draft-ietf-rtcweb-ip-handling-04.txt   draft-ietf-rtcweb-ip-handling-05.txt 
Network Working Group J. Uberti Network Working Group J. Uberti
Internet-Draft G. Shieh Internet-Draft Google
Intended status: Standards Track Google Intended status: Standards Track G. Shieh
Expires: January 4, 2018 July 3, 2017 Expires: August 15, 2018 Facebook
February 11, 2018
WebRTC IP Address Handling Requirements WebRTC IP Address Handling Requirements
draft-ietf-rtcweb-ip-handling-04 draft-ietf-rtcweb-ip-handling-05
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
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provisions of BCP 78 and BCP 79. provisions of BCP 78 and BCP 79.
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This Internet-Draft will expire on January 4, 2018. This Internet-Draft will expire on August 15, 2018.
Copyright Notice Copyright Notice
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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 . . . . . . . . . . . . . . . . . . . . . . . 4
6. Application Guidance . . . . . . . . . . . . . . . . . . . . 6 5.1. Principles . . . . . . . . . . . . . . . . . . . . . . . 4
7. Security Considerations . . . . . . . . . . . . . . . . . . . 6 5.2. Modes and Recommendations . . . . . . . . . . . . . . . . 5
8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 6 6. Implementation Guidance . . . . . . . . . . . . . . . . . . . 6
9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 7 7. Application Guidance . . . . . . . . . . . . . . . . . . . . 7
10. References . . . . . . . . . . . . . . . . . . . . . . . . . 7 8. Security Considerations . . . . . . . . . . . . . . . . . . . 7
10.1. Normative References . . . . . . . . . . . . . . . . . . 7 9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 7
10.2. Informative References . . . . . . . . . . . . . . . . . 7 10. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 7
Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 8 11. References . . . . . . . . . . . . . . . . . . . . . . . . . 7
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 9 11.1. Normative References . . . . . . . . . . . . . . . . . . 7
11.2. Informative References . . . . . . . . . . . . . . . . . 7
Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 9
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 10
1. Introduction 1. Introduction
One of WebRTC's key features is its support of peer-to-peer One of WebRTC's key features is its support of peer-to-peer
connections. However, when establishing such a connection, which connections. However, when establishing such a connection, which
involves connectivity tests using various IP addresses, WebRTC may involves connection attempts from various IP addresses, WebRTC may
allow a web application to learn additional information about the allow a web application to learn additional information about the
user compared to an application that only uses the Hypertext Transfer user compared to an application that only uses the Hypertext Transfer
Protocol (HTTP) [RFC7230]. This may be problematic in certain cases. Protocol (HTTP) [RFC7230]. This may be problematic in certain cases.
This document summarizes the concerns, and makes recommendations on This document summarizes the concerns, and makes recommendations on
how WebRTC implementations should best handle the tradeoff between how WebRTC implementations should best handle the 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", "MAY", and "OPTIONAL" in this "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in [RFC2119]. document are to be interpreted as described in [RFC2119].
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)
[RFC5245], which gathers and exchanges all the IP addresses it can [RFC5245], which attempts to discover multiple IP addresses using
discover, using techniques like Session Traversal Utilities for NAT techniques such as Session Traversal Utilities for NAT (STUN)
(STUN) [RFC5389] and Traversal Using Relays around NAT (TURN) [RFC5389] and Traversal Using Relays around NAT (TURN) [RFC5766], and
[RFC5766], in order to check the connectivity of each local-address- then checks the connectivity of each local-address-remote-address
remote-address pair and select the best one. The addresses that are pair in order to select the best one. The addresses that are
gathered usually consist of an endpoint's private physical/virtual collected usually consist of an endpoint's private physical/virtual
addresses and its public Internet addresses. addresses and its public Internet addresses.
