path: root/doc/rfc/rfc8829.txt

Internet Engineering Task Force (IETF)                         J. Uberti
Request for Comments: 8829                                        Google
Category: Standards Track                                    C. Jennings
ISSN: 2070-1721                                                    Cisco
                                                        E. Rescorla, Ed.
                                                                 Mozilla
                                                            January 2021


            JavaScript Session Establishment Protocol (JSEP)

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 is an Internet Standards Track document.

   This document is a product of the Internet Engineering Task Force
   (IETF).  It represents the consensus of the IETF community.  It has
   received public review and has been approved for publication by the
   Internet Engineering Steering Group (IESG).  Further information on
   Internet Standards is available in Section 2 of RFC 7841.

   Information about the current status of this document, any errata,
   and how to provide feedback on it may be obtained at
   https://www.rfc-editor.org/info/rfc8829.

Copyright Notice

   Copyright (c) 2021 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
   (https://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
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

Table of Contents

   1.  Introduction
     1.1.  General Design of JSEP
     1.2.  Other Approaches Considered
     1.3.  Contradiction regarding bundle-only "m=" sections
   2.  Terminology
   3.  Semantics and Syntax
     3.1.  Signaling Model
     3.2.  Session Descriptions and State Machine
     3.3.  Session Description Format
     3.4.  Session Description Control
       3.4.1.  RtpTransceivers
       3.4.2.  RtpSenders
       3.4.3.  RtpReceivers
     3.5.  ICE
       3.5.1.  ICE Gathering Overview
       3.5.2.  ICE Candidate Trickling
         3.5.2.1.  ICE Candidate Format
       3.5.3.  ICE Candidate Policy
       3.5.4.  ICE Candidate Pool
       3.5.5.  ICE Versions
     3.6.  Video Size Negotiation
       3.6.1.  Creating an imageattr Attribute
       3.6.2.  Interpreting imageattr Attributes
     3.7.  Simulcast
     3.8.  Interactions with Forking
       3.8.1.  Sequential Forking
       3.8.2.  Parallel Forking
   4.  Interface
     4.1.  PeerConnection
       4.1.1.  Constructor
       4.1.2.  addTrack
       4.1.3.  removeTrack
       4.1.4.  addTransceiver
       4.1.5.  onaddtrack Event
       4.1.6.  createDataChannel
       4.1.7.  ondatachannel Event
       4.1.8.  createOffer
       4.1.9.  createAnswer
       4.1.10. SessionDescriptionType
         4.1.10.1.  Use of Provisional Answers
         4.1.10.2.  Rollback
       4.1.11. setLocalDescription
       4.1.12. setRemoteDescription
       4.1.13. currentLocalDescription
       4.1.14. pendingLocalDescription
       4.1.15. currentRemoteDescription
       4.1.16. pendingRemoteDescription
       4.1.17. canTrickleIceCandidates
       4.1.18. setConfiguration
       4.1.19. addIceCandidate
       4.1.20. onicecandidate Event
     4.2.  RtpTransceiver
       4.2.1.  stop
       4.2.2.  stopped
       4.2.3.  setDirection
       4.2.4.  direction
       4.2.5.  currentDirection
       4.2.6.  setCodecPreferences
   5.  SDP Interaction Procedures
     5.1.  Requirements Overview
       5.1.1.  Usage Requirements
       5.1.2.  Profile Names and Interoperability
     5.2.  Constructing an Offer
       5.2.1.  Initial Offers
       5.2.2.  Subsequent Offers
       5.2.3.  Options Handling
         5.2.3.1.  IceRestart
         5.2.3.2.  VoiceActivityDetection
     5.3.  Generating an Answer
       5.3.1.  Initial Answers
       5.3.2.  Subsequent Answers
       5.3.3.  Options Handling
         5.3.3.1.  VoiceActivityDetection
     5.4.  Modifying an Offer or Answer
     5.5.  Processing a Local Description
     5.6.  Processing a Remote Description
     5.7.  Processing a Rollback
     5.8.  Parsing a Session Description
       5.8.1.  Session-Level Parsing
       5.8.2.  Media Section Parsing
       5.8.3.  Semantics Verification
     5.9.  Applying a Local Description
     5.10. Applying a Remote Description
     5.11. Applying an Answer
   6.  Processing RTP/RTCP
   7.  Examples
     7.1.  Simple Example
     7.2.  Detailed Example
     7.3.  Early Transport Warmup Example
   8.  Security Considerations
   9.  IANA Considerations
   10. References
     10.1.  Normative References
     10.2.  Informative References
   Appendix A.  SDP ABNF Syntax
   Acknowledgements
   Authors' Addresses

1.  Introduction

   This document describes how the W3C Web Real-Time Communication
   (WebRTC) RTCPeerConnection interface [W3C.webrtc] is used to control
   the setup, management, and teardown of a multimedia session.

1.1.  General Design of JSEP

   WebRTC call setup has been designed to focus on controlling the media
   plane, leaving signaling-plane behavior 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 call signaling
   protocol, 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 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), which allows for
   full control of the signaling state machine from JavaScript.  As
   described above, JSEP assumes a model in which a JavaScript
   application executes inside a runtime containing WebRTC APIs (the
   "JSEP implementation").  The JSEP implementation is almost entirely
   divorced 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 Interactive
   Connectivity Establishment (ICE) state machine [RFC8445].  The
   combination of the JSEP implementation and the JavaScript application
   is referred to throughout this document as a "JSEP endpoint".

   In this document, the use of JSEP is described as if it always occurs
   between two JSEP endpoints.  Note, though, that in many cases it will
   actually be between a JSEP endpoint and some kind of server, such as
   a gateway or Multipoint Control Unit (MCU).  This distinction is
   invisible to the JSEP endpoint; 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 configuration
   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
   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 [RFC8445], JSEP decouples the ICE state machine from
   the overall signaling state machine.  The ICE state machine must
   remain in the JSEP implementation because only the implementation has
   the necessary knowledge of candidates and other transport
   information.  Performing this separation provides additional
   flexibility in protocols that decouple session descriptions from
   transport.  For instance, in traditional SIP, each offer or answer is
   self-contained, including both the session descriptions and the
   transport information.  However, [RFC8840] allows SIP to be used with
   Trickle ICE [RFC8838], in which the session description can be sent
   immediately and the transport information can be sent when available.
   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.  JSEP's
   decoupling of the ICE and signaling state machines allows it to
   accommodate either model.

   Although it abstracts signaling, the JSEP approach requires that the
   application 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 make life easier for the application 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 provide implementations of the SIP [RFC3261] and
   Extensible Messaging and Presence Protocol (XMPP) [RFC6120] signaling
   protocols atop the JSEP API.  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 JSEP implementation, forcing it
   to have to understand and handle concepts like signaling glare (see
   [RFC3264], Section 4).

   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 the argument that requiring
   exposure of this level of complexity to the application programmer
   would not be beneficial; it would (1) result in an API where even a
   simple example would require a significant amount of code to
   orchestrate all the needed interactions and (2) create a large API
   surface that would need 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 JSEP implementation.
   Instead of providing createOffer/createAnswer methods within the
   implementation, 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.

1.3.  Contradiction regarding bundle-only "m=" sections

   Since the approval of the WebRTC specification documents, the IETF
   has become aware of an inconsistency between the document specifying
   JSEP and the document specifying BUNDLE (this RFC and [RFC8843],
   respectively).  Rather than delaying publication further to come to a
   resolution, the documents are being published as they were originally
   approved.  The IETF intends to restart work on these technologies,
   and revised versions of these documents will be published as soon as
   a resolution becomes available.

   The specific issue involves the handling of "m=" sections that are
   designated as bundle-only, as discussed in Section 4.1.1 of this RFC.
   Currently, there is divergence between JSEP and BUNDLE, as well as
   between these specifications and existing browser implementations:

   *  JSEP prescribes that said "m=" sections should use port zero and
      add an "a=bundle-only" attribute in initial offers, but not in
      answers or subsequent offers.

   *  BUNDLE prescribes that these "m=" sections should be marked as
      described in the previous point, but in all offers and answers.

   *  Most current browsers do not mark any "m=" sections with port zero
      and instead use the same port for all bundled "m=" sections; some
      others follow the JSEP behavior.

2.  Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
   "OPTIONAL" in this document are to be interpreted as described in
   BCP 14 [RFC2119] [RFC8174] when, and only when, they appear in all
   capitals, as shown here.

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 JSEP implementation is totally decoupled
   from the actual 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 implementation, and
   when.

         +-----------+                               +-----------+
         |  Web App  |<--- App-Specific Signaling -->|  Web App  |
         +-----------+                               +-----------+
               ^                                            ^
               |  SDP                                       |  SDP
               V                                            V
         +-----------+                                +-----------+
         |   JSEP    |<----------- Media ------------>|   JSEP    |
         |   Impl.   |                                |   Impl.   |
         +-----------+                                +-----------+

                       Figure 1: JSEP Signaling Model

3.2.  Session Descriptions and State Machine

   In order to establish the media plane, the JSEP implementation 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; some parameters are
   declarative, and the remote side must either accept them or reject
   them altogether.  An example of such a parameter is the TLS
   fingerprints [RFC8122] as used in the context of DTLS [RFC6347];
   these fingerprints are calculated based on the local certificate(s)
   offered and are not subject to negotiation.

   In addition, various RFCs put different conditions on the format of
   offers versus answers.  For example, an offer may propose an
   arbitrary number of "m=" sections (i.e., media descriptions as
   described in [RFC4566], Section 5.14), but an answer must contain the
   exact same number as the offer.

   Lastly, while the exact media parameters are known only after an
   offer and an answer have been exchanged, the offerer may receive ICE
   checks, and possibly media (e.g., in the case of a re-offer after a
   connection has been established) before it receives 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 JSEP
   implementation 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]), and the answerer, who first
   calls setRemoteDescription(sdp [offer]) and then later
   setLocalDescription(sdp [answer]).

   During the offer/answer exchange, the outstanding offer is considered
   to be "pending" at the offerer and the answerer, as it may be either
   accepted or rejected.  If this is a re-offer, each side will also
   have "current" local and remote descriptions, which reflect the
   result of the last offer/answer exchange.  Sections 4.1.14, 4.1.16,
   4.1.13, and 4.1.15 provide more detail on pending and current
   descriptions.

   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, Traversal Using Relays around NAT (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 JSEP level,
   a new offer can be generated at any point.  For example, when using
   SIP for signaling, if one offer is sent and is then canceled 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 then finally end the
   offer/answer exchange by sending a final answer.  The state machine
   for this is as follows:

                       setRemote(OFFER)               setLocal(PRANSWER)
                           /-----\                               /-----\
                           |     |                               |     |
                           v     |                               v     |
            +---------------+    |                +---------------+    |
            |               |----/                |               |----/
            |  have-        | setLocal(PRANSWER)  | have-         |
            |  remote-offer |------------------- >| local-pranswer|
            |               |                     |               |
            |               |                     |               |
            +---------------+                     +---------------+
                 ^   |                                   |
                 |   | setLocal(ANSWER)                  |
   setRemote(OFFER)  |                                   |
                 |   V                  setLocal(ANSWER) |
            +---------------+                            |
            |               |                            |
            |               |<---------------------------+
            |    stable     |
            |               |<---------------------------+
            |               |                            |
            +---------------+          setRemote(ANSWER) |
                 ^   |                                   |
                 |   | setLocal(OFFER)                   |
   setRemote(ANSWER) |                                   |
                 |   V                                   |
            +---------------+                     +---------------+
            |               |                     |               |
            |  have-        | setRemote(PRANSWER) |have-          |
            |  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.

3.3.  Session Description Format

   JSEP's session descriptions use Session Description Protocol (SDP)
   syntax for their internal representation.  While this format is not
   optimal for manipulation from JavaScript, it is widely accepted and
   is 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 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.

   As detailed below, 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 that tell the JSEP
   implementation 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/createAnswer 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 an "m="
   section, or if RtpTransceivers have been stopped and disassociated
   from "m=" sections.  An RtpTransceiver is said to be associated with
   an "m=" section if its media identification (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, an "m=" section
   is always associated with exactly one RtpTransceiver.  However, in
   certain cases where an "m=" section has been rejected, as discussed
   in Section 5.2.2 below, that "m=" section will be "recycled" and
   associated with a new RtpTransceiver with a new MID value.

   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 the RTP Control Protocol (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 by 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
   JSEP implementation 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 locally or remotely initiated ICE restart.

   When a new gathering phase starts, the ICE agent will notify the
   application that gathering is occurring through a state change event.
   Then, when each new ICE candidate becomes available, the ICE agent
   will supply it to the application via an onicecandidate event; these
   candidates will also automatically be added to the current and/or
   pending local session description.  Finally, when all candidates have
   been gathered, a final onicecandidate 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, if an "m=" section is
   bundled (either by a successful bundle negotiation or by being marked
   as bundle-only), then candidates will be gathered and exchanged for
   that "m=" section if and only if its MID item is a BUNDLE-tag, as
   described in [RFC8843].

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 [RFC8838].  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 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 that 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.

   The candidate details are specified in an IceCandidate field, using
   the same SDP syntax as the "candidate-attribute" field defined in
   [RFC8839], Section 5.1.  Note that this field does not contain an
   "a=" prefix, as indicated in the following example:

   candidate:1 1 UDP 1694498815 192.0.2.33 10000 typ host

   The IceCandidate object contains a field to indicate which ICE
   username fragment (ufrag) it is associated with, as defined in
   [RFC8839], Section 5.4.  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 an "m=" section index or by 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 5.10
   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.

   Implementations MUST be prepared to receive objects with some fields
   missing, as mentioned above.

3.5.3.  ICE Candidate Policy

   Typically, when gathering ICE candidates, the JSEP implementation
   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
   implementation chooses to enforce regarding which IP addresses are
   permitted for the application, as discussed in [RFC8828].

   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 that they want to take the call.
   For this scenario, the JSEP implementation 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 JSEP implementation will
   determine the current setting at the start of each gathering phase.
   Then, during the gathering phase, the implementation 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 JSEP implementation to begin
   ICE gathering via the information supplied to setLocalDescription, as
   the local description indicates the number of ICE components that
   will be needed and for which candidates must be gathered.  However,
   to accelerate cases where the application knows the number of ICE
   components to use ahead of time, it may ask the implementation to
   gather a pool of potential ICE candidates to help ensure rapid media
   setup.

   When setLocalDescription is eventually called and the JSEP
   implementation prepares 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 are 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
   will be kept alive as long as they may be needed, the application
   will consume resources on the STUN/TURN servers it is using.  ("STUN"
   stands for "Session Traversal Utilities for NAT".)

3.5.5.  ICE Versions

   While this specification formally relies on [RFC8445], at the time of
   its publication, the majority of WebRTC implementations support the
   version of ICE described in [RFC5245].  The "ice2" attribute defined
   in [RFC8445] can be used to detect the version in use by a remote
   endpoint and to provide a smooth transition from the older
   specification to the newer one.  Implementations MUST be able to
   accept remote descriptions that do not have the "ice2" attribute.

