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Versions: (draft-uberti-rtcweb-jsep) 00 01 02 03 04 05 06 07 08 09 10 11 12 13 14 15 16 17 18 19 20

Network Working Group                                          J. Uberti
Internet-Draft                                                    Google
Intended status: Standards Track                             C. Jennings
Expires: July 20, 2017                                             Cisco
                                                        E. Rescorla, Ed.
                                                                 Mozilla
                                                        January 16, 2017


               Javascript Session Establishment Protocol
                       draft-ietf-rtcweb-jsep-18

Abstract

   This document describes the mechanisms for allowing a Javascript
   application to control the signaling plane of a multimedia session
   via the interface specified in the W3C RTCPeerConnection API, and
   discusses how this relates to existing signaling protocols.

Status of This Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on July 20, 2017.

Copyright Notice

   Copyright (c) 2017 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of



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   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   4
     1.1.  General Design of JSEP  . . . . . . . . . . . . . . . . .   4
     1.2.  Other Approaches Considered . . . . . . . . . . . . . . .   5
   2.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . .   6
   3.  Semantics and Syntax  . . . . . . . . . . . . . . . . . . . .   6
     3.1.  Signaling Model . . . . . . . . . . . . . . . . . . . . .   6
     3.2.  Session Descriptions and State Machine  . . . . . . . . .   7
     3.3.  Session Description Format  . . . . . . . . . . . . . . .  10
     3.4.  Session Description Control . . . . . . . . . . . . . . .  10
       3.4.1.  RtpTransceivers . . . . . . . . . . . . . . . . . . .  10
       3.4.2.  RtpSenders  . . . . . . . . . . . . . . . . . . . . .  11
       3.4.3.  RtpReceivers  . . . . . . . . . . . . . . . . . . . .  11
     3.5.  ICE . . . . . . . . . . . . . . . . . . . . . . . . . . .  11
       3.5.1.  ICE Gathering Overview  . . . . . . . . . . . . . . .  11
       3.5.2.  ICE Candidate Trickling . . . . . . . . . . . . . . .  12
         3.5.2.1.  ICE Candidate Format  . . . . . . . . . . . . . .  12
       3.5.3.  ICE Candidate Policy  . . . . . . . . . . . . . . . .  13
       3.5.4.  ICE Candidate Pool  . . . . . . . . . . . . . . . . .  14
     3.6.  Video Size Negotiation  . . . . . . . . . . . . . . . . .  15
       3.6.1.  Creating an imageattr Attribute . . . . . . . . . . .  15
       3.6.2.  Interpreting an imageattr Attribute . . . . . . . . .  16
     3.7.  Simulcast . . . . . . . . . . . . . . . . . . . . . . . .  17
     3.8.  Interactions With Forking . . . . . . . . . . . . . . . .  18
       3.8.1.  Sequential Forking  . . . . . . . . . . . . . . . . .  19
       3.8.2.  Parallel Forking  . . . . . . . . . . . . . . . . . .  19
   4.  Interface . . . . . . . . . . . . . . . . . . . . . . . . . .  20
     4.1.  PeerConnection  . . . . . . . . . . . . . . . . . . . . .  20
       4.1.1.  Constructor . . . . . . . . . . . . . . . . . . . . .  20
       4.1.2.  addTrack  . . . . . . . . . . . . . . . . . . . . . .  22
       4.1.3.  removeTrack . . . . . . . . . . . . . . . . . . . . .  23
       4.1.4.  addTransceiver  . . . . . . . . . . . . . . . . . . .  23
       4.1.5.  createDataChannel . . . . . . . . . . . . . . . . . .  23
       4.1.6.  createOffer . . . . . . . . . . . . . . . . . . . . .  24
       4.1.7.  createAnswer  . . . . . . . . . . . . . . . . . . . .  25
       4.1.8.  SessionDescriptionType  . . . . . . . . . . . . . . .  25
         4.1.8.1.  Use of Provisional Answers  . . . . . . . . . . .  26
         4.1.8.2.  Rollback  . . . . . . . . . . . . . . . . . . . .  27
       4.1.9.  setLocalDescription . . . . . . . . . . . . . . . . .  28
       4.1.10. setRemoteDescription  . . . . . . . . . . . . . . . .  28
       4.1.11. currentLocalDescription . . . . . . . . . . . . . . .  29
       4.1.12. pendingLocalDescription . . . . . . . . . . . . . . .  29
       4.1.13. currentRemoteDescription  . . . . . . . . . . . . . .  29
       4.1.14. pendingRemoteDescription  . . . . . . . . . . . . . .  29



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       4.1.15. canTrickleIceCandidates . . . . . . . . . . . . . . .  30
       4.1.16. setConfiguration  . . . . . . . . . . . . . . . . . .  30
       4.1.17. addIceCandidate . . . . . . . . . . . . . . . . . . .  31
     4.2.  RtpTransceiver  . . . . . . . . . . . . . . . . . . . . .  32
       4.2.1.  stop  . . . . . . . . . . . . . . . . . . . . . . . .  32
       4.2.2.  stopped . . . . . . . . . . . . . . . . . . . . . . .  32
       4.2.3.  setDirection  . . . . . . . . . . . . . . . . . . . .  32
       4.2.4.  direction . . . . . . . . . . . . . . . . . . . . . .  32
       4.2.5.  currentDirection  . . . . . . . . . . . . . . . . . .  33
       4.2.6.  setCodecPreferences . . . . . . . . . . . . . . . . .  33
   5.  SDP Interaction Procedures  . . . . . . . . . . . . . . . . .  33
     5.1.  Requirements Overview . . . . . . . . . . . . . . . . . .  34
       5.1.1.  Implementation Requirements . . . . . . . . . . . . .  34
       5.1.2.  Usage Requirements  . . . . . . . . . . . . . . . . .  35
       5.1.3.  Profile Names and Interoperability  . . . . . . . . .  36
     5.2.  Constructing an Offer . . . . . . . . . . . . . . . . . .  37
       5.2.1.  Initial Offers  . . . . . . . . . . . . . . . . . . .  37
       5.2.2.  Subsequent Offers . . . . . . . . . . . . . . . . . .  42
       5.2.3.  Options Handling  . . . . . . . . . . . . . . . . . .  46
         5.2.3.1.  IceRestart  . . . . . . . . . . . . . . . . . . .  46
         5.2.3.2.  VoiceActivityDetection  . . . . . . . . . . . . .  46
     5.3.  Generating an Answer  . . . . . . . . . . . . . . . . . .  47
       5.3.1.  Initial Answers . . . . . . . . . . . . . . . . . . .  47
       5.3.2.  Subsequent Answers  . . . . . . . . . . . . . . . . .  51
       5.3.3.  Options Handling  . . . . . . . . . . . . . . . . . .  53
         5.3.3.1.  VoiceActivityDetection  . . . . . . . . . . . . .  53
     5.4.  Modifying an Offer or Answer  . . . . . . . . . . . . . .  53
     5.5.  Processing a Local Description  . . . . . . . . . . . . .  54
     5.6.  Processing a Remote Description . . . . . . . . . . . . .  54
     5.7.  Parsing a Session Description . . . . . . . . . . . . . .  55
       5.7.1.  Session-Level Parsing . . . . . . . . . . . . . . . .  55
       5.7.2.  Media Section Parsing . . . . . . . . . . . . . . . .  57
       5.7.3.  Semantics Verification  . . . . . . . . . . . . . . .  59
     5.8.  Applying a Local Description  . . . . . . . . . . . . . .  60
     5.9.  Applying a Remote Description . . . . . . . . . . . . . .  62
     5.10. Applying an Answer  . . . . . . . . . . . . . . . . . . .  65
   6.  Processing RTP/RTCP . . . . . . . . . . . . . . . . . . . . .  68
   7.  Examples  . . . . . . . . . . . . . . . . . . . . . . . . . .  68
     7.1.  Simple Example  . . . . . . . . . . . . . . . . . . . . .  68
     7.2.  Normal Examples . . . . . . . . . . . . . . . . . . . . .  72
   8.  Security Considerations . . . . . . . . . . . . . . . . . . .  81
   9.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  81
   10. Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  81
   11. References  . . . . . . . . . . . . . . . . . . . . . . . . .  82
     11.1.  Normative References . . . . . . . . . . . . . . . . . .  82
     11.2.  Informative References . . . . . . . . . . . . . . . . .  85
   Appendix A.  Appendix A . . . . . . . . . . . . . . . . . . . . .  87
   Appendix B.  Appendix B . . . . . . . . . . . . . . . . . . . . .  88



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   Appendix C.  Change log . . . . . . . . . . . . . . . . . . . . .  91
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  99

1.  Introduction

   This document describes how the W3C WEBRTC RTCPeerConnection
   interface [W3C.WD-webrtc-20140617] is used to control the setup,
   management and teardown of a multimedia session.

1.1.  General Design of JSEP

   The thinking behind WebRTC call setup has been to fully specify and
   control the media plane, but to leave the signaling plane up to the
   application as much as possible.  The rationale is that different
   applications may prefer to use different protocols, such as the
   existing SIP or Jingle call signaling protocols, or something custom
   to the particular application, perhaps for a novel use case.  In this
   approach, the key information that needs to be exchanged is the
   multimedia session description, which specifies the necessary
   transport and media configuration information necessary to establish
   the media plane.

   With these considerations in mind, this document describes the
   Javascript Session Establishment Protocol (JSEP) that allows for full
   control of the signaling state machine from Javascript.  JSEP removes
   the browser almost entirely from the core signaling flow, which is
   instead handled by the Javascript making use of two interfaces: (1)
   passing in local and remote session descriptions and (2) interacting
   with the ICE state machine.

   In this document, the use of JSEP is described as if it always occurs
   between two browsers.  Note though in many cases it will actually be
   between a browser and some kind of server, such as a gateway or MCU.
   This distinction is invisible to the browser; it just follows the
   instructions it is given via the API.

   JSEP's handling of session descriptions is simple and
   straightforward.  Whenever an offer/answer exchange is needed, the
   initiating side creates an offer by calling a createOffer() API.  The
   application then uses that offer to set up its local config via the
   setLocalDescription() API.  The offer is finally sent off to the
   remote side over its preferred signaling mechanism (e.g.,
   WebSockets); upon receipt of that offer, the remote party installs it
   using the setRemoteDescription() API.

   To complete the offer/answer exchange, the remote party uses the
   createAnswer() API to generate an appropriate answer, applies it
   using the setLocalDescription() API, and sends the answer back to the



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   initiator over the signaling channel.  When the initiator gets that
   answer, it installs it using the setRemoteDescription() API, and
   initial setup is complete.  This process can be repeated for
   additional offer/answer exchanges.

   Regarding ICE [RFC5245], JSEP decouples the ICE state machine from
   the overall signaling state machine, as the ICE state machine must
   remain in the browser, because only the browser has the necessary
   knowledge of candidates and other transport info.  Performing this
   separation also provides additional flexibility; in protocols that
   decouple session descriptions from transport, such as Jingle, the
   session description can be sent immediately and the transport
   information can be sent when available.  In protocols that don't,
   such as SIP, the information can be used in the aggregated form.
   Sending transport information separately can allow for faster ICE and
   DTLS startup, since ICE checks can start as soon as any transport
   information is available rather than waiting for all of it.

   Through its abstraction of signaling, the JSEP approach does require
   the application to be aware of the signaling process.  While the
   application does not need to understand the contents of session
   descriptions to set up a call, the application must call the right
   APIs at the right times, convert the session descriptions and ICE
   information into the defined messages of its chosen signaling
   protocol, and perform the reverse conversion on the messages it
   receives from the other side.

   One way to mitigate this is to provide a Javascript library that
   hides this complexity from the developer; said library would
   implement a given signaling protocol along with its state machine and
   serialization code, presenting a higher level call-oriented interface
   to the application developer.  For example, libraries exist to adapt
   the JSEP API into an API suitable for a SIP or XMPP.  Thus, JSEP
   provides greater control for the experienced developer without
   forcing any additional complexity on the novice developer.

1.2.  Other Approaches Considered

   One approach that was considered instead of JSEP was to include a
   lightweight signaling protocol.  Instead of providing session
   descriptions to the API, the API would produce and consume messages
   from this protocol.  While providing a more high-level API, this put
   more control of signaling within the browser, forcing the browser to
   have to understand and handle concepts like signaling glare.  In
   addition, it prevented the application from driving the state machine
   to a desired state, as is needed in the page reload case.





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   A second approach that was considered but not chosen was to decouple
   the management of the media control objects from session
   descriptions, instead offering APIs that would control each component
   directly.  This was rejected based on a feeling that requiring
   exposure of this level of complexity to the application programmer
   would not be beneficial; it would result in an API where even a
   simple example would require a significant amount of code to
   orchestrate all the needed interactions, as well as creating a large
   API surface that needed to be agreed upon and documented.  In
   addition, these API points could be called in any order, resulting in
   a more complex set of interactions with the media subsystem than the
   JSEP approach, which specifies how session descriptions are to be
   evaluated and applied.

   One variation on JSEP that was considered was to keep the basic
   session description-oriented API, but to move the mechanism for
   generating offers and answers out of the browser.  Instead of
   providing createOffer/createAnswer methods within the browser, this
   approach would instead expose a getCapabilities API which would
   provide the application with the information it needed in order to
   generate its own session descriptions.  This increases the amount of
   work that the application needs to do; it needs to know how to
   generate session descriptions from capabilities, and especially how
   to generate the correct answer from an arbitrary offer and the
   supported capabilities.  While this could certainly be addressed by
   using a library like the one mentioned above, it basically forces the
   use of said library even for a simple example.  Providing
   createOffer/createAnswer avoids this problem, but still allows
   applications to generate their own offers/answers (to a large extent)
   if they choose, using the description generated by createOffer as an
   indication of the browser's capabilities.

2.  Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in [RFC2119].

3.  Semantics and Syntax

3.1.  Signaling Model

   JSEP does not specify a particular signaling model or state machine,
   other than the generic need to exchange session descriptions in the
   fashion described by [RFC3264](offer/answer) in order for both sides
   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
   a session.  However, the browser is totally decoupled from the actual



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   mechanism by which these offers and answers are communicated to the
   remote side, including addressing, retransmission, forking, and glare
   handling.  These issues are left entirely up to the application; the
   application has complete control over which offers and answers get
   handed to the browser, and when.


       +-----------+                               +-----------+
       |  Web App  |<--- App-Specific Signaling -->|  Web App  |
       +-----------+                               +-----------+
             ^                                            ^
             |  SDP                                       |  SDP
             V                                            V
       +-----------+                                +-----------+
       |  Browser  |<----------- Media ------------>|  Browser  |
       +-----------+                                +-----------+


                      Figure 1: JSEP Signaling Model

3.2.  Session Descriptions and State Machine

   In order to establish the media plane, the user agent needs specific
   parameters to indicate what to transmit to the remote side, as well
   as how to handle the media that is received.  These parameters are
   determined by the exchange of session descriptions in offers and
   answers, and there are certain details to this process that must be
   handled in the JSEP APIs.

   Whether a session description applies to the local side or the remote
   side affects the meaning of that description.  For example, the list
   of codecs sent to a remote party indicates what the local side is
   willing to receive, which, when intersected with the set of codecs
   the remote side supports, specifies what the remote side should send.
   However, not all parameters follow this rule; for example, the DTLS-
   SRTP parameters [RFC5763] sent to a remote party indicate what
   certificate the local side will use in DTLS setup, and thereby what
   the remote party should expect to receive; the remote party will have
   to accept these parameters, with no option to choose different
   values.

   In addition, various RFCs put different conditions on the format of
   offers versus answers.  For example, an offer may propose an
   arbitrary number of media streams (i.e. m= sections), but an answer
   must contain the exact same number as the offer.

   Lastly, while the exact media parameters are only known only after an
   offer and an answer have been exchanged, it is possible for the



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   offerer to receive media after they have sent an offer and before
   they have received an answer.  To properly process incoming media in
   this case, the offerer's media handler must be aware of the details
   of the offer before the answer arrives.

   Therefore, in order to handle session descriptions properly, the user
   agent needs:

   1.  To know if a session description pertains to the local or remote
       side.

   2.  To know if a session description is an offer or an answer.

   3.  To allow the offer to be specified independently of the answer.

   JSEP addresses this by adding both setLocalDescription and
   setRemoteDescription methods and having session description objects
   contain a type field indicating the type of session description being
   supplied.  This satisfies the requirements listed above for both the
   offerer, who first calls setLocalDescription(sdp [offer]) and then
   later setRemoteDescription(sdp [answer]), as well as for the
   answerer, who first calls setRemoteDescription(sdp [offer]) and then
   later setLocalDescription(sdp [answer]).

   JSEP also allows for an answer to be treated as provisional by the
   application.  Provisional answers provide a way for an answerer to
   communicate initial session parameters back to the offerer, in order
   to allow the session to begin, while allowing a final answer to be
   specified later.  This concept of a final answer is important to the
   offer/answer model; when such an answer is received, any extra
   resources allocated by the caller can be released, now that the exact
   session configuration is known.  These "resources" can include things
   like extra ICE components, TURN candidates, or video decoders.
   Provisional answers, on the other hand, do no such deallocation; as a
   result, multiple dissimilar provisional answers, with their own codec
   choices, transport parameters, etc., can be received and applied
   during call setup.  Note that the final answer itself may be
   different than any received provisional answers.

   In [RFC3264], the constraint at the signaling level is that only one
   offer can be outstanding for a given session, but at the media stack
   level, a new offer can be generated at any point.  For example, when
   using SIP for signaling, if one offer is sent, then cancelled using a
   SIP CANCEL, another offer can be generated even though no answer was
   received for the first offer.  To support this, the JSEP media layer
   can provide an offer via the createOffer() method whenever the
   Javascript application needs one for the signaling.  The answerer can
   send back zero or more provisional answers, and finally end the



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   offer-answer exchange by sending a final answer.  The state machine
   for this is as follows:


                       setRemote(OFFER)               setLocal(PRANSWER)
                           /-----\                               /-----\
                           |     |                               |     |
                           v     |                               v     |
            +---------------+    |                +---------------+    |
            |               |----/                |               |----/
            |               | setLocal(PRANSWER)  |               |
            |  Remote-Offer |------------------- >| Local-Pranswer|
            |               |                     |               |
            |               |                     |               |
            +---------------+                     +---------------+
                 ^   |                                   |
                 |   | setLocal(ANSWER)                  |
   setRemote(OFFER)  |                                   |
                 |   V                  setLocal(ANSWER) |
            +---------------+                            |
            |               |                            |
            |               |<---------------------------+
            |    Stable     |
            |               |<---------------------------+
            |               |                            |
            +---------------+          setRemote(ANSWER) |
                 ^   |                                   |
                 |   | setLocal(OFFER)                   |
   setRemote(ANSWER) |                                   |
                 |   V                                   |
            +---------------+                     +---------------+
            |               |                     |               |
            |               | setRemote(PRANSWER) |               |
            |  Local-Offer  |------------------- >|Remote-Pranswer|
            |               |                     |               |
            |               |----\                |               |----\
            +---------------+    |                +---------------+    |
                           ^     |                               ^     |
                           |     |                               |     |
                           \-----/                               \-----/
                       setLocal(OFFER)               setRemote(PRANSWER)


                       Figure 2: JSEP State Machine

   Aside from these state transitions there is no other difference
   between the handling of provisional ("pranswer") and final ("answer")
   answers.



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3.3.  Session Description Format

   JSEP's session descriptions use SDP syntax for their internal
   representation.  While this format is not optimal for manipulation
   from Javascript, it is widely accepted, and frequently updated with
   new features; any alternate encoding of session descriptions would
   have to keep pace with the changes to SDP, at least until the time
   that this new encoding eclipsed SDP in popularity.

   However, to simplify Javascript processing, and provide for future
   flexibility, the SDP syntax is encapsulated within a
   SessionDescription object, which can be constructed from SDP, and be
   serialized out to SDP.  If future specifications agree on a JSON
   format for session descriptions, we could easily enable this object
   to generate and consume that JSON.

   Other methods may be added to SessionDescription in the future to
   simplify handling of SessionDescriptions from Javascript.  In the
   meantime, Javascript libraries can be used to perform these
   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.

3.4.  Session Description Control

   In order to give the application control over various common session
   parameters, JSEP provides control surfaces which tell the browser how
   to generate session descriptions.  This avoids the need for
   Javascript to modify session descriptions in most cases.

   Changes to these objects result in changes to the session
   descriptions generated by subsequent createOffer/Answer calls.

3.4.1.  RtpTransceivers

   RtpTransceivers allow the application to control the RTP media
   associated with one m= section.  Each RtpTransceiver has an RtpSender
   and an RtpReceiver, which an application can use to control the
   sending and receiving of RTP media.  The application may also modify
   the RtpTransceiver directly, for instance, by stopping it.

   RtpTransceivers generally have a 1:1 mapping with m= sections,
   although there may be more RtpTransceivers than m= sections when
   RtpTransceivers are created but not yet associated with a m= section,
   or if RtpTransceivers have been stopped and disassociated from m=



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   sections.  An RtpTransceiver is said to be associated with an m=
   section if its mid property is non-null; otherwise it is said to be
   disassociated.  The associated m= section is determined using a
   mapping between transceivers and m= section indices, formed when
   creating an offer or applying a remote offer.  An RtpTransceiver is
   never associated with more than one m= section, and once a session
   description is applied, a m= section is always associated with
   exactly one RtpTransceiver.

   RtpTransceivers can be created explicitly by the application or
   implicitly by calling setRemoteDescription with an offer that adds
   new m= sections.

3.4.2.  RtpSenders

   RtpSenders allow the application to control how RTP media is sent.
   An RtpSender is conceptually responsible for the outgoing RTP
   stream(s) described by an m= section.  This includes encoding the
   attached MediaStreamTrack, sending RTP media packets, and generating/
   processing RTCP for the outgoing RTP streams(s).

