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<!DOCTYPE rfc SYSTEM "rfc2629.dtd">
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<?rfc iprnotified="no" ?>
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<?rfc colonspace="yes" ?>
<?rfc rfcedstyle="no" ?>
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<rfc xmlns:xi="http://www.w3.org/2001/XInclude" category="std"
     docName="draft-ietf-rtcweb-jsep-26"
ipr="trust200902"> number="8829" consensus="true"
     ipr="trust200902" obsoletes="" updates="" submissionType="IETF"
     xml:lang="en" tocInclude="true" symRefs="true" sortRefs="true"
     tocDepth="4" version="3">
  <!-- xml2rfc v2v3 conversion 2.34.0 -->
  <front>
    <title abbrev="JSEP">JavaScript Session Establishment
    Protocol</title> Protocol (JSEP)</title>
    <seriesInfo name="RFC" value="8829"/>

<!-- [rfced] Please insert any keywords (beyond those that appear in the
title) for use on https://www.rfc-editor.org/search -->

<!-- [rfced] We have updated the title to include "JSEP"; please let us know
any objections.

Original:
 JavaScript Session Establishment Protocol

Currently:
 JavaScript Session Establishment Protocol (JSEP) -->

    <author fullname="Justin Uberti" initials="J." surname="Uberti">
      <organization>Google</organization>
      <address>
        <postal>
          <street>747 6th St S</street> Street South</street>
          <city>Kirkland</city>
          <region>WA</region>
          <code>98033</code>
          <country>USA</country>
          <country>United States of America</country>
        </postal>
        <email>justin@uberti.name</email>
      </address>
    </author>
    <author fullname="Cullen Jennings" initials="C." surname="Jennings">
      <organization>Cisco</organization>
      <address>
        <postal>
          <street>400 3rd Avenue SW</street>
          <city>Calgary</city>
          <region>AB</region>
          <code>T2P 4H2</code>
          <country>Canada</country>
        </postal>
        <email>fluffy@iii.ca</email>
      </address>
    </author>
    <author fullname="Eric Rescorla" initials="E.K." initials="E." surname="Rescorla" role="editor">
      <organization>Mozilla</organization>
      <address>
        <postal>
          <street>331 E. Evelyn Ave</street> Ave.</street>
          <city>Mountain View</city>
          <region>CA</region>
          <code>94041</code>
          <country>USA</country>
          <country>United States of America</country>
        </postal>
        <email>ekr@rtfm.com</email>
      </address>
    </author>

<!-- [rfced] Eric, we note that Mozilla and RTFM are listed as your
affiliation in various documents within the cluster.  Please review and let us
know if any updates are needed.
-->

    <date />
    <area>RAI</area> month="July" year="2020"/>

    <abstract>
      <t>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, API and discusses how this relates to existing
      signaling protocols.</t>
    </abstract>
  </front>
  <middle>

<!-- [rfced] Adam Roach previously requested that "All other documents in
Cluster 238 that currently reference RFC 5245 should be updated to
reference RFC 8445. ... any other references to RFC 5245 that I may
have overlooked should also be updated."

We believe this document intentionally refers to both RFCs.  Please let us
know if any updates are needed.
-->

    <section title="Introduction" anchor="sec.introduction"> anchor="sec.introduction" numbered="true" toc="default">
      <name>Introduction</name>
      <t>This document describes how the W3C WEBRTC Web Real-Time Communication (WebRTC) RTCPeerConnection
      interface
      <xref target="W3C.webrtc"></xref> target="W3C.webrtc" format="default"/> is used to control the setup,
      management
      management, and teardown of a multimedia session.</t>
      <section title="General anchor="sec.general-design-of-jsep" numbered="true" toc="default">
        <name>General Design of JSEP"
      anchor="sec.general-design-of-jsep"> JSEP</name>
        <t>WebRTC call setup has been designed to focus on controlling
        the media plane, leaving signaling plane 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 necessary transport and media configuration
        information necessary to establish the media plane.</t>
        <t>With these considerations in mind, this document describes
        the JavaScript Session Establishment Protocol (JSEP) that (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 ICE Interactive
	Connectivity Establishment (ICE) state
        machine.
        machine <xref target="RFC8445"/>. The combination of the JSEP implementation and the
        JavaScript application is referred to throughout this document
        as a "JSEP endpoint".</t>
        <t>In this document, the use of JSEP is described as if it
        always occurs between two JSEP endpoints. Note though Note, though, that in many
        cases it will actually be between a JSEP endpoint and some kind
        of server, such as a gateway or MCU. Multipoint Control Unit (MCU). This distinction is
        invisible to the JSEP endpoint; it just follows the
        instructions it is given via the API.</t>
        <t>JSEP's handling of session descriptions is simple and
        straightforward. Whenever an offer/answer exchange is needed,
        the initiating side creates an offer by calling a createOffer()
        API. The application then uses that offer to set up its local
        config via the setLocalDescription() API. The offer is finally
        sent off to the remote side over its preferred signaling
        mechanism (e.g., WebSockets); upon receipt of that offer, the
        remote party installs it using the setRemoteDescription()
        API.</t>
        <t>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.</t>
        <t>Regarding ICE
        <xref target="RFC8445"></xref>, target="RFC8445" format="default"/>, JSEP decouples the ICE state
        machine from the overall signaling state machine, as 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,
        <xref target="I-D.ietf-mmusic-trickle-ice-sip" /> target="RFC8840" format="default"/> allows SIP to
        be used with trickle Trickle ICE
        <xref target="I-D.ietf-ice-trickle" />, target="RFC8838" format="default"/>, 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.</t>
        <t>Through its abstraction of signaling, the JSEP approach does
        require the application to be aware of the signaling process.
        While the application does not need to understand the contents
        of session descriptions to set up a call, the application must
        call the right APIs at the right times, convert the session
        descriptions and ICE information into the defined messages of
        its chosen signaling protocol, and perform the reverse
        conversion on the messages it receives from the other side.</t>
        <t>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 higher-level call-oriented interface to the
        application developer. For example, libraries exist to adapt
        the JSEP API into an API suitable for a SIP interface or an Extensible Messaging
	and Presence Protocol (XMPP) interface <xref target="RFC6120"/>.