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 have some
   maximum set by policy.  By specifying these limits in an
   "a=imageattr" attribute, JSEP endpoints can attempt to ensure that
   the remote sender transmits video at an acceptable resolution.
   However, when communicating with a non-JSEP endpoint that does not
   understand this attribute, any signaled limits may be exceeded, and
   the JSEP implementation MUST handle this gracefully, e.g., by
   discarding the video.

   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

   The receiver will first combine any known local limits (e.g.,
   hardware decoder capabilities or local policy) to determine the
   absolute minimum and maximum sizes it can receive.  If there are no
   known local limits, the "a=imageattr" attribute SHOULD be omitted.
   If these local limits preclude receiving any video, i.e., the
   degenerate case of no permitted resolutions, the "a=imageattr"
   attribute MUST be omitted, and the "m=" section MUST be marked as
   sendonly/inactive, as appropriate.

   Otherwise, an "a=imageattr" attribute is created with a "recv"
   direction, and the resulting resolution space formed from the
   aforementioned intersection is used to specify its minimum and
   maximum "x=" and "y=" values.

   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 different limitations,
   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.  In
   this case, the implementation would generate this attribute:

             a=imageattr:* recv [x=[48:1280],y=[48:720],q=1.0]

   This declaration indicates that the receiver is capable of decoding
   any image resolution from 48x48 up to 1280x720 pixels.

3.6.2.  Interpreting imageattr Attributes

   [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 behavior that is more specific.  When a
   MediaStreamTrack, which is producing video of a certain resolution
   (the "track resolution"), is attached to an RtpSender, which is
   encoding the track video at the same or lower resolution(s) (the
   "encoder resolutions"), and a remote description is applied that
   references the sender and contains valid "a=imageattr recv"
   attributes, it MUST follow the rules below to ensure that the sender
   does not transmit a resolution that would exceed the size criteria
   specified in the attributes.  These rules MUST be followed as long as
   the attributes remain present in the remote description, including
   cases in which the track changes its resolution or is replaced with a
   different track.

   Depending on how the RtpSender is configured, it may be producing a
   single encoding at a certain resolution or, if simulcast
   (Section 3.7) has been negotiated, multiple encodings, each at their
   own specific resolution.  In addition, depending on the
   configuration, each encoding may have the flexibility to reduce
   resolution when needed or may be locked to a specific output
   resolution.

   For each encoding being produced by the RtpSender, the set of
   "a=imageattr recv" attributes in the corresponding "m=" section of
   the remote description is processed to determine what should be
   transmitted.  Only attributes that reference the media format
   selected for the encoding are considered; each such attribute is
   evaluated individually, starting with the attribute with the highest
   "q=" value.  If multiple attributes have the same "q=" value, they
   are evaluated in the order they appear in their containing "m="
   section.  Note that while JSEP endpoints will include at most one
   "a=imageattr recv" attribute per media format, JSEP endpoints may
   receive session descriptions from non-JSEP endpoints with "m="
   sections that contain multiple such attributes.

   For each "a=imageattr recv" attribute, the following rules are
   applied.  If this processing is successful, the encoding is
   transmitted accordingly, and no further attributes are considered for
   that encoding.  Otherwise, the next attribute is evaluated, in the
   aforementioned order.  If none of the supplied attributes can be
   processed successfully, the encoding MUST NOT be transmitted, and an
   error SHOULD be raised to the application.

   *  The limits from the attribute are compared to the encoder
      resolution.  Only the specific limits mentioned below are
      considered; any other values, such as picture aspect ratio, MUST
      be ignored.  When considering a MediaStreamTrack that is producing
      rotated video, the unrotated resolution MUST be used for the
      checks.  This is required regardless of whether the receiver
      supports performing receive-side rotation (e.g., through
      Coordination of Video Orientation (CVO) [TS26.114]), as it
      significantly simplifies the matching logic.

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

   *  If the encoder resolution exceeds the maximum size permitted by
      the attribute and the encoder is allowed to adjust its resolution,
      the encoder SHOULD apply downscaling in order to satisfy the
      limits.  Downscaling MUST NOT change the picture aspect ratio of
      the encoding, ignoring any trivial differences due to rounding.
      For example, if the encoder resolution is 1280x720 and the
      attribute specified a maximum of 640x480, the expected output
      resolution would be 640x360.  If downscaling cannot be applied,
      the attribute MUST NOT be used.

   *  If the encoder resolution is less than the minimum size permitted
      by the attribute, the attribute MUST NOT be used; the encoder MUST
      NOT apply upscaling.  JSEP implementations SHOULD avoid this
      situation by allowing receipt of arbitrarily small resolutions,
      perhaps via fallback to a software decoder.

   *  If the encoder resolution is within the maximum and minimum sizes,
      no action is needed.

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.  Upon generation of an offer or answer,
   these encodings are indicated via 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 that 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
   setting up simulcast in the case where the JSEP endpoint receives the
   initial offer requires out-of-band signaling or SDP inspection.
   However, in the case where the JSEP endpoint sets up simulcast in its
   initial offer, any established simulcast streams will continue to
   work upon receipt of an incoming re-offer.  Future versions of this
   specification may add additional APIs to handle the incoming initial
   offer scenario.

   When using JSEP to transmit multiple encodings from an RtpSender, the
   techniques from [RFC8853] and [RFC8851] are used.  Specifically, when
   multiple encodings have been configured for an RtpSender, the "m="
   section for the RtpSender will include an "a=simulcast" attribute, as
   defined in [RFC8853], Section 5.1, 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 [RFC8851], Section 4;
   the use of Restriction Identifiers (RIDs, also called rid-ids or
   RtpStreamIds) 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
   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 in 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 as follows:

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

   *  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 as either (1) a provisional answer, leaving open
   the possibility of using a different answer in the future or (2) 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 [RFC3960],
   Section 3.1.  Most SIP devices today only support exchanging media
   with a single device at a time and do not try to mix multiple early
   media 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
   (see Section 4.1) and making sure that the same local media streams
   have been added into this new PeerConnection.  Then the new
   PeerConnection object would produce an 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 JSEP implementation to only surface the
   permitted candidates (including any implementation-internal
   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 implementation 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 the 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 [RFC8843].  Regardless
   of policy, the application will always try to negotiate bundle onto a
   single transport and will offer a single bundle group across all "m="
   sections; 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 "m=" 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 "m=" sections 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 N media streams.  This policy
      balances desire to multiplex with the need to ensure that basic
      audio 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 "m=" 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 will
      require separate candidates to be gathered for each media stream.

   max-bundle:  Only the first "m=" 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 trade-off 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 JSEP implementation 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 JSEP implementation will only gather RTP candidates and
      will insert an "a=rtcp-mux-only" indication into any new "m="
      sections in offers it generates.  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" (Lip Synchronization) group when creating an offer or answer.
   Adding tracks to the same "LS" group indicates that the playback of
   these tracks should be synchronized for proper lip sync, as described
   in [RFC5888], Section 7.  addTrack attempts to 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
   "m=" section associated with the sender as recvonly (if
   transceiver.direction is sendrecv) or as inactive (if
   transceiver.direction 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 that 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.  onaddtrack Event

   The onaddtrack event is dispatched to the application when a new
   remote track has been signaled as a result of a setRemoteDescription
   call.  The new track is supplied as a MediaStreamTrack object in the
   event, along with the MediaStream(s) the track is part of.

4.1.6.  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 ("SCTP" stands for "Stream Control Transmission
   Protocol") 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 that
   are used by the PeerConnection (e.g., maxPacketLifetime) but are not
   reflected in the SDP and do not affect the JSEP state.

4.1.7.  ondatachannel Event

   The ondatachannel event is dispatched to the application when a new
   data channel has been negotiated by the remote side, which can occur
   at any time after the underlying SCTP/DTLS association has been
   established.  The new data channel object is supplied in the event.

4.1.8.  createOffer

   The createOffer method generates a blob of SDP that contains an offer
   per [RFC3264] with the supported configurations for the session,
   including descriptions of the media added to this PeerConnection, the
   codec, RTP, and 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 specification that defines 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 generating new ICE
   credentials, but it does not change the PeerConnection state, trigger
   candidate gathering, or cause media to start or stop flowing.
   Specifically, the offer is not applied, and does not become the
   pending local description, until setLocalDescription is called.

4.1.9.  createAnswer

   The createAnswer method generates a blob of SDP that contains an SDP
   answer per [RFC3264] with the supported configuration for the session
   that is compatible with the parameters supplied in the most recent
   call to setRemoteDescription; setRemoteDescription 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 specification that defines 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 generating new ICE
   credentials, but it does not change the PeerConnection state, trigger
   candidate gathering, or cause a media state change.  Specifically,
   the answer is not applied, and does not become the current local
   description, until setLocalDescription is called.

4.1.10.  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 MUST be parsed as an offer; said
   description may include many possible media configurations.  A
   description used as an "offer" may be applied any time the
   PeerConnection is in a "stable" state or applied as an update to a
   previously supplied but unanswered "offer".

   "pranswer" indicates that a description MUST be parsed as an answer,
   but not a final answer, and so MUST 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 as an update to a previously sent
   "pranswer".

   "answer" indicates that a description MUST be parsed as an answer,
   the offer/answer exchange MUST be considered complete, and any
   resources (decoders, candidates) that are no longer needed SHOULD be
   released.  A description used as an "answer" may be applied as a
   response to an "offer" or as 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 indicating that the
   state machine MUST be rolled back to the previous "stable" state, as
   described in Section 4.1.10.2.  The contents MUST be empty.

4.1.10.1.  Use of Provisional Answers

   Most 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 JSEP application
   is to create and send a "sendonly" final answer with a null
   MediaStreamTrack immediately after receiving the offer, which will
   prevent media from being sent by the caller and allow media to be
   sent immediately upon answer by the callee.  Later, when the callee
   actually accepts the call, the application can plug in the real
   MediaStreamTrack and create a new "sendrecv" offer to update the
   previous offer/answer pair and start bidirectional media flow.  While
   this could also be done with a "sendonly" pranswer followed by a
   "sendrecv" answer, the initial pranswer leaves the offer/answer
   exchange open, which means that the caller cannot send an updated
   offer during this time.

   As an example, consider a typical JSEP application that wants to set
   up audio and video as quickly as possible.  When the callee receives
   an offer with audio and video MediaStreamTracks, it will send an
   immediate answer accepting these tracks as sendonly (meaning that the
   caller will not send the callee any media yet, and because the callee
   has not yet added its own MediaStreamTracks, the callee will not send
   any media either).  It will then ask the user to accept the call and
   acquire the needed local tracks.  Upon acceptance by the user, the
   application will plug in the tracks it has acquired, which, because
   ICE handshaking and DTLS handshaking have likely completed by this
   point, can start transmitting immediately.  The application will also
   send a new offer to the remote side indicating call acceptance and
   moving the audio and video to be two-way media.  A detailed example
   flow along these lines is shown in Section 7.3.

   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.

4.1.10.2.  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, needs to return to a
   "stable" state and the previous local/remote description.  To support
   this, we introduce the concept of "rollback", which discards any
   proposed changes to the session, returning the state machine to the
   "stable" state.  A rollback is performed by supplying a session
   description of type "rollback" with empty contents to either
   setLocalDescription or setRemoteDescription.

4.1.11.  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, final answer, or rollback; 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 it continues
   until setRemoteDescription is called with either (1) a final answer,
   at which point the PeerConnection can fully adopt the pending local
   description or (2) 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 (1) setRemoteDescription was previously called with an offer, (2)
   setLocalDescription is called with an answer (provisional or final),
   (3) the media directions are compatible, and (4) media is available
   to send, this will result in the starting of media transmission.

4.1.12.  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.

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

   If (1) setLocalDescription was previously called with an offer, (2)
   setRemoteDescription is called with an answer (provisional or final),
   (3) the media directions are compatible, and (4) media is available
   to send, this will result in the starting of media transmission.

4.1.13.  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.

4.1.14.  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".

4.1.15.  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.

4.1.16.  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.

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

4.1.17.  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 that 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.

4.1.18.  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 calling this method
   depend on when it is invoked, and they will differ, depending on
   which specific parameters are changed:

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

   *  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 made available to the
      application until a gathering phase occurs, and so any necessary
      filtering can still be done on any pooled candidates.

   *  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.

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

   Calling this method may result in a change to the state of the ICE
   agent.

4.1.19.  addIceCandidate

   The addIceCandidate method provides an update to the ICE agent via an
   IceCandidate object (Section 3.5.2.1).  If the IceCandidate's
   candidate field is non-null, the IceCandidate is treated as a new
   remote ICE candidate, which will be added to the current and/or
   pending remote description according to the rules defined for Trickle
   ICE.  Otherwise, the IceCandidate is treated as an end-of-candidates
   indication, as defined in [RFC8838], Section 14.

   In either case, the "m=" section index, MID, and ufrag fields from
   the supplied IceCandidate are used to determine which "m=" section
   and ICE candidate generation the IceCandidate belongs to, as
   described in Section 3.5.2.1 above.  In the case of an end-of-
   candidates indication, null values for the "m=" section index and MID
   fields are interpreted to mean that the indication applies to all
   "m=" sections in the specified ICE candidate generation.  However, if
   both fields are null for a new remote candidate, this MUST be treated
   as an invalid condition, as specified below.

   If any IceCandidate fields contain invalid values or an error occurs
   during the processing of the IceCandidate object, the supplied
   IceCandidate MUST be ignored and an error MUST be returned.

   Otherwise, the new remote candidate or end-of-candidates indication
   is supplied to the ICE agent.  In the case of a new remote candidate,
   connectivity checks will be sent to the new candidate, assuming
   setLocalDescription has already been called to initialize the ICE
   gathering process.

4.1.20.  onicecandidate Event

   The onicecandidate event is dispatched to the application in two
   situations: (1) when the ICE agent has discovered a new allowed local
   ICE candidate during ICE gathering, as outlined in Section 3.5.1 and
   subject to the restrictions discussed in Section 3.5.3, or (2) when
   an ICE gathering phase completes.  The event contains a single
   IceCandidate object, as defined in Section 3.5.2.1.

   In the first case, the newly discovered candidate is reflected in the
   IceCandidate object, and all of its fields MUST be non-null.  This
   candidate will also be added to the current and/or pending local
   description according to the rules defined for Trickle ICE.

   In the second case, the event's IceCandidate object MUST have its
   candidate field set to null to indicate that the current gathering
   phase is complete, i.e., there will be no further onicecandidate
   events in this phase.  However, the IceCandidate's ufrag field MUST
   be specified to indicate which ICE candidate generation is ending.
   The IceCandidate's "m=" section index and MID fields MAY be specified
   to indicate that the event applies to a specific "m=" section, or set
   to null to indicate it applies to all "m=" sections in the current
   ICE candidate generation.  This event can be used by the application
   to generate an end-of-candidates indication, as defined in [RFC8838],
   Section 13.

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 stop 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.  The permitted values
   for direction are "recvonly", "sendrecv", "sendonly", and "inactive",
   mirroring the identically named direction attributes defined in
   [RFC4566], Section 6.

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

4.2.5.  currentDirection

   The currentDirection property indicates the last negotiated direction
   for the transceiver's associated "m=" section.  More specifically, it
   indicates the direction attribute [RFC3264] of the associated "m="
   section in the last applied answer (including provisional answers),
   with "send" and "recv" directions reversed if it was a remote answer.
   For example, if the direction 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.

