draft-ietf-rtcweb-ip-handling-07.txt		draft-ietf-rtcweb-ip-handling-08.txt
	Network Working Group J. Uberti		Network Working Group J. Uberti
	Internet-Draft Google		Internet-Draft Google
	Intended status: Standards Track G. Shieh		Intended status: Standards Track G. Shieh
	Expires: October 20, 2018 Facebook		Expires: December 5, 2018 Facebook
	April 18, 2018		June 3, 2018
	WebRTC IP Address Handling Requirements		WebRTC IP Address Handling Requirements
	draft-ietf-rtcweb-ip-handling-07		draft-ietf-rtcweb-ip-handling-08
	Abstract		Abstract
	This document provides information and requirements for how IP		This document provides information and requirements for how IP
	addresses should be handled by WebRTC implementations.		addresses should be handled by WebRTC implementations.
	Status of This Memo		Status of This Memo
	This Internet-Draft is submitted in full conformance with the		This Internet-Draft is submitted in full conformance with the
	provisions of BCP 78 and BCP 79.		provisions of BCP 78 and BCP 79.
	skipping to change at page 1, line 32 ¶		skipping to change at page 1, line 32 ¶
	Internet-Drafts are working documents of the Internet Engineering		Internet-Drafts are working documents of the Internet Engineering
	Task Force (IETF). Note that other groups may also distribute		Task Force (IETF). Note that other groups may also distribute
	working documents as Internet-Drafts. The list of current Internet-		working documents as Internet-Drafts. The list of current Internet-
	Drafts is at https://datatracker.ietf.org/drafts/current/.		Drafts is at https://datatracker.ietf.org/drafts/current/.
	Internet-Drafts are draft documents valid for a maximum of six months		Internet-Drafts are draft documents valid for a maximum of six months
	and may be updated, replaced, or obsoleted by other documents at any		and may be updated, replaced, or obsoleted by other documents at any
	time. It is inappropriate to use Internet-Drafts as reference		time. It is inappropriate to use Internet-Drafts as reference
	material or to cite them other than as "work in progress."		material or to cite them other than as "work in progress."
	This Internet-Draft will expire on October 20, 2018.		This Internet-Draft will expire on December 5, 2018.
	Copyright Notice		Copyright Notice
	Copyright (c) 2018 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
	(https://trustee.ietf.org/license-info) in effect on the date of		(https://trustee.ietf.org/license-info) in effect on the date of
	publication of this document. Please review these documents		publication of this document. Please review these documents
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	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
	5.1. Principles . . . . . . . . . . . . . . . . . . . . . . . 4		5.1. Principles . . . . . . . . . . . . . . . . . . . . . . . 4
	5.2. Modes and Recommendations . . . . . . . . . . . . . . . . 5		5.2. Modes and Recommendations . . . . . . . . . . . . . . . . 5
	6. Implementation Guidance . . . . . . . . . . . . . . . . . . . 6		6. Implementation Guidance . . . . . . . . . . . . . . . . . . . 6
	6.1. Ensuring Normal Routing . . . . . . . . . . . . . . . . . 6		6.1. Ensuring Normal Routing . . . . . . . . . . . . . . . . . 6
	6.2. Determining Host Candidates . . . . . . . . . . . . . . . 6		6.2. Determining Host Candidates . . . . . . . . . . . . . . . 7
	7. Application Guidance . . . . . . . . . . . . . . . . . . . . 7		7. Application Guidance . . . . . . . . . . . . . . . . . . . . 7
	8. Security Considerations . . . . . . . . . . . . . . . . . . . 7		8. Security Considerations . . . . . . . . . . . . . . . . . . . 7
	9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 7		9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 8
	10. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 7		10. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 8
	11. References . . . . . . . . . . . . . . . . . . . . . . . . . 8		11. References . . . . . . . . . . . . . . . . . . . . . . . . . 8
	11.1. Normative References . . . . . . . . . . . . . . . . . . 8		11.1. Normative References . . . . . . . . . . . . . . . . . . 8
	11.2. Informative References . . . . . . . . . . . . . . . . . 8		11.2. Informative References . . . . . . . . . . . . . . . . . 8
	Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 9		Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 9
	Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 10		Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 11
	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 connection attempts from 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
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	interfaces (a "split-tunnel" VPN), WebRTC will discover not only		interfaces (a "split-tunnel" VPN), WebRTC will discover not only
	the public address for the VPN, but also the ISP public address		the public address for the VPN, but also the ISP public address
	over which the VPN is running.		over which the VPN is running.
