draft-ietf-rtcweb-ip-handling-11.txt   draft-ietf-rtcweb-ip-handling-12.txt 
Network Working Group J. Uberti Network Working Group J. Uberti
Internet-Draft Google Internet-Draft Google
Intended status: Standards Track November 2, 2018 Intended status: Standards Track July 2, 2019
Expires: May 6, 2019 Expires: January 3, 2020
WebRTC IP Address Handling Requirements WebRTC IP Address Handling Requirements
draft-ietf-rtcweb-ip-handling-11 draft-ietf-rtcweb-ip-handling-12
Abstract Abstract
This document provides information and requirements for how IP This document provides information and requirements for how IP
addresses should be handled by WebRTC implementations. addresses should be handled by WebRTC implementations.
Status of This Memo Status of This Memo
This Internet-Draft is submitted in full conformance with the This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79. provisions of BCP 78 and BCP 79.
skipping to change at page 1, line 31 skipping to change at page 1, line 31
Internet-Drafts are working documents of the Internet Engineering Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet- working documents as Internet-Drafts. The list of current Internet-
Drafts is at https://datatracker.ietf.org/drafts/current/. Drafts is at https://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress." material or to cite them other than as "work in progress."
This Internet-Draft will expire on May 6, 2019. This Internet-Draft will expire on January 3, 2020.
Copyright Notice Copyright Notice
Copyright (c) 2018 IETF Trust and the persons identified as the Copyright (c) 2019 IETF Trust and the persons identified as the
document authors. All rights reserved. document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents Provisions Relating to IETF Documents
(https://trustee.ietf.org/license-info) in effect on the date of (https://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License. described in the Simplified BSD License.
Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 2 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 2
3. Problem Statement . . . . . . . . . . . . . . . . . . . . . . 2 3. Problem Statement . . . . . . . . . . . . . . . . . . . . . . 2
4. Goals . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4 4. Goals . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4
5. Detailed Design . . . . . . . . . . . . . . . . . . . . . . . 4 5. Detailed Design . . . . . . . . . . . . . . . . . . . . . . . 5
5.1. Principles . . . . . . . . . . . . . . . . . . . . . . . 5 5.1. Principles . . . . . . . . . . . . . . . . . . . . . . . 5
5.2. Modes and Recommendations . . . . . . . . . . . . . . . . 5 5.2. Modes and Recommendations . . . . . . . . . . . . . . . . 5
6. Implementation Guidance . . . . . . . . . . . . . . . . . . . 7 6. Implementation Guidance . . . . . . . . . . . . . . . . . . . 7
6.1. Ensuring Normal Routing . . . . . . . . . . . . . . . . . 7 6.1. Ensuring Normal Routing . . . . . . . . . . . . . . . . . 7
6.2. Determining Host Candidates . . . . . . . . . . . . . . . 7 6.2. Determining Associated Local Addresses . . . . . . . . . 7
7. Application Guidance . . . . . . . . . . . . . . . . . . . . 8 7. Application Guidance . . . . . . . . . . . . . . . . . . . . 8
8. Security Considerations . . . . . . . . . . . . . . . . . . . 8 8. Security Considerations . . . . . . . . . . . . . . . . . . . 8
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 8 9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 8
10. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 8 10. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 8
11. References . . . . . . . . . . . . . . . . . . . . . . . . . 8 11. References . . . . . . . . . . . . . . . . . . . . . . . . . 8
11.1. Normative References . . . . . . . . . . . . . . . . . . 8 11.1. Normative References . . . . . . . . . . . . . . . . . . 8
11.2. Informative References . . . . . . . . . . . . . . . . . 9 11.2. Informative References . . . . . . . . . . . . . . . . . 9
Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 10 Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 10
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 12 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 12
skipping to change at page 2, line 44 skipping to change at page 2, line 44
Protocol (HTTP) [RFC7230]. This may be problematic in certain cases. Protocol (HTTP) [RFC7230]. This may be problematic in certain cases.
This document summarizes the concerns, and makes recommendations on This document summarizes the concerns, and makes recommendations on
how WebRTC implementations should best handle the tradeoff between how WebRTC implementations should best handle the tradeoff between
privacy and media performance. privacy and media performance.
2. Terminology 2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
"OPTIONAL" in this document are to be interpreted as described in BCP "OPTIONAL" in this document are to be interpreted as described in BCP
14 [RFC2119] [RFC8174] when, and only when, they appear in all 14 [RFC2119][RFC8174] when, and only when, they appear in all
capitals, as shown here. capitals, as shown here.