   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 follow the specified preferences.

5.  SDP Interaction Procedures

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

5.1.  Requirements Overview

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

5.1.1.  Usage Requirements

   All session descriptions handled by JSEP implementations, 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.

   *  ICE, as specified in [RFC8445], MUST be used.  Note that the
      remote endpoint may use a lite implementation; implementations
      MUST properly handle remote endpoints that use ICE-lite.  The
      remote endpoint may also use an older version of ICE;
      implementations MUST properly handle remote endpoints that use ICE
      as specified in [RFC5245].

   *  DTLS [RFC6347] or DTLS-SRTP [RFC5763] MUST be used, as appropriate
      for the media type, as specified in [RFC8827].

   The SDP security descriptions mechanism for SRTP keying [RFC4568]
   MUST NOT be used, as discussed in [RFC8827].

5.1.2.  Profile Names and Interoperability

   For media "m=" sections, JSEP implementations MUST support the
   "UDP/TLS/RTP/SAVPF" profile specified in [RFC5764] as well as the
   "TCP/DTLS/RTP/SAVPF" profile specified in [RFC7850] and MUST indicate
   one of these profiles for each media "m=" line they produce in an
   offer.  For data "m=" sections, implementations MUST support the
   "UDP/DTLS/SCTP" profile as well as the "TCP/DTLS/SCTP" profile and
   MUST indicate one of these profiles for each data "m=" line they
   produce in an offer.  The exact profile to use is determined by the
   protocol associated with the current default or selected ICE
   candidate, as described in [RFC8839], Section 4.2.1.2.

   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/TLS/RTP/SAVPF" or "TCP/DTLS/RTP/SAVPF".
   In order to simplify compatibility with such endpoints, JSEP
   implementations MUST follow the following rules when processing the
   media "m=" sections in a received offer:

   *  Any profile in the offer matching one of the following MUST be
      accepted:

      -  "RTP/AVP" (defined in [RFC4566], Section 8.2.2)

      -  "RTP/AVPF" (defined in [RFC4585], Section 9)

      -  "RTP/SAVP" (defined in [RFC3711], Section 12)

      -  "RTP/SAVPF" (defined in [RFC5124], Section 6)

      -  "TCP/DTLS/RTP/SAVP" (defined in [RFC7850], Section 3.4)

      -  "TCP/DTLS/RTP/SAVPF" (defined in [RFC7850], Section 3.5)

      -  "UDP/TLS/RTP/SAVP" (defined in [RFC5764], Section 9)

      -  "UDP/TLS/RTP/SAVPF" (defined in [RFC5764], Section 9)

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

   *  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" attributes.  Note that lack of an
      "a=fingerprint" attribute will lead to negotiation failure.

   *  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".

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

   Note that re-offers by JSEP implementations MUST use the correct
   profile strings even if the initial offer/answer exchange used an
   (incorrect) older profile string.  This simplifies JSEP behavior,
   with minimal downside, as any remote endpoint that fails to handle
   such a re-offer will also fail to handle a JSEP endpoint's initial
   offer.

5.2.  Constructing an Offer

   When createOffer is called, a new SDP description MUST be created
   that includes the functionality specified in [RFC8834].  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:

   *  The first SDP line MUST be "v=0" as defined in [RFC4566],
      Section 5.1.

   *  The second SDP line MUST be an "o=" line as defined in [RFC4566],
      Section 5.2.  The value of the <username> field SHOULD be "-".
      The <sess-id> MUST be representable by a 64-bit signed integer,
      and the value MUST be less than 2^(63)-1.  It is RECOMMENDED that
      the <sess-id> be constructed by generating a 64-bit quantity with
      the highest bit set 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 a local IP address in
      this field; this problem is discussed in [RFC8828].  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.

   *  The third SDP line MUST be a "s=" line as defined 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 in JSEP, having the same meaningless
      value seems clearer.

   *  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.

   *  Encryption Keys ("k=") lines do not provide sufficient security
      and MUST NOT be included.

   *  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".

   *  An "a=ice-options" line with the "trickle" and "ice2" options MUST
      be added, as specified in [RFC8840], Section 4.1.1 and [RFC8445],
      Section 10.

   *  If WebRTC identity is being used, an "a=identity" line MUST be
      added, as described in [RFC8827], Section 5.

   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.  If there are no
   such RtpTransceivers, no "m=" sections are generated; more can be
   added later, as discussed in [RFC3264], Section 5.

   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
   contain a unique set of ICE credentials and a 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 any and all certificates that
   have been specified for the PeerConnection; as a result, they MUST
   all have the same fingerprint value or values [RFC8122], or these
   values MUST be session-level attributes.

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

   *  If the "m=" section is marked as bundle-only, then the <port>
      value MUST be set to zero.  Otherwise, 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 default port value
      of 9 (Discard) MUST be used, as indicated in [RFC8840],
      Section 4.1.1.

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

   *  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.

   *  Unless excluded by the above restrictions, the media formats MUST
      include the mandatory audio/video codecs as specified in
      [RFC7874], Section 3 and [RFC7742], 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 default value "IN IP4
   0.0.0.0", as defined in [RFC8840], Section 4.1.1.

   [RFC8859] groups SDP attributes into different categories.  To avoid
   unnecessary duplication when bundling, attributes of category
   IDENTICAL or TRANSPORT MUST NOT be repeated in bundled "m=" sections,
   repeating the guidance from [RFC8843], Section 7.1.3.  This includes
   "m=" sections for which bundling has been negotiated and is still
   desired, as well as "m=" sections marked as bundle-only.

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

   *  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 [RFC8843], Section 15.2.
      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.

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

   *  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.

   *  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.

   *  For each supported Forward Error Correction (FEC) mechanism,
      "a=rtpmap" and "a=fmtp" lines, as specified in [RFC4566],
      Section 6.  The FEC mechanisms that MUST be supported are
      specified in [RFC8854], Section 7, and specific usage for each
      media type is outlined in Sections 4 and 5.

   *  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.

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

   *  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 [RFC8834],
      Section 5.2.  Any header extensions that require encryption MUST
      be specified as indicated in [RFC6904], Section 4.

   *  For each supported RTCP feedback mechanism, an "a=rtcp-fb" line,
      as specified in [RFC4585], Section 4.2.  The list of RTCP feedback
      mechanisms that SHOULD/MUST be supported is specified in
      [RFC8834], Section 5.1.

   *  If the RtpTransceiver has a sendrecv or sendonly direction:

      -  For each MediaStream that was associated with the transceiver
         when it was created via addTrack or addTransceiver, an "a=msid"
         line, as specified in [RFC8830], Section 2, but omitting the
         "appdata" field.

   *  If the RtpTransceiver has a sendrecv or sendonly direction, and
      the application has specified a rid-id for an encoding, 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 [RFC8851], and its direction MUST be "send".  If the
      application has chosen a rid-id, it MUST be used; otherwise, a
      rid-id MUST be generated by the implementation. rid-ids MUST be
      generated in a fashion that does not leak user information, e.g.,
      randomly or using a per-PeerConnection counter (see guidance at
      the end of [RFC8852], Section 3.3), and SHOULD be 3 bytes or less,
      to allow them to efficiently fit into the RTP header extensions
      defined in [RFC8852], Section 3.3.  If no encodings have been
      specified, or only one encoding is specified but without a rid-id,
      then no "a=rid" lines are generated.

   *  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 [RFC8853], Section 5.1.
      The associated set of rid-ids MUST include all of the rid-ids used
      in the "a=rid" lines for this "m=" section.

   *  If (1) the bundle policy for this PeerConnection is set to "max-
      bundle" and this is not the first "m=" section or (2) the bundle
      policy is set to "balanced" and this is not the first "m=" section
      for this media type, an "a=bundle-only" line.

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

   *  "a=ice-ufrag" and "a=ice-pwd" lines, as specified in [RFC8839],
      Section 5.4.

   *  For each desired digest algorithm, one or more "a=fingerprint"
      lines for each of the endpoint's certificates, as specified in
      [RFC8122], Section 5.

   *  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".

   *  An "a=tls-id" line, as specified in [RFC8842], Section 5.2.

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

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

   *  If the RTP/RTCP multiplexing policy is "require", an "a=rtcp-mux-
      only" line, as specified in [RFC8858], Section 4.

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

   Lastly, if a data channel has been created, an "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" [RFC8841].  The
   <fmt> value MUST be set to "webrtc-datachannel" as specified in
   [RFC8841], Section 4.2.2.

   Within the data "m=" section, an "a=mid" line MUST be generated and
   included as described above, along with an "a=sctp-port" line
   referencing the SCTP port number, as defined in [RFC8841],
   Section 5.1; and, if appropriate, an "a=max-message-size" line, as
   defined in [RFC8841], Section 6.1.

   As discussed above, the following attributes of category IDENTICAL or
   TRANSPORT are included only if the data "m=" section either has a
   unique address or is associated with the BUNDLE-tag (e.g., if it is
   the only "m=" section):

   *  "a=ice-ufrag"

   *  "a=ice-pwd"

   *  "a=fingerprint"

   *  "a=setup"

   *  "a=tls-id"

   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 JSEP
   implementation offers all "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 that SDP permits to be at either the session level or the
   media level SHOULD generally be at the media level even if they are
   identical.  This assists development and debugging by making it
   easier to understand individual media sections, especially if one of
   a set of initially identical attributes is subsequently changed.
   However, implementations MAY choose to aggregate attributes at the
   session level, and JSEP implementations MUST be prepared to receive
   attributes in either location.

   Attributes other than the ones specified above MAY be included,
   except for the following attributes, which are specifically
   incompatible with the requirements of [RFC8834] and MUST NOT be
   included:

   *  "a=crypto"

   *  "a=key-mgmt"

   *  "a=ice-lite"

   Note that when bundle is used, any additional attributes that are
   added MUST follow the advice in [RFC8859] 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 previous offer was not applied using setLocalDescription,
   meaning the PeerConnection is still in the "stable" state, the steps
   for generating an initial offer MUST be followed, subject to the
   following restriction:

   *  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
      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 previous offer was applied using setLocalDescription, but a
   corresponding answer from the remote side has not yet been applied,
   meaning the PeerConnection is still in the "have-local-offer" state,
   an offer is generated by following the steps in the "stable" state
   above, along with these exceptions:

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

   *  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
      (including a new MID value) and placing it at the same index as
      the "m=" section with a zero port.

   *  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.

   *  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 all "a=msid" lines removed.

   *  For RtpTransceivers that are not stopped, the "a=msid" line or
      lines MUST stay the same if they are present in the current
      description, regardless of changes to the transceiver's direction
      or track.  If no "a=msid" line is present in the current
      description, "a=msid" line(s) MUST be generated according to the
      same rules as for an initial offer.

   *  Each "m=" and "c=" line MUST be filled in with the port, relevant
      RTP profile, and address of the default candidate for the "m="
      section, as described in [RFC8839], Section 4.2.1.2 and clarified
      in Section 5.1.2.  If no RTP candidates have yet been gathered,
      default values MUST still be used, as described above.

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

   *  Each "a=ice-ufrag" and "a=ice-pwd" line MUST stay the same, unless
      the ICE configuration has changed (e.g., changes to either 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.

   *  If the "m=" section is not bundled into another "m=" section, its
      "a=rtcp" attribute line MUST be filled in with the port and
      address of the default RTCP candidate, as indicated in [RFC5761],
      Section 5.1.3.  If no RTCP candidates have yet been gathered,
      default values MUST be used, as described in Section 5.2.1 above.

   *  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 [RFC8839], Section 5.1.  If candidate
      gathering for the section has completed, an "a=end-of-candidates"
      attribute MUST be added, as described in [RFC8840], Section 8.2.
      If the "m=" section is bundled into another "m=" section, both
      "a=candidate" and "a=end-of-candidates" MUST be omitted.

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

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

   If the previous offer was applied using setLocalDescription, and a
   corresponding answer from the remote side has been applied using
   setRemoteDescription, meaning the PeerConnection is in the "have-
   remote-pranswer" state or the "stable" state, an offer is generated
   based on the negotiated session descriptions by following the steps
   mentioned for the "have-local-offer" state above.

   In addition, for each existing, 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 local
   or remote description, as appropriate:

   *  The "m=" line and corresponding "a=rtpmap" and "a=fmtp" lines MUST
      only include media formats that have not been excluded by the
      codec preferences of the associated transceiver and also MUST
      include all currently available formats.  Media formats that were
      previously offered but are no longer available (e.g., a shared
      hardware codec) MAY be excluded.

   *  Unless codec preferences have been set for the associated
      transceiver, the media formats on the "m=" line MUST be generated
      in the same order as in the most recent answer.  Any media formats
      that were not present in the most recent answer MUST be added
      after all existing formats.

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

   *  The RTCP feedback mechanisms MUST only include those that are
      present in the most recent answer, except for the case of format-
      specific mechanisms that are referencing a newly added media
      format.

   *  The "a=rtcp" line MUST NOT be added if the most recent answer
      included an "a=rtcp-mux" line.

   *  The "a=rtcp-mux" line MUST be the same as that in the most recent
      answer.

   *  The "a=rtcp-mux-only" line MUST NOT be added.

   *  The "a=rtcp-rsize" line MUST NOT be added unless present in the
      most recent answer.

   *  An "a=bundle-only" line, as defined in [RFC8843], Section 6, MUST
      NOT be added.  Instead, JSEP implementations MUST simply omit
      parameters in the IDENTICAL and TRANSPORT categories for bundled
      "m=" sections, as described in [RFC8843], Section 7.1.3.

   *  Note that if media "m=" sections are bundled into a data "m="
      section, then certain TRANSPORT and IDENTICAL attributes may
      appear in the data "m=" section even if they would otherwise only
      be appropriate for a media "m=" section (e.g., "a=rtcp-mux").
      This cannot happen in initial offers because in the initial offer
      JSEP implementations always list media "m=" sections (if any)
      before the data "m=" section (if any), and at least one of those
      media "m=" sections will not have the "a=bundle-only" attribute.
      Therefore, in initial offers, any "a=bundle-only" "m=" sections
      will be bundled into a preceding non-bundle-only media "m="
      section.

   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.

   "a=group:LS" attributes are generated in the same way as for initial
   offers, with the additional stipulation that any lip sync groups that
   were present in the most recent answer MUST continue to exist and
   MUST contain any previously existing MID identifiers, as long as the
   identified "m=" sections still exist and are not rejected, and the
   group still contains at least two MID identifiers.  This ensures that
   any synchronized "recvonly" "m=" sections continue to be synchronized
   in the new offer.

5.2.3.  Options Handling

   The createOffer method takes as a parameter an RTCOfferOptions
   object.  Special processing is performed when generating an 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 [RFC8839], Section 4.4.3.1.1.  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

   Silence suppression, also known as discontinuous transmission
   ("DTX"), can reduce the bandwidth used for audio by switching to a
   special encoding when voice activity is not detected, at the cost of
   some fidelity.

   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 [RFC6716], the
   "usedtx=1" parameter, specified in [RFC7587], would be used in the
   offer.

