draft-ietf-rtcweb-security-01.txt		draft-ietf-rtcweb-security-02.txt
	RTC-Web E. Rescorla		RTC-Web E. Rescorla
	Internet-Draft RTFM, Inc.		Internet-Draft RTFM, Inc.
	Intended status: Standards Track October 30, 2011		Intended status: Standards Track March 12, 2012
	Expires: May 2, 2012		Expires: September 13, 2012
	Security Considerations for RTC-Web		Security Considerations for RTC-Web
	draft-ietf-rtcweb-security-01		draft-ietf-rtcweb-security-02
	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 May 2, 2012.		This Internet-Draft will expire on September 13, 2012.
	Copyright Notice		Copyright Notice
	Copyright (c) 2011 IETF Trust and the persons identified as the		Copyright (c) 2012 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
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	carefully, as they describe your rights and restrictions with respect		carefully, as they describe your rights and restrictions with respect
	to this document. Code Components extracted from this document must		to this document. Code Components extracted from this document must
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	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 . . . . . . . . . . . . . . . . . . . . . . . . . 5		1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4
	2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 6		2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 5
	3. The Browser Threat Model . . . . . . . . . . . . . . . . . . . 6		3. The Browser Threat Model . . . . . . . . . . . . . . . . . . . 5
	3.1. Access to Local Resources . . . . . . . . . . . . . . . . 7		3.1. Access to Local Resources . . . . . . . . . . . . . . . . 6
	3.2. Same Origin Policy . . . . . . . . . . . . . . . . . . . . 7		3.2. Same Origin Policy . . . . . . . . . . . . . . . . . . . . 6
	3.3. Bypassing SOP: CORS, WebSockets, and consent to		3.3. Bypassing SOP: CORS, WebSockets, and consent to
	communicate . . . . . . . . . . . . . . . . . . . . . . . 8		communicate . . . . . . . . . . . . . . . . . . . . . . . 7
	4. Security for RTC-Web Applications . . . . . . . . . . . . . . 8		4. Security for RTC-Web Applications . . . . . . . . . . . . . . 7
	4.1. Access to Local Devices . . . . . . . . . . . . . . . . . 8		4.1. Access to Local Devices . . . . . . . . . . . . . . . . . 8
	4.1.1. Calling Scenarios and User Expectations . . . . . . . 9		4.1.1. Calling Scenarios and User Expectations . . . . . . . 8
	4.1.1.1. Dedicated Calling Services . . . . . . . . . . . . 9		4.1.1.1. Dedicated Calling Services . . . . . . . . . . . . 8
	4.1.1.2. Calling the Site You're On . . . . . . . . . . . . 9		4.1.1.2. Calling the Site You're On . . . . . . . . . . . . 9
	4.1.1.3. Calling to an Ad Target . . . . . . . . . . . . . 10		4.1.1.3. Calling to an Ad Target . . . . . . . . . . . . . 9
	4.1.2. Origin-Based Security . . . . . . . . . . . . . . . . 10		4.1.2. Origin-Based Security . . . . . . . . . . . . . . . . 10
	4.1.3. Security Properties of the Calling Page . . . . . . . 12		4.1.3. Security Properties of the Calling Page . . . . . . . 11
	4.2. Communications Consent Verification . . . . . . . . . . . 13		4.2. Communications Consent Verification . . . . . . . . . . . 12
	4.2.1. ICE . . . . . . . . . . . . . . . . . . . . . . . . . 13		4.2.1. ICE . . . . . . . . . . . . . . . . . . . . . . . . . 12
	4.2.2. Masking . . . . . . . . . . . . . . . . . . . . . . . 14		4.2.2. Masking . . . . . . . . . . . . . . . . . . . . . . . 13
	4.2.3. Backward Compatibility . . . . . . . . . . . . . . . . 14		4.2.3. Backward Compatibility . . . . . . . . . . . . . . . . 13
	4.2.4. IP Location Privacy . . . . . . . . . . . . . . . . . 15		4.2.4. IP Location Privacy . . . . . . . . . . . . . . . . . 14
	4.3. Communications Security . . . . . . . . . . . . . . . . . 15		4.3. Communications Security . . . . . . . . . . . . . . . . . 14
	4.3.1. Protecting Against Retrospective Compromise . . . . . 16		4.3.1. Protecting Against Retrospective Compromise . . . . . 15
	4.3.2. Protecting Against During-Call Attack . . . . . . . . 17		4.3.2. Protecting Against During-Call Attack . . . . . . . . 16
	4.3.2.1. Key Continuity . . . . . . . . . . . . . . . . . . 17		4.3.2.1. Key Continuity . . . . . . . . . . . . . . . . . . 16
	4.3.2.2. Short Authentication Strings . . . . . . . . . . . 18		4.3.2.2. Short Authentication Strings . . . . . . . . . . . 17