3. Problem Statement 3. Problem Statement
In order to establish a peer-to-peer connection, WebRTC In order to establish a peer-to-peer connection, WebRTC
implementations use Interactive Connectivity Establishment (ICE) implementations use Interactive Connectivity Establishment (ICE)
[RFC8445], which attempts to discover multiple IP addresses using [RFC8445], which attempts to discover multiple IP addresses using
techniques such as Session Traversal Utilities for NAT (STUN) techniques such as Session Traversal Utilities for NAT (STUN)
[RFC5389] and Traversal Using Relays around NAT (TURN) [RFC5766], and [RFC5389] and Traversal Using Relays around NAT (TURN) [RFC5766], and
then checks the connectivity of each local-address-remote-address then checks the connectivity of each local-address-remote-address
pair in order to select the best one. The addresses that are pair in order to select the best one. The addresses that are
collected usually consist of an endpoint's private physical/virtual collected usually consist of an endpoint's private physical or
addresses and its public Internet addresses. virtual addresses and its public Internet addresses.
These addresses are exposed upwards to the web application, so that These addresses are provided to the web application so that they can
they can be communicated to the remote endpoint for its checks. This be communicated to the remote endpoint for its checks. This allows
allows the application to learn more about the local network the application to learn more about the local network configuration
configuration than it would from a typical HTTP scenario, in which than it would from a typical HTTP scenario, in which the web server
the web server would only see a single public Internet address, i.e., would only see a single public Internet address, i.e., the address
the address from which the HTTP request was sent. from which the HTTP request was sent.
The information revealed falls into three categories: The information revealed falls into three categories:
1. If the client is multihomed, additional public IP addresses for 1. If the client is multihomed, additional public IP addresses for
the client can be learned. In particular, if the client tries to the client can be learned. In particular, if the client tries to
hide its physical location through a Virtual Private Network hide its physical location through a Virtual Private Network
(VPN), and the VPN and local OS support routing over multiple (VPN), and the VPN and local OS support routing over multiple
interfaces (a "split-tunnel" VPN), WebRTC will discover not only interfaces (a "split-tunnel" VPN), WebRTC can discover not only
the public address for the VPN, but also the ISP public address the public address for the VPN, but also the ISP public address
over which the VPN is running. over which the VPN is running.
2. If the client is behind a Network Address Translator (NAT), the 2. If the client is behind a Network Address Translator (NAT), the
client's private IP addresses, often [RFC1918] addresses, can be client's private IP addresses, often [RFC1918] addresses, can be
learned. learned.
3. If the client is behind a proxy (a client-configured "classical 3. If the client is behind a proxy (a client-configured "classical
application proxy", as defined in [RFC1919], Section 3), but application proxy", as defined in [RFC1919], Section 3), but
direct access to the Internet is permitted, WebRTC's STUN checks direct access to the Internet is permitted, WebRTC's STUN checks
will bypass the proxy and reveal the public IP address of the will bypass the proxy and reveal the public IP address of the
client. This concern also applies to the "enterprise TURN client. This concern also applies to the "enterprise TURN
server" scenario described in [RFC7478], Section 2.3.5.1, if, as server" scenario described in [RFC7478], Section 2.3.5.1, if, as
above, direct Internet access is permitted. However, when the above, direct Internet access is permitted. However, when the
term "proxy" is used in this document, it is always in reference term "proxy" is used in this document, it is always in reference
to an [RFC1919] proxy server. to an [RFC1919] proxy server.
Of these three concerns, #1 is the most significant, because for some Of these three concerns, the first is the most significant, because
users, the purpose of using a VPN is for anonymity. However, for some users, the purpose of using a VPN is for anonymity.
different VPN users will have different needs, and some VPN users However, different VPN users will have different needs, and some VPN
(e.g., corporate VPN users) may in fact prefer WebRTC to send media users (e.g., corporate VPN users) may in fact prefer WebRTC to send
traffic directly, i.e., not through the VPN. media traffic directly, i.e., not through the VPN.
