draft-ietf-rtcweb-jsep-03.txt		draft-ietf-rtcweb-jsep-04.txt
	Network Working Group J. Uberti		Network Working Group J. Uberti
	Internet-Draft Google		Internet-Draft Google
	Intended status: Standards Track C. Jennings		Intended status: Standards Track C. Jennings
	Expires: August 29, 2013 Cisco		Expires: March 22, 2014 Cisco
	February 25, 2013		September 18, 2013
	Javascript Session Establishment Protocol		Javascript Session Establishment Protocol
	draft-ietf-rtcweb-jsep-03		draft-ietf-rtcweb-jsep-04
	Abstract		Abstract
	This document describes the mechanisms for allowing a Javascript		This document describes the mechanisms for allowing a Javascript
	application to fully control the signaling plane of a multimedia		application to control the signaling plane of a multimedia session
	session via the interface specified in the W3C RTCPeerConnection API,		via the interface specified in the W3C RTCPeerConnection API, and
	and discusses how this relates to existing signaling protocols.		discusses how this relates to existing signaling protocols.
	Status of this Memo		Status of This Memo
	This Internet-Draft is submitted in full conformance with the		This Internet-Draft is submitted in full conformance with the
	provisions of BCP 78 and BCP 79.		provisions of BCP 78 and BCP 79.
	Internet-Drafts are working documents of the Internet Engineering		Internet-Drafts are working documents of the Internet Engineering
	Task Force (IETF). Note that other groups may also distribute		Task Force (IETF). Note that other groups may also distribute
	working documents as Internet-Drafts. The list of current Internet-		working documents as Internet-Drafts. The list of current Internet-
	Drafts is at http://datatracker.ietf.org/drafts/current/.		Drafts is at http://datatracker.ietf.org/drafts/current/.
	Internet-Drafts are draft documents valid for a maximum of six months		Internet-Drafts are draft documents valid for a maximum of six months
	and may be updated, replaced, or obsoleted by other documents at any		and may be updated, replaced, or obsoleted by other documents at any
	time. It is inappropriate to use Internet-Drafts as reference		time. It is inappropriate to use Internet-Drafts as reference
	material or to cite them other than as "work in progress."		material or to cite them other than as "work in progress."
	This Internet-Draft will expire on August 29, 2013.		This Internet-Draft will expire on March 22, 2014.
	Copyright Notice		Copyright Notice
	Copyright (c) 2013 IETF Trust and the persons identified as the		Copyright (c) 2013 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
	carefully, as they describe your rights and restrictions with respect		carefully, as they describe your rights and restrictions with respect
	to this document. Code Components extracted from this document must		to this document. Code Components extracted from this document must
	include Simplified BSD License text as described in Section 4.e of		include Simplified BSD License text as described in Section 4.e of
	the Trust Legal Provisions and are provided without warranty as		the Trust Legal Provisions and are provided without warranty as
	described in the Simplified BSD License.		described in the Simplified BSD License.
	Table of Contents		Table of Contents
	1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4		1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
	1.1. General Design of JSEP . . . . . . . . . . . . . . . . . . 4		1.1. General Design of JSEP . . . . . . . . . . . . . . . . . 3
	1.2. Other Approaches Considered . . . . . . . . . . . . . . . 6		1.2. Other Approaches Considered . . . . . . . . . . . . . . . 5
	2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 6		2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 6
	3. Semantics and Syntax . . . . . . . . . . . . . . . . . . . . . 7		3. Semantics and Syntax . . . . . . . . . . . . . . . . . . . . 6
	3.1. Signaling Model . . . . . . . . . . . . . . . . . . . . . 7		3.1. Signaling Model . . . . . . . . . . . . . . . . . . . . . 6
	3.2. Session Descriptions and State Machine . . . . . . . . . . 7		3.2. Session Descriptions and State Machine . . . . . . . . . 7
	3.3. Session Description Format . . . . . . . . . . . . . . . . 10		3.3. Session Description Format . . . . . . . . . . . . . . . 9
	3.4. ICE . . . . . . . . . . . . . . . . . . . . . . . . . . . 10		3.4. ICE . . . . . . . . . . . . . . . . . . . . . . . . . . . 10
	3.4.1. ICE Candidate Trickling . . . . . . . . . . . . . . . 10		3.4.1. ICE Candidate Trickling . . . . . . . . . . . . . . . 10
	3.4.1.1. ICE Candidate Format . . . . . . . . . . . . . . . 11		3.4.1.1. ICE Candidate Format . . . . . . . . . . . . . . 10
	3.5. Interactions With Forking . . . . . . . . . . . . . . . . 11		3.5. Interactions With Forking . . . . . . . . . . . . . . . . 11
	3.5.1. Sequential Forking . . . . . . . . . . . . . . . . . . 12		3.5.1. Sequential Forking . . . . . . . . . . . . . . . . . 11
	3.5.2. Parallel Forking . . . . . . . . . . . . . . . . . . . 12		3.5.2. Parallel Forking . . . . . . . . . . . . . . . . . . 12
	3.6. Session Rehydration . . . . . . . . . . . . . . . . . . . 13		3.6. Session Rehydration . . . . . . . . . . . . . . . . . . . 13
	4. Interface . . . . . . . . . . . . . . . . . . . . . . . . . . 14		4. Interface . . . . . . . . . . . . . . . . . . . . . . . . . . 13
	4.1. SDP Requirements . . . . . . . . . . . . . . . . . . . . . 14		4.1. Methods . . . . . . . . . . . . . . . . . . . . . . . . . 14
	4.2. Methods . . . . . . . . . . . . . . . . . . . . . . . . . 15		4.1.1. createOffer . . . . . . . . . . . . . . . . . . . . . 14
	4.2.1. createOffer . . . . . . . . . . . . . . . . . . . . . 15		4.1.2. createAnswer . . . . . . . . . . . . . . . . . . . . 15
	4.2.2. createAnswer . . . . . . . . . . . . . . . . . . . . . 16		4.1.3. SessionDescriptionType . . . . . . . . . . . . . . . 15
	4.2.3. SessionDescriptionType . . . . . . . . . . . . . . . . 17		4.1.3.1. Use of Provisional Answers . . . . . . . . . . . 16
	4.2.3.1. Use of Provisional Answers . . . . . . . . . . . . 18		4.1.3.2. Rollback . . . . . . . . . . . . . . . . . . . . 17
	4.2.3.2. Rollback . . . . . . . . . . . . . . . . . . . . . 18		4.1.4. setLocalDescription . . . . . . . . . . . . . . . . . 17
	4.2.4. setLocalDescription . . . . . . . . . . . . . . . . . 19		4.1.5. setRemoteDescription . . . . . . . . . . . . . . . . 18
	4.2.5. setRemoteDescription . . . . . . . . . . . . . . . . . 19		4.1.6. localDescription . . . . . . . . . . . . . . . . . . 18
	4.2.6. localDescription . . . . . . . . . . . . . . . . . . . 20		4.1.7. remoteDescription . . . . . . . . . . . . . . . . . . 18
	4.2.7. remoteDescription . . . . . . . . . . . . . . . . . . 20		4.1.8. updateIce . . . . . . . . . . . . . . . . . . . . . . 19
	4.2.8. updateIce . . . . . . . . . . . . . . . . . . . . . . 20		4.1.9. addIceCandidate . . . . . . . . . . . . . . . . . . . 19
	4.2.9. addIceCandidate . . . . . . . . . . . . . . . . . . . 21		5. SDP Interaction Procedures . . . . . . . . . . . . . . . . . 19
	5. SDP Interaction Procedures . . . . . . . . . . . . . . . . . . 21		5.1. SDP Requirements Overview . . . . . . . . . . . . . . . . 19
	5.1. Constructing an Offer . . . . . . . . . . . . . . . . . . 21		5.2. Constructing an Offer . . . . . . . . . . . . . . . . . . 21
	5.2. Generating an Answer . . . . . . . . . . . . . . . . . . . 21		5.2.1. Initial Offers . . . . . . . . . . . . . . . . . . . 21
	5.3. Parsing an Offer . . . . . . . . . . . . . . . . . . . . . 21		5.2.2. Subsequent Offers . . . . . . . . . . . . . . . . . . 25
	5.4. Parsing an Answer . . . . . . . . . . . . . . . . . . . . 21		5.2.3. Constraints Handling . . . . . . . . . . . . . . . . 26
	5.5. Applying a Local Description . . . . . . . . . . . . . . . 21		5.2.3.1. OfferToReceiveAudio . . . . . . . . . . . . . . . 26
	5.6. Applying a Remote Description . . . . . . . . . . . . . . 21		5.2.3.2. OfferToReceiveVideo . . . . . . . . . . . . . . . 27
	6. Configurable SDP Parameters . . . . . . . . . . . . . . . . . 21		5.2.3.3. VoiceActivityDetection . . . . . . . . . . . . . 27
	7. Security Considerations . . . . . . . . . . . . . . . . . . . 22		5.2.3.4. IceRestart . . . . . . . . . . . . . . . . . . . 27
	8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 23		5.3. Generating an Answer . . . . . . . . . . . . . . . . . . 27
	9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 23		5.3.1. Initial Answers . . . . . . . . . . . . . . . . . . . 27
	10. References . . . . . . . . . . . . . . . . . . . . . . . . . . 23		5.3.2. Subsequent Answers . . . . . . . . . . . . . . . . . 31
	10.1. Normative References . . . . . . . . . . . . . . . . . . . 23		5.3.3. Constraints Handling . . . . . . . . . . . . . . . . 31
	10.2. Informative References . . . . . . . . . . . . . . . . . . 24		5.4. Parsing an Offer . . . . . . . . . . . . . . . . . . . . 31
	Appendix A. JSEP Implementation Examples . . . . . . . . . . . . 25		5.5. Parsing an Answer . . . . . . . . . . . . . . . . . . . . 31
	A.1. Example API Flows . . . . . . . . . . . . . . . . . . . . 25		5.6. Applying a Local Description . . . . . . . . . . . . . . 31
	A.1.1. Call using ROAP . . . . . . . . . . . . . . . . . . . 26		5.7. Applying a Remote Description . . . . . . . . . . . . . . 31
	A.1.2. Call using XMPP . . . . . . . . . . . . . . . . . . . 26
	A.1.3. Adding video to a call, using XMPP . . . . . . . . . . 28		6. Configurable SDP Parameters . . . . . . . . . . . . . . . . . 31
	A.1.4. Simultaneous add of video streams, using XMPP . . . . 28		7. Security Considerations . . . . . . . . . . . . . . . . . . . 33
	A.1.5. Call using SIP . . . . . . . . . . . . . . . . . . . . 29		8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 33
	A.1.6. Handling early media (e.g. 1-800-GO FEDEX), using		9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 33
	SIP . . . . . . . . . . . . . . . . . . . . . . . . . 30		10. References . . . . . . . . . . . . . . . . . . . . . . . . . 33
	Appendix B. Change log . . . . . . . . . . . . . . . . . . . . . 31		10.1. Normative References . . . . . . . . . . . . . . . . . . 33
	Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 32		10.2. Informative References . . . . . . . . . . . . . . . . . 35
			Appendix A. JSEP Implementation Examples . . . . . . . . . . . . 36
			A.1. Example API Flows . . . . . . . . . . . . . . . . . . . . 36
			A.1.1. Call using ROAP . . . . . . . . . . . . . . . . . . . 36
			A.1.2. Call using XMPP . . . . . . . . . . . . . . . . . . . 37
			A.1.3. Adding video to a call, using XMPP . . . . . . . . . 38
			A.1.4. Simultaneous add of video streams, using XMPP . . . . 39
			A.1.5. Call using SIP . . . . . . . . . . . . . . . . . . . 40
			A.1.6. Handling early media (e.g. 1-800-GO FEDEX), using SIP 40
			A.2. Example Session Descriptions . . . . . . . . . . . . . . 41
			A.2.1. createOffer . . . . . . . . . . . . . . . . . . . . . 41
			A.2.2. createAnswer . . . . . . . . . . . . . . . . . . . . 43
			Appendix B. Change log . . . . . . . . . . . . . . . . . . . . . 44
			Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 45
	1. Introduction		1. Introduction
	This document describes how the W3C WEBRTC RTCPeerConnection		This document describes how the W3C WEBRTC RTCPeerConnection
	interface[W3C.WD-webrtc-20111027] is used to control the setup,		interface[W3C.WD-webrtc-20111027] is used to control the setup,
	management and teardown of a multimedia session.		management and teardown of a multimedia session.