	4.3.2.3. Recommendations . . . . . . . . . . . . . . . . . 19		4.3.2.3. Third Party Identity . . . . . . . . . . . . . . . 18
	5. Security Considerations . . . . . . . . . . . . . . . . . . . 19		5. Security Considerations . . . . . . . . . . . . . . . . . . . 18
	6. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 19		6. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 18
	7. References . . . . . . . . . . . . . . . . . . . . . . . . . . 19		7. References . . . . . . . . . . . . . . . . . . . . . . . . . . 19
	7.1. Normative References . . . . . . . . . . . . . . . . . . . 19		7.1. Normative References . . . . . . . . . . . . . . . . . . . 19
	7.2. Informative References . . . . . . . . . . . . . . . . . . 20		7.2. Informative References . . . . . . . . . . . . . . . . . . 19
	Appendix A. A Proposed Security Architecture [No Consensus on		Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 21
	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 11		skipping to change at page 5, line 11
	particular, it needs to contend with malicious calling services. For		particular, it needs to contend with malicious calling services. For
	example, if the calling service can cause the browser to make a call		example, if the calling service can cause the browser to make a call
	at any time to any callee of its choice, then this facility can be		at any time to any callee of its choice, then this facility can be
	used to bug a user's computer without their knowledge, simply by		used to bug a user's computer without their knowledge, simply by
	placing a call to some recording service. More subtly, if the		placing a call to some recording service. More subtly, if the
	exposed APIs allow the server to instruct the browser to send		exposed APIs allow the server to instruct the browser to send
	arbitrary content, then they can be used to bypass firewalls or mount		arbitrary content, then they can be used to bypass firewalls or mount
	denial of service attacks. Any successful system will need to be		denial of service attacks. Any successful system will need to be
	resistant to this and other attacks.		resistant to this and other attacks.
			A companion document [I-D.ietf-rtcweb-security-arch] describes a
			security architecture intended to address the issues raised in this
			document.
	2. Terminology		2. Terminology
	The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",		The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
	"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this		"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
	document are to be interpreted as described in RFC 2119 [RFC2119].		document are to be interpreted as described in RFC 2119 [RFC2119].
	3. The Browser Threat Model		3. The Browser Threat Model
	The security requirements for RTC-Web follow directly from the		The security requirements for RTC-Web follow directly from the
	requirement that the browser's job is to protect the user. Huang et		requirement that the browser's job is to protect the user. Huang et
	skipping to change at page 7, line 38		skipping to change at page 6, line 43
	at all. For instance, there is no real way to run specific		at all. For instance, there is no real way to run specific
	executables directly from a script (though the user can of course be		executables directly from a script (though the user can of course be
	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.[RFC6454] (although CORS [CORS] and WebSockets [RFC6455]
	[I-D.ietf-hybi-thewebsocketprotocol] provides a escape hatch from		provides a escape hatch from this restriction, as described below.)
	this restriction, as described below.) This SAME ORIGIN POLICY (SOP)		This SAME ORIGIN POLICY (SOP) prevents server A from mounting attacks
	prevents server A from mounting attacks on server B via the user's		on server B via the user's browser, which protects both the user
	browser, which protects both the user (e.g., from misuse of his		(e.g., from misuse of his credentials) and the server (e.g., from DoS
	credentials) and the server (e.g., from DoS attack).		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
	iframes from different origins on the same page--cannot.		iframes from different origins on the same page--cannot.