<!-- [rfced] Section 1.1:  For ease of the reader, and per other
documents in this cluster (cluster C238), we expanded "XMPP," cited
RFC 6120, and added RFC 6120 to the list of Informative References.
Please let us know any concerns.

Original:
 For example,
 libraries exist to adapt the JSEP API into an API suitable for a SIP
 or XMPP.

Currently:
 For example,
 libraries exist to adapt the JSEP API into an API suitable for a SIP
 interface or an Extensible Messaging and Presence Protocol (XMPP)
 interface [RFC6120].
...
 [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>. -->

 Thus, JSEP
        provides greater control for the experienced developer without
        forcing any additional complexity on the novice developer.</t>
      </section>
      <section title="Other anchor="sec.other-approaches-consider" numbered="true" toc="default">
        <name>Other Approaches Considered"
      anchor="sec.other-approaches-consider"> Considered</name>
        <t>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
        <xref target="RFC3264" />, Section 4).</t> sectionFormat="comma" section="4"/>).</t>
        <t>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
        result in an API where even a simple example would require a
        significant amount of code to orchestrate all the needed
        interactions, as well as creating a large API surface that
        needed to be agreed upon and documented. In addition, these API
        points could be called in any order, resulting in a more
        complex set

<!-- [rfced] Section 1.2:  We had trouble parsing this sentence.
If the suggested text is not correct, please clarify "as well as
creating ..."

Original (the previous sentence is included for context):
 A second approach that was considered but not chosen was to decouple
 the management of interactions with the media subsystem than 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 result in an API where even a
 simple example would require a significant amount of code to
 orchestrate all the needed interactions, as well as creating a large
 API surface that needed to be agreed upon and documented.

Suggested:
 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 needed to be agreed upon and documented. -->

 In addition, these API
        points could be called in any order, resulting in a more
        complex set of interactions with the media subsystem than the
        JSEP approach, which specifies how session descriptions are to
        be evaluated and applied.</t>
        <t>One variation on JSEP that was considered was to keep the
        basic session description-oriented API, 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 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.</t>
      </section>
    </section>
    <section title="Terminology" anchor="sec.terminology"> anchor="sec.terminology" numbered="true" toc="default">
      <name>Terminology</name>
    <t>The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL
      NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", "<bcp14>MUST</bcp14>", "<bcp14>MUST NOT</bcp14>",
    "<bcp14>REQUIRED</bcp14>", "<bcp14>SHALL</bcp14>",
    "<bcp14>SHALL NOT</bcp14>", "<bcp14>SHOULD</bcp14>",
    "<bcp14>SHOULD NOT</bcp14>",
    "<bcp14>RECOMMENDED</bcp14>", "<bcp14>NOT RECOMMENDED</bcp14>",
    "<bcp14>MAY</bcp14>", and
      "OPTIONAL" "<bcp14>OPTIONAL</bcp14>" in this document are
    to be interpreted as described in BCP&nbsp;14 <xref target="RFC2119"/>
    <xref target="RFC2119"></xref>.</t> target="RFC8174"/> when, and only when, they appear in all capitals,
    as shown here.</t>
    </section>
    <section title="Semantics anchor="sec.semantics-and-syntax" numbered="true" toc="default">
      <name>Semantics and Syntax"
    anchor="sec.semantics-and-syntax"> Syntax</name>
      <section title="Signaling Model" anchor="sec.signaling-model"> anchor="sec.signaling-model" numbered="true" toc="default">
        <name>Signaling Model</name>
        <t>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
        <xref target="RFC3264"></xref> target="RFC3264" format="default"/> (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.</t>
        <figure anchor="fig-sigModel" title="JSEP anchor="fig-sigModel">
          <name>JSEP Signaling Model">
          <artwork>
            <![CDATA[ Model</name>
          <artwork name="" type="" align="left" alt=""><![CDATA[
      +-----------+                               +-----------+
      |  Web App  |<--- App-Specific Signaling -->|  Web App  |
      +-----------+                               +-----------+
            ^                                            ^
            |  SDP                                       |  SDP
            V                                            V
      +-----------+                                +-----------+
      |   JSEP    |<----------- Media ------------>|   JSEP    |
      |   Impl.   |                                |   Impl.   |
      +-----------+                                +-----------+
]]>
</artwork> ]]></artwork>
        </figure>
      </section>
      <section title="Session anchor="sec.session-descriptions-and-state-machine" numbered="true" toc="default">
        <name>Session Descriptions and State Machine"
      anchor="sec.session-descriptions-and-state-machine"> Machine</name>
        <t>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.</t>
        <t>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 declarative,
        and the remote side MUST <bcp14>MUST</bcp14> either accept them or reject them
        altogether. An example of such a parameter is the DTLS
        fingerprints
        <xref target="RFC8122"></xref>, target="RFC8122" format="default"/>, which are calculated based on
        the local certificate(s) offered, offered and are not subject to
        negotiation.</t>

        <t>In addition, various RFCs put different conditions on the
        format of offers versus answers. For example, an offer may
        propose an arbitrary number
        negotiation.

<!-- [rfced] Section 3.2:  We do not see any mention of m= sections (i.e., media
        descriptions as described DTLS in
        <xref target="RFC4566" />, Section 5.14), but an answer must
        contain the exact same number as the offer.</t>

        <t>Lastly, while the exact media parameters are only known
RFC 8122; we only
        after an offer and an see "TLS."  Will this citation be clear to
readers?