	1.1. General Design of JSEP		1.1. General Design of JSEP
	The thinking behind WebRTC call setup has been to fully specify and		The thinking behind WebRTC call setup has been to fully specify and
	skipping to change at page 6, line 42		skipping to change at page 6, line 12
	generating offers and answers out of the browser. Instead of		generating offers and answers out of the browser. Instead of
	providing createOffer/createAnswer methods within the browser, this		providing createOffer/createAnswer methods within the browser, this
	approach would instead expose a getCapabilities API which would		approach would instead expose a getCapabilities API which would
	provide the application with the information it needed in order to		provide the application with the information it needed in order to
	generate its own session descriptions. This increases the amount of		generate its own session descriptions. This increases the amount of
	work that the application needs to do; it needs to know how to		work that the application needs to do; it needs to know how to
	generate session descriptions from capabilities, and especially how		generate session descriptions from capabilities, and especially how
	to generate the correct answer from an arbitrary offer and the		to generate the correct answer from an arbitrary offer and the
	supported capabilities. While this could certainly be addressed by		supported capabilities. While this could certainly be addressed by
	using a library like the one mentioned above, it basically forces the		using a library like the one mentioned above, it basically forces the
	use of said library even for a simple example. Providing		use of said library even for a simple example. Providing createOffer
	createOffer/createAnswer avoids this problem, but still allows		/createAnswer avoids this problem, but still allows applications to
	applications to generate their own offers/answers (to a large extent)		generate their own offers/answers (to a large extent) if they choose,
	if they choose, using the description generated by createOffer as an		using the description generated by createOffer as an indication of
	indication of the browser's capabilities.		the browser's capabilities.
	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 [RFC2119].		document are to be interpreted as described in [RFC2119].
	3. Semantics and Syntax		3. Semantics and Syntax
	3.1. Signaling Model		3.1. Signaling Model
	skipping to change at page 7, line 22		skipping to change at page 6, line 40
	fashion described by [RFC3264] (offer/answer) in order for both sides		fashion described by [RFC3264] (offer/answer) in order for both sides
	of the session to know how to conduct the session. JSEP provides		of the session to know how to conduct the session. JSEP provides
	mechanisms to create offers and answers, as well as to apply them to		mechanisms to create offers and answers, as well as to apply them to
	a session. However, the browser is totally decoupled from the actual		a session. However, the browser is totally decoupled from the actual
	mechanism by which these offers and answers are communicated to the		mechanism by which these offers and answers are communicated to the
	remote side, including addressing, retransmission, forking, and glare		remote side, including addressing, retransmission, forking, and glare
	handling. These issues are left entirely up to the application; the		handling. These issues are left entirely up to the application; the
	application has complete control over which offers and answers get		application has complete control over which offers and answers get
	handed to the browser, and when.		handed to the browser, and when.
	+-----------+ +-----------+		+-----------+ +-----------+
	| Web App |<--- App-Specific Signaling -->| Web App |		| Web App |<--- App-Specific Signaling -->| Web App |
	+-----------+ +-----------+		+-----------+ +-----------+
	^ ^		^ ^
	| SDP | SDP		| SDP | SDP
	V V		V V
	+-----------+ +-----------+		+-----------+ +-----------+
	| Browser |<----------- Media ------------>| Browser |		| Browser |<----------- Media ------------>| Browser |
	+-----------+ +-----------+		+-----------+ +-----------+
	Figure 1: JSEP Signaling Model		Figure 1: JSEP Signaling Model
	3.2. Session Descriptions and State Machine		3.2. Session Descriptions and State Machine
	In order to establish the media plane, the user agent needs specific		In order to establish the media plane, the user agent needs specific
	parameters to indicate what to transmit to the remote side, as well		parameters to indicate what to transmit to the remote side, as well
	as how to handle the media that is received. These parameters are		as how to handle the media that is received. These parameters are
	determined by the exchange of session descriptions in offers and		determined by the exchange of session descriptions in offers and
	answers, and there are certain details to this process that must be		answers, and there are certain details to this process that must be
	skipping to change at page 9, line 23		skipping to change at page 8, line 44
	v | v |		v | v |
	+---------------+ | +---------------+ |		+---------------+ | +---------------+ |
	| |----/ | |----/		| |----/ | |----/
	| | setLocal(PRANSWER) | |		| | setLocal(PRANSWER) | |
	| Remote-Offer |------------------- >| Local-Pranswer|		| Remote-Offer |------------------- >| Local-Pranswer|
	| | | |		| | | |
	| | | |		| | | |
	+---------------+ +---------------+		+---------------+ +---------------+
	^ | |		^ | |
	| | setLocal(ANSWER) |		| | setLocal(ANSWER) |
	setRemote(OFFER) | | |		setRemote(OFFER) | |
	| V setLocal(ANSWER) |		| V setLocal(ANSWER) |
	+---------------+ |		+---------------+ |
	| | |		| | |
	| | |		| | |
	| Stable |<---------------------------+		| Stable |<---------------------------+
	| | |		| | |
	| | |		| | |
	+---------------+ setRemote(ANSWER) |		+---------------+ setRemote(ANSWER) |
	^ | |		^ | |
	| | setLocal(OFFER) |		| | setLocal(OFFER) |
	setRemote(ANSWER)| | |		setRemote(ANSWER) | |
	| V |		| V |
	+---------------+ +---------------+		+---------------+ +---------------+
	| | | |		| | | |
	| | setRemote(PRANSWER) | |		| | setRemote(PRANSWER) | |
	| Local-Offer |------------------- >|Remote-Pranswer|		| Local-Offer |------------------- >|Remote-Pranswer|
	| | | |		| | | |
	| |----\ | |----\		| |----\ | |----\
	+---------------+ | +---------------+ |		+---------------+ | +---------------+ |
	^ | ^ |		^ | ^ |
	| | | |		| | | |
	skipping to change at page 10, line 26		skipping to change at page 9, line 47
	However, to simplify Javascript processing, and provide for future		However, to simplify Javascript processing, and provide for future
	flexibility, the SDP syntax is encapsulated within a		flexibility, the SDP syntax is encapsulated within a
	SessionDescription object, which can be constructed from SDP, and be		SessionDescription object, which can be constructed from SDP, and be
	serialized out to SDP. If future specifications agree on a JSON		serialized out to SDP. If future specifications agree on a JSON
	format for session descriptions, we could easily enable this object		format for session descriptions, we could easily enable this object
	to generate and consume that JSON.		to generate and consume that JSON.
	Other methods may be added to SessionDescription in the future to		Other methods may be added to SessionDescription in the future to
	simplify handling of SessionDescriptions from Javascript. In the		simplify handling of SessionDescriptions from Javascript. In the
	meantime, it would be simple to write a Javascript library to perform		meantime, Javascript libraries can be used to perform these
	these manipulations.		manipulations.
			Note that most applications should be able to treat the
			SessionDescriptions produced and consumed by these various API calls
			as opaque blobs; that is, the application will not need to read or
			change them. The W3C API will provide appropriate APIs to allow the
			application to control various session parameters, which will provide
			the necessary information to the browser about what sort of
			SessionDescription to produce.
	3.4. ICE		3.4. ICE
	When a new ICE candidate is available, the ICE Agent will notify the		When a new ICE candidate is available, the ICE Agent will notify the
	application via a callback; these candidates will automatically be		application via a callback; these candidates will automatically be
	added to the local session description. When all candidates have		added to the local session description. When all candidates have
	been gathered, the callback will also be invoked to signal that the		been gathered, the callback will also be invoked to signal that the
	gathering process is complete.		gathering process is complete.
	3.4.1. ICE Candidate Trickling		3.4.1. ICE Candidate Trickling
	Candidate trickling is a technique through which a caller may		Candidate trickling is a technique through which a caller may
	incrementally provide candidates to the callee after the initial		incrementally provide candidates to the callee after the initial
	offer has been dispatched; the semantics of "Trickle ICE" are defined		offer has been dispatched; the semantics of "Trickle ICE" are defined
	in [I-D.rescorla-mmusic-ice-trickle]. This process allows the callee		in [I-D.ivov-mmusic-trickle-ice]. This process allows the callee to
	to begin acting upon the call and setting up the ICE (and perhaps		begin acting upon the call and setting up the ICE (and perhaps DTLS)
	DTLS) connections immediately, without having to wait for the caller		connections immediately, without having to wait for the caller to
	to gather all possible candidates. This results in faster call		gather all possible candidates. This results in faster call startup
	startup in cases where gathering is not performed prior to initating		in cases where gathering is not performed prior to initiating the
	the call.		call.
	JSEP supports optional candidate trickling by providing APIs that		JSEP supports optional candidate trickling by providing APIs that
	provide control and feedback on the ICE candidate gathering process.		provide control and feedback on the ICE candidate gathering process.
	Applications that support candidate trickling can send the initial		Applications that support candidate trickling can send the initial
	offer immediately and send individual candidates when they get the		offer immediately and send individual candidates when they get the
	notified of a new candidate; applications that do not support this		notified of a new candidate; applications that do not support this
	feature can simply wait for the indication that gathering is		feature can simply wait for the indication that gathering is
	complete, and then create and send their offer, with all the		complete, and then create and send their offer, with all the
	candidates, at this time.		candidates, at this time.
	skipping to change at page 11, line 22		skipping to change at page 10, line 51
	using the new remote candidates for connectivity checks.		using the new remote candidates for connectivity checks.
	3.4.1.1. ICE Candidate Format		3.4.1.1. ICE Candidate Format
	As with session descriptions, the syntax of the IceCandidate object		As with session descriptions, the syntax of the IceCandidate object
	provides some abstraction, but can be easily converted to and from		provides some abstraction, but can be easily converted to and from
	the SDP candidate lines.		the SDP candidate lines.
	The candidate lines are the only SDP information that is contained		The candidate lines are the only SDP information that is contained
	within IceCandidate, as they represent the only information needed		within IceCandidate, as they represent the only information needed
	that is not present in the initial offer (i.e. for trickle		that is not present in the initial offer (i.e. for trickle
	candidates). This information is carried with the same syntax as the		candidates). This information is carried with the same syntax as the
	"candidate-attribute" field defined for ICE. For example:		"candidate-attribute" field defined for ICE. For example:
	candidate:1 1 UDP 1694498815 192.0.2.33 10000 typ host		candidate:1 1 UDP 1694498815 192.0.2.33 10000 typ host
	The IceCandidate object also contains fields to indicate which m=		The IceCandidate object also contains fields to indicate which m=
	line it should be associated with. The m line can be identified in		line it should be associated with. The m line can be identified in
	one of two ways; either by a m-line index, or a MID. The m-line		one of two ways; either by a m-line index, or a MID. The m-line
	index is a zero-based index, referring to the Nth m-line in the SDP.		index is a zero-based index, referring to the Nth m-line in the SDP.
	The MID uses the "media stream identification", as defined in		The MID uses the "media stream identification", as defined in
	[RFC3388] , to identify the m-line. WebRTC implementations creating		[RFC5888] , to identify the m-line. WebRTC implementations creating
	an ICE Candidate object MUST populate both of these fields.		an ICE Candidate object MUST populate both of these fields.
	Implementations receiving an ICE Candidate object SHOULD use the MID		Implementations receiving an ICE Candidate object SHOULD use the MID
	if they implement that functionality, or the m-line index, if not.		if they implement that functionality, or the m-line index, if not.
	3.5. Interactions With Forking		3.5. Interactions With Forking
	Some call signaling systems allow various types of forking where an		Some call signaling systems allow various types of forking where an
	SDP Offer may be provided to more than one device. For example, SIP		SDP Offer may be provided to more than one device. For example, SIP
	[RFC3261] defines both a "Parallel Search" and "Sequential Search".		[RFC3261] defines both a "Parallel Search" and "Sequential Search".