	3.3. Bypassing SOP: CORS, WebSockets, and consent to communicate		3.3. Bypassing SOP: CORS, WebSockets, and consent to communicate
	skipping to change at page 8, line 21		skipping to change at page 7, line 25
	The W3C Cross-Origin Resource Sharing (CORS) spec [CORS] is a		The W3C Cross-Origin Resource Sharing (CORS) spec [CORS] is a
	response to this demand. In CORS, when a script from origin A		response to this demand. In CORS, when a script from origin A
	executes what would otherwise be a forbidden cross-origin request,		executes what would otherwise be a forbidden cross-origin request,
	the browser instead contacts the target server to determine whether		the browser instead contacts the target server to determine whether
	it is willing to allow cross-origin requests from A. If it is so		it is willing to allow cross-origin requests from A. If it is so
	willing, the browser then allows the request. This consent		willing, the browser then allows the request. This consent
	verification process is designed to safely allow cross-origin		verification process is designed to safely allow cross-origin
	requests.		requests.
	While CORS is designed to allow cross-origin HTTP requests,		While CORS is designed to allow cross-origin HTTP requests,
	WebSockets [I-D.ietf-hybi-thewebsocketprotocol] allows cross-origin		WebSockets [RFC6455] allows cross-origin establishment of transparent
	establishment of transparent channels. Once a WebSockets connection		channels. Once a WebSockets connection has been established from a
	has been established from a script to a site, the script can exchange		script to a site, the script can exchange any traffic it likes
	any traffic it likes without being required to frame it as a series		without being required to frame it as a series of HTTP request/
	of HTTP request/response transactions. As with CORS, a WebSockets		response transactions. As with CORS, a WebSockets transaction starts
	transaction starts with a consent verification stage to avoid		with a consent verification stage to avoid allowing scripts to simply
	allowing scripts to simply send arbitrary data to another origin.		send arbitrary data to another origin.
	While consent verification is conceptually simple--just do a		While consent verification is conceptually simple--just do a
	handshake before you start exchanging the real data--experience has		handshake before you start exchanging the real data--experience has
	shown that designing a correct consent verification system is		shown that designing a correct consent verification system is
	difficult. In particular, Huang et al. [huang-w2sp] have shown		difficult. In particular, Huang et al. [huang-w2sp] have shown
	vulnerabilities in the existing Java and Flash consent verification		vulnerabilities in the existing Java and Flash consent verification
	techniques and in a simplified version of the WebSockets handshake.		techniques and in a simplified version of the WebSockets handshake.
	In particular, it is important to be wary of CROSS-PROTOCOL attacks		In particular, it is important to be wary of CROSS-PROTOCOL attacks
	in which the attacking script generates traffic which is acceptable		in which the attacking script generates traffic which is acceptable
	to some non-Web protocol state machine. In order to resist this form		to some non-Web protocol state machine. In order to resist this form
	skipping to change at page 11, line 40		skipping to change at page 10, line 48
	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
	necessary 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. Callee-oriented consent provided by the calling site not
	because a malicious site can claim that the user is calling any user		work well because a malicious site can claim that the user is calling
	of his choice. One fix for this is to tie calls to a		any user 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 tying 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. [I-D.ietf-rtcweb-security-arch] and
			[I-D.rescorla-rtcweb-generic-idp] describe mechanisms which
			facilitate this sort of consent.
	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. Naturally, it is somewhat challenging to design UI
	interface, since managing individual user permissions can be painful.		primitives which express this sort of policy.
	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
	the site which you have authorized for calling not to bug you then
	your security situation is not really ideal. It is RECOMMENDED that
	browsers have explicit (and obvious) indicators that they are in a
	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 secure 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
	skipping to change at page 13, line 8		skipping to change at page 12, line 12
	Even if calls are only possible from HTTPS sites, if the site embeds		Even if calls are only possible from HTTPS sites, if the site embeds
	active content (e.g., JavaScript) that is fetched over HTTP or from		active content (e.g., JavaScript) that is fetched over HTTP or from
	an untrusted site, because that JavaScript is executed in the		an untrusted site, because that JavaScript is executed in the
	security context of the page [finer-grained]. Thus, it is also		security context of the page [finer-grained]. Thus, it is also
	dangerous to allow RTC-Web functionality from HTTPS origins that		dangerous to allow RTC-Web functionality from HTTPS origins that
	embed mixed content. Note: this issue is not restricted to PAGES		embed mixed content. Note: this issue is not restricted to PAGES
	which contain mixed content. If a page from a given origin ever		which contain mixed content. If a page from a given origin ever
	loads mixed content then it is possible for a network attacker to		loads mixed content then it is possible for a network attacker to
	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)
	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
	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
	network access via the browser introduces the risk of using the		network access via the browser introduces the risk of using 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
	skipping to change at page 14, line 20		skipping to change at page 13, line 19
	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.
