RFC1918" target="https://www.rfc-editor.org/info/rfc1
<?rfc include='reference.RFC.4941.xml'?> 918" quoteTitle="true" derivedAnchor="RFC1918">
<?rfc include='reference.RFC.6146.xml'?> <front>
<?rfc include='reference.RFC.7016.xml'?> <title>Address Allocation for Private Internets</title>
<?rfc include='reference.RFC.7230.xml'?> <author initials="Y." surname="Rekhter" fullname="Y. Rekhter">
<?rfc include='reference.RFC.7478.xml'?> <organization showOnFrontPage="true"/>
</author>
<!-- <?rfc include='reference.I-D.ietf-rtcweb-security-arch'?>; In MISSREF as of <author initials="B." surname="Moskowitz" fullname="B. Moskowitz">
7/26/19 --> <organization showOnFrontPage="true"/>
</author>
<reference anchor='WEBRTC-SECURITY'> <author initials="D." surname="Karrenberg" fullname="D. Karrenberg">
<front> <organization showOnFrontPage="true"/>
<title>WebRTC Security Architecture</title> </author>
<author initials="G. J." surname="de Groot" fullname="G. J. de Groot
<author initials='E' surname='Rescorla' fullname='Eric Rescorla'> ">
<organization /> <organization showOnFrontPage="true"/>
</author> </author>
<author initials="E." surname="Lear" fullname="E. Lear">
<date month='July' day='22' year='2019' /> <organization showOnFrontPage="true"/>
</author>
<abstract><t>This document defines the security architecture for WebRTC, a proto <date year="1996" month="February"/>
col suite intended for use with real-time applications that can be deployed in b <abstract>
rowsers - "real time communication on the Web".</t></abstract> <t indent="0">This document describes address allocation for priva
te internets. This document specifies an Internet Best Current Practices for th
</front> e Internet Community, and requests discussion and suggestions for improvements.<
/t>
<seriesInfo name='Work in Progress,' value='draft-ietf-rtcweb-security-arch-20' </abstract>
/> </front>
</reference> <seriesInfo name="BCP" value="5"/>
<seriesInfo name="RFC" value="1918"/>
<!-- <?rfc include='reference.I-D.ietf-rtcweb-transports'?>; In MISSREF as of 7/ <seriesInfo name="DOI" value="10.17487/RFC1918"/>
26/19 --> </reference>
<reference anchor="RFC1919" target="https://www.rfc-editor.org/info/rfc1
<reference anchor='WEBRTC-TRANSPORTS'> 919" quoteTitle="true" derivedAnchor="RFC1919">
<front> <front>
<title>Transports for WebRTC</title> <title>Classical versus Transparent IP Proxies</title>
<author initials="M." surname="Chatel" fullname="M. Chatel">
<author initials='H' surname='Alvestrand' fullname='Harald Alvestrand'> <organization showOnFrontPage="true"/>
<organization /> </author>
</author> <date year="1996" month="March"/>
<abstract>
<date month='October' day='26' year='2016' /> <t indent="0">This document explains "classical" and "transparent"
proxy techniques and attempts to provide rules to help determine when each prox
<abstract><t>This document describes the data transport protocols used by WebRTC y system may be used without causing problems. This memo provides information f
, including the protocols used for interaction with intermediate boxes such as f or the Internet community. This memo does not specify an Internet standard of a
irewalls, relays and NAT boxes.</t></abstract> ny kind.</t>
</abstract>
</front> </front>
<seriesInfo name="RFC" value="1919"/>
<seriesInfo name='Work in Progress,' value='draft-ietf-rtcweb-transports-17' /> <seriesInfo name="DOI" value="10.17487/RFC1919"/>
</reference> </reference>
<reference anchor="RFC1928" target="https://www.rfc-editor.org/info/rfc1
928" quoteTitle="true" derivedAnchor="RFC1928">
<front>
<title>SOCKS Protocol Version 5</title>
<author initials="M." surname="Leech" fullname="M. Leech">
<organization showOnFrontPage="true"/>
</author>
<author initials="M." surname="Ganis" fullname="M. Ganis">
<organization showOnFrontPage="true"/>
