draft-ietf-rtcweb-security-00.txt		draft-ietf-rtcweb-security-01.txt
	RTC-Web E. Rescorla		RTC-Web E. Rescorla
	Internet-Draft RTFM, Inc.		Internet-Draft RTFM, Inc.
	Intended status: Standards Track September 21, 2011		Intended status: Standards Track October 30, 2011
	Expires: March 24, 2012		Expires: May 2, 2012
	Security Considerations for RTC-Web		Security Considerations for RTC-Web
	draft-ietf-rtcweb-security-00		draft-ietf-rtcweb-security-01
	Abstract		Abstract
	The Real-Time Communications on the Web (RTC-Web) working group is		The Real-Time Communications on the Web (RTC-Web) working group is
	tasked with standardizing protocols for real-time communications		tasked with standardizing protocols for real-time communications
	between Web browsers. The major use cases for RTC-Web technology are		between Web browsers. The major use cases for RTC-Web technology are
	real-time audio and/or video calls, Web conferencing, and direct data		real-time audio and/or video calls, Web conferencing, and direct data
	transfer. Unlike most conventional real-time systems (e.g., SIP-		transfer. Unlike most conventional real-time systems (e.g., SIP-
	based soft phones) RTC-Web communications are directly controlled by		based soft phones) RTC-Web communications are directly controlled by
	some Web server, which poses new security challenges. For instance,		some Web server, which poses new security challenges. For instance,
	skipping to change at page 2, line 7		skipping to change at page 2, line 7
	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 March 24, 2012.		This Internet-Draft will expire on May 2, 2012.
	Copyright Notice		Copyright Notice
	Copyright (c) 2011 IETF Trust and the persons identified as the		Copyright (c) 2011 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
	publication of this document. Please review these documents		publication of this document. Please review these documents
	skipping to change at page 3, line 7		skipping to change at page 3, line 7
	modifications of such material outside the IETF Standards Process.		modifications of such material outside the IETF Standards Process.
	Without obtaining an adequate license from the person(s) controlling		Without obtaining an adequate license from the person(s) controlling
	the copyright in such materials, this document may not be modified		the copyright in such materials, this document may not be modified
	outside the IETF Standards Process, and derivative works of it may		outside the IETF Standards Process, and derivative works of it may
	not be created outside the IETF Standards Process, except to format		not be created outside the IETF Standards Process, except to format
	it for publication as an RFC or to translate it into languages other		it for publication as an RFC or to translate it into languages other
	than English.		than English.
	Table of Contents		Table of Contents
	1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4		1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 5
	2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 5		2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 6
	3. The Browser Threat Model . . . . . . . . . . . . . . . . . . . 5		3. The Browser Threat Model . . . . . . . . . . . . . . . . . . . 6
	3.1. Access to Local Resources . . . . . . . . . . . . . . . . 6		3.1. Access to Local Resources . . . . . . . . . . . . . . . . 7
	3.2. Same Origin Policy . . . . . . . . . . . . . . . . . . . . 6		3.2. Same Origin Policy . . . . . . . . . . . . . . . . . . . . 7
	3.3. Bypassing SOP: CORS, WebSockets, and consent to		3.3. Bypassing SOP: CORS, WebSockets, and consent to
	communicate . . . . . . . . . . . . . . . . . . . . . . . 7		communicate . . . . . . . . . . . . . . . . . . . . . . . 8
	4. Security for RTC-Web Applications . . . . . . . . . . . . . . 7		4. Security for RTC-Web Applications . . . . . . . . . . . . . . 8
	4.1. Access to Local Devices . . . . . . . . . . . . . . . . . 7		4.1. Access to Local Devices . . . . . . . . . . . . . . . . . 8
	4.1.1. Calling Scenarios and User Expectations . . . . . . . 8		4.1.1. Calling Scenarios and User Expectations . . . . . . . 9
	4.1.1.1. Dedicated Calling Services . . . . . . . . . . . . 8		4.1.1.1. Dedicated Calling Services . . . . . . . . . . . . 9
	4.1.1.2. Calling the Site You're On . . . . . . . . . . . . 8		4.1.1.2. Calling the Site You're On . . . . . . . . . . . . 9
	4.1.1.3. Calling to an Ad Target . . . . . . . . . . . . . 9		4.1.1.3. Calling to an Ad Target . . . . . . . . . . . . . 10
	4.1.2. Origin-Based Security . . . . . . . . . . . . . . . . 9		4.1.2. Origin-Based Security . . . . . . . . . . . . . . . . 10
	4.1.3. Security Properties of the Calling Page . . . . . . . 11		4.1.3. Security Properties of the Calling Page . . . . . . . 12
	4.2. Communications Consent Verification . . . . . . . . . . . 12		4.2. Communications Consent Verification . . . . . . . . . . . 13
	4.2.1. ICE . . . . . . . . . . . . . . . . . . . . . . . . . 12		4.2.1. ICE . . . . . . . . . . . . . . . . . . . . . . . . . 13
	4.2.2. Masking . . . . . . . . . . . . . . . . . . . . . . . 12		4.2.2. Masking . . . . . . . . . . . . . . . . . . . . . . . 14
	4.2.3. Backward Compatibility . . . . . . . . . . . . . . . . 13		4.2.3. Backward Compatibility . . . . . . . . . . . . . . . . 14
	4.3. Communications Security . . . . . . . . . . . . . . . . . 14		4.2.4. IP Location Privacy . . . . . . . . . . . . . . . . . 15
	4.3.1. Protecting Against Retrospective Compromise . . . . . 15		4.3. Communications Security . . . . . . . . . . . . . . . . . 15
	4.3.2. Protecting Against During-Call Attack . . . . . . . . 15		4.3.1. Protecting Against Retrospective Compromise . . . . . 16
	4.3.2.1. Key Continuity . . . . . . . . . . . . . . . . . . 16		4.3.2. Protecting Against During-Call Attack . . . . . . . . 17
	4.3.2.2. Short Authentication Strings . . . . . . . . . . . 16		4.3.2.1. Key Continuity . . . . . . . . . . . . . . . . . . 17
	4.3.2.3. Recommendations . . . . . . . . . . . . . . . . . 17		4.3.2.2. Short Authentication Strings . . . . . . . . . . . 18
	5. Security Considerations . . . . . . . . . . . . . . . . . . . 17		4.3.2.3. Recommendations . . . . . . . . . . . . . . . . . 19
	6. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 18		5. Security Considerations . . . . . . . . . . . . . . . . . . . 19
	7. References . . . . . . . . . . . . . . . . . . . . . . . . . . 18		6. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 19
	7.1. Normative References . . . . . . . . . . . . . . . . . . . 18		7. References . . . . . . . . . . . . . . . . . . . . . . . . . . 19
	7.2. Informative References . . . . . . . . . . . . . . . . . . 18		7.1. Normative References . . . . . . . . . . . . . . . . . . . 19
	Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 20		7.2. Informative References . . . . . . . . . . . . . . . . . . 20
			Appendix A. A Proposed Security Architecture [No Consensus on
			This] . . . . . . . . . . . . . . . . . . . . . . . . 22
			A.1. Trust Hierarchy . . . . . . . . . . . . . . . . . . . . . 22
			A.1.1. Authenticated Entities . . . . . . . . . . . . . . . . 22
			A.1.2. Unauthenticated Entities . . . . . . . . . . . . . . . 23
			A.2. Overview . . . . . . . . . . . . . . . . . . . . . . . . . 23
			A.2.1. Initial Signaling . . . . . . . . . . . . . . . . . . 24
			A.2.2. Media Consent Verification . . . . . . . . . . . . . . 26
			A.2.3. DTLS Handshake . . . . . . . . . . . . . . . . . . . . 26
			A.2.4. Communications and Consent Freshness . . . . . . . . . 27
			A.3. Detailed Technical Description . . . . . . . . . . . . . . 27
			A.3.1. Origin and Web Security Issues . . . . . . . . . . . . 27
			A.3.2. Device Permissions Model . . . . . . . . . . . . . . . 28
			A.3.3. Communications Consent . . . . . . . . . . . . . . . . 29
			A.3.4. IP Location Privacy . . . . . . . . . . . . . . . . . 29
			A.3.5. Communications Security . . . . . . . . . . . . . . . 30
			A.3.6. Web-Based Peer Authentication . . . . . . . . . . . . 31
			A.3.6.1. Generic Concepts . . . . . . . . . . . . . . . . . 31
			A.3.6.2. BrowserID . . . . . . . . . . . . . . . . . . . . 32
			A.3.6.3. OAuth . . . . . . . . . . . . . . . . . . . . . . 35
			A.3.6.4. Generic Identity Support . . . . . . . . . . . . . 36
			Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 36
	1. Introduction		1. Introduction
	The Real-Time Communications on the Web (RTC-Web) working group is		The Real-Time Communications on the Web (RTC-Web) working group is
	tasked with standardizing protocols for real-time communications		tasked with standardizing protocols for real-time communications
	between Web browsers. The major use cases for RTC-Web technology are		between Web browsers. The major use cases for RTC-Web technology are
	real-time audio and/or video calls, Web conferencing, and direct data		real-time audio and/or video calls, Web conferencing, and direct data
	transfer. Unlike most conventional real-time systems, (e.g., SIP-		transfer. Unlike most conventional real-time systems, (e.g., SIP-
	based[RFC3261] soft phones) RTC-Web communications are directly		based[RFC3261] soft phones) RTC-Web communications are directly
	controlled by some Web server. A simple case is shown below.		controlled by some Web server. A simple case is shown below.
