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
	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.
	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 http://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 January 4, 2018.		This Internet-Draft will expire on August 15, 2018.
	Copyright Notice		Copyright Notice
	Copyright (c) 2017 IETF Trust and the persons identified as the		Copyright (c) 2018 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
	(http://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 . . . . . . . . . . . . . . . . . . . . . . . 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
This html diff was produced by rfcdiff 1.46. The latest version is available from http://tools.ietf.org/tools/rfcdiff/