#2 is a less significant but valid concern. While the [RFC4941] IPv6 The second concern is less significant but valid nonetheless. The
addresses recommended by [I-D.ietf-rtcweb-transports] are fairly core issue is that web applications can learn about addresses that
benign due to their intentionally short lifetimes, IPv4 addresses are not exposed to the internet; typically these address are IPv4,
present some challenges. Although they typically contain minimal but they can also be IPv6, as in the case of NAT64 [RFC6146]. While
disclosure of the [RFC4941] IPv6 addresses recommended by
[I-D.ietf-rtcweb-transports] is fairly benign due to their
intentionally short lifetimes, IPv4 addresses present some
challenges. Although private IPv4 addresses often contain minimal
entropy (e.g., 192.168.0.2, a fairly common address), in the worst 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 case, they can contain 24 bits of entropy with an indefinite
lifetime. As such, they can be a fairly significant fingerprinting lifetime. As such, they can be a fairly significant fingerprinting
surface. In addition, intranet web sites can be attacked more easily surface. In addition, intranet web sites can be attacked more easily
when their IPv4 address range is externally known. when their IPv4 address range is externally known.
Private local IP addresses can also act as an identifier that allows Private IP addresses can also act as an identifier that allows web
web applications running in isolated browsing contexts (e.g., normal applications running in isolated browsing contexts (e.g., normal and
and private browsing) to learn that they are running on the same private browsing) to learn that they are running on the same device.
device. This could allow the application sessions to be correlated, This could allow the application sessions to be correlated, defeating
defeating some of the privacy protections provided by isolation. It some of the privacy protections provided by isolation. It should be
should be noted that local addresses are just one potential mechanism noted that private addresses are just one potential mechanism for
for this correlation and this is an area for further study. this correlation and this is an area for further study.
#3 is the least common concern, as proxy administrators can already The third concern is the least common, as proxy administrators can
control this behavior through organizational firewall policy, and already control this behavior through organizational firewall policy,
generally, forcing WebRTC traffic through a proxy server will have and generally, forcing WebRTC traffic through a proxy server will
negative effects on both the proxy and on media quality. have negative effects on both the proxy and on media quality.
Note also that these concerns predate WebRTC; Adobe Flash Player has Note also that these concerns predate WebRTC; Adobe Flash Player has
provided similar functionality since the introduction of RTMFP provided similar functionality since the introduction of Real-Time
[RFC7016] in 2008. Media Flow Protocol (RTMFP) support [RFC7016] in 2008.
4. Goals 4. Goals
WebRTC's support of secure peer-to-peer connections facilitates WebRTC's support of secure peer-to-peer connections facilitates
deployment of decentralized systems, which can have privacy benefits. deployment of decentralized systems, which can have privacy benefits.
As a result, we want to avoid blunt solutions that disable WebRTC or As a result, blunt solutions that disable WebRTC or make it
make it significantly harder to use. This document takes a more significantly harder to use are undesirable. This document takes a
nuanced approach, with the following goals: more nuanced approach, with the following goals:
o Provide a framework for understanding the problem so that controls o Provide a framework for understanding the problem so that controls
might be provided to make different tradeoffs regarding might be provided to make different tradeoffs regarding
performance and privacy concerns with WebRTC. performance and privacy concerns with WebRTC.
o Using that framework, define settings that enable peer-to-peer o Using that framework, define settings that enable peer-to-peer
communications, each with a different balance between performance communications, each with a different balance between performance
and privacy. and privacy.
o Finally, provide recommendations for default settings that provide o Finally, provide recommendations for default settings that provide
skipping to change at page 5, line 18 skipping to change at page 5, line 21
1. By default, WebRTC traffic should follow typical IP routing, 1. By default, WebRTC traffic should follow typical IP routing,
i.e., WebRTC should use the same interface used for HTTP traffic, i.e., WebRTC should use the same interface used for HTTP traffic,
and only the system's 'typical' public addresses (or those of an and only the system's 'typical' public addresses (or those of an
enterprise TURN server, if present) should be visible to the enterprise TURN server, if present) should be visible to the
application. However, in the interest of optimal media quality, application. However, in the interest of optimal media quality,
it should be possible to enable WebRTC to make use of all network it should be possible to enable WebRTC to make use of all network
interfaces to determine the ideal route. interfaces to determine the ideal route.
2. By default, WebRTC should be able to negotiate direct peer-to- 2. By default, WebRTC should be able to negotiate direct peer-to-
peer connections between endpoints (i.e., without traversing a peer connections between endpoints (i.e., without traversing a
NAT or relay server). This ensures that applications that need NAT or relay server) when such connections are possible. This
true peer-to-peer routing for bandwidth or latency reasons can ensures that applications that need true peer-to-peer routing for
operate successfully. bandwidth or latency reasons can operate successfully.