	skipping to change at page 6, line 40		skipping to change at page 7, line 40
	induced to download executable files and run them).		induced to download executable files and run them).
	3.2. Same Origin Policy		3.2. Same Origin Policy
	Many other resources are accessible but isolated. For instance,		Many other resources are accessible but isolated. For instance,
	while scripts are allowed to make HTTP requests via the		while scripts are allowed to make HTTP requests via the
	XMLHttpRequest() API those requests are not allowed to be made to any		XMLHttpRequest() API those requests are not allowed to be made to any
	server, but rather solely to the same ORIGIN from whence the script		server, but rather solely to the same ORIGIN from whence the script
	came.[I-D.abarth-origin] (although CORS [CORS] and WebSockets		came.[I-D.abarth-origin] (although CORS [CORS] and WebSockets
	[I-D.ietf-hybi-thewebsocketprotocol] provides a escape hatch from		[I-D.ietf-hybi-thewebsocketprotocol] provides a escape hatch from
	this restriction, as described below. This SAME ORIGIN POLICY (SOP)		this restriction, as described below.) This SAME ORIGIN POLICY (SOP)
	prevents server A from mounting attacks on server B via the user's		prevents server A from mounting attacks on server B via the user's
	browser, which protects both the user (e.g., from misuse of his		browser, which protects both the user (e.g., from misuse of his
	credentials) and the server (e.g., from DoS attack).		credentials) and the server (e.g., from DoS attack).
	More generally, SOP forces scripts from each site to run in their		More generally, SOP forces scripts from each site to run in their
	own, isolated, sandboxes. While there are techniques to allow them		own, isolated, sandboxes. While there are techniques to allow them
	to interact, those interactions generally must be mutually consensual		to interact, those interactions generally must be mutually consensual
	(by each site) and are limited to certain channels. For instance,		(by each site) and are limited to certain channels. For instance,
	multiple pages/browser panes from the same origin can read each		multiple pages/browser panes from the same origin can read each
	other's JS variables, but pages from the different origins--or even		other's JS variables, but pages from the different origins--or even
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	generate traffic which resembles a given protocol.		generate traffic which resembles a given protocol.
	4. Security for RTC-Web Applications		4. Security for RTC-Web Applications
	4.1. Access to Local Devices		4.1. Access to Local Devices
	As discussed in Section 1, allowing arbitrary sites to initiate calls		As discussed in Section 1, allowing arbitrary sites to initiate calls
	violates the core Web security guarantee; without some access		violates the core Web security guarantee; without some access
	restrictions on local devices, any malicious site could simply bug a		restrictions on local devices, any malicious site could simply bug a
	user. At minimum, then, it MUST NOT be possible for arbitrary sites		user. At minimum, then, it MUST NOT be possible for arbitrary sites
	to initiate calls to arbitrary location without user consent. This		to initiate calls to arbitrary locations without user consent. This
	immediately raises the question, however, of what should be the scope		immediately raises the question, however, of what should be the scope
	of user consent.		of user consent.
	For the rest of this discussion we assume that the user is somehow		For the rest of this discussion we assume that the user is somehow
	going to grant consent to some entity (e.g., a social networking		going to grant consent to some entity (e.g., a social networking
	site) to initiate a call on his behalf. This consent may be limited		site) to initiate a call on his behalf. This consent may be limited
	to a single call or may be a general consent. In order for the user		to a single call or may be a general consent. In order for the user
	to make an intelligent decision about whether to allow a call (and		to make an intelligent decision about whether to allow a call (and
	hence his camera and microphone input to be routed somewhere), he		hence his camera and microphone input to be routed somewhere), he
	must understand either who is request access, where the media is		must understand either who is requesting access, where the media is
	going, or both. So, for instance, one might imagine that at the time		going, or both. So, for instance, one might imagine that at the time
	access to camera and microphone is requested, the user is shown a		access to camera and microphone is requested, the user is shown a
	dialog that says "site X has requested access to camera and		dialog that says "site X has requested access to camera and
	microphone, yes or no" (though note that this type of in-flow		microphone, yes or no" (though note that this type of in-flow
	interface violates one of the guidelines in Section Section 3). The		interface violates one of the guidelines in Section 3). The user's
	user's decision will of course be based on his opinion of Site X.		decision will of course be based on his opinion of Site X. However,
	However, as discussed below, this is a complicated concept.		as discussed below, this is a complicated concept.
	4.1.1. Calling Scenarios and User Expectations		4.1.1. Calling Scenarios and User Expectations
	While a large number of possible calling scenarios are possible, the		While a large number of possible calling scenarios are possible, the
	scenarios discussed in this section illustrate many of the		scenarios discussed in this section illustrate many of the
	difficulties of identifying the relevant scope of consent.		difficulties of identifying the relevant scope of consent.
	4.1.1.1. Dedicated Calling Services		4.1.1.1. Dedicated Calling Services
	The first scenario we consider is a dedicated calling service. In		The first scenario we consider is a dedicated calling service. In
	skipping to change at page 9, line 12		skipping to change at page 10, line 12
	The paradigmatic case here is the "click here to talk to a		The paradigmatic case here is the "click here to talk to a
	representative" windows that appear on many shopping sites. In this		representative" windows that appear on many shopping sites. In this
	case, the user's expectation is that they are calling the site		case, the user's expectation is that they are calling the site
	they're actually visiting. However, it is unlikely that they want to		they're actually visiting. However, it is unlikely that they want to
	provide a general consent to such a site; just because I want some		provide a general consent to such a site; just because I want some
	information on a car doesn't mean that I want the car manufacturer to		information on a car doesn't mean that I want the car manufacturer to
	be able to activate my microphone whenever they please. Thus, this		be able to activate my microphone whenever they please. Thus, this
	suggests the need for a second consent mechanism where I only grant		suggests the need for a second consent mechanism where I only grant
	consent for the duration of a given call. As described in		consent for the duration of a given call. As described in
	Section 3.1, great care must be taken in the design of this interface		Section 3.1, great care must be taken in the design of this interface
	to avoid the users just clicking through.		to avoid the users just clicking through. Note also that the user
			interface chrome must clearly display elements showing that the call
			is continuing in order to avoid attacks where the calling site just
			leaves it up indefinitely but shows a Web UI that implies otherwise.