   If the "VoiceActivityDetection" option is specified, with a value of
   "false", the JSEP implementation 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 using the Opus codec, the "usedtx=0" parameter
   would be specified in the offer.  In addition, the implementation
   MUST NOT use silence suppression for media it generates, regardless
   of whether the "CN" codecs or related fmtp parameters appear in the
   peer's description.  The impact of these rules is that silence
   suppression in JSEP depends on mutual agreement of both sides, which
   ensures consistent handling regardless of which codec is used.

   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 [RFC8834].  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 Section 5.2.1 above, except that the "a=ice-options"
   line, with the "trickle" option as specified in [RFC8840],
   Section 4.1.3 and the "ice2" option as specified in [RFC8445],
   Section 10, 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, select the local RtpTransceivers that are referenced by
   the MID values in the specified group, and determine which of them
   either reference a common local MediaStream (specified in the calls
   to addTrack/addTransceiver used to create them) or have no
   MediaStream to reference because they were not created by addTrack/
   addTransceiver.  If at least two such RtpTransceivers exist, a group
   of type "LS" with the MID values of these RtpTransceivers MUST be
   added.  Otherwise, the offered "LS" group MUST be ignored and no
   corresponding group generated in the answer.

   As a simple example, consider the following offer of a single audio
   and single video track contained in the same MediaStream.  SDP lines
   not relevant to this example have been removed for clarity.  As
   explained in Section 5.2, a group of type "LS" has been added that
   references each track's RtpTransceiver.

             a=group:LS a1 v1
             m=audio 10000 UDP/TLS/RTP/SAVPF 0
             a=mid:a1
             a=msid:ms1
             m=video 10001 UDP/TLS/RTP/SAVPF 96
             a=mid:v1
             a=msid:ms1

   If the answerer uses a single MediaStream when it adds its tracks,
   both of its transceivers will reference this stream, and so the
   subsequent answer will contain a "LS" group identical to that in the
   offer, as shown below:

             a=group:LS a1 v1
             m=audio 20000 UDP/TLS/RTP/SAVPF 0
             a=mid:a1
             a=msid:ms2
             m=video 20001 UDP/TLS/RTP/SAVPF 96
             a=mid:v1
             a=msid:ms2

   However, if the answerer groups its tracks into separate
   MediaStreams, its transceivers will reference different streams, and
   so the subsequent answer will not contain a "LS" group.

             m=audio 20000 UDP/TLS/RTP/SAVPF 0
             a=mid:a1
             a=msid:ms2a
             m=video 20001 UDP/TLS/RTP/SAVPF 96
             a=mid:v1
             a=msid:ms2b

   Finally, if the answerer does not add any tracks, its transceivers
   will not reference any MediaStreams, causing the preferences of the
   offerer to be maintained, and so the subsequent answer will contain
   an identical "LS" group.

             a=group:LS a1 v1
             m=audio 20000 UDP/TLS/RTP/SAVPF 0
             a=mid:a1
             a=recvonly
             m=video 20001 UDP/TLS/RTP/SAVPF 96
             a=mid:v1
             a=recvonly

   The example in Section 7.2 shows a more involved case of "LS" group
   generation.

   The next step is to generate an "m=" section 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
   are considered to be present in each "m=" section.  Each offered "m="
   section will have an associated RtpTransceiver, as described in
   Section 5.10.  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:

   *  The associated RtpTransceiver has been stopped.

   *  There is no offered media format that is both supported and, if
      applicable, allowed by codec preferences.

   *  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.

   *  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.

   *  This "m=" section is in a bundle group, and the group's offerer
      tagged "m=" section is being rejected due to one of the above
      reasons.  This requires all "m=" sections in the bundle group to
      be rejected, as specified in [RFC8843], Section 7.3.3.

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

   *  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 default <port> value of 9 (Discard) MUST be
      used, as indicated in [RFC8840], Section 4.1.1.

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

   *  If codec preferences have been set for the associated transceiver,
      media formats MUST be generated in the corresponding order,
      regardless of what was offered, and MUST exclude any codecs not
      present in the codec preferences.

   *  Otherwise, the media formats on the "m=" line MUST be generated in
      the same order as those offered in the current remote description,
      excluding any currently unsupported formats.  Any currently
      available media formats that are not present in the current remote
      description MUST be added after all existing formats.

   *  In either case, the media formats in the answer MUST include at
      least one format that is present in the offer but MAY include
      formats that are locally supported but not present in the offer,
      as mentioned in [RFC3264], Section 6.1.  If no common format
      exists, the "m=" section is rejected as described above.

   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 default value "IN IP4
   0.0.0.0", as defined in [RFC8840], Section 4.1.3.

   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 or TRANSPORT):

   *  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.

   *  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.

   *  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.

   *  If "rtx" is present in the offer, for each primary codec where RTP
      retransmission should be used, a corresponding "a=rtpmap" line
      indicating "rtx" with the clock rate of the primary codec and an
      "a=fmtp" line that references the payload type of the primary
      codec, as specified in [RFC4588], Section 8.1.

   *  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 [RFC8854], Section 7, and
      specific usage for each media type is outlined in Sections 4 and
      5.

   *  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.

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

   *  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 [RFC8834], Section 5.2.  Any header extensions that
      require encryption MUST be specified as indicated in [RFC6904],
      Section 4.

   *  For each supported RTCP feedback mechanism that is present in the
      offer, an "a=rtcp-fb" line, as specified in [RFC4585],
      Section 4.2.  The list of RTCP feedback mechanisms that SHOULD/
      MUST be supported is specified in [RFC8834], Section 5.1.

   *  If the RtpTransceiver has a sendrecv or sendonly direction:

      -  For each MediaStream that was associated with the transceiver
         when it was created via addTrack or addTransceiver, an "a=msid"
         line, as specified in [RFC8830], Section 2, but omitting the
         "appdata" field.

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

   *  "a=ice-ufrag" and "a=ice-pwd" lines, as specified in [RFC8839],
      Section 5.4.

   *  For each desired digest algorithm, one or more "a=fingerprint"
      lines for each of the endpoint's certificates, as specified in
      [RFC8122], Section 5.

   *  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".  When
      the offer contains the "actpass" value, as will always be the case
      with JSEP endpoints, the answerer SHOULD use the "active" role.
      Offers from non-JSEP endpoints MAY send other values for
      "a=setup", in which case the answer MUST use a value consistent
      with the value in the offer.

   *  An "a=tls-id" line, as specified in [RFC8842], Section 5.3.

   *  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 default value
      "9 IN IP4 0.0.0.0" (because no candidates have yet been gathered).

   *  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, an "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, an "a=mid" line MUST be generated and
   included as described above, along with an "a=sctp-port" line
   referencing the SCTP port number, as defined in [RFC8841],
   Section 5.1; and, if appropriate, an "a=max-message-size" line, as
   defined in [RFC8841], Section 6.1.

   As discussed above, the following attributes of category IDENTICAL or
   TRANSPORT are included only if the data "m=" section is not bundled
   into another "m=" section:

   *  "a=ice-ufrag"

   *  "a=ice-pwd"

   *  "a=fingerprint"

   *  "a=setup"

   *  "a=tls-id"

   Note that if media "m=" sections are bundled into a data "m="
   section, then certain TRANSPORT and IDENTICAL attributes may also
   appear in the data "m=" section even if they would otherwise only be
   appropriate for a media "m=" section (e.g., "a=rtcp-mux").

   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 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, all "m=" sections in the
   answer MUST NOT have an "a=bundle-only" line.

   Attributes that are common between all "m=" sections MAY be moved to
   the session level if explicitly defined to be valid at the 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 previous answer was not applied using setLocalDescription,
   meaning the PeerConnection is still in the "have-remote-offer" state,
   the steps for generating an initial answer MUST be followed, subject
   to the following restriction:

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

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

   *  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 [RFC8839], Section 4.2.1.2.  Note that in certain
      cases, the "m=" line protocol may not match that of the default
      candidate, because the "m=" line protocol value MUST match what
      was supplied in the offer, as described above.

   *  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 [RFC8839], Section 4.4.1.1.1.  If
      the "m=" section is bundled into another "m=" section, it still
      MUST NOT contain any ICE credentials.

   *  Each "a=tls-id" line MUST stay the same, unless the offerer's
      "a=tls-id" line changed, in which case a new tls-id value MUST be
      created, as described in [RFC8842], Section 5.2.

   *  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.

   *  RTCP multiplexing MUST be used, and an "a=rtcp-mux" line inserted
      if and only if the "m=" section previously used RTCP multiplexing.

   *  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, default
      values MUST be used, as described in Section 5.3.1 above.

   *  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 [RFC8839], Section 5.1.  If candidate
      gathering for the section has completed, an "a=end-of-candidates"
      attribute MUST be added, as described in [RFC8840], Section 8.2.
      If the "m=" section is bundled into another "m=" section, both
      "a=candidate" and "a=end-of-candidates" MUST be omitted.

   *  For RtpTransceivers that are not stopped, the "a=msid" line(s)
      MUST stay the same, regardless of changes to the transceiver's
      direction or track.  If no "a=msid" line is present in the current
      description, "a=msid" line(s) MUST be generated according to the
      same rules as for an initial answer.

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 MUST
   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").  The impact of this rule is that an answerer
   will not try to use silence suppression with any endpoint that does
   not offer it, making silence suppression support bilateral even with
   non-JSEP endpoints.

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.

   After calling setLocalDescription with an offer or answer, the
   application MAY modify the SDP to reduce its capabilities before
   sending it to the far side, as long as it follows the rules above
   that define a valid JSEP offer or answer.  Likewise, an application
   that has received an offer or answer from a peer MAY modify the
   received SDP, subject to the same constraints, before calling
   setRemoteDescription.

   As always, the application is solely responsible for what it sends to
   the other party, and all incoming SDP will be processed by the JSEP
   implementation 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
   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:

   *  If the description is of type "rollback", follow the processing
      defined in Section 5.7 and skip the processing described in the
      rest of this section.

   *  Otherwise, 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".

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

   *  The SessionDescription is then checked to ensure that its contents
      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, the SessionDescription is parsed into a data structure, as
      described in Section 5.8 below.

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

5.6.  Processing a Remote Description

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

   *  If the description is of type "rollback", follow the processing
      defined in Section 5.7 and skip the processing described in the
      rest of this section.

   *  Otherwise, 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".

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

   *  Next, the SessionDescription is parsed into a data structure, as
      described in Section 5.8 below.  If parsing fails for any reason,
      processing MUST stop and an error MUST be returned.

   *  Finally, the parsed SessionDescription is applied as described in
      Section 5.10 below.

5.7.  Processing a Rollback

   A rollback may be performed if the PeerConnection is in any state
   except for "stable".  This means that both offers and provisional
   answers can be rolled back.  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.  If a rollback is
   attempted in the "stable" state, processing MUST stop and an error
   MUST be returned.  Note that this implies that once the answerer has
   performed setLocalDescription with its answer, this cannot be rolled
   back.

   The effect of rollback MUST be the same regardless of whether
   setLocalDescription or setRemoteDescription is called.

   In order to process rollback, a JSEP implementation abandons the
   current offer/answer transaction, sets the signaling state to
   "stable", and sets the pending local and/or remote description (see
   Sections 4.1.14 and 4.1.16) to "null".  Any resources or candidates
   that were allocated by the abandoned local description are discarded;
   any media that is received is processed according to the previous
   local and remote descriptions.

   A rollback disassociates any RtpTransceivers that were associated
   with "m=" sections by the application of the rolled-back session
   description (see Sections 5.10 and 5.9).  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, an 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 an "m=" section for the added track appear
   in the generated offer.

5.8.  Parsing a Session Description

   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 that it is syntactically correct
   according to its defining ABNF, check that it conforms to the
   semantics used in [RFC4566] and [RFC3264], 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.

5.8.1.  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.

   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 only
      used for ICE mismatch detection, as defined in [RFC8445],
      Section 5.4.  Note that JSEP implementations should never
      encounter this condition because ICE is required for WebRTC.

   *  The "i=", "u=", "e=", "p=", "t=", "r=", "z=", and "k=" lines MUST
      be checked for syntax, but their values are not otherwise used.

   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:

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

   *  If present, a single "a=ice-lite" line is parsed as specified in
      [RFC8839], Section 5.3, and a value indicating the presence of
      ice-lite is stored.

   *  If present, a single "a=ice-ufrag" line is parsed as specified in
      [RFC8839], Section 5.4, and the ufrag value is stored.

   *  If present, a single "a=ice-pwd" line is parsed as specified in
      [RFC8839], Section 5.4, and the password value is stored.

   *  If present, a single "a=ice-options" line is parsed as specified
      in [RFC8839], Section 5.6, and the set of specified options is
      stored.

   *  Any "a=fingerprint" lines are parsed as specified in [RFC8122],
      Section 5, and the set of fingerprint and algorithm values is
      stored.

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

   *  If present, a single "a=tls-id" line is parsed as specified in
      [RFC8842], Section 5, and the attribute value is stored.

   *  Any "a=identity" lines are parsed and the identity values stored
      for subsequent verification, as specified in [RFC8827], Section 5.

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

   Other attributes that are not relevant to JSEP may also be present,
   and implementations SHOULD process any that they recognize.  As
   required by [RFC4566], Section 5.13, unknown attribute lines MUST be
   ignored.

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

5.8.2.  Media Section Parsing

   Like the session-level lines, the media section 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:

   *  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.

   *  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:

   *  If present, a single "a=ice-ufrag" line is parsed as specified in
      [RFC8839], Section 5.4, and the ufrag value is stored.

   *  If present, a single "a=ice-pwd" line is parsed as specified in
      [RFC8839], Section 5.4, and the password value is stored.

   *  If present, a single "a=ice-options" line is parsed as specified
      in [RFC8839], Section 5.6, and the set of specified options is
      stored.

   *  Any "a=candidate" attributes MUST be parsed as specified in
      [RFC8839], Section 5.1, and their values stored.

   *  Any "a=remote-candidates" attributes MUST be parsed as specified
      in [RFC8839], Section 5.2, but their values are ignored.

   *  If present, a single "a=end-of-candidates" attribute MUST be
      parsed as specified in [RFC8840], Section 8.1, and its presence or
      absence flagged and stored.

   *  Any "a=fingerprint" lines are parsed as specified in [RFC8122],
      Section 5, and the set of fingerprint and algorithm values is
      stored.

   If the "m=" <proto> value indicates use of RTP, as described in
   Section 5.1.2 above, the following attribute lines MUST be processed:

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

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

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

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

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

   *  Any "a=ssrc" attributes MUST be parsed as specified in [RFC5576],
      Section 4.1, and their values stored.

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

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

   *  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.

   *  If present, a single "a=rtcp-mux-only" attribute MUST be parsed as
      specified in [RFC8858], Section 3, and its presence or absence
      flagged and stored.

   *  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.

   *  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.

   *  If present, "a=msid" attributes MUST be parsed as specified in
      [RFC8830], Section 3.2, and their values stored, ignoring any
      "appdata" field.  If no "a=msid" attributes are present, a random
      msid-id value is generated for a "default" MediaStream for the
      session, if not already present, and this value is stored.

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

   *  Any "a=rid" lines MUST be parsed as specified in [RFC8851],
      Section 10, and their values stored.