	2. If the client is behind a Network Address Translator (NAT), the		2. If the client is behind a Network Address Translator (NAT), the
	client's private IP addresses, often [RFC1918] addresses, can be		client's private IP addresses, often [RFC1918] addresses, can be
	learned.		learned.
	3. If the client is behind a proxy (a client-configured "classical		3. If the client is behind a proxy (a client-configured "classical
	application proxy", as defined in [RFC1919], Section 3), but		application proxy", as defined in [RFC1919], Section 3), but
	direct access to the Internet is also supported, WebRTC's STUN		direct access to the Internet is permitted, WebRTC's STUN checks
	checks will bypass the proxy and reveal the public IP address of		will bypass the proxy and reveal the public IP address of the
	the client.		client. This concern also applies to the "enterprise TURN
			server" scenario described in [RFC7478], Section 2.3.5.1, if, as
			above, direct Internet access is permitted. However, when the
			term "proxy" is used in this document, it is always in reference
			to an [RFC1919] proxy server.
	Of these three concerns, #1 is the most significant, because for some		Of these three concerns, #1 is the most significant, because for some
	users, the purpose of using a VPN is for anonymity. However,		users, the purpose of using a VPN is for anonymity. However,
	different VPN users will have different needs, and some VPN users		different VPN users will have different needs, and some VPN users
	(e.g., corporate VPN users) may in fact prefer WebRTC to send media		(e.g., corporate VPN users) may in fact prefer WebRTC to send media
	traffic directly, i.e., not through the VPN.		traffic directly, i.e., not through the VPN.
	#2 is considered to be a less significant concern, given that the		#2 is considered to be a less 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., the		192.168.0.2), or have built-in privacy protection (e.g., the
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	information in a way that might violate user expectations.		information in a way that might violate user expectations.
	5. Detailed Design		5. Detailed Design
	5.1. Principles		5.1. Principles
	The key principles for our framework are stated below:		The key principles for our framework are stated below:
	1. By default, WebRTC traffic should follow typical IP routing,		1. By default, WebRTC traffic should follow typical IP routing,
	i.e., WebRTC should use the same interface used for HTTP traffic,		i.e., WebRTC should use the same interface used for HTTP traffic,
	and only the system's 'typical' public addresses should be		and only the system's 'typical' public addresses (or those of an
	visible to the application. However, in the interest of optimal		enterprise TURN server, if present) should be visible to the
	media quality, it should be possible to enable WebRTC to make use		application. However, in the interest of optimal media quality,
	of all network interfaces to determine the ideal route.		it should be possible to enable WebRTC to make use of all network
			interfaces to determine the ideal route.
	2. By default, WebRTC should be able to negotiate direct peer-to-		2. By default, WebRTC should be able to negotiate direct peer-to-
	peer connections between endpoints (i.e., without traversing a		peer connections between endpoints (i.e., without traversing a
	NAT or relay server), by providing a minimal set of local IP		NAT or relay server), by providing a minimal set of local IP
	addresses to the application for use in the ICE process. This		addresses to the application for use in the ICE process. This
	ensures that applications that need true peer-to-peer routing for		ensures that applications that need true peer-to-peer routing for
	bandwidth or latency reasons can operate successfully. However,		bandwidth or latency reasons can operate successfully. However,
	it should be possible to suppress these addresses (with the		it should be possible to suppress these addresses (with the
	resultant impact on direct connections) if desired.		resultant impact on direct connections) if desired.
	3. By default, WebRTC traffic should not be sent through proxy		3. By default, WebRTC traffic should not be sent through application
	servers, due to the media quality problems associated with		proxy servers, due to the media quality problems associated with
	sending WebRTC traffic over TCP, which is almost always used when		sending WebRTC traffic over TCP, which is almost always used when
	communicating with proxies, as well as proxy performance issues		communicating with such proxies, as well as proxy performance
	that may result from proxying WebRTC's long-lived, high-bandwidth		issues that may result from proxying WebRTC's long-lived, high-
	connections. However, it should be possible to force WebRTC to		bandwidth connections. However, it should be possible to force
	send its traffic through a configured proxy if desired.		WebRTC to send its traffic through a configured proxy if desired.