Original:
 An example of such a parameter is the DTLS
 fingerprints [RFC8122], which are calculated based on the local
 certificate(s) offered, and are not subject to negotiation.

Possibly:
 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. -->

</t>
        <t>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
        <xref target="RFC4566" sectionFormat="comma" section="5.14"/>), but an answer must
        contain the exact same number as the offer.</t>
        <t>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.</t>
        <t>Therefore, in order to handle session descriptions properly,
        the JSEP implementation needs:
        <list style="numbers">

          <t>To
        </t>
        <ol spacing="normal" type="1">
          <li>To know if a session description pertains to the local or
          remote side.</t>

          <t>To side.</li>
          <li>To know if a session description is an offer or an
          answer.</t>

          <t>To
          answer.</li>
          <li>To allow the offer to be specified independently of the
          answer.</t>
        </list>JSEP
          answer.</li>
        </ol>
        <t>JSEP addresses this by adding both setLocalDescription
        and setRemoteDescription methods and having session description
        objects contain a type field indicating the type of session
        description being supplied. This satisfies the requirements
        listed above for both the offerer, who first calls
        setLocalDescription(sdp [offer]) and then later
        setRemoteDescription(sdp [answer]), as well as for and the
        answerer, who first calls setRemoteDescription(sdp [offer]) and
        then later setLocalDescription(sdp [answer]).</t>
        <t>During the offer/answer exchange, the outstanding offer is
        considered to be "pending" at the offerer and the answerer, as
        it may either 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
        <xref target="sec.pendinglocaldescription" />, format="counter"/>,
        <xref target="sec.pendingremotedescription" />, format="counter"/>,
        <xref target="sec.currentlocaldescription" />, format="counter"/>, and
        <xref target="sec.currentremotedescription" />, format="counter"/> provide more
        detail on pending and current descriptions.</t>
        <t>JSEP also allows for an answer to be treated as provisional
        by the application. Provisional answers provide a way for an
        answerer to communicate initial session parameters back to the
        offerer, in order to allow the session to begin, while allowing
        a final answer to be specified later. This concept of a final
        answer is important to the offer/answer model; when such an
        answer is received, any extra resources allocated by the caller
        can be released, now that the exact session configuration is
        known. These "resources" can include things like extra ICE
        components, TURN 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.</t>
        <t>In
        <xref target="RFC3264"></xref>, target="RFC3264" format="default"/>, the constraint at the signaling
        level is that only one offer can be outstanding for a given
        session, but at the media stack level, a new offer can be
        generated at any point. For example, when using SIP for
        signaling, if one offer is sent, sent and is then cancelled 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,
        answers and then finally end the offer-answer offer/answer exchange by sending a
        final answer. The state machine for this is as follows:</t>