   *  If present, a single "a=simulcast" line MUST be parsed as
      specified in [RFC8853], and its values stored.

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

   *  The "m=" <fmt> value MUST be parsed as specified in [RFC8841],
      Section 4.3, and the application protocol value stored.

   *  An "a=sctp-port" attribute MUST be present, and it MUST be parsed
      as specified in [RFC8841], Section 5.2, and the value stored.

   *  If present, a single "a=max-message-size" attribute MUST be parsed
      as specified in [RFC8841], Section 6, and the value stored.
      Otherwise, use the specified default.

   Other attributes that are not relevant to JSEP may also be present,
   and implementations SHOULD process any that they recognize.  As
   required by [RFC4566], Section 5.13, unknown attribute lines MUST be
   ignored.

5.8.3.  Semantics Verification

   Assuming that 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:

   *  For each "m=" section, valid values for each of the mandatory-to-
      use features enumerated in Section 5.1.1 MUST be present.  These
      values MAY be either 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 [RFC8839], Section 5.4.

      -  A tls-id value, which MUST be set according to [RFC8842],
         Section 5.  If this is a re-offer or a response to a re-offer
         and the tls-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.

      -  A 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.

   *  All rid-ids referenced in an "a=simulcast" line MUST exist as
      "a=rid" lines.

   *  Each "m=" section is also checked to ensure that prohibited
      features are not used.

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

   *  If an "m=" section was present in the previous answer, the state
      of RTP/RTCP multiplexing MUST match what was previously
      negotiated.

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

   *  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.

   *  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.

5.9.  Applying a Local Description

   The following steps are performed at the media engine level to apply
   a local description.  If an error is returned, the session MUST be
   restored to the state it was in before performing these steps.

   First, "m=" sections are processed.  For each "m=" 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.

   *  If this "m=" section is new, begin gathering candidates for it, as
      defined in [RFC8445], Section 5.1.1, unless it is definitively
      being bundled (either (1) this is an offer and the "m=" section is
      marked bundle-only or (2) it is an answer and the "m=" section is
      bundled into another "m=" section).

   *  Or, if the ICE ufrag and password values have changed, trigger the
      ICE agent to start an ICE restart as described in [RFC8445],
      Section 9, and begin gathering new candidates for the "m="
      section.  If this description is an answer, also start checks on
      that media section.

   *  If the "m=" section <proto> value indicates use of RTP:

      -  If there is no RtpTransceiver associated with this "m="
         section, find one and associate it with this "m=" section
         according to the following steps.  Note that this situation
         will only occur when applying an offer.

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

         o  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.

      -  For each specified RTP header extension, establish a mapping
         between the extension ID and URI, as described in [RFC5285],
         Section 6.

      -  If the MID header extension is supported, prepare to demux RTP
         streams intended for this "m=" section based on the MID header
         extension, as described in [RFC8843], Section 15.

      -  For each specified media format, establish a mapping between
         the payload type and the actual media format, as described in
         [RFC3264], Section 6.1.  In addition, prepare to demux RTP
         streams intended for this "m=" section based on the media
         formats supported by this "m=" section, as described in
         [RFC8843], Section 9.2.

      -  For each specified "rtx" media format, establish a mapping
         between the RTX payload type and its associated primary payload
         type, as described in Sections 8.6 and 8.7 of [RFC4588].

      -  If the direction 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 Section 5.11 below.

5.10.  Applying a Remote Description

   The following steps are performed to apply a remote description.  If
   an error is returned, the session MUST be restored to the state it
   was in before performing these steps.

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

   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.

   *  For any specified "CT" bandwidth value, set this value as the
      limit for the maximum total bitrate for all "m=" sections, as
      specified in [RFC4566], Section 5.8.  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.

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

   *  Any "AS" bandwidth value ([RFC4566], Section 5.8) MUST be ignored,
      as the meaning of this construct at the session level is not well
      defined.

   For each "m=" 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.

   *  If the ICE ufrag or password changed from the previous remote
      description:

      -  If the description is of type "offer", the implementation MUST
         note that an ICE restart is needed, as described in [RFC8839],
         Section 4.4.1.1.1.

      -  If the description is of type "answer" or "pranswer", then
         check to see if the current local description is an ICE
         restart, and if not, generate an error.  If the PeerConnection
         state is "have-remote-pranswer" and the ICE ufrag or password
         changed from the previous provisional answer, then signal the
         ICE agent to discard any previous ICE checklist state for the
         "m=" section.  Finally, signal the ICE agent to begin checks.

   *  If the current local description indicates an ICE restart but
      neither the ICE ufrag nor the password has changed from the
      previous remote description (as prescribed by [RFC8445],
      Section 9), generate an error.

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

   *  Pair any supplied ICE candidates with any gathered local
      candidates, as described in [RFC8445], Section 6.1.2, and start
      connectivity checks with the appropriate credentials.

   *  If an "a=end-of-candidates" attribute is present, process the end-
      of-candidates indication as described in [RFC8838], Section 14.

   *  If the "m=" section <proto> value indicates use of RTP:

      -  If the "m=" section is being recycled (see Section 5.2.2),
         disassociate 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 disassociated in the previous step),
         either find an RtpTransceiver or create one according to the
         following steps:

         o  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.

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

         o  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.

      -  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.  Note that RTX payload types may refer to primary
         payload types that are not supported by the local media
         implementation, in which case the RTX payload type MUST also be
         ignored.

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

      -  For each specified Synchronization Source (SSRC) that is
         signaled in the "m=" section, prepare to demux RTP streams
         intended for this "m=" section using that SSRC, as described in
         [RFC8843], Section 9.2.

      -  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" ("Transport Independent Application
         Specific Maximum") 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 1000 changes the unit from kbps to bps (as required by
         TIAS), and the 0.95 is to allocate 5% to RTCP.  An estimate of
         header overhead is then subtracted out, in which the 50 is
         based on 50 packets per second, the 40 is based on typical
         header size (in bytes), and the 8 converts bytes to bits.  Note
         that "TIAS" is preferred over "AS" because it provides more
         accurate control of bandwidth.

      -  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 "m=" section is of type "audio":

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

         o  For each specified "telephone-event" media format, enable
            dual-tone multifrequency (DTMF) transmission for all
            supported media formats with the same clock rate, as
            described in [RFC4733], Section 2.5.1.2.  If there are any
            supported media formats that do not have a corresponding
            telephone-event format, disable DTMF transmission for those
            formats.

         o  For any specified "ptime" value, configure the available
            media formats to use the specified packet size when sending.
            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 Section 5.11 below.

5.11.  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 "m=" section, the following steps MUST be performed:

   *  If the "m=" section has been rejected (i.e., the <port> value is
      set to zero in the answer), stop any reception or transmission of
      media for this section, and, unless a non-rejected "m=" section is
      bundled with this "m=" section, discard any associated ICE
      components, as described in [RFC8839], Section 4.4.3.1.

   *  If the remote DTLS fingerprint has been changed or the value of
      the "a=tls-id" attribute has changed, tear down the DTLS
      connection.  This includes the case when the PeerConnection state
      is "have-remote-pranswer".  If a DTLS connection needs to be torn
      down but the answer does not indicate an ICE restart or, in the
      case of "have-remote-pranswer", new ICE credentials, an error MUST
      be generated.  If an ICE restart is performed without a change in
      the tls-id value or fingerprint, then the same DTLS connection is
      continued over the new ICE channel.  Note that although JSEP
      requires that answerers change the tls-id value if and only if the
      offerer does, non-JSEP answerers are permitted to change the tls-
      id value as long as the offer contained an ICE restart.  Thus,
      JSEP implementations that process DTLS data prior to receiving an
      answer MUST be prepared to receive either a ClientHello or data
      from the previous DTLS connection.

   *  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.

   *  If the "m=" section <proto> value indicates use of RTP:

      -  If the "m=" section references RTCP feedback mechanisms that
         were not present in the corresponding "m=" section in the
         offer, this indicates a negotiation problem and MUST result in
         an error.  However, new media formats and new RTP header
         extension values are permitted in the answer, as described in
         [RFC3264], Section 7 and [RFC5285], Section 6.

      -  If the "m=" 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 "m=" section has Reduced-Size RTCP enabled, configure
         the RTCP transmission for this "m=" section to use Reduced-Size
         RTCP, as specified in [RFC5506].

      -  If the direction attribute in the answer indicates that the
         JSEP implementation should be sending media ("sendonly" for
         local answers, "recvonly" for remote answers, or "sendrecv" for
         either type of answer), choose the media format to send as the
         most preferred media format from the remote description that is
         also locally supported, as discussed in Sections 6.1 and 7 of
         [RFC3264], and start transmitting RTP media using that format
         once the underlying transport layers have been established.  If
         an SSRC has not already been chosen for this outgoing RTP
         stream, choose a unique random one.  If media is already being
         transmitted, the same SSRC SHOULD be used unless the clock rate
         of the new codec is different, in which case a new SSRC MUST be
         chosen, as specified in [RFC7160], Section 4.1.

      -  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 MID header extension has been
         negotiated, include it in the outgoing RTP streams, as
         indicated in [RFC8843], Section 15.  If the RtpStreamId or
         RepairedRtpStreamId header extensions have been negotiated and
         rid-ids have been established, include these header extensions
         in the outgoing RTP streams, as indicated in [RFC8851],
         Section 4.

      -  If the "m=" section is of type "audio", and silence suppression
         was (1) configured for the send media format as a result of
         processing the remote description and (2) also enabled for that
         format in the local description, use silence suppression for
         outgoing media, in accordance with the guidance in
         Section 5.2.3.2.  If these conditions are not met, silence
         suppression MUST NOT be used for outgoing media.

      -  If simulcast has been negotiated, send the appropriate number
         of Source RTP Streams as specified in [RFC8853], Section 5.3.3.

      -  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 unique 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 clock rate, allow redundant encoding
         using the specified format for resiliency purposes, as
         discussed in [RFC8854], 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 [RFC8108], Section 5.4.1 to
         select which SSRC to use.

      -  If the direction attribute in the answer indicates that the
         JSEP implementation should not be sending media ("recvonly" for
         local answers, "sendonly" for remote answers, or "inactive" for
         either type of answer), stop transmitting all RTP media, but
         continue sending RTCP, as described in [RFC3264], Section 5.1.

   *  If the "m=" section <proto> value indicates use of SCTP:

      -  If an SCTP association exists and the remote SCTP port has
         changed, discard the existing SCTP association.  This includes
         the case when the PeerConnection state is "have-remote-
         pranswer".

      -  If no valid SCTP association exists, prepare to initiate an
         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 [RFC8841], 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 [RFC8843], Section 7.4.

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

6.  Processing RTP/RTCP

   When bundling, associating incoming RTP/RTCP with the proper "m="
   section is defined in [RFC8843], Section 9.2.  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 or sections 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 the RID mechanism
   [RFC8851] and its associated RtpStreamId and RepairedRtpStreamId
   identifiers to distinguish between multiple encoded streams and
   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
   accommodate RFC line-length restrictions, some of the SDP lines 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, including examples with
   IPv6 addresses, can be found in [SDP4WebRTC].

7.1.  Simple Example

   This section shows a very simple example that sets up a minimal
   audio/video call between two JSEP endpoints without using Trickle
   ICE.  The example in the following section provides a more detailed
   example of what could happen in a JSEP session.

   The code flow below shows Alice's endpoint initiating the session to
   Bob's endpoint.  The messages from the JavaScript application in
   Alice's browser to the JavaScript in Bob's browser, abbreviated as
   "AliceJS" and "BobJS", respectively, are assumed to flow over some
   signaling protocol via a web server.  The JavaScript 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.  The user agents (JSEP implementations) in Alice's and Bob's
   browsers, abbreviated as "AliceUA" and "BobUA", respectively, are
   both using the default bundle policy of "balanced" and the default
   RTCP mux policy of "require".

   //                  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:       ontrack events for audio and video tracks

   //                  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:   ontrack events for audio and video tracks

   //                  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
   s=-
   t=0 0
   a=ice-options:trickle ice2
   a=group:BUNDLE a1 v1
   a=group:LS a1 v1

   m=audio 10100 UDP/TLS/RTP/SAVPF 96 0 8 97 98
   c=IN IP4 203.0.113.100
   a=mid:a1
   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=fmtp:97 0-15
   a=fmtp:98 0-15
   a=maxptime:120
   a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
   a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
   a=msid:47017fee-b6c1-4162-929c-a25110252400
   a=ice-ufrag:ETEn
   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=tls-id:91bbf309c0990a6bec11e38ba2933cee
   a=rtcp:10101 IN IP4 203.0.113.100
   a=rtcp-mux
   a=rtcp-rsize
   a=candidate:1 1 udp 2113929471 203.0.113.100 10100 typ host
   a=candidate:1 2 udp 2113929470 203.0.113.100 10101 typ host
   a=end-of-candidates

   m=video 10102 UDP/TLS/RTP/SAVPF 100 101 102 103
   c=IN IP4 203.0.113.100
   a=mid:v1
   a=sendrecv
   a=rtpmap:100 VP8/90000
   a=rtpmap:101 H264/90000
   a=fmtp:101 packetization-mode=1;profile-level-id=42e01f
   a=rtpmap:102 rtx/90000
   a=fmtp:102 apt=100
   a=rtpmap:103 rtx/90000
   a=fmtp:103 apt=101
   a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
   a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
   a=rtcp-fb:100 ccm fir
   a=rtcp-fb:100 nack
   a=rtcp-fb:100 nack pli
   a=msid:47017fee-b6c1-4162-929c-a25110252400
   a=ice-ufrag:BGKk
   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
   a=setup:actpass
   a=tls-id:91bbf309c0990a6bec11e38ba2933cee
   a=rtcp:10103 IN IP4 203.0.113.100
   a=rtcp-mux
   a=rtcp-rsize
   a=candidate:1 1 udp 2113929471 203.0.113.100 10102 typ host
   a=candidate:1 2 udp 2113929470 203.0.113.100 10103 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=ice-options:trickle ice2
   a=group:BUNDLE a1 v1
   a=group:LS a1 v1

   m=audio 10200 UDP/TLS/RTP/SAVPF 96 0 8 97 98
   c=IN IP4 203.0.113.200
   a=mid:a1
   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=fmtp:97 0-15
   a=fmtp:98 0-15
   a=maxptime:120
   a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
   a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
   a=msid:61317484-2ed4-49d7-9eb7-1414322a7aae
   a=ice-ufrag:6sFv
   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=tls-id:eec3392ab83e11ceb6a0990c903fbb19
   a=rtcp-mux
   a=rtcp-rsize
   a=candidate:1 1 udp 2113929471 203.0.113.200 10200 typ host
   a=end-of-candidates

   m=video 10200 UDP/TLS/RTP/SAVPF 100 101 102 103
   c=IN IP4 203.0.113.200
   a=mid:v1
   a=sendrecv
   a=rtpmap:100 VP8/90000
   a=rtpmap:101 H264/90000
   a=fmtp:101 packetization-mode=1;profile-level-id=42e01f
   a=rtpmap:102 rtx/90000
   a=fmtp:102 apt=100
   a=rtpmap:103 rtx/90000
   a=fmtp:103 apt=101
   a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
   a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
   a=rtcp-fb:100 ccm fir
   a=rtcp-fb:100 nack
   a=rtcp-fb:100 nack pli
   a=msid:61317484-2ed4-49d7-9eb7-1414322a7aae

7.2.  Detailed Example

   This section shows a more involved example of a session between two
   JSEP endpoints.  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.  Initially, both Alice and Bob establish an audio
   channel and a data channel.  Later, Bob adds two video flows -- one
   for his video feed and one for screen sharing, both supporting FEC --
   with the video feed configured for simulcast.  Alice accepts these
   video flows but does not add video flows of her own, so they are
   handled as recvonly.  Alice also specifies a maximum video decoder
   resolution.