	4.1.1.3. Calling to an Ad Target		4.1.1.3. Calling to an Ad Target
	In both of the previous cases, the user has a direct relationship		In both of the previous cases, the user has a direct relationship
	(though perhaps a transient one) with the target of the call.		(though perhaps a transient one) with the target of the call.
	Moreover, in both cases he is actually visiting the site of the		Moreover, in both cases he is actually visiting the site of the
	person he is being asked to trust. However, this is not always so.		person he is being asked to trust. However, this is not always so.
	Consider the case where a user is a visiting a content site which		Consider the case where a user is a visiting a content site which
	hosts an advertisement with an invitation to call for more		hosts an advertisement with an invitation to call for more
	information. When the user clicks the ad, they are connected with		information. When the user clicks the ad, they are connected with
	skipping to change at page 10, line 31		skipping to change at page 11, line 34
	Cryptographic Consent		Cryptographic Consent
	Only allow calls to a given set of peer keying material or to a		Only allow calls to a given set of peer keying material or to a
	cryptographically established identity.		cryptographically established identity.
	Unfortunately, none of these approaches is satisfactory for all		Unfortunately, none of these approaches is satisfactory for all
	cases. As discussed above, individual consent puts the user's		cases. As discussed above, individual consent puts the user's
	approval in the UI flow for every call. Not only does this quickly		approval in the UI flow for every call. Not only does this quickly
	become annoying but it can train the user to simply click "OK", at		become annoying but it can train the user to simply click "OK", at
	which point the consent becomes useless. Thus, while it may be		which point the consent becomes useless. Thus, while it may be
	necssary to have individual consent in some case, this is not a		necessary to have individual consent in some case, this is not a
	suitable solution for (for instance) the calling service case. Where		suitable solution for (for instance) the calling service case. Where
	necessary, in-flow user interfaces must be carefully designed to		necessary, in-flow user interfaces must be carefully designed to
	avoid the risk of the user blindly clicking through.		avoid the risk of the user blindly clicking through.
	The other two options are designed to restrict calls to a given		The other two options are designed to restrict calls to a given
	target. Unfortunately, Callee-oriented consent does not work well		target. Unfortunately, Callee-oriented consent does not work well
	because a malicious site can claim that the user is calling any user		because a malicious site can claim that the user is calling any user
	of his choice. One fix for this is to tie calls to a		of his choice. One fix for this is to tie calls to a
	cryptographically established identity. While not suitable for all		cryptographically established identity. While not suitable for all
	cases, this approach may be useful for some. If we consider the		cases, this approach may be useful for some. If we consider the
	advertising case described in Section 4.1.1.3, it's not particularly		advertising case described in Section 4.1.1.3, it's not particularly
	convenient to require the advertiser to instantiate an iframe on the		convenient to require the advertiser to instantiate an iframe on the
	hosting site just to get permission; a more convenient approach is to		hosting site just to get permission; a more convenient approach is to
	cryptographically tie the advertiser's certificate to the		cryptographically tie the advertiser's certificate to the
	communication directly. We're still tieing permissions to origin		communication directly. We're still tying permissions to origin
	here, but to the media origin (and-or destination) rather than to the		here, but to the media origin (and-or destination) rather than to the
	Web origin.		Web origin.
	Another case where media-level cryptographic identity makes sense is		Another case where media-level cryptographic identity makes sense is
	when a user really does not trust the calling site. For instance, I		when a user really does not trust the calling site. For instance, I
	might be worried that the calling service will attempt to bug my		might be worried that the calling service will attempt to bug my
	computer, but I also want to be able to conveniently call my friends.		computer, but I also want to be able to conveniently call my friends.
	If consent is tied to particular communications endpoints, then my		If consent is tied to particular communications endpoints, then my
	risk is limited. However, this is also not that convenient an		risk is limited. However, this is also not that convenient an
	interface, since managing individual user permissions can be painful.		interface, since managing individual user permissions can be painful.
	While this is primarily a question not for IETF, it should be clear		While this is primarily a question not for IETF, it should be clear
	that there is no really good answer. In general, if you cannot trust		that there is no really good answer. In general, if you cannot trust
	the site which you have authorized for calling not to bug you then		the site which you have authorized for calling not to bug you then
	your security situation is not really ideal. It is RECOMMENDED that		your security situation is not really ideal. It is RECOMMENDED that
	browsers have explicit (and obvious) indicators that they are in a		browsers have explicit (and obvious) indicators that they are in a
	call in order to mitigate this risk.		call in order to mitigate this risk.
	4.1.3. Security Properties of the Calling Page		4.1.3. Security Properties of the Calling Page
	Origin-based security is intended to security against web attackers.		Origin-based security is intended to secure against web attackers.
	However, we must also consider the case of network attackers.		However, we must also consider the case of network attackers.
	Consider the case where I have granted permission to a calling		Consider the case where I have granted permission to a calling
	service by an origin that has the HTTP scheme, e.g.,		service by an origin that has the HTTP scheme, e.g.,
	http://calling-service.example.com. If I ever use my computer on an		http://calling-service.example.com. If I ever use my computer on an
	unsecured network (e.g., a hotspot or if my own home wireless network		unsecured network (e.g., a hotspot or if my own home wireless network
	is insecure), and browse any HTTP site, then an attacker can bug my		is insecure), and browse any HTTP site, then an attacker can bug my
	computer. The attack proceeds like this:		computer. The attack proceeds like this:
	1. I connect to http://anything.example.org/. Note that this site		1. I connect to http://anything.example.org/. Note that this site
	is unaffiliated with the calling service.		is unaffiliated with the calling service.
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	infect the browser's notion of that origin semi-permanently.		infect the browser's notion of that origin semi-permanently.
	[[ OPEN ISSUE: What recommendation should IETF make about (a)		[[ OPEN ISSUE: What recommendation should IETF make about (a)
	whether RTCWeb long-term consent should be available over HTTP pages		whether RTCWeb long-term consent should be available over HTTP pages
	and (b) How to handle origins where the consent is to an HTTPS URL		and (b) How to handle origins where the consent is to an HTTPS URL
	but the page contains active mixed content? ]]		but the page contains active mixed content? ]]
	4.2. Communications Consent Verification		4.2. Communications Consent Verification
	As discussed in Section 3.3, allowing web applications unrestricted		As discussed in Section 3.3, allowing web applications unrestricted
	access to the via the browser network introduces the risk of using		network access via the browser introduces the risk of using the
	the browser as an attack platform against machines which would not		browser as an attack platform against machines which would not
	otherwise be accessible to the malicious site, for instance because		otherwise be accessible to the malicious site, for instance because
	they are topologically restricted (e.g., behind a firewall or NAT).		they are topologically restricted (e.g., behind a firewall or NAT).