	Although these are primarily signaling level issues that are outside		Although these are primarily signaling level issues that are outside
	skipping to change at page 13, line 14		skipping to change at page 12, line 46
	to simultaneously exchange media with multiple peers, the flow is		to simultaneously exchange media with multiple peers, the flow is
	slightly more complex, but the Javascript application can follow the		slightly more complex, but the Javascript application can follow the
	strategy that [RFC3960] describes using UPDATE. (It is worth noting		strategy that [RFC3960] describes using UPDATE. (It is worth noting
	that use cases where this is the desired behavior are very unusual.)		that use cases where this is the desired behavior are very unusual.)
	The UPDATE approach allows the signaling to set up a separate media		The UPDATE approach allows the signaling to set up a separate media
	flow for each peer that it wishes to exchange media with. In JSEP,		flow for each peer that it wishes to exchange media with. In JSEP,
	this offer used in the UPDATE would be formed by simply creating a		this offer used in the UPDATE would be formed by simply creating a
	new PeerConnection and making sure that the same local media streams		new PeerConnection and making sure that the same local media streams
	have been added into this new PeerConnection. Then the new		have been added into this new PeerConnection. Then the new
	PeerConnection object would produce a SDP offer that could be used by		PeerConnection object would produce a SDP offer that could be used by
	the signaling to perform the UPDATE strategy discussed in [RFC3960] .		the signaling to perform the UPDATE strategy discussed in [RFC3960].
	As a result of sharing the media streams, the application will end up		As a result of sharing the media streams, the application will end up
	with N parallel PeerConnection sessions, each with a local and remote		with N parallel PeerConnection sessions, each with a local and remote
	description and their own local and remote addresses. The media flow		description and their own local and remote addresses. The media flow
	from these sessions can be managed by specifying SDP direction		from these sessions can be managed by specifying SDP direction
	attributes in the descriptions, or the application can choose to play		attributes in the descriptions, or the application can choose to play
	out the media from all sessions mixed together. Of course, if the		out the media from all sessions mixed together. Of course, if the
	application wants to only keep a single session, it can simply		application wants to only keep a single session, it can simply
	terminate the sessions that it no longer needs.		terminate the sessions that it no longer needs.
	skipping to change at page 13, line 43		skipping to change at page 13, line 27
	than normal call signaling messages.		than normal call signaling messages.
	With rehydration, the current signaling state is persisted somewhere		With rehydration, the current signaling state is persisted somewhere
	outside of the page, perhaps on the application server, or in browser		outside of the page, perhaps on the application server, or in browser
	local storage. The page is then reloaded, the saved signaling state		local storage. The page is then reloaded, the saved signaling state
	is retrieved, and a new PeerConnection object is created for the		is retrieved, and a new PeerConnection object is created for the
	session. The previously obtained MediaStreams are re-acquired, and		session. The previously obtained MediaStreams are re-acquired, and
	are given the same IDs as the original session; this ensures the IDs		are given the same IDs as the original session; this ensures the IDs
	in use by the remote side continue to work. Next, a new offer is		in use by the remote side continue to work. Next, a new offer is
	generated by the new PeerConnection; this offer will have new ICE and		generated by the new PeerConnection; this offer will have new ICE and
	possibly new SDES credentials (since the old ICE and SRTP state has		possibly new DTLS-SRTP certificate fingerprints (since the old ICE
	been lost). Finally, this offer is used to re-initiate the session		and SRTP state has been lost). Finally, this offer is used to re-
	with the existing remote endpoint, who simply sees the new offer as		initiate the session with the existing remote endpoint, who simply
	an in-call renegotiation, and replies with an answer that can be		sees the new offer as an in-call renegotiation, and replies with an
	supplied to setRemoteDescription. ICE processing proceeds as usual,		answer that can be supplied to setRemoteDescription. ICE processing
	and as soon as connectivity is established, the session will be back		proceeds as usual, and as soon as connectivity is established, the
	up and running again.		session will be back up and running again.
	[OPEN ISSUE: EKR proposed an alternative rehydration approach where		[OPEN ISSUE: EKR proposed an alternative rehydration approach where
	the actual internal PeerConnection object in the browser was kept		the actual internal PeerConnection object in the browser was kept
	alive for some time after the web page was killed and provided some		alive for some time after the web page was killed and provided some
	way for a new page to acquire the old PeerConnection object.]		way for a new page to acquire the old PeerConnection object.]
	4. Interface		4. Interface
	This section details the basic operations that must be present to		This section details the basic operations that must be present to
	implement JSEP functionality. The actual API exposed in the W3C API		implement JSEP functionality. The actual API exposed in the W3C API
	may have somewhat different syntax, but should map easily to these		may have somewhat different syntax, but should map easily to these
	concepts.		concepts.
	4.1. SDP Requirements		4.1. Methods
	Note: The text in this section may not represent working group
	consensus and is put here so that the working group can discuss it
	and find out how to change it such that it does have consensus.
	When generating SDP blobs, either for offers or answers, the
	generated SDP needs to conform to the following specifications.
	Similarly, in order to properly process received SDP blobs,
	implementations need to implement the functionality described in the
	following specifications. This list is derived from
	[I-D.ietf-rtcweb-rtp-usage].
	R-1 [RFC4566] is the base SDP specification and MUST be
	implemented.
	R-2 The [RFC5888] grouping framework MUST be implemented for
	signaling grouping information, and MUST be used to identify m=
	lines via the a=mid attribute.
	R-3 [RFC5124] MUST be supported for signaling RTP/SAVPF RTP
	profile.
	R-4 [RFC4585] MUST be implemented to signal RTCP based feedback.
	R-5 [RFC5245] MUST be implemented for signaling the ICE candidate
	lines corresponding to each media stream.
	R-6 [RFC5761] MUST be implemented to signal multiplexing of RTP and
	RTCP.
	R-7 The SDP atributes of "sendonly", "recvonly", "inactive", and
	"sendrecv" from [RFC4566] MUST be implemented to signal
	information about media direction.
	R-8 [RFC5576] MUST be implemented to signal RTP SSRC values. [OPEN
	ISSUE; depends on BUNDLE and how we choose to represent
	multiple media sources]
	R-9 [RFC5763] MUST be implemented to signal DTLS certificate
	fingerprints.
	R-10 [RFC5506] MAY be implemented to signal Reduced-Size RTCP
	messages.
	R-11 [RFC3556] with bandwidth modifiers MAY be supported for
	specifying RTCP bandwidth as a fraction of the media bandwidth,
	RTCP fraction allocated to the senders and setting maximum
	media bit-rate boundaries.
	R-12 [RFC4568] MAY be implemented to signal SDES SRTP keying
	information.
	R-13 A TBD-draft MUST be supported, in order to signal associations
	between RTP objects and W3C MediaStreams and MediaStreamTracks
	in a standard way. Though there is not yet WG consensus in
	this area, this TBD-draft is very likely to be
	[I-D.alvestrand-mmusic-msid].
	R-14 A TBD-draft MUST be supported to signal the use or multiplexing
	RTP somethings on a single UDP port, in order to avoid
	excessive use of port number resources. Though there is not
	yet WG consensus in this area, this TBD-draft is very likely to
	be [I-D.holmberg-mmusic-sdp-bundle-negotiation].
	As required by [RFC4566] Section 5.13 JSEP implementations MUST
	ignore unknown attributes (a=) lines.
	Example SDP for RTCWeb call flows can be found in
	[I-D.nandakumar-rtcweb-sdp]. [TODO: since we are starting to
	specify how to handle SDP in this document, should these call flows
	be merged into this document, or this link moved to the examples
	section?]
	4.2. Methods
	4.2.1. createOffer		4.1.1. createOffer
	The createOffer method generates a blob of SDP that contains a		The createOffer method generates a blob of SDP that contains a
	[RFC3264] offer with the supported configurations for the session,		[RFC3264] offer with the supported configurations for the session,
	including descriptions of the local MediaStreams attached to this		including descriptions of the local MediaStreams attached to this
	PeerConnection, the codec/RTP/RTCP options supported by this		PeerConnection, the codec/RTP/RTCP options supported by this
	implementation, and any candidates that have been gathered by the ICE		implementation, and any candidates that have been gathered by the ICE
	Agent. A constraints parameters may be supplied to provide		Agent. A constraints parameters may be supplied to provide
	additional control over the generated offer. This constraints		additional control over the generated offer. This constraints
	parameter should allow for the following manipulations to be		parameter should allow for the following manipulations to be
	performed:		performed:
	skipping to change at page 16, line 12		skipping to change at page 14, line 33
	o To trigger an ICE restart, for the purpose of reestablishing		o To trigger an ICE restart, for the purpose of reestablishing
	connectivity.		connectivity.
	o For re-offer cases, to request an offer that contains the full set		o For re-offer cases, to request an offer that contains the full set
	of supported capabilities, as opposed to just the currently		of supported capabilities, as opposed to just the currently
	negotiated parameters.		negotiated parameters.
	In the initial offer, the generated SDP will contain all desired		In the initial offer, the generated SDP will contain all desired
	functionality for the session (certain parts that are supported but		functionality for the session (certain parts that are supported but
	not desired by default may be omitted); for each SDP line, the		not desired by default may be omitted); for each SDP line, the
	generation of the SDP must follow the process defined for generating		generation of the SDP will follow the process defined for generating
	an initial offer from the document (listed in Section 4.1) that		an initial offer from the document that specifies the given SDP line.
	specifies the given SDP line.		The exact handling of initial offer generation is detailed in
			Section 5.2.1. below.
	In the event createOffer is called after the session is established,		In the event createOffer is called after the session is established,
	createOffer will generate an offer to modify the current session		createOffer will generate an offer to modify the current session
	based on any changes that have been made to the session, e.g. adding		based on any changes that have been made to the session, e.g. adding
	or removing MediaStreams, or requesting an ICE restart. For each		or removing MediaStreams, or requesting an ICE restart. For each
	existing stream, the generation of each SDP line must follow the		existing stream, the generation of each SDP line must follow the
	process defined for generating an updated offer from the document		process defined for generating an updated offer from the document
	that specfies the given SDP line. For each new stream, the		that specifies the given SDP line. For each new stream, the
	generation of the SDP must follow the process of generating an		generation of the SDP must follow the process of generating an
	initial offer, as mentioned above. If no changes have been made, or		initial offer, as mentioned above. If no changes have been made, or
	for SDP lines that are unaffected by the requested changes, the offer		for SDP lines that are unaffected by the requested changes, the offer
	will only contain the parameters negotiated by the last offer-answer		will only contain the parameters negotiated by the last offer-answer
	exchange.		exchange. The exact handling of subsequent offer generation is
			detailed in Section 5.2.2. below.
	Session descriptions generated by createOffer must be immediately		Session descriptions generated by createOffer must be immediately
	usable by setLocalDescription; if a system has limited resources		usable by setLocalDescription; if a system has limited resources
	(e.g. a finite number of decoders), createOffer should return an		(e.g. a finite number of decoders), createOffer should return an
	offer that reflects the current state of the system, so that		offer that reflects the current state of the system, so that
	setLocalDescription will succeed when it attempts to acquire those		setLocalDescription will succeed when it attempts to acquire those
	resources. Because this method may need to inspect the system state		resources. Because this method may need to inspect the system state
	to determine the currently available resources, it may be implemented		to determine the currently available resources, it may be implemented
	as an async operation.		as an async operation.
	Calling this method may do things such as generate new ICE		Calling this method may do things such as generate new ICE
	credentials, but does not result in candidate gathering, or cause		credentials, but does not result in candidate gathering, or cause
	media to start or stop flowing.		media to start or stop flowing.
	4.2.2. createAnswer		4.1.2. createAnswer
	The createAnswer method generates a blob of SDP that contains a		The createAnswer method generates a blob of SDP that contains a
	[RFC3264] SDP answer with the supported configuration for the session		[RFC3264] SDP answer with the supported configuration for the session
	that is compatible with the parameters supplied in the offer. Like		that is compatible with the parameters supplied in the offer. Like
	createOffer, the returned blob contains descriptions of the local		createOffer, the returned blob contains descriptions of the local
	MediaStreams attached to this PeerConnection, the codec/RTP/RTCP		MediaStreams attached to this PeerConnection, the codec/RTP/RTCP
	options negotiated for this session, and any candidates that have		options negotiated for this session, and any candidates that have
	been gathered by the ICE Agent. A constraints parameter may be		been gathered by the ICE Agent. A constraints parameter may be
	supplied to provide additional control over the generated answer.		supplied to provide additional control over the generated answer.