3. It should be possible to configure WebRTC to not disclose private 3. It should be possible to configure WebRTC to not disclose private
local IP addresses, to avoid the issues associated with web local IP addresses, to avoid the issues associated with web
applications learning such addresses. This document does not applications learning such addresses. This document does not
require this to be the default state, as there is no currently require this to be the default state, as there is no currently
defined mechanism that can satisfy this requirement as well as defined mechanism that can satisfy this requirement as well as
the aforementioned requirement to allow direct peer-to-peer the aforementioned requirement to allow direct peer-to-peer
connections. connections.
4. By default, WebRTC traffic should not be sent through proxy 4. By default, WebRTC traffic should not be sent through proxy
skipping to change at page 6, line 8 skipping to change at page 6, line 12
path, and is ideal when media performance is the highest path, and is ideal when media performance is the highest
priority, but it discloses the most information. priority, but it discloses the most information.
Mode 2: Default route + associated local addresses: WebRTC MUST Mode 2: Default route + associated local addresses: WebRTC MUST
follow the kernel routing table rules, which will typically follow the kernel routing table rules, which will typically
cause media packets to take the same route as the cause media packets to take the same route as the
application's HTTP traffic. If an enterprise TURN server is application's HTTP traffic. If an enterprise TURN server is
present, the preferred route MUST be through this TURN present, the preferred route MUST be through this TURN
server. Once an interface has been chosen, the private IPv4 server. Once an interface has been chosen, the private IPv4
and IPv6 addresses associated with this interface MUST be and IPv6 addresses associated with this interface MUST be
discovered and provided to the application. This ensures discovered and provided to the application as host
that direct connections can still be established in this candidates. This ensures that direct connections can still
mode. be established in this mode.
Mode 3: Default route only: This is the the same as Mode 2, except Mode 3: Default route only: This is the the same as Mode 2, except
that the associated private addresses MUST NOT be provided; that the associated private addresses MUST NOT be provided;
the only IP addresses gathered are those discovered via the only IP addresses gathered are those discovered via
mechanisms like STUN and TURN (on the default route). This mechanisms like STUN and TURN (on the default route). This
may cause traffic to hairpin through a NAT, fall back to an may cause traffic to hairpin through a NAT, fall back to an
application TURN server, or fail altogether, with resulting application TURN server, or fail altogether, with resulting
quality implications. quality implications.
Mode 4: Force proxy: This is the same as Mode 3, but when the Mode 4: Force proxy: This is the same as Mode 3, but when the
application's HTTP traffic is sent through a proxy, WebRTC application's HTTP traffic is sent through a proxy, WebRTC
media traffic MUST also be proxied. If the proxy does not media traffic MUST also be proxied. If the proxy does not
support UDP (as is the case for all HTTP and most SOCKS support UDP (as is the case for all HTTP and most SOCKS
[RFC1928] proxies), or the WebRTC implementation does not [RFC1928] proxies), or the WebRTC implementation does not
support UDP proxying, the use of UDP will be disabled, and support UDP proxying, the use of UDP will be disabled, and
TCP will be used to send and receive media through the TCP will be used to send and receive media through the
proxy. Use of TCP will result in reduced media quality, in proxy. Use of TCP will result in reduced media quality, in
addition to any performance considerations associated with addition to any performance considerations associated with
sending all WebRTC media through the proxy server. sending all WebRTC media through the proxy server.
Mode 1 MUST only be used when user consent has been provided. The Mode 1 MUST NOT be used unless user consent has been provided. The
details of this consent are left to the implementation; one potential details of this consent are left to the implementation; one potential
mechanism is to tie this consent to getUserMedia consent. mechanism is to tie this consent to getUserMedia (device permissions)
consent, described in [I-D.ietf-rtcweb-security-arch], Section 6.2.
Alternatively, implementations can provide a specific mechanism to Alternatively, implementations can provide a specific mechanism to
obtain user consent. obtain user consent.
In cases where user consent has not been obtained, Mode 2 SHOULD be In cases where user consent has not been obtained, Mode 2 SHOULD be
used. used.