3.4.3.  RtpReceivers

   RtpReceivers allow the application to inspect how RTP media is
   received.  An RtpReceiver is conceptually responsible for the
   incoming RTP stream(s) described by an m= section.  This includes
   processing received RTP media packets, decoding the incoming
   stream(s) to produce a remote MediaStreamTrack, and generating/
   processing RTCP for the incoming RTP stream(s).

3.5.  ICE

3.5.1.  ICE Gathering Overview

   JSEP gathers ICE candidates as needed by the application.  Collection
   of ICE candidates is referred to as a gathering phase, and this is
   triggered either by the addition of a new or recycled m= section to
   the local session description, or new ICE credentials in the
   description, indicating an ICE restart.  Use of new ICE credentials
   can be triggered explicitly by the application, or implicitly by the
   browser in response to changes in the ICE configuration.

   When the ICE configuration changes in a way that requires a new
   gathering phase, a 'needs-ice-restart' bit is set.  When this bit is
   set, calls to the createOffer API will generate new ICE credentials.
   This bit is cleared by a call to the setLocalDescription API with new
   ICE credentials from either an offer or an answer, i.e., from either
   a local- or remote-initiated ICE restart.



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   When a new gathering phase starts, the ICE Agent will notify the
   application that gathering is occurring through an event.  Then, when
   each new ICE candidate becomes available, the ICE Agent will supply
   it to the application via an additional event; these candidates will
   also automatically be added to the current and/or pending local
   session description.  Finally, when all candidates have been
   gathered, an event will be dispatched to signal that the gathering
   process is complete.

   Note that gathering phases only gather the candidates needed by
   new/recycled/restarting m= sections; other m= sections continue to
   use their existing candidates.  Also, when bundling is active,
   candidates are only gathered (and exchanged) for the m= sections
   referenced in BUNDLE-tags, as described in
   [I-D.ietf-mmusic-sdp-bundle-negotiation].

3.5.2.  ICE Candidate Trickling

   Candidate trickling is a technique through which a caller may
   incrementally provide candidates to the callee after the initial
   offer has been dispatched; the semantics of "Trickle ICE" are defined
   in [I-D.ietf-ice-trickle].  This process allows the callee to begin
   acting upon the call and setting up the ICE (and perhaps DTLS)
   connections immediately, without having to wait for the caller to
   gather all possible candidates.  This results in faster media setup
   in cases where gathering is not performed prior to initiating the
   call.

   JSEP supports optional candidate trickling by providing APIs, as
   described above, that provide control and feedback on the ICE
   candidate gathering process.  Applications that support candidate
   trickling can send the initial offer immediately and send individual
   candidates when they get the notified of a new candidate;
   applications that do not support this feature can simply wait for the
   indication that gathering is complete, and then create and send their
   offer, with all the candidates, at this time.

   Upon receipt of trickled candidates, the receiving application will
   supply them to its ICE Agent.  This triggers the ICE Agent to start
   using the new remote candidates for connectivity checks.

3.5.2.1.  ICE Candidate Format

   In JSEP, ICE candidates are abstracted by an IceCandidate object, and
   as with session descriptions, SDP syntax is used for the internal
   representation.





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   The candidate details are specified in an IceCandidate field, using
   the same SDP syntax as the "candidate-attribute" field defined in
   [RFC5245], Section 15.1.  For example:


   candidate:1 1 UDP 1694498815 192.0.2.33 10000 typ host


   The IceCandidate object contains a field to indicate which ICE ufrag
   it is associated with, as defined in [RFC5245], Section 15.4.  This
   value is used to determine which session description (and thereby
   which gathering phase) this IceCandidate belongs to, which helps
   resolve ambiguities during ICE restarts.  If this field is absent in
   a received IceCandidate (perhaps when communicating with a non-JSEP
   endpoint), the most recently received session description is assumed.

   The IceCandidate object also contains fields to indicate which m=
   section it is associated with, which can be identified in one of two
   ways, either by a m= section index, or a MID.  The m= section index
   is a zero-based index, with index N referring to the N+1th m= section
   in the session description referenced by this IceCandidate.  The MID
   is a "media stream identification" value, as defined in [RFC5888],
   Section 4, which provides a more robust way to identify the m=
   section in the session description, using the MID of the associated
   RtpTransceiver object (which may have been locally generated by the
   answerer when interacting with a non-JSEP endpoint that does not
   support the MID attribute, as discussed in Section 5.9 below).  If
   the MID field is present in a received IceCandidate, it MUST be used
   for identification; otherwise, the m= section index is used instead.

   When creating an IceCandidate object, JSEP implementations MUST
   populate all of these fields.

3.5.3.  ICE Candidate Policy

   Typically, when gathering ICE candidates, the browser will gather all
   possible forms of initial candidates - host, server reflexive, and
   relay.  However, in certain cases, applications may want to have more
   specific control over the gathering process, due to privacy or
   related concerns.  For example, one may want to only use relay
   candidates, to leak as little location information as possible
   (keeping in mind that this choice comes with corresponding
   operational costs).  To accomplish this, JSEP allows the application
   to restrict which ICE candidates are used in a session.  Note that
   this filtering is applied on top of any restrictions the browser
   chooses to enforce regarding which IP addresses are permitted for the
   application, as discussed in [I-D.ietf-rtcweb-ip-handling].




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   There may also be cases where the application wants to change which
   types of candidates are used while the session is active.  A prime
   example is where a callee may initially want to use only relay
   candidates, to avoid leaking location information to an arbitrary
   caller, but then change to use all candidates (for lower operational
   cost) once the user has indicated they want to take the call.  For
   this scenario, the browser MUST allow the candidate policy to be
   changed in mid-session, subject to the aforementioned interactions
   with local policy.

   To administer the ICE candidate policy, the browser will determine
   the current setting at the start of each gathering phase.  Then,
   during the gathering phase, the browser MUST NOT expose candidates
   disallowed by the current policy to the application, use them as the
   source of connectivity checks, or indirectly expose them via other
   fields, such as the raddr/rport attributes for other ICE candidates.
   Later, if a different policy is specified by the application, the
   application can apply it by kicking off a new gathering phase via an
   ICE restart.

3.5.4.  ICE Candidate Pool

   JSEP applications typically inform the browser to begin ICE gathering
   via the information supplied to setLocalDescription, as this is where
   the app specifies the number of media streams, and thereby ICE
   components, for which to gather candidates.  However, to accelerate
   cases where the application knows the number of ICE components to use
   ahead of time, it may ask the browser to gather a pool of potential
   ICE candidates to help ensure rapid media setup.

   When setLocalDescription is eventually called, and the browser goes
   to gather the needed ICE candidates, it SHOULD start by checking if
   any candidates are available in the pool.  If there are candidates in
   the pool, they SHOULD be handed to the application immediately via
   the ICE candidate event.  If the pool becomes depleted, either
   because a larger-than-expected number of ICE components is used, or
   because the pool has not had enough time to gather candidates, the
   remaining candidates are gathered as usual.  This only occurs for the
   first offer/answer exchange, after which the candidate pool is
   emptied and no longer used.

   One example of where this concept is useful is an application that
   expects an incoming call at some point in the future, and wants to
   minimize the time it takes to establish connectivity, to avoid
   clipping of initial media.  By pre-gathering candidates into the
   pool, it can exchange and start sending connectivity checks from
   these candidates almost immediately upon receipt of a call.  Note
   though that by holding on to these pre-gathered candidates, which



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   will be kept alive as long as they may be needed, the application
   will consume resources on the STUN/TURN servers it is using.

3.6.  Video Size Negotiation

   Video size negotiation is the process through which a receiver can
   use the "a=imageattr" SDP attribute [RFC6236] to indicate what video
   frame sizes it is capable of receiving.  A receiver may have hard
   limits on what its video decoder can process, or it may wish to
   constrain what it receives due to application preferences, e.g. a
   specific size for the window in which the video will be displayed.

   Note that certain codecs support transmission of samples with aspect
   ratios other than 1.0 (i.e., non-square pixels).  JSEP
   implementations will not transmit non-square pixels, but SHOULD
   receive and render such video with the correct aspect ratio.
   However, sample aspect ratio has no impact on the size negotiation
   described below; all dimensions are measured in pixels, whether
   square or not.

3.6.1.  Creating an imageattr Attribute

   In order to determine the limits on what video resolution a receiver
   wants to receive, it will intersect its decoder hard limits with any
   mandatory constraints that have been applied to the associated
   MediaStreamTrack.  If the decoder limits are unknown, e.g. when using
   a software decoder, the mandatory constraints are used directly.  For
   the answerer, these mandatory constraints can be applied to the
   remote MediaStreamTracks that are created by a setRemoteDescription
   call, and will affect the output of the ensuing createAnswer call.
   Any constraints set after setLocalDescription is used to set the
   answer will result in a new offer-answer exchange.  For the offerer,
   because it does not know about any remote MediaStreamTracks until it
   receives the answer, the offer can only reflect decoder hard limits.
   If the offerer wishes to set mandatory constraints on video
   resolution, it must do so after receiving the answer, and the result
   will be a new offer-answer to communicate them.

   If there are no known decoder limits or mandatory constraints, the
   "a=imageattr" attribute SHOULD be omitted.

   Otherwise, an "a=imageattr" attribute is created with "recv"
   direction, and the resulting resolution space formed by intersecting
   the decoder limits and constraints is used to specify its minimum and
   maximum x= and y= values.  If the intersection is the null set, i.e.,
   there are no resolutions that are permitted by both the decoder and
   the mandatory constraints, this MUST be represented by x=0 and y=0
   values.



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   The rules here express a single set of preferences, and therefore,
   the "a=imageattr" q= value is not important.  It SHOULD be set to
   1.0.

   The "a=imageattr" field is payload type specific.  When all video
   codecs supported have the same capabilities, use of a single
   attribute, with the wildcard payload type (*), is RECOMMENDED.
   However, when the supported video codecs have differing capabilities,
   specific "a=imageattr" attributes MUST be inserted for each payload
   type.

   As an example, consider a system with a multiformat video decoder,
   which is capable of decoding any resolution from 48x48 to 720p, and
   where the application has constrained the received track to at most
   360p.  In this case, the implementation would generate this
   attribute:

   a=imageattr:* recv [x=[48:640],y=[48:360],q=1.0]

   This declaration indicates that the receiver is capable of decoding
   any image resolution from 48x48 up to 640x360 pixels.

3.6.2.  Interpreting an imageattr Attribute

   [RFC6236] defines "a=imageattr" to be an advisory field.  This means
   that it does not absolutely constrain the video formats that the
   sender can use, but gives an indication of the preferred values.

   This specification prescribes more specific behavior.  When a sender
   of a given MediaStreamTrack, which is producing video of a certain
   resolution, receives an "a=imageattr recv" attribute, it MUST check
   to see if the original resolution meets the size criteria specified
   in the attribute, and adapt the resolution accordingly by scaling (if
   appropriate).  Note that when considering a MediaStreamTrack that is
   producing rotated video, the unrotated resolution MUST be used.  This
   is required regardless of whether the receiver supports performing
   receive-side rotation (e.g., through CVO), as it significantly
   simplifies the matching logic.

   For the purposes of resolution negotiation, only size limits are
   considered.  Any other values, e.g. picture or sample aspect ratio,
   MUST be ignored.

   When communicating with a non-JSEP endpoint, multiple relevant
   "a=imageattr recv" attributes may be present in a received m=
   section.  If this occurs, attributes other than the one with the
   highest "q=" value MUST be ignored.  If multiple attributes have the




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   same "q=" value, those that appear after the first such attribute in
   the m= section MUST be ignored.

   If an "a=imageattr recv" attribute references a different video
   payload type than what has been selected for sending the
   MediaStreamTrack, it MUST be ignored.

   If the original resolution matches the size limits in the attribute,
   the track MUST be transmitted untouched.

   If the original resolution exceeds the size limits in the attribute,
   the sender SHOULD apply downscaling to the output of the
   MediaStreamTrack in order to satisfy the limits.  Downscaling MUST
   NOT change the track aspect ratio.

   If the original resolution is less than the size limits in the
   attribute, upscaling is needed, but this may not be appropriate in
   all cases.  To address this concern, the application can set an
   upscaling policy for each sent track.  For this case, if upscaling is
   permitted by policy, the sender SHOULD apply upscaling in order to
   provide the desired resolution.  Otherwise, the sender MUST NOT apply
   upscaling.  The sender SHOULD NOT upscale in other cases, even if the
   policy permits it.  Upscaling MUST NOT change the track aspect ratio.

   If there is no appropriate and permitted scaling mechanism that
   allows the received size limits to be satisfied, the sender MUST NOT
   transmit the track.

   If the attribute includes a "sar=" (sample aspect ratio) value set to
   something other than "1.0", indicating the receiver wants to receive
   non-square pixels, this cannot be satisfied and the sender MUST NOT
   transmit the track.

   In the special case of receiving a maximum resolution of [0, 0], as
   described above, the sender MUST NOT transmit the track.

3.7.  Simulcast

   JSEP supports simulcast transmission of a MediaStreamTrack, where
   multiple encodings of the source media can be transmitted within the
   context of a single m= section.  The current JSEP API is designed to
   allow applications to send simulcasted media but only to receive a
   single encoding.  This allows for multi-user scenarios where each
   sending client sends multiple encodings to a server, which then, for
   each receiving client, chooses the appropriate encoding to forward.

   Applications request support for simulcast by configuring multiple
   encodings on an RtpSender, which, upon generation of an offer or



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   answer, are indicated in SDP markings on the corresponding m=
   section, as described below.  Receivers that understand simulcast and
   are willing to receive it will also include SDP markings to indicate
   their support, and JSEP endpoints will use these markings to
   determine whether simulcast is permitted for a given RtpSender.  If
   simulcast support is not negotiated, the RtpSender will only use the
   first configured encoding.

   Note that the exact simulcast parameters are up to the sending
   application.  While the aforementioned SDP markings are provided to
   ensure the remote side can receive and demux multiple simulcast
   encodings, the specific resolutions and bitrates to be used for each
   encoding are purely a send-side decision in JSEP.

   JSEP currently does not provide a mechanism to configure receipt of
   simulcast.  This means that if simulcast is offered by the remote
   endpoint, the answer generated by a JSEP endpoint will not indicate
   support for receipt of simulcast, and as such the remote endpoint
   will only send a single encoding per m= section.

   In addition, JSEP does not provide a mechanism to handle an incoming
   offer requesting simulcast from the JSEP endpoint.  This means that
   established simulcast streams will continue to work through a
   received re-offer, but setting up initial simulcast by way of a
   received offer requires out-of-band signaling or SDP inspection.
   Future versions of this specification may add additional APIs to
   provide direct control.

   When using JSEP to transmit multiple encodings from a RtpSender, the
   techniques from [I-D.ietf-mmusic-sdp-simulcast] and
   [I-D.ietf-mmusic-rid] are used.  Specifically, when multiple
   encodings have been configured for a RtpSender, the m= section for
   the RtpSender will include an "a=simulcast" attribute, as defined in
   [I-D.ietf-mmusic-sdp-simulcast], Section 6.2, with a "send" simulcast
   stream description that lists each desired encoding, and no "recv"
   simulcast stream description.  The m= section will also include an
   "a=rid" attribute for each encoding, as specified in
   [I-D.ietf-mmusic-rid], Section 4; the use of RID identifiers allows
   the individual encodings to be disambiguated even though they are all
   part of the same m= section.

3.8.  Interactions With Forking

   Some call signaling systems allow various types of forking where an
   SDP Offer may be provided to more than one device.  For example, SIP
   [RFC3261] defines both a "Parallel Search" and "Sequential Search".
   Although these are primarily signaling level issues that are outside
   the scope of JSEP, they do have some impact on the configuration of



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   the media plane that is relevant.  When forking happens at the
   signaling layer, the Javascript application responsible for the
   signaling needs to make the decisions about what media should be sent
   or received at any point of time, as well as which remote endpoint it
   should communicate with; JSEP is used to make sure the media engine
   can make the RTP and media perform as required by the application.
   The basic operations that the applications can have the media engine
   do are:

   o  Start exchanging media with a given remote peer, but keep all the
      resources reserved in the offer.

   o  Start exchanging media with a given remote peer, and free any
      resources in the offer that are not being used.

3.8.1.  Sequential Forking

   Sequential forking involves a call being dispatched to multiple
   remote callees, where each callee can accept the call, but only one
   active session ever exists at a time; no mixing of received media is
   performed.

   JSEP handles sequential forking well, allowing the application to
   easily control the policy for selecting the desired remote endpoint.
   When an answer arrives from one of the callees, the application can
   choose to apply it either as a provisional answer, leaving open the
   possibility of using a different answer in the future, or apply it as
   a final answer, ending the setup flow.

   In a "first-one-wins" situation, the first answer will be applied as
   a final answer, and the application will reject any subsequent
   answers.  In SIP parlance, this would be ACK + BYE.

   In a "last-one-wins" situation, all answers would be applied as
   provisional answers, and any previous call leg will be terminated.
   At some point, the application will end the setup process, perhaps
   with a timer; at this point, the application could reapply the
   pending remote description as a final answer.

3.8.2.  Parallel Forking

   Parallel forking involves a call being dispatched to multiple remote
   callees, where each callee can accept the call, and multiple
   simultaneous active signaling sessions can be established as a
   result.  If multiple callees send media at the same time, the
   possibilities for handling this are described in Section 3.1 of
   [RFC3960].  Most SIP devices today only support exchanging media with
   a single device at a time, and do not try to mix multiple early media



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   audio sources, as that could result in a confusing situation.  For
   example, consider having a European ringback tone mixed together with
   the North American ringback tone - the resulting sound would not be
   like either tone, and would confuse the user.  If the signaling
   application wishes to only exchange media with one of the remote
   endpoints at a time, then from a media engine point of view, this is
   exactly like the sequential forking case.

   In the parallel forking case where the Javascript application wishes
   to simultaneously exchange media with multiple peers, the flow is
   slightly more complex, but the Javascript application can follow the
   strategy that [RFC3960] describes using UPDATE.  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, this offer used in
   the UPDATE would be formed by simply creating a new PeerConnection
   and making sure that the same local media streams have been added
   into this new PeerConnection.  Then the new PeerConnection object
   would produce a SDP offer that could be used by the signaling to
   perform the UPDATE strategy discussed in [RFC3960].

   As a result of sharing the media streams, the application will end up
   with N parallel PeerConnection sessions, each with a local and remote
   description and their own local and remote addresses.  The media flow
   from these sessions can be managed using setDirection (see
   Section 4.2.3), or the application can choose to play out the media
   from all sessions mixed together.  Of course, if the application
   wants to only keep a single session, it can simply terminate the
   sessions that it no longer needs.

4.  Interface

   This section details the basic operations that must be present to
   implement JSEP functionality.  The actual API exposed in the W3C API
   may have somewhat different syntax, but should map easily to these
   concepts.

4.1.  PeerConnection

4.1.1.  Constructor

   The PeerConnection constructor allows the application to specify
   global parameters for the media session, such as the STUN/TURN
   servers and credentials to use when gathering candidates, as well as
   the initial ICE candidate policy and pool size, and also the bundle
   policy to use.

   If an ICE candidate policy is specified, it functions as described in
   Section 3.5.3, causing the browser to only surface the permitted



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   candidates (including any internal browser filtering) to the
   application, and only use those candidates for connectivity checks.
   The set of available policies is as follows:

   all:  All candidates permitted by browser policy will be gathered and
      used.



   relay:  All candidates except relay candidates will be filtered out.
      This obfuscates the location information that might be ascertained
      by the remote peer from the received candidates.  Depending on how
      the application deploys and chooses relay servers, this could
      obfuscate location to a metro or possibly even global level.

   The default ICE candidate policy MUST be set to "all" as this is
   generally the desired policy, and also typically reduces use of
   application TURN server resources significantly.

   If a size is specified for the ICE candidate pool, this indicates the
   number of ICE components to pre-gather candidates for.  Because pre-
   gathering results in utilizing STUN/TURN server resources for
   potentially long periods of time, this must only occur upon
   application request, and therefore the default candidate pool size
   MUST be zero.

   The application can specify its preferred policy regarding use of
   bundle, the multiplexing mechanism defined in
   [I-D.ietf-mmusic-sdp-bundle-negotiation].  Regardless of policy, the
   application will always try to negotiate bundle onto a single
   transport, and will offer a single bundle group across all media
   section; use of this single transport is contingent upon the answerer
   accepting bundle.  However, by specifying a policy from the list
   below, the application can control exactly how aggressively it will
   try to bundle media streams together, which affects how it will
   interoperate with a non-bundle-aware endpoint.  When negotiating with
   a non-bundle-aware endpoint, only the streams not marked as bundle-
   only streams will be established.

   The set of available policies is as follows:

   balanced:  The first media section of each type (audio, video, or
      application) will contain transport parameters, which will allow
      an answerer to unbundle that section.  The second and any
      subsequent media section of each type will be marked bundle-only.
      The result is that if there are N distinct media types, then
      candidates will be gathered for for N media streams.  This policy
      balances desire to multiplex with the need to ensure basic audio



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      and video can still be negotiated in legacy cases.  When acting as
      answerer, if there is no bundle group in the offer, the
      implementation will reject all but the first m= section of each
      type.



   max-compat:  All media sections will contain transport parameters;
      none will be marked as bundle-only.  This policy will allow all
      streams to be received by non-bundle-aware endpoints, but require
      separate candidates to be gathered for each media stream.



   max-bundle:  Only the first media section will contain transport
      parameters; all streams other than the first will be marked as
      bundle-only.  This policy aims to minimize candidate gathering and
      maximize multiplexing, at the cost of less compatibility with
      legacy endpoints.  When acting as answerer, the implementation
      will reject any m= sections other than the first m= section,
      unless they are in the same bundle group as that m= section.

   As it provides the best tradeoff between performance and
   compatibility with legacy endpoints, the default bundle policy MUST
   be set to "balanced".