	As an answer, the generated SDP will contain a specific configuration		As an answer, the generated SDP will contain a specific configuration
	that specifies how the media plane should be established; for each		that specifies how the media plane should be established; for each
	SDP line, the generation of the SDP must follow the process defined		SDP line, the generation of the SDP must follow the process defined
	for generating an answer from the document that specifies the given		for generating an answer from the document that specifies the given
	SDP line.		SDP line. The exact handling of answer generation is detailed in
			Section 5.3. below.
	Session descriptions generated by createAnswer must be immediately		Session descriptions generated by createAnswer must be immediately
	usable by setLocalDescription; like createOffer, the returned		usable by setLocalDescription; like createOffer, the returned
	description should reflect the current state of the system. Because		description should reflect the current state of the system. Because
	this method may need to inspect the system state to determine the		this method may need to inspect the system state to determine the
	currently available resources, it may need to be implemented as an		currently available resources, it may need to be implemented as an
	async operation.		async operation.
	Calling this method may do things such as generate new ICE		Calling this method may do things such as generate new ICE
	credentials, but does not trigger candidate gathering or change media		credentials, but does not trigger candidate gathering or change media
	state.		state.
	4.2.3. SessionDescriptionType		4.1.3. SessionDescriptionType
	Session description objects (RTCSessionDescription) may be of type		Session description objects (RTCSessionDescription) may be of type
	"offer", "pranswer", and "answer". These types provide information		"offer", "pranswer", and "answer". These types provide information
	as to how the description parameter should be parsed, and how the		as to how the description parameter should be parsed, and how the
	media state should be changed.		media state should be changed.
	"offer" indicates that a description should be parsed as an offer;		"offer" indicates that a description should be parsed as an offer;
	said description may include many possible media configurations. A		said description may include many possible media configurations. A
	description used as an "offer" may be applied anytime the		description used as an "offer" may be applied anytime the
	PeerConnection is in a stable state, or as an update to a previously		PeerConnection is in a stable state, or as an update to a previously
	skipping to change at page 18, line 11		skipping to change at page 16, line 36
	were allocated as a result of the offer. As such, the application		were allocated as a result of the offer. As such, the application
	can use some discretion on whether an answer should be applied as		can use some discretion on whether an answer should be applied as
	provisional or final, and can change the type of the session		provisional or final, and can change the type of the session
	description as needed. For example, in a serial forking scenario, an		description as needed. For example, in a serial forking scenario, an
	application may receive multiple "final" answers, one from each		application may receive multiple "final" answers, one from each
	remote endpoint. The application could choose to accept the initial		remote endpoint. The application could choose to accept the initial
	answers as provisional answers, and only apply an answer as final		answers as provisional answers, and only apply an answer as final
	when it receives one that meets its criteria (e.g. a live user		when it receives one that meets its criteria (e.g. a live user
	instead of voicemail).		instead of voicemail).
	4.2.3.1. Use of Provisional Answers		4.1.3.1. Use of Provisional Answers
	Most web applications will not need to create answers using the		Most web applications will not need to create answers using the
	"pranswer" type. The preferred handling for a web application would		"pranswer" type. The preferred handling for a web application would
	be to create and send an "inactive" answer more or less immediately		be to create and send an "inactive" answer more or less immediately
	after receiving the offer, instead of waiting for a human user to		after receiving the offer, instead of waiting for a human user to
	physically answer the call. Later, when the human input is received,		physically answer the call. Later, when the human input is received,
	the application can create a new "sendrecv" offer to update the		the application can create a new "sendrecv" offer to update the
	previous offer/answer pair and start the media flow. This approach		previous offer/answer pair and start the media flow. This approach
	is preferred because it minimizes the amount of time that the offer-		is preferred because it minimizes the amount of time that the offer-
	answer exchange is left open, in addition to avoiding media clipping		answer exchange is left open, in addition to avoiding media clipping
	by ensuring the transport is ready to go by the time the call is		by ensuring the transport is ready to go by the time the call is
	phyiscally answered. However, some applications may not be able to		physically answered. However, some applications may not be able to
	do this, particularly ones that are attempting to gateway to other		do this, particularly ones that are attempting to gateway to other
	signaling protocols. In these cases, "pranswer" can still allow the		signaling protocols. In these cases, "pranswer" can still allow the
	application to warm up the transport.		application to warm up the transport.
	Consider a typical web application that will set up a data channel,		Consider a typical web application that will set up a data channel,
	an audio channel, and a video channel. When an endpoint receives an		an audio channel, and a video channel. When an endpoint receives an
	offer with these channels, it could send an answer accepting the data		offer with these channels, it could send an answer accepting the data
	channel for two-way data, and accepting the audio and video tracks as		channel for two-way data, and accepting the audio and video tracks as
	inactive or receive-only. It could then ask the user to accept the		inactive or receive-only. It could then ask the user to accept the
	call, acquire the local media streams, and send a new offer to the		call, acquire the local media streams, and send a new offer to the
	remote side moving the audio and video to be two-way media. By the		remote side moving the audio and video to be two-way media. By the
	time the human has accepted the call and sent the new offer, it is		time the human has accepted the call and sent the new offer, it is
	likely that the ICE and DTLS handshaking for all the channels will		likely that the ICE and DTLS handshaking for all the channels will
	already be set up.		already be set up.
	4.2.3.2. Rollback		4.1.3.2. Rollback
	In certain situations it may be desirable to "undo" a change made to		In certain situations it may be desirable to "undo" a change made to
	setLocalDescription or setRemoteDescription. Consider a case where a		setLocalDescription or setRemoteDescription. Consider a case where a
	call is ongoing, and one side wants to change some of the session		call is ongoing, and one side wants to change some of the session
	parameters; that side generates an updated offer and then calls		parameters; that side generates an updated offer and then calls
	setLocalDescription. However, the remote side, either before or		setLocalDescription. However, the remote side, either before or
	after setRemoteDescription, decides it does not want to accept the		after setRemoteDescription, decides it does not want to accept the
	new parameters, and sends a reject message back to the offerer. Now,		new parameters, and sends a reject message back to the offerer. Now,
	the offerer, and possibly the answerer as well, need to return to a		the offerer, and possibly the answerer as well, need to return to a
	stable state and the previous local/remote description. To support		stable state and the previous local/remote description. To support
	skipping to change at page 19, line 17		skipping to change at page 17, line 41
	candidates that were allocated by the new local description are		candidates that were allocated by the new local description are
	discarded; any media that is received will be processed according to		discarded; any media that is received will be processed according to
	the previous session description.		the previous session description.
	A rollback is performed by supplying a session description of type		A rollback is performed by supplying a session description of type
	"rollback" to either setLocalDescription or setRemoteDescription,		"rollback" to either setLocalDescription or setRemoteDescription,
	depending on which needs to be rolled back (i.e. if the new offer was		depending on which needs to be rolled back (i.e. if the new offer was
	supplied to setLocalDescription, the rollback should be done on		supplied to setLocalDescription, the rollback should be done on
	setLocalDescription as well.)		setLocalDescription as well.)
	4.2.4. setLocalDescription		4.1.4. setLocalDescription
	The setLocalDescription method instructs the PeerConnection to apply		The setLocalDescription method instructs the PeerConnection to apply
	the supplied SDP blob as its local configuration. The type field		the supplied SDP blob as its local configuration. The type field
	indicates whether the blob should be processed as an offer,		indicates whether the blob should be processed as an offer,
	provisional answer, or final answer; offers and answers are checked		provisional answer, or final answer; offers and answers are checked
	differently, using the various rules that exist for each SDP line.		differently, using the various rules that exist for each SDP line.
	This API changes the local media state; among other things, it sets		This API changes the local media state; among other things, it sets
	up local resources for receiving and decoding media. In order to		up local resources for receiving and decoding media. In order to
	successfully handle scenarios where the application wants to offer to		successfully handle scenarios where the application wants to offer to
	skipping to change at page 19, line 46		skipping to change at page 18, line 21
	local description is supplied, and the number of transports currently		local description is supplied, and the number of transports currently
	in use does not match the number of transports needed by the local		in use does not match the number of transports needed by the local
	description, the PeerConnection will create transports as needed and		description, the PeerConnection will create transports as needed and
	begin gathering candidates for them.		begin gathering candidates for them.
	If setRemoteDescription was previous called with an offer, and		If setRemoteDescription was previous called with an offer, and
	setLocalDescription is called with an answer (provisional or final),		setLocalDescription is called with an answer (provisional or final),
	and the media directions are compatible, and media are available to		and the media directions are compatible, and media are available to
	send, this will result in the starting of media transmission.		send, this will result in the starting of media transmission.
	4.2.5. setRemoteDescription		4.1.5. setRemoteDescription
	The setRemoteDescription method instructs the PeerConnection to apply		The setRemoteDescription method instructs the PeerConnection to apply
	the supplied SDP blob as the desired remote configuration. As in		the supplied SDP blob as the desired remote configuration. As in
	setLocalDescription, the type field of the indicates how the blob		setLocalDescription, the type field of the indicates how the blob
	should be processed.		should be processed.
	This API changes the local media state; among other things, it sets		This API changes the local media state; among other things, it sets
	up local resources for sending and encoding media.		up local resources for sending and encoding media.
	If setRemoteDescription was previously called with an offer, and		If setRemoteDescription was previously called with an offer, and
	setLocalDescription is called with an answer (provisional or final),		setLocalDescription is called with an answer (provisional or final),
	and the media directions are compatible, and media are available to		and the media directions are compatible, and media are available to
	send, this will result in the starting of media transmission.		send, this will result in the starting of media transmission.
	4.2.6. localDescription		4.1.6. localDescription
	The localDescription method returns a copy of the current local		The localDescription method returns a copy of the current local
	configuration, i.e. what was most recently passed to		configuration, i.e. what was most recently passed to
	setLocalDescription, plus any local candidates that have been		setLocalDescription, plus any local candidates that have been
	generated by the ICE Agent.		generated by the ICE Agent.
	TODO: Do we need to expose accessors for both the current and		TODO: Do we need to expose accessors for both the current and
	proposed local description?		proposed local description?
	A null object will be returned if the local description has not yet		A null object will be returned if the local description has not yet
	been established, or if the PeerConnection has been closed.		been established, or if the PeerConnection has been closed.
	4.2.7. remoteDescription		4.1.7. remoteDescription
	The remoteDescription method returns a copy of the current remote		The remoteDescription method returns a copy of the current remote
	configuration, i.e. what was most recently passed to		configuration, i.e. what was most recently passed to
	setRemoteDescription, plus any remote candidates that have been		setRemoteDescription, plus any remote candidates that have been
	supplied via processIceMessage.		supplied via processIceMessage.
	TODO: Do we need to expose accessors for both the current and		TODO: Do we need to expose accessors for both the current and
	proposed remote description?		proposed remote description?
	A null object will be returned if the remote description has not yet		A null object will be returned if the remote description has not yet
	been established, or if the PeerConnection has been closed.		been established, or if the PeerConnection has been closed.
	4.2.8. updateIce		4.1.8. updateIce
	The updateIce method allows the configuration of the ICE Agent to be		The updateIce method allows the configuration of the ICE Agent to be
	changed during the session, primarily for changing which types of		changed during the session, primarily for changing which types of
	local candidates are provided to the application and used for		local candidates are provided to the application and used for
	connectivity checks. A callee may initially configure the ICE Agent		connectivity checks. A callee may initially configure the ICE Agent
	to use only relay candidates, to avoid leaking location information,		to use only relay candidates, to avoid leaking location information,
	but update this configuration to use all candidates once the call is		but update this configuration to use all candidates once the call is
	accepted.		accepted.
