draft-ietf-rtcweb-ip-handling-03.txt		draft-ietf-rtcweb-ip-handling-04.txt
	Network Working Group J. Uberti		Network Working Group J. Uberti
	Internet-Draft G. Shieh		Internet-Draft G. Shieh
	Intended status: Standards Track Google		Intended status: Standards Track Google
	Expires: July 18, 2017 January 14, 2017		Expires: January 4, 2018 July 3, 2017
	WebRTC IP Address Handling Requirements		WebRTC IP Address Handling Requirements
	draft-ietf-rtcweb-ip-handling-03		draft-ietf-rtcweb-ip-handling-04
	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 applications.		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 http://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 July 18, 2017.		This Internet-Draft will expire on January 4, 2018.
	Copyright Notice		Copyright Notice
	Copyright (c) 2017 IETF Trust and the persons identified as the		Copyright (c) 2017 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		(http://trustee.ietf.org/license-info) in effect on the date of
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	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. Problem Statement . . . . . . . . . . . . . . . . . . . . . . 2		2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 2
	3. Goals . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3		3. Problem Statement . . . . . . . . . . . . . . . . . . . . . . 2
	4. Detailed Design . . . . . . . . . . . . . . . . . . . . . . . 4		4. Goals . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4
	5. Application Guidance . . . . . . . . . . . . . . . . . . . . 6		5. Detailed Design . . . . . . . . . . . . . . . . . . . . . . . 4
	6. Security Considerations . . . . . . . . . . . . . . . . . . . 6		6. Application Guidance . . . . . . . . . . . . . . . . . . . . 6
	7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 6		7. Security Considerations . . . . . . . . . . . . . . . . . . . 6
	8. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 6		8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 6
	9. Informative References . . . . . . . . . . . . . . . . . . . 6		9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 7
			10. References . . . . . . . . . . . . . . . . . . . . . . . . . 7
			10.1. Normative References . . . . . . . . . . . . . . . . . . 7
			10.2. Informative References . . . . . . . . . . . . . . . . . 7
	Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 8		Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 8
	Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 8		Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 9
	1. Introduction		1. Introduction
	As a technology that supports peer-to-peer connections, WebRTC may		One of WebRTC's key features is its support of peer-to-peer
	send data over different network paths than the path used for HTTP		connections. However, when establishing such a connection, which
	traffic. This may allow a web application to learn additional		involves connectivity tests using various IP addresses, WebRTC may
	information about the user, which may be problematic in certain		allow a web application to learn additional information about the
	cases. This document summarizes the concerns, and makes		user compared to an application that only uses the Hypertext Transfer
	recommendations on how best to handle the tradeoff between privacy		Protocol (HTTP) [RFC7230]. This may be problematic in certain cases.
	and media performance.		This document summarizes the concerns, and makes recommendations on
			how WebRTC implementations should best handle the tradeoff between
			privacy and media performance.
	2. Problem Statement		2. Terminology
	WebRTC enables real-time peer-to-peer communications by enumerating		The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
	network interfaces and discovering the best route through the ICE		"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
	[RFC5245]protocol. During the ICE process, the peers involved in a		document are to be interpreted as described in [RFC2119].
	session gather and exchange all the IP addresses they can discover,
	so that the connectivity of each IP pair can be checked, and the best		3. Problem Statement
	path chosen. The addresses that are gathered usually consist of an
	endpoint's private physical/virtual addresses, and its public		In order to establish a peer-to-peer connection, WebRTC
	Internet addresses.		implementations use Interactive Connectivity Establishment (ICE)
			[RFC5245], which gathers and exchanges all the IP addresses it can
			discover, using techniques like Session Traversal Utilities for NAT
			(STUN) [RFC5389] and Traversal Using Relays around NAT (TURN)
			[RFC5766], in order to check the connectivity of each local-address-
			remote-address pair and select the best one. The addresses that are
			gathered usually consist of an endpoint's private physical/virtual
			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. This allows the
	application to learn more about the local network configuration than		application to learn more about the local network configuration than
	it would from a typical HTTP scenario, in which the web server would		it would from a typical HTTP scenario, in which the web server would
	only see a single public Internet address, i.e. the address from		only see a single public Internet address, i.e. the address from
	which the HTTP request was sent.		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 NAT, the client's private IP addresses,		1. If the client is behind a Network Address Translator (NAT), the
	typically [RFC1918]addresses, can be learned.		client's private IP addresses, typically [RFC1918] addresses, can
			be learned.