   //                  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:       ontrack event with audio track from Alice

   //                  candidates are sent to Bob
   AliceUA->AliceJS:   onicecandidate (host) |offer-B1-candidate-1|
   AliceJS->WebServer: signaling with |offer-B1-candidate-1|
   AliceUA->AliceJS:   onicecandidate (srflx) |offer-B1-candidate-2|
   AliceJS->WebServer: signaling with |offer-B1-candidate-2|
   AliceUA->AliceJS:   onicecandidate (relay) |offer-B1-candidate-3|
   AliceJS->WebServer: signaling with |offer-B1-candidate-3|

   WebServer->BobJS:   signaling with |offer-B1-candidate-1|
   BobJS->BobUA:       addIceCandidate with |offer-B1-candidate-1|
   WebServer->BobJS:   signaling with |offer-B1-candidate-2|
   BobJS->BobUA:       addIceCandidate with |offer-B1-candidate-2|
   WebServer->BobJS:   signaling with |offer-B1-candidate-3|
   BobJS->BobUA:       addIceCandidate with |offer-B1-candidate-3|

   //                  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:   ontrack event with audio track from Bob

   //                  candidates are sent to Alice
   BobUA->BobJS:       onicecandidate (host) |answer-B1-candidate-1|
   BobJS->WebServer:   signaling with |answer-B1-candidate-1|
   BobUA->BobJS:       onicecandidate (srflx) |answer-B1-candidate-2|
   BobJS->WebServer:   signaling with |answer-B1-candidate-2|
   BobUA->BobJS:       onicecandidate (relay) |answer-B1-candidate-3|
   BobJS->WebServer:   signaling with |answer-B1-candidate-3|

   WebServer->AliceJS: signaling with |answer-B1-candidate-1|
   AliceJS->AliceUA:   addIceCandidate with |answer-B1-candidate-1|
   WebServer->AliceJS: signaling with |answer-B1-candidate-2|
   AliceJS->AliceUA:   addIceCandidate with |answer-B1-candidate-2|
   WebServer->AliceJS: signaling with |answer-B1-candidate-3|
   AliceJS->AliceUA:   addIceCandidate with |answer-B1-candidate-3|

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

   //                  media is flowing between endpoints
   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|
   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:   ontrack event with first video track
   AliceUA->AliceJS:   ontrack event with second video track
   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 endpoints
   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:

   v=0
   o=- 4962303333179871723 1 IN IP4 0.0.0.0
   s=-
   t=0 0
   a=ice-options:trickle ice2
   a=group:BUNDLE a1 d1

   m=audio 9 UDP/TLS/RTP/SAVPF 96 0 8 97 98
   c=IN IP4 0.0.0.0
   a=mid:a1
   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=fmtp:97 0-15
   a=fmtp:98 0-15
   a=maxptime:120
   a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
   a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
   a=msid:57017fee-b6c1-4162-929c-a25110252400
   a=ice-ufrag:ATEn
   a=ice-pwd:AtSK0WpNtpUjkY4+86js7ZQl
   a=fingerprint:sha-256
                 29: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=tls-id:17f0f4ba8a5f1213faca591b58ba52a7
   a=rtcp-mux
   a=rtcp-mux-only
   a=rtcp-rsize

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

   |offer-B1-candidate-1| looks like:

   ufrag ATEn
   index 0
   mid   a1
   attr  candidate:1 1 udp 2113929471 203.0.113.100 10100 typ host

   |offer-B1-candidate-2| looks like:

   ufrag ATEn
   index 0
   mid   a1
   attr  candidate:1 1 udp 1845494015 198.51.100.100 11100 typ srflx
                   raddr 203.0.113.100 rport 10100

   |offer-B1-candidate-3| looks like:

   ufrag ATEn
   index 0
   mid   a1
   attr  candidate:1 1 udp 255 192.0.2.100 12100 typ relay
                   raddr 198.51.100.100 rport 11100

   The SDP for |answer-B1| looks like:

   v=0
   o=- 7729291447651054566 1 IN IP4 0.0.0.0
   s=-
   t=0 0
   a=ice-options:trickle ice2
   a=group:BUNDLE a1 d1

   m=audio 9 UDP/TLS/RTP/SAVPF 96 0 8 97 98
   c=IN IP4 0.0.0.0
   a=mid:a1
   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=fmtp:97 0-15
   a=fmtp:98 0-15
   a=maxptime:120
   a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
   a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
   a=msid:71317484-2ed4-49d7-9eb7-1414322a7aae
   a=ice-ufrag:7sFv
   a=ice-pwd:dOTZKZNVlO9RSGsEGM63JXT2
   a=fingerprint:sha-256
                 7B: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=tls-id:7a25ab85b195acaf3121f5a8ab4f0f71
   a=rtcp-mux
   a=rtcp-mux-only
   a=rtcp-rsize

   m=application 9 UDP/DTLS/SCTP webrtc-datachannel
   c=IN IP4 0.0.0.0
   a=mid:d1
   a=sctp-port:5000
   a=max-message-size:65536

   |answer-B1-candidate-1| looks like:

   ufrag 7sFv
   index 0
   mid   a1
   attr  candidate:1 1 udp 2113929471 203.0.113.200 10200 typ host

   |answer-B1-candidate-2| looks like:

   ufrag 7sFv
   index 0
   mid   a1
   attr  candidate:1 1 udp 1845494015 198.51.100.200 11200 typ srflx
                   raddr 203.0.113.200 rport 10200

   |answer-B1-candidate-3| looks like:

   ufrag 7sFv
   index 0
   mid   a1
   attr  candidate:1 1 udp 255 192.0.2.200 12200 typ relay
                   raddr 198.51.100.200 rport 11200

   The SDP for |offer-B2| is shown below.  In addition to the new "m="
   sections for video, both of which are offering FEC and one of which
   is offering simulcast, note the increment of the version number in
   the "o=" line; changes to the "c=" line, indicating the local
   candidate that was selected; and the inclusion of gathered candidates
   as a=candidate lines.

   v=0
   o=- 7729291447651054566 2 IN IP4 0.0.0.0
   s=-
   t=0 0
   a=ice-options:trickle ice2
   a=group:BUNDLE a1 d1 v1 v2
   a=group:LS a1 v1

   m=audio 12200 UDP/TLS/RTP/SAVPF 96 0 8 97 98
   c=IN IP4 192.0.2.200
   a=mid:a1
   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=fmtp:97 0-15
   a=fmtp:98 0-15
   a=maxptime:120
   a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
   a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
   a=msid:71317484-2ed4-49d7-9eb7-1414322a7aae
   a=ice-ufrag:7sFv
   a=ice-pwd:dOTZKZNVlO9RSGsEGM63JXT2
   a=fingerprint:sha-256
                 7B: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=tls-id:7a25ab85b195acaf3121f5a8ab4f0f71
   a=rtcp-mux
   a=rtcp-mux-only
   a=rtcp-rsize
   a=candidate:1 1 udp 2113929471 203.0.113.200 10200 typ host
   a=candidate:1 1 udp 1845494015 198.51.100.200 11200 typ srflx
               raddr 203.0.113.200 rport 10200
   a=candidate:1 1 udp 255 192.0.2.200 12200 typ relay
               raddr 198.51.100.200 rport 11200
   a=end-of-candidates

   m=application 12200 UDP/DTLS/SCTP webrtc-datachannel
   c=IN IP4 192.0.2.200
   a=mid:d1
   a=sctp-port:5000
   a=max-message-size:65536

   m=video 12200 UDP/TLS/RTP/SAVPF 100 101 102 103 104
   c=IN IP4 192.0.2.200
   a=mid:v1
   a=sendrecv
   a=rtpmap:100 VP8/90000
   a=rtpmap:101 H264/90000
   a=fmtp:101 packetization-mode=1;profile-level-id=42e01f
   a=rtpmap:102 rtx/90000
   a=fmtp:102 apt=100
   a=rtpmap:103 rtx/90000
   a=fmtp:103 apt=101
   a=rtpmap:104 flexfec/90000
   a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
   a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
   a=rtcp-fb:100 ccm fir
   a=rtcp-fb:100 nack
   a=rtcp-fb:100 nack pli
   a=msid:71317484-2ed4-49d7-9eb7-1414322a7aae
   a=rid:1 send
   a=rid:2 send
   a=rid:3 send
   a=simulcast:send 1;2;3

   m=video 12200 UDP/TLS/RTP/SAVPF 100 101 102 103 104
   c=IN IP4 192.0.2.200
   a=mid:v2
   a=sendrecv
   a=rtpmap:100 VP8/90000
   a=rtpmap:101 H264/90000
   a=fmtp:101 packetization-mode=1;profile-level-id=42e01f
   a=rtpmap:102 rtx/90000
   a=fmtp:102 apt=100
   a=rtpmap:103 rtx/90000
   a=fmtp:103 apt=101
   a=rtpmap:104 flexfec/90000
   a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
   a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
   a=rtcp-fb:100 ccm fir
   a=rtcp-fb:100 nack
   a=rtcp-fb:100 nack pli
   a=msid:81317484-2ed4-49d7-9eb7-1414322a7aae

   The SDP for |answer-B2| is shown below.  In addition to the
   acceptance of the video "m=" sections, the use of a=recvonly to
   indicate one-way video, and the use of a=imageattr to limit the
   received resolution, note the use of setup:passive to maintain the
   existing DTLS roles.

   v=0
   o=- 4962303333179871723 2 IN IP4 0.0.0.0
   s=-
   t=0 0
   a=ice-options:trickle ice2
   a=group:BUNDLE a1 d1 v1 v2
   a=group:LS a1 v1

   m=audio 12100 UDP/TLS/RTP/SAVPF 96 0 8 97 98
   c=IN IP4 192.0.2.100
   a=mid:a1
   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=fmtp:97 0-15
   a=fmtp:98 0-15
   a=maxptime:120
   a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
   a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
   a=msid:57017fee-b6c1-4162-929c-a25110252400
   a=ice-ufrag:ATEn
   a=ice-pwd:AtSK0WpNtpUjkY4+86js7ZQl
   a=fingerprint:sha-256
                 29: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=tls-id:17f0f4ba8a5f1213faca591b58ba52a7
   a=rtcp-mux
   a=rtcp-mux-only
   a=rtcp-rsize
   a=candidate:1 1 udp 2113929471 203.0.113.100 10100 typ host
   a=candidate:1 1 udp 1845494015 198.51.100.100 11100 typ srflx
               raddr 203.0.113.100 rport 10100
   a=candidate:1 1 udp 255 192.0.2.100 12100 typ relay
               raddr 198.51.100.100 rport 11100
   a=end-of-candidates

   m=application 12100 UDP/DTLS/SCTP webrtc-datachannel
   c=IN IP4 192.0.2.100
   a=mid:d1
   a=sctp-port:5000
   a=max-message-size:65536

   m=video 12100 UDP/TLS/RTP/SAVPF 100 101 102 103
   c=IN IP4 192.0.2.100
   a=mid:v1
   a=recvonly
   a=rtpmap:100 VP8/90000
   a=rtpmap:101 H264/90000
   a=fmtp:101 packetization-mode=1;profile-level-id=42e01f
   a=rtpmap:102 rtx/90000
   a=fmtp:102 apt=100
   a=rtpmap:103 rtx/90000
   a=fmtp:103 apt=101
   a=imageattr:100 recv [x=[48:1920],y=[48:1080],q=1.0]
   a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
   a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
   a=rtcp-fb:100 ccm fir
   a=rtcp-fb:100 nack
   a=rtcp-fb:100 nack pli

   m=video 12100 UDP/TLS/RTP/SAVPF 100 101 102 103
   c=IN IP4 192.0.2.100
   a=mid:v2
   a=recvonly
   a=rtpmap:100 VP8/90000
   a=rtpmap:101 H264/90000
   a=fmtp:101 packetization-mode=1;profile-level-id=42e01f
   a=rtpmap:102 rtx/90000
   a=fmtp:102 apt=100
   a=rtpmap:103 rtx/90000
   a=fmtp:103 apt=101
   a=imageattr:100 recv [x=[48:1920],y=[48:1080],q=1.0]
   a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
   a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
   a=rtcp-fb:100 ccm fir
   a=rtcp-fb:100 nack
   a=rtcp-fb:100 nack pli

7.3.  Early Transport Warmup Example

   This example demonstrates the early-warmup technique described in
   Section 4.1.10.1.  Here, Alice's endpoint sends an offer to Bob's
   endpoint to start an audio/video call.  Bob immediately responds with
   an answer that accepts the audio/video "m=" sections but marks them
   as sendonly (from his perspective), meaning that Alice will not yet
   send media.  This allows the JSEP implementation to start negotiating
   ICE and DTLS immediately.  Bob's endpoint then prompts him to answer
   the call, and when he does, his endpoint sends a second offer, which
   enables the audio and video "m=" sections, and thereby bidirectional
   media transmission.  The advantage of such a flow is that as soon as
   the first answer is received, the implementation can proceed with ICE
   and DTLS negotiation and establish the session transport.  If the
   transport setup completes before the second offer is sent, then media
   can be transmitted by the callee immediately upon answering the call,
   minimizing perceived post-dial delay.  The second offer/answer
   exchange can also change the preferred codecs or other session
   parameters.

   This example also makes use of the "relay" ICE candidate policy
   described in Section 3.5.3 to minimize the ICE gathering and checking
   needed.