	Regardless of the configuration, the gathering process collects all		Regardless of the configuration, the gathering process collects all
	available candidates, but excluded candidates will not be surfaced in		available candidates, but excluded candidates will not be surfaced in
	onicecandidate callback or used for connectivity checks.		onicecandidate callback or used for connectivity checks.
	This call may result in a change to the state of the ICE Agent, and		This call may result in a change to the state of the ICE Agent, and
	may result in a change to media state if it results in connectivity		may result in a change to media state if it results in connectivity
	being established.		being established.
	4.2.9. addIceCandidate		4.1.9. addIceCandidate
	The addIceCandidate method provides a remote candidate to the ICE		The addIceCandidate method provides a remote candidate to the ICE
	Agent, which, if parsed successfully, will be added to the remote		Agent, which, if parsed successfully, will be added to the remote
	description according to the rules defined for Trickle ICE.		description according to the rules defined for Trickle ICE.
	Connectivity checks will be sent to the new candidate.		Connectivity checks will be sent to the new candidate.
	This call will result in a change to the state of the ICE Agent, and		This call will result in a change to the state of the ICE Agent, and
	may result in a change to media state if it results in connectivity		may result in a change to media state if it results in connectivity
	being established.		being established.
	5. SDP Interaction Procedures		5. SDP Interaction Procedures
	This section describes the specific procedures to be followed when		This section describes the specific procedures to be followed when
	creating and parsing SDP objects. [Work In Progress]		creating and parsing SDP objects.
	5.1. Constructing an Offer		5.1. SDP Requirements Overview
			The key specifications that govern creation and processing of offers
			and answers are listed below. This list is derived from
			[I-D.ietf-rtcweb-rtp-usage].
	5.2. Generating an Answer		R-1 [RFC4566] is the base SDP specification and MUST be
			implemented.
	5.3. Parsing an Offer		R-2 The [RFC5888] grouping framework MUST be implemented for
			signaling grouping information, and MUST be used to identify m=
			lines via the a=mid attribute.
	5.4. Parsing an Answer		R-3 [RFC5124] MUST be supported for signaling RTP/SAVPF RTP
			profile.
	5.5. Applying a Local Description		R-4 [RFC4585] MUST be implemented to signal RTCP based feedback.
	5.6. Applying a Remote Description		R-5 [RFC5245] MUST be implemented for signaling the ICE candidate
			lines corresponding to each media stream.
			R-6 [RFC5761] MUST be implemented to signal multiplexing of RTP and
			RTCP.
			R-7 The SDP atributes of "sendonly", "recvonly", "inactive", and
			"sendrecv" from [RFC4566] MUST be implemented to signal
			information about media direction.
			R-8 [RFC5576] MUST be implemented to signal RTP SSRC values.
			R-9 [RFC5763] MUST be implemented to signal DTLS certificate
			fingerprints.
			R-10 [RFC5506] MAY be implemented to signal Reduced-Size RTCP
			messages.
			R-11 [RFC3556] with bandwidth modifiers MAY be supported for
			specifying RTCP bandwidth as a fraction of the media bandwidth,
			RTCP fraction allocated to the senders and setting maximum media
			bit-rate boundaries.
			R-12 [RFC4568] MUST NOT be implemented to signal SDES SRTP keying
			information.
			R-13 A [I-D.ietf-mmusic-msid] MUST be supported, in order to signal
			associations between RTP objects and W3C MediaStreams and
			MediaStreamTracks in a standard way.
			R-14 The bundle mechanism in
			[I-D.ietf-mmusic-sdp-bundle-negotiation] MUST be supported to
			signal the use or multiplexing RTP somethings on a single UDP
			port, in order to avoid excessive use of port number resources.
			As required by [RFC4566] Section 5.13 JSEP implementations MUST
			ignore unknown attributes (a=) lines.
			Example SDP for RTCWeb call flows can be found in
			[I-D.nandakumar-rtcweb-sdp]. [TODO: since we are starting to specify
			how to handle SDP in this document, should these call flows be merged
			into this document, or this link moved to the examples section?]
			5.2. Constructing an Offer
			When createOffer is called, a new SDP description must be created
			that includes the functionality specified in
			[I-D.ietf-rtcweb-rtp-usage]. The exact details of this process are
			explained below.
			5.2.1. Initial Offers
			When createOffer is called for the first time, the result is known as
			the initial offer.
			The first step in generating an initial offer is to generate session-
			level attributes, as specified in [RFC4566], Section 5.
			Specifically:
			o The first SDP line MUST be "v=0", as specified in [RFC4566],
			Section 5.1
			o The second SDP line MUST be an "o=" line, as specified in
			[RFC4566], Section 5.2. The value of the <username> field SHOULD
			be "-". The value of the <sess-id> field SHOULD be a
			cryptographically random number. To ensure uniqueness, this
			number SHOULD be at least 64 bits long. The value of the <sess-
			version> field SHOULD be zero. The value of the <nettype>
			<addrtype> <unicast-address> tuple SHOULD be set to a non-
			meaningful address, such as IN IP4 0.0.0.0, to prevent leaking the
			local address in this field. As mentioned in [RFC4566], the
			entire o= line needs to be unique, but selecting a random number
			for <sess-id> is sufficient to accomplish this.
			o The third SDP line MUST be a "s=" line, as specified in [RFC4566],
			Section 5.3; a single space SHOULD be used as the session name,
			e.g. "s= "
			o Session Information ("i="), URI ("u="), Email Address ("e="),
			Phone Number ("p="), Bandwidth ("b="), Repeat Times ("r="), and
			Time Zones ("z=") lines are not useful in this context and SHOULD
			NOT be included.
			o Encryption Keys ("k=") lines do not provide sufficient security
			and MUST NOT be included.
			o A "t=" line MUST be added, as specified in [RFC4566], Section 5.9;
			both <start-time> and <stop-time> SHOULD be set to zero, e.g. "t=0
			0".
			The next step is to generate m= sections for each MediaStreamTrack
			that has been added to the PeerConnection via the addStream method.
			Note that this method takes a MediaStream, which can contain multiple
			MediaStreamTracks, and therefore multiple m= sections can be
			generated even if addStream is only called once.
			Each m= section should be generated as specified in [RFC4566],
			Section 5.14. The <proto> field MUST be set to "RTP/SAVPF". If a m=
			section is not being bundled into another m= section, it MUST
			generate a unique set of ICE credentials and gather its own set of
			candidates. Otherwise, it MUST use the same ICE credentials and
			candidates that were used in the m= section that it is being bundled
			into. For DTLS, all m= sections MUST use the same certificate [OPEN
			ISSUE: how this is configured] and will therefore have the same
			fingerprint values.
			Each m= section MUST include the following:
			o An "a=mid" line, as specified in [RFC5888], Section 4.
			o An "a=msid" line, as specified in [I-D.ietf-mmusic-msid],
			Section 2.
			o [OPEN ISSUE: Use of App Token versus stream-correlator ]
			o An "a=sendrecv" line, as specified in [RFC3264], Section 5.1.
			o For each supported codec, "a=rtpmap" and "a=fmtp" lines, as
			specified in [RFC4566], Section 6. For audio, the codecs
			specified in [I-D.ietf-rtcweb-audio], Section 3, MUST be be
			supported.
			o For each primary codec where RTP retransmission should be used, a
			corresponding "a=rtpmap" line indicating "rtx" with the clock rate
			of the primary codec and an "a=fmtp" line that references the
			payload type fo the primary codec, as specified in [RFC4588],
			Section 8.1.
			o For each supported FEC mechanism, a corresponding "a=rtpmap" line
			indicating the desired FEC codec.
			o "a=ice-ufrag" and "a=ice-passwd" lines, as specified in [RFC5245],
			Section 15.4.
			o An "a=ice-options" line, with the "trickle" option, as specified
			in [I-D.ivov-mmusic-trickle-ice], Section 4.
			o For each candidate that has been gathered during the most recent
			gathering phase, an "a=candidate" line, as specified in [RFC5245],
			Section 4.3., paragraph 3.
			o For the current default candidate, a "c=" line, as specific in
			[RFC5245], Section 4.3., paragraph 6. [OPEN ISSUE, pending
			resolution in mmusic: If no candidates have yet been gathered yet,
			the default candidate should be set to the null value defined in
			[I-D.ivov-mmusic-trickle-ice], Section 5.1.]
			o An "a=fingerprint" line, as specified in [RFC4572], Section 5.
			Use of the SHA-256 algorithm for the fingerprint is REQUIRED; if
			the browser also supports stronger hashes, additional
			"a=fingerprint" lines with these hashes MAY also be added.
			o An "a=setup" line, as specified in [RFC4145], Section 4, and
			clarified for use in DTLS-SRTP scenarios in [RFC5763], Section 5.
			The role value in the offer MUST be "actpass".
			o An "a=rtcp-mux" line, as specified in [RFC5761], Section 5.1.1.
			o An "a=rtcp-rsize" line, as specified in [RFC5506], Section 5.
			o For each supported RTP header extension, an "a=extmap" line, as
			specified in [RFC5285], Section 5. The list of header extensions
			that SHOULD/MUST be supported is specified in
			[I-D.ietf-rtcweb-rtp-usage], Section 5.2. Any header extensions
			that require encryption MUST be specified as indicated in
			[RFC6904], Section 4.
			o For each supported RTCP feedback mechanism, an "a=rtcp-fb"
			mechanism, as specified in [RFC4585], Section 4.2. The list of
			RTCP feedback mechanisms that SHOULD/MUST be supported is
			specified in [I-D.ietf-rtcweb-rtp-usage], Section 5.1.
			o An "a=ssrc" line, as specified in [RFC5576], Section 4.1,
			indicating the SSRC to be used for sending media.
			o If RTX is supported for this media type, another "a=ssrc" line
			with the RTX SSRC, and an "a=ssrc-group" line, as specified in
			[RFC5576], section 4.2, with semantics set to "FID" and including
			the primary and RTX SSRCs.
			o If FEC is supported for this media type, another "a=ssrc" line
			with the FEC SSRC, and an "a=ssrc-group" line, as specified in
			[RFC5576], section 4.2, with semantics set to "FEC" and including
			the primary and FEC SSRCs.
			o [OPEN ISSUE: Handling of a=imageattr]
			o [TODO: bundle-only]
			Lastly, if a data channel has been created, a m= section MUST be
			generated for data. The <media> field MUST be set to "application"
			and the <proto> field MUST be set to "DTLS/SCTP", as specified in
			[I-D.ietf-mmusic-sctp-sdp], Section 3. The "a=mid", "a=ice-ufrag",
			"a=ice-passwd", "a=ice-options", "a=candidate", "a=fingerprint", and
			"a=setup" lines MUST be included as mentioned above. [OPEN ISSUE:
			additional SCTP-specific stuff to be included, as indicated in
			[I-D.jesup-rtcweb-data-protocol] (currently none)]
			Once all m= sections have been generated, a session-level "a=group"
			attribute MUST be added as specified in [RFC5888]. This attribute
			MUST have semantics "BUNDLE", and identify the m= sections to be
			bundled. [OPEN ISSUE: Need to determine exactly how this decision is
			made.]
			Attributes that are common between all m= sections MAY be moved to
			session-level, if desired.
			Attributes other than the ones specified above MAY be included,
			except for the following attributes which are specifically
			incompatible with the requirements of [I-D.ietf-rtcweb-rtp-usage],
			and MUST NOT be included:
			o "a=crypto"
			o "a=key-mgmt"
			o "a=ice-lite"
			Note that when BUNDLE is used, any additional attributes that are
			added MUST follow the advice in
			[I-D.nandakumar-mmusic-sdp-mux-attributes] on how those attributes
			interact with BUNDLE.
			5.2.2. Subsequent Offers
			When createOffer is called a second (or later) time, the processing
			is different, depending on the current signaling state.
			If the initial offer was not applied using setLocalDescription,
			meaning the PeerConnection is still in the "stable" state, the steps
			for generating an initial offer should be followed, with this
			exception:
			o The "o=" line MUST stay the same.