	5.2. Modes and Recommendations		5.2. Modes and Recommendations
	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 use all network		Mode 1: Enumerate all addresses: WebRTC MUST use all network
	interfaces to attempt communication with STUN servers, TURN		interfaces to attempt communication with STUN servers, TURN
	servers, or peers. This will converge on the best media		servers, or peers. This will converge on the best media
	path, and is ideal when media performance is the highest		path, and is ideal when media performance is the highest
	priority, but it discloses the most information.		priority, but it discloses the most information.
	Mode 2: Default route + associated local addresses: WebRTC MUST		Mode 2: Default route + associated local addresses: WebRTC MUST
	follow the kernel routing table rules, which will typically		follow the kernel routing table rules, which will typically
	cause media packets to take the same route as the		cause media packets to take the same route as the
	application's HTTP traffic. In addition, the private IPv4		application's HTTP traffic. If an application TURN server
	and IPv6 addresses associated with the kernel-chosen		is present, the preferred route MUST be through this TURN
	interface MUST be discovered and provided to the		server. Once an interface has been chosen, the private IPv4
	application. This ensures that direct connections can still		and IPv6 addresses associated with this interface MUST be
	be established in this mode.		discovered and provided to the application. This ensures
			that direct 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 addresses MUST NOT be provided;		that the associated private addresses MUST NOT be provided;
	the only IP addresses gathered are those discovered via		the only IP addresses gathered are those discovered via
	mechanisms like STUN and TURN (on the default route). This		mechanisms like STUN and TURN (on the default route). This
	may cause traffic to hairpin through a NAT, fall back to an		may cause traffic to hairpin through a NAT, fall back to an
	application TURN server, or fail altogether, with resulting		application TURN server, or fail altogether, with resulting
	quality implications.		quality implications.
	Mode 4: Force proxy: This is the same as Mode 3, but all WebRTC		Mode 4: Force proxy: This is the same as Mode 3, but when the
	media traffic is forced through a proxy, if one is		application's HTTP traffic is sent through an application
	configured. If the proxy does not support UDP (as is the		proxy, WebRTC media traffic MUST also be proxied. If the
	case for all HTTP and most SOCKS [RFC1928] proxies), or the		proxy does not support UDP (as is the case for all HTTP and
	WebRTC implementation does not support UDP proxying, the use		most SOCKS [RFC1928] proxies), or the WebRTC implementation
	of UDP will be disabled, and TCP will be used to send and		does not support UDP proxying, the use of UDP will be
	receive media through the proxy. Use of TCP will result in		disabled, and TCP will be used to send and receive media
	reduced media quality, in addition to any performance		through the proxy. Use of TCP will result in reduced media
	considerations associated with sending all WebRTC media		quality, in addition to any performance considerations
	through the proxy server.		associated with sending all WebRTC media through the proxy
			server.
	Mode 1 MUST only be used when user consent has been provided. The		Mode 1 MUST only be used when user consent has been provided. The
	details of this consent are left to the implementation; one potential		details of this consent are left to the implementation; one potential
	mechanism is to tie this consent to getUserMedia consent.		mechanism is to tie this consent to getUserMedia consent.
	Alternatively, implementations can provide a specific mechanism to		Alternatively, implementations can provide a specific mechanism to
	obtain user consent.		obtain user consent.
	In cases where user consent has not been obtained, Mode 2 SHOULD be		In cases where user consent has not been obtained, Mode 2 SHOULD be
	used.		used.