        <t>
        <figure anchor="fig-state-machine"
          title="JSEP anchor="fig-state-machine">
          <name>JSEP State Machine">
            <artwork>
              <![CDATA[ Machine</name>
          <artwork name="" type="" align="left" alt=""><![CDATA[
                    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)
]]>
</artwork> ]]></artwork>
        </figure>
        </t>
        <t>Aside from these state transitions transitions, there is no other
        difference between the handling of provisional ("pranswer") and
        final ("answer") answers.</t>
      </section>
      <section title="Session anchor="sec.session-description-forma" numbered="true" toc="default">
        <name>Session Description Format"
      anchor="sec.session-description-forma"> Format</name>
        <t>JSEP's session descriptions use SDP Session Description Protocol (SDP) syntax for their
        internal representation. While this format is not optimal for
        manipulation from JavaScript, it is widely accepted, 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.</t>
        <t>However, to provide for future flexibility, the SDP syntax
        is encapsulated within a SessionDescription object, which can
        be constructed from SDP, 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.</t>
        <t>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.</t>
      </section>
      <section title="Session anchor="sec.session-description-ctrl" numbered="true" toc="default">
        <name>Session Description Control"
      anchor="sec.session-description-ctrl"> Control</name>
        <t>In order to give the application control over various common
        session parameters, JSEP provides control surfaces which that tell
        the JSEP implementation how to generate session descriptions.
        This avoids the need for JavaScript to modify session
        descriptions in most cases.</t>
        <t>Changes to these objects result in changes to the session
        descriptions generated by subsequent createOffer/Answer createOffer/createAnswer
        calls.</t>
        <section title="RtpTransceivers" anchor="sec.rtptransceivers"> anchor="sec.rtptransceivers" numbered="true" toc="default">
          <name>RtpTransceivers</name>
          <t>RtpTransceivers allow the application to control the RTP
          media associated with one m= "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.</t>
          <t>RtpTransceivers generally have a 1:1 mapping with m= "m="
          sections, although there may be more RtpTransceivers than m= "m="
          sections when RtpTransceivers are created but not yet
          associated with a m= an "m=" section, or if RtpTransceivers have been
          stopped and disassociated from m= "m=" sections. An RtpTransceiver
          is said to be associated with an m= "m=" section if its mid
          media identification (mid) property is non-null; otherwise otherwise, it is said to be
          disassociated. The associated m= "m=" section is determined using
          a mapping between transceivers and m= "m=" section indices, formed
          when creating an offer or applying a remote offer.</t>
          <t>An RtpTransceiver is never associated with more than one
          m=
          "m=" section, and once a session description is applied, a m= an "m="
          section is always associated with exactly one RtpTransceiver.
          However, in certain cases where a m= an "m=" section has been
          rejected, as discussed in
          <xref target="sec.subsequent-offers" /> format="default"/> below, that m= "m=" section
          will be "recycled" and associated with a new RtpTransceiver
          with a new mid value.</t>
          <t>RtpTransceivers can be created explicitly by the
          application or implicitly by calling setRemoteDescription
          with an offer that adds new m= "m=" sections.</t>
        </section>
        <section title="RtpSenders" anchor="sec.rtpsenders"> anchor="sec.rtpsenders" numbered="true" toc="default">
          <name>RtpSenders</name>
          <t>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= "m=" section. This
          includes encoding the attached MediaStreamTrack, sending RTP
          media packets, and generating/processing RTCP the RTP Control Protocol (RTCP) for the
          outgoing RTP streams(s).</t>
        </section>
        <section title="RtpReceivers" anchor="sec.rtpreceivers"> anchor="sec.rtpreceivers" numbered="true" toc="default">
          <name>RtpReceivers</name>
          <t>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= "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).</t>
        </section>
      </section>
      <section title="ICE" anchor="sec.ice">
        <section title="ICE anchor="sec.ice" numbered="true" toc="default">
        <name>ICE</name>
        <section anchor="sec.ice-gather-overview" numbered="true" toc="default">
          <name>ICE Gathering Overview"
        anchor="sec.ice-gather-overview"> Overview</name>
          <t>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= "m=" section to the local session description, 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, application or implicitly by the JSEP
          implementation in response to changes in the ICE
          configuration.</t>
          <t>When the ICE configuration changes in a way that requires
          a new gathering phase, a 'needs-ice-restart' bit is set. When
          this bit is set, calls to the createOffer API will generate
          new ICE credentials. This bit is cleared by a call to the
          setLocalDescription API with new ICE credentials from either
          an offer or an answer, i.e., from either a local- locally or
          remote-initiated
          remotely initiated ICE restart.</t>
          <t>When a new gathering phase starts, the ICE agent will
          notify the application that gathering is occurring through an
          event. Then, when each new ICE candidate becomes available,
          the ICE agent will supply it to the application via an
          additional event; these candidates will also automatically be
          added to the current and/or pending local session
          description. Finally, when all candidates have been gathered,
          an event will be dispatched to signal that the gathering
          process is complete.</t>
          <t>Note that gathering phases only gather the candidates
          needed by new/recycled/restarting m= "m=" sections; other m= "m="
          sections continue to use their existing candidates. Also, if
          an m= "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= "m=" section
          if and only if its MID item is a BUNDLE-tag, as described in
          <xref target="I-D.ietf-mmusic-sdp-bundle-negotiation" />.</t> target="RFC8843" format="default"/>.</t>
        </section>
        <section title="ICE anchor="sec.ice-candidate-trickling" numbered="true" toc="default">
          <name>ICE Candidate Trickling"
        anchor="sec.ice-candidate-trickling"> Trickling</name>
          <t>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
          <xref target="I-D.ietf-ice-trickle"></xref>. target="RFC8838" format="default"/>. 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.</t>
          <t>JSEP supports optional candidate trickling by providing
          APIs, as described above, that provide control and feedback
          on the ICE candidate gathering process. Applications that
          support candidate trickling can send the initial offer
          immediately and send individual candidates when they get the
          notified of a new candidate; applications that do not support
          this feature can simply wait for the indication that
          gathering is complete, and then create and send their offer,
          with all the candidates, at this that time.</t>
          <t>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.</t>
          <section title="ICE anchor="sec.ice-candidate-format" numbered="true" toc="default">
            <name>ICE Candidate Format"
          anchor="sec.ice-candidate-format"> Format</name>
            <t>In JSEP, ICE candidates are abstracted by an
            IceCandidate object, and as with session descriptions, SDP
            syntax is used for the internal representation.</t>
            <t>The candidate details are specified in an IceCandidate
            field, using the same SDP syntax as the
            "candidate-attribute" field defined in
            <xref target="I-D.ietf-mmusic-ice-sip-sdp" />,
            Section 4.1. target="RFC8839" sectionFormat="comma" section="5.1"/>. Note that this
            field does not contain an "a=" prefix, as indicated in the
            following example:</t>
            <figure>
              <artwork>
                <![CDATA[

            <sourcecode name="" type="sdp"><![CDATA[
candidate:1 1 UDP 1694498815 192.0.2.33 10000 typ host
]]>
</artwork>
            </figure> ]]></sourcecode>
            <t>The IceCandidate object contains a field to indicate
            which ICE ufrag it is associated with, as defined in
            <xref target="I-D.ietf-mmusic-ice-sip-sdp" />,
            Section 4.4. target="RFC8839" sectionFormat="comma" 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.</t>
            <t>The IceCandidate object also contains fields to indicate
            which m= "m=" section it is associated with, which can be
            identified in one of two ways, ways: either by a m= an "m=" section
            index,
            index or by a MID. The m= "m=" section index is a zero-based
            index, with index N referring to the N+1th m= "m=" section in
            the session description referenced by this IceCandidate.
            The MID is a "media stream identification" value, as
            defined in
            <xref target="RFC5888"></xref>, Section 4, target="RFC5888" sectionFormat="comma" section="4"/>, which provides a
            more robust way to identify the m= "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
            <xref target="sec.applying-a-remote-desc" /> format="default"/> below). If the
            MID field is present in a received IceCandidate, it MUST <bcp14>MUST</bcp14> be
            used for identification; otherwise, the m= "m=" section index is
            used instead.</t>
            <t>When creating an IceCandidate object, JSEP
            implementations MUST <bcp14>MUST</bcp14> populate each of the candidate, ufrag,
            m=
            "m=" section index, and MID fields. Implementations MUST <bcp14>MUST</bcp14> also
            be prepared to receive objects with some fields missing, as
            mentioned above.</t>
          </section>
        </section>
        <section title="ICE anchor="sec.ice-candidate-policy" numbered="true" toc="default">
          <name>ICE Candidate Policy"
        anchor="sec.ice-candidate-policy"> Policy</name>
          <t>Typically, when gathering ICE candidates, the JSEP
          implementation will gather all possible forms of initial
          candidates -- host, server-reflexive, and relay.