	In order to prevent this form of attack as well as cross-protocol		In order to prevent this form of attack as well as cross-protocol
	attacks it is important to require that the target of traffic		attacks it is important to require that the target of traffic
	explicitly consent to receiving the traffic in question. Until that		explicitly consent to receiving the traffic in question. Until that
	consent has been verified for a given endpoint, traffic other than		consent has been verified for a given endpoint, traffic other than
	the consent handshake MUST NOT be sent to that endpoint.		the consent handshake MUST NOT be sent to that endpoint.
	4.2.1. ICE		4.2.1. ICE
	Verifying receiver consent requires some sort of explicit handshake,		Verifying receiver consent requires some sort of explicit handshake,
	but conveniently we already need one in order to do NAT hole-		but conveniently we already need one in order to do NAT hole-
	punching. ICE [RFC5245] includes a handshake designed to verify that		punching. ICE [RFC5245] includes a handshake designed to verify that
	the receiving element wishes to receive traffic from the sender. It		the receiving element wishes to receive traffic from the sender. It
	is important to remember here that the site initiating ICE is		is important to remember here that the site initiating ICE is
	presumed malicious; in order for the handshake to be secure the		presumed malicious; in order for the handshake to be secure the
	receiving element MUST demonstrate receipt/knowledge of some value		receiving element MUST demonstrate receipt/knowledge of some value
	not available to the site (thus preventing it from forging		not available to the site (thus preventing the site from forging
	responses). In order to achieve this objective with ICE, the STUN		responses). In order to achieve this objective with ICE, the STUN
	transaction IDs must be generated by the browser and MUST NOT be made		transaction IDs must be generated by the browser and MUST NOT be made
	available to the initiating script, even via a diagnostic interface.		available to the initiating script, even via a diagnostic interface.
			Verifying receiver consent also requires verifying the receiver wants
			to receive traffic from a particular sender, and at this time; for
			example a malicious site may simply attempt ICE to known servers that
			are using ICE for other sessions. ICE provides this verification as
			well, by using the STUN credentials as a form of per-session shared
			secret. Those credentials are known to the Web application, but
			would need to also be known and used by the STUN-receiving element to
			be useful.
			There also needs to be some mechanism for the browser to verify that
			the target of the traffic continues to wish to receive it.
			Obviously, some ICE-based mechanism will work here, but it has been
			observed that because ICE keepalives are indications, they will not
			work here, so some other mechanism is needed.
	4.2.2. Masking		4.2.2. Masking
	Once consent is verified, there still is some concern about		Once consent is verified, there still is some concern about
	misinterpretation attacks as described by Huang et al.[huang-w2sp].		misinterpretation attacks as described by Huang et al.[huang-w2sp].
	As long as communication is limited to UDP, then this risk is		As long as communication is limited to UDP, then this risk is
	probably limited, thus masking is not required for UDP. I.e., once		probably limited, thus masking is not required for UDP. I.e., once
	communications consent has been verified, it is most likely safe to		communications consent has been verified, it is most likely safe to
	allow the implementation to send arbitrary UDP traffic to the chosen		allow the implementation to send arbitrary UDP traffic to the chosen
	destination, provided that the STUN keepalives continue to succeed.		destination, provided that the STUN keepalives continue to succeed.
	However, with TCP the risk of transparent proxies becomes much more		In particular, this is true for the data channel if DTLS is used
	severe. If TCP is to be used, then WebSockets style masking MUST be		because DTLS (with the anti-chosen plaintext mechanisms required by
	employed.		TLS 1.1) does not allow the attacker to generate predictable
			ciphertext. However, with TCP the risk of transparent proxies
			becomes much more severe. If TCP is to be used, then WebSockets
			style masking MUST be employed.
	4.2.3. Backward Compatibility		4.2.3. Backward Compatibility
	A requirement to use ICE limits compatibility with legacy non-ICE		A requirement to use ICE limits compatibility with legacy non-ICE
	clients. It seems unsafe to completely remove the requirement for		clients. It seems unsafe to completely remove the requirement for
	some check. All proposed checks have the common feature that the		some check. All proposed checks have the common feature that the
	browser sends some message to the candidate traffic recipient and		browser sends some message to the candidate traffic recipient and
	refuses to send other traffic until that message has been replied to.		refuses to send other traffic until that message has been replied to.
	The message/reply pair must be generated in such a way that an		The message/reply pair must be generated in such a way that an
	attacker who controls the Web application cannot forge them,		attacker who controls the Web application cannot forge them,
	skipping to change at page 13, line 51		skipping to change at page 15, line 20
	responds with responses authenticated with ICE credentials) then this		responds with responses authenticated with ICE credentials) then this
	issue does not exist. However, such a design is very close to an		issue does not exist. However, such a design is very close to an
	ICE-Lite implementation (indeed, arguably is one). An intermediate		ICE-Lite implementation (indeed, arguably is one). An intermediate
	approach would be to have a STUN extension that indicated that one		approach would be to have a STUN extension that indicated that one
	was responding to RTC-Web checks but not computing integrity checks		was responding to RTC-Web checks but not computing integrity checks
	based on the ICE credentials. This would allow the use of standalone		based on the ICE credentials. This would allow the use of standalone
	STUN servers without the risk of confusing them with legacy STUN		STUN servers without the risk of confusing them with legacy STUN
	servers. If a non-ICE legacy solution is needed, then this is		servers. If a non-ICE legacy solution is needed, then this is
	probably the best choice.		probably the best choice.
	[TODO: Write something about consent freshness and RTCP].		Once initial consent is verified, we also need to verify continuing
			consent, in order to avoid attacks where two people briefly share an
			IP (e.g., behind a NAT in an Internet cafe) and the attacker arranges
			for a large, unstoppable, traffic flow to the network and then
			leaves. The appropriate technologies here are fairly similar to
			those for initial consent, though are perhaps weaker since the
			threats is less severe.
	[[ OPEN ISSUE: Exactly what should be the requirements here?		[[ OPEN ISSUE: Exactly what should be the requirements here?
	Proposals include ICE all the time or ICE but with allowing one of		Proposals include ICE all the time or ICE but with allowing one of
	these non-ICE things for legacy. ]]		these non-ICE things for legacy. ]]
			4.2.4. IP Location Privacy
			Note that as soon as the callee sends their ICE candidates, the
			callee learns the callee's IP addresses. The callee's server
			reflexive address reveals a lot of information about the callee's
			location. In order to avoid tracking, implementations may wish to
			suppress the start of ICE negotiation until the callee has answered.
			In addition, either side may wish to hide their location entirely by
			forcing all traffic through a TURN server.
	4.3. Communications Security		4.3. Communications Security
	Finally, we consider a problem familiar from the SIP world:		Finally, we consider a problem familiar from the SIP world:
	communications security. For obvious reasons, it MUST be possible		communications security. For obvious reasons, it MUST be possible
	for the communicating parties to establish a channel which is secure		for the communicating parties to establish a channel which is secure
	against both message recovery and message modification. (See		against both message recovery and message modification. (See
	[RFC5479] for more details.) This service must be provided for both		[RFC5479] for more details.) This service must be provided for both
	data and voice/video. Ideally the same security mechanisms would be		data and voice/video. Ideally the same security mechanisms would be
	used for both types of content. Technology for providing this		used for both types of content. Technology for providing this
	service (for instance, DTLS [RFC4347] and DTLS-SRTP [RFC5763]) is		service (for instance, DTLS [RFC4347] and DTLS-SRTP [RFC5763]) is
	skipping to change at page 18, line 7		skipping to change at page 19, line 39
	should be specified and/or mandatory to implement? In particular,		should be specified and/or mandatory to implement? In particular,
	should we allow DTLS-SRTP only, or both DTLS-SRTP and SDES. Should		should we allow DTLS-SRTP only, or both DTLS-SRTP and SDES. Should
	we allow RTP for backward compatibility? ]]		we allow RTP for backward compatibility? ]]
	5. Security Considerations		5. Security Considerations
	This entire document is about security.		This entire document is about security.