			If the initial offer was applied using setLocalDescription, but an
			answer from the remote side has not yet been applied, meaning the
			PeerConnection is still in the "local-offer" state, the steps for
			generating an initial offer should be followed, with these
			exceptions:
			o The "o=" line MUST stay the same, except for the <session-version>
			field, which MUST increase by 1 from the previously applied local
			description.
			o The "s=" and "t=" lines MUST stay the same.
			o Each "a=mid" line MUST stay the same.
			o Each "a=ice-ufrag" and "a=ice-pwd" line MUST stay the same.
			o For MediaStreamTracks that are still present, the "a=msid",
			"a=ssrc", and "a=ssrc-group" lines MUST stay the same.
			o If any MediaStreamTracks have been removed, either through the
			removeStream method or by removing them from an added MediaStream,
			their m= sections MUST be marked as recvonly by changing the value
			of the [RFC3264] directional attribute to "a=recvonly". The
			"a=msid", "a=ssrc", and "a=ssrc-group" lines MUST be removed from
			the associated m= sections.
			If the initial offer was applied using setLocalDescription, and an
			answer from the remote side has been applied using
			setRemoteDescription, meaning the PeerConnection is in the "remote-
			pranswer" or "stable" states, an offer is generated based on the
			negotiated session descriptions by following the steps mentioned for
			the "local-offer" state above, along with these exceptions: [OPEN
			ISSUE: should this be permitted in the remote-pranswer state?]
			o If a m= section was rejected, i.e. has had its port set to zero in
			either the local or remote description, it MUST remain rejected
			and have a zero port in the new offer, as indicated in RFC3264,
			Section 5.1.
			o If a m= section exists in the current local description, but has
			its state set to inactive or recvonly, and a new MediaStreamTrack
			is added, the previously existing m= section MUST be recycled
			instead of creating a new m= section. [OPEN ISSUE: Nail down
			exactly what this means. Should the codecs remain the same?
			(No.) Should ICE restart? (No.) Can the "a=mid" attribute be
			changed? (Yes?)]
			o If a m= section exists in the current local description, but does
			not have an associated MediaStreamTrack (i.e. it is inactive or
			recvonly), a corresponding m= section MUST be generated in the new
			offer, but without "a=msid", "a=ssrc", or "a=ssrc-group"
			attributes, and the appropriate directional attribute must be
			specified.
			In addition, for each previously existing, non-rejected m= section in
			the new offer, the following adjustments are made based on the
			contents of the corresponding m= section in the current remote
			description:
			o The m= line and corresponding "a=rtpmap" and "a=fmtp" lines MUST
			only include codecs present in the remote description.
			o The RTP header extensions MUST only include those that are present
			in the remote description.
			o The RTCP feedback extensions MUST only include those that are
			present in the remote description.
			o The "a=rtcp-mux" line MUST only be added if present in the remote
			description.
			o The "a=rtcp-rsize" line MUST only be added if present in the
			remote description.
			5.2.3. Constraints Handling
			The createOffer method takes as a parameter a MediaConstraints
			object. Special processing is performed when generating a SDP
			description if the following constraints are present.
			5.2.3.1. OfferToReceiveAudio
			If the "OfferToReceiveAudio" constraint is specified, with a value of
			"true", the offer MUST include a non-rejected m= section with media
			type "audio", even if no audio MediaStreamTrack has been added to the
			PeerConnection. This allows the offerer to receive audio even when
			not sending it; accordingly, the directional attribute on the audio
			m= section MUST be set to recvonly. If this constraint is specified
			when an audio MediaStreamTrack has already been added to the
			PeerConnection, or a non-rejected m= section with media type "audio"
			previously existed, it has no effect.
			5.2.3.2. OfferToReceiveVideo
			If the "OfferToReceiveAudio" constraint is specified, with a value of
			"true", the offer MUST include a m= section with media type "video",
			even if no video MediaStreamTrack has been added to the
			PeerConnection. This allows the offerer to receive video even when
			not sending it; accordingly, the directional attribute on the video
			m= section MUST be set to recvonly. If this constraint is specified
			when an video MediaStreamTrack has already been added to the
			PeerConnection, or a non-rejected m= section with media type "video"
			previously existed, it has no effect.
			5.2.3.3. VoiceActivityDetection
			If the "VoiceActivityDetection" constraint is specified, with a value
			of "true", the offer MUST indicate support for silence suppression by
			including comfort noise ("CN") codecs for each supported clock rate,
			as specified in [RFC3389], Section 5.1. [OPEN issue: should this do
			anything in signaling, or should it just control built-in DTX modes
			in audio codecs? Opus has built-in DTX, but G.711 does not.]
			5.2.3.4. IceRestart
			If the "IceRestart" constraint is specified, with a value of "true",
			the offer MUST indicate an ICE restart by generating new ICE ufrag
			and pwd attributes, as specified in RFC5245, Section 9.1.1.1. If
			this constraint is specified on an initial offer, it has no effect
			(since a new ICE ufrag and pwd are already generated).
			5.3. Generating an Answer
			When createAnswer is called, a new SDP description must be created
			that is compatible with the supplied remote description as well as
			the requirements specified in [I-D.ietf-rtcweb-rtp-usage]. The exact
			details of this process are explained below.
			5.3.1. Initial Answers
			When createAnswer is called for the first time after a remote
			description has been provided, the result is known as the initial
			answer. If no remote description has been installed, an answer
			cannot be generated, and an error MUST be returned.
			Note that the remote description SDP may not have been created by a
			WebRTC endpoint and may not conform to all the requirements listed in
			Section 5.2. For many cases, this is not a problem. However, if any
			mandatory SDP attributes are missing, or functionality listed as
			mandatory-to-use is not present (e.g. ICE, DTLS) [TODO: find
			reference for this], this MUST be treated as an error. [OPEN ISSUE:
			Should this cause setRemoteDescription to fail, or should this cause
			createAnswer to reject those particular m= sections?]
			The first step in generating an initial answer is to generate
			session-level attributes. The process here is identical to that
			indicated in the Initial Offers section above, with the addition that
			The next step is to generate m= sections for each m= section that is
			present in the remote offer, as specified in [RFC3264], Section 6.
			For the purposes of this discussion, any session-level attributes in
			the offer that are also valid as media-level attributes SHALL be
			considered to be present in each m= section.
			If any of the offered m= sections have been rejected, by stopping the
			associated remote MediaStreamTrack, the corresponding m= section in
			the answer MUST be marked as rejected by setting the port in the m=
			line to zero, as indicated in [RFC3264], Section 6., and processing
			continues with the next m= section.
			For each non-rejected m= section of a given media type, if there is a
			local MediaStreamTrack of the specified type which has been added to
			the PeerConnection via addStream and not yet associated with a m=
			section, the MediaStreamTrack is associated with the m= section at
			this time. If there are more m= sections of a certain type than
			MediaStreamTracks, some m= sections will not have an associated
			MediaStreamTrack. If there are more MediaStreamTracks of a certain
			type than m= sections, only the first N MediaStreamTracks will be
			able to be associated in the constructed answer. The remainder will
			need to be associated in a subsequent offer.
			Each m= section should then generated as specified in [RFC3264],
			Section 6.1. The <proto> field MUST be set to "RTP/SAVPF". If the
			offer supports BUNDLE, all m= sections to be BUNDLEd must use the
			same ICE credentials and candidates; all m= sections not being
			BUNDLEd must use unique ICE credentials and candidates. Each m=
			section MUST include the following:
			o If present in the offer, an "a=mid" line, as specified in
			[RFC5888], Section 9.1. The "mid" value MUST match that specified
			in the offer.
			o If a local MediaStreamTrack has been associated, an "a=msid" line,
			as specified in [I-D.ietf-mmusic-msid], Section 2.
			o [OPEN ISSUE: Use of App Token versus stream-correlator ]
			o If a local MediaStreamTrack has been associated, an "a=sendrecv"
			line, as specified in [RFC3264], Section 6.1. If no local
			MediaStreamTrack has been associated, an "a=recvonly" line.
			[TODO: handle non-sendrecv offered m= sections]
			o For each supported codec that is present in the offer, "a=rtpmap"
			and "a=fmtp" lines, as specified in [RFC4566], Section 6, and
			[RFC3264], Section 6.1. For audio, the codecs specified in
			[I-D.ietf-rtcweb-audio], Section 3, MUST be be supported. Note
			that for simplicity, the answerer MAY use different payload types
			for codecs than the offerer, as it is not prohibited by
			Section 6.1.
			o If "rtx" is present in the offer, for each primary codec where RTP
			retransmission should be used, a corresponding "a=rtpmap" line
			indicating "rtx" with the clock rate of the primary codec and an
			"a=fmtp" line that references the payload type fo the primary
			codec, as specified in [RFC4588], Section 8.1.
			o For each supported FEC mechanism that is present in the offer, a
			corresponding "a=rtpmap" line indicating the desired FEC codec.
			o "a=ice-ufrag" and "a=ice-passwd" lines, as specified in [RFC5245],
			Section 15.4.
			o If the "trickle" ICE option is present in the offer, an "a=ice-
			options" line, with the "trickle" option, as specified in
			[I-D.ivov-mmusic-trickle-ice], Section 4.
			o For each candidate that has been gathered during the most recent
			gathering phase, an "a=candidate" line, as specified in [RFC5245],
			Section 4.3., paragraph 3.
			o For the current default candidate, a "c=" line, as specific in
			[RFC5245], Section 4.3., paragraph 6. [OPEN ISSUE, pending
			resolution in mmusic: If no candidates have yet been gathered yet,
			the default candidate should be set to the null value defined in
			[I-D.ivov-mmusic-trickle-ice], Section 5.1.]
			o An "a=fingerprint" line, as specified in [RFC4572], Section 5.
			Use of the SHA-256 algorithm for the fingerprint is REQUIRED; if
			the browser also supports stronger hashes, additional
			"a=fingerprint" lines with these hashes MAY also be added.
			o An "a=setup" line, as specified in [RFC4145], Section 4, and
			clarified for use in DTLS-SRTP scenarios in [RFC5763], Section 5.
			The role value in the answer MUST be "active" or "passive"; the
			"active" role is RECOMMENDED.
			o If present in the offer, an "a=rtcp-mux" line, as specified in
			[RFC5761], Section 5.1.1.
			o If present in the offer, an "a=rtcp-rsize" line, as specified in
			[RFC5506], Section 5.
			o For each supported RTP header extension that is present in the
			offer, an "a=extmap" line, as specified in [RFC5285], Section 5.
			The list of header extensions that SHOULD/MUST be supported is
			specified in [I-D.ietf-rtcweb-rtp-usage], Section 5.2. Any header
			extensions that require encryption MUST be specified as indicated
			in [RFC6904], Section 4.
			o For each supported RTCP feedback mechanism that is present in the
			offer, an "a=rtcp-fb" mechanism, as specified in [RFC4585],
			Section 4.2. The list of RTCP feedback mechanisms that SHOULD/
			MUST be supported is specified in [I-D.ietf-rtcweb-rtp-usage],
			Section 5.1.
			o If a local MediaStreamTrack has been associated, an "a=ssrc" line,
			as specified in [RFC5576], Section 4.1, indicating the SSRC to be
			used for sending media.
			o If a local MediaStreamTrack has been associated, and RTX has been
			negotiated for this m= section, another "a=ssrc" line with the RTX
			SSRC, and an "a=ssrc-group" line, as specified in [RFC5576],
			section 4.2, with semantics set to "FID" and including the primary
			and RTX SSRCs.
			o If a local MediaStreamTrack has been associated, and FEC has been
			negotiated for this m= section, another "a=ssrc" line with the FEC
			SSRC, and an "a=ssrc-group" line, as specified in [RFC5576],
			section 4.2, with semantics set to "FEC" and including the primary
			and FEC SSRCs.
			o [OPEN ISSUE: Handling of a=imageattr]
			o [TODO: bundle-only]
			If a data channel m= section has been offered, a m= section MUST also
			be generated for data. The <media> field MUST be set to
			"application" and the <proto> field MUST be set to "DTLS/SCTP", as
			specified in [I-D.ietf-mmusic-sctp-sdp], Section 3. The "a=mid", "a
			=ice-ufrag", "a=ice-passwd", "a=ice-options", "a=candidate",
			"a=fingerprint", and "a=setup" lines MUST be included as mentioned
			above. [OPEN ISSUE: additional SCTP-specific stuff to be included,
			as indicated in [I-D.jesup-rtcweb-data-protocol] (currently none)]
			[TODO: processing of BUNDLE group]
			Attributes that are common between all m= sections MAY be moved to
			session-level, if desired.