	2. If the client tries to hide its physical location through a VPN,		2. If the client tries to hide its physical location through a
	and the VPN and local OS support routing over multiple interfaces		Virtual Private Network (VPN), and the VPN and local OS support
	(i.e., a "split-tunnel" VPN), WebRTC will discover the public		routing over multiple interfaces (i.e., a "split-tunnel" VPN),
	address for the VPN as well as the ISP public address that the		WebRTC will discover the public address for the VPN as well as
	VPN runs over.		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
	[RFC5389]checks will bypass the proxy and reveal the public		checks will bypass the proxy and reveal the public address of the
	address of the client.		client.
	Of these three concerns, #2 is the most significant concern, since		Of these three concerns, #2 is the most significant concern, since
	for some users, the purpose of using a VPN is for anonymity.		for some users, the purpose of using a VPN is for anonymity.
	However, different VPN users will have different needs, and some VPN		However, different VPN users will have different needs, and some VPN
	users (e.g. corporate VPN users) may in fact prefer WebRTC to send		users (e.g. corporate VPN users) may in fact prefer WebRTC to send
	media traffic directly, i.e., not through the VPN.		media traffic directly, i.e., not through the VPN.
	#3 is a less common concern, as proxy administrators can control this		#3 is a less common concern, as proxy administrators can control this
	behavior through organization firewall policy if desired, coupled		behavior through organization firewall policy if desired, coupled
	with the fact that forcing WebRTC traffic through a proxy will have		with the fact that forcing WebRTC traffic through a proxy will have
	negative effects on both the proxy and on media quality. For		negative effects on both the proxy and on media quality. For
	situations where this is an important consideration, use of a RETURN		situations where this is an important consideration, use of a RETURN
	proxy, as described below, can be an effective solution.		proxy, as described below, can be an effective solution.
	#1 is considered to be the least significant concern, given that the		#1 is considered to be the least significant concern, given that the
	local address values often contain minimal information (e.g.		local address values often contain minimal information (e.g.
	192.168.0.2), or have built-in privacy protection (e.g.		192.168.0.2), or have built-in privacy protection (e.g. [RFC4941]
	[RFC4941]IPv6 addresses).		IPv6 addresses).
	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.
	3. Goals		4. Goals
	Being peer-to-peer, WebRTC represents a privacy-enabling technology,		Being peer-to-peer, WebRTC represents a privacy-enabling technology,
	and therefore we want to avoid solutions that disable WebRTC or make		and therefore we want to avoid solutions that disable WebRTC or make
	it harder to use. This means that WebRTC should be configured by		it harder to use. This means that WebRTC should be configured by
	default to only reveal the minimum amount of information needed to		default to only reveal the minimum amount of information needed to
	establish a performant WebRTC session, while providing options to		establish a performant WebRTC session, while providing options to
	reveal additional information upon user consent, or further limit		reveal additional information upon user consent, or further limit
	this information if the user has specifically requested this.		this information if the user has specifically requested this.
	Specifically, WebRTC should:		Specifically, WebRTC should:
	o Provide a privacy-friendly default behavior which strikes the		o Provide a privacy-friendly default behavior which strikes the
	right balance between privacy and media performance for most users		right balance between privacy and media performance for most users
	and use cases.		and use cases.
	o For users who care more about one versus the other, provide a		o For users who care more about one versus the other, provide a
	means to customize the experience.		means to customize the experience.
	4. Detailed Design		5. Detailed Design
	The key principles for the design are listed below:		The key principles for the design are listed below:
	1. By default, WebRTC should follow normal IP routing rules, to the		1. By default, WebRTC should follow normal IP routing rules, to the
	extent that this is easy to determine (i.e., not considering		extent that this is easy to determine (i.e., not considering
	proxies). This can be accomplished by binding local sockets to		proxies). This can be accomplished by binding local sockets to
	the wildcard addresses (0.0.0.0 for IPv4, :: for IPv6), which		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		allows the OS to route WebRTC traffic the same way as it would
	HTTP traffic, and allows only the 'typical' public addresses to		HTTP traffic, and allows only the 'typical' public addresses to
	be discovered.		be discovered.
	2. By default, support for direct connections between hosts (i.e.,		2. By default, support for direct connections between hosts (i.e.,
	without traversing a NAT or relay server) should be maintained.		without traversing a NAT or relay server) should be maintained.