These defaults provide a reasonable tradeoff that permits trusted These defaults provide a reasonable 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
skipping to change at page 7, line 22 skipping to change at page 7, line 25
associated with forcing media through said proxy) for intra- associated with forcing media through said proxy) for intra-
organization traffic. organization traffic.
6. Implementation Guidance 6. Implementation Guidance
This section provides guidance to WebRTC implementations on how to This section provides guidance to WebRTC implementations on how to
implement the policies described above. implement the policies described above.
6.1. Ensuring Normal Routing 6.1. Ensuring Normal Routing
When trying to follow typical IP routing, the simplest approach is to When trying to follow typical IP routing, as required by Modes 2 and
bind the sockets used for peer-to-peer connections to the wildcard 3, the simplest approach is to bind() the sockets used for peer-to-
addresses (0.0.0.0 for IPv4, :: for IPv6), which allows the OS to peer connections to the wildcard addresses (0.0.0.0 for IPv4, :: for
route WebRTC traffic the same way as it would HTTP traffic. STUN and IPv6), which allows the OS to route WebRTC traffic the same way as it
TURN will work as usual, and host candidates can still be determined would HTTP traffic. STUN and TURN will work as usual, and host
as mentioned below. candidates can still be determined as mentioned below.
6.2. Determining Host Candidates 6.2. Determining Associated Local Addresses
When binding to a wildcard address, some extra work is needed to When binding to a wildcard address, some extra work is needed to
determine a suitable host candidate, which we define as the source determine the associated local address required by Mode 2, which we
address that would be used for any packets sent to the web define as the source address that would be used for any packets sent
application host (assuming that UDP and TCP get the same routing). to the web application host (assuming that UDP and TCP get the same
Use of the web application host as a destination ensures the right routing treatment). Use of the web application host as a destination
source address is selected, regardless of where the application ensures the right source address is selected, regardless of where the
resides (e.g., on an intranet). application resides (e.g., on an intranet).
First, the appropriate remote IPv4/IPv6 address is obtained by First, the appropriate remote IPv4/IPv6 address is obtained by
resolving the host component of the web application URI [RFC3986]. resolving the host component of the web application URI [RFC3986].
If the client is behind a proxy and cannot resolve these IPs via DNS, If the client is behind a proxy and cannot resolve these IPs via DNS,
the address of the proxy can be used instead. Or, if the web the address of the proxy can be used instead. Or, if the web
application was loaded from a file:// URI [RFC8089], rather than over application was loaded from a file:// URI [RFC8089], rather than over
the network, the implementation can fall back to a well-known DNS the network, the implementation can fall back to a well-known DNS
name or IP address. name or IP address.
Once a suitable remote IP has been determined, the implementation can Once a suitable remote IP has been determined, the implementation can
create a UDP socket, bind it to the appropriate wildcard address, and create a UDP socket, bind() it to the appropriate wildcard address,
tell it to connect to the remote IP. Generally, this results in the and then connect() to the remote IP. Generally, this results in the
socket being assigned a local address based on the kernel routing socket being assigned a local address based on the kernel routing
table, without sending any packets over the network. table, without sending any packets over the network.
Finally, the socket can be queried using getsockname() or the Finally, the socket can be queried using getsockname() or the
equivalent to determine the appropriate host candidate. equivalent to determine the appropriate local address.
7. Application Guidance 7. Application Guidance
The recommendations mentioned in this document may cause certain The recommendations mentioned in this document may cause certain
WebRTC applications to malfunction. In order to be robust in all WebRTC applications to malfunction. In order to be robust in all
scenarios, the following guidelines are provided for applications: scenarios, the following guidelines are provided for applications:
o Applications SHOULD deploy a TURN server with support for both UDP o Applications SHOULD deploy a TURN server with support for both UDP
and TCP connections to the server. This ensures that connectivity and TCP connections to the server. This ensures that connectivity
can still be established, even when Mode 3 or 4 are in use, can still be established, even when Mode 3 or 4 are in use,
assuming the TURN server can be reached. assuming the TURN server can be reached.
o Applications SHOULD detect when they don't have access to the full o Applications SHOULD detect when they don't have access to the full
set of ICE candidates by checking for the presence of host set of ICE candidates by checking for the presence of host
candidates. If no host candidates are present, Mode 3 or 4 above candidates. If no host candidates are present, Mode 3 or 4 above
is in use; this knowledge can be useful for diagnostic purposes. is in use; this knowledge can be useful for diagnostic purposes.
