rfc8828xml2.original.xml   rfc8828.xml 
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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</title> Requirements</title>
<seriesInfo name="RFC" value="8828"/>
<author fullname="Justin Uberti" initials="J." surname="Uberti"> <author fullname="Justin Uberti" initials="J." surname="Uberti">
<organization>Google</organization> <organization>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" /> <date month="July" year="2020"/>
<area>RAI</area> <area>RAI</area>
<!-- [rfced] Please insert any keywords (beyond those that appear in <!-- [rfced] Please insert any keywords (beyond those that appear in
the title) for use on https://www.rfc-editor.org/search. --> the title) for use on https://www.rfc-editor.org/search. -->
<keyword>example</keyword>
<keyword>example</keyword>
<abstract> <abstract>
<t>This document provides information and requirements for how IP <t>This document provides information and requirements for how IP
addresses should be handled by WebRTC implementations.</t> addresses should be handled by Web Real-Time Communication (WebRTC) implem entations.</t>
</abstract> </abstract>
</front> </front>
<middle> <middle>
<section title="Introduction"> <section numbered="true" toc="default">
<name>Introduction</name>
<t>One of WebRTC's key features is its support of peer-to-peer <t>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"/>. This may be problematic in
document summarizes the concerns, and makes recommendations on how WebRTC certain cases. This
implementations should best handle the tradeoff between privacy and media document summarizes the concerns and makes recommendations on how WebRTC
performance.</t> implementations should best handle the trade-off between privacy and medi
</section> a
<section title="Terminology"> performance.</t>
</section>
<section numbered="true" toc="default">
<name>Terminology</name>
<t>
The key words "<bcp14>MUST</bcp14>", "<bcp14>MUST NOT</bcp14>",
"<bcp14>REQUIRED</bcp14>", "<bcp14>SHALL</bcp14>", "<bcp14>SHALL NOT</bcp14>
",
"<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"/> <xref
target="RFC8174"/> when, and only when, they appear in all capitals, as
shown here.
</t>
<t>The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
"OPTIONAL" in this document are to be interpreted as described in BCP 14
<xref target="RFC2119"></xref><xref target="RFC8174"></xref>
when, and only when, they appear in all capitals, as shown here.</t>
</section> </section>
<section title="Problem Statement"> <section numbered="true" toc="default">
<name>Problem Statement</name>
<t>In order to establish a peer-to-peer connection, WebRTC <t>In order to establish a peer-to-peer connection, 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"/>, which attempts to discover mult iple 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"/> and Traversal Using Relays
<xref target="RFC5766" />, and then checks the connectivity of each around NAT (TURN)
<xref target="RFC5766" format="default"/>. It 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>These addresses are provided to the web application so that <t>These addresses are provided to the web application 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>The information revealed falls into three categories: <t>The information revealed falls into one of three categories:
<list style="numbers"> </t>
<ol spacing="normal" type="1">
<t>If the client is multihomed, additional public IP addresses for the <li>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>If the client is behind a Network Address Translator (NAT), the
<t>If the client is behind a Network 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"/> addresses, can be learned.</li
>
<t>If the client is behind a proxy (a client-configured "classical <li>If the client is behind a proxy (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"/>), 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"/> 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"/> proxy server.</li>
</list></t> </ol>
<t>Of these three concerns, the first is the most significant, because for some <t>Of these three concerns, the first is the most 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>The second concern is less significant but valid nonetheless. The core <t>The second concern is less significant but 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"/>.