   The application can specify its preferred policy regarding use of
   RTP/RTCP multiplexing [RFC5761]  using one of the following policies:

   negotiate:  The browser will gather both RTP and RTCP candidates but
      also will offer "a=rtcp-mux", thus allowing for compatibility with
      either multiplexing or non-multiplexing endpoints.

   require:  The browser will only gather RTP candidates.  This halves
      the number of candidates that the offerer needs to gather.
      Applying a description with an m= section that does not contain an
      "a=rtcp-mux" attribute will cause an error to be returned.

   The default multiplexing policy MUST be set to "require".
   Implementations MAY choose to reject attempts by the application to
   set the multiplexing policy to "negotiate".

4.1.2.  addTrack

   The addTrack method adds a MediaStreamTrack to the PeerConnection,
   using the MediaStream argument to associate the track with other
   tracks in the same MediaStream, so that they can be added to the same
   "LS" group when creating an offer or answer. addTrack attempts to



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   minimize the number of transceivers as follows: If the PeerConnection
   is in the "have-remote-offer" state, the track will be attached to
   the first compatible transceiver that was created by the most recent
   call to setRemoteDescription() and does not have a local track.
   Otherwise, a new transceiver will be created, as described in
   Section 4.1.4.

4.1.3.  removeTrack

   The removeTrack method removes a MediaStreamTrack from the
   PeerConnection, using the RtpSender argument to indicate which sender
   should have its track removed.  The sender's track is cleared, and
   the sender stops sending.  Future calls to createOffer will mark the
   media description associated with the sender as recvonly (if
   transceiver.currentDirection is sendrecv) or as inactive (if
   transceiver.currentDirection is sendonly).

4.1.4.  addTransceiver

   The addTransceiver method adds a new RtpTransceiver to the
   PeerConnection.  If a MediaStreamTrack argument is provided, then the
   transceiver will be configured with that media type and the track
   will be attached to the transceiver.  Otherwise, the application MUST
   explicitly specify the type; this mode is useful for creating
   recvonly transceivers as well as for creating transceivers to which a
   track can be attached at some later point.

   At the time of creation, the application can also specify a
   transceiver direction attribute, a set of MediaStreams which the
   transceiver is associated with (allowing LS group assignments), and a
   set of encodings for the media (used for simulcast as described in
   Section 3.7).

4.1.5.  createDataChannel

   The createDataChannel method creates a new data channel and attaches
   it to the PeerConnection.  If no data channel currently exists for
   this PeerConnection, then a new offer/answer exchange is required.
   All data channels on a given PeerConnection share the same SCTP/DTLS
   association and therefore the same m= section, so subsequent creation
   of data channels does not have any impact on the JSEP state.

   The createDataChannel method also includes a number of arguments
   which are used by the PeerConnection (e.g., maxPacketLifetime) but
   are not reflected in the SDP and do not affect the JSEP state.






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4.1.6.  createOffer

   The createOffer method generates a blob of SDP that contains a
   [RFC3264] offer with the supported configurations for the session,
   including descriptions of the media added to this PeerConnection, the
   codec/RTP/RTCP options supported by this implementation, and any
   candidates that have been gathered by the ICE Agent.  An options
   parameter may be supplied to provide additional control over the
   generated offer.  This options parameter allows an application to
   trigger an ICE restart, for the purpose of reestablishing
   connectivity.

   In the initial offer, the generated SDP will contain all desired
   functionality for the session (functionality that is supported but
   not desired by default may be omitted); for each SDP line, the
   generation of the SDP will follow the process defined for generating
   an initial offer from the document that 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,
   createOffer will generate an offer to modify the current session
   based on any changes that have been made to the session, e.g., adding
   or stopping RtpTransceivers, or requesting an ICE restart.  For each
   existing stream, the generation of each SDP line must follow the
   process defined for generating an updated offer from the RFC that
   specifies the given SDP line.  For each new stream, the generation of
   the SDP must follow the process of generating an initial offer, as
   mentioned above.  If no changes have been made, or for SDP lines that
   are unaffected by the requested changes, the offer will only contain
   the parameters negotiated by the last offer-answer exchange.  The
   exact handling of subsequent offer generation is detailed in
   Section 5.2.2. below.

   Session descriptions generated by createOffer must be immediately
   usable by setLocalDescription; if a system has limited resources
   (e.g. a finite number of decoders), createOffer should return an
   offer that reflects the current state of the system, so that
   setLocalDescription will succeed when it attempts to acquire those
   resources.

   Calling this method may do things such as generate new ICE
   credentials, but does not result in candidate gathering, or cause
   media to start or stop flowing.







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4.1.7.  createAnswer

   The createAnswer method generates a blob of SDP that contains a
   [RFC3264] SDP answer with the supported configuration for the session
   that is compatible with the parameters supplied in the most recent
   call to setRemoteDescription, which MUST have been called prior to
   calling createAnswer.  Like createOffer, the returned blob contains
   descriptions of the media added to this PeerConnection, the
   codec/RTP/RTCP options negotiated for this session, and any
   candidates that have been gathered by the ICE Agent.  An options
   parameter may be supplied to provide additional control over the
   generated answer.

   As an answer, the generated SDP will contain a specific configuration
   that specifies how the media plane should be established; for each
   SDP line, the generation of the SDP must follow the process defined
   for generating an answer from the document that specifies the given
   SDP line.  The exact handling of answer generation is detailed in
   Section 5.3. below.

   Session descriptions generated by createAnswer must be immediately
   usable by setLocalDescription; like createOffer, the returned
   description should reflect the current state of the system.

   Calling this method may do things such as generate new ICE
   credentials, but does not trigger candidate gathering or change media
   state.

4.1.8.  SessionDescriptionType

   Session description objects (RTCSessionDescription) may be of type
   "offer", "pranswer", "answer" or "rollback".  These types provide
   information as to how the description parameter should be parsed, and
   how the media state should be changed.

   "offer" indicates that a description should be parsed as an offer;
   said description may include many possible media configurations.  A
   description used as an "offer" may be applied anytime the
   PeerConnection is in a stable state, or as an update to a previously
   supplied but unanswered "offer".

   "pranswer" indicates that a description should be parsed as an
   answer, but not a final answer, and so should not result in the
   freeing of allocated resources.  It may result in the start of media
   transmission, if the answer does not specify an inactive media
   direction.  A description used as a "pranswer" may be applied as a
   response to an "offer", or an update to a previously sent "pranswer".




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   "answer" indicates that a description should be parsed as an answer,
   the offer-answer exchange should be considered complete, and any
   resources (decoders, candidates) that are no longer needed can be
   released.  A description used as an "answer" may be applied as a
   response to an "offer", or an update to a previously sent "pranswer".

   The only difference between a provisional and final answer is that
   the final answer results in the freeing of any unused resources that
   were allocated as a result of the offer.  As such, the application
   can use some discretion on whether an answer should be applied as
   provisional or final, and can change the type of the session
   description as needed.  For example, in a serial forking scenario, an
   application may receive multiple "final" answers, one from each
   remote endpoint.  The application could choose to accept the initial
   answers as provisional answers, and only apply an answer as final
   when it receives one that meets its criteria (e.g. a live user
   instead of voicemail).

   "rollback" is a special session description type implying that the
   state machine should be rolled back to the previous stable state, as
   described in Section 4.1.8.2.  The contents MUST be empty.

4.1.8.1.  Use of Provisional Answers

   Most web applications will not need to create answers using the
   "pranswer" type.  While it is good practice to send an immediate
   response to an "offer", in order to warm up the session transport and
   prevent media clipping, the preferred handling for a web application
   would be to create and send an "inactive" final answer immediately
   after receiving the offer.  Later, when the called user actually
   accepts the call, the application can create a new "sendrecv" offer
   to update the previous offer/answer pair and start the media flow.
   While this could also be done with an inactive "pranswer", followed
   by a sendrecv "answer", the initial "pranswer" leaves the offer-
   answer exchange open, which means that neither side can send an
   updated offer during this time.

   As an example, consider a typical web application that will set up a
   data channel, an audio channel, and a video channel.  When an
   endpoint receives an 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 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 remote side moving the audio and video to be
   two-way media.  By the time the human has accepted the call and
   triggered the new offer, it is likely that the ICE and DTLS
   handshaking for all the channels will already have finished.




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   Of course, some applications may not be able to perform this double
   offer-answer exchange, particularly ones that are attempting to
   gateway to legacy signaling protocols.  In these cases, "pranswer"
   can still provide the application with a mechanism to warm up the
   transport.

4.1.8.2.  Rollback

   In certain situations it may be desirable to "undo" a change made to
   setLocalDescription or setRemoteDescription.  Consider a case where a
   call is ongoing, and one side wants to change some of the session
   parameters; that side generates an updated offer and then calls
   setLocalDescription.  However, the remote side, either before or
   after setRemoteDescription, decides it does not want to accept the
   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
   stable state and the previous local/remote description.  To support
   this, we introduce the concept of "rollback".

   A rollback discards any proposed changes to the session, returning
   the state machine to the stable state, and setting the pending local
   and/or remote description (see Section 4.1.12 and Section 4.1.14) to
   null.  Any resources or candidates that were allocated by the
   abandoned local description are discarded; any media that is received
   will be processed according to the previous local and remote
   descriptions.  Rollback can only be used to cancel proposed changes;
   there is no support for rolling back from a stable state to a
   previous stable state.  Note that this implies that once the answerer
   has performed setLocalDescription with his answer, this cannot be
   rolled back.

   A rollback will disassociate any RtpTransceivers that were associated
   with m= sections by the application of the rolled-back session
   description (see Section 5.9 and Section 5.8).  This means that some
   RtpTransceivers that were previously associated will no longer be
   associated with any m= section; in such cases, the value of the
   RtpTransceiver's mid property MUST be set to null, and the mapping
   between the transceiver and its m= section index MUST be discarded.
   RtpTransceivers that were created by applying a remote offer that was
   subsequently rolled back MUST be stopped and removed from the
   PeerConnection.  However, a RtpTransceiver MUST NOT be removed if a
   track was attached to the RtpTransceiver via the addTrack method.
   This is so that an application may call addTrack, then call
   setRemoteDescription with an offer, then roll back that offer, then
   call createOffer and have a m= section for the added track appear in
   the generated offer.





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   A rollback is performed by supplying a session description of type
   "rollback" with empty contents to either setLocalDescription or
   setRemoteDescription, depending on which was most recently used (i.e.
   if the new offer was supplied to setLocalDescription, the rollback
   should be done using setLocalDescription as well).

4.1.9.  setLocalDescription

   The setLocalDescription method instructs the PeerConnection to apply
   the supplied session description as its local configuration.  The
   type field indicates whether the description should be processed as
   an offer, provisional answer, or final answer; offers and answers are
   checked differently, using the various rules that exist for each SDP
   line.

   This API changes the local media state; among other things, it sets
   up local resources for receiving and decoding media.  In order to
   successfully handle scenarios where the application wants to offer to
   change from one media format to a different, incompatible format, the
   PeerConnection must be able to simultaneously support use of both the
   current and pending local descriptions (e.g., support the codecs that
   exist in either description).  This dual processing begins when the
   PeerConnection enters the have-local-offer state, and continues until
   setRemoteDescription is called with either a final answer, at which
   point the PeerConnection can fully adopt the pending local
   description, or a rollback, which results in a revert to the current
   local description.

   This API indirectly controls the candidate gathering process.  When a
   local description is supplied, and the number of transports currently
   in use does not match the number of transports needed by the local
   description, the PeerConnection will create transports as needed and
   begin gathering candidates for each transport, using ones from the
   candidate pool if available.

   If setRemoteDescription was previously called with an offer, and
   setLocalDescription is called with an answer (provisional or final),
   and the media directions are compatible, and media is available to
   send, this will result in the starting of media transmission.

4.1.10.  setRemoteDescription

   The setRemoteDescription method instructs the PeerConnection to apply
   the supplied session description as the desired remote configuration.
   As in setLocalDescription, the type field of the description
   indicates how it should be processed.





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   This API changes the local media state; among other things, it sets
   up local resources for sending and encoding media.

   If setLocalDescription was previously called with an offer, and
   setRemoteDescription is called with an answer (provisional or final),
   and the media directions are compatible, and media is available to
   send, this will result in the starting of media transmission.

4.1.11.  currentLocalDescription

   The currentLocalDescription method returns the current negotiated
   local description - i.e., the local description from the last
   successful offer/answer exchange - in addition to any local
   candidates that have been generated by the ICE Agent since the local
   description was set.

   A null object will be returned if an offer/answer exchange has not
   yet been completed.

4.1.12.  pendingLocalDescription

   The pendingLocalDescription method returns a copy of the local
   description currently in negotiation - i.e., a local offer set
   without any corresponding remote answer - in addition to any local
   candidates that have been generated by the ICE Agent since the local
   description was set.

   A null object will be returned if the state of the PeerConnection is
   "stable" or "have-remote-offer".

4.1.13.  currentRemoteDescription

   The currentRemoteDescription method returns a copy of the current
   negotiated remote description - i.e., the remote description from the
   last successful offer/answer exchange - in addition to any remote
   candidates that have been supplied via processIceMessage since the
   remote description was set.

   A null object will be returned if an offer/answer exchange has not
   yet been completed.

4.1.14.  pendingRemoteDescription

   The pendingRemoteDescription method returns a copy of the remote
   description currently in negotiation - i.e., a remote offer set
   without any corresponding local answer - in addition to any remote
   candidates that have been supplied via processIceMessage since the
   remote description was set.



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   A null object will be returned if the state of the PeerConnection is
   "stable" or "have-local-offer".

4.1.15.  canTrickleIceCandidates

   The canTrickleIceCandidates property indicates whether the remote
   side supports receiving trickled candidates.  There are three
   potential values:

   null:  No SDP has been received from the other side, so it is not
      known if it can handle trickle.  This is the initial value before
      setRemoteDescription() is called.

   true:  SDP has been received from the other side indicating that it
      can support trickle.

   false:  SDP has been received from the other side indicating that it
      cannot support trickle.

   As described in Section 3.5.2, JSEP implementations always provide
   candidates to the application individually, consistent with what is
   needed for Trickle ICE.  However, applications can use the
   canTrickleIceCandidates property to determine whether their peer can
   actually do Trickle ICE, i.e., whether it is safe to send an initial
   offer or answer followed later by candidates as they are gathered.
   As "true" is the only value that definitively indicates remote
   Trickle ICE support, an application which compares
   canTrickleIceCandidates against "true" will by default attempt Half
   Trickle on initial offers and Full Trickle on subsequent interactions
   with a Trickle ICE-compatible agent.

4.1.16.  setConfiguration

   The setConfiguration method allows the global configuration of the
   PeerConnection, which was initially set by constructor parameters, to
   be changed during the session.  The effects of this method call
   depend on when it is invoked, and differ depending on which specific
   parameters are changed:

   o  Any changes to the STUN/TURN servers to use affect the next
      gathering phase.  If an ICE gathering phase has already started or
      completed, the 'needs-ice-restart' bit mentioned in Section 3.5.1
      will be set.  This will cause the next call to createOffer to
      generate new ICE credentials, for the purpose of forcing an ICE
      restart and kicking off a new gathering phase, in which the new
      servers will be used.  If the ICE candidate pool has a nonzero
      size, and a local description has not yet been applied, any




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      existing candidates will be discarded, and new candidates will be
      gathered from the new servers.

   o  Any change to the ICE candidate policy affects the next gathering
      phase.  If an ICE gathering phase has already started or
      completed, the 'needs-ice-restart' bit will be set.  Either way,
      changes to the policy have no effect on the candidate pool,
      because pooled candidates are not surfaced to the application
      until a gathering phase occurs, and so any necessary filtering can
      still be done on any pooled candidates.

   o  The ICE candidate pool size MUST NOT be changed after applying a
      local description.  If a local description has not yet been
      applied, any changes to the ICE candidate pool size take effect
      immediately; if increased, additional candidates are pre-gathered;
      if decreased, the now-superfluous candidates are discarded.

   o  The bundle and RTCP-multiplexing policies MUST NOT be changed
      after the construction of the PeerConnection.

   This call may result in a change to the state of the ICE Agent.

4.1.17.  addIceCandidate

   The addIceCandidate method provides a remote candidate to the ICE
   Agent, which, if parsed successfully, will be added to the current
   and/or pending remote description according to the rules defined for
   Trickle ICE.  The pair of MID and ufrag is used to determine the m=
   section and ICE candidate generation to which the candidate belongs.
   If the MID is not present, the m= section index is used to look up
   the locally generated MID (see Section 5.9), which is used in place
   of a supplied MID.  If these values or the candidate string are
   invalid, an error is generated.

   The purpose of the ufrag is to resolve ambiguities when trickle ICE
   is in progress during an ICE restart.  If the ufrag is absent, the
   candidate MUST be assumed to belong to the most recently applied
   remote description.  Connectivity checks will be sent to the new
   candidate.

   This method can also be used to provide an end-of-candidates
   indication to the ICE Agent, as defined in [I-D.ietf-ice-trickle]).
   The MID and ufrag are used as described above to determine the m=
   section and ICE generation for which candidate gathering is complete.
   If the ufrag is not present, then the end-of-candidates indication
   MUST be assumed to apply to the relevant m= section in the most
   recently applied remote description.  If neither the MID nor the m=




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   index is present, then the indication MUST be assumed to apply to all
   m= sections in the most recently applied remote description.

   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
   being established.

4.2.  RtpTransceiver

4.2.1.  stop

   The stop method stops an RtpTransceiver.  This will cause future
   calls to createOffer to generate a zero port for the associated m=
   section.  See below for more details.

4.2.2.  stopped

   The stopped property indicates whether the transceiver has been
   stopped, either by a call to stopTransceiver or by applying an answer
   that rejects the associated m= section.  In either of these cases, it
   is set to "true", and otherwise will be set to "false".

   A stopped RtpTransceiver does not send any outgoing RTP or RTCP or
   process any incoming RTP or RTCP.  It cannot be restarted.

4.2.3.  setDirection

   The setDirection method sets the direction of a transceiver, which
   affects the direction property of the associated m= section on future
   calls to createOffer and createAnswer.

   When creating offers, the transceiver direction is directly reflected
   in the output, even for reoffers.  When creating answers, the
   transceiver direction is intersected with the offered direction, as
   explained in the Section 5.3 section below.

   Note that while setDirection sets the direction property of the
   transceiver immediately (Section 4.2.4), this property does not
   immediately affect whether the transceiver's RtpSender will send or
   its RtpReceiver will receive.  The direction in effect is represented
   by the currentDirection property, which is only updated when an
   answer is applied.

4.2.4.  direction

   The direction property indicates the last value passed into
   setDirection.  If setDirection has never been called, it is set to
   the direction the transceiver was initialized with.



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4.2.5.  currentDirection

   The currentDirection property indicates the last negotiated direction
   for the transceiver's associated m= section.  More specifically, it
   indicates the [RFC3264] directional attribute of the associated m=
   section in the last applied answer, with "send" and "recv" directions
   reversed if it was a remote answer.  For example, if the directional
   attribute for the associated m= section in a remote answer is
   "recvonly", currentDirection is set to "sendonly".

   If an answer that references this transceiver has not yet been
   applied, or if the transceiver is stopped, currentDirection is set to
   null.

4.2.6.  setCodecPreferences

   The setCodecPreferences method sets the codec preferences of a
   transceiver, which in turn affect the presence and order of codecs of
   the associated m= section on future calls to createOffer and
   createAnswer.  Note that setCodecPreferences does not directly affect
   which codec the implementation decides to send.  It only affects
   which codecs the implementation indicates that it prefers to receive,
   via the offer or answer.  Even when a codec is excluded by
   setCodecPreferences, it still may be used to send until the next
   offer/answer exchange discards it.

   The codec preferences of an RtpTransceiver can cause codecs to be
   excluded by subsequent calls to createOffer and createAnswer, in
   which case the corresponding media formats in the associated m=
   section will be excluded.  The codec preferences cannot add media
   formats that would otherwise not be present.  This includes codecs
   that were not negotiated in a previous offer/answer exchange that
   included the transceiver.

   The codec preferences of an RtpTransceiver can also determine the
   order of codecs in subsequent calls to createOffer and createAnswer,
   in which case the order of the media formats in the associated m=
   section will match.  However, the codec preferences cannot change the
   order of the media formats after an answer containing the transceiver
   has been applied.  At this point, codecs can only be removed, not
   reordered.

5.  SDP Interaction Procedures

   This section describes the specific procedures to be followed when
   creating and parsing SDP objects.





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5.1.  Requirements Overview

   JSEP implementations must comply with the specifications listed below
   that govern the creation and processing of offers and answers.

   The first set of specifications is the "mandatory-to-implement" set.
   All implementations must support these behaviors, but may not use all
   of them if the remote side, which may not be a JSEP endpoint, does
   not support them.

   The second set of specifications is the "mandatory-to-use" set.  The
   local JSEP endpoint and any remote endpoint must indicate support for
   these specifications in their session descriptions.

5.1.1.  Implementation Requirements

   Implementations of JSEP MUST conform to [I-D.ietf-rtcweb-rtp-usage].
   This list of mandatory-to-implement specifications is derived from
   the requirements outlined in that document and from
   [I-D.ietf-rtcweb-security-arch].

   R-1   [RFC4566] is the base SDP specification and MUST be
         implemented.

   R-2   [RFC5764] MUST be supported for signaling the UDP/TLS/RTP/SAVPF
         [RFC5764], TCP/DTLS/RTP/SAVPF [RFC7850], "UDP/DTLS/SCTP"
         [I-D.ietf-mmusic-sctp-sdp], and "TCP/DTLS/SCTP"
         [I-D.ietf-mmusic-sctp-sdp] RTP profiles.

   R-3   [RFC5245] MUST be implemented for signaling the ICE credentials
         and candidate lines corresponding to each media stream.  The
         ICE implementation MUST be a Full implementation, not a Lite
         implementation.

   R-4   [RFC5763] MUST be implemented to signal DTLS certificate
         fingerprints.