8. Security Considerations 8. Security Considerations
This document is entirely devoted to security considerations. This document describes several potential privacy and security
concerns associated with WebRTC peer-to-peer connections, and
provides mechanisms and recommendations for WebRTC implementations to
address these concerns.
9. IANA Considerations 9. IANA Considerations
This document requires no actions from IANA. This document requires no actions from IANA.
10. Acknowledgements 10. Acknowledgements
Several people provided input into this document, including Bernard Several people provided input into this document, including Bernard
Aboba, Harald Alvestrand, Youenn Fablet, Ted Hardie, Matthew Aboba, Harald Alvestrand, Youenn Fablet, Ted Hardie, Matthew
Kaufmann, Eric Rescorla, Adam Roach, and Martin Thomson. Kaufmann, Eric Rescorla, Adam Roach, and Martin Thomson.
skipping to change at page 9, line 5 skipping to change at page 9, line 10
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997, DOI 10.17487/RFC2119, March 1997,
<https://www.rfc-editor.org/info/rfc2119>. <https://www.rfc-editor.org/info/rfc2119>.
[RFC3986] Berners-Lee, T., Fielding, R., and L. Masinter, "Uniform [RFC3986] Berners-Lee, T., Fielding, R., and L. Masinter, "Uniform
Resource Identifier (URI): Generic Syntax", STD 66, Resource Identifier (URI): Generic Syntax", STD 66,
RFC 3986, DOI 10.17487/RFC3986, January 2005, RFC 3986, DOI 10.17487/RFC3986, January 2005,
<https://www.rfc-editor.org/info/rfc3986>. <https://www.rfc-editor.org/info/rfc3986>.
[RFC5389] Rosenberg, J., Mahy, R., Matthews, P., and D. Wing,
"Session Traversal Utilities for NAT (STUN)", RFC 5389,
DOI 10.17487/RFC5389, October 2008,
<https://www.rfc-editor.org/info/rfc5389>.
[RFC5766] Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using
Relays around NAT (TURN): Relay Extensions to Session
Traversal Utilities for NAT (STUN)", RFC 5766,
DOI 10.17487/RFC5766, April 2010,
<https://www.rfc-editor.org/info/rfc5766>.
[RFC8089] Kerwin, M., "The "file" URI Scheme", RFC 8089, [RFC8089] Kerwin, M., "The "file" URI Scheme", RFC 8089,
DOI 10.17487/RFC8089, February 2017, DOI 10.17487/RFC8089, February 2017,
<https://www.rfc-editor.org/info/rfc8089>. <https://www.rfc-editor.org/info/rfc8089>.
[RFC8174] Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC [RFC8174] Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC
2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174, 2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174,
May 2017, <https://www.rfc-editor.org/info/rfc8174>. May 2017, <https://www.rfc-editor.org/info/rfc8174>.
[RFC8445] Keranen, A., Holmberg, C., and J. Rosenberg, "Interactive [RFC8445] Keranen, A., Holmberg, C., and J. Rosenberg, "Interactive
Connectivity Establishment (ICE): A Protocol for Network Connectivity Establishment (ICE): A Protocol for Network
Address Translator (NAT) Traversal", RFC 8445, Address Translator (NAT) Traversal", RFC 8445,
DOI 10.17487/RFC8445, July 2018, DOI 10.17487/RFC8445, July 2018,
<https://www.rfc-editor.org/info/rfc8445>. <https://www.rfc-editor.org/info/rfc8445>.
11.2. Informative References 11.2. Informative References
[I-D.ietf-rtcweb-security-arch]
Rescorla, E., "WebRTC Security Architecture", draft-ietf-
rtcweb-security-arch-18 (work in progress), February 2019.
[I-D.ietf-rtcweb-transports] [I-D.ietf-rtcweb-transports]
Alvestrand, H., "Transports for WebRTC", draft-ietf- Alvestrand, H., "Transports for WebRTC", draft-ietf-
rtcweb-transports-17 (work in progress), October 2016. rtcweb-transports-17 (work in progress), October 2016.
[RFC1918] Rekhter, Y., Moskowitz, B., Karrenberg, D., de Groot, G., [RFC1918] Rekhter, Y., Moskowitz, B., Karrenberg, D., de Groot, G.,
and E. Lear, "Address Allocation for Private Internets", and E. Lear, "Address Allocation for Private Internets",
BCP 5, RFC 1918, DOI 10.17487/RFC1918, February 1996, BCP 5, RFC 1918, DOI 10.17487/RFC1918, February 1996,
<https://www.rfc-editor.org/info/rfc1918>. <https://www.rfc-editor.org/info/rfc1918>.