	6. Acknowledgements		6. Acknowledgements
			Bernard Aboba, Harald Alvestrand, Cullen Jennings, Hadriel Kaplan (S
			4.2.1), Matthew Kaufman, Magnus Westerland.
	7. References		7. References
	7.1. Normative References		7.1. 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, March 1997.		Requirement Levels", BCP 14, RFC 2119, March 1997.
	7.2. Informative References		7.2. Informative References
	[CORS] van Kesteren, A., "Cross-Origin Resource Sharing".		[CORS] van Kesteren, A., "Cross-Origin Resource Sharing".
	[I-D.abarth-origin]		[I-D.abarth-origin]
	Barth, A., "The Web Origin Concept",		Barth, A., "The Web Origin Concept",
	draft-abarth-origin-09 (work in progress), November 2010.		draft-abarth-origin-09 (work in progress), November 2010.
	[I-D.ietf-hybi-thewebsocketprotocol]		[I-D.ietf-hybi-thewebsocketprotocol]
	Fette, I. and A. Melnikov, "The WebSocket protocol",		Fette, I. and A. Melnikov, "The WebSocket protocol",
	draft-ietf-hybi-thewebsocketprotocol-15 (work in		draft-ietf-hybi-thewebsocketprotocol-17 (work in
	progress), September 2011.		progress), September 2011.
	[I-D.kaufman-rtcweb-security-ui]		[I-D.kaufman-rtcweb-security-ui]
	Kaufman, M., "Client Security User Interface Requirements		Kaufman, M., "Client Security User Interface Requirements
	for RTCWEB", draft-kaufman-rtcweb-security-ui-00 (work in		for RTCWEB", draft-kaufman-rtcweb-security-ui-00 (work in
	progress), June 2011.		progress), June 2011.
	[RFC2818] Rescorla, E., "HTTP Over TLS", RFC 2818, May 2000.		[RFC2818] Rescorla, E., "HTTP Over TLS", RFC 2818, May 2000.
	[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,		[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
	skipping to change at page 20, line 12		skipping to change at page 22, line 5
	Kain, A. and M. Macon, "Design and Evaluation of a Voice		Kain, A. and M. Macon, "Design and Evaluation of a Voice
	Conversion Algorithm based on Spectral Envelope Mapping		Conversion Algorithm based on Spectral Envelope Mapping
	and Residual Prediction", Proceedings of ICASSP, May		and Residual Prediction", Proceedings of ICASSP, May
	2001.		2001.
	[whitten-johnny]		[whitten-johnny]
	Whitten, A. and J. Tygar, "Why Johnny Can't Encrypt: A		Whitten, A. and J. Tygar, "Why Johnny Can't Encrypt: A
	Usability Evaluation of PGP 5.0", Proceedings of the 8th		Usability Evaluation of PGP 5.0", Proceedings of the 8th
	USENIX Security Symposium, 1999.		USENIX Security Symposium, 1999.
			Appendix A. A Proposed Security Architecture [No Consensus on This]
			This section contains a proposed security architecture, based on the
			considerations discussed in the main body of this memo. This section
			is currently the opinion of the author and does not have consensus
			though some (many?) elements of this proposal do seem to have general
			consensus.
			A.1. Trust Hierarchy
			The basic assumption of this proposal is that network resources exist
			in a hierarchy of trust, rooted in the browser, which serves as the
			user's TRUSTED COMPUTING BASE (TCB). Any security property which the
			user wishes to have enforced must be ultimately guaranteed by the
			browser (or transitively by some property the browser verifies).
			Conversely, if the browser is compromised, then no security
			guarantees are possible. Note that there are cases (e.g., Internet
			kiosks) where the user can't really trust the browser that much. In
			these cases, the level of security provided is limited by how much
			they trust the browser.
			Optimally, we would not rely on trust in any entities other than the
			browser. However, this is unfortunately not possible if we wish to
			have a functional system. Other network elements fall into two
			categories: those which can be authenticated by the browser and thus
			are partly trusted--though to the minimum extent necessary--and those
			which cannot be authenticated and thus are untrusted. This is a
			natural extension of the end-to-end principle.
			A.1.1. Authenticated Entities
			There are two major classes of authenticated entities in the system:
			o Calling services: Web sites whose origin we can verify (optimally
			via HTTPS).
			o Other users: RTC-Web peers whose origin we can verify
			cryptographically (optimally via DTLS-SRTP).
			Note that merely being authenticated does not make these entities
			trusted. For instance, just because we can verify that
			https://www.evil.org/ is owned by Dr. Evil does not mean that we can
			trust Dr. Evil to access our camera an microphone. However, it gives
			the user an opportunity to determine whether he wishes to trust Dr.
			Evil or not; after all, if he desires to contact Dr. Evil, it's safe
			to temporarily give him access to the camera and microphone for the
			purpose of the call. The point here is that we must first identify
			other elements before we can determine whether to trust them.
			It's also worth noting that there are settings where authentication
			is non-cryptographic, such as other machines behind a firewall.
			Naturally, the level of trust one can have in identities verified in
			this way depends on how strong the topology enforcement is.
			A.1.2. Unauthenticated Entities
			Other than the above entities, we are not generally able to identify
			other network elements, thus we cannot trust them. This does not
			mean that it is not possible to have any interaction with them, but
			it means that we must assume that they will behave maliciously and
			design a system which is secure even if they do so.
			A.2. Overview
			This section describes a typical RTCWeb session and shows how the
			various security elements interact and what guarantees are provided
			to the user. The example in this section is a "best case" scenario
			in which we provide the maximal amount of user authentication and
			media privacy with the minimal level of trust in the calling service.
			Simpler versions with lower levels of security are also possible and
			are noted in the text where applicable. It's also important to
			recognize the tension between security (or performance) and privacy.
			The example shown here is aimed towards settings where we are more
			concerned about secure calling than about privacy, but as we shall
			see, there are settings where one might wish to make different
			tradeoffs--this architecture is still compatible with those settings.
			For the purposes of this example, we assume the topology shown in the
			figure below. This topology is derived from the topology shown in
			Figure 1, but separates Alice and Bob's identities from the process
			of signaling. Specifically, Alice and Bob have relationships with
			some Identity Provider (IDP) that supports a protocol such OpenID or
			BrowserID) that can be used to attest to their identity. This
			separation isn't particularly important in "closed world" cases where
			Alice and Bob are users on the same social network and have
			identities based on that network. However, there are important
			settings where that is not the case, such as federation (calls from
			one network to another) and calling on untrusted sites, such as where
			two users who have a relationship via a given social network want to
			call each other on another, untrusted, site, such as a poker site.
			+----------------+
			| |
			| Signaling |
			| Server |
			| |
			+----------------+
			^ ^
			/ \
			HTTPS / \ HTTPS
			/ \
			/ \
			v v
			JS API JS API
			+-----------+ +-----------+
			| | Media | |
			Alice | Browser |<---------->| Browser | Bob
			| | (DTLS-SRTP)| |
			+-----------+ +-----------+
			^ ^--+ +--^ ^
			| | | |
			v | | v
			+-----------+ | | +-----------+
			| |<--------+ | |
			| IDP | | | IDP |
			| | +------->| |
			+-----------+ +-----------+
			Figure 2: A call with IDP-based identity
			A.2.1. Initial Signaling
			Alice and Bob are both users of a common calling service; they both
			have approved the calling service to make calls (we defer the
			discussion of device access permissions till later). They are both
			connected to the calling service via HTTPS and so know the origin
			with some level of confidence. They also have accounts with some
			identity provider. This sort of identity service is becoming
			increasingly common in the Web environment in technologies such
			(BrowserID, Federated Google Login, Facebook Connect, OAuth, OpenID,
			WebFinger), and is often provided as a side effect service of your
			ordinary accounts with some service. In this example, we show Alice
			and Bob using a separate identity service, though they may actually
			be using the same identity service as calling service or have no
			identity service at all.