   R-5   [RFC5888] MUST be implemented for signaling grouping
         information, and MUST be used to identify m= lines via the
         a=mid attribute.

   R-6   [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-7   The bundle mechanism in
         [I-D.ietf-mmusic-sdp-bundle-negotiation] MUST be supported to




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         signal the ability to multiplex RTP streams on a single UDP
         port, in order to avoid excessive use of port number resources.

   R-8   The SDP attributes of "sendonly", "recvonly", "inactive", and
         "sendrecv" from [RFC4566] MUST be implemented to signal
         information about media direction.

   R-9   [RFC5576] MUST be implemented to signal RTP SSRC values and
         grouping semantics.

   R-10  [RFC4585] MUST be implemented to signal RTCP based feedback.

   R-11  [RFC5761] MUST be implemented to signal multiplexing of RTP and
         RTCP.

   R-12  [RFC5506] MUST be implemented to signal reduced-size RTCP
         messages.

   R-13  [RFC4588] MUST be implemented to signal RTX payload type
         associations.

   R-14  [RFC3556] MUST be supported for control of RTCP bandwidth
         limits.

   The SDES SRTP keying mechanism from [RFC4568] MUST NOT be
   implemented, as discussed in [I-D.ietf-rtcweb-security-arch].

   As required by [RFC4566], Section 5.13, JSEP implementations MUST
   ignore unknown attribute (a=) lines.

5.1.2.  Usage Requirements

   All session descriptions handled by JSEP endpoints, both local and
   remote, MUST indicate support for the following specifications.  If
   any of these are absent, this omission MUST be treated as an error.

   U-1  ICE, as specified in [RFC5245], MUST be used.  Note that the
        remote endpoint may use a Lite implementation; implementations
        MUST properly handle remote endpoints which do ICE-Lite.

   U-2  DTLS [RFC6347] or DTLS-SRTP [RFC5763], MUST be used, as
        appropriate for the media type, as specified in
        [I-D.ietf-rtcweb-security-arch]








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5.1.3.  Profile Names and Interoperability

   For media m= sections, JSEP endpoints MUST support both the "UDP/TLS/
   RTP/SAVPF" and "TCP/DTLS/RTP/SAVPF" profiles and MUST indicate one of
   these two profiles for each media m= line they produce in an offer.
   For data m= sections, JSEP endpoints must support both the "UDP/DTLS/
   SCTP" and "TCP/DTLS/SCTP" profiles and MUST indicate one of these two
   profiles for each data m= line they produce in an offer.  Because ICE
   can select either TCP or UDP transport depending on network
   conditions, both advertisements are consistent with ICE eventually
   selecting either either UDP or TCP.

   Unfortunately, in an attempt at compatibility, some endpoints
   generate other profile strings even when they mean to support one of
   these profiles.  For instance, an endpoint might generate "RTP/AVP"
   but supply "a=fingerprint" and "a=rtcp-fb" attributes, indicating its
   willingness to support "(UDP,TCP)/TLS/RTP/SAVPF".  In order to
   simplify compatibility with such endpoints, JSEP endpoints MUST
   follow the following rules when processing the media m= sections in
   an offer:

   o  The profile in any "m=" line in any answer MUST exactly match the
      profile provided in the offer.

   o  Any profile matching the following patterns MUST be accepted:
      "RTP/[S]AVP[F]" and "(UDP/TCP)/TLS/RTP/SAVP[F]"

   o  Because DTLS-SRTP is REQUIRED, the choice of SAVP or AVP has no
      effect; support for DTLS-SRTP is determined by the presence of one
      or more "a=fingerprint" attribute.  Note that lack of an
      "a=fingerprint" attribute will lead to negotiation failure.

   o  The use of AVPF or AVP simply controls the timing rules used for
      RTCP feedback.  If AVPF is provided, or an "a=rtcp-fb" attribute
      is present, assume AVPF timing, i.e., a default value of "trr-
      int=0".  Otherwise, assume that AVPF is being used in an AVP
      compatible mode and use a value of "trr-int=4000".

   o  For data m= sections, JSEP endpoints MUST support receiving the
      "UDP/DTLS/SCTP", "TCP/DTLS/SCTP", or "DTLS/SCTP" (for backwards
      compatibility) profiles.

   Note that re-offers by JSEP endpoints MUST use the correct profile
   strings even if the initial offer/answer exchange used an (incorrect)
   older profile string.






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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 "-".  [RFC3264] requires that the <sess-id> be representable as
      a 64-bit signed integer.  It is RECOMMENDED that the <sess-id> be
      generated as a 64-bit quantity with the high bit being sent to
      zero and the remaining 63 bits being cryptographically random.
      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; to match the "o=" line, a single dash SHOULD be used
      as the session name, e.g. "s=-".  Note that this differs from the
      advice in [RFC4566] which proposes a single space, but as both
      "o=" and "s=" are meaningless, having the same meaningless value
      seems clearer.

   o  Session Information ("i="), URI ("u="), Email Address ("e="),
      Phone Number ("p="), 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".



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   o  An "a=ice-options" line with the "trickle" option MUST be added,
      as specified in [I-D.ietf-ice-trickle], Section 4.

   The next step is to generate m= sections, as specified in [RFC4566]
   Section 5.14.  An m= section is generated for each RtpTransceiver
   that has been added to the PeerConnection, excluding any stopped
   RtpTransceivers.  This is done in the order the RtpTransceivers were
   added to the PeerConnection.

   For each m= section generated for an RtpTransceiver, establish a
   mapping between the transceiver and the index of the generated m=
   section.

   Each m= section, provided it is not marked as bundle-only, MUST
   generate a unique set of ICE credentials and gather its own unique
   set of ICE candidates.  Bundle-only m= sections MUST NOT contain any
   ICE credentials and MUST NOT gather any candidates.

   For DTLS, all m= sections MUST use all the certificate(s) that have
   been specified for the PeerConnection; as a result, they MUST all
   have the same [I-D.ietf-mmusic-4572-update] fingerprint value(s), or
   these value(s) MUST be session-level attributes.

   Each m= section should be generated as specified in [RFC4566],
   Section 5.14.  For the m= line itself, the following rules MUST be
   followed:

   o  The port value is set to the port of the default ICE candidate for
      this m= section, but given that no candidates are available yet,
      the "dummy" port value of 9 (Discard) MUST be used, as indicated
      in [I-D.ietf-ice-trickle], Section 5.1.

   o  To properly indicate use of DTLS, the <proto> field MUST be set to
      "UDP/TLS/RTP/SAVPF", as specified in [RFC5764], Section 8.

   o  If codec preferences have been set for the associated transceiver,
      media formats MUST be generated in the corresponding order, and
      MUST exclude any codecs not present in the codec preferences.

   o  The media formats in the answer MAY include codecs present in the
      offer that were discarded in a previous offer/answer exchange.
      This is necessary for compatibility with third- party call control
      and SIP use cases.

   o  Unless excluded by the above restrictions, the media formats MUST
      include the mandatory audio/video codecs as specified in
      [I-D.ietf-rtcweb-audio](see Section 3) and
      [I-D.ietf-rtcweb-video](see Section 5).



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   The m= line MUST be followed immediately by a "c=" line, as specified
   in [RFC4566], Section 5.7.  Again, as no candidates are available
   yet, the "c=" line must contain the "dummy" value "IN IP4 0.0.0.0",
   as defined in [I-D.ietf-ice-trickle], Section 5.1.

   [I-D.ietf-mmusic-sdp-mux-attributes] groups SDP attributes into
   different categories.  To avoid unnecessary duplication when
   bundling, Section 8.1 of [I-D.ietf-mmusic-sdp-bundle-negotiation]
   specifies that attributes of category IDENTICAL or TRANSPORT should
   not be repeated in bundled m= sections.

   The following attributes, which are of a category other than
   IDENTICAL or TRANSPORT, MUST be included in each m= section:

   o  An "a=mid" line, as specified in [RFC5888], Section 4.  All MID
      values MUST be generated in a fashion that does not leak user
      information, e.g., randomly or using a per-PeerConnection counter,
      and SHOULD be 3 bytes or less, to allow them to efficiently fit
      into the RTP header extension defined in
      [I-D.ietf-mmusic-sdp-bundle-negotiation], Section 14.  Note that
      this does not set the RtpTransceiver mid property, as that only
      occurs when the description is applied.  The generated MID value
      can be considered a "proposed" MID at this point.

   o  A direction attribute which is the same as that of the associated
      transceiver.

   o  For each media format on the m= line, "a=rtpmap" and "a=fmtp"
      lines, as specified in [RFC4566], Section 6, and [RFC3264],
      Section 5.1.

   o  If this m= section is for media with configurable durations of
      media per packet, e.g., audio, an "a=maxptime" line, indicating
      the maximum amount of media, specified in milliseconds, that can
      be encapsulated in each packet, as specified in [RFC4566],
      Section 6.  This value is set to the smallest of the maximum
      duration values across all the codecs included in the m= section.

   o  If this m= section is for video media, and there are known
      limitations on the size of images which can be decoded, an
      "a=imageattr" line, as specified in Section 3.6.

   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 of the primary codec, as specified in [RFC4588],
      Section 8.1.




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   o  For each supported FEC mechanism, "a=rtpmap" and "a=fmtp" lines,
      as specified in [RFC4566], Section 6.  The FEC mechanisms that
      MUST be supported are specified in [I-D.ietf-rtcweb-fec],
      Section 6, and specific usage for each media type is outlined in
      Sections 4 and 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  If the bundle policy for this PeerConnection is set to "max-
      bundle", and this is not the first m= section, or the bundle
      policy is set to "balanced", and this is not the first m= section
      for this media type, an "a=bundle-only" line.

   o  If the RtpTransceiver has a sendrecv or sendonly direction:

      *  An "a=msid" line, as specified in [I-D.ietf-mmusic-msid],
         Section 2.

   o  If the RtpTransceiver has a sendrecv or sendonly direction, and
      the application has specified RID values or has specified more
      than one encoding in the RtpSenders's parameters, an "a=rid" line
      for each encoding specified.  The "a=rid" line is specified in
      [I-D.ietf-mmusic-rid], and its direction MUST be "send".  If the
      application has chosen a RID value, it MUST be used as the rid-
      identifier; otherwise a RID value MUST be generated by the
      implementation.  RID values MUST be generated in a fashion that
      does not leak user information, e.g., randomly or using a per-
      PeerConnection counter, and SHOULD be 3 bytes or less, to allow
      them to efficiently fit into the RTP header extension defined in
      [I-D.ietf-avtext-rid], Section 3.  If no encodings have been
      specified, or only one encoding is specified but without a RID
      value, then no "a=rid" lines are generated.

   o  If the RtpTransceiver has a sendrecv or sendonly direction and
      more than one "a=rid" line has been generated, an "a=simulcast"
      line, with direction "send", as defined in
      [I-D.ietf-mmusic-sdp-simulcast], Section 6.2.  The list of RIDs




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      MUST include all of the RID identifiers used in the "a=rid" lines
      for this m= section.

   The following attributes, which are of category IDENTICAL or
   TRANSPORT, MUST appear only in "m=" sections which either have a
   unique address or which are associated with the bundle-tag.  (In
   initial offers, this means those "m=" sections which do not contain
   an "a=bundle-only" attribute.

   o  "a=ice-ufrag" and "a=ice-pwd" lines, as specified in [RFC5245],
      Section 15.4.

   o  An "a=fingerprint" line for each of the endpoint's certificates,
      as specified in [RFC4572], Section 5; the digest algorithm used
      for the fingerprint MUST match that used in the certificate
      signature.

   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=dtls-id" line, as specified in [I-D.ietf-mmusic-dtls-sdp]
      Section 5.2.

   o  An "a=rtcp" line, as specified in [RFC3605], Section 2.1,
      containing the dummy value "9 IN IP4 0.0.0.0", because no
      candidates have yet been gathered.

   o  An "a=rtcp-mux" line, as specified in [RFC5761], Section 5.1.3.

   o  An "a=rtcp-rsize" line, as specified in [RFC5506], Section 5.

   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 "UDP/DTLS/SCTP"
   [I-D.ietf-mmusic-sctp-sdp].  The "fmt" value MUST be set to "webrtc-
   datachannel" as specified in [I-D.ietf-mmusic-sctp-sdp], Section 4.1.

   Within the data m= section, the "a=mid", "a=ice-ufrag", "a=ice-pwd",
   "a=fingerprint", "a=dtls-id", and "a=setup" lines MUST be included as
   mentioned above, along with an "a=fmtp:webrtc-datachannel" line and
   an "a=sctp-port" line referencing the SCTP port number as defined in
   [I-D.ietf-mmusic-sctp-sdp], Section 4.1.

   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 MUST include the mid identifiers of
   each m= section.  The effect of this is that the browser offers all



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   m= sections as one bundle group.  However, whether the m= sections
   are bundle-only or not depends on the bundle policy.

   The next step is to generate session-level lip sync groups as defined
   in [RFC5888], Section 7.  For each MediaStream referenced by more
   than one RtpTransceiver (by passing those MediaStreams as arguments
   to the addTrack and addTransceiver methods), a group of type "LS"
   MUST be added that contains the mid values for each RtpTransceiver.

   Attributes which SDP permits to either be at the session level or the
   media level SHOULD generally be at the media level even if they are
   identical.  This promotes readability, especially if one of a set of
   initially identical attributes is subsequently changed.

   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.ietf-mmusic-sdp-mux-attributes]
   on how those attributes interact with bundle.

   Note that these requirements are in some cases stricter than those of
   SDP.  Implementations MUST be prepared to accept compliant SDP even
   if it would not conform to the requirements for generating SDP in
   this specification.

5.2.2.  Subsequent Offers

   When createOffer is called a second (or later) time, or is called
   after a local description has already been installed, the processing
   is somewhat different than for an initial offer.

   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, subject to the
   following restriction:

   o  The fields of the "o=" line MUST stay the same except for the
      <session-version> field, which MUST increment by one on each call
      to createOffer if the offer might differ from the output of the



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      previous call to createOffer; implementations MAY opt to increment
      <session-version> on every call.  The value of the generated
      <session-version> is independent of the <session-version> of the
      current local description; in particular, in the case where the
      current version is N, an offer is created and applied with version
      N+1, and then that offer is rolled back so that the current
      version is again N, the next generated offer will still have
      version N+2.

   Note that if the application creates an offer by reading
   currentLocalDescription instead of calling createOffer, the returned
   SDP may be different than when setLocalDescription was originally
   called, due to the addition of gathered ICE candidates, but the
   <session-version> will not have changed.  There are no known
   scenarios in which this causes problems, but if this is a concern,
   the solution is simply to use createOffer to ensure a unique
   <session-version>.

   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, an offer is
   generated by following the steps in the "stable" state above, along
   with these exceptions:

   o  The "s=" and "t=" lines MUST stay the same.

   o  If any RtpTransceiver has been added, and there exists an m=
      section with a zero port in the current local description or the
      current remote description, that m= section MUST be recycled by
      generating an m= section for the added RtpTransceiver as if the m=
      section were being added to the session description, placed at the
      same index as the m= section with a zero port.

   o  If an RtpTransceiver is stopped and is not associated with an m=
      section, an m= section MUST NOT be generated for it.  This
      prevents adding back RtpTransceivers whose m= sections were
      recycled and used for a new RtpTransceiver in a previous offer/
      answer exchange, as described above.

   o  If an RtpTransceiver has been stopped and is associated with an m=
      section, and the m= section is not being recycled as described
      above, an m= section MUST be generated for it with the port set to
      zero and the "a=msid" line removed.

   o  For RtpTransceivers that are not stopped, the "a=msid" line MUST
      stay the same if they are present in the current description.





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   o  Each "m=" and c=" line MUST be filled in with the port, protocol,
      and address of the default candidate for the m= section, as
      described in [RFC5245], Section 4.3.  If ICE checking has already
      completed for one or more candidate pairs and a candidate pair is
      in active use, then that pair MUST be used, even if ICE has not
      yet completed.  Note that this differs from the guidance in
      [RFC5245], Section 9.1.2.2, which only refers to offers created
      when ICE has completed.  In each case, if no RTP candidates have
      yet been gathered, dummy values MUST be used, as described above.

   o  Each "a=mid" line MUST stay the same.

   o  Each "a=ice-ufrag" and "a=ice-pwd" line MUST stay the same, unless
      the ICE configuration has changed (either changes to the supported
      STUN/TURN servers, or the ICE candidate policy), or the
      "IceRestart" option ( Section 5.2.3.1 was specified.  If the m=
      section is bundled into another m= section, it still MUST NOT
      contain any ICE credentials.

   o  If the m= section is not bundled into another m= section, an
      "a=rtcp" attribute line MUST be added with of the default RTCP
      candidate, as indicated in [RFC5761], Section 5.1.3.

   o  If the m= section is not bundled into another m= section, for each
      candidate that has been gathered during the most recent gathering
      phase (see Section 3.5.1), an "a=candidate" line MUST be added, as
      defined in [RFC5245], Section 4.3., paragraph 3.  If candidate
      gathering for the section has completed, an "a=end-of-candidates"
      attribute MUST be added, as described in [I-D.ietf-ice-trickle],
      Section 9.3.  If the m= section is bundled into another m=
      section, both "a=candidate" and "a=end-of-candidates" MUST be
      omitted.

   o  For RtpTransceivers that are still present, the "a=msid" line MUST
      stay the same.

   o  For RtpTransceivers that are still present, the "a=rid" lines MUST
      stay the same.

   o  For RtpTransceivers that are still present, any "a=simulcast" line
      MUST stay the same.

   o  If any RtpTransceiver has been stopped, the port MUST be set to
      zero and the "a=msid" line MUST be removed.

   o  If any RtpTransceiver has been added, and there exists a m=
      section with a zero port in the current local description or the
      current remote description, that m= section MUST be recycled by



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      generating a m= section for the added RtpTransceiver as if the m=
      section were being added to session description, except that
      instead of adding it, the generated m= section replaces the m=
      section with a zero port.  The new m= section MUST contain a new
      MID.

   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.

   In addition, for each non-recycled, 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, if
   any:

   o  The m= line and corresponding "a=rtpmap" and "a=fmtp" lines MUST
      only include codecs present in the most recent answer which have
      not been excluded by the codec preferences of the associated
      transceiver.  Note that non-JSEP endpoints are not subject to
      these restrictions, and might offer media formats that were not
      present in the most recent answer, as specified in [RFC3264],
      Section 8.  Therefore, JSEP endpoints MUST be prepared to receive
      such offers.

   o  The media formats on the m= line MUST be generated in the same
      order as in the current local description.

   o  The RTP header extensions MUST only include those that are present
      in the most recent answer.

   o  The RTCP feedback extensions MUST only include those that are
      present in the most recent answer.

   o  The "a=rtcp" line MUST only be added if the most recent answer did
      not include an "a=rtcp-mux" line.

   o  The "a=rtcp-mux" line MUST only be added if present in the most
      recent answer.

   o  The "a=rtcp-mux-only" line MUST only be added if present in the
      most recent answer.

   o  The "a=rtcp-rsize" line MUST only be added if present in the most
      recent answer.




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   The "a=group:BUNDLE" attribute MUST include the mid identifiers
   specified in the bundle group in the most recent answer, minus any m=
   sections that have been marked as rejected, plus any newly added or
   re-enabled m= sections.  In other words, the bundle attribute must
   contain all m= sections that were previously bundled, as long as they
   are still alive, as well as any new m= sections.

   The "LS" groups are generated in the same way as with initial offers.

5.2.3.  Options Handling

   The createOffer method takes as a parameter an RTCOfferOptions
   object.  Special processing is performed when generating a SDP
   description if the following options are present.

5.2.3.1.  IceRestart

   If the "IceRestart" option 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
   option is specified on an initial offer, it has no effect (since a
   new ICE ufrag and pwd are already generated).  Similarly, if the ICE
   configuration has changed, this option has no effect, since new ufrag
   and pwd attributes will be generated automatically.  This option is
   primarily useful for reestablishing connectivity in cases where
   failures are detected by the application.

5.2.3.2.  VoiceActivityDetection

   If the "VoiceActivityDetection" option is specified, with a value of
   "true", the offer MUST indicate support for silence suppression in
   the audio it receives by including comfort noise ("CN") codecs for
   each offered audio codec, as specified in [RFC3389], Section 5.1,
   except for codecs that have their own internal silence suppression
   support.  For codecs that have their own internal silence suppression
   support, the appropriate fmtp parameters for that codec MUST be
   specified to indicate that silence suppression for received audio is
   desired.  For example, when using the Opus codec, the "usedtx=1"
   parameter would be specified in the offer.  This option allows the
   endpoint to significantly reduce the amount of audio bandwidth it
   receives, at the cost of some fidelity, depending on the quality of
   the remote VAD algorithm.

   If the "VoiceActivityDetection" option is specified, with a value of
   "false", the browser MUST NOT emit "CN" codecs.  For codecs that have
   their own internal silence suppression support, the appropriate fmtp
   parameters for that codec MUST be specified to indicate that silence
   suppression for received audio is not desired.  For example, when



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   using the Opus codec, the "usedtx=0" parameter would be specified in
   the offer.

   Note that setting the "VoiceActivityDetection" parameter when
   generating an offer is a request to receive audio with silence
   suppression.  It has no impact on whether the local endpoint does
   silence suppression for the audio it sends.

   The "VoiceActivityDetection" option does not have any impact on the
   setting of the "vad" value in the signaling of the client to mixer
   audio level header extension described in [RFC6464], Section 4.

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
   JSEP 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 above is not present, this MUST be treated as an
   error, and MUST cause the affected m= sections to be marked as
   rejected.

   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, except that the
   "a=ice-options" line, with the "trickle" option as specified in
   [I-D.ietf-ice-trickle], Section 4, is only included if such an option
   was present in the offer.