</author>
<author initials="Y." surname="Lee" fullname="Y. Lee">
<organization showOnFrontPage="true"/>
</author>
<author initials="R." surname="Kuris" fullname="R. Kuris">
<organization showOnFrontPage="true"/>
</author>
<author initials="D." surname="Koblas" fullname="D. Koblas">
<organization showOnFrontPage="true"/>
</author>
<author initials="L." surname="Jones" fullname="L. Jones">
<organization showOnFrontPage="true"/>
</author>
<date year="1996" month="March"/>
<abstract>
<t indent="0">This memo describes a protocol that is an evolution
of the previous version of the protocol, version 4 [1]. This new protocol stems
from active discussions and prototype implementations. [STANDARDS-TRACK]</t>
</abstract>
</front>
<seriesInfo name="RFC" value="1928"/>
<seriesInfo name="DOI" value="10.17487/RFC1928"/>
</reference>
<reference anchor="RFC4941" target="https://www.rfc-editor.org/info/rfc4
941" quoteTitle="true" derivedAnchor="RFC4941">
<front>
<title>Privacy Extensions for Stateless Address Autoconfiguration in
IPv6</title>
<author initials="T." surname="Narten" fullname="T. Narten">
<organization showOnFrontPage="true"/>
</author>
<author initials="R." surname="Draves" fullname="R. Draves">
<organization showOnFrontPage="true"/>
</author>
<author initials="S." surname="Krishnan" fullname="S. Krishnan">
<organization showOnFrontPage="true"/>
</author>
<date year="2007" month="September"/>
<abstract>
<t indent="0">Nodes use IPv6 stateless address autoconfiguration t
o generate addresses using a combination of locally available information and in
formation advertised by routers. Addresses are formed by combining network pref
ixes with an interface identifier. On an interface that contains an embedded IE
EE Identifier, the interface identifier is typically derived from it. On other
interface types, the interface identifier is generated through other means, for
example, via random number generation. This document describes an extension to
IPv6 stateless address autoconfiguration for interfaces whose interface identifi
er is derived from an IEEE identifier. Use of the extension causes nodes to gen
erate global scope addresses from interface identifiers that change over time, e
ven in cases where the interface contains an embedded IEEE identifier. Changing
the interface identifier (and the global scope addresses generated from it) ove
r time makes it more difficult for eavesdroppers and other information collector
s to identify when different addresses used in different transactions actually c
orrespond to the same node. [STANDARDS-TRACK]</t>
</abstract>
</front>
<seriesInfo name="RFC" value="4941"/>
<seriesInfo name="DOI" value="10.17487/RFC4941"/>
</reference>
<reference anchor="RFC6146" target="https://www.rfc-editor.org/info/rfc6
146" quoteTitle="true" derivedAnchor="RFC6146">
<front>
<title>Stateful NAT64: Network Address and Protocol Translation from
IPv6 Clients to IPv4 Servers</title>
<author initials="M." surname="Bagnulo" fullname="M. Bagnulo">
<organization showOnFrontPage="true"/>
</author>
<author initials="P." surname="Matthews" fullname="P. Matthews">
<organization showOnFrontPage="true"/>
</author>
<author initials="I." surname="van Beijnum" fullname="I. van Beijnum
">
<organization showOnFrontPage="true"/>
</author>
<date year="2011" month="April"/>
</front>
<seriesInfo name="RFC" value="6146"/>
<seriesInfo name="DOI" value="10.17487/RFC6146"/>
</reference>
<reference anchor="RFC7016" target="https://www.rfc-editor.org/info/rfc7
016" quoteTitle="true" derivedAnchor="RFC7016">
<front>
<title>Adobe's Secure Real-Time Media Flow Protocol</title>
<author initials="M." surname="Thornburgh" fullname="M. Thornburgh">
<organization showOnFrontPage="true"/>
</author>
<date year="2013" month="November"/>
<abstract>
