draft-ietf-rtcweb-ip-handling-08.txt		draft-ietf-rtcweb-ip-handling-09.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: December 5, 2018 Facebook		Expires: December 15, 2018 Facebook
	June 3, 2018		June 13, 2018
	WebRTC IP Address Handling Requirements		WebRTC IP Address Handling Requirements
	draft-ietf-rtcweb-ip-handling-08		draft-ietf-rtcweb-ip-handling-09
	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 December 5, 2018.		This Internet-Draft will expire on December 15, 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
	skipping to change at page 5, line 8 ¶		skipping to change at page 5, line 8 ¶
	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 application		3. By default, WebRTC traffic should not be sent through proxy
	proxy servers, due to the media quality problems associated with		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 such proxies, as well as proxy performance		communicating with such proxies, as well as proxy performance
	issues that may result from proxying WebRTC's long-lived, high-		issues that may result from proxying WebRTC's long-lived, high-
	bandwidth connections. However, it should be possible to force		bandwidth connections. However, it should be possible to force
	WebRTC to 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. If an application TURN server		application's HTTP traffic. If an enterprise TURN server is
	is present, the preferred route MUST be through this TURN		present, the preferred route MUST be through this TURN
	server. Once an interface has been chosen, the private IPv4		server. Once an interface has been chosen, the private IPv4
	and IPv6 addresses associated with this interface MUST be		and IPv6 addresses associated with this interface MUST be
	discovered and provided to the application. This ensures		discovered and provided to the application. This ensures
	that direct connections can still be established in this		that direct connections can still be established in this
	mode.		mode.
	Mode 3: Default route only: This is the the same as Mode 2, except		Mode 3: Default route only: This is the the same as Mode 2, except
	that the associated private addresses MUST NOT be provided;		that the associated private addresses MUST NOT be provided;
	the only IP addresses gathered are those discovered via		the only IP addresses gathered are those discovered via
	mechanisms like STUN and TURN (on the default route). This		mechanisms like STUN and TURN (on the default route). This
	may cause traffic to hairpin through a NAT, fall back to an		may cause traffic to hairpin through a NAT, fall back to an
	application TURN server, or fail altogether, with resulting		application TURN server, or fail altogether, with resulting
	quality implications.		quality implications.
	Mode 4: Force proxy: This is the same as Mode 3, but when the		Mode 4: Force proxy: This is the same as Mode 3, but when the
	application's HTTP traffic is sent through an application		application's HTTP traffic is sent through a proxy, WebRTC
	proxy, WebRTC media traffic MUST also be proxied. If the		media traffic MUST also be proxied. If the proxy does not
	proxy does not support UDP (as is the case for all HTTP and		support UDP (as is the case for all HTTP and most SOCKS
	most SOCKS [RFC1928] proxies), or the WebRTC implementation		[RFC1928] proxies), or the WebRTC implementation does not
	does not support UDP proxying, the use of UDP will be		support UDP proxying, the use of UDP will be disabled, and
	disabled, and TCP will be used to send and receive media		TCP will be used to send and receive media through the
	through the proxy. Use of TCP will result in reduced media		proxy. Use of TCP will result in reduced media quality, in
	quality, in addition to any performance considerations		addition to any performance considerations associated with
	associated with sending all WebRTC media through the proxy		sending all WebRTC media through the proxy server.
	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.
	skipping to change at page 9, line 42 ¶		skipping to change at page 9, line 42 ¶
	Time Communication Use Cases and Requirements", RFC 7478,		Time Communication Use Cases and Requirements", RFC 7478,
	DOI 10.17487/RFC7478, March 2015,		DOI 10.17487/RFC7478, March 2015,
	<https://www.rfc-editor.org/info/rfc7478>.		<https://www.rfc-editor.org/info/rfc7478>.
	[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 -09:
			o Fixed confusing text regarding enterprise TURN servers.
	Changes in draft -08:		Changes in draft -08:
	o Discuss how enterprise TURN servers should be handled.		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:
End of changes. 7 change blocks.
	18 lines changed or deleted		21 lines changed or added
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