   The next step is to generate session-level lip sync groups as defined
   in [RFC5888], Section 7.  For each group of type "LS" present in the
   offer, determine which of the local RtpTransceivers identified by
   that group's mid values reference a common local MediaStream (as
   specified in the addTrack and addTransceiver methods).  If at least
   two such RtpTransceivers exist, a group of type "LS" with the mid
   values of these RtpTransceivers MUST be added.  Otherwise, this



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   indicates a difference of opinion between the offerer and answerer
   regarding lip sync status, and as such, the offered group MUST be
   ignored and no corresponding "LS" group generated.

   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.

   The next step is to go through each offered m= section.  Each offered
   m= section will have an associated RtpTransceiver, as described in
   Section 5.9.  If there are more RtpTransceivers than there are m=
   sections, the unmatched RtpTransceivers will need to be associated in
   a subsequent offer.

   For each offered m= section, if any of the following conditions are
   true, 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 further processing for this m= section can
   be skipped:

   o  The associated RtpTransceiver has been stopped.

   o  No supported codec is present in the offer.

   o  The bundle policy is "max-bundle", and this is not the first m=
      section or in the same bundle group as the first m= section.

   o  The bundle policy is "balanced", and this is not the first m=
      section for this media type or in the same bundle group as the
      first m= section for this media type.

   Otherwise, each m= section in the answer should then be generated as
   specified in [RFC3264], Section 6.1.  For the m= line itself, the
   following rules must be followed:

   o  The port value would normally be set to the port of the default
      ICE candidate for this m= section, but given that no candidates
      are available yet, the "dummy" port value of 9 (Discard) MUST be
      used, as indicated in [I-D.ietf-ice-trickle], Section 5.1.

   o  The <proto> field MUST be set to exactly match the <proto> field
      for the corresponding m= line in the offer.

   o  If codec preferences have been set for the associated transceiver,
      media formats MUST be generated in the corresponding order, and
      MUST exclude any codecs not present in the codec preferences or



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      not present in the offer.  Note that non-JSEP endpoints are not
      subject to this restriction, and might add media formats in the
      answer that are not present in the offer, as specified in
      [RFC3264], Section 6.1.  Therefore, JSEP endpoints MUST be
      prepared to receive such answers.

   o  Unless excluded by the above restrictions, the media formats MUST
      include the mandatory audio/video codecs as specified in
      [I-D.ietf-rtcweb-audio](see Section 3) and
      [I-D.ietf-rtcweb-video](see Section 5).

   The m= line MUST be followed immediately by a "c=" line, as specified
   in [RFC4566], Section 5.7.  Again, as no candidates are available
   yet, the "c=" line must contain the "dummy" value "IN IP4 0.0.0.0",
   as defined in [I-D.ietf-ice-trickle], Section 5.1.

   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 contain the following attributes (which are of attribute
   types other than IDENTICAL and TRANSPORT):

   o  If and only 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  A direction attribute, determined by applying the rules regarding
      the offered direction specified in [RFC3264], Section 6.1, and
      then intersecting with the direction of the associated
      RtpTransceiver.  For example, in the case where an m= section is
      offered as "sendonly", and the local transceiver is set to
      "sendrecv", the result in the answer is a "recvonly" direction.

   o  For each media format on the m= line, "a=rtpmap" and "a=fmtp"
      lines, as specified in [RFC4566], Section 6, and [RFC3264],
      Section 6.1.

   o  If this m= section is for media with configurable durations of
      media per packet, e.g., audio, an "a=maxptime" line, as described
      in Section 5.2.

   o  If this m= section is for video media, and there are known
      limitations on the size of images which can be decoded, an
      "a=imageattr" line, as specified in Section 3.6.

   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



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      "a=fmtp" line that references the payload type of the primary
      codec, as specified in [RFC4588], Section 8.1.

   o  For each supported FEC mechanism, "a=rtpmap" and "a=fmtp" lines,
      as specified in [RFC4566], Section 6.  The FEC mechanisms that
      MUST be supported are specified in [I-D.ietf-rtcweb-fec],
      Section 6, and specific usage for each media type is outlined in
      Sections 4 and 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 the RtpTransceiver has a sendrecv or sendonly direction:

      *  An "a=msid" line, as specified in [I-D.ietf-mmusic-msid],
         Section 2.

   Each m= section which is not bundled into another m= section, MUST
   contain the following attributes (which are of category IDENTICAL or
   TRANSPORT):

   o  "a=ice-ufrag" and "a=ice-pwd" lines, as specified in [RFC5245],
      Section 15.4.

   o  An "a=fingerprint" line for each of the endpoint's certificates,
      as specified in [RFC4572], Section 5; the digest algorithm used
      for the fingerprint MUST match that used in the certificate
      signature.

   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  An "a=dtls-id" line, as specified in [I-D.ietf-mmusic-dtls-sdp]
      Section 5.3.





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   o  If present in the offer, an "a=rtcp-mux" line, as specified in
      [RFC5761], Section 5.1.3.  Otherwise, an "a=rtcp" line, as
      specified in [RFC3605], Section 2.1, containing the dummy value "9
      IN IP4 0.0.0.0" (because no candidates have yet been gathered).

   o  If present in the offer, an "a=rtcp-rsize" line, as specified in
      [RFC5506], Section 5.

   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> and "fmt" fields MUST be set to exactly
   match the fields in the offer.

   Within the data m= section, the "a=mid", "a=ice-ufrag", "a=ice-pwd",
   "a=candidate", "a=fingerprint", "a=dtls-id", and "a=setup" lines MUST
   be included under the conditions described above, along with an
   "a=fmtp:webrtc-datachannel" line and an "a=sctp-port" line
   referencing the SCTP port number as defined in
   [I-D.ietf-mmusic-sctp-sdp], Section 4.1.

   If "a=group" attributes with semantics of "BUNDLE" are offered,
   corresponding session-level "a=group" attributes MUST be added as
   specified in [RFC5888].  These attributes MUST have semantics
   "BUNDLE", and MUST include the all mid identifiers from the offered
   bundle groups that have not been rejected.  Note that regardless of
   the presence of "a=bundle-only" in the offer, no m= sections in the
   answer should have an "a=bundle-only" line.

   Attributes that are common between all m= sections MAY be moved to
   session-level, if explicitly defined to be valid at session-level.

   The attributes prohibited in the creation of offers are also
   prohibited in the creation of answers.

5.3.2.  Subsequent Answers

   When createAnswer is called a second (or later) time, or is called
   after a local description has already been installed, the processing
   is somewhat different than for an initial answer.

   If the initial answer was not applied using setLocalDescription,
   meaning the PeerConnection is still in the "have-remote-offer" state,
   the steps for generating an initial answer should be followed,
   subject to the following restriction:

   o  The fields of the "o=" line MUST stay the same except for the
      <session-version> field, which MUST increment if the session




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      description changes in any way from the previously generated
      answer.

   If any session description was previously supplied to
   setLocalDescription, an answer is generated by following the steps in
   the "have-remote-offer" state above, along with these exceptions:

   o  The "s=" and "t=" lines MUST stay the same.

   o  Each "m=" and c=" line MUST be filled in with the port and address
      of the default candidate for the m= section, as described in
      [RFC5245], Section 4.3.  Note, however, that the m= line protocol
      need not match the default candidate, because this protocol value
      must instead match what was supplied in the offer, as described
      above.

   o  The media formats on the m= line MUST be generated in the same
      order as in the current local description.

   o  Each "a=ice-ufrag" and "a=ice-pwd" line MUST stay the same, unless
      the m= section is restarting, in which case new ICE credentials
      must be created as specified in [RFC5245], Section 9.2.1.1.  If
      the m= section is bundled into another m= section, it still MUST
      NOT contain any ICE credentials.

   o  Each "a=setup" line MUST use an "active" or "passive" role value
      consistent with the existing DTLS association, if the association
      is being continued by the offerer.

   o  If the m= section is not bundled into another m= section and RTCP
      multiplexing is not active, an "a=rtcp" attribute line MUST be
      filled in with the port and address of the default RTCP candidate.
      If no RTCP candidates have yet been gathered, dummy values MUST be
      used, as described in the initial answer section above.

   o  If the m= section is not bundled into another m= section, for each
      candidate that has been gathered during the most recent gathering
      phase (see Section 3.5.1), an "a=candidate" line MUST be added, as
      defined in [RFC5245], Section 4.3., paragraph 3.  If candidate
      gathering for the section has completed, an "a=end-of-candidates"
      attribute MUST be added, as described in [I-D.ietf-ice-trickle],
      Section 9.3.  If the m= section is bundled into another m=
      section, both "a=candidate" and "a=end-of-candidates" MUST be
      omitted.

   o  For RtpTransceivers that are not stopped, the "a=msid" line MUST
      stay the same.




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5.3.3.  Options Handling

   The createAnswer method takes as a parameter an RTCAnswerOptions
   object.  The set of parameters for RTCAnswerOptions is different than
   those supported in RTCOfferOptions; the IceRestart option is
   unnecessary, as ICE credentials will automatically be changed for all
   m= sections where the offerer chose to perform ICE restart.

   The following options are supported in RTCAnswerOptions.

5.3.3.1.  VoiceActivityDetection

   Silence suppression in the answer is handled as described in
   Section 5.2.3.2, with one exception: if support for silence
   suppression was not indicated in the offer, the
   VoiceActivityDetection parameter has no effect, and the answer should
   be generated as if VoiceActivityDetection was set to false.  This is
   done on a per-codec basis (e.g., if the offerer somehow offered
   support for CN but set "usedtx=0" for Opus, setting
   VoiceActivityDetection to true would result in an answer with CN
   codecs and "usedtx=0").

5.4.  Modifying an Offer or Answer

   The SDP returned from createOffer or createAnswer MUST NOT be changed
   before passing it to setLocalDescription.  If precise control over
   the SDP is needed, the aforementioned createOffer/createAnswer
   options or RtpTransceiver APIs MUST be used.

   Note that the application MAY modify the SDP to reduce the
   capabilities in the offer it sends to the far side (post-
   setLocalDescription) or the offer that it installs from the far side
   (pre-setRemoteDescription), as long as it remains a valid SDP offer
   and specifies a subset of what was in the original offer.  This is
   safe because the answer is not permitted to expand capabilities, and
   therefore will just respond to what is present in the offer.

   The application SHOULD NOT modify the SDP in the answer it transmits,
   as the answer contains the negotiated capabilities, and this can
   cause the two sides to have different ideas about what exactly was
   negotiated.

   As always, the application is solely responsible for what it sends to
   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
   that all SDP is well-formed; however, one should be able to assume
   that any implementation of this specification will be able to




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   process, as a remote offer or answer, unmodified SDP coming from any
   other implementation of this specification.

5.5.  Processing a Local Description

   When a SessionDescription is supplied to setLocalDescription, the
   following steps MUST be performed:

   o  First, the type of the SessionDescription is checked against the
      current state of the PeerConnection:

      *  If the type is "offer", the PeerConnection state MUST be either
         "stable" or "have-local-offer".

      *  If the type is "pranswer" or "answer", the PeerConnection state
         MUST be either "have-remote-offer" or "have-local-pranswer".

   o  If the type is not correct for the current state, processing MUST
      stop and an error MUST be returned.

   o  Next, the SessionDescription is parsed into a data structure, as
      described in the Section 5.7 section below.  If parsing fails for
      any reason, processing MUST stop and an error MUST be returned.

   o  Finally, the parsed SessionDescription is applied as described in
      the Section 5.8 section below.

5.6.  Processing a Remote Description

   When a SessionDescription is supplied to setRemoteDescription, the
   following steps MUST be performed:

   o  First, the type of the SessionDescription is checked against the
      current state of the PeerConnection:

      *  If the type is "offer", the PeerConnection state MUST be either
         "stable" or "have-remote-offer".

      *  If the type is "pranswer" or "answer", the PeerConnection state
         MUST be either "have-local-offer" or "have-remote-pranswer".

   o  If the type is not correct for the current state, processing MUST
      stop and an error MUST be returned.

   o  Next, the SessionDescription is parsed into a data structure, as
      described in the Section 5.7 section below.  If parsing fails for
      any reason, processing MUST stop and an error MUST be returned.




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   o  Finally, the parsed SessionDescription is applied as described in
      the Section 5.9 section below.

5.7.  Parsing a Session Description

   When a SessionDescription of any type is supplied to setLocal/
   RemoteDescription, the implementation must parse it and reject it if
   it is invalid.  The exact details of this process are explained
   below.

   The SDP contained in the session description object consists of a
   sequence of text lines, each containing a key-value expression, as
   described in [RFC4566], Section 5.  The SDP is read, line-by-line,
   and converted to a data structure that contains the deserialized
   information.  However, SDP allows many types of lines, not all of
   which are relevant to JSEP applications.  For each line, the
   implementation will first ensure it is syntactically correct
   according to its defining ABNF, check that it conforms to [RFC4566]
   and [RFC3264] semantics, and then either parse and store or discard
   the provided value, as described below.

   If any line is not well-formed, or cannot be parsed as described, the
   parser MUST stop with an error and reject the session description,
   even if the value is to be discarded.  This ensures that
   implementations do not accidentally misinterpret ambiguous SDP.

5.7.1.  Session-Level Parsing

   First, the session-level lines are checked and parsed.  These lines
   MUST occur in a specific order, and with a specific syntax, as
   defined in [RFC4566], Section 5.  Note that while the specific line
   types (e.g. "v=", "c=") MUST occur in the defined order, lines of the
   same type (typically "a=") can occur in any order, and their ordering
   is not meaningful.

   The following non-attribute lines are not meaningful in the JSEP
   context and MAY be discarded once they have been checked.

      The "c=" line MUST be checked for syntax but its value is not
      used.  This supersedes the guidance in [RFC5245], Section 6.1, to
      use "ice-mismatch" to indicate mismatches between "c=" and the
      candidate lines; because JSEP always uses ICE, "ice-mismatch" is
      not useful in this context.

      The "i=", "u=", "e=", "p=", "t=", "r=", "z=", and "k=" lines are
      not used by this specification; they MUST be checked for syntax
      but their values are not used.




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   The remaining non-attribute lines are processed as follows:

      The "v=" line MUST have a version of 0, as specified in [RFC4566],
      Section 5.1.

      The "o=" line MUST be parsed as specified in [RFC4566],
      Section 5.2.

      The "b=" line, if present, MUST be parsed as specified in
      [RFC4566], Section 5.8, and the bwtype and bandwidth values
      stored.

   Finally, the attribute lines are processed.  Specific processing MUST
   be applied for the following session-level attribute ("a=") lines:

   o  Any "a=group" lines are parsed as specified in [RFC5888],
      Section 5, and the group's semantics and mids are stored.

   o  If present, a single "a=ice-lite" line is parsed as specified in
      [RFC5245], Section 15.3, and a value indicating the presence of
      ice-lite is stored.

   o  If present, a single "a=ice-ufrag" line is parsed as specified in
      [RFC5245], Section 15.4, and the ufrag value is stored.

   o  If present, a single "a=ice-pwd" line is parsed as specified in
      [RFC5245], Section 15.4, and the password value is stored.

   o  If present, a single "a=ice-options" line is parsed as specified
      in [RFC5245], Section 15.5, and the set of specified options is
      stored.

   o  Any "a=fingerprint" lines are parsed as specified in [RFC4572],
      Section 5, and the set of fingerprint and algorithm values is
      stored.

   o  If present, a single "a=setup" line is parsed as specified in
      [RFC4145], Section 4, and the setup value is stored.

   o  If present, a single "a=dtls-id" line is parsed as specified in
      [I-D.ietf-mmusic-dtls-sdp] Section 5, and the dtls-id value is
      stored.

   o  Any "a=extmap" lines are parsed as specified in [RFC5285],
      Section 5, and their values are stored.

   Once all the session-level lines have been parsed, processing
   continues with the lines in media sections.



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5.7.2.  Media Section Parsing

   Like the session-level lines, the media session lines MUST occur in
   the specific order and with the specific syntax defined in [RFC4566],
   Section 5.

   The "m=" line itself MUST be parsed as described in [RFC4566],
   Section 5.14, and the media, port, proto, and fmt values stored.

   Following the "m=" line, specific processing MUST be applied for the
   following non-attribute lines:

   o  As with the "c=" line at the session level, the "c=" line MUST be
      parsed according to [RFC4566], Section 5.7, but its value is not
      used.

   o  The "b=" line, if present, MUST be parsed as specified in
      [RFC4566], Section 5.8, and the bwtype and bandwidth values
      stored.

   Specific processing MUST also be applied for the following attribute
   lines:

   o  If present, a single "a=ice-ufrag" line is parsed as specified in
      [RFC5245], Section 15.4, and the ufrag value is stored.

   o  If present, a single "a=ice-pwd" line is parsed as specified in
      [RFC5245], Section 15.4, and the password value is stored.

   o  If present, a single "a=ice-options" line is parsed as specified
      in [RFC5245], Section 15.5, and the set of specified options is
      stored.

   o  Any "a=candidate" attributes MUST be parsed as specified in
      [RFC5245], Section 15.1, and their values stored.

   o  Any "a=remote-candidates" attributes MUST be parsed as specified
      in [RFC5245], Section 15.2, but their values are ignored.

   o  If present, a single "a=end-of-candidates" attribute MUST be
      parsed as specified in [I-D.ietf-ice-trickle], Section 8.2, and
      its presence or absence flagged and stored.

   o  Any "a=fingerprint" lines are parsed as specified in [RFC4572],
      Section 5, and the set of fingerprint and algorithm values is
      stored.





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   If the "m=" proto value indicates use of RTP, as described in the
   Section 5.1.3 section above, the following attribute lines MUST be
   processed:

   o  The "m=" fmt value MUST be parsed as specified in [RFC4566],
      Section 5.14, and the individual values stored.

   o  Any "a=rtpmap" or "a=fmtp" lines MUST be parsed as specified in
      [RFC4566], Section 6, and their values stored.

   o  If present, a single "a=ptime" line MUST be parsed as described in
      [RFC4566], Section 6, and its value stored.

   o  If present, a single "a=maxptime" line MUST be parsed as described
      in [RFC4566], Section 6, and its value stored.

   o  If present, a single direction attribute line (e.g.  "a=sendrecv")
      MUST be parsed as described in [RFC4566], Section 6, and its value
      stored.

   o  Any "a=ssrc" or "a=ssrc-group" attributes MUST be parsed as
      specified in [RFC5576], Sections 4.1-4.2, and their values stored.

   o  Any "a=extmap" attributes MUST be parsed as specified in
      [RFC5285], Section 5, and their values stored.

   o  Any "a=rtcp-fb" attributes MUST be parsed as specified in
      [RFC4585], Section 4.2., and their values stored.

   o  If present, a single "a=rtcp-mux" attribute MUST be parsed as
      specified in [RFC5761], Section 5.1.3, and its presence or absence
      flagged and stored.

   o  If present, a single "a=rtcp-mux-only" attribute MUST be parsed as
      specified in [I-D.ietf-mmusic-mux-exclusive], Section 3, and its
      presence or absence flagged and stored.

   o  If present, a single "a=rtcp-rsize" attribute MUST be parsed as
      specified in [RFC5506], Section 5, and its presence or absence
      flagged and stored.

   o  If present, a single "a=rtcp" attribute MUST be parsed as
      specified in [RFC3605], Section 2.1, but its value is ignored, as
      this information is superfluous when using ICE.

   o  If present, a single "a=msid" attribute MUST be parsed as
      specified in [I-D.ietf-mmusic-msid], Section 3.2, and its value
      stored.



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   o  Any "a=imageattr" attributes MUST be parsed as specified in
      [RFC6236], Section 3, and their values stored.

   o  Any "a=rid" lines MUST be parsed as specified in
      [I-D.ietf-mmusic-rid], Section 10, and their values stored.

   o  If present, a single "a=simulcast" line MUST be parsed as
      specified in [I-D.ietf-mmusic-sdp-simulcast], and its values
      stored.

   Otherwise, if the "m=" proto value indicates use of SCTP, the
   following attribute lines MUST be processed:

   o  The "m=" fmt value MUST be parsed as specified in
      [I-D.ietf-mmusic-sctp-sdp], Section 4.3, and the application
      protocol value stored.

   o  An "a=sctp-port" attribute MUST be present, and it MUST be parsed
      as specified in [I-D.ietf-mmusic-sctp-sdp], Section 5.2, and the
      value stored.

   o  If present, a single "a=max-message-size" attribute MUST be parsed
      as specified in [I-D.ietf-mmusic-sctp-sdp], Section 6, and the
      value stored.  Otherwise, use the specified default.

5.7.3.  Semantics Verification

   Assuming parsing completes successfully, the parsed description is
   then evaluated to ensure internal consistency as well as proper
   support for mandatory features.  Specifically, the following checks
   are performed:

   o  For each m= section, valid values for each of the mandatory-to-use
      features enumerated in Section 5.1.2 MUST be present.  These
      values MAY either be present at the media level, or inherited from
      the session level.

      *  ICE ufrag and password values, which MUST comply with the size
         limits specified in [RFC5245], Section 15.4.

      *  dtls-id value, which MUST be set according to
         [I-D.ietf-mmusic-dtls-sdp] Section 5.  If this is a re-offer
         and the dtls-id value is different from that presently in use,
         the DTLS connection is not being continued and the remote
         description MUST be part of an ICE restart, together with new
         ufrag and password values.  If this is an answer, the dtls-id
         value, if present, MUST be the same as in the offer.




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      *  DTLS setup value, which MUST be set according to the rules
         specified in [RFC5763], Section 5 and MUST be consistent with
         the selected role of the current DTLS connection, if one exists
         and is being continued.

      *  DTLS fingerprint values, where at least one fingerprint MUST be
         present.

   o  All RID values referenced in an "a=simulcast" line MUST exist as
      "a=rid" lines.

   o  Each m= section is also checked to ensure prohibited features are
      not used.  If this is a local description, the "ice-lite"
      attribute MUST NOT be specified.

   o  If the RTP/RTCP multiplexing policy is "require", each m= section
      MUST contain an "a=rtcp-mux" attribute.