			The attributes prohibited in creation of offers are also prohibited
			in the creation of answers.
			5.3.2. Subsequent Answers
			5.3.3. Constraints Handling
			5.4. Parsing an Offer
			5.5. Parsing an Answer
			5.6. Applying a Local Description
			5.7. Applying a Remote Description
	6. Configurable SDP Parameters		6. Configurable SDP Parameters
	Note: This section is still very early and is likely to		Note: This section is still very early and is likely to significantly
	significantly change as we get a better understanding of a) the use		change as we get a better understanding of a) the use cases for this
	cases for this b) the implications at the protocol level c) feedback		b) the implications at the protocol level c) feedback from
	from implementors on what they can do.		implementors on what they can do.
	The following elements of the SDP media description MUST NOT be		The following elements of the SDP media description MUST NOT be
	changed between the createOffer and the setLocalDescription, since		changed between the createOffer and the setLocalDescription, since
	they reflect transport attributes that are solely under browser		they reflect transport attributes that are solely under browser
	control, and the browser MUST NOT honor an attempt to change them:		control, and the browser MUST NOT honor an attempt to change them:
	o The number, type and port number of m-lines.		o The number, type and port number of m-lines.
	o The generated ICE credentials (a=ice-ufrag and a=ice-pwd).		o The generated ICE credentials (a=ice-ufrag and a=ice-pwd).
	o The set of ICE candidates and their parameters (a=candidate).		o The set of ICE candidates and their parameters (a=candidate).
	The following modifications, if done by the browser to a description		The following modifications, if done by the browser to a description
	between createOffer/createAnswer and the setLocalDescription, MUST be		between createOffer/createAnswer and the setLocalDescription, MUST be
	honored by the browser:		honored by the browser:
	o Remove or reorder codecs (m=)		o Remove or reorder codecs (m=)
	The following parameters may be controlled by constraints passed into		The following parameters may be controlled by constraints passed into
	createOffer/createAnswer. As an open issue, these changes may also		createOffer/createAnswer. As an open issue, these changes may also
	skipping to change at page 22, line 23		skipping to change at page 32, line 19
	o Remove or reorder codecs (m=)		o Remove or reorder codecs (m=)
	The following parameters may be controlled by constraints passed into		The following parameters may be controlled by constraints passed into
	createOffer/createAnswer. As an open issue, these changes may also		createOffer/createAnswer. As an open issue, these changes may also
	be be performed by manipulating the SDP returned from createOffer/		be be performed by manipulating the SDP returned from createOffer/
	createAnswer, as indicated above, as long as the capabilities of the		createAnswer, as indicated above, as long as the capabilities of the
	endpoint are not exceeded (e.g. asking for a resolution greater than		endpoint are not exceeded (e.g. asking for a resolution greater than
	what the endpoint can encode):		what the endpoint can encode):
	o disable BUNDLE (a=group)		o disable BUNDLE (a=group)
	o disable RTCP mux (a=rtcp-mux)		o disable RTCP mux (a=rtcp-mux)
	o change send resolution or framerate
	o change desired recv resolution or framerate		o change send resolution or frame rate
	o change maximum total bandwidth (b=) [OPEN ISSUE: need to clarify
	if this is CT or AS - see section 5.8 of RFC4566]		o change desired recv resolution or frame rate
			o change maximum total bandwidth (b=) [OPEN ISSUE: need to clarify
			if this is CT or AS - see section 5.8 of [RFC4566]]
	o remove desired AVPF mechanisms (a=rtcp-fb)		o remove desired AVPF mechanisms (a=rtcp-fb)
	o remove RTP header extensions (a=extmap)		o remove RTP header extensions (a=extmap)
	o change media send/recv state (a=sendonly/recvonly/inactive)		o change media send/recv state (a=sendonly/recvonly/inactive)
	For example, an application could implement call hold by adding an		For example, an application could implement call hold by adding an
	a=inactive attribute to its local description, and then applying and		a=inactive attribute to its local description, and then applying and
	signaling that description.		signaling that description.
	The application can also modify the SDP to reduce the capabilities in		The application can also modify the SDP to reduce the capabilities in
	the offer it sends to the far side in any way the application sees		the offer it sends to the far side in any way the application sees
	fit, as long as it is a valid SDP offer and specifies a subset of		fit, as long as it is a valid SDP offer and specifies a subset of
	what the browser is expecting to do.		what the browser is expecting to do.
	skipping to change at page 23, line 4		skipping to change at page 33, line 8
	the other party, and all incoming SDP will be processed by the		the other party, and all incoming SDP will be processed by the
	browser to the extent of its capabilities. It is an error to assume		browser to the extent of its capabilities. It is an error to assume
	that all SDP is well-formed; however, one should be able to assume		that all SDP is well-formed; however, one should be able to assume
	that any implementation of this specification will be able to		that any implementation of this specification will be able to
	process, as a remote offer or answer, unmodified SDP coming from any		process, as a remote offer or answer, unmodified SDP coming from any
	other implementation of this specification.		other implementation of this specification.
	7. Security Considerations		7. Security Considerations
	The intent of the WebRTC protocol suite is to provide an environment		The intent of the WebRTC protocol suite is to provide an environment
	that is securable by default: all media is encrypted, keys are		that is securable by default: all media is encrypted, keys are
	exchanged in a secure fashion, and the Javascript API includes		exchanged in a secure fashion, and the Javascript API includes
	functions that can be used to verify the identity of communication		functions that can be used to verify the identity of communication
	partners.		partners.
	8. IANA Considerations		8. IANA Considerations
	This document requires no actions from IANA.		This document requires no actions from IANA.
	9. Acknowledgements		9. Acknowledgements
	skipping to change at page 23, line 27		skipping to change at page 33, line 31
	Neil Stratford, Eric Rescorla, Anant Narayanan, Andrew Hutton,		Neil Stratford, Eric Rescorla, Anant Narayanan, Andrew Hutton,
	Richard Ejzak, and Adam Bergkvist all provided valuable feedback on		Richard Ejzak, and Adam Bergkvist all provided valuable feedback on
	this proposal. Matthew Kaufman provided the observation that keeping		this proposal. Matthew Kaufman provided the observation that keeping
	state out of the browser allows a call to continue even if the page		state out of the browser allows a call to continue even if the page
	is reloaded.		is reloaded.
	10. References		10. References
	10.1. Normative References		10.1. Normative References
	[I-D.rescorla-mmusic-ice-trickle]		[I-D.ietf-mmusic-msid]
	Rescorla, E., Uberti, J., and E. Ivov, "Trickle ICE:		Alvestrand, H., "Cross Session Stream Identification in
	Incremental Provisioning of Candidates for the Interactive		the Session Description Protocol", draft-ietf-mmusic-
	Connectivity Establishment (ICE) Protocol",		msid-01 (work in progress), August 2013.
	draft-rescorla-mmusic-ice-trickle-00 (work in progress),
	October 2012.		[I-D.ietf-mmusic-sctp-sdp]
			Loreto, S. and G. Camarillo, "Stream Control Transmission
			Protocol (SCTP)-Based Media Transport in the Session
			Description Protocol (SDP)", draft-ietf-mmusic-sctp-sdp-04
			(work in progress), June 2013.
			[I-D.ietf-mmusic-sdp-bundle-negotiation]
			Holmberg, C., Alvestrand, H., and C. Jennings,
			"Multiplexing Negotiation Using Session Description
			Protocol (SDP) Port Numbers", draft-ietf-mmusic-sdp-
			bundle-negotiation-04 (work in progress), June 2013.
			[I-D.ietf-rtcweb-audio]
			Valin, J. and C. Bran, "WebRTC Audio Codec and Processing
			Requirements", draft-ietf-rtcweb-audio-02 (work in
			progress), August 2013.
			[I-D.ietf-rtcweb-rtp-usage]
			Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time
			Communication (WebRTC): Media Transport and Use of RTP",
			draft-ietf-rtcweb-rtp-usage-09 (work in progress),
			September 2013.
			[I-D.nandakumar-mmusic-sdp-mux-attributes]
			Nandakumar, S., "A Framework for SDP Attributes when
			Multiplexing", draft-nandakumar-mmusic-sdp-mux-
			attributes-03 (work in progress), July 2013.
	[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.
	[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.
	[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model		[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
	with Session Description Protocol (SDP)", RFC 3264,		with Session Description Protocol (SDP)", RFC 3264, June
	June 2002.		2002.
	[RFC3388] Camarillo, G., Eriksson, G., Holler, J., and H.
	Schulzrinne, "Grouping of Media Lines in the Session
	Description Protocol (SDP)", RFC 3388, December 2002.
	[RFC3960] Camarillo, G. and H. Schulzrinne, "Early Media and Ringing		[RFC4145] Yon, D. and G. Camarillo, "TCP-Based Media Transport in
	Tone Generation in the Session Initiation Protocol (SIP)",		the Session Description Protocol (SDP)", RFC 4145,
	RFC 3960, December 2004.		September 2005.
	[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session		[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
	Description Protocol", RFC 4566, July 2006.		Description Protocol", RFC 4566, July 2006.
			[RFC4572] Lennox, J., "Connection-Oriented Media Transport over the
			Transport Layer Security (TLS) Protocol in the Session
			Description Protocol (SDP)", RFC 4572, July 2006.
	[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,		[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
	"Extended RTP Profile for Real-time Transport Control		"Extended RTP Profile for Real-time Transport Control
	Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585,		Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July
	July 2006.		2006.
	[RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for		[RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for
	Real-time Transport Control Protocol (RTCP)-Based Feedback		Real-time Transport Control Protocol (RTCP)-Based Feedback
	(RTP/SAVPF)", RFC 5124, February 2008.		(RTP/SAVPF)", RFC 5124, February 2008.
	[RFC5245] Rosenberg, J., "Interactive Connectivity Establishment		[RFC5245] Rosenberg, J., "Interactive Connectivity Establishment
	(ICE): A Protocol for Network Address Translator (NAT)		(ICE): A Protocol for Network Address Translator (NAT)
	Traversal for Offer/Answer Protocols", RFC 5245,		Traversal for Offer/Answer Protocols", RFC 5245, April
	April 2010.		2010.
			[RFC5285] Singer, D. and H. Desineni, "A General Mechanism for RTP
			Header Extensions", RFC 5285, July 2008.
	[RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and		[RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
	Control Packets on a Single Port", RFC 5761, April 2010.		Control Packets on a Single Port", RFC 5761, April 2010.
	[RFC5888] Camarillo, G. and H. Schulzrinne, "The Session Description		[RFC5888] Camarillo, G. and H. Schulzrinne, "The Session Description
	Protocol (SDP) Grouping Framework", RFC 5888, June 2010.		Protocol (SDP) Grouping Framework", RFC 5888, June 2010.
	10.2. Informative References		[RFC6904] Lennox, J., "Encryption of Header Extensions in the Secure
			Real-time Transport Protocol (SRTP)", RFC 6904, April
	[I-D.alvestrand-mmusic-msid]		2013.
	Alvestrand, H., "Cross Session Stream Identification in
	the Session Description Protocol",
	draft-alvestrand-mmusic-msid-01 (work in progress),
	October 2012.
	[I-D.holmberg-mmusic-sdp-bundle-negotiation]		10.2. Informative References
	Holmberg, C. and H. Alvestrand, "Multiplexing Negotiation
	Using Session Description Protocol (SDP) Port Numbers",
	draft-holmberg-mmusic-sdp-bundle-negotiation-00 (work in
	progress), October 2011.
	[I-D.ietf-rtcweb-rtp-usage]		[I-D.ivov-mmusic-trickle-ice]
	Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time		Ivov, E., Rescorla, E., and J. Uberti, "Trickle ICE:
	Communication (WebRTC): Media Transport and Use of RTP",		Incremental Provisioning of Candidates for the Interactive
	draft-ietf-rtcweb-rtp-usage-04 (work in progress),		Connectivity Establishment (ICE) Protocol", draft-ivov-
	July 2012.		mmusic-trickle-ice-01 (work in progress), March 2013.