	These defaults provide a reasonable tradeoff that permits trusted		These defaults provide a reasonable tradeoff that permits trusted
	WebRTC applications to achieve optimal network performance, but gives		WebRTC applications to achieve optimal network performance, but gives
	applications without consent (e.g., 1-way streaming or data channel		applications without consent (e.g., 1-way streaming or data channel
	applications) only the minimum information needed to achieve direct		applications) only the minimum information needed to achieve direct
	skipping to change at page 6, line 21 ¶		skipping to change at page 6, line 29 ¶
	These defaults provide a reasonable tradeoff that permits trusted		These defaults provide a reasonable tradeoff that permits trusted
	WebRTC applications to achieve optimal network performance, but gives		WebRTC applications to achieve optimal network performance, but gives
	applications without consent (e.g., 1-way streaming or data channel		applications without consent (e.g., 1-way streaming or data channel
	applications) only the minimum information needed to achieve direct		applications) only the minimum information needed to achieve direct
	connections, as defined in Mode 2. However, implementations MAY		connections, as defined in Mode 2. However, implementations MAY
	choose stricter modes if desired, e.g., if a user indicates they want		choose stricter modes if desired, e.g., if a user indicates they want
	all WebRTC traffic to follow the default route.		all WebRTC traffic to follow the default route.
	Note that the suggested defaults can still be used even for		Note that the suggested defaults can still be used even for
	organizations that want all external WebRTC traffic to traverse a		organizations that want all external WebRTC traffic to traverse a
	proxy, simply by setting an organizational firewall policy that		proxy or enterprise TURN server, simply by setting an organizational
	allows WebRTC traffic to only leave through the proxy. This provides		firewall policy that allows WebRTC traffic to only leave through the
	a way to ensure the proxy is used for any external traffic, but		proxy or TURN server. This provides a way to ensure the proxy or
	avoids the performance issues associated with Mode 4 (where all media		TURN server is used for any external traffic, but still allows direct
	is forced through said proxy) for intra-organization traffic.		connections (and, in the proxy case, avoids the performance issues
			associated with forcing media through said proxy) for intra-
			organization traffic.
	6. Implementation Guidance		6. Implementation Guidance
	This section provides guidance to WebRTC implementations on how to		This section provides guidance to WebRTC implementations on how to
	implement the policies described above.		implement the policies described above.
	6.1. Ensuring Normal Routing		6.1. Ensuring Normal Routing
	When trying to follow typical IP routing, the simplest approach is to		When trying to follow typical IP routing, the simplest approach is to
	bind the sockets used for peer-to-peer connections to the wildcard		bind the sockets used for peer-to-peer connections to the wildcard
	skipping to change at page 9, line 25 ¶		skipping to change at page 9, line 31 ¶
	[RFC7016] Thornburgh, M., "Adobe's Secure Real-Time Media Flow		[RFC7016] Thornburgh, M., "Adobe's Secure Real-Time Media Flow
	Protocol", RFC 7016, DOI 10.17487/RFC7016, November 2013,		Protocol", RFC 7016, DOI 10.17487/RFC7016, November 2013,
	<https://www.rfc-editor.org/info/rfc7016>.		<https://www.rfc-editor.org/info/rfc7016>.
	[RFC7230] Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer		[RFC7230] Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer
	Protocol (HTTP/1.1): Message Syntax and Routing",		Protocol (HTTP/1.1): Message Syntax and Routing",
	RFC 7230, DOI 10.17487/RFC7230, June 2014,		RFC 7230, DOI 10.17487/RFC7230, June 2014,
	<https://www.rfc-editor.org/info/rfc7230>.		<https://www.rfc-editor.org/info/rfc7230>.
			[RFC7478] Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-
			Time Communication Use Cases and Requirements", RFC 7478,
			DOI 10.17487/RFC7478, March 2015,
			<https://www.rfc-editor.org/info/rfc7478>.
	[RFC8089] Kerwin, M., "The "file" URI Scheme", RFC 8089,		[RFC8089] Kerwin, M., "The "file" URI Scheme", RFC 8089,
	DOI 10.17487/RFC8089, February 2017,		DOI 10.17487/RFC8089, February 2017,
	<https://www.rfc-editor.org/info/rfc8089>.		<https://www.rfc-editor.org/info/rfc8089>.
	Appendix A. Change log		Appendix A. Change log
			Changes in draft -08:
			o Discuss how enterprise TURN servers should be handled.
	Changes in draft -07:		Changes in draft -07:
	o Clarify consent guidance.		o Clarify consent guidance.
	Changes in draft -06:		Changes in draft -06:
	o Clarify recommendations.		o Clarify recommendations.
	o Split implementation guidance into two sections.		o Split implementation guidance into two sections.
End of changes. 16 change blocks.
	42 lines changed or deleted		60 lines changed or added
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