These addresses are exposed upwards to the web application, so that These addresses are exposed upwards to the web application, so that
they can be communicated to the remote endpoint. This allows the they can be communicated to the remote endpoint for its checks. This
application to learn more about the local network configuration than allows the application to learn more about the local network
it would from a typical HTTP scenario, in which the web server would configuration than it would from a typical HTTP scenario, in which
only see a single public Internet address, i.e. the address from the web server would only see a single public Internet address, i.e.,
which the HTTP request was sent. the address 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 behind a Network Address Translator (NAT), the 1. If the client is multihomed, additional public IP addresses for
client's private IP addresses, typically [RFC1918] addresses, can the client can be learned. In particular, if the client tries to
be learned. hide its physical location through a Virtual Private Network
(VPN), and the VPN and local OS support routing over multiple
interfaces (a "split-tunnel" VPN), WebRTC will discover not only
the public address for the VPN, but also the ISP public address
over which the VPN is running.
2. If the client tries to hide its physical location through a 2. If the client is behind a Network Address Translator (NAT), the
Virtual Private Network (VPN), and the VPN and local OS support client's private IP addresses, often [RFC1918] addresses, can be
routing over multiple interfaces (i.e., a "split-tunnel" VPN), learned.
WebRTC will discover the public address for the VPN as well as
the ISP public address that the VPN runs over.
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 also supported, WebRTC's STUN direct access to the Internet is also supported, WebRTC's STUN
checks will bypass the proxy and reveal the public address of the checks will bypass the proxy and reveal the public IP address of
client. the client.
Of these three concerns, #2 is the most significant concern, since Of these three concerns, #1 is the most significant, because for some
for some users, the purpose of using a VPN is for anonymity. users, the purpose of using a VPN is for anonymity. However,
However, different VPN users will have different needs, and some VPN different VPN users will have different needs, and some VPN users
users (e.g. corporate VPN users) may in fact prefer WebRTC to send (e.g., corporate VPN users) may in fact prefer WebRTC to send media
media traffic directly, i.e., not through the VPN. traffic directly, i.e., not through the VPN.
#3 is a less common concern, as proxy administrators can control this #2 is considered to be a less significant concern, given that the
behavior through organization firewall policy if desired, coupled local address values often contain minimal information (e.g.,
with the fact that forcing WebRTC traffic through a proxy will have 192.168.0.2), or have built-in privacy protection (e.g., the
negative effects on both the proxy and on media quality. For [RFC4941] IPv6 addresses recommended by
situations where this is an important consideration, use of a RETURN [I-D.ietf-rtcweb-transports]).
proxy, as described below, can be an effective solution.
#1 is considered to be the least significant concern, given that the #3 is the least common concern, as proxy administrators can already
local address values often contain minimal information (e.g. control this behavior through organizational firewall policy, and
192.168.0.2), or have built-in privacy protection (e.g. [RFC4941] generally, forcing WebRTC traffic through a proxy server will have
IPv6 addresses). 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 RTMFP
[RFC7016] in 2008. [RFC7016] in 2008.
4. Goals 4. Goals
Being peer-to-peer, WebRTC represents a privacy-enabling technology, WebRTC's support of secure peer-to-peer connections facilitates
and therefore we want to avoid solutions that disable WebRTC or make deployment of decentralized systems, which can have privacy benefits.
it harder to use. This means that WebRTC should be configured by As a result, we want to avoid blunt solutions that disable WebRTC or
default to only reveal the minimum amount of information needed to make it significantly harder to use. This document takes a more
establish a performant WebRTC session, while providing options to nuanced approach, with the following goals:
reveal additional information upon user consent, or further limit
this information if the user has specifically requested this.
Specifically, WebRTC should:
o Provide a privacy-friendly default behavior which strikes the o Provide a framework for understanding the problem so that controls
right balance between privacy and media performance for most users might be provided to make different tradeoffs regarding
and use cases. performance and privacy concerns with WebRTC.
o For users who care more about one versus the other, provide a o Using that framework, define settings that enable peer-to-peer
means to customize the experience. communications, each with a different balance between performance
and privacy.
o Finally, provide recommendations for default settings that provide
reasonable performance without also exposing addressing
information in a way that might violate user expectations.