	In particular, this is true for the data channel if DTLS is used		In particular, this is true for the data channel if DTLS is used
	because DTLS (with the anti-chosen plaintext mechanisms required by		because DTLS (with the anti-chosen plaintext mechanisms required by
	TLS 1.1) does not allow the attacker to generate predictable		TLS 1.1) does not allow the attacker to generate predictable
	ciphertext. However, with TCP the risk of transparent proxies		ciphertext. However, with TCP the risk of transparent proxies
	becomes much more severe. If TCP is to be used, then WebSockets		becomes much more severe. If TCP is to be used, then WebSockets
	style masking MUST be employed.		style masking MUST be employed. [Note: current thinking in the
			RTCWEB WG is not to support TCP and to support SCTP over DTLS, thus
			removing the need for masking.]
	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 15, line 28		skipping to change at page 14, line 28
	probably the best choice.		probably the best choice.
	Once initial consent is verified, we also need to verify continuing		Once initial consent is verified, we also need to verify continuing
	consent, in order to avoid attacks where two people briefly share an		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		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		for a large, unstoppable, traffic flow to the network and then
	leaves. The appropriate technologies here are fairly similar to		leaves. The appropriate technologies here are fairly similar to
	those for initial consent, though are perhaps weaker since the		those for initial consent, though are perhaps weaker since the
	threats is less severe.		threats is less severe.
	[[ OPEN ISSUE: Exactly what should be the requirements here?
	Proposals include ICE all the time or ICE but with allowing one of
	these non-ICE things for legacy. ]]
	4.2.4. IP Location Privacy		4.2.4. IP Location Privacy
	Note that as soon as the callee sends their ICE candidates, the		Note that as soon as the callee sends their ICE candidates, the
	callee learns the callee's IP addresses. The callee's server		callee learns the callee's IP addresses. The callee's server
	reflexive address reveals a lot of information about the callee's		reflexive address reveals a lot of information about the callee's
	location. In order to avoid tracking, implementations may wish to		location. In order to avoid tracking, implementations may wish to
	suppress the start of ICE negotiation until the callee has answered.		suppress the start of ICE negotiation until the callee has answered.
	In addition, either side may wish to hide their location entirely by		In addition, either side may wish to hide their location entirely by
	forcing all traffic through a TURN server.		forcing all traffic through a TURN server.
	skipping to change at page 19, line 22		skipping to change at page 18, line 19
	avoid them needing to check it on every call. However, this is		avoid them needing to check it on every call. However, this is
	problematic for reasons indicated in Section 4.3.2.1. In principle		problematic for reasons indicated in Section 4.3.2.1. In principle
	it is of course possible to render a different UI element to indicate		it is of course possible to render a different UI element to indicate
	that calls are using an unauthenticated set of keying material		that calls are using an unauthenticated set of keying material
	(recall that the attacker can just present a slightly different name		(recall that the attacker can just present a slightly different name
	so that the attack shows the same UI as a call to a new device or to		so that the attack shows the same UI as a call to a new device or to
	someone you haven't called before) but as a practical matter, users		someone you haven't called before) but as a practical matter, users
	simply ignore such indicators even in the rather more dire case of		simply ignore such indicators even in the rather more dire case of
	mixed content warnings.		mixed content warnings.
	4.3.2.3. Recommendations		4.3.2.3. Third Party Identity
	[[ OPEN ISSUE: What are the best UI recommendations to make?
	Proposal: take the text from [I-D.kaufman-rtcweb-security-ui]
	Section 2]]
	[[ OPEN ISSUE: Exactly what combination of media security primitives		The conventional approach to providing communications identity has of
	should be specified and/or mandatory to implement? In particular,		course been to have some third party identity system (e.g., PKI) to
	should we allow DTLS-SRTP only, or both DTLS-SRTP and SDES. Should		authenticate the endpoints. Such mechanisms have proven to be too
	we allow RTP for backward compatibility? ]]		cumbersome for use by typical users (and nearly too cumbersome for
			administrators). However, a new generation of Web-based identity
			providers (BrowserID, Federated Google Login, Facebook Connect,
			OAuth, OpenID, WebFinger), has recently been developed and use Web
			technologies to provide lightweight (from the user's perspective)
			third-party authenticated transactions. It is possible (see
			[I-D.rescorla-rtcweb-generic-idp]) to use systems of this type to
			authenticate RTCWEB calls, linking them to existing user notions of
			identity (e.g., Facebook adjacencies). Calls which are authenticated
			in this fashion are naturally resistant even to active MITM attack by
			the calling site.