	To accomplish this, the local IPv4 and IPv6 addresses of the		To accomplish this, the local IPv4 and IPv6 addresses of the
	interface used for outgoing STUN traffic should still be surfaced		interface used for outgoing STUN traffic should still be surfaced
	as candidates, even when binding to the wildcard addresses as		as candidates, even when binding to the wildcard addresses as
	mentioned above. The appropriate addresses here can be		mentioned above. The appropriate addresses here can be
	discovered by the common trick of binding sockets to the wildcard		discovered by the common trick of binding sockets to the wildcard
	addresses, connect()ing those sockets to some well-known public		addresses, connect()ing those sockets to some well-known public
	IP address (one particular example being "8.8.8.8"), and then		IP address, and then reading the bound local addresses via
	reading the bound local addresses via getsockname(). This		getsockname(). This approach requires no data exchange; it
	approach requires no data exchange; it simply provides a		simply provides a mechanism for applications to retrieve the
	mechanism for applications to retrieve the desired information		desired information from the kernel routing table.
	from the kernel routing table.
	3. Determining whether a web proxy is in use is a complex process,		3. Determining whether a web proxy is in use is a complex process,
	as the answer can depend on the exact site or address being		as the answer can depend on the exact site or address being
	contacted. Furthermore, web proxies that support UDP are not		contacted. Furthermore, web proxies that support UDP are not
	widely deployed today. As a result, when WebRTC is made to go		widely deployed today. As a result, when WebRTC is made to go
	through a proxy, it typically needs to use TCP, either ICE-TCP		through a proxy, it typically needs to use TCP, either ICE-TCP
	[RFC6544]or TURN-over-TCP [RFC5766]. Naturally, this has
			[RFC6544] or TURN-over-TCP [RFC5766]. Naturally, this has
	attendant costs on media quality as well as proxy performance,		attendant costs on media quality as well as proxy performance,
	and should be avoided where possible.		and should be avoided where possible.
	4. RETURN [I-D.ietf-rtcweb-return]is a proposal for explicit		4. RETURN [I-D.ietf-rtcweb-return] is a proposal for explicit
	proxying of WebRTC media traffic. When RETURN proxies are		proxying of WebRTC media traffic. When RETURN proxies are
	deployed, media and STUN checks will go through the proxy, but		deployed, media and STUN checks will go through the proxy, but
	without the performance issues associated with sending through a		without the performance issues associated with sending through a
	typical web proxy.		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 bind to all interfaces
	individually and use them all to attempt communication with		individually and use them all to attempt communication with
	skipping to change at page 5, line 29 ¶		skipping to change at page 5, line 42 ¶
	connections can still 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 address MUST NOT be provided.
	This may cause traffic to hairpin through a NAT, fall back		This may cause traffic to hairpin through a NAT, fall back
	to the application TURN server, or fail altogether, with		to the application TURN server, or fail altogether, with
	resulting quality implications.		resulting quality implications.
	Mode 4: Force proxy: This forces all WebRTC media traffic through a		Mode 4: Force proxy: This forces all WebRTC media traffic through a
	proxy, if one is configured. If the proxy does not support		proxy, if one is configured. If the proxy does not support
	UDP (as is the case for all HTTP and most SOCKS		UDP (as is the case for all HTTP and most SOCKS [RFC1928]
	[RFC1928]proxies), or the WebRTC implementation does not		proxies), or the WebRTC implementation does not support UDP
	support UDP proxying, the use of UDP will be disabled, and		proxying, the use of UDP will be disabled, and TCP will be
	TCP will be used to send and receive media through the		used to send and receive media through the proxy. Use of
	proxy. Use of TCP will result in reduced quality, in		TCP will result in reduced quality, in addition to any
	addition to any performance considerations associated with		performance considerations associated with sending all
	sending all WebRTC media through the proxy server.		WebRTC media through the proxy server.