5. Detailed Design 5. Detailed Design
The key principles for the design are listed below: 5.1. Principles
1. By default, WebRTC should follow normal IP routing rules, to the The key principles for our framework are stated below:
extent that this is easy to determine (i.e., not considering
proxies). This can be accomplished by binding local sockets to
the wildcard addresses (0.0.0.0 for IPv4, :: for IPv6), which
allows the OS to route WebRTC traffic the same way as it would
HTTP traffic, and allows only the 'typical' public addresses to
be discovered.
2. By default, support for direct connections between hosts (i.e., 1. By default, WebRTC traffic should follow typical IP routing,
without traversing a NAT or relay server) should be maintained. i.e., WebRTC should use the same interface used for HTTP traffic,
To accomplish this, the local IPv4 and IPv6 addresses of the and only the system's 'typical' public addresses should be
interface used for outgoing STUN traffic should still be surfaced visible to the application. However, in the interest of optimal
as candidates, even when binding to the wildcard addresses as media quality, it should be possible to enable WebRTC to make use
mentioned above. The appropriate addresses here can be of all network interfaces to determine the ideal route.
discovered by the common trick of binding sockets to the wildcard
addresses, connect()ing those sockets to some well-known public
IP address, and then reading the bound local addresses via
getsockname(). This approach requires no data exchange; it
simply provides a mechanism for applications to retrieve the
desired information from the kernel routing table.
3. Determining whether a web proxy is in use is a complex process, 2. By default, WebRTC should be able to negotiate direct peer-to-
as the answer can depend on the exact site or address being peer connections between endpoints (i.e., without traversing a
contacted. Furthermore, web proxies that support UDP are not NAT or relay server), by providing a minimal set of local IP
widely deployed today. As a result, when WebRTC is made to go addresses to the application for use in the ICE process. This
through a proxy, it typically needs to use TCP, either ICE-TCP ensures that applications that need true peer-to-peer routing for
bandwidth or latency reasons can operate successfully. However,
it should be possible to suppress these addresses (with the
resultant impact on direct connections) if desired.
[RFC6544] or TURN-over-TCP [RFC5766]. Naturally, this has 3. By default, WebRTC traffic should not be sent through proxy
attendant costs on media quality as well as proxy performance, servers, due to the media quality problems associated with
and should be avoided where possible. sending WebRTC traffic over TCP, which is almost always used when
communicating with proxies, as well as proxy performance issues
that may result from proxying WebRTC's long-lived, high-bandwidth
connections. However, it should be possible to force WebRTC to
send its traffic through a configured proxy if desired.
4. RETURN [I-D.ietf-rtcweb-return] is a proposal for explicit 5.2. Modes and Recommendations
proxying of WebRTC media traffic. When RETURN proxies are
deployed, media and STUN checks will go through the proxy, but
without the performance issues associated with sending through a
typical web proxy.
Based on these ideas, we define four specific modes of WebRTC Based on these ideas, we define four specific modes of WebRTC
behavior, reflecting different media quality/privacy tradeoffs: behavior, reflecting different media quality/privacy tradeoffs:
Mode 1: Enumerate all addresses: WebRTC MUST bind to all interfaces Mode 1: Enumerate all addresses: WebRTC MUST use all network
individually and use them all to attempt communication with interfaces to attempt communication with STUN servers, TURN
STUN servers, TURN servers, or peers. This will converge on servers, or peers. This will converge on the best media
the best media path, and is ideal when media performance is path, and is ideal when media performance is the highest
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 (e.g., by binding follow the kernel routing table rules, which will typically
solely to the wildcard address), which will typically cause cause media packets to take the same route as the
media packets to take the same route as the application's application's HTTP traffic. In addition, the private IPv4
HTTP traffic. In addition, any private IPv4 and IPv6 and IPv6 addresses associated with the kernel-chosen
addresses associated with the kernel-chosen interface MUST interface MUST be discovered and provided to the
be discovered through getsockname, as mentioned above, and application. This ensures that direct connections can still
provided to the application. This ensures that direct be established in this mode.
connections can still be established in this mode.