<!-- [rfced] Sections 3.5.3 and 8:  Per author feedback for
RFC 8839 and per other documents in this cluster, we hyphenated the
term "server reflexive".  Please let us know any objections.

Original:
 Typically, when gathering ICE candidates, the JSEP implementation
 will gather all possible forms of initial candidates - host, server
 reflexive, and relay.
...
 Thus,
 for instance, it is not possible to prevent the remote peer from
 learning your public IP address by removing server reflexive
 candidates.

Currently:
 Typically, when gathering ICE candidates, the JSEP implementation
 will gather all possible forms of initial candidates - host, server-
 reflexive, and relay.
...
 Thus,
 for instance, it is not possible to prevent the remote peer from
 learning your public IP address by removing server-reflexive
 candidates. -->

 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
          <xref target="I-D.ietf-rtcweb-ip-handling" />.</t> target="RFC8828" format="default"/>.</t>
          <t>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 <bcp14>MUST</bcp14> allow the candidate policy to be
          changed in mid-session, subject to the aforementioned
          interactions with local policy.</t>
          <t>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 <bcp14>MUST NOT</bcp14> 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.</t>
        </section>
        <section title="ICE anchor="sec.ice-candidate-pool" numbered="true" toc="default">
          <name>ICE Candidate Pool"
        anchor="sec.ice-candidate-pool"> Pool</name>
          <t>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 which 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.</t>

          <t>When
<!-- [rfced] We suggest clarifying "goes to gather" - perhaps "gathers",
"starts to gather", "prepares to gather", etc.  "goes to gather" may be
confusing for some readers.

Original:
   When setLocalDescription is eventually called, and the JSEP
   implementation goes to gather the needed ICE candidates, it SHOULD
   start by checking if any candidates are available in the pool.
-->
          <t>When setLocalDescription is eventually called and the
          JSEP implementation goes to gather the needed ICE candidates,
          it <bcp14>SHOULD</bcp14> start by checking if any candidates are available
          in the pool. If there are candidates in the pool, they SHOULD <bcp14>SHOULD</bcp14>
          be handed to the application immediately via the ICE
          candidate event. If the pool becomes depleted, either because
          a larger-than-expected number of ICE components is used, 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.</t>
          <t>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 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.</t>
          using. ("STUN" stands for "Session Traversal Utilities for NAT".)</t>
        </section>
        <section title="ICE Versions"> numbered="true" toc="default">
          <name>ICE Versions</name>
          <t>While this specification formally relies on <xref
          target="RFC8445"></xref>, target="RFC8445" format="default"/>, at the time of its publication, the
          majority of WebRTC implementations support the version
          of ICE described in <xref target="RFC5245"></xref>. target="RFC5245" format="default"/>. The use of
          the "ice2" attribute defined in <xref target="RFC8445"></xref> target="RFC8445" format="default"/>
          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 <bcp14>MUST</bcp14> be able to accept remote
          descriptions that do not have the "ice2" attribute.</t>
        </section>
      </section>
      <section anchor="sec.imageattr" title="Video numbered="true" toc="default">
        <name>Video Size Negotiation"> Negotiation</name>
        <t>Video size negotiation is the process through which a
        receiver can use the "a=imageattr" SDP attribute
        <xref target="RFC6236" /> format="default"/> 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 <bcp14>MUST</bcp14> handle this gracefully, e.g.,
        by discarding the video.</t>
        <t>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, pixels but
        SHOULD
        <bcp14>SHOULD</bcp14> 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.</t>
        <section anchor="sec.creating-imageattr"
        title="Creating numbered="true" toc="default">
          <name>Creating an imageattr Attribute"> Attribute</name>
          <t>The receiver will first intersect any known local limits
          (e.g., hardware decoder capababilities, capabilities, 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 <bcp14>SHOULD</bcp14> be omitted. If these local
          limits preclude receiving any video, i.e., the degenerate
          case of no permitted resolutions, the "a=imageattr" attribute
          MUST
          <bcp14>MUST</bcp14> be omitted, and the m= "m=" section MUST <bcp14>MUST</bcp14> be marked as
          sendonly/inactive, as appropriate.</t>
          <t>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= "x=" and y= "y=" values.</t>
          <t>The rules here express a single set of preferences, and
          therefore, the "a=imageattr" q= "q=" value is not important. It
          SHOULD
          <bcp14>SHOULD</bcp14> be set to 1.0.</t> "1.0".</t>
          <t>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.
          <bcp14>RECOMMENDED</bcp14>. However, when the supported video codecs have
          different limitations, specific "a=imageattr" attributes MUST <bcp14>MUST</bcp14>
          be inserted for each payload type.</t>
          <t>As an example, consider a system with a multiformat video
          decoder, which is capable of decoding any resolution from
          48x48 to 720p, 720p. In this case, the implementation would
          generate this attribute:</t>
          <t>a=imageattr:* recv [x=[48:1280],y=[48:720],q=1.0]</t>
          <t>This declaration indicates that the receiver is capable of
          decoding any image resolution from 48x48 up to 1280x720
          pixels.</t>
        </section>
        <section anchor="sec.interpreting-imageattr"
        title="Interpreting numbered="true" toc="default">
          <name>Interpreting imageattr Attributes"> Attributes</name>
          <t>
          <xref target="RFC6236" /> format="default"/> defines "a=imageattr" to be an
          advisory field. This means that it does not absolutely
          constrain the video formats that the sender can use, use but
          gives an indication of the preferred values.</t>
          <t>This specification prescribes behavior that is more specific behavior. specific. When
          a MediaStreamTrack, which is producing video of a certain
          resolution (the "track resolution"), is attached to a 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 <bcp14>MUST</bcp14> 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 <bcp14>MUST</bcp14> be followed as long as the
          attributes remain present in the remote description,
          including cases in which the track changes its resolution, resolution or
          is replaced with a different track.</t>
          <t>Depending on how the RtpSender is configured, it may be
          producing a single encoding at a certain resolution, resolution or, if
          simulcast
          <xref
          (<xref target="sec.simulcast" /> format="default"/>) 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, needed or may
          be locked to a specific output resolution.</t>
          <t>For each encoding being produced by the RtpSender, the set
          of "a=imageattr recv" attributes in the corresponding m= "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= "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=
          "m=" sections that contain multiple such attributes.</t>
          <t>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 <bcp14>MUST NOT</bcp14> be transmitted, and an error SHOULD <bcp14>SHOULD</bcp14> be
          raised to the application.
          <list style="symbols">