			Alice is logged onto the calling service and decides to call Bob. She
			can see from the calling service that he is online and the calling
			service presents a JS UI in the form of a button next to Bob's name
			which says "Call". Alice clicks the button, which initiates a JS
			callback that instantiates a PeerConnection object. This does not
			require a security check: JS from any origin is allowed to get this
			far.
			Once the PeerConnection is created, the calling service JS needs to
			set up some media. Because this is an audio/video call, it creates
			two MediaStreams, one connected to an audio input and one connected
			to a video input. At this point the first security check is
			required: untrusted origins are not allowed to access the camera and
			microphone. In this case, because Alice is a long-term user of the
			calling service, she has made a permissions grant (i.e., a setting in
			the browser) to allow the calling service to access her camera and
			microphone any time it wants. The browser checks this setting when
			the camera and microphone requests are made and thus allows them.
			In the current W3C API, once some streams have been added, Alice's
			browser + JS generates a signaling message The format of this data is
			currently undefined. It may be a complete message as defined by ROAP
			[REF] or may be assembled piecemeal by the JS. In either case, it
			will contain:
			o Media channel information
			o ICE candidates
			o A fingerprint attribute binding the message to Alice's public key
			[RFC5763]
			Prior to sending out the signaling message, the PeerConnection code
			contacts the identity service and obtains an assertion binding
			Alice's identity to her fingerprint. The exact details depend on the
			identity service (though as discussed in Appendix A.3.6.4 I believe
			PeerConnection can be agnostic to them), but for now it's easiest to
			think of as a BrowserID assertion.
			This message is sent to the signaling server, e.g., by XMLHttpRequest
			[REF] or by WebSockets [I-D.ietf-hybi-thewebsocketprotocol]. The
			signaling server processes the message from Alice's browser,
			determines that this is a call to Bob and sends a signaling message
			to Bob's browser (again, the format is currently undefined). The JS
			on Bob's browser processes it, and alerts Bob to the incoming call
			and to Alice's identity. In this case, Alice has provided an
			identity assertion and so Bob's browser contacts Alice's identity
			provider (again, this is done in a generic way so the browser has no
			specific knowledge of the IDP) to verity the assertion. This allows
			the browser to display a trusted element indicating that a call is
			coming in from Alice. If Alice is in Bob's address book, then this
			interface might also include her real name, a picture, etc. The
			calling site will also provide some user interface element (e.g., a
			button) to allow Bob to answer the call, though this is most likely
			not part of the trusted UI.
			If Bob agrees [I am ignoring early media for now], a PeerConnection
			is instantiated with the message from Alice's side. Then, a similar
			process occurs as on Alice's browser: Bob's browser verifies that
			the calling service is approved, the media streams are created, and a
			return signaling message containing media information, ICE
			candidates, and a fingerprint is sent back to Alice via the signaling
			service. If Bob has a relationship with an IDP, the message will
			also come with an identity assertion.
			At this point, Alice and Bob each know that the other party wants to
			have a secure call with them. Based purely on the interface provided
			by the signaling server, they know that the signaling server claims
			that the call is from Alice to Bob. Because the far end sent an
			identity assertion along with their message, they know that this is
			verifiable from the IDP as well. Of course, the call works perfectly
			well if either Alice or Bob doesn't have a relationship with an IDP;
			they just get a lower level of assurance. Moreover, Alice might wish
			to make an anonymous call through an anonymous calling site, in which
			case she would of course just not provide any identity assertion and
			the calling site would mask her identity from Bob.
			A.2.2. Media Consent Verification
			As described in Section 4.2. This proposal specifies that that be
			performed via ICE. Thus, Alice and Bob perform ICE checks with each
			other. At the completion of these checks, they are ready to send
			non-ICE data.
			At this point, Alice knows that (a) Bob (assuming he is verified via
			his IDP) or someone else who the signaling service is claiming is Bob
			is willing to exchange traffic with her and (b) that either Bob is at
			the IP address which she has verified via ICE or there is an attacker
			who is on-path to that IP address detouring the traffic. Note that
			it is not possible for an attacker who is on-path but not attached to
			the signaling service to spoof these checks because they do not have
			the ICE credentials. Bob's security guarantees with respect to Alice
			are the converse of this.
			A.2.3. DTLS Handshake
			Once the ICE checks have completed [more specifically, once some ICE
			checks have completed], Alice and Bob can set up a secure channel.
			This is performed via DTLS [RFC4347] (for the data channel) and DTLS-
			SRTP [RFC5763] for the media channel. Specifically, Alice and Bob
			perform a DTLS handshake on every channel which has been established
			by ICE. The total number of channels depends on the amount of
			muxing; in the most likely case we are using both RTP/RTCP mux and
			muxing multiple media streams on the same channel, in which case
			there is only one DTLS handshake. Once the DTLS handshake has
			completed, the keys are extracted and used to key SRTP for the media
			channels.
			At this point, Alice and Bob know that they share a set of secure
			data and/or media channels with keys which are not known to any
			third-party attacker. If Alice and Bob authenticated via their IDPs,
			then they also know that the signaling service is not attacking them.
			Even if they do not use an IDP, as long as they have minimal trust in
			the signaling service not to perform a man-in-the-middle attack, they
			know that their communications are secure against the signaling
			service as well.
			A.2.4. Communications and Consent Freshness
			From a security perspective, everything from here on in is a little
			anticlimactic: Alice and Bob exchange data protected by the keys
			negotiated by DTLS. Because of the security guarantees discussed in
			the previous sections, they know that the communications are
			encrypted and authenticated.
			The one remaining security property we need to establish is "consent
			freshness", i.e., allowing Alice to verify that Bob is still prepared
			to receive her communications. ICE specifies periodic STUN
			keepalizes but only if media is not flowing. Because the consent
			issue is more difficult here, we require RTCWeb implementations to
			periodically send keepalives. If a keepalive fails and no new ICE
			channels can be established, then the session is terminated.
			A.3. Detailed Technical Description
			A.3.1. Origin and Web Security Issues
			The basic unit of permissions for RTC-Web is the origin
			[I-D.abarth-origin]. Because the security of the origin depends on
			being able to authenticate content from that origin, the origin can
			only be securely established if data is transferred over HTTPS.
			Thus, clients MUST treat HTTP and HTTPS origins as different
			permissions domains and SHOULD NOT permit access to any RTC-Web
			functionality from scripts fetched over non-secure (HTTP) origins.
			If an HTTPS origin contains mixed active content (regardless of
			whether it is present on the specific page attempting to access RTC-
			Web functionality), any access MUST be treated as if it came from the
			HTTP origin. For instance, if a https://www.example.com/example.html
			loads https://www.example.com/example.js and
			http://www.example.org/jquery.js, any attempt by example.js to access
			RTCWeb functionality MUST be treated as if it came from
			http://www.example.com/. Note that many browsers already track mixed
			content and either forbid it by default or display a warning.
			A.3.2. Device Permissions Model
			Implementations MUST obtain explicit user consent prior to providing
			access to the camera and/or microphone. Implementations MUST at
			minimum support the following two permissions models:
			o Requests for one-time camera/microphone access.
			o Requests for permanent access.