Mode 3: Default route only: This is the the same as Mode 2, except Mode 3: Default route only: This is the the same as Mode 2, except
that the associated private address MUST NOT be provided. that the associated private addressses MUST NOT be provided;
This may cause traffic to hairpin through a NAT, fall back the only IP addresses gathered are those discovered via
to the application TURN server, or fail altogether, with mechanisms like STUN and TURN (on the default route). This
resulting quality implications. may cause traffic to hairpin through a NAT, fall back to an
application TURN server, or fail altogether, with resulting
quality implications.
Mode 4: Force proxy: This forces all WebRTC media traffic through a Mode 4: Force proxy: This is the same as Mode 3, but all WebRTC
proxy, if one is configured. If the proxy does not support media traffic is forced through a proxy, if one is
UDP (as is the case for all HTTP and most SOCKS [RFC1928] configured. If the proxy does not support UDP (as is the
proxies), or the WebRTC implementation does not support UDP case for all HTTP and most SOCKS [RFC1928] proxies), or the
proxying, the use of UDP will be disabled, and TCP will be WebRTC implementation does not support UDP proxying, the use
used to send and receive media through the proxy. Use of of UDP will be disabled, and TCP will be used to send and
TCP will result in reduced quality, in addition to any receive media through the proxy. Use of TCP will result in
performance considerations associated with sending all reduced media quality, in addition to any performance
WebRTC media through the proxy server. considerations associated with sending all WebRTC media
through the proxy server.
The recommended defaults are as follows:
Mode 1 MUST only be used when user consent has been provided; this Mode 1 MUST only be used when user consent has been provided; this
thwarts the typical drive-by enumeration attacks. The details of allows trusted WebRTC applications to achieve optimal network
this consent are left to the implementation; one potential mechanism performance, but significanly limites the network information exposed
is to tie this consent to getUserMedia consent. to arbitrary web pages. The details of this consent are left to the
implementation; one potential mechanism is to tie this consent to
getUserMedia 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. This allows applications to still achieve direct connections used. This allows applications to still achieve direct connections
in many cases, even without consent (e.g., data channel in many cases, even without consent (e.g., streaming or data channel
applications). However, user agents MAY choose a stricter default applications). However, implementations MAY choose a stricter
policy in certain circumstances. default policy in certain circumstances.
Note that when a RETURN proxy is configured for the interface Note that these defaults can still be used even for organizations
associated with the default route, Mode 2 and 3 will cause any that want all external WebRTC traffic to traverse a proxy, simply by
external media traffic to go through the RETURN proxy. While the setting an organizational firewall policy that allows WebRTC traffic
RETURN approach gives the best performance, a similar result can be to only leave through the proxy. This provides a way to ensure the
achieved for non-RETURN proxies via an organization firewall policy proxy is used for any external traffic, but avoids the performance
that only allows external WebRTC traffic to leave through the proxy issues of Mode 4 (where all media is forced through said proxy) for
(typically, over TCP). This provides a way to ensure the proxy is intra-organization traffic.
used for any external traffic, but avoids the performance issues of
Mode 4, where all media is forced through said proxy, for intra-
organization traffic.
6. Application Guidance 6. Implementation Guidance
This section provides guidance to WebRTC implementations on how to
implement the policies described above.
When trying to follow typical IP routing, the simplest approach is to
bind the sockets used for p2p connections to the wildcard addresses
(0.0.0.0 for IPv4, :: for IPv6), which allows the OS to route WebRTC
traffic the same way as it would HTTP traffic. STUN and TURN will
work as usual, and host candidates can be determined as mentioned
below.
In order to discover the correct local IP addresses, implementations
can use the common trick of binding sockets to the wildcard
addresses, connect()ing those sockets to the IPv4/IPv6 addresses of
the web application (obtained by resolving the host component of its
URI [RFC3986]) and then reading the bound local addresses via
getsockname(). This requires no data exchange; it simply provides a
mechanism for applications to retrieve the desired information from
the kernel routing table.