            <t>The
          </t>
          <ul spacing="normal">
            <li>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 <bcp14>MUST</bcp14> be ignored. When considering a
            MediaStreamTrack that is producing rotated video, the
            unrotated resolution MUST <bcp14>MUST</bcp14> be used for the checks. This is
            required regardless of whether the receiver supports
            performing receive-side rotation (e.g., through CVO Coordination of
	    Video Orientation (CVO)
            <xref target="TS26.114" />), format="default"/>), as it significantly simplifies
            the matching logic.</t>

            <t>If logic.</li>
            <li>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 <bcp14>MUST NOT</bcp14> be used.</t>

            <t>If used.</li>
            <li>If the encoder resolution exceeds the maximum size
            permitted by the attribute, attribute and the encoder is allowed to
            adjust its resolution, the encoder SHOULD <bcp14>SHOULD</bcp14> apply downscaling
            in order to satisfy the limits. Downscaling MUST NOT <bcp14>MUST NOT</bcp14> change
            the picture aspect ratio of the encoding, ignoring any
            trivial differences due to rounding. For example, if the
            encoder resolution is 1280x720, 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
            <bcp14>MUST NOT</bcp14> be used.</t>

            <t>If used.</li>
            <li>If the encoder resolution is less than the minimum size
            permitted by the attribute, the attribute MUST NOT <bcp14>MUST NOT</bcp14> be used;
            the encoder MUST NOT <bcp14>MUST NOT</bcp14> apply upscaling. JSEP implementations
            SHOULD
            <bcp14>SHOULD</bcp14> avoid this situation by allowing receipt of
            arbitrarily small resolutions, perhaps via fallback to a
            software decoder.</t>

            <t>If decoder.</li>
            <li>If the encoder resolution is within the maximum and
            minimum sizes, no action is needed.</t>
          </list></t> needed.</li>
          </ul>
        </section>
      </section>
      <section title="Simulcast" anchor="sec.simulcast"> anchor="sec.simulcast" numbered="true" toc="default">
        <name>Simulcast</name>
        <t>JSEP supports simulcast transmission of a MediaStreamTrack,
        where multiple encodings of the source media can be transmitted
        within the context of a single m= "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.</t>
        <t>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= "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.</t>
        <t>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.</t>
        <t>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= "m="
        section.</t>
        <t>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 in 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.</t>
        <t>When using JSEP to transmit multiple encodings from a an
        RtpSender, the techniques from
        <xref target="I-D.ietf-mmusic-sdp-simulcast" /> target="RFC8853" format="default"/> and
        <xref target="I-D.ietf-mmusic-rid" /> target="RFC8851" format="default"/> 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
        <xref target="RFC8853" sectionFormat="comma" section="6.2"/>,
        with a "send" simulcast stream description that lists each
        desired encoding, and no "recv" simulcast stream description.

<!-- [rfced] Sections 3.7 and 5.2.1:  We do not see "a=simulcast" or
"send" mentioned anywhere in Section 6.2 of
RFC 8853 [I-D.ietf-mmusic-sdp-simulcast].  Please confirm that these citations
are correct and will be clear to readers.

Original:
 Specifically, when multiple
 encodings have been configured for a RtpSender, the m= section for
 the RtpSender will include an "a=simulcast" attribute, as defined in
        <xref target="I-D.ietf-mmusic-sdp-simulcast" />,
 [I-D.ietf-mmusic-sdp-simulcast], Section 6.2, with a "send" simulcast
 stream description that lists each desired encoding, and no "recv"
 simulcast stream description.
...
 o  If the RtpTransceiver has a sendrecv or sendonly direction and
    more than one "a=rid" line has been generated, an "a=simulcast"
    line, with direction "send", as defined in
    [I-D.ietf-mmusic-sdp-simulcast], Section 6.2. -->

        The m= "m=" section will also include an "a=rid" attribute for each
        encoding, as specified in
        <xref target="I-D.ietf-mmusic-rid" />, Section 4; target="RFC8851" sectionFormat="comma" section="4"/>; the use of
        RID identifiers
        Restriction Identifiers (RIDs) allows the individual encodings to be
        disambiguated even though they are all part of the same m= "m="
        section.</t>
      </section>
      <section title="Interactions With Forking"
      anchor="sec.interactions-with-forking"> anchor="sec.interactions-with-forking" numbered="true" toc="default">
        <name>Interactions with Forking</name>
        <t>Some call signaling systems allow various types of forking
        where an SDP Offer may be provided to more than one device. For
        example, SIP
        <xref target="RFC3261"></xref> target="RFC3261" format="default"/> defines both a "Parallel Search" "parallel search"
        and "Sequential Search". "sequential search". Although these are primarily signaling
        level 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 of 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:
        <list style="symbols">