   If this session description is of type "pranswer" or "answer", the
   following additional checks are applied:

   o  The session description must follow the rules defined in
      [RFC3264], Section 6, including the requirement that the number of
      m= sections MUST exactly match the number of m= sections in the
      associated offer.

   o  For each m= section, the media type and protocol values MUST
      exactly match the media type and protocol values in the
      corresponding m= section in the associated offer.

   If any of the preceding checks failed, processing MUST stop and an
   error MUST be returned.

5.8.  Applying a Local Description

   The following steps are performed at the media engine level to apply
   a local description.

   First, the parsed parameters are checked to ensure that they are
   identical to those generated in the last call to createOffer/
   createAnswer, and thus have not been altered, as discussed in
   Section 5.4; otherwise, processing MUST stop and an error MUST be
   returned.

   Next, media sections are processed.  For each media section, the
   following steps MUST be performed; if any parameters are out of
   bounds, or cannot be applied, processing MUST stop and an error MUST
   be returned.



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   o  If this media section is new, begin gathering candidates for it,
      as defined in [RFC5245], Section 4.1.1, unless it has been marked
      as bundle-only.

   o  Or, if the ICE ufrag and password values have changed, and it has
      not been marked as bundle-only, trigger the ICE Agent to start an
      ICE restart, and begin gathering new candidates for the media
      section as described in [RFC5245], Section 9.1.1.1.  If this
      description is an answer, also start checks on that media section
      as defined in [RFC5245], Section 9.3.1.1.

   o  If the media section proto value indicates use of RTP:

      *  If there is no RtpTransceiver associated with this m= section
         (which will only happen when applying an offer), find one and
         associate it with this m= section according to the following
         steps:

         +  Find the RtpTransceiver that corresponds to this m= section,
            using the mapping between transceivers and m= section
            indices established when creating the offer.

         +  Set the value of this RtpTransceiver's mid property to the
            MID of the m= section.

      *  If RTCP mux is indicated, prepare to demux RTP and RTCP from
         the RTP ICE component, as specified in [RFC5761],
         Section 5.1.3.  If RTCP mux is not indicated, but was
         previously negotiated, i.e., the RTCP ICE component no longer
         exists, this MUST result in an error.

      *  For each specified RTP header extension, establish a mapping
         between the extension ID and URI, as described in section 6 of
         [RFC5285].  If any indicated RTP header extension is not
         supported, this MUST result in an error.

      *  If the MID header extension is supported, prepare to demux RTP
         streams intended for this media section based on the MID header
         extension, as described in
         [I-D.ietf-mmusic-sdp-bundle-negotiation], Section 14.

      *  For each specified media format, establish a mapping between
         the payload type and the actual media format, as described in
         [RFC3264], Section 6.1.  If any indicated media format is not
         supported, this MUST result in an error.

      *  For each specified "rtx" media format, establish a mapping
         between the RTX payload type and its associated primary payload



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         type, as described in [RFC4588], Sections 8.6 and 8.7.  If any
         referenced primary payload types are not present, this MUST
         result in an error.

      *  If the directional attribute is of type "sendrecv" or
         "recvonly", enable receipt and decoding of media.

   Finally, if this description is of type "pranswer" or "answer",
   follow the processing defined in the Section 5.10 section below.

5.9.  Applying a Remote Description

   If the answer contains any "a=ice-options" attributes where "trickle"
   is listed as an attribute, update the PeerConnection canTrickle
   property to be true.  Otherwise, set this property to false.

   The following steps are performed at the media engine level to apply
   a remote description.

   The following steps MUST be performed for attributes at the session
   level; if any parameters are out of bounds, or cannot be applied,
   processing MUST stop and an error MUST be returned.

   o  For any specified "CT" bandwidth value, set this as the limit for
      the maximum total bitrate for all m= sections, as specified in
      Section 5.8 of [RFC4566].  Within this overall limit, the
      implementation can dynamically decide how to best allocate the
      available bandwidth between m= sections, respecting any specific
      limits that have been specified for individual m= sections.

   o  For any specified "RR" or "RS" bandwidth values, handle as
      specified in [RFC3556], Section 2.

   o  Any "AS" bandwidth value MUST be ignored, as the meaning of this
      construct at the session level is not well defined.

   For each media section, the following steps MUST be performed; if any
   parameters are out of bounds, or cannot be applied, processing MUST
   stop and an error MUST be returned.

   o  If the ICE ufrag or password changed from the previous remote
      description, then an ICE restart is needed, as described in
      Section 9.1.1.1 of [RFC5245] If the description is of type
      "offer", mark that an ICE restart is needed.  If the description
      is of type "answer" and the current local description is also an
      ICE restart, then signal the ICE agent to begin checks as
      described in Section 9.3.1.1 of [RFC5245].  An answer MUST change




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      the ufrag and password in an answer if and only if ICE is
      restarting, as described in Section 9.2.1.1 of [RFC5245].

   o  Configure the ICE components associated with this media section to
      use the supplied ICE remote ufrag and password for their
      connectivity checks.

   o  Pair any supplied ICE candidates with any gathered local
      candidates, as described in Section 5.7 of [RFC5245] and start
      connectivity checks with the appropriate credentials.

   o  If an "a=end-of-candidates" attribute is present, process the end-
      of-candidates indication as described in [I-D.ietf-ice-trickle]
      Section 11.

   o  If the media section proto value indicates use of RTP:

      *  If the m= section is being recycled (see Section 5.2.2),
         dissociate the currently associated RtpTransceiver by setting
         its mid property to null, and discard the mapping between the
         transceiver and its m= section index.

      *  If the m= section is not associated with any RtpTransceiver
         (possibly because it was dissociated in the previous step),
         either find an RtpTransceiver or create one according to the
         following steps:

         +  If the m= section is sendrecv or recvonly, and there are
            RtpTransceivers of the same type that were added to the
            PeerConnection by addTrack and are not associated with any
            m= section and are not stopped, find the first (according to
            the canonical order described in Section 5.2.1) such
            RtpTransceiver.

         +  If no RtpTransceiver was found in the previous step, create
            one with a recvonly direction.

         +  Associate the found or created RtpTransceiver with the m=
            section by setting the value of the RtpTransceiver's mid
            property to the MID of the m= section, and establish a
            mapping between the transceiver and the index of the m=
            section.  If the m= section does not include a MID (i.e.,
            the remote endpoint does not support the MID extension),
            generate a value for the RtpTransceiver mid property,
            following the guidance for "a=mid" mentioned in
            Section 5.2.1.





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      *  For each specified media format that is also supported by the
         local implementation, establish a mapping between the specified
         payload type and the media format, as described in [RFC3264],
         Section 6.1.  Specifically, this means that the implementation
         records the payload type to be used in outgoing RTP packets
         when sending each specified media format, as well as the
         relative preference for each format that is indicated in their
         ordering.  If any indicated media format is not supported by
         the local implementation, it MUST be ignored.

      *  For each specified "rtx" media format, establish a mapping
         between the RTX payload type and its associated primary payload
         type, as described in [RFC4588], Section 4.  If any referenced
         primary payload types are not present, this MUST result in an
         error.

      *  For each specified fmtp parameter that is supported by the
         local implementation, enable them on the associated media
         formats.

      *  For each specified RTP header extension that is also supported
         by the local implementation, establish a mapping between the
         extension ID and URI, as described in [RFC5285], Section 5.
         Specifically, this means that the implementation records the
         extension ID to be used in outgoing RTP packets when sending
         each specified header extension.  If any indicated RTP header
         extension is not supported by the local implementation, it MUST
         be ignored.

      *  For each specified RTCP feedback mechanism that is supported by
         the local implementation, enable them on the associated media
         formats.

      *  For any specified "TIAS" bandwidth value, set this value as a
         constraint on the maximum RTP bitrate to be used when sending
         media, as specified in [RFC3890].  If a "TIAS" value is not
         present, but an "AS" value is specified, generate a "TIAS"
         value using this formula:

         TIAS = AS * 1000 * 0.95 - 50 * 40 * 8

         The 50 is based on 50 packets per second, the 40 is based on an
         estimate of total header size, the 1000 changes the unit from
         kbps to bps (as required by TIAS), and the 0.95 is to allocate
         5% to RTCP.  "TIAS" is used in preference to "AS" because it
         provides more accurate control of bandwidth.





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      *  For any "RR" or "RS" bandwidth values, handle as specified in
         [RFC3556], Section 2.

      *  Any specified "CT" bandwidth value MUST be ignored, as the
         meaning of this construct at the media level is not well
         defined.

      *  If the media section is of type audio:

         +  For each specified "CN" media format, enable DTX for all
            supported media formats with the same clockrate, as
            described in [RFC3389], Section 5, except for formats that
            have their own internal DTX mechanisms.  DTX for such
            formats (e.g., Opus) is controlled via fmtp parameters, as
            discussed in Section 5.2.3.2.

         +  For each specified "telephone-event" media format, enable
            DTMF transmission for all supported media formats with the
            same clockrate, as described in [RFC4733], Section 2.5.1.2.
            If the application attempts to transmit DTMF when using a
            media format that does not have a corresponding telephone-
            event format, this MUST result in an error.

         +  For any specified "ptime" value, configure the available
            media formats to use the specified packet size.  If the
            specified size is not supported for a media format, use the
            next closest value instead.

   Finally, if this description is of type "pranswer" or "answer",
   follow the processing defined in the Section 5.10 section below.

5.10.  Applying an Answer

   In addition to the steps mentioned above for processing a local or
   remote description, the following steps are performed when processing
   a description of type "pranswer" or "answer".

   For each media section, the following steps MUST be performed:

   o  If the media section has been rejected (i.e. port is set to zero
      in the answer), stop any reception or transmission of media for
      this section, and, unless a non-rejected media section is bundled
      with this media section, discard any associated ICE components, as
      described in Section 9.2.1.3 of [RFC5245].

   o  If the remote DTLS fingerprint has been changed or the dtls-id has
      changed, tear down the DTLS connection.  If a DTLS connection
      needs to be torn down but the answer does not indicate an ICE



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      restart, an error MUST be generated.  If an ICE restart is
      performed without a change in dtls-id or fingerprint, then the
      same DTLS connection is continued over the new ICE channel.

   o  If no valid DTLS connection exists, prepare to start a DTLS
      connection, using the specified roles and fingerprints, on any
      underlying ICE components, once they are active.

   o  If the media section proto value indicates use of RTP:

      *  If the media section references any media formats, RTP header
         extensions, or RTCP feedback mechanisms that were not present
         in the corresponding media section in the offer, this indicates
         a negotiation problem and MUST result in an error.

      *  If the media section has RTCP mux enabled, discard the RTCP ICE
         component, if one exists, and begin or continue muxing RTCP
         over the RTP ICE component, as specified in [RFC5761],
         Section 5.1.3.  Otherwise, prepare to transmit RTCP over the
         RTCP ICE component; if no RTCP ICE component exists, because
         RTCP mux was previously enabled, this MUST result in an error.

      *  If the media section has reduced-size RTCP enabled, configure
         the RTCP transmission for this media section to use reduced-
         size RTCP, as specified in [RFC5506].

      *  If the directional attribute in the answer is of type
         "sendrecv" or "sendonly", choose the media format to send as
         the most preferred media format from the remote description
         that is also present in the answer, as described in [RFC3264],
         Sections 6.1 and 7, and start transmitting RTP media once the
         underlying transport layers have been established.  If a SSRC
         has not already been chosen for this outgoing RTP stream,
         choose a random one.

      *  The payload type mapping from the remote description is used to
         determine payload types for the outgoing RTP streams, including
         the payload type for the send media format chosen above.  Any
         RTP header extensions that were negotiated should be included
         in the outgoing RTP streams, using the extension mapping from
         the remote description; if the RID header extension has been
         negotiated, and RID values are specified, include the RID
         header extension in the outgoing RTP streams, as indicated in
         [I-D.ietf-mmusic-rid], Section 4.

      *  If simulcast has been negotiated, send the number of Source RTP
         Streams as specified in [I-D.ietf-mmusic-sdp-simulcast],
         Section 6.2.2.



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      *  If the send media format chosen above has a corresponding "rtx"
         media format, or a FEC mechanism has been negotiated, establish
         a Redundancy RTP Stream with a random SSRC for each Source RTP
         Stream, and start or continue transmitting RTX/FEC packets as
         needed.

      *  If the send media format chosen above has a corresponding "red"
         media format of the same clockrate, allow redundant encoding
         using the specified format for resiliency purposes, as
         discussed in [I-D.ietf-rtcweb-fec], Section 3.2.  Note that
         unlike RTX or FEC media formats, the "red" format is
         transmitted on the Source RTP Stream, not the Redundancy RTP
         Stream.

      *  Enable the RTCP feedback mechanisms referenced in the media
         section for all Source RTP Streams using the specified media
         formats.  Specifically, begin or continue sending the requested
         feedback types and reacting to received feedback, as specified
         in [RFC4585], Section 4.2.  When sending RTCP feedback, follow
         the rules and recommendations from
         [I-D.ietf-avtcore-rtp-multi-stream], Section 5.4.1 to select
         which SSRC to use.

      *  If the directional attribute is of type "recvonly" or
         "inactive", stop transmitting all RTP media, but continue
         sending RTCP, as described in [RFC3264], Section 5.1.

   o  If the media section proto value indicates use of SCTP:

      *  If no SCTP association yet exists, prepare to initiate a SCTP
         association over the associated ICE component and DTLS
         connection, using the local SCTP port value from the local
         description, and the remote SCTP port value from the remote
         description, as described in [I-D.ietf-mmusic-sctp-sdp],
         Section 10.2.

   If the answer contains valid bundle groups, discard any ICE
   components for the m= sections that will be bundled onto the primary
   ICE components in each bundle, and begin muxing these m= sections
   accordingly, as described in
   [I-D.ietf-mmusic-sdp-bundle-negotiation], Section 8.2.

   If the description is of type "answer", and there are still remaining
   candidates in the ICE candidate pool, discard them.







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6.  Processing RTP/RTCP

   When bundling, associating incoming RTP/RTCP with the proper m=
   section is defined in [I-D.ietf-mmusic-sdp-bundle-negotiation].  [The
   BUNDLE draft does not currently contain the necessary text to
   describe this demux, but when it does it will contain text like that
   contained in Appendix B.]  When not bundling, the proper m= section
   is clear from the ICE component over which the RTP/RTCP is received.

   Once the proper m= section(s) are known, RTP/RTCP is delivered to the
   RtpTransceiver(s) associated with the m= section(s) and further
   processing of the RTP/RTCP is done at the RtpTransceiver level.  This
   includes using RID [I-D.ietf-mmusic-rid] to distinguish between
   multiple Encoded Streams, as well as determine which Source RTP
   stream should be repaired by a given Redundancy RTP stream.

7.  Examples

   Note that this example section shows several SDP fragments.  To
   format in 72 columns, some of the lines in SDP have been split into
   multiple lines, where leading whitespace indicates that a line is a
   continuation of the previous line.  In addition, some blank lines
   have been added to improve readability but are not valid in SDP.

   More examples of SDP for WebRTC call flows can be found in
   [I-D.nandakumar-rtcweb-sdp].

7.1.  Simple Example

   This section shows a very simple example that sets up a minimal audio
   / video call between two browsers and does not use trickle ICE.  The
   example in the following section provides a more realistic example of
   what would happen in a normal browser to browser connection.

   The flow shows Alice's browser initiating the session to Bob's
   browser.  The messages from Alice's JS to Bob's JS are assumed to
   flow over some signaling protocol via a web server.  The JS on both
   Alice's side and Bob's side waits for all candidates before sending
   the offer or answer, so the offers and answers are complete.  Trickle
   ICE is not used.  Both Alice and Bob are using the default policy of
   balanced.










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//                  set up local media state
AliceJS->AliceUA:   create new PeerConnection
AliceJS->AliceUA:   addTrack with two tracks: audio and video
AliceJS->AliceUA:   createOffer to get offer
AliceJS->AliceUA:   setLocalDescription with offer
AliceUA->AliceJS:   multiple onicecandidate events with candidates

//                  wait for ICE gathering to complete
AliceUA->AliceJS:   onicecandidate event with null candidate
AliceJS->AliceUA:   get |offer-A1| from pendingLocalDescription

//                  |offer-A1| is sent over signaling protocol to Bob
AliceJS->WebServer: signaling with |offer-A1|
WebServer->BobJS:   signaling with |offer-A1|

//                  |offer-A1| arrives at Bob
BobJS->BobUA:       create a PeerConnection
BobJS->BobUA:       setRemoteDescription with |offer-A1|
BobUA->BobJS:       onaddstream event with remoteStream

//                  Bob accepts call
BobJS->BobUA:       addTrack with local tracks
BobJS->BobUA:       createAnswer
BobJS->BobUA:       setLocalDescription with answer
BobUA->BobJS:       multiple onicecandidate events with candidates

//                  wait for ICE gathering to complete
BobUA->BobJS:       onicecandidate event with null candidate
BobJS->BobUA:       get |answer-A1| from currentLocalDescription

//                  |answer-A1| is sent over signaling protocol to Alice
BobJS->WebServer:   signaling with |answer-A1|
WebServer->AliceJS: signaling with |answer-A1|

//                  |answer-A1| arrives at Alice
AliceJS->AliceUA:   setRemoteDescription with |answer-A1|
AliceUA->AliceJS:   onaddstream event with remoteStream

//                  media flows
BobUA->AliceUA:     media sent from Bob to Alice
AliceUA->BobUA:     media sent from Alice to Bob


   The SDP for |offer-A1| looks like:


   v=0
   o=- 4962303333179871722 1 IN IP4 0.0.0.0



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   s=-
   t=0 0
   a=group:BUNDLE a1 v1
   a=ice-options:trickle
   m=audio 56500 UDP/TLS/RTP/SAVPF 96 0 8 97 98
   c=IN IP4 192.0.2.1
   a=mid:a1
   a=rtcp:56501 IN IP4 192.0.2.1
   a=msid:47017fee-b6c1-4162-929c-a25110252400
          f83006c5-a0ff-4e0a-9ed9-d3e6747be7d9
   a=sendrecv
   a=rtpmap:96 opus/48000/2
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:97 telephone-event/8000
   a=rtpmap:98 telephone-event/48000
   a=maxptime:120
   a=ice-ufrag:ETEn1v9DoTMB9J4r
   a=ice-pwd:OtSK0WpNtpUjkY4+86js7ZQl
   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=rtcp-mux
   a=rtcp-rsize
   a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
   a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:mid
   a=candidate:3348148302 1 udp 2113937151 192.0.2.1 56500
               typ host
   a=candidate:3348148302 2 udp 2113937151 192.0.2.1 56501
               typ host
   a=end-of-candidates

   m=video 56502 UDP/TLS/RTP/SAVPF 100 101
   c=IN IP4 192.0.2.1
   a=rtcp:56503 IN IP4 192.0.2.1
   a=mid:v1
   a=msid:61317484-2ed4-49d7-9eb7-1414322a7aae
          f30bdb4a-5db8-49b5-bcdc-e0c9a23172e0
   a=sendrecv
   a=rtpmap:100 VP8/90000
   a=rtpmap:101 rtx/90000
   a=fmtp:101 apt=100
   a=ice-ufrag:BGKkWnG5GmiUpdIV
   a=ice-pwd:mqyWsAjvtKwTGnvhPztQ9mIf
   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



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   a=setup:actpass
   a=rtcp-mux
   a=rtcp-rsize
   a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid
   a=rtcp-fb:100 ccm fir
   a=rtcp-fb:100 nack
   a=rtcp-fb:100 nack pli
   a=candidate:3348148302 1 udp 2113937151 192.0.2.1 56502
               typ host
   a=candidate:3348148302 2 udp 2113937151 192.0.2.1 56503
               typ host
   a=end-of-candidates


   The SDP for |answer-A1| looks like:


   v=0
   o=- 6729291447651054566 1 IN IP4 0.0.0.0
   s=-
   t=0 0
   a=group:BUNDLE a1 v1
   m=audio 20000 UDP/TLS/RTP/SAVPF 96 0 8 97 98
   c=IN IP4 192.0.2.2
   a=mid:a1
   a=rtcp:20000 IN IP4 192.0.2.2
   a=msid:PI39StLS8W7ZbQl1sJsWUXkr3Zf12fJUvzQ1
          PI39StLS8W7ZbQl1sJsWUXkr3Zf12fJUvzQ1a0
   a=sendrecv
   a=rtpmap:96 opus/48000/2
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:97 telephone-event/8000
   a=rtpmap:98 telephone-event/48000
   a=maxptime:120
   a=ice-ufrag:6sFvz2gdLkEwjZEr
   a=ice-pwd:cOTZKZNVlO9RSGsEGM63JXT2
   a=fingerprint:sha-256 6B:8B:F0:65:5F:78:E2:51:3B:AC:6F:F3:3F:46:1B:35
               :DC:B8:5F:64:1A:24:C2:43:F0:A1:58:D0:A1:2C:19:08
   a=setup:active
   a=rtcp-mux
   a=rtcp-rsize
   a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
   a=candidate:2299743422 1 udp 2113937151 192.0.2.2 20000
               typ host
   a=end-of-candidates

   m=video 20000 UDP/TLS/RTP/SAVPF 100 101



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   c=IN IP4 192.0.2.2
   a=rtcp 20001 IN IP4 192.0.2.2
   a=mid:v1
   a=msid:PI39StLS8W7ZbQl1sJsWUXkr3Zf12fJUvzQ1
          PI39StLS8W7ZbQl1sJsWUXkr3Zf12fJUvzQ1v0
   a=sendrecv
   a=rtpmap:100 VP8/90000
   a=rtpmap:101 rtx/90000
   a=fmtp:101 apt=100
   a=fingerprint:sha-256 6B:8B:F0:65:5F:78:E2:51:3B:AC:6F:F3:3F:46:1B:35
                        :DC:B8:5F:64:1A:24:C2:43:F0:A1:58:D0:A1:2C:19:08
   a=setup:active
   a=rtcp-mux
   a=rtcp-rsize
   a=rtcp-fb:100 ccm fir
   a=rtcp-fb:100 nack
   a=rtcp-fb:100 nack pli


7.2.  Normal Examples

   This section shows a typical example of a session between two
   browsers setting up an audio channel and a data channel.  Trickle ICE
   is used in full trickle mode with a bundle policy of max-bundle, an
   RTCP mux policy of require, and a single TURN server.  Later, two
   video flows, one for the presenter and one for screen sharing, are
   added to the session.  This example shows Alice's browser initiating
   the session to Bob's browser.  The messages from Alice's JS to Bob's
   JS are assumed to flow over some signaling protocol via a web server.