Use of the web application IPs ensures the right local IPs are
selected, regardless of where the application is hosted (e.g., on an
intranet). If the client is behind a proxy and cannot resolve the
IPs via DNS, the IPv4/v6 addresses of the proxy can be used instead.
If the web application was loaded from a file:// URI [RFC8089], the
implementation can fall back to a well-known DNS name or IP address.
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.
7. Security Considerations 8. Security Considerations
This document is entirely devoted to security considerations. This document is entirely devoted to security considerations.
8. IANA Considerations 9. IANA Considerations
This document requires no actions from IANA. This document requires no actions from IANA.
9. Acknowledgements 10. Acknowledgements
Several people provided input into this document, including Bernard Several people provided input into this document, including Bernard
Aboba, Harald Alvestrand, Ted Hardie, Matthew Kaufmann, Eric Aboba, Harald Alvestrand, Ted Hardie, Matthew Kaufmann, Eric
Rescorla, Adam Roach, and Martin Thomson. Rescorla, Adam Roach, and Martin Thomson.
10. References 11. References
10.1. Normative References 11.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997, DOI 10.17487/RFC2119, March 1997,
<http://www.rfc-editor.org/info/rfc2119>. <https://www.rfc-editor.org/info/rfc2119>.
10.2. Informative References 11.2. Informative References
[I-D.ietf-rtcweb-return] [I-D.ietf-rtcweb-transports]
Schwartz, B. and J. Uberti, "Recursively Encapsulated TURN Alvestrand, H., "Transports for WebRTC", draft-ietf-
(RETURN) for Connectivity and Privacy in WebRTC", draft- rtcweb-transports-17 (work in progress), October 2016.
ietf-rtcweb-return-02 (work in progress), March 2017.
[RFC1918] Rekhter, Y., Moskowitz, B., Karrenberg, D., de Groot, G., [RFC1918] Rekhter, Y., Moskowitz, B., Karrenberg, D., de Groot, G.,
and E. Lear, "Address Allocation for Private Internets", and E. Lear, "Address Allocation for Private Internets",
BCP 5, RFC 1918, DOI 10.17487/RFC1918, February 1996, BCP 5, RFC 1918, DOI 10.17487/RFC1918, February 1996,
<http://www.rfc-editor.org/info/rfc1918>. <https://www.rfc-editor.org/info/rfc1918>.
[RFC1919] Chatel, M., "Classical versus Transparent IP Proxies", [RFC1919] Chatel, M., "Classical versus Transparent IP Proxies",
RFC 1919, DOI 10.17487/RFC1919, March 1996, RFC 1919, DOI 10.17487/RFC1919, March 1996,
<http://www.rfc-editor.org/info/rfc1919>. <https://www.rfc-editor.org/info/rfc1919>.
[RFC1928] Leech, M., Ganis, M., Lee, Y., Kuris, R., Koblas, D., and [RFC1928] Leech, M., Ganis, M., Lee, Y., Kuris, R., Koblas, D., and
L. Jones, "SOCKS Protocol Version 5", RFC 1928, L. Jones, "SOCKS Protocol Version 5", RFC 1928,
DOI 10.17487/RFC1928, March 1996, DOI 10.17487/RFC1928, March 1996,
<http://www.rfc-editor.org/info/rfc1928>. <https://www.rfc-editor.org/info/rfc1928>.
[RFC3986] Berners-Lee, T., Fielding, R., and L. Masinter, "Uniform
Resource Identifier (URI): Generic Syntax", STD 66,
RFC 3986, DOI 10.17487/RFC3986, January 2005,
<https://www.rfc-editor.org/info/rfc3986>.
[RFC4941] Narten, T., Draves, R., and S. Krishnan, "Privacy [RFC4941] Narten, T., Draves, R., and S. Krishnan, "Privacy
Extensions for Stateless Address Autoconfiguration in Extensions for Stateless Address Autoconfiguration in
IPv6", RFC 4941, DOI 10.17487/RFC4941, September 2007, IPv6", RFC 4941, DOI 10.17487/RFC4941, September 2007,
<http://www.rfc-editor.org/info/rfc4941>. <https://www.rfc-editor.org/info/rfc4941>.