recommended by <xref target="WEBRTC-TRANSPORTS" /> is fairly While disclosure of the <xref target="RFC4941" format="default"/> IPv6 add
resses
recommended by <xref target="RFC8835"
format="default"/> 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>Private IP addresses can also act as an identifier that allows <t>Private IP addresses can also act as an identifier 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>The third concern is the least common, as proxy administrators can alre ady <t>The third concern is the least common, as proxy administrators can alre 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>Note also that these concerns predate WebRTC; Adobe Flash Player has <t>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 Media Flow Protocol (RTMFP) support
<xref target="RFC7016" /> in 2008.</t> <xref target="RFC7016" format="default"/> in 2008.</t>
</section> </section>
<section title="Goals"> <section numbered="true" toc="default">
<name>Goals</name>
<t>WebRTC's support of secure peer-to-peer connections facilitates <t>WebRTC's support of secure peer-to-peer connections 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">
<t>Provide a framework for understanding the problem so that controls <li>Provide a framework for understanding the problem so that controls
might be provided to make different tradeoffs regarding performance and might be provided to make different trade-offs regarding performance and
privacy concerns with WebRTC.</t> privacy concerns with WebRTC.</li>
<li>Using that framework, define settings that enable peer-to-peer
<t>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>Finally, provide recommendations for default settings that provide
<t>Finally, provide recommendations for default settings 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="default">
<section title="Principles"> <name>Detailed Design</name>
<section numbered="true" toc="default">
<name>Principles</name>
<t>The key principles for our framework are stated below: <t>The key principles for our framework are stated below:
<list style="numbers"> </t>
<ol spacing="normal" type="1">
<t>By default, WebRTC traffic should follow typical IP routing, i.e., <li>By default, WebRTC traffic should follow typical IP routing (i.e.,
WebRTC should use the same interface used for HTTP traffic, and only 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>By default, WebRTC should be able to negotiate direct peer-to-peer
<t>By default, WebRTC should 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>It should be possible to configure WebRTC to not disclose private
<t>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 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>By default, WebRTC traffic should not be sent through proxy
<t>By default, WebRTC traffic 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="default">
<name>Modes and Recommendations</name>
<t>Based on these ideas, we define four specific modes of WebRTC <t>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:
<list style="format Mode %d:"> </t>
<dl newline="true">
<t>Enumerate all addresses: WebRTC MUST use all network interfaces to <dt>Mode 1 - Enumerate all addresses:</dt>
<dd>WebRTC <bcp14>MUST</bcp14> use all network interfaces 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>Mode 2 - Default route + associated local addresses:</dt>
<t>Default route + associated local addresses: WebRTC MUST follow the <dd>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>Mode 3 - Default route only: </dt>
<t>Default route only: This is the the same as Mode 2, except that <dd>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>Mode 4 - Force proxy:</dt>
<t>Force proxy: This is the same as Mode 3, but when the <dd>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"/> proxies), or the WebRTC impl ementation 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>Mode 1 <bcp14>MUST NOT</bcp14> be used unless user consent has been p
<t>Mode 1 MUST NOT be used unless user consent has been provided. The rovided. 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"
Section 6.2. Alternatively, implementations can provide a specific format="default" sectionFormat="comma" section="6.2"/>.
Alternatively, implementations can provide a specific
mechanism to obtain user consent.</t> mechanism to obtain user consent.</t>
<t>In cases where user consent has not been obtained, Mode 2 <bcp14>SHOU
<t>In cases where user consent has not been obtained, Mode 2 SHOULD be LD</bcp14> be
used.</t> used.</t>
<t>These defaults provide a reasonable trade-off that permits trusted
<t>These defaults provide a reasonable tradeoff 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>Future documents may define additional modes and/or update the <t>Future documents may define additional modes and/or update the
recommended default modes.</t> recommended default modes.</t>
<t>Note that the suggested defaults can still be used even for <t>Note that the suggested defaults can still 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="default">
<name>Implementation Guidance</name>
<t>This section provides guidance to WebRTC implementations on how to <t>This section provides guidance to WebRTC implementations on how to
implement the policies described above.</t> implement the policies described above.</t>
<section title="Ensuring Normal Routing"> <section numbered="true" toc="default">
<name>Ensuring Normal Routing</name>
<t>When trying to follow typical IP routing, as required by Modes 2 <t>When trying to follow typical IP routing, 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="default">
<name>Determining Associated Local Addresses</name>
<t>When binding to a wildcard address, some extra work is needed to <t>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 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>First, the appropriate remote IPv4/IPv6 address is obtained by <t>First, the appropriate remote IPv4/IPv6 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"/>. If the client is behind a pro xy 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"/> 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>Once a suitable remote IP has been determined, the implementation <t>Once a suitable remote IP has been determined, 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>Finally, the socket can be queried using getsockname() or the <t>Finally, the socket can be queried using 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="default">