          <t>Start are as follows:
        </t>
        <ul spacing="normal">
          <li>Start exchanging media with a given remote peer, but keep
          all the resources reserved in the offer.</t>

          <t>Start offer.</li>
          <li>Start exchanging media with a given remote peer, and free
          any resources in the offer that are not being used.</t>
        </list></t> used.</li>
        </ul>
        <section title="Sequential Forking"
        anchor="sec.sequential-forking"> anchor="sec.sequential-forking" numbered="true" toc="default">
          <name>Sequential Forking</name>
          <t>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.</t>
          <t>JSEP handles sequential forking well, allowing the
          application to easily control the policy for selecting the
          desired remote endpoint. When an answer arrives from one of
          the callees, the application can choose to apply it either as
          a either
          (1)&nbsp;a provisional answer, leaving open the possibility of using a
          different answer in the future, future or apply it as a (2)&nbsp;a final
          answer, ending the setup flow.</t>
          <t>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.</t>
          <t>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.</t>
        </section>
        <section title="Parallel Forking"
        anchor="sec.parallel-forking"> anchor="sec.parallel-forking" numbered="true" toc="default">
          <name>Parallel Forking</name>
          <t>Parallel forking involves a call being dispatched to
          multiple remote callees, where each callee can accept the
          call,
          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
          <xref target="RFC3960"></xref>, Section 3.1. target="RFC3960" sectionFormat="comma" section="3.1"/>. Most SIP devices
          today only support exchanging media with a single device at a
          time,
          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, 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.</t>
          <t>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
          <xref target="RFC3960"></xref> describes target="RFC3960" format="default"/> 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
          <xref target="sec.peerconnection" />) format="default"/>) and making sure that
          the same local media streams have been added into this new
          PeerConnection. Then the new PeerConnection object would
          produce a an SDP offer that could be used by the signaling to
          perform the UPDATE strategy discussed in
          <xref target="RFC3960"></xref>.</t> target="RFC3960" format="default"/>.</t>
          <t>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
          <xref target="sec.transceiver-set-direction" />), format="default"/>), 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.</t>
        </section>
      </section>
    </section>
    <section title="Interface" anchor="sec.interface"> anchor="sec.interface" numbered="true" toc="default">
      <name>Interface</name>
      <t>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.

<!-- [rfced]  Sections 4 and 7:  For ease of the reader, please let
us know if we may cite [W3C.webrtc] as a reminder regarding
"addTrack," "removeTrack," etc. (Section 4) as well as
"onicecandidate," "addIceCandidate," and "ontrack" (Section 7),
as follows.

Original:
 The actual API exposed in the W3C API
 may have somewhat different syntax, but should map easily to these concepts.</t>
      <section title="PeerConnection" anchor="sec.peerconnection">
 concepts.
...
 More examples of SDP for WebRTC call flows, including examples with
 IPv6 addresses, can be found in [I-D.ietf-rtcweb-sdp].

Suggested:
 The actual API exposed in the W3C API [W3C.webrtc]
 may have somewhat different syntax but should map easily to these
 concepts.
...
 More examples of SDP for WebRTC call flows, including examples with
 IPv6 addresses, can be found in [SDP4WebRTC].  See [W3C.webrtc] for
 information regarding "onicecandidate," "addIceCandidate," and
 "ontrack". -->

</t>
      <section title="Constructor" anchor="sec.pc-constructor"> anchor="sec.peerconnection" numbered="true" toc="default">
        <name>PeerConnection</name>
        <section anchor="sec.pc-constructor" numbered="true" toc="default">
          <name>Constructor</name>
          <t>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.</t>
          <t>If an ICE candidate policy is specified, it functions as
          described in
          <xref target="sec.ice-candidate-policy" />, format="default"/>, causing the JSEP
          implementation to only surface the permitted candidates
          (including any implementation-internal filtering) to the
          application,
          application and only use those candidates for connectivity
          checks. The set of available policies is as follows:
          <list style="hanging">
            <t hangText="all:">All
          </t>
          <dl newline="false" spacing="normal">
            <dt>all:</dt>
            <dd>All candidates permitted by
            implementation policy will be gathered and used.</t>

            <t></t>
            <t hangText="relay:">All used.</dd>
            <dt>relay:</dt>
            <dd>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.</t>
          </list></t>
            level.</dd>
          </dl>
          <t>The default ICE candidate policy MUST <bcp14>MUST</bcp14> be set to "all" "all", as
          this is generally the desired policy, policy and also typically
          reduces the use of application TURN server resources
          significantly.</t>
          <t>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 pre&nbhy;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 <bcp14>MUST</bcp14> be zero.</t>
          <t>The application can specify its preferred policy regarding
          use of bundle, the multiplexing mechanism defined in
          <xref target="I-D.ietf-mmusic-sdp-bundle-negotiation"> target="RFC8843" format="default">
            </xref>. Regardless of policy, the application will always
          try to negotiate bundle onto a single transport, transport and will
          offer a single bundle group across all m= "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.</t>
          <t>The set of available policies is as follows:
          <list style="hanging">
            <t hangText="balanced:">The
          </t>
          <dl newline="false" spacing="normal">
            <dt>balanced:</dt>
            <dd>The first m= "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= section "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 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= "m=" section of
            each type.</t>

            <t></t>
            <t hangText="max-compat:">All m= type.</dd>
            <dt>max-compat:</dt>
            <dd>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, endpoints but will require separate candidates
            to be gathered for each media stream.</t>