			In addition, they SHOULD support requests for access to a single
			communicating peer. E.g., "Call customerservice@ford.com". Browsers
			servicing such requests SHOULD clearly indicate that identity to the
			user when asking for permission.
			API Requirement: The API MUST provide a mechanism for the requesting
			JS to indicate which of these forms of permissions it is
			requesting. This allows the client to know what sort of user
			interface experience to provide. In particular, browsers might
			display a non-invasive door hanger ("some features of this site
			may not work..." when asking for long-term permissions) but a more
			invasive UI ("here is your own video") for single-call
			permissions. The API MAY grant weaker permissions than the JS
			asked for if the user chooses to authorize only those permissions,
			but if it intends to grant stronger ones SHOULD display the
			appropriate UI for those permissions.
			API Requirement: The API MUST provide a mechanism for the requesting
			JS to relinquish the ability to see or modify the media (e.g., via
			MediaStream.record()). Combined with secure authentication of the
			communicating peer, this allows a user to be sure that the calling
			site is not accessing or modifying their conversion.
			UI Requirement: The UI MUST clearly indicate when the user's camera
			and microphone are in use. This indication MUST NOT be
			suppressable by the JS and MUST clearly indicate how to terminate
			a call, and provide a UI means to immediately stop camera/
			microphone input without the JS being able to prevent it.
			UI Requirement: If the UI indication of camera/microphone use are
			displayed in the browser such that minimizing the browser window
			would hide the indication, or the JS creating an overlapping
			window would hide the indication, then the browser SHOULD stop
			camera and microphone input.
			Clients MAY permit the formation of data channels without any direct
			user approval. Because sites can always tunnel data through the
			server, further restrictions on the data channel do not provide any
			additional security. (though see Appendix A.3.3 for a related issue).
			Implementations which support some form of direct user authentication
			SHOULD also provide a policy by which a user can authorize calls only
			to specific counterparties. Specifically, the implementation SHOULD
			provide the following interfaces/controls:
			o Allow future calls to this verified user.
			o Allow future calls to any verified user who is in my system
			address book (this only works with address book integration, of
			course).
			Implementations SHOULD also provide a different user interface
			indication when calls are in progress to users whose identities are
			directly verifiable. Appendix A.3.5 provides more on this.
			A.3.3. Communications Consent
			Browser client implementations of RTC-Web MUST implement ICE. Server
			gateway implementations which operate only at public IP addresses may
			implement ICE-Lite.
			Browser implementations MUST verify reachability via ICE prior to
			sending any non-ICE packets to a given destination. Implementations
			MUST NOT provide the ICE transaction ID to JavaScript. [Note: this
			document takes no position on the split between ICE in JS and ICE in
			the browser. The above text is written the way it is for editorial
			convenience and will be modified appropriately if the WG decides on
			ICE in the JS.]
			Implementations MUST send keepalives no less frequently than every 30
			seconds regardless of whether traffic is flowing or not. If a
			keepalive fails then the implementation MUST either attempt to find a
			new valid path via ICE or terminate media for that ICE component.
			Note that ICE [RFC5245]; Section 10 keepalives use STUN Binding
			Indications which are one-way and therefore not sufficient. We will
			need to define a new mechanism for this. [OPEN ISSUE: what to do
			here.]
			A.3.4. IP Location Privacy
			As mentioned in Section 4.2.4 above, a side effect of the default ICE
			behavior is that the peer learns one's IP address, which leaks large
			amounts of location information, especially for mobile devices. This
			has negative privacy consequences in some circumstances. The
			following two API requirements are intended to mitigate this issue:
			API Requirement: The API MUST provide a mechanism to suppress ICE
			negotiation (though perhaps to allow candidate gathering) until
			the user has decided to answer the call [note: determining when
			the call has been answered is a question for the JS.] This
			enables a user to prevent a peer from learning their IP address if
			they elect not to answer a call.
			API Requirement: The API MUST provide a mechanism for the calling
			application to indicate that only TURN candidates are to be used.
			This prevents the peer from learning one's IP address at all.
			A.3.5. Communications Security
			Implementations MUST implement DTLS and DTLS-SRTP. All data channels
			MUST be secured via DTLS. DTLS-SRTP MUST be offered for every media
			channel and MUST be the default; i.e., if an implementation receives
			an offer for DTLS-SRTP and SDES and/or plain RTP, DTLS-SRTP MUST be
			selected.
			[OPEN ISSUE: What should the settings be here? MUST?]
			Implementations MAY support SDES and RTP for media traffic for
			backward compatibility purposes.
			API Requirement: The API MUST provide a mechanism to indicate that a
			fresh DTLS key pair is to be generated for a specific call. This
			is intended to allow for unlinkability. Note that there are also
			settings where it is attractive to use the same keying material
			repeatedly, especially those with key continuity-based
			authentication.
			API Requirement: The API MUST provide a mechanism to indicate that a
			fresh DTLS key pair is to be generated for a specific call. This
			is intended to allow for unlinkability.
			API Requirement: When DTLS-SRTP is used, the API MUST NOT permit the
			JS to obtain the negotiated keying material. This requirement
			preserves the end-to-end security of the media.
			UI Requirements: A user-oriented client MUST provide an
			"inspector" interface which allows the user to determine the
			security characteristics of the media. [largely derived from
			[I-D.kaufman-rtcweb-security-ui]
			The following properties SHOULD be displayed "up-front" in the
			browser chrome, i.e., without requiring the user to ask for them:
			* A client MUST provide a user interface through which a user may
			determine the security characteristics for currently-displayed
			audio and video stream(s)
			* A client MUST provide a user interface through which a user may
			determine the security characteristics for transmissions of
			their microphone audio and camera video.
			* The "security characteristics" MUST include an indication as to
			whether or not the transmission is cryptographically protected
			and whether that protection is based on a key that was
			delivered out-of-band (from a server) or was generated as a
			result of a pairwise negotiation.
			* If the far endpoint was directly verified Appendix A.3.6 the
			"security characteristics" MUST include the verified
			information.
			The following properties are more likely to require some "drill-
			down" from the user:
			* If the transmission is cryptographically protected, the The
			algorithms in use (For example: "AES-CBC" or "Null Cipher".)
			* If the transmission is cryptographically protected, the
			"security characteristics" MUST indicate whether PFS is
			provided.
			* If the transmission is cryptographically protected via an end-
			to-end mechanism the "security characteristics" MUST include
			some mechanism to allow an out-of-band verification of the
			peer, such as a certificate fingerprint or an SAS.
			A.3.6. Web-Based Peer Authentication
			A.3.6.1. Generic Concepts
			In a number of cases, it is desirable for the endpoint (i.e., the
			browser) to be able to directly identity the endpoint on the other
			side without trusting only the signaling service to which they are
			connected. For instance, users may be making a call via a federated
			system where they wish to get direct authentication of the other
			side. Alternately, they may be making a call on a site which they
			minimally trust (such as a poker site) but to someone who has an
			identity on a site they do trust (such as a social network.)
			Recently, a number of Web-based identity technologies (OAuth,
			BrowserID, Facebook Connect), etc. have been developed. While the
			details vary, what these technologies share is that they have a Web-
			based (i.e., HTTP/HTTPS identity provider) which attests to your
			identity. For instance, if I have an account at example.org, I could
			use the example.org identity provider to prove to others that I was
			alice@example.org. The development of these technologies allows us
			to separate calling from identity provision: I could call you on
			Poker Galaxy but identify myself as alice@example.org.