<name>Application Guidance</name>
<t>The recommendations mentioned in this document may cause certain <t>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:
<list style="symbols"> </t>
<ul spacing="normal">
<t>Applications SHOULD deploy a TURN server with support for both UDP <li>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>Applications <bcp14>SHOULD</bcp14> detect when they don't have acces
<t>Applications SHOULD detect when they don't have access to the full s 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="default">
<name>Security Considerations</name>
<t>This document describes several potential privacy and security concerns <t>This document describes several potential privacy and security concerns
associated with WebRTC peer-to-peer connections, and provides mechanisms 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="default">
<name>IANA Considerations</name>
<t>This document requires no actions from IANA.</t> <t>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">
<?rfc include='reference.RFC.2119.xml'?>
<?rfc include='reference.RFC.3986.xml'?>
<?rfc include='reference.RFC.5389.xml'?>
<?rfc include='reference.RFC.5766.xml'?>
<?rfc include='reference.RFC.8089.xml'?>
<?rfc include='reference.RFC.8174.xml'?>
<?rfc include='reference.RFC.8445.xml'?>
</references>
<references title="Informative References">
<?rfc include='reference.RFC.1918.xml'?>
<?rfc include='reference.RFC.1919.xml'?>
<?rfc include='reference.RFC.1928.xml'?>
<?rfc include='reference.RFC.4941.xml'?>
<?rfc include='reference.RFC.6146.xml'?>
<?rfc include='reference.RFC.7016.xml'?>
<?rfc include='reference.RFC.7230.xml'?>
<?rfc include='reference.RFC.7478.xml'?>
<!-- <?rfc include='reference.I-D.ietf-rtcweb-security-arch'?>; In MISSREF as of
7/26/19 -->
<reference anchor='WEBRTC-SECURITY'>
<front>
<title>WebRTC Security Architecture</title>
<author initials='E' surname='Rescorla' fullname='Eric Rescorla'>
<organization />
</author>
<date month='July' day='22' year='2019' /> <references>
<name>References</name>
<abstract><t>This document defines the security architecture for WebRTC, a proto <references>
col suite intended for use with real-time applications that can be deployed in b <name>Normative References</name>
rowsers - "real time communication on the Web".</t></abstract> <xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer
ence.RFC.2119.xml"/>
</front> <xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer
ence.RFC.3986.xml"/>
<seriesInfo name='Work in Progress,' value='draft-ietf-rtcweb-security-arch-20' <xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer
/> ence.RFC.5389.xml"/>
</reference> <xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer
ence.RFC.5766.xml"/>
<!-- <?rfc include='reference.I-D.ietf-rtcweb-transports'?>; In MISSREF as of 7/ <xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer
26/19 --> ence.RFC.8089.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer
ence.RFC.8174.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer
ence.RFC.8445.xml"/>
</references>
<references>
<name>Informative References</name>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer
ence.RFC.1918.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer
ence.RFC.1919.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer
ence.RFC.1928.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer
ence.RFC.4941.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer
ence.RFC.6146.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer
ence.RFC.7016.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer
ence.RFC.7230.xml"/>
<xi:include href="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/refer
ence.RFC.7478.xml"/>
<reference anchor='WEBRTC-TRANSPORTS'> <!--draft-ietf-rtcweb-security-arch: 8827 -->
<front> <reference anchor="RFC8827" target="https://www.rfc-editor.org/info/rfc8827">
<title>Transports for WebRTC</title> <front>
<title>WebRTC Security Architecture</title>
<author initials='E.' surname='Rescorla' fullname='Eric Rescorla'>
<organization/>
</author>
<date month='July' year='2020'/>
</front>
<seriesInfo name="RFC" value="8827"/>
<seriesInfo name="DOI" value="10.17487/RFC8827"/>
</reference>
<author initials='H' surname='Alvestrand' fullname='Harald Alvestrand'> <!-- draft-ietf-rtcweb-transports-17: 8835 -->
<organization /> <reference anchor="RFC8835" target="https://www.rfc-editor.org/info/rfc8835">
</author>
<date month='October' day='26' year='2016' /> <front>
<title>Transports for WebRTC</title>
<abstract><t>This document describes the data transport protocols used by WebRTC <author initials="H." surname="Alvestrand" fullname="Harald Alvestrand">
, including the protocols used for interaction with intermediate boxes such as f <organization />
irewalls, relays and NAT boxes.</t></abstract> </author>
</front> <date month="July" year="2020" />
</front>
<seriesInfo name="RFC" value="8835" />
<seriesInfo name="DOI" value="10.17487/RFC8835"/>
<seriesInfo name='Work in Progress,' value='draft-ietf-rtcweb-transports-17' />
</reference> </reference>
</references> </references>
<section title="Change log"> </references>
<t>Changes in draft -12: <section numbered="false" toc="default">
<list style="symbols"> <name>Acknowledgements</name>
<t>Several people provided input into this document, including <contact fu
<t>Editorial updates from IETF LC review.</t> llname="Bernard
</list></t> Aboba"/>, <contact fullname="Harald Alvestrand"/>, <contact fullname="Youe
nn
<t>Changes in draft -11: Fablet"/>, <contact fullname="Ted Hardie"/>, <contact fullname="Matthew Ka
<list style="symbols"> ufmann"/>,
<contact fullname="Eric Rescorla"/>, <contact fullname="Adam Roach"/>, and
<t>Editorial updates from AD review.</t> <contact fullname="Martin Thomson"/>.</t>
</list></t>
<t>Changes in draft -10:
<list style="symbols">
<t>Incorporate feedback from IETF 102 on the problem space.</t>
<t>Note that future versions of the document may define new modes.</t>
</list></t>
<t>Changes in draft -09:
<list style="symbols">
<t>Fixed confusing text regarding enterprise TURN servers.</t>
</list></t>
<t>Changes in draft -08:
<list style="symbols">
<t>Discuss how enterprise TURN servers should be handled.</t>
</list></t>
<t>Changes in draft -07:
<list style="symbols">
<t>Clarify consent guidance.</t>
</list></t>
<t>Changes in draft -06:
<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. 84 change blocks. 
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