  //                  set up local media state
  AliceJS->AliceUA:   create new PeerConnection
  AliceJS->AliceUA:   addTrack with an audio track
  AliceJS->AliceUA:   createDataChannel to get data channel
  AliceJS->AliceUA:   createOffer to get |offer-B1|
  AliceJS->AliceUA:   setLocalDescription with |offer-B1|

  //                  |offer-B1| is sent over signaling protocol to Bob
  AliceJS->WebServer: signaling with |offer-B1|
  WebServer->BobJS:   signaling with |offer-B1|

  //                  |offer-B1| arrives at Bob
  BobJS->BobUA:       create a PeerConnection
  BobJS->BobUA:       setRemoteDescription with |offer-B1|
  BobUA->BobJS:       onaddstream with audio track from Alice

  //                  candidates are sent to Bob



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  AliceUA->AliceJS:   onicecandidate event with |candidate-B1| (host)
  AliceJS->WebServer: signaling with |candidate-B1|
  AliceUA->AliceJS:   onicecandidate event with |candidate-B2| (srflx)
  AliceJS->WebServer: signaling with |candidate-B2|

  WebServer->BobJS:   signaling with |candidate-B1|
  BobJS->BobUA:       addIceCandidate with |candidate-B1|
  WebServer->BobJS:   signaling with |candidate-B2|
  BobJS->BobUA:       addIceCandidate with |candidate-B2|

  //                  Bob accepts call
  BobJS->BobUA:       addTrack with local audio
  BobJS->BobUA:       createDataChannel to get data channel
  BobJS->BobUA:       createAnswer to get |answer-B1|
  BobJS->BobUA:       setLocalDescription with |answer-B1|

  //                  |answer-B1| is sent to Alice
  BobJS->WebServer:   signaling with |answer-B1|
  WebServer->AliceJS: signaling with |answer-B1|
  AliceJS->AliceUA:   setRemoteDescription with |answer-B1|
  AliceUA->AliceJS:   onaddstream event with audio track from Bob

  //                  candidates are sent to Alice
  BobUA->BobJS:       onicecandidate event with |candidate-B3| (host)
  BobJS->WebServer:   signaling with |candidate-B3|
  BobUA->BobJS:       onicecandidate event with |candidate-B4| (srflx)
  BobJS->WebServer:   signaling with |candidate-B4|

  WebServer->AliceJS: signaling with |candidate-B3|
  AliceJS->AliceUA:   addIceCandidate with |candidate-B3|
  WebServer->AliceJS: signaling with |candidate-B4|
  AliceJS->AliceUA:   addIceCandidate with |candidate-B4|

  //                  data channel opens
  BobUA->BobJS:       ondatachannel event
  AliceUA->AliceJS:   ondatachannel event
  BobUA->BobJS:       onopen
  AliceUA->AliceJS:   onopen

  //                  media is flowing between browsers
  BobUA->AliceUA:     audio+data sent from Bob to Alice
  AliceUA->BobUA:     audio+data sent from Alice to Bob

  //                  some time later Bob adds two video streams
  //                  note, no candidates exchanged, because of bundle
  BobJS->BobUA:       addTrack with first video stream
  BobJS->BobUA:       addTrack with second video stream
  BobJS->BobUA:       createOffer to get |offer-B2|



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  BobJS->BobUA:       setLocalDescription with |offer-B2|

  //                  |offer-B2| is sent to Alice
  BobJS->WebServer:   signaling with |offer-B2|
  WebServer->AliceJS: signaling with |offer-B2|
  AliceJS->AliceUA:   setRemoteDescription with |offer-B2|
  AliceUA->AliceJS:   onaddstream event with first video stream
  AliceUA->AliceJS:   onaddstream event with second video stream
  AliceJS->AliceUA:   createAnswer to get |answer-B2|
  AliceJS->AliceUA:   setLocalDescription with |answer-B2|

  //                  |answer-B2| is sent over signaling protocol to Bob
  AliceJS->WebServer: signaling with |answer-B2|
  WebServer->BobJS:   signaling with |answer-B2|
  BobJS->BobUA:       setRemoteDescription with |answer-B2|

  //                  media is flowing between browsers
  BobUA->AliceUA:     audio+video+data sent from Bob to Alice
  AliceUA->BobUA:     audio+video+data sent from Alice to Bob


   The SDP for |offer-B1| looks like:





























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   v=0
   o=- 4962303333179871723 1 IN IP4 0.0.0.0
   s=-
   t=0 0
   a=group:BUNDLE a1 d1
   a=ice-options:trickle
   m=audio 9 UDP/TLS/RTP/SAVPF 96 0 8 97 98
   c=IN IP4 0.0.0.0
   a=rtcp:9 IN IP4 0.0.0.0
   a=mid:a1
   a=msid:57017fee-b6c1-4162-929c-a25110252400
          e83006c5-a0ff-4e0a-9ed9-d3e6747be7d9
   a=sendrecv
   a=rtpmap:96 opus/48000/2
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:97 telephone-event/8000
   a=rtpmap:98 telephone-event/48000
   a=maxptime:120
   a=ice-ufrag:ATEn1v9DoTMB9J4r
   a=ice-pwd:AtSK0WpNtpUjkY4+86js7ZQl
   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=rtcp-mux
   a=rtcp-rsize
   a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
   a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:mid

   m=application 0 UDP/DTLS/SCTP webrtc-datachannel
   c=IN IP4 0.0.0.0
   a=bundle-only
   a=mid:d1
   a=fmtp:webrtc-datachannel max-message-size=65536
   a=sctp-port 5000
   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


   The SDP for |candidate-B1| looks like:


   candidate:109270923 1 udp 2122194687 192.168.1.2 51556 typ host


   The SDP for |candidate-B2| looks like:



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   candidate:4036177503 1 udp 1685987071 11.22.33.44 52546 typ srflx
             raddr 192.168.1.2 rport 51556


   The SDP for |answer-B1| looks like:


   v=0
   o=- 7729291447651054566 1 IN IP4 0.0.0.0
   s=-
   t=0 0
   a=group:BUNDLE a1 d1
   a=ice-options:trickle
   m=audio 9 UDP/TLS/RTP/SAVPF 96 0 8 97 98
   c=IN IP4 0.0.0.0
   a=rtcp:9 IN IP4 0.0.0.0
   a=mid:a1
   a=msid:QI39StLS8W7ZbQl1sJsWUXkr3Zf12fJUvzQ1
          QI39StLS8W7ZbQl1sJsWUXkr3Zf12fJUvzQ1a0
   a=sendrecv
   a=rtpmap:96 opus/48000/2
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:97 telephone-event/8000
   a=rtpmap:98 telephone-event/48000
   a=maxptime:120
   a=ice-ufrag:7sFvz2gdLkEwjZEr
   a=ice-pwd:dOTZKZNVlO9RSGsEGM63JXT2
   a=fingerprint:sha-256 6B:8B:F0:65:5F:78:E2:51:3B:AC:6F:F3:3F:46:1B:35
                        :DC:B8:5F:64:1A:24:C2:43:F0:A1:58:D0:A1:2C:19:08
   a=setup:active
   a=rtcp-mux
   a=rtcp-rsize
   a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
   a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:mid

   m=application 9 UDP/DTLS/SCTP webrtc-datachannel
   c=IN IP4 0.0.0.0
   a=mid:d1
   a=fmtp:webrtc-datachannel max-message-size=65536
   a=sctp-port 5000
   a=fingerprint:sha-256 6B:8B:F0:65:5F:78:E2:51:3B:AC:6F:F3:3F:46:1B:35
                        :DC:B8:5F:64:1A:24:C2:43:F0:A1:58:D0:A1:2C:19:08
   a=setup:active


   The SDP for |candidate-B3| looks like:




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   candidate:109270924 1 udp 2122194687 192.168.2.3 61665 typ host


   The SDP for |candidate-B4| looks like:


   candidate:4036177504 1 udp 1685987071 55.66.77.88 64532 typ srflx
             raddr 192.168.2.3 rport 61665


   The SDP for |offer-B2| looks like: (note the increment of the version
   number in the o= line, and the c= and a=rtcp lines, which indicate
   the local candidate that was selected)


   v=0
   o=- 7729291447651054566 2 IN IP4 0.0.0.0
   s=-
   t=0 0
   a=group:BUNDLE a1 d1 v1 v2
   a=ice-options:trickle
   m=audio 64532 UDP/TLS/RTP/SAVPF 96 0 8 97 98
   c=IN IP4 55.66.77.88
   a=rtcp:64532 IN IP4 55.66.77.88
   a=mid:a1
   a=msid:QI39StLS8W7ZbQl1sJsWUXkr3Zf12fJUvzQ1
          QI39StLS8W7ZbQl1sJsWUXkr3Zf12fJUvzQ1a0
   a=sendrecv
   a=rtpmap:96 opus/48000/2
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:97 telephone-event/8000
   a=rtpmap:98 telephone-event/48000
   a=maxptime:120
   a=ice-ufrag:7sFvz2gdLkEwjZEr
   a=ice-pwd:dOTZKZNVlO9RSGsEGM63JXT2
   a=fingerprint:sha-256 6B:8B:F0:65:5F:78:E2:51:3B:AC:6F:F3:3F:46:1B:35
                        :DC:B8:5F:64:1A:24:C2:43:F0:A1:58:D0:A1:2C:19:08
   a=setup:actpass
   a=rtcp-mux
   a=rtcp-rsize
   a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
   a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:mid
   a=candidate:109270924 1 udp 2122194687 192.168.2.3 61665 typ host
   a=candidate:4036177504 1 udp 1685987071 55.66.77.88 64532 typ srflx
               raddr 192.168.2.3 rport 61665
   a=candidate:3671762467 1 udp 41819903 66.77.88.99 50416 typ relay
               raddr 55.66.77.88 rport 64532



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   a=end-of-candidates

   m=application 64532 UDP/DTLS/SCTP webrtc-datachannel
   c=IN IP4 55.66.77.88
   a=mid:d1
   a=fmtp:webrtc-datachannel max-message-size=65536
   a=sctp-port 5000
   a=ice-ufrag:7sFvz2gdLkEwjZEr
   a=ice-pwd:dOTZKZNVlO9RSGsEGM63JXT2
   a=fingerprint:sha-256 6B:8B:F0:65:5F:78:E2:51:3B:AC:6F:F3:3F:46:1B:35
                        :DC:B8:5F:64:1A:24:C2:43:F0:A1:58:D0:A1:2C:19:08
   a=setup:actpass
   a=candidate:109270924 1 udp 2122194687 192.168.2.3 61665 typ host
   a=candidate:4036177504 1 udp 1685987071 55.66.77.88 64532 typ srflx
               raddr 192.168.2.3 rport 61665
   a=candidate:3671762467 1 udp 41819903 66.77.88.99 50416 typ relay
               raddr 55.66.77.88 rport 64532
   a=end-of-candidates

   m=video 0 UDP/TLS/RTP/SAVPF 100 101
   c=IN IP4 55.66.77.88
   a=bundle-only
   a=rtcp:64532 IN IP4 55.66.77.88
   a=mid:v1
   a=msid:61317484-2ed4-49d7-9eb7-1414322a7aae
          f30bdb4a-5db8-49b5-bcdc-e0c9a23172e0
   a=sendrecv
   a=rtpmap:100 VP8/90000
   a=rtpmap:101 rtx/90000
   a=fmtp:101 apt=100
   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=rtcp-mux
   a=rtcp-rsize
   a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:mid
   a=rtcp-fb:100 ccm fir
   a=rtcp-fb:100 nack
   a=rtcp-fb:100 nack pli

   m=video 0 UDP/TLS/RTP/SAVPF 100 101
   c=IN IP4 55.66.77.88
   a=bundle-only
   a=rtcp:64532 IN IP4 55.66.77.88
   a=mid:v1
   a=msid:71317484-2ed4-49d7-9eb7-1414322a7aae
          f30bdb4a-5db8-49b5-bcdc-e0c9a23172e0



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   a=sendrecv
   a=rtpmap:100 VP8/90000
   a=rtpmap:101 rtx/90000
   a=fmtp:101 apt=100
   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=rtcp-mux
   a=rtcp-rsize
   a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:mid
   a=rtcp-fb:100 ccm fir
   a=rtcp-fb:100 nack
   a=rtcp-fb:100 nack pli


   The SDP for |answer-B2| looks like: (note the use of setup:passive to
   maintain the existing DTLS roles, and the use of a=recvonly to
   indicate that the video streams are one-way)


   v=0
   o=- 4962303333179871723 2 IN IP4 0.0.0.0
   s=-
   t=0 0
   a=group:BUNDLE a1 d1 v1 v2
   a=ice-options:trickle
   m=audio 52546 UDP/TLS/RTP/SAVPF 96 0 8 97 98
   c=IN IP4 11.22.33.44
   a=rtcp:52546 IN IP4 11.22.33.44
   a=mid:a1
   a=msid:57017fee-b6c1-4162-929c-a25110252400
          e83006c5-a0ff-4e0a-9ed9-d3e6747be7d9
   a=sendrecv
   a=rtpmap:96 opus/48000/2
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:97 telephone-event/8000
   a=rtpmap:98 telephone-event/48000
   a=maxptime:120
   a=ice-ufrag:ATEn1v9DoTMB9J4r
   a=ice-pwd:AtSK0WpNtpUjkY4+86js7ZQl
   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:passive
   a=rtcp-mux
   a=rtcp-rsize



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   a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
   a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:mid
   a=candidate:109270923 1 udp 2122194687 192.168.1.2 51556 typ host
   a=candidate:4036177503 1 udp 1685987071 11.22.33.44 52546 typ srflx
               raddr 192.168.1.2 rport 51556
   a=candidate:3671762466 1 udp 41819903 22.33.44.55 61405 typ relay
               raddr 11.22.33.44 rport 52546
   a=end-of-candidates

   m=application 52546 UDP/DTLS/SCTP webrtc-datachannel
   c=IN IP4 11.22.33.44
   a=mid:d1
   a=fmtp:webrtc-datachannel max-message-size=65536
   a=sctp-port 5000
   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:passive

   m=video 52546 UDP/TLS/RTP/SAVPF 100 101
   c=IN IP4 11.22.33.44
   a=rtcp:52546 IN IP4 11.22.33.44
   a=mid:v1
   a=recvonly
   a=rtpmap:100 VP8/90000
   a=rtpmap:101 rtx/90000
   a=fmtp:101 apt=100
   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:passive
   a=rtcp-mux
   a=rtcp-rsize
   a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:mid
   a=rtcp-fb:100 ccm fir
   a=rtcp-fb:100 nack
   a=rtcp-fb:100 nack pli

   m=video 52546 UDP/TLS/RTP/SAVPF 100 101
   c=IN IP4 11.22.33.44
   a=rtcp:52546 IN IP4 11.22.33.44
   a=mid:v2
   a=recvonly
   a=rtpmap:100 VP8/90000
   a=rtpmap:101 rtx/90000
   a=fmtp:101 apt=100
   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



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   a=setup:passive
   a=rtcp-mux
   a=rtcp-rsize
   a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:mid
   a=rtcp-fb:100 ccm fir
   a=rtcp-fb:100 nack
   a=rtcp-fb:100 nack pli


8.  Security Considerations

   The IETF has published separate documents
   [I-D.ietf-rtcweb-security-arch] [I-D.ietf-rtcweb-security] describing
   the security architecture for WebRTC as a whole.  The remainder of
   this section describes security considerations for this document.

   While formally the JSEP interface is an API, it is better to think of
   it is an Internet protocol, with the JS being untrustworthy from the
   perspective of the browser.  Thus, the threat model of [RFC3552]
   applies.  In particular, JS can call the API in any order and with
   any inputs, including malicious ones.  This is particularly relevant
   when we consider the SDP which is passed to setLocalDescription().
   While correct API usage requires that the application pass in SDP
   which was derived from createOffer() or createAnswer(), there is no
   guarantee that applications do so.  The browser MUST be prepared for
   the JS to pass in bogus data instead.

   Conversely, the application programmer MUST recognize that the JS
   does not have complete control of browser behavior.  One case that
   bears particular mention is that editing ICE candidates out of the
   SDP or suppressing trickled candidates does not have the expected
   behavior: implementations will still perform checks from those
   candidates even if they are not sent to the other side.  Thus, for
   instance, it is not possible to prevent the remote peer from learning
   your public IP address by removing server reflexive candidates.
   Applications which wish to conceal their public IP address should
   instead configure the ICE agent to use only relay candidates.

9.  IANA Considerations

   This document requires no actions from IANA.

10.  Acknowledgements

   Significant text incorporated in the draft as well and review was
   provided by Peter Thatcher, Taylor Brandstetter, Harald Alvestrand
   and Suhas Nandakumar.  Dan Burnett, Neil Stratford, Anant Narayanan,




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   Andrew Hutton, Richard Ejzak, Adam Bergkvist and Matthew Kaufman all
   provided valuable feedback on this proposal.

11.  References

11.1.  Normative References

   [I-D.ietf-avtcore-rtp-multi-stream]
              Lennox, J., Westerlund, M., Wu, Q., and C. Perkins,
              "Sending Multiple RTP Streams in a Single RTP Session",
              draft-ietf-avtcore-rtp-multi-stream-11 (work in progress),
              December 2015.

   [I-D.ietf-avtext-rid]
              Roach, A., Nandakumar, S., and P. Thatcher, "RTP Stream
              Identifier (RID) Source Description (SDES)", draft-ietf-
              avtext-rid-00 (work in progress), February 2016.

   [I-D.ietf-ice-trickle]
              Ivov, E., Rescorla, E., Uberti, J., and P. Saint-Andre,
              "Trickle ICE: Incremental Provisioning of Candidates for
              the Interactive Connectivity Establishment (ICE)
              Protocol".

   [I-D.ietf-mmusic-4572-update]
              Holmberg, C., "Updates to RFC 4572", draft-ietf-mmusic-
              4572-update-05 (work in progress), June 2016.

   [I-D.ietf-mmusic-dtls-sdp]
              Holmberg, C. and R. Shpount, "Using the SDP Offer/Answer
              Mechanism for DTLS", draft-ietf-mmusic-dtls-sdp-14 (work
              in progress), July 2016.

   [I-D.ietf-mmusic-msid]
              Alvestrand, H., "Cross Session Stream Identification in
              the Session Description Protocol", draft-ietf-mmusic-
              msid-01 (work in progress), August 2013.

   [I-D.ietf-mmusic-mux-exclusive]
              Holmberg, C., "Indicating Exclusive Support of RTP/RTCP
              Multiplexing using SDP", draft-ietf-mmusic-mux-
              exclusive-08 (work in progress), June 2016.

   [I-D.ietf-mmusic-rid]
              Thatcher, P., Zanaty, M., Nandakumar, S., Burman, B.,
              Roach, A., and B. Campen, "RTP Payload Format
              Constraints", draft-ietf-mmusic-rid-04 (work in progress),
              February 2016.



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   [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-mmusic-sdp-mux-attributes]
              Nandakumar, S., "A Framework for SDP Attributes when
              Multiplexing", draft-ietf-mmusic-sdp-mux-attributes-01
              (work in progress), February 2014.

   [I-D.ietf-mmusic-sdp-simulcast]
              Burman, B., Westerlund, M., Nandakumar, S., and M. Zanaty,
              "Using Simulcast in SDP and RTP Sessions", draft-ietf-
              mmusic-sdp-simulcast-04 (work in progress), February 2016.

   [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-fec]
              Uberti, J., "WebRTC Forward Error Correction
              Requirements", draft-ietf-rtcweb-fec-00 (work in
              progress), February 2015.

   [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.ietf-rtcweb-security]
              Rescorla, E., "Security Considerations for WebRTC", draft-
              ietf-rtcweb-security-06 (work in progress), January 2014.

   [I-D.ietf-rtcweb-security-arch]
              Rescorla, E., "WebRTC Security Architecture", draft-ietf-
              rtcweb-security-arch-09 (work in progress), February 2014.






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   [I-D.ietf-rtcweb-video]
              Roach, A., "WebRTC Video Processing and Codec
              Requirements", draft-ietf-rtcweb-video-00 (work in
              progress), July 2014.

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.

   [RFC3261]  Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
              A., Peterson, J., Sparks, R., Handley, M., and E.
              Schooler, "SIP: Session Initiation Protocol", RFC 3261,
              June 2002.

   [RFC3264]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
              with Session Description Protocol (SDP)", RFC 3264, June
              2002.

   [RFC3552]  Rescorla, E. and B. Korver, "Guidelines for Writing RFC
              Text on Security Considerations", BCP 72, RFC 3552, July
              2003.

   [RFC3605]  Huitema, C., "Real Time Control Protocol (RTCP) attribute
              in Session Description Protocol (SDP)", RFC 3605, October
              2003.

   [RFC3890]  Westerlund, M., "A Transport Independent Bandwidth
              Modifier for the Session Description Protocol (SDP)",
              RFC 3890, DOI 10.17487/RFC3890, September 2004,
              <http://www.rfc-editor.org/info/rfc3890>.

   [RFC4145]  Yon, D. and G. Camarillo, "TCP-Based Media Transport in
              the Session Description Protocol (SDP)", RFC 4145,
              September 2005.