	[I-D.jennings-rtcweb-signaling]		[I-D.jennings-rtcweb-signaling]
	Jennings, C., Rosenberg, J., and R. Jesup, "RTCWeb Offer/		Jennings, C., Rosenberg, J., and R. Jesup, "RTCWeb Offer/
	Answer Protocol (ROAP)",		Answer Protocol (ROAP)", draft-jennings-rtcweb-
	draft-jennings-rtcweb-signaling-01 (work in progress),		signaling-01 (work in progress), October 2011.
	October 2011.
			[I-D.jesup-rtcweb-data-protocol]
			Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel
			Protocol", draft-jesup-rtcweb-data-protocol-04 (work in
			progress), February 2013.
	[I-D.nandakumar-rtcweb-sdp]		[I-D.nandakumar-rtcweb-sdp]
	Nandakumar, S. and C. Jennings, "SDP for the WebRTC",		Nandakumar, S. and C. Jennings, "SDP for the WebRTC",
	draft-nandakumar-rtcweb-sdp-00 (work in progress),		draft-nandakumar-rtcweb-sdp-02 (work in progress), July
	October 2012.		2013.
			[RFC3389] Zopf, R., "Real-time Transport Protocol (RTP) Payload for
			Comfort Noise (CN)", RFC 3389, September 2002.
	[RFC3556] Casner, S., "Session Description Protocol (SDP) Bandwidth		[RFC3556] Casner, S., "Session Description Protocol (SDP) Bandwidth
	Modifiers for RTP Control Protocol (RTCP) Bandwidth",		Modifiers for RTP Control Protocol (RTCP) Bandwidth", RFC
	RFC 3556, July 2003.		3556, July 2003.
			[RFC3960] Camarillo, G. and H. Schulzrinne, "Early Media and Ringing
			Tone Generation in the Session Initiation Protocol (SIP)",
			RFC 3960, December 2004.
	[RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session		[RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session
	Description Protocol (SDP) Security Descriptions for Media		Description Protocol (SDP) Security Descriptions for Media
	Streams", RFC 4568, July 2006.		Streams", RFC 4568, July 2006.
			[RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
			Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
			July 2006.
	[RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size		[RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size
	Real-Time Transport Control Protocol (RTCP): Opportunities		Real-Time Transport Control Protocol (RTCP): Opportunities
	and Consequences", RFC 5506, April 2009.		and Consequences", RFC 5506, April 2009.
	[RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific		[RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific
	Media Attributes in the Session Description Protocol		Media Attributes in the Session Description Protocol
	(SDP)", RFC 5576, June 2009.		(SDP)", RFC 5576, June 2009.
	[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.
	[W3C.WD-webrtc-20111027]		[W3C.WD-webrtc-20111027]
	Bergkvist, A., Burnett, D., Narayanan, A., and C.		Bergkvist, A., Burnett, D., Narayanan, A., and C.
	Jennings, "WebRTC 1.0: Real-time Communication Between		Jennings, "WebRTC 1.0: Real-time Communication Between
	Browsers", World Wide Web Consortium WD WD-webrtc-		Browsers", World Wide Web Consortium WD WD-
	20111027, October 2011,		webrtc-20111027, October 2011,
	<http://www.w3.org/TR/2011/WD-webrtc-20111027>.		<http://www.w3.org/TR/2011/WD-webrtc-20111027>.
	Appendix A. JSEP Implementation Examples		Appendix A. JSEP Implementation Examples
	A.1. Example API Flows		A.1. Example API Flows
	Below are several sample flows for the new PeerConnection and library		Below are several sample flows for the new PeerConnection and library
	APIs, demonstrating when the various APIs are called in different		APIs, demonstrating when the various APIs are called in different
	situations and with various transport protocols. For clarity and		situations and with various transport protocols. For clarity and
	simplicity, the createOffer/createAnswer calls are assumed to be		simplicity, the createOffer/createAnswer calls are assumed to be
	skipping to change at page 31, line 27		skipping to change at page 41, line 24
	OffererUA->OffererJS: onaddstream(remoteStream);		OffererUA->OffererJS: onaddstream(remoteStream);
	// ICE Completes (at Offerer)		// ICE Completes (at Offerer)
	OffererUA->AnswererUA: Media		OffererUA->AnswererUA: Media
	// 200 OK arrives at Offerer		// 200 OK arrives at Offerer
	OffererJS: answer = parseResponse(sip);		OffererJS: answer = parseResponse(sip);
	OffererJS->OffererUA: pc.setRemoteDescription("answer", answer);		OffererJS->OffererUA: pc.setRemoteDescription("answer", answer);
	OffererJS->AnswererJS: ACK		OffererJS->AnswererJS: ACK
			A.2. Example Session Descriptions
			A.2.1. createOffer
			This SDP shows a typical initial offer, created by createOffer for a
			PeerConnection with a single audio MediaStreamTrack, a single video
			MediaStreamTrack, and a single data channel. Host candidates have
			also already been gathered. Note some lines have been broken into
			two lines for formatting reasons.
			v=0
			o=- 4962303333179871722 1 IN IP4 0.0.0.0
			s=-
			t=0 0
			a=group:BUNDLE audio video data
			m=audio 56500 RTP/SAVPF 111 0 8 126
			c=IN IP4 192.0.2.1
			a=rtcp:56501 IN IP4 192.0.2.1
			a=candidate:3348148302 1 udp 2113937151 192.0.2.1 56500
			typ host generation 0
			a=candidate:3348148302 2 udp 2113937151 192.0.2.1 56501
			typ host generation 0
			a=ice-ufrag:ETEn1v9DoTMB9J4r
			a=ice-pwd:OtSK0WpNtpUjkY4+86js7ZQl
			a=ice-options:trickle
			a=mid:audio
			a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
			a=sendrecv
			a=rtcp-mux
			a=rtcp-rsize
			a=fingerprint:sha-256
			19:E2:1C:3B:4B:9F:81:E6:B8:5C:F4:A5:A8:D8:73:04
			:BB:05:2F:70:9F:04:A9:0E:05:E9:26:33:E8:70:88:A2
			a=setup:actpass
			a=rtpmap:111 opus/48000/2
			a=fmtp:111 minptime=10
			a=rtpmap:0 PCMU/8000
			a=rtpmap:8 PCMA/8000
			a=rtpmap:126 telephone-event/8000
			a=maxptime:60
			a=ssrc:1732846380 cname:EocUG1f0fcg/yvY7
			a=msid:47017fee-b6c1-4162-929c-a25110252400
			f83006c5-a0ff-4e0a-9ed9-d3e6747be7d9
			m=video 56502 RTP/SAVPF 100 115 116 117
			c=IN IP4 192.0.2.1
			a=rtcp:56503 IN IP4 192.0.2.1
			a=candidate:3348148302 1 udp 2113937151 192.0.2.1 56502
			typ host generation 0
			a=candidate:3348148302 2 udp 2113937151 192.0.2.1 56503
			typ host generation 0
			a=ice-ufrag:BGKkWnG5GmiUpdIV
			a=ice-pwd:mqyWsAjvtKwTGnvhPztQ9mIf
			a=ice-options:trickle
			a=mid:video
			a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
			a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
			a=sendrecv
			a=rtcp-mux
			a=rtcp-rsize
			a=fingerprint:sha-256
			19:E2:1C:3B:4B:9F:81:E6:B8:5C:F4:A5:A8:D8:73:04
			:BB:05:2F:70:9F:04:A9:0E:05:E9:26:33:E8:70:88:A2
			a=setup:actpass
			a=rtpmap:100 VP8/90000
			a=rtcp-fb:100 ccm fir
			a=rtcp-fb:100 nack
			a=rtcp-fb:100 goog-remb
			a=rtpmap:115 rtx/90000
			a=fmtp:115 apt=100
			a=rtpmap:116 red/90000
			a=rtpmap:117 ulpfec/90000
			a=ssrc:1366781083 cname:EocUG1f0fcg/yvY7
			a=ssrc:1366781084 cname:EocUG1f0fcg/yvY7
			a=ssrc:1366781085 cname:EocUG1f0fcg/yvY7
			a=ssrc-group:FID 1366781083 1366781084
			a=ssrc-group:FEC 1366781083 1366781085
			a=msid:61317484-2ed4-49d7-9eb7-1414322a7aae
			f30bdb4a-5db8-49b5-bcdc-e0c9a23172e0
			m=application 56504 DTLS/SCTP 5000
			c=IN IP4 192.0.2.1
			a=candidate:3348148302 1 udp 2113937151 192.0.2.1 56504
			typ host generation 0
			a=ice-ufrag:VD5v2BnbZm3mgP3d
			a=ice-pwd:+Jlkuox+VVIUDqxcfIDuTZMH
			a=ice-options:trickle
			a=mid:data
			a=fingerprint:sha-256 19:E2:1C:3B:4B:9F:81:E6:B8:5C:F4:A5:A8:D8:73:04
			:BB:05:2F:70:9F:04:A9:0E:05:E9:26:33:E8:70:88:A2
			a=setup:actpass
			a=fmtp:5000 protocol=webrtc-datachannel; streams=10
			A.2.2. createAnswer
			This SDP shows a typical initial answer to the above offer, created
			by createAnswer for a PeerConnection with a single audio
			MediaStreamTrack, a single video MediaStreamTrack, and a single data
			channel. Host candidates have also already been gathered. Note some
			lines have been broken into two lines for formatting reasons.
			v=0
			o=- 6729291447651054566 1 IN IP4 0.0.0.0
			s=-
			t=0 0
			a=group:BUNDLE audio video data
			m=audio 20000 RTP/SAVPF 111 0 8 126
			c=IN IP4 192.0.2.2
			a=candidate:2299743422 1 udp 2113937151 192.0.2.2 20000
			typ host generation 0
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	Appendix B. Change log		Appendix B. Change log
			Changes in draft-04:
			o Filled in sections on createOffer and createAnswer.
			o Added SDP examples.
			o Fixed references.
	Changes in draft-03:		Changes in draft-03:
	o Added text describing relationship to W3C specification		o Added text describing relationship to W3C specification
	Changes in draft -02:		Changes in draft-02:
	o Converted from nroff		o Converted from nroff
	o Removed comparisons to old approaches abandoned by the working		o Removed comparisons to old approaches abandoned by the working
	group		group
	o Removed stuff that has moved to W3C specificaiton
			o Removed stuff that has moved to W3C specification
	o Align SDP handling with W3C draft		o Align SDP handling with W3C draft
	o Clarified section on forking.		o Clarified section on forking.
	Changes in draft -01:		Changes in draft-01:
	o Added diagrams for architecture and state machine.		o Added diagrams for architecture and state machine.
	o Added sections on forking and rehydration.		o Added sections on forking and rehydration.
	o Clarified meaning of "pranswer" and "answer".		o Clarified meaning of "pranswer" and "answer".
	o Reworked how ICE restarts and media directions are controlled.		o Reworked how ICE restarts and media directions are controlled.
	o Added list of parameters that can be changed in a description.		o Added list of parameters that can be changed in a description.
	o Updated suggested API and examples to match latest thinking.		o Updated suggested API and examples to match latest thinking.
	o Suggested API and examples have been moved to an appendix.		o Suggested API and examples have been moved to an appendix.
	Changes in draft -00:		Changes in draft -00:
	o Migrated from draft-uberti-rtcweb-jsep-02.		o Migrated from draft-uberti-rtcweb-jsep-02.
	Authors' Addresses		Authors' Addresses
	Justin Uberti		Justin Uberti
	Google		Google
	747 6th Ave S		747 6th Ave S
	Kirkland, WA 98033		Kirkland, WA 98033
	USA		USA
	Email: justin@uberti.name		Email: justin@uberti.name
	Cullen Jennings		Cullen Jennings
	Cisco		Cisco
	170 West Tasman Drive		170 West Tasman Drive
	San Jose, CA 95134		San Jose, CA 95134
	USA		USA
	Email: fluffy@iii.ca		Email: fluffy@iii.ca
End of changes. 82 change blocks.
	248 lines changed or deleted		983 lines changed or added
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