			Whatever the underlying technology, the general principle is that the
			party which is being authenticated is NOT the signaling site but
			rather the user (and their browser). Similarly, the relying party is
			the browser and not the signaling site. This means that the
			PeerConnection API MUST arrange to talk directly to the identity
			provider in a way that cannot be impersonated by the calling site.
			The following sections provide two examples of this.
			A.3.6.2. BrowserID
			BrowserID [https://browserid.org/] is a technology which allows a
			user with a verified email address to generate an assertion
			(authenticated by their identity provider) attesting to their
			identity (phrased as an email address). The way that this is used in
			practice is that the relying party embeds JS in their site which
			talks to the BrowserID code (either hosted on a trusted intermediary
			or embedded in the browser). That code generates the assertion which
			is passed back to the relying party for verification. The assertion
			can be verified directly or with a Web service provided by the
			identity provider. It's relatively easy to extend this functionality
			to authenticate RTC-Web calls, as shown below.
			+----------------------+ +----------------------+
			| | | |
			| Alice's Browser | | Bob's Browser |
			| | OFFER ------------> | |
			| Calling JS Code | | Calling JS Code |
			| ^ | | ^ |
			| | | | | |
			| v | | v |
			| PeerConnection | | PeerConnection |
			| | ^ | | | ^ |
			| Finger| |Signed | |Signed | | |
			| print | |Finger | |Finger | |"Alice"|
			| | |print | |print | | |
			| v | | | v | |
			| +--------------+ | | +---------------+ |
			| | BrowserID | | | | BrowserID | |
			| | Signer | | | | Verifier | |
			| +--------------+ | | +---------------+ |
			| ^ | | ^ |
			+-----------|----------+ +----------|-----------+
			| |
			| Get certificate |
			v | Check
			+----------------------+ | certificate
			| | |
			| Identity |/-------------------------------+
			| Provider |
			| |
			+----------------------+
			The way this mechanism works is as follows. On Alice's side, Alice
			goes to initiate a call.
			1. The calling JS instantiates a PeerConnection and tells it that it
			is interested in having it authenticated via BrowserID.
			2. The PeerConnection instantiates the BrowserID signer in an
			invisible IFRAME. The IFRAME is tagged with an origin that
			indicates that it was generated by the PeerConnection (this
			prevents ordinary JS from implementing it). The BrowserID signer
			is provided with Alice's fingerprint. Note that the IFRAME here
			does not render any UI. It is being used solely to allow the
			browser to load the BrowserID signer in isolation, especially
			from the calling site.
			3. The BrowserID signer contacts Alice's identity provider,
			authenticating as Alice (likely via a cookie).
			4. The identity provider returns a short-term certificate attesting
			to Alice's identity and her short-term public key.
			5. The Browser-ID code signs the fingerprint and returns the signed
			assertion + certificate to the PeerConnection. [Note: there are
			well-understood Web mechanisms for this that I am excluding here
			for simplicity.]
			6. The PeerConnection returns the signed information to the calling
			JS code.
			7. The signed assertion gets sent over the wire to Bob's browser
			(via the signaling service) as part of the call setup.
			Obviously, the format of the signed assertion varies depending on
			what signaling style the WG ultimately adopts. However, for
			concreteness, if something like ROAP were adopted, then the entire
			message might look like:
			{
			"messageType":"OFFER",
			"callerSessionId":"13456789ABCDEF",
			"seq": 1
			"sdp":"
			v=0\n
			o=- 2890844526 2890842807 IN IP4 192.0.2.1\n
			s= \n
			c=IN IP4 192.0.2.1\n
			t=2873397496 2873404696\n
			m=audio 49170 RTP/AVP 0\n
			a=fingerprint: SHA-1 \
			4A:AD:B9:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB\n",
			"identity":{
			"identityType":"browserid",
			"assertion": {
			"digest":"<hash of fingerprint and session IDs>",
			"audience": "[TBD]"
			"valid-until": 1308859352261,
			}, // signed using user's key
			"certificate": {
			"email": "rescorla@gmail.com",
			"public-key": "<ekrs-public-key>",
			"valid-until": 1308860561861,
			} // certificate is signed by gmail.com
			}
			}
			Note that we only expect to sign the fingerprint values and the
			session IDs, in order to allow the JS or calling service to modify
			the rest of the SDP, while protecting the identity binding. [OPEN
			ISSUE: should we sign seq too?]
			[TODO: NEed to talk about Audience a bit.]
			On Bob's side, he receives the signed assertion as part of the call
			setup message and a similar procedure happens to verify it.
			1. The calling JS instantiates a PeerConnection and provides it the
			relevant signaling information, including the signed assertion.
			2. The PeerConnection instantiates a BrowserID verifier in an IFRAME
			and provides it the signed assertion.
			3. The BrowserID verifier contacts the identity provider to verify
			the certificate and then uses the key to verify the signed
			fingerprint.
			4. Alice's verified identity is returned to the PeerConnection (it
			already has the fingerprint).
			5. At this point, Bob's browser can display a trusted UI indication
			that Alice is on the other end of the call.
			When Bob returns his answer, he follows the converse procedure, which
			provides Alice with a signed assertion of Bob's identity and keying
			material.
			A.3.6.3. OAuth
			While OAuth is not directly designed for user-to-user authentication,
			with a little lateral thinking it can be made to serve. We use the
			following mapping of OAuth concepts to RTC-Web concepts:
			+----------------------+----------------------+
			| OAuth | RTCWeb |
			+----------------------+----------------------+
			| Client | Relying party |
			| Resource owner | Authenticating party |
			| Authorization server | Identity service |
			| Resource server | Identity service |
			+----------------------+----------------------+
			Table 1
			The idea here is that when Alice wants to authenticate to Bob (i.e.,
			for Bob to be aware that she is calling). In order to do this, she
			allows Bob to see a resource on the identity provider that is bound
			to the call, her identity, and her public key. Then Bob retrieves
			the resource from the identity provider, thus verifying the binding
			between Alice and the call.
			Alice IDP Bob
			---------------------------------------------------------
			Call-Id, Fingerprint ------->
			<------------------- Auth Code
			Auth Code ---------------------------------------------->
			<----- Get Token + Auth Code
			Token --------------------->
			<------------- Get call-info
			Call-Id, Fingerprint ------>
			This is a modified version of a common OAuth flow, but omits the
			redirects required to have the client point the resource owner to the
			IDP, which is acting as both the resource server and the
			authorization server, since Alice already has a handle to the IDP.
			Above, we have referred to "Alice", but really what we mean is the
			PeerConnection. Specifically, the PeerConnection will instantiate an
			IFRAME with JS from the IDP and will use that IFRAME to communicate
			with the IDP, authenticating with Alice's identity (e.g., cookie).
			Similarly, Bob's PeerConnection instantiates an IFRAME to talk to the
			IDP.
			A.3.6.4. Generic Identity Support
			I believe it's possible to build a generic interface between the
			PeerConnection and any identity sub-module so that the PeerConnection
			just gets pointed to the IDP (which the relying party either trusts
			or not) and JS from the IDP provides the concrete interfaces.
			However, I need to work out the details, so I'm not specifying this
			yet. If it works, the previous two sections will just be examples.
	Author's Address		Author's Address
	Eric Rescorla		Eric Rescorla
	RTFM, Inc.		RTFM, Inc.
	2064 Edgewood Drive		2064 Edgewood Drive
	Palo Alto, CA 94303		Palo Alto, CA 94303
	USA		USA
	Phone: +1 650 678 2350		Phone: +1 650 678 2350
	Email: ekr@rtfm.com		Email: ekr@rtfm.com
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