[RFC5245] Rosenberg, J., "Interactive Connectivity Establishment [RFC5245] Rosenberg, J., "Interactive Connectivity Establishment
(ICE): A Protocol for Network Address Translator (NAT) (ICE): A Protocol for Network Address Translator (NAT)
Traversal for Offer/Answer Protocols", RFC 5245, Traversal for Offer/Answer Protocols", RFC 5245,
DOI 10.17487/RFC5245, April 2010, DOI 10.17487/RFC5245, April 2010,
<http://www.rfc-editor.org/info/rfc5245>. <https://www.rfc-editor.org/info/rfc5245>.
[RFC5389] Rosenberg, J., Mahy, R., Matthews, P., and D. Wing, [RFC5389] Rosenberg, J., Mahy, R., Matthews, P., and D. Wing,
"Session Traversal Utilities for NAT (STUN)", RFC 5389, "Session Traversal Utilities for NAT (STUN)", RFC 5389,
DOI 10.17487/RFC5389, October 2008, DOI 10.17487/RFC5389, October 2008,
<http://www.rfc-editor.org/info/rfc5389>. <https://www.rfc-editor.org/info/rfc5389>.
[RFC5766] Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using [RFC5766] Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using
Relays around NAT (TURN): Relay Extensions to Session Relays around NAT (TURN): Relay Extensions to Session
Traversal Utilities for NAT (STUN)", RFC 5766, Traversal Utilities for NAT (STUN)", RFC 5766,
DOI 10.17487/RFC5766, April 2010, DOI 10.17487/RFC5766, April 2010,
<http://www.rfc-editor.org/info/rfc5766>. <https://www.rfc-editor.org/info/rfc5766>.
[RFC6544] Rosenberg, J., Keranen, A., Lowekamp, B., and A. Roach,
"TCP Candidates with Interactive Connectivity
Establishment (ICE)", RFC 6544, DOI 10.17487/RFC6544,
March 2012, <http://www.rfc-editor.org/info/rfc6544>.
[RFC7016] Thornburgh, M., "Adobe's Secure Real-Time Media Flow [RFC7016] Thornburgh, M., "Adobe's Secure Real-Time Media Flow
Protocol", RFC 7016, DOI 10.17487/RFC7016, November 2013, Protocol", RFC 7016, DOI 10.17487/RFC7016, November 2013,
<http://www.rfc-editor.org/info/rfc7016>. <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,
<http://www.rfc-editor.org/info/rfc7230>. <https://www.rfc-editor.org/info/rfc7230>.
[RFC8089] Kerwin, M., "The "file" URI Scheme", RFC 8089,
DOI 10.17487/RFC8089, February 2017,
<https://www.rfc-editor.org/info/rfc8089>.
Appendix A. Change log Appendix A. Change log
Changes in draft -05:
o Separated framework definition from implementation techniques.
o Removed RETURN references.
o Use origin when determining local IPs, rather than a well-known
IP.
Changes in draft -04: Changes in draft -04:
o Rewording and cleanup in abstract, intro, and problem statement. o Rewording and cleanup in abstract, intro, and problem statement.
o Added 2119 boilerplate. o Added 2119 boilerplate.
o Fixed weird reference spacing. o Fixed weird reference spacing.
o Expanded acronyms on first use. o Expanded acronyms on first use.
skipping to change at page 9, line 36 skipping to change at page 10, line 29
Justin Uberti Justin Uberti
Google Google
747 6th St S 747 6th St S
Kirkland, WA 98033 Kirkland, WA 98033
USA USA
Email: justin@uberti.name Email: justin@uberti.name
Guo-wei Shieh Guo-wei Shieh
Google Facebook
747 6th St S 1101 Dexter Ave
Kirkland, WA 98033 Seattle, WA 98109
USA USA
Email: guoweis@google.com Email: guoweis@facebook.com
 End of changes. 53 change blocks. 
171 lines changed or deleted 212 lines changed or added

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