<t indent="0">This memo describes Adobe's Secure Real-Time Media F
low Protocol (RTMFP), an endpoint-to-endpoint communication protocol designed to
securely transport parallel flows of real-time video, audio, and data messages,
as well as bulk data, over IP networks. RTMFP has features that make it effect
ive for peer-to-peer (P2P) as well as client-server communications, even when Ne
twork Address Translators (NATs) are used.</t>
</abstract>
</front>
<seriesInfo name="RFC" value="7016"/>
<seriesInfo name="DOI" value="10.17487/RFC7016"/>
</reference>
<reference anchor="RFC7230" target="https://www.rfc-editor.org/info/rfc7
230" quoteTitle="true" derivedAnchor="RFC7230">
<front>
<title>Hypertext Transfer Protocol (HTTP/1.1): Message Syntax and Ro
uting</title>
<author initials="R." surname="Fielding" fullname="R. Fielding" role
="editor">
<organization showOnFrontPage="true"/>
</author>
<author initials="J." surname="Reschke" fullname="J. Reschke" role="
editor">
<organization showOnFrontPage="true"/>
</author>
<date year="2014" month="June"/>
<abstract>
<t indent="0">The Hypertext Transfer Protocol (HTTP) is a stateles
s application-level protocol for distributed, collaborative, hypertext informati
on systems. This document provides an overview of HTTP architecture and its ass
ociated terminology, defines the "http" and "https" Uniform Resource Identifier
(URI) schemes, defines the HTTP/1.1 message syntax and parsing requirements, and
describes related security concerns for implementations.</t>
</abstract>
</front>
<seriesInfo name="RFC" value="7230"/>
<seriesInfo name="DOI" value="10.17487/RFC7230"/>
</reference>
<reference anchor="RFC7478" target="https://www.rfc-editor.org/info/rfc7
478" quoteTitle="true" derivedAnchor="RFC7478">
<front>
<title>Web Real-Time Communication Use Cases and Requirements</title
>
<author initials="C." surname="Holmberg" fullname="C. Holmberg">
<organization showOnFrontPage="true"/>
</author>
<author initials="S." surname="Hakansson" fullname="S. Hakansson">
<organization showOnFrontPage="true"/>
</author>
<author initials="G." surname="Eriksson" fullname="G. Eriksson">
<organization showOnFrontPage="true"/>
</author>
<date year="2015" month="March"/>
<abstract>
<t indent="0">This document describes web-based real-time communic
ation use cases. Requirements on the browser functionality are derived from the
use cases.</t>
<t indent="0">This document was developed in an initial phase of t
he work with rather minor updates at later stages. It has not really served as
a tool in deciding features or scope for the WG's efforts so far. It is being p
ublished to record the early conclusions of the WG. It will not be used as a se
t of rigid guidelines that specifications and implementations will be held to in
the future.</t>
</abstract>
</front>
<seriesInfo name="RFC" value="7478"/>
<seriesInfo name="DOI" value="10.17487/RFC7478"/>
</reference>
<reference anchor="RFC8827" target="https://www.rfc-editor.org/info/rfc8
827" quoteTitle="true" derivedAnchor="RFC8827">
<front>
<title>WebRTC Security Architecture</title>
<author initials="E." surname="Rescorla" fullname="Eric Rescorla">
<organization showOnFrontPage="true"/>
</author>
<date month="January" year="2021"/>
</front>
<seriesInfo name="RFC" value="8827"/>
<seriesInfo name="DOI" value="10.17487/RFC8827"/>
</reference>
<reference anchor="RFC8835" target="https://www.rfc-editor.org/info/rfc8
835" quoteTitle="true" derivedAnchor="RFC8835">
<front>
<title>Transports for WebRTC</title>
<author initials="H." surname="Alvestrand" fullname="Harald Alvestra
nd">
<organization showOnFrontPage="true"/>
</author>
<date month="January" year="2021"/>
</front>
<seriesInfo name="RFC" value="8835"/>
<seriesInfo name="DOI" value="10.17487/RFC8835"/>
</reference>
</references> </references>
<section title="Change log"> </references>