   //                  set up local media state
   AliceJS->AliceUA:   create new PeerConnection with "relay" ICE policy
   AliceJS->AliceUA:   addTrack with two tracks: audio and video
   AliceJS->AliceUA:   createOffer to get |offer-C1|
   AliceJS->AliceUA:   setLocalDescription with |offer-C1|

   //                  |offer-C1| is sent over signaling protocol to Bob
   AliceJS->WebServer: signaling with |offer-C1|
   WebServer->BobJS:   signaling with |offer-C1|

   //                  |offer-C1| arrives at Bob
   BobJS->BobUA:       create new PeerConnection with "relay" ICE policy
   BobJS->BobUA:       setRemoteDescription with |offer-C1|
   BobUA->BobJS:       ontrack events for audio and video

   //                  a relay candidate is sent to Bob
   AliceUA->AliceJS:   onicecandidate (relay) |offer-C1-candidate-1|
   AliceJS->WebServer: signaling with |offer-C1-candidate-1|

   WebServer->BobJS:   signaling with |offer-C1-candidate-1|
   BobJS->BobUA:       addIceCandidate with |offer-C1-candidate-1|

   //                  Bob prepares an early answer to warm up the
   //                  transport
   BobJS->BobUA:       addTransceiver with null audio and video tracks
   BobJS->BobUA:       transceiver.setDirection(sendonly) for both
   BobJS->BobUA:       createAnswer
   BobJS->BobUA:       setLocalDescription with answer

   //                  |answer-C1| is sent over signaling protocol
   //                  to Alice
   BobJS->WebServer:   signaling with |answer-C1|
   WebServer->AliceJS: signaling with |answer-C1|

   //                  |answer-C1| (sendonly) arrives at Alice
   AliceJS->AliceUA:   setRemoteDescription with |answer-C1|
   AliceUA->AliceJS:   ontrack events for audio and video

   //                  a relay candidate is sent to Alice
   BobUA->BobJS:       onicecandidate (relay) |answer-B1-candidate-1|
   BobJS->WebServer:   signaling with |answer-B1-candidate-1|

   WebServer->AliceJS: signaling with |answer-B1-candidate-1|
   AliceJS->AliceUA:   addIceCandidate with |answer-B1-candidate-1|

   //                  ICE and DTLS establish while call is ringing

   //                  Bob accepts call, starts media, and sends
   //                  new offer
   BobJS->BobUA:       transceiver.setTrack with audio and video tracks
   BobUA->AliceUA:     media sent from Bob to Alice
   BobJS->BobUA:       transceiver.setDirection(sendrecv) for both
                       transceivers
   BobJS->BobUA:       createOffer
   BobJS->BobUA:       setLocalDescription with offer

   //                  |offer-C2| is sent over signaling protocol
   //                  to Alice
   BobJS->WebServer:   signaling with |offer-C2|
   WebServer->AliceJS: signaling with |offer-C2|

   //                  |offer-C2| (sendrecv) arrives at Alice
   AliceJS->AliceUA:   setRemoteDescription with |offer-C2|
   AliceJS->AliceUA:   createAnswer
   AliceJS->AliceUA:   setLocalDescription with |answer-C2|
   AliceUA->BobUA:     media sent from Alice to Bob

   //                  |answer-C2| is sent over signaling protocol
   //                  to Bob
   AliceJS->WebServer: signaling with |answer-C2|
   WebServer->BobJS:   signaling with |answer-C2|
   BobJS->BobUA:       setRemoteDescription with |answer-C2|

   The SDP for |offer-C1| looks like:

   v=0
   o=- 1070771854436052752 1 IN IP4 0.0.0.0
   s=-
   t=0 0
   a=ice-options:trickle ice2
   a=group:BUNDLE a1 v1
   a=group:LS a1 v1

   m=audio 9 UDP/TLS/RTP/SAVPF 96 0 8 97 98
   c=IN IP4 0.0.0.0
   a=mid:a1
   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=fmtp:97 0-15
   a=fmtp:98 0-15
   a=maxptime:120
   a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
   a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
   a=msid:bbce3ba6-abfc-ac63-d00a-e15b286f8fce
   a=ice-ufrag:4ZcD
   a=ice-pwd:ZaaG6OG7tCn4J/lehAGz+HHD
   a=fingerprint:sha-256
                 C4:68:F8:77:6A:44:F1:98:6D:7C:9F:47:EB:E3:34:A4:
                 0A:AA:2D:49:08:28:70:2E:1F:AE:18:7D:4E:3E:66:BF
   a=setup:actpass
   a=tls-id:9e5b948ade9c3d41de6617b68f769e55
   a=rtcp-mux
   a=rtcp-mux-only
   a=rtcp-rsize

   m=video 0 UDP/TLS/RTP/SAVPF 100 101 102 103
   c=IN IP4 0.0.0.0
   a=mid:v1
   a=sendrecv
   a=rtpmap:100 VP8/90000
   a=rtpmap:101 H264/90000
   a=fmtp:101 packetization-mode=1;profile-level-id=42e01f
   a=rtpmap:102 rtx/90000
   a=fmtp:102 apt=100
   a=rtpmap:103 rtx/90000
   a=fmtp:103 apt=101
   a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
   a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
   a=rtcp-fb:100 ccm fir
   a=rtcp-fb:100 nack
   a=rtcp-fb:100 nack pli
   a=msid:bbce3ba6-abfc-ac63-d00a-e15b286f8fce
   a=bundle-only

   |offer-C1-candidate-1| looks like:

   ufrag 4ZcD
   index 0
   mid   a1
   attr  candidate:1 1 udp 255 192.0.2.100 12100 typ relay
                   raddr 0.0.0.0 rport 0

   The SDP for |answer-C1| looks like:

   v=0
   o=- 6386516489780559513 1 IN IP4 0.0.0.0
   s=-
   t=0 0
   a=ice-options:trickle ice2
   a=group:BUNDLE a1 v1
   a=group:LS a1 v1

   m=audio 9 UDP/TLS/RTP/SAVPF 96 0 8 97 98
   c=IN IP4 0.0.0.0
   a=mid:a1
   a=sendonly
   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=fmtp:97 0-15
   a=fmtp:98 0-15
   a=maxptime:120
   a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
   a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
   a=msid:751f239e-4ae0-c549-aa3d-890de772998b
   a=ice-ufrag:TpaA
   a=ice-pwd:t2Ouhc67y8JcCaYZxUUTgKw/
   a=fingerprint:sha-256
                 A2:F3:A5:6D:4C:8C:1E:B2:62:10:4A:F6:70:61:C4:FC:
                 3C:E0:01:D6:F3:24:80:74:DA:7C:3E:50:18:7B:CE:4D
   a=setup:active
   a=tls-id:55e967f86b7166ed14d3c9eda849b5e9
   a=rtcp-mux
   a=rtcp-mux-only
   a=rtcp-rsize

   m=video 9 UDP/TLS/RTP/SAVPF 100 101 102 103
   c=IN IP4 0.0.0.0
   a=mid:v1
   a=sendonly
   a=rtpmap:100 VP8/90000
   a=rtpmap:101 H264/90000
   a=fmtp:101 packetization-mode=1;profile-level-id=42e01f
   a=rtpmap:102 rtx/90000
   a=fmtp:102 apt=100
   a=rtpmap:103 rtx/90000
   a=fmtp:103 apt=101
   a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
   a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
   a=rtcp-fb:100 ccm fir
   a=rtcp-fb:100 nack
   a=rtcp-fb:100 nack pli
   a=msid:751f239e-4ae0-c549-aa3d-890de772998b

   |answer-C1-candidate-1| looks like:

   ufrag TpaA
   index 0
   mid   a1
   attr  candidate:1 1 udp 255 192.0.2.200 12200 typ relay
                   raddr 0.0.0.0 rport 0

   The SDP for |offer-C2| looks like:

   v=0
   o=- 6386516489780559513 2 IN IP4 0.0.0.0
   s=-
   t=0 0
   a=ice-options:trickle ice2
   a=group:BUNDLE a1 v1
   a=group:LS a1 v1

   m=audio 12200 UDP/TLS/RTP/SAVPF 96 0 8 97 98
   c=IN IP4 192.0.2.200
   a=mid:a1
   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=fmtp:97 0-15
   a=fmtp:98 0-15
   a=maxptime:120
   a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
   a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
   a=msid:751f239e-4ae0-c549-aa3d-890de772998b
   a=ice-ufrag:TpaA
   a=ice-pwd:t2Ouhc67y8JcCaYZxUUTgKw/
   a=fingerprint:sha-256
                 A2:F3:A5:6D:4C:8C:1E:B2:62:10:4A:F6:70:61:C4:FC:
                 3C:E0:01:D6:F3:24:80:74:DA:7C:3E:50:18:7B:CE:4D
   a=setup:actpass
   a=tls-id:55e967f86b7166ed14d3c9eda849b5e9
   a=rtcp-mux
   a=rtcp-mux-only
   a=rtcp-rsize
   a=candidate:1 1 udp 255 192.0.2.200 12200 typ relay
               raddr 0.0.0.0 rport 0
   a=end-of-candidates

   m=video 12200 UDP/TLS/RTP/SAVPF 100 101 102 103
   c=IN IP4 192.0.2.200
   a=mid:v1
   a=sendrecv
   a=rtpmap:100 VP8/90000
   a=rtpmap:101 H264/90000
   a=fmtp:101 packetization-mode=1;profile-level-id=42e01f
   a=rtpmap:102 rtx/90000
   a=fmtp:102 apt=100
   a=rtpmap:103 rtx/90000
   a=fmtp:103 apt=101
   a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
   a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
   a=rtcp-fb:100 ccm fir
   a=rtcp-fb:100 nack
   a=rtcp-fb:100 nack pli
   a=msid:751f239e-4ae0-c549-aa3d-890de772998b

   The SDP for |answer-C2| looks like:

   v=0
   o=- 1070771854436052752 2 IN IP4 0.0.0.0
   s=-
   t=0 0
   a=ice-options:trickle ice2
   a=group:BUNDLE a1 v1
   a=group:LS a1 v1

   m=audio 12100 UDP/TLS/RTP/SAVPF 96 0 8 97 98
   c=IN IP4 192.0.2.100
   a=mid:a1
   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=fmtp:97 0-15
   a=fmtp:98 0-15
   a=maxptime:120
   a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
   a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
   a=msid:bbce3ba6-abfc-ac63-d00a-e15b286f8fce
   a=ice-ufrag:4ZcD
   a=ice-pwd:ZaaG6OG7tCn4J/lehAGz+HHD
   a=fingerprint:sha-256
                 C4:68:F8:77:6A:44:F1:98:6D:7C:9F:47:EB:E3:34:A4:
                 0A:AA:2D:49:08:28:70:2E:1F:AE:18:7D:4E:3E:66:BF
   a=setup:passive
   a=tls-id:9e5b948ade9c3d41de6617b68f769e55
   a=rtcp-mux
   a=rtcp-mux-only
   a=rtcp-rsize
   a=candidate:1 1 udp 255 192.0.2.100 12100 typ relay
               raddr 0.0.0.0 rport 0
   a=end-of-candidates

   m=video 12100 UDP/TLS/RTP/SAVPF 100 101 102 103
   c=IN IP4 192.0.2.100
   a=mid:v1
   a=sendrecv
   a=rtpmap:100 VP8/90000
   a=rtpmap:101 H264/90000
   a=fmtp:101 packetization-mode=1;profile-level-id=42e01f
   a=rtpmap:102 rtx/90000
   a=fmtp:102 apt=100
   a=rtpmap:103 rtx/90000
   a=fmtp:103 apt=101
   a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
   a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
   a=rtcp-fb:100 ccm fir
   a=rtcp-fb:100 nack
   a=rtcp-fb:100 nack pli
   a=msid:bbce3ba6-abfc-ac63-d00a-e15b286f8fce

8.  Security Considerations

   The IETF has published separate documents [RFC8827] [RFC8826]
   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 as an Internet protocol, with the application JavaScript being
   untrustworthy from the perspective of the JSEP implementation.  Thus,
   the threat model of [RFC3552] applies.  In particular, JavaScript can
   call the API in any order and with any inputs, including malicious
   ones.  This is particularly relevant when we consider the SDP that is
   passed to setLocalDescription.  While correct API usage requires that
   the application pass in SDP that was derived from createOffer or
   createAnswer, there is no guarantee that applications do so.  The
   JSEP implementation MUST be prepared for the JavaScript to pass in
   bogus data instead.

   Conversely, the application programmer needs to be aware that the
   JavaScript does not have complete control of endpoint 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 that wish to conceal their public IP
   address MUST instead configure the ICE agent to use only relay
   candidates.

9.  IANA Considerations

   This document has no IANA actions.

10.  References

10.1.  Normative References

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119,
              DOI 10.17487/RFC2119, March 1997,
              <https://www.rfc-editor.org/info/rfc2119>.

   [RFC3261]  Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
              A., Peterson, J., Sparks, R., Handley, M., and E.
              Schooler, "SIP: Session Initiation Protocol", RFC 3261,
              DOI 10.17487/RFC3261, June 2002,
              <https://www.rfc-editor.org/info/rfc3261>.

   [RFC3264]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
              with Session Description Protocol (SDP)", RFC 3264,
              DOI 10.17487/RFC3264, June 2002,
              <https://www.rfc-editor.org/info/rfc3264>.

   [RFC3552]  Rescorla, E. and B. Korver, "Guidelines for Writing RFC
              Text on Security Considerations", BCP 72, RFC 3552,
              DOI 10.17487/RFC3552, July 2003,
              <https://www.rfc-editor.org/info/rfc3552>.

   [RFC3605]  Huitema, C., "Real Time Control Protocol (RTCP) attribute
              in Session Description Protocol (SDP)", RFC 3605,
              DOI 10.17487/RFC3605, October 2003,
              <https://www.rfc-editor.org/info/rfc3605>.

   [RFC3711]  Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
              Norrman, "The Secure Real-time Transport Protocol (SRTP)",
              RFC 3711, DOI 10.17487/RFC3711, March 2004,
              <https://www.rfc-editor.org/info/rfc3711>.

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

   [RFC4145]  Yon, D. and G. Camarillo, "TCP-Based Media Transport in
              the Session Description Protocol (SDP)", RFC 4145,
              DOI 10.17487/RFC4145, September 2005,
              <https://www.rfc-editor.org/info/rfc4145>.

   [RFC4566]  Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
              Description Protocol", RFC 4566, DOI 10.17487/RFC4566,
              July 2006, <https://www.rfc-editor.org/info/rfc4566>.

   [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,
              DOI 10.17487/RFC4585, July 2006,
              <https://www.rfc-editor.org/info/rfc4585>.

   [RFC5124]  Ott, J. and E. Carrara, "Extended Secure RTP Profile for
              Real-time Transport Control Protocol (RTCP)-Based Feedback
              (RTP/SAVPF)", RFC 5124, DOI 10.17487/RFC5124, February
              2008, <https://www.rfc-editor.org/info/rfc5124>.

   [RFC5285]  Singer, D. and H. Desineni, "A General Mechanism for RTP
              Header Extensions", RFC 5285, DOI 10.17487/RFC5285, July
              2008, <https://www.rfc-editor.org/info/rfc5285>.

   [RFC5761]  Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
              Control Packets on a Single Port", RFC 5761,
              DOI 10.17487/RFC5761, April 2010,
              <https://www.rfc-editor.org/info/rfc5761>.

   [RFC5888]  Camarillo, G. and H. Schulzrinne, "The Session Description
              Protocol (SDP) Grouping Framework", RFC 5888,
              DOI 10.17487/RFC5888, June 2010,
              <https://www.rfc-editor.org/info/rfc5888>.

   [RFC6236]  Johansson, I. and K. Jung, "Negotiation of Generic Image
              Attributes in the Session Description Protocol (SDP)",
              RFC 6236, DOI 10.17487/RFC6236, May 2011,
              <https://www.rfc-editor.org/info/rfc6236>.

   [RFC6347]  Rescorla, E. and N. Modadugu, "Datagram Transport Layer
              Security Version 1.2", RFC 6347, DOI 10.17487/RFC6347,
              January 2012, <https://www.rfc-editor.org/info/rfc6347>.