   [RFC4566]  Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
              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,
              "Extended RTP Profile for Real-time Transport Control
              Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July
              2006.






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   [RFC5245]  Rosenberg, J., "Interactive Connectivity Establishment
              (ICE): A Protocol for Network Address Translator (NAT)
              Traversal for Offer/Answer Protocols", RFC 5245, April
              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
              Control Packets on a Single Port", RFC 5761, April 2010.

   [RFC5888]  Camarillo, G. and H. Schulzrinne, "The Session Description
              Protocol (SDP) Grouping Framework", RFC 5888, June 2010.

   [RFC6236]  Johansson, I. and K. Jung, "Negotiation of Generic Image
              Attributes in the Session Description Protocol (SDP)",
              RFC 6236, May 2011.

   [RFC6347]  Rescorla, E. and N. Modadugu, "Datagram Transport Layer
              Security Version 1.2", RFC 6347, January 2012.

   [RFC6904]  Lennox, J., "Encryption of Header Extensions in the Secure
              Real-time Transport Protocol (SRTP)", RFC 6904, April
              2013.

   [RFC7850]  Nandakumar, S., "Registering Values of the SDP 'proto'
              Field for Transporting RTP Media over TCP under Various
              RTP Profiles", RFC 7850, DOI 10.17487/RFC7850, April 2016,
              <http://www.rfc-editor.org/info/rfc7850>.

11.2.  Informative References

   [I-D.ietf-rtcweb-ip-handling]
              Uberti, J. and G. Shieh, "WebRTC IP Address Handling
              Recommendations", draft-ietf-rtcweb-ip-handling-01 (work
              in progress), March 2016.

   [I-D.nandakumar-rtcweb-sdp]
              Nandakumar, S. and C. Jennings, "SDP for the WebRTC",
              draft-nandakumar-rtcweb-sdp-02 (work in progress), July
              2013.

   [RFC3389]  Zopf, R., "Real-time Transport Protocol (RTP) Payload for
              Comfort Noise (CN)", RFC 3389, September 2002.







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   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
              July 2003, <http://www.rfc-editor.org/info/rfc3550>.

   [RFC3556]  Casner, S., "Session Description Protocol (SDP) Bandwidth
              Modifiers for RTP Control Protocol (RTCP) Bandwidth",
              RFC 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
              Description Protocol (SDP) Security Descriptions for Media
              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.

   [RFC4733]  Schulzrinne, H. and T. Taylor, "RTP Payload for DTMF
              Digits, Telephony Tones, and Telephony Signals", RFC 4733,
              DOI 10.17487/RFC4733, December 2006,
              <http://www.rfc-editor.org/info/rfc4733>.

   [RFC5506]  Johansson, I. and M. Westerlund, "Support for Reduced-Size
              Real-Time Transport Control Protocol (RTCP): Opportunities
              and Consequences", RFC 5506, April 2009.

   [RFC5576]  Lennox, J., Ott, J., and T. Schierl, "Source-Specific
              Media Attributes in the Session Description Protocol
              (SDP)", RFC 5576, June 2009.

   [RFC5763]  Fischl, J., Tschofenig, H., and E. Rescorla, "Framework
              for Establishing a Secure Real-time Transport Protocol
              (SRTP) Security Context Using Datagram Transport Layer
              Security (DTLS)", RFC 5763, May 2010.

   [RFC5764]  McGrew, D. and E. Rescorla, "Datagram Transport Layer
              Security (DTLS) Extension to Establish Keys for the Secure
              Real-time Transport Protocol (SRTP)", RFC 5764, May 2010.

   [RFC6464]  Lennox, J., Ed., Ivov, E., and E. Marocco, "A Real-time
              Transport Protocol (RTP) Header Extension for Client-to-
              Mixer Audio Level Indication", RFC 6464,
              DOI 10.17487/RFC6464, December 2011,
              <http://www.rfc-editor.org/info/rfc6464>.



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   [RFC7656]  Lennox, J., Gross, K., Nandakumar, S., Salgueiro, G., and
              B. Burman, Ed., "A Taxonomy of Semantics and Mechanisms
              for Real-Time Transport Protocol (RTP) Sources", RFC 7656,
              DOI 10.17487/RFC7656, November 2015,
              <http://www.rfc-editor.org/info/rfc7656>.

   [W3C.WD-webrtc-20140617]
              Bergkvist, A., Burnett, D., Narayanan, A., and C.
              Jennings, "WebRTC 1.0: Real-time Communication Between
              Browsers", World Wide Web Consortium WD WD-webrtc-
              20140617, June 2014,
              <http://www.w3.org/TR/2011/WD-webrtc-20140617>.

Appendix A.  Appendix A

   For the syntax validation performed in Section 5.7, the following
   list of ABNF definitions is used:


































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   +-----------------------+-------------------------------------------+
   | Attribute             | Reference                                 |
   +-----------------------+-------------------------------------------+
   | ptime                 | [RFC4566] Section 9                       |
   | maxptime              | [RFC4566] Section 9                       |
   | rtpmap                | [RFC4566] Section 9                       |
   | recvonly              | [RFC4566] Section 9                       |
   | sendrecv              | [RFC4566] Section 9                       |
   | sendonly              | [RFC4566] Section 9                       |
   | inactive              | [RFC4566] Section 9                       |
   | framerate             | [RFC4566] Section 9                       |
   | fmtp                  | [RFC4566] Section 9                       |
   | quality               | [RFC4566] Section 9                       |
   | rtcp                  | [RFC3605] Section 2.1                     |
   | setup                 | [RFC4145] Sections 3, 4, and 5            |
   | connection            | [RFC4145] Sections 3, 4, and 5            |
   | fingerprint           | [RFC4572] Section 5                       |
   | rtcp-fb               | [RFC4585] Section 4.2                     |
   | candidate             | [RFC5245] Section 15.1                    |
   | remote-candidates     | [RFC5245] Section 15.2                    |
   | ice-lite              | [RFC5245] Section 15.3                    |
   | ice-ufrag             | [RFC5245] Section 15.4                    |
   | ice-pwd               | [RFC5245] Section 15.4                    |
   | ice-options           | [RFC5245] Section 15.5                    |
   | extmap                | [RFC5285] Section 7                       |
   | mid                   | [RFC5888] Section 4 and 5                 |
   | group                 | [RFC5888] Section 4 and 5                 |
   | imageattr             | [RFC6236] Section 3.1                     |
   | extmap (encrypt       | [RFC6904] Section 4                       |
   | option)               |                                           |
   | msid                  | [I-D.ietf-mmusic-msid] Section 2          |
   | rid                   | [I-D.ietf-mmusic-rid] Section 10          |
   | simulcast             | [I-D.ietf-mmusic-sdp-simulcast]Section    |
   |                       | 6.1                                       |
   | dtls-id               | [I-D.ietf-mmusic-dtls-sdp]Section 4       |
   +-----------------------+-------------------------------------------+

                       Table 1: SDP ABNF References

Appendix B.  Appendix B

   The following text is meant to completely replace section
   "Associating RTP/RTCP Streams With Correct SDP Media Description" of
   [I-D.ietf-mmusic-sdp-bundle-negotiation].

   As described in [RFC3550], RTP/RTCP packets are associated with RTP
   streams as defined in [RFC7656].  Each RTP stream is identified by an
   SSRC value, and each RTP/RTCP packet carries an SSRC value that is



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   used to associate the packet with the correct RTP stream.  An RTCP
   packet can carry multiple SSRC values, and might therefore be
   associated with multiple RTP streams.

   In order to be able to process received RTP/RTCP packets correctly it
   must be possible to associate an RTP stream with the correct "m="
   line, as the "m=" line and SDP attributes associated with the "m="
   line contain information needed to process the packets.

   As all RTP streams associated with a BUNDLE group use the same
   address:port combination for sending and receiving RTP/RTCP packets,
   the local address:port combination cannot be used to associate an RTP
   stream with the correct "m=" line.  In addition, multiple RTP streams
   might be associated with the same "m=" line.

   An offerer and answerer can inform each other which SSRC values they
   will use for an RTP stream by using the SDP 'ssrc' attribute
   [RFC5576].  However, an offerer will not know which SSRC values the
   answerer will use until the offerer has received the answer providing
   that information.  Due to this, before the offerer has received the
   answer, the offerer will not be able to associate an RTP stream with
   the correct "m=" line using the SSRC value associated with the RTP
   stream.  In addition, the offerer and answerer may start using new
   SSRC values mid-session, without informing each other using the SDP
   'ssrc' attribute.

   In order for an offerer and answerer to always be able to associate
   an RTP stream with the correct "m=" line, the offerer and answerer
   using the BUNDLE extension MUST support the mechanism defined in
   [I-D.ietf-mmusic-sdp-bundle-negotiation] section 14.  where the
   offerer and answerer insert the identification-tag associated with an
   "m=" line (provided by the remote peer) into RTP and RTCP packets
   associated with a BUNDLE group.

   The mapping from an SSRC to an identification-tag is carried in RTCP
   SDES packets or in RTP header extensions
   ([I-D.ietf-mmusic-sdp-bundle-negotiation] section 14).  Since a
   compound RTCP packet can contain multiple RTCP SDES packets, and each
   RTCP SDES packet can contain multiple chunks, an RTCP packet can
   contain several SSRC to identification-tag mappings.  The offerer and
   answerer maintain tables used for routing that are updated each time
   an RTP/RTCP packet contains new information that affects how packets
   should be routed.

   To prepare for demultiplexing RTP packets to the correct "m=" line,
   the following steps MUST be followed for each BUNDLE group.





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      Construct a table mapping MID to "m=" line for each "m=" line in
      this BUNDLE group.  Note that an "m=" line may only have one MID.

      Construct a table mapping incoming SSRC to "m=" line for each "m="
      line in this BUNDLE group and for each SSRC configured for
      receiving in that "m=" line.

      Construct a table mapping outgoing SSRC to "m=line" for each "m="
      line in this BUNDLE group and for each SSRC configured for sending
      in that "m=" line.

      Construct a table mapping payload type to "m=" line for each "m="
      line in the BUNDLE group and for each payload type configured for
      receiving in that "m=" line.  If any payload type is configured
      for receiving in more than one "m=" line in the BUNDLE group, do
      not it include it in the table.

      Note that for each of these tables, there can only be one mapping
      for any given key (MID, SSRC, or PT).  In other words, the tables
      are not multimaps.

   As "m=" lines are added or removed from the BUNDLE groups, or their
   configurations are changed, the tables above MUST also be updated.

   For each RTP packet received, the following steps MUST be followed to
   route the packet to the correct "m=" section within a BUNDLE group.
   Note that the phrase 'deliver a packet to the "m=" line' means to
   further process the packet as would normally happen with RTP/RTCP, if
   it were received on a transport associated with that "m=" line
   outside of a BUNDLE group (i.e., if the "m=" line were not BUNDLEd),
   including dropping an RTP packet if the packet's PT does not match
   any PT in the "m=" line.

      If the packet has a MID and that MID is not in the table mapping
      MID to "m=" line, drop the packet and stop.

      If the packet has a MID and that MID is in the table mapping MID
      to "m=" line, update the incoming SSRC mapping table to include an
      entry that maps the packet's SSRC to the "m=" line for that MID.

      If the packet's SSRC is in the incoming SSRC mapping table, route
      the packet to the associated "m=" line and stop.

      If the packet's payload type is in the payload type table, update
      the the incoming SSRC mapping table to include an entry that maps
      the packet's SSRC to the "m=" line for that payload type.  In
      addition, route the packet to the associated "m=" line and stop.




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      Otherwise, drop the packet.

   For each RTCP packet received (including each RTCP packet that is
   part of a compound RTCP packet), the packet MUST be routed to the
   appropriate handler for the SSRCs it contains information about.
   Some examples of such handling are given below.

      If the packet is of type SR, and the sender SSRC for the packet is
      found in the incoming SSRC table, deliver a copy of the packet to
      the "m=" line associated with that SSRC.  In addition, for each
      report block in the report whose SSRC is found in the outgoing
      SSRC table, deliver a copy of the RTCP packet to the "m=" line
      associated with that SSRC.

      If the packet is of type RR, for each report block in the packet
      whose SSRC is found in the outgoing SSRC table, deliver a copy of
      the RTCP packet to the "m=" line associated with that SSRC.

      If the packet is of type SDES, and the sender SSRC for the packet
      is found in the incoming SSRC table, deliver the packet to the
      "m=" line associated with that SSRC.  In addition, for each chunk
      in the packet that contains a MID that is in the table mapping MID
      to "m=" line, update the incoming SSRC mapping table to include an
      entry that maps the SSRC for that chunk to the "m=" line
      associated with that MID.  (This case can occur when RTCP for a
      source is received before any RTP packets.)

      If the packet is of type BYE, for each SSRC indicated in the
      packet that is found in the incoming SSRC table, deliver a copy of
      the packet to the "m=" line associated with that SSRC.

      If the packet is of type RTPFB or PSFB, as defined in [RFC4585],
      and the media source SSRC for the packet is found in the outgoing
      SSRC table, deliver the packet to the "m=" line associated with
      that SSRC.

Appendix C.  Change log

   Note: This section will be removed by RFC Editor before publication.

   Changes in draft-18:

   o  Update demux algorithm and move it to an appendix in preparation
      for merging it into BUNDLE.

   o  Clarify why we can't handle an incoming offer to send simulcast.

   o  Expand IceCandidate object text.



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   o  Further document use of ICE candidate pool.

   o  Document removeTrack.

   o  Update requirements to only accept the last generated offer/answer
      as an argument to setLocalDescription.

   o  Allow round pixels.

   o  Fix code around default timing when AVPF is not specified.

   o  Clean up terminology around m= line and m=section.

   o  Provide a more realistic example for minimum decoder capabilities.

   o  Document behavior when rtcp-mux policy is require but rtcp-mux
      attribute not provided.

   o  Expanded discussion of RtpSender and RtpReceiver.

   o  Add RtpTransceiver.currentDirection and document setDirection.

   o  Require imageattr x=0, y=0 to indicate that there are no valid
      resolutions.

   o  Require a privacy-preserving MID/RID construction.

   o  Require support for RFC 3556 bandwidth modifiers.

   o  Update maxptime description.

   o  Note that endpoints may encounter extra codecs in answers and
      subsequent offers from non-JSEP peers.

   o  Update references.

   Changes in draft-17:

   o  Split createOffer and createAnswer sections to clearly indicate
      attributes which always appear and which only appear when not
      bundled into another m= section.

   o  Add descriptions of RtpTransceiver methods.

   o  Describe how to process RTCP feedback attributes.

   o  Clarify transceiver directions and their interaction with 3264.




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   o  Describe setCodecPreferences.

   o  Update RTP demux algorithm.  Include RTCP.

   o  Update requirements for when a=rtcp is included, limiting to cases
      where it is needed for backward compatibility.

   o  Clarify SAR handling.

   o  Updated addTrack matching algorithm.

   o  Remove a=ssrc requirements.

   o  Handle a=setup in reoffers.

   o  Discuss how RTX/FEC should be handled.

   o  Discuss how telephone-event should be handled.

   o  Discuss how CN/DTX should be handled.

   o  Add missing references to ABNF table.

   Changes in draft-16:

   o  Update addIceCandidate to indicate ICE generation and allow per-m=
      section end-of-candidates.

   o  Update fingerprint handling to use draft-ietf-mmusic-4572-update.

   o  Update text around SDP processing of RTP header extensions and
      payload formats.

   o  Add sections on simulcast, addTransceiver, and createDataChannel.

   o  Clarify text to ensure that the session ID is a positive 63 bit
      integer.

   o  Clarify SDP processing for direction indication.

   o  Describe SDP processing for rtcp-mux-only.

   o  Specify how SDP session version in o= line.

   o  Require that when doing an re-offer, the capabilities of the new
      session are mostly required to be a subset of the previously
      negotiated session.




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   o  Clarified ICE restart interaction with bundle-only.

   o  Remove support for changing SDP before calling
      setLocalDescription.

   o  Specify algorithm for demuxing RTP based on MID, PT, and SSRC.

   o  Clarify rules for rejecting m= lines when bundle policy is
      balanced or max-bundle.

   Changes in draft-15:

   o  Clarify text around codecs offered in subsequent transactions to
      refer to what's been negotiated.

   o  Rewrite LS handling text to indicate edge cases and that we're
      living with them.

   o  Require that answerer reject m= lines when there are no codecs in
      common.

   o  Enforce max-bundle on offer processing.

   o  Fix TIAS formula to handle bits vs. kilobits.

   o  Describe addTrack algorithm.

   o  Clean up references.

   Changes in draft-14:

   o  Added discussion of RtpTransceivers + RtpSenders + RtpReceivers,
      and how they interact with createOffer/createAnswer.

   o  Removed obsolete OfferToReceiveX options.

   o  Explained how addIceCandidate can be used for end-of-candidates.

   Changes in draft-13:

   o  Clarified which SDP lines can be ignored.

   o  Clarified how to handle various received attributes.

   o  Revised how attributes should be generated for bundled m= lines.

   o  Remove unused references.




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   o  Remove text advocating use of unilateral PTs.

   o  Trigger an ICE restart even if the ICE candidate policy is being
      made more strict.

   o  Remove the 'public' ICE candidate policy.

   o  Move open issues into GitHub issues.

   o  Split local/remote description accessors into current/pending.

   o  Clarify a=imageattr handling.

   o  Add more detail on VoiceActivityDetection handling.

   o  Reference draft-shieh-rtcweb-ip-handling.

   o  Make it clear when an ICE restart should occur.

   o  Resolve changes needed in references.

   o  Remove MSID semantics.

   o  ice-options are now at session level.

   o  Default RTCP mux policy is now 'require'.

   Changes in draft-12:

   o  Filled in sections on applying local and remote descriptions.

   o  Discussed downscaling and upscaling to fulfill imageattr
      requirements.

   o  Updated what SDP can be modified by the application.

   o  Updated to latest datachannel SDP.

   o  Allowed multiple fingerprint lines.

   o  Switched back to IPv4 for dummy candidates.

   o  Added additional clarity on ICE default candidates.

   Changes in draft-11:

   o  Clarified handling of RTP CNAMEs.




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   o  Updated what SDP lines should be processed or ignored.

   o  Specified how a=imageattr should be used.

   Changes in draft-10:

   o  Described video size negotiation with imageattr.

   o  Clarified rejection of sections that do not have mux-only.

   o  Add handling of LS groups

   Changes in draft-09:

   o  Don't return null for {local,remote}Description after close().

   o  Changed TCP/TLS to UDP/DTLS in RTP profile names.

   o  Separate out bundle and mux policy.

   o  Added specific references to FEC mechanisms.

   o  Added canTrickle mechanism.

   o  Added section on subsequent answers and, answer options.

   o  Added text defining set{Local,Remote}Description behavior.

   Changes in draft-08:

   o  Added new example section and removed old examples in appendix.

   o  Fixed <proto> field handling.

   o  Added text describing a=rtcp attribute.

   o  Reworked handling of OfferToReceiveAudio and OfferToReceiveVideo
      per discussion at IETF 90.

   o  Reworked trickle ICE handling and its impact on m= and c= lines
      per discussion at interim.

   o  Added max-bundle-and-rtcp-mux policy.

   o  Added description of maxptime handling.

   o  Updated ICE candidate pool default to 0.




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   o  Resolved open issues around AppID/receiver-ID.

   o  Reworked and expanded how changes to the ICE configuration are
      handled.

   o  Some reference updates.

   o  Editorial clarification.

   Changes in draft-07:

   o  Expanded discussion of VAD and Opus DTX.

   o  Added a security considerations section.

   o  Rewrote the section on modifying SDP to require implementations to
      clearly indicate whether any given modification is allowed.

   o  Clarified impact of IceRestart on CreateOffer in local-offer
      state.

   o  Guidance on whether attributes should be defined at the media
      level or the session level.

   o  Renamed "default" bundle policy to "balanced".

   o  Removed default ICE candidate pool size and clarify how it works.

   o  Defined a canonical order for assignment of MSTs to m= lines.

   o  Removed discussion of rehydration.

   o  Added Eric Rescorla as a draft editor.

   o  Cleaned up references.

   o  Editorial cleanup

   Changes in draft-06:

   o  Reworked handling of m= line recycling.

   o  Added handling of BUNDLE and bundle-only.

   o  Clarified handling of rollback.

   o  Added text describing the ICE Candidate Pool and its behavior.




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   o  Allowed OfferToReceiveX to create multiple recvonly m= sections.

   Changes in draft-05:

   o  Fixed several issues identified in the createOffer/Answer sections
      during document review.

   o  Updated references.

   Changes in draft-04:

   o  Filled in sections on createOffer and createAnswer.

   o  Added SDP examples.

   o  Fixed references.

   Changes in draft-03:

   o  Added text describing relationship to W3C specification

   Changes in draft-02:

   o  Converted from nroff

   o  Removed comparisons to old approaches abandoned by the working
      group

   o  Removed stuff that has moved to W3C specification

   o  Align SDP handling with W3C draft

   o  Clarified section on forking.

   Changes in draft-01:

   o  Added diagrams for architecture and state machine.

   o  Added sections on forking and rehydration.

   o  Clarified meaning of "pranswer" and "answer".

   o  Reworked how ICE restarts and media directions are controlled.

   o  Added list of parameters that can be changed in a description.

   o  Updated suggested API and examples to match latest thinking.




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   o  Suggested API and examples have been moved to an appendix.

   Changes in draft -00:

   o  Migrated from draft-uberti-rtcweb-jsep-02.

Authors' Addresses

   Justin Uberti
   Google
   747 6th St S
   Kirkland, WA  98033
   USA

   Email: justin@uberti.name


   Cullen Jennings
   Cisco
   400 3rd Avenue SW
   Calgary, AB  T2P 4H2
   Canada

   Email: fluffy@iii.ca


   Eric Rescorla (editor)
   Mozilla
   331 Evelyn Ave
   Mountain View, CA  94041
   USA

   Email: ekr@rtfm.com


















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