	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		Bernard Aboba, Harald Alvestrand, Cullen Jennings, Hadriel Kaplan (S
	4.2.1), Matthew Kaufman, Magnus Westerland.		4.2.1), Matthew Kaufman, Magnus Westerland.
	skipping to change at page 19, line 43		skipping to change at page 19, line 4
	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		Bernard Aboba, Harald Alvestrand, Cullen Jennings, Hadriel Kaplan (S
	4.2.1), Matthew Kaufman, Magnus Westerland.		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.ietf-rtcweb-security-arch]
	Barth, A., "The Web Origin Concept",		Rescorla, E., "RTCWEB Security Architecture",
	draft-abarth-origin-09 (work in progress), November 2010.		draft-ietf-rtcweb-security-arch-00 (work in progress),
			January 2012.
	[I-D.ietf-hybi-thewebsocketprotocol]
	Fette, I. and A. Melnikov, "The WebSocket protocol",
	draft-ietf-hybi-thewebsocketprotocol-17 (work in
	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.
			[I-D.rescorla-rtcweb-generic-idp]
			Rescorla, E., "RTCWEB Generic Identity Provider
			Interface", draft-rescorla-rtcweb-generic-idp-00 (work in
			progress), January 2012.
	[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,
	A., Peterson, J., Sparks, R., Handley, M., and E.		A., Peterson, J., Sparks, R., Handley, M., and E.
	Schooler, "SIP: Session Initiation Protocol", RFC 3261,		Schooler, "SIP: Session Initiation Protocol", RFC 3261,
	June 2002.		June 2002.
	[RFC3552] Rescorla, E. and B. Korver, "Guidelines for Writing RFC		[RFC3552] Rescorla, E. and B. Korver, "Guidelines for Writing RFC
	Text on Security Considerations", BCP 72, RFC 3552,		Text on Security Considerations", BCP 72, RFC 3552,
	July 2003.		July 2003.
	skipping to change at page 21, line 18		skipping to change at page 20, line 23
	[RFC5763] Fischl, J., Tschofenig, H., and E. Rescorla, "Framework		[RFC5763] Fischl, J., Tschofenig, H., and E. Rescorla, "Framework
	for Establishing a Secure Real-time Transport Protocol		for Establishing a Secure Real-time Transport Protocol
	(SRTP) Security Context Using Datagram Transport Layer		(SRTP) Security Context Using Datagram Transport Layer
	Security (DTLS)", RFC 5763, May 2010.		Security (DTLS)", RFC 5763, May 2010.
	[RFC6189] Zimmermann, P., Johnston, A., and J. Callas, "ZRTP: Media		[RFC6189] Zimmermann, P., Johnston, A., and J. Callas, "ZRTP: Media
	Path Key Agreement for Unicast Secure RTP", RFC 6189,		Path Key Agreement for Unicast Secure RTP", RFC 6189,
	April 2011.		April 2011.
			[RFC6454] Barth, A., "The Web Origin Concept", RFC 6454,
			December 2011.
			[RFC6455] Fette, I. and A. Melnikov, "The WebSocket Protocol",
			RFC 6455, December 2011.
	[abarth-rtcweb]		[abarth-rtcweb]
	Barth, A., "Prompting the user is security failure", RTC-		Barth, A., "Prompting the user is security failure", RTC-
	Web Workshop.		Web Workshop.
	[cranor-wolf]		[cranor-wolf]
	Sunshine, J., Egelman, S., Almuhimedi, H., Atri, N., and		Sunshine, J., Egelman, S., Almuhimedi, H., Atri, N., and
	L. cranor, "Crying Wolf: An Empirical Study of SSL Warning		L. cranor, "Crying Wolf: An Empirical Study of SSL Warning
	Effectiveness", Proceedings of the 18th USENIX Security		Effectiveness", Proceedings of the 18th USENIX Security
	Symposium, 2009.		Symposium, 2009.
	skipping to change at page 22, line 5		skipping to change at page 21, line 14
	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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