   [RFC6716]  Valin, JM., Vos, K., and T. Terriberry, "Definition of the
              Opus Audio Codec", RFC 6716, DOI 10.17487/RFC6716,
              September 2012, <https://www.rfc-editor.org/info/rfc6716>.

   [RFC6904]  Lennox, J., "Encryption of Header Extensions in the Secure
              Real-time Transport Protocol (SRTP)", RFC 6904,
              DOI 10.17487/RFC6904, April 2013,
              <https://www.rfc-editor.org/info/rfc6904>.

   [RFC7160]  Petit-Huguenin, M. and G. Zorn, Ed., "Support for Multiple
              Clock Rates in an RTP Session", RFC 7160,
              DOI 10.17487/RFC7160, April 2014,
              <https://www.rfc-editor.org/info/rfc7160>.

   [RFC7587]  Spittka, J., Vos, K., and JM. Valin, "RTP Payload Format
              for the Opus Speech and Audio Codec", RFC 7587,
              DOI 10.17487/RFC7587, June 2015,
              <https://www.rfc-editor.org/info/rfc7587>.

   [RFC7742]  Roach, A.B., "WebRTC Video Processing and Codec
              Requirements", RFC 7742, DOI 10.17487/RFC7742, March 2016,
              <https://www.rfc-editor.org/info/rfc7742>.

   [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,
              <https://www.rfc-editor.org/info/rfc7850>.

   [RFC7874]  Valin, JM. and C. Bran, "WebRTC Audio Codec and Processing
              Requirements", RFC 7874, DOI 10.17487/RFC7874, May 2016,
              <https://www.rfc-editor.org/info/rfc7874>.

   [RFC8108]  Lennox, J., Westerlund, M., Wu, Q., and C. Perkins,
              "Sending Multiple RTP Streams in a Single RTP Session",
              RFC 8108, DOI 10.17487/RFC8108, March 2017,
              <https://www.rfc-editor.org/info/rfc8108>.

   [RFC8122]  Lennox, J. and C. Holmberg, "Connection-Oriented Media
              Transport over the Transport Layer Security (TLS) Protocol
              in the Session Description Protocol (SDP)", RFC 8122,
              DOI 10.17487/RFC8122, March 2017,
              <https://www.rfc-editor.org/info/rfc8122>.

   [RFC8174]  Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC
              2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174,
              May 2017, <https://www.rfc-editor.org/info/rfc8174>.

   [RFC8445]  Keranen, A., Holmberg, C., and J. Rosenberg, "Interactive
              Connectivity Establishment (ICE): A Protocol for Network
              Address Translator (NAT) Traversal", RFC 8445,
              DOI 10.17487/RFC8445, July 2018,
              <https://www.rfc-editor.org/info/rfc8445>.

   [RFC8826]  Rescorla, E., "Security Considerations for WebRTC",
              RFC 8826, DOI 10.17487/RFC8826, January 2021,
              <https://www.rfc-editor.org/info/rfc8826>.

   [RFC8827]  Rescorla, E., "WebRTC Security Architecture", RFC 8827,
              DOI 10.17487/RFC8827, January 2021,
              <https://www.rfc-editor.org/info/rfc8827>.

   [RFC8830]  Alvestrand, H., "WebRTC MediaStream Identification in the
              Session Description Protocol", RFC 8830,
              DOI 10.17487/RFC8830, January 2021,
              <https://www.rfc-editor.org/info/rfc8830>.

   [RFC8834]  Perkins, C., Westerlund, M., and J. Ott, "Media Transport
              and Use of RTP in WebRTC", RFC 8834, DOI 10.17487/RFC8834,
              January 2021, <https://www.rfc-editor.org/info/rfc8834>.

   [RFC8838]  Ivov, E., Uberti, J., and P. Saint-Andre, "Trickle ICE:
              Incremental Provisioning of Candidates for the Interactive
              Connectivity Establishment (ICE) Protocol", RFC 8838,
              DOI 10.17487/RFC8838, January 2021,
              <https://www.rfc-editor.org/info/rfc8838>.

   [RFC8839]  Petit-Huguenin, M., Nandakumar, S., Holmberg, C., Keränen,
              A., and R. Shpount, "Session Description Protocol (SDP)
              Offer/Answer Procedures for Interactive Connectivity
              Establishment (ICE)", RFC 8839, DOI 10.17487/RFC8839,
              January 2021, <https://www.rfc-editor.org/info/rfc8839>.

   [RFC8840]  Ivov, E., Stach, T., Marocco, E., and C. Holmberg, "A
              Session Initiation Protocol (SIP) Usage for Incremental
              Provisioning of Candidates for the Interactive
              Connectivity Establishment (Trickle ICE)", RFC 8840,
              DOI 10.17487/RFC8840, January 2021,
              <https://www.rfc-editor.org/info/rfc8840>.

   [RFC8841]  Holmberg, C., Shpount, R., Loreto, S., and G. Camarillo,
              "Session Description Protocol (SDP) Offer/Answer
              Procedures for Stream Control Transmission Protocol (SCTP)
              over Datagram Transport Layer Security (DTLS) Transport",
              RFC 8841, DOI 10.17487/RFC8841, January 2021,
              <https://www.rfc-editor.org/info/rfc8841>.

   [RFC8842]  Holmberg, C. and R. Shpount, "Session Description Protocol
              (SDP) Offer/Answer Considerations for Datagram Transport
              Layer Security (DTLS) and Transport Layer Security (TLS)",
              RFC 8842, DOI 10.17487/RFC8842, January 2021,
              <https://www.rfc-editor.org/info/rfc8842>.

   [RFC8843]  Holmberg, C., Alvestrand, H., and C. Jennings,
              "Negotiating Media Multiplexing Using the Session
              Description Protocol (SDP)", RFC 8843,
              DOI 10.17487/RFC8843, January 2021,
              <https://www.rfc-editor.org/info/rfc8843>.

   [RFC8851]  Roach, A.B., Ed., "RTP Payload Format Restrictions",
              RFC 8851, DOI 10.17487/RFC8851, January 2021,
              <https://www.rfc-editor.org/info/rfc8851>.

   [RFC8852]  Roach, A.B., Nandakumar, S., and P. Thatcher, "RTP Stream
              Identifier Source Description (SDES)", RFC 8852,
              DOI 10.17487/RFC8852, January 2021,
              <https://www.rfc-editor.org/info/rfc8852>.

   [RFC8853]  Burman, B., Westerlund, M., Nandakumar, S., and M. Zanaty,
              "Using Simulcast in Session Description Protocol (SDP) and
              RTP Sessions", RFC 8853, DOI 10.17487/RFC8853, January
              2021, <https://www.rfc-editor.org/info/rfc8853>.

   [RFC8854]  Uberti, J., "WebRTC Forward Error Correction
              Requirements", RFC 8854, DOI 10.17487/RFC8854, January
              2021, <https://www.rfc-editor.org/info/rfc8854>.

   [RFC8858]  Holmberg, C., "Indicating Exclusive Support of RTP and RTP
              Control Protocol (RTCP) Multiplexing Using the Session
              Description Protocol (SDP)", RFC 8858,
              DOI 10.17487/RFC8858, January 2021,
              <https://www.rfc-editor.org/info/rfc8858>.

   [RFC8859]  Nandakumar, S., "A Framework for Session Description
              Protocol (SDP) Attributes When Multiplexing", RFC 8859,
              DOI 10.17487/RFC8859, January 2021,
              <https://www.rfc-editor.org/info/rfc8859>.

10.2.  Informative References

   [RFC3389]  Zopf, R., "Real-time Transport Protocol (RTP) Payload for
              Comfort Noise (CN)", RFC 3389, DOI 10.17487/RFC3389,
              September 2002, <https://www.rfc-editor.org/info/rfc3389>.

   [RFC3556]  Casner, S., "Session Description Protocol (SDP) Bandwidth
              Modifiers for RTP Control Protocol (RTCP) Bandwidth",
              RFC 3556, DOI 10.17487/RFC3556, July 2003,
              <https://www.rfc-editor.org/info/rfc3556>.

   [RFC3960]  Camarillo, G. and H. Schulzrinne, "Early Media and Ringing
              Tone Generation in the Session Initiation Protocol (SIP)",
              RFC 3960, DOI 10.17487/RFC3960, December 2004,
              <https://www.rfc-editor.org/info/rfc3960>.

   [RFC4568]  Andreasen, F., Baugher, M., and D. Wing, "Session
              Description Protocol (SDP) Security Descriptions for Media
              Streams", RFC 4568, DOI 10.17487/RFC4568, July 2006,
              <https://www.rfc-editor.org/info/rfc4568>.

   [RFC4588]  Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
              Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
              DOI 10.17487/RFC4588, July 2006,
              <https://www.rfc-editor.org/info/rfc4588>.

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

   [RFC5245]  Rosenberg, J., "Interactive Connectivity Establishment
              (ICE): A Protocol for Network Address Translator (NAT)
              Traversal for Offer/Answer Protocols", RFC 5245,
              DOI 10.17487/RFC5245, April 2010,
              <https://www.rfc-editor.org/info/rfc5245>.

   [RFC5506]  Johansson, I. and M. Westerlund, "Support for Reduced-Size
              Real-Time Transport Control Protocol (RTCP): Opportunities
              and Consequences", RFC 5506, DOI 10.17487/RFC5506, April
              2009, <https://www.rfc-editor.org/info/rfc5506>.

   [RFC5576]  Lennox, J., Ott, J., and T. Schierl, "Source-Specific
              Media Attributes in the Session Description Protocol
              (SDP)", RFC 5576, DOI 10.17487/RFC5576, June 2009,
              <https://www.rfc-editor.org/info/rfc5576>.

   [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, DOI 10.17487/RFC5763, May
              2010, <https://www.rfc-editor.org/info/rfc5763>.

   [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,
              DOI 10.17487/RFC5764, May 2010,
              <https://www.rfc-editor.org/info/rfc5764>.

   [RFC6120]  Saint-Andre, P., "Extensible Messaging and Presence
              Protocol (XMPP): Core", RFC 6120, DOI 10.17487/RFC6120,
              March 2011, <https://www.rfc-editor.org/info/rfc6120>.

   [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,
              <https://www.rfc-editor.org/info/rfc6464>.

   [RFC8828]  Uberti, J. and G. Shieh, "WebRTC IP Address Handling
              Requirements", RFC 8828, DOI 10.17487/RFC8828, January
              2021, <https://www.rfc-editor.org/info/rfc8828>.

   [SDP4WebRTC]
              Nandakumar, S. and C. Jennings, "Annotated Example SDP for
              WebRTC", Work in Progress, Internet-Draft, draft-ietf-
              rtcweb-sdp-14, 17 December 2020,
              <https://tools.ietf.org/html/draft-ietf-rtcweb-sdp-14>.

   [TS26.114] 3GPP, "3rd Generation Partnership Project; Technical
              Specification Group Services and System Aspects; IP
              Multimedia Subsystem (IMS); Multimedia Telephony; Media
              handling and interaction (Release 16)", 3GPP TS 26.114
              V16.3.0, September 2019,
              <https://www.3gpp.org/DynaReport/26114.htm>.

   [W3C.webrtc]
              Jennings, C., Ed., Boström, H., Ed., and J. Bruaroey, Ed.,
              "WebRTC 1.0: Real-time Communication Between Browsers",
              World Wide Web Consortium PR PR-webrtc-20201215, December
              2020, <https://www.w3.org/TR/2020/PR-webrtc-20201215/>.

Appendix A.  SDP ABNF Syntax

   For the syntax validation performed in Section 5.8, the following
   list of ABNF definitions is used:

          +=========================+==========================+
          | Attribute               | Reference                |
          +=========================+==========================+
          | ptime                   | Section 6 of [RFC4566]   |
          +-------------------------+--------------------------+
          | maxptime                | Section 6 of [RFC4566]   |
          +-------------------------+--------------------------+
          | rtpmap                  | Section 6 of [RFC4566]   |
          +-------------------------+--------------------------+
          | recvonly                | Section 9 of [RFC4566]   |
          +-------------------------+--------------------------+
          | sendrecv                | Section 9 of [RFC4566]   |
          +-------------------------+--------------------------+
          | sendonly                | Section 9 of [RFC4566]   |
          +-------------------------+--------------------------+
          | inactive                | Section 9 of [RFC4566]   |
          +-------------------------+--------------------------+
          | fmtp                    | Section 9 of [RFC4566]   |
          +-------------------------+--------------------------+
          | rtcp                    | Section 2.1 of [RFC3605] |
          +-------------------------+--------------------------+
          | setup                   | Section 4 of [RFC4145]   |
          +-------------------------+--------------------------+
          | fingerprint             | Section 5 of [RFC8122]   |
          +-------------------------+--------------------------+
          | rtcp-fb                 | Section 4.2 of [RFC4585] |
          +-------------------------+--------------------------+
          | extmap                  | Section 7 of [RFC5285]   |
          +-------------------------+--------------------------+
          | mid                     | Section 4 of [RFC5888]   |
          +-------------------------+--------------------------+
          | group                   | Section 5 of [RFC5888]   |
          +-------------------------+--------------------------+
          | imageattr               | Section 3.1 of [RFC6236] |
          +-------------------------+--------------------------+
          | extmap (encrypt option) | Section 4 of [RFC6904]   |
          +-------------------------+--------------------------+
          | candidate               | Section 5.1 of [RFC8839] |
          +-------------------------+--------------------------+
          | remote-candidates       | Section 5.2 of [RFC8839] |
          +-------------------------+--------------------------+
          | ice-lite                | Section 5.3 of [RFC8839] |
          +-------------------------+--------------------------+
          | ice-ufrag               | Section 5.4 of [RFC8839] |
          +-------------------------+--------------------------+
          | ice-pwd                 | Section 5.4 of [RFC8839] |
          +-------------------------+--------------------------+
          | ice-options             | Section 5.6 of [RFC8839] |
          +-------------------------+--------------------------+
          | msid                    | Section 3 of [RFC8830]   |
          +-------------------------+--------------------------+
          | rid                     | Section 10 of [RFC8851]  |
          +-------------------------+--------------------------+
          | simulcast               | Section 5.1 of [RFC8853] |
          +-------------------------+--------------------------+
          | tls-id                  | Section 4 of [RFC8842]   |
          +-------------------------+--------------------------+

                       Table 1: SDP ABNF References

Acknowledgements

   Harald Alvestrand, Taylor Brandstetter, Suhas Nandakumar, and Peter
   Thatcher provided significant text for this document.  Bernard Aboba,
   Adam Bergkvist, Jan-Ivar Bruaroey, Dan Burnett, Ben Campbell, Alissa
   Cooper, Richard Ejzak, Stefan Håkansson, Ted Hardie, Christer
   Holmberg, Andrew Hutton, Randell Jesup, Matthew Kaufman, Anant
   Narayanan, Adam Roach, Robert Sparks, Neil Stratford, Martin Thomson,
   Sean Turner, and Magnus Westerlund all provided valuable feedback on
   this document.

Authors' Addresses

   Justin Uberti
   Google
   747 6th Street South
   Kirkland, WA 98033
   United States of America

   Email: justin@uberti.name


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

   Email: fluffy@iii.ca


   Eric Rescorla (editor)
   Mozilla

   Email: ekr@rtfm.com