<t>Changes in draft -12: <section numbered="false" toc="include" removeInRFC="false" pn="section-appe
<list style="symbols"> ndix.a">
<name slugifiedName="name-acknowledgements">Acknowledgements</name>
<t>Editorial updates from IETF LC review.</t> <t indent="0" pn="section-appendix.a-1">Several people provided input into
</list></t> this document, including <contact fullname="Bernard Aboba"/>, <contact fu
llname="Harald Alvestrand"/>, <contact fullname="Youenn Fablet"/>, <contac
<t>Changes in draft -11: t fullname="Ted Hardie"/>, <contact fullname="Matthew Kaufmann"/>,
<list style="symbols"> <contact fullname="Eric Rescorla"/>, <contact fullname="Adam Roach"/>, and
<contact fullname="Martin Thomson"/>.</t>
<t>Editorial updates from AD review.</t> </section>
</list></t> <section anchor="authors-addresses" numbered="false" removeInRFC="false" toc
="include" pn="section-appendix.b">
<t>Changes in draft -10: <name slugifiedName="name-authors-addresses">Authors' Addresses</name>
<list style="symbols"> <author fullname="Justin Uberti" initials="J." surname="Uberti">
<organization showOnFrontPage="true">Google</organization>
<t>Incorporate feedback from IETF 102 on the problem space.</t> <address>
<postal>
<t>Note that future versions of the document may define new modes.</t> <street>747 6th St S</street>
</list></t> <city>Kirkland</city>
<region>WA</region>
<t>Changes in draft -09: <code>98033</code>
<list style="symbols"> <country>United States of America</country>
</postal>
<t>Fixed confusing text regarding enterprise TURN servers.</t> <email>justin@uberti.name</email>
</list></t> </address>
</author>
<t>Changes in draft -08: <author fullname="Guo-wei Shieh" initials="G." surname="Shieh">
<list style="symbols"> <organization showOnFrontPage="true"/>
<address>
<t>Discuss how enterprise TURN servers should be handled.</t> <postal>
</list></t> <street>333 Elliott Ave W #500</street>
<city>Seattle</city>
<t>Changes in draft -07: <region>WA</region>
<list style="symbols"> <code>98119</code>
<country>United States of America</country>
<t>Clarify consent guidance.</t> </postal>
</list></t> <email>guoweis@gmail.com</email>
</address>
<t>Changes in draft -06: </author>
<list style="symbols">
<t>Clarify recommendations.</t>
<t>Split implementation guidance into two sections.</t>
</list></t>
<t>Changes in draft -05:
<list style="symbols">
<t>Separated framework definition from implementation techniques.</t>
<t>Removed RETURN references.</t>
<t>Use origin when determining local IPs, rather than a well-known
IP.</t>
</list></t>
<t>Changes in draft -04:
<list style="symbols">
<t>Rewording and cleanup in abstract, intro, and problem statement.</t>
<t>Added 2119 boilerplate.</t>
<t>Fixed weird reference spacing.</t>
<t>Expanded acronyms on first use.</t>
<t>Removed 8.8.8.8 mention.</t>
<t>Removed mention of future browser considerations.</t>
</list></t>
<t>Changes in draft -03:
<list style="symbols">
<t>Clarified when to use which modes.</t>
<t>Added 2119 qualifiers to make normative statements.</t>
<t>Defined 'proxy'.</t>
<t>Mentioned split tunnels in problem statement.</t>
</list></t>
<t>Changes in draft -02:
<list style="symbols">
<t>Recommendations -&gt; Requirements</t>
<t>Updated text regarding consent.</t>
</list></t>
<t>Changes in draft -01:
<list style="symbols">
<t>Incorporated feedback from Adam Roach; changes to discussion of
cam/mic permission, as well as use of proxies, and various editorial
changes.</t>
<t>Added several more references.</t>
</list></t>
<t>Changes in draft -00:
<list style="symbols">
<t>Published as WG draft.</t>
</list></t>
</section> </section>
</back> </back>
</rfc> </rfc>
 End of changes. 71 change blocks. 
384 lines changed or deleted 905 lines changed or added

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