[RFC1919] Chatel, M., "Classical versus Transparent IP Proxies", [RFC1919] Chatel, M., "Classical versus Transparent IP Proxies",
skipping to change at page 9, line 44 skipping to change at page 10, line 15
[RFC1928] Leech, M., Ganis, M., Lee, Y., Kuris, R., Koblas, D., and [RFC1928] Leech, M., Ganis, M., Lee, Y., Kuris, R., Koblas, D., and
L. Jones, "SOCKS Protocol Version 5", RFC 1928, L. Jones, "SOCKS Protocol Version 5", RFC 1928,
DOI 10.17487/RFC1928, March 1996, DOI 10.17487/RFC1928, March 1996,
<https://www.rfc-editor.org/info/rfc1928>. <https://www.rfc-editor.org/info/rfc1928>.
[RFC4941] Narten, T., Draves, R., and S. Krishnan, "Privacy [RFC4941] Narten, T., Draves, R., and S. Krishnan, "Privacy
Extensions for Stateless Address Autoconfiguration in Extensions for Stateless Address Autoconfiguration in
IPv6", RFC 4941, DOI 10.17487/RFC4941, September 2007, IPv6", RFC 4941, DOI 10.17487/RFC4941, September 2007,
<https://www.rfc-editor.org/info/rfc4941>. <https://www.rfc-editor.org/info/rfc4941>.
[RFC5389] Rosenberg, J., Mahy, R., Matthews, P., and D. Wing, [RFC6146] Bagnulo, M., Matthews, P., and I. van Beijnum, "Stateful
"Session Traversal Utilities for NAT (STUN)", RFC 5389, NAT64: Network Address and Protocol Translation from IPv6
DOI 10.17487/RFC5389, October 2008, Clients to IPv4 Servers", RFC 6146, DOI 10.17487/RFC6146,
<https://www.rfc-editor.org/info/rfc5389>. April 2011, <https://www.rfc-editor.org/info/rfc6146>.
[RFC5766] Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using
Relays around NAT (TURN): Relay Extensions to Session
Traversal Utilities for NAT (STUN)", RFC 5766,
DOI 10.17487/RFC5766, April 2010,
<https://www.rfc-editor.org/info/rfc5766>.
[RFC7016] Thornburgh, M., "Adobe's Secure Real-Time Media Flow [RFC7016] Thornburgh, M., "Adobe's Secure Real-Time Media Flow
Protocol", RFC 7016, DOI 10.17487/RFC7016, November 2013, Protocol", RFC 7016, DOI 10.17487/RFC7016, November 2013,
<https://www.rfc-editor.org/info/rfc7016>. <https://www.rfc-editor.org/info/rfc7016>.
[RFC7230] Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer [RFC7230] Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer
Protocol (HTTP/1.1): Message Syntax and Routing", Protocol (HTTP/1.1): Message Syntax and Routing",
RFC 7230, DOI 10.17487/RFC7230, June 2014, RFC 7230, DOI 10.17487/RFC7230, June 2014,
<https://www.rfc-editor.org/info/rfc7230>. <https://www.rfc-editor.org/info/rfc7230>.
[RFC7478] Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real- [RFC7478] Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-
Time Communication Use Cases and Requirements", RFC 7478, Time Communication Use Cases and Requirements", RFC 7478,
DOI 10.17487/RFC7478, March 2015, DOI 10.17487/RFC7478, March 2015,
<https://www.rfc-editor.org/info/rfc7478>. <https://www.rfc-editor.org/info/rfc7478>.
Appendix A. Change log Appendix A. Change log
Changes in draft -12:
o Editorial updates from IETF LC review.
Changes in draft -11: Changes in draft -11:
o Editorial updates from AD review. o Editorial updates from AD review.
Changes in draft -10: Changes in draft -10:
o Incorporate feedback from IETF 102 on the problem space. o Incorporate feedback from IETF 102 on the problem space.
o Note that future versions of the document may define new modes. o Note that future versions of the document may define new modes.
 End of changes. 30 change blocks. 
77 lines changed or deleted 99 lines changed or added

This html diff was produced by rfcdiff 1.47. The latest version is available from http://tools.ietf.org/tools/rfcdiff/