	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		thwarts the typical drive-by enumeration attacks. The details of
	this consent are left to the implementation; one potential mechanism		this consent are left to the implementation; one potential mechanism
	is to tie this consent to getUserMedia consent.		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., data channel
	applications). However, user agents MAY choose a stricter default		applications). However, user agents MAY choose a stricter default
	skipping to change at page 6, line 10 ¶		skipping to change at page 6, line 24 ¶
	associated with the default route, Mode 2 and 3 will cause any		associated with the default route, Mode 2 and 3 will cause any
	external media traffic to go through the RETURN proxy. While the		external media traffic to go through the RETURN proxy. While the
	RETURN approach gives the best performance, a similar result can be		RETURN approach gives the best performance, a similar result can be
	achieved for non-RETURN proxies via an organization firewall policy		achieved for non-RETURN proxies via an organization firewall policy
	that only allows external WebRTC traffic to leave through the proxy		that only allows external WebRTC traffic to leave through the proxy
	(typically, over TCP). This provides a way to ensure the proxy is		(typically, over TCP). This provides a way to ensure the proxy is
	used for any external traffic, but avoids the performance issues of		used for any external traffic, but avoids the performance issues of
	Mode 4, where all media is forced through said proxy, for intra-		Mode 4, where all media is forced through said proxy, for intra-
	organization traffic.		organization traffic.
	5. Application Guidance		6. 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.
	o Future versions of browsers may present an indicator to signify		7. Security Considerations
	that the page is using WebRTC to set up a peer-to-peer connection.
	Applications MUST only use WebRTC in a fashion that is consistent
	with user expectations.
	6. Security Considerations
	This document is entirely devoted to security considerations.		This document is entirely devoted to security considerations.
	7. IANA Considerations		8. IANA Considerations
	This document requires no actions from IANA.		This document requires no actions from IANA.
	8. Acknowledgements		9. Acknowledgements
	Several people provided input into this document, including Harald		Several people provided input into this document, including Bernard
	Alvestrand, Ted Hardie, Matthew Kaufmann, Eric Rescorla, Adam Roach,		Aboba, Harald Alvestrand, Ted Hardie, Matthew Kaufmann, Eric
	and Martin Thomson.		Rescorla, Adam Roach, and Martin Thomson.
	9. Informative References		10. References
			10.1. Normative References
			[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
			Requirement Levels", BCP 14, RFC 2119,
			DOI 10.17487/RFC2119, March 1997,
			<http://www.rfc-editor.org/info/rfc2119>.
			10.2. Informative References
	[I-D.ietf-rtcweb-return]		[I-D.ietf-rtcweb-return]
	Schwartz, B. and J. Uberti, "Recursively Encapsulated TURN		Schwartz, B. and J. Uberti, "Recursively Encapsulated TURN
	(RETURN) for Connectivity and Privacy in WebRTC", draft-		(RETURN) for Connectivity and Privacy in WebRTC", draft-
	ietf-rtcweb-return-01 (work in progress), January 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>.		<http://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>.		<http://www.rfc-editor.org/info/rfc1919>.
	skipping to change at page 8, line 5 ¶		skipping to change at page 8, line 25 ¶
	[RFC6544] Rosenberg, J., Keranen, A., Lowekamp, B., and A. Roach,		[RFC6544] Rosenberg, J., Keranen, A., Lowekamp, B., and A. Roach,
	"TCP Candidates with Interactive Connectivity		"TCP Candidates with Interactive Connectivity
	Establishment (ICE)", RFC 6544, DOI 10.17487/RFC6544,		Establishment (ICE)", RFC 6544, DOI 10.17487/RFC6544,
	March 2012, <http://www.rfc-editor.org/info/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>.		<http://www.rfc-editor.org/info/rfc7016>.
			[RFC7230] Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer
			Protocol (HTTP/1.1): Message Syntax and Routing",
			RFC 7230, DOI 10.17487/RFC7230, June 2014,
			<http://www.rfc-editor.org/info/rfc7230>.
	Appendix A. Change log		Appendix A. Change log
			Changes in draft -04:
			o Rewording and cleanup in abstract, intro, and problem statement.
			o Added 2119 boilerplate.
			o Fixed weird reference spacing.
			o Expanded acronyms on first use.
			o Removed 8.8.8.8 mention.
			o Removed mention of future browser considerations.
	Changes in draft -03:		Changes in draft -03:
	o Clarified when to use which modes.		o Clarified when to use which modes.
	o Use 2119 qualifiers to make normative statements.		o Added 2119 qualifiers to make normative statements.
	o Defined 'proxy'.		o Defined 'proxy'.
	o Mentioned split tunnels in problem statement.		o Mentioned split tunnels in problem statement.
	Changes in draft -02:		Changes in draft -02:
	o Recommendations -> Requirements		o Recommendations -> Requirements
	o Updated text regarding consent.		o Updated text regarding consent.
End of changes. 30 change blocks.
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