            <t></t>
            <t hangText="max-bundle:">Only stream.</dd>
            <dt>max-bundle:</dt>
            <dd>Only the first m= "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= "m="
            sections other than the first m= "m=" section, unless they are
            in the same bundle group as that m= section.</t>
          </list></t> "m=" section.</dd>
          </dl>
          <t>As it provides the best tradeoff trade-off between performance and
          compatibility with legacy endpoints, the default bundle
          policy MUST <bcp14>MUST</bcp14> be set to "balanced".</t>
          <t>The application can specify its preferred policy regarding
          use of RTP/RTCP multiplexing
          <xref target="RFC5761"></xref> target="RFC5761" format="default"/> using one of the following
          policies:
          <list style="hanging">
            <t hangText="negotiate:">The
          </t>
          <dl newline="false" spacing="normal">
            <dt>negotiate:</dt>
            <dd>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.</t>
            <t hangText="require:">The endpoints.</dd>
            <dt>require:</dt>
            <dd>The JSEP implementation will only
            gather RTP candidates and will insert an "a=rtcp-mux-only"
            indication into any new m= "m=" sections in offers it generates.
            This halves the number of candidates that the offerer needs
            to gather. Applying a description with an m= "m=" section that
            does not contain an "a=rtcp-mux" attribute will cause an
            error to be returned.</t>
          </list></t> returned.</dd>
          </dl>
          <t>The default multiplexing policy MUST <bcp14>MUST</bcp14> be set to "require".
          Implementations MAY <bcp14>MAY</bcp14> choose to reject attempts by the
          application to set the multiplexing policy to
          "negotiate".</t>
        </section>
        <section title="addTrack" anchor="sec.addTrack"> anchor="sec.addTrack" numbered="true" toc="default">
          <name>addTrack</name>
          <t>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
          <xref target="RFC5888"></xref>, Section 7. addTrack target="RFC5888" sectionFormat="comma" section="7"/>. &nbsp;addTrack attempts
          to minimize the number of transceivers as follows: If if the
          PeerConnection is in the "have-remote-offer" "have&nbhy;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
          <xref target="sec.addTransceiver" />.</t> format="default"/>.</t>
        </section>
        <section title="removeTrack" anchor="sec.removeTrack"> anchor="sec.removeTrack" numbered="true" toc="default">
          <name>removeTrack</name>
          <t>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= "m=" section associated with the
          sender as recvonly (if transceiver.direction is sendrecv) or
          as inactive (if transceiver.direction is sendonly).</t>
        </section>
        <section title="addTransceiver" anchor="sec.addTransceiver"> anchor="sec.addTransceiver" numbered="true" toc="default">
          <name>addTransceiver</name>
          <t>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 <bcp14>MUST</bcp14> 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.</t>
          <t>At the time of creation, the application can also specify
          a transceiver direction attribute, a set of MediaStreams
          which
          that the transceiver is associated with (allowing LS "LS" group
          assignments), and a set of encodings for the media (used for
          simulcast as described in
          <xref target="sec.simulcast" />).</t> format="default"/>).</t>
        </section>
        <section title="createDataChannel"
        anchor="sec.createDataChannel"> anchor="sec.createDataChannel" numbered="true" toc="default">
          <name>createDataChannel</name>
          <t>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= "m=" section, so subsequent creation of data
          channels does not have any impact on the JSEP state.</t>
          <t>The createDataChannel method also includes a number of
          arguments which that are used by the PeerConnection (e.g.,
          maxPacketLifetime) but are not reflected in the SDP and do
          not affect the JSEP state.</t>
        </section>
        <section title="createOffer" anchor="sec.createoffer"> anchor="sec.createoffer" numbered="true" toc="default">
          <name>createOffer</name>
          <t>The createOffer method generates a blob of SDP that
          contains a
          <xref target="RFC3264"></xref> an offer per <xref target="RFC3264" format="default"/> with the supported
          configurations for the session, including descriptions of the
          media added to this PeerConnection, the codec/RTP/RTCP
          options supported by this implementation, and any candidates
          that have been gathered by the ICE agent. An options
          parameter may be supplied to provide additional control over
          the generated offer. This options parameter allows an
          application to trigger an ICE restart, for the purpose of
          reestablishing connectivity.</t>
          <t>In the initial offer, the generated SDP will contain all
          desired functionality for the session (functionality that is
          supported but not desired by default may be omitted); for
          each SDP line, the generation of the SDP will follow the
          process defined for generating an initial offer from the
          document that specifies the given SDP line. The exact
          handling of initial offer generation is detailed in
          <xref target="sec.initial-offers" /> format="default"/> below.</t>
          <t>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 offer/answer exchange. The exact handling of subsequent
          offer generation is detailed in
          <xref target="sec.subsequent-offers" />. format="default"/> below.</t>
          <t>Session descriptions generated by createOffer must be
          immediately usable by setLocalDescription; if a system has
          limited resources (e.g. (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.</t>
          <t>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.</t>
        </section>
        <section title="createAnswer" anchor="sec.createanswer"> anchor="sec.createanswer" numbered="true" toc="default">
          <name>createAnswer</name>
          <t>The createAnswer method generates a blob of SDP that
          contains a
          <xref target="RFC3264"></xref> an SDP answer per <xref target="RFC3264" format="default"/> with the supported
          configuration for the session that is compatible with the
          parameters supplied in the most recent call to
          setRemoteDescription, which MUST <bcp14>MUST</bcp14> 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.</t>
          <t>As an answer, the generated SDP will contain a specific
          configuration that specifies how the media plane should be
          established; for each SDP line, the generation of the SDP
          must follow the process defined for generating an answer from
          the document that specifies the given SDP line. The exact
          handling of answer generation is detailed in
          <xref target="sec.generating-an-answer" />. format="default"/> below.</t>
          <t>Session descriptions generated by createAnswer must be
          immediately usable by setLocalDescription; like createOffer,
          the returned description should reflect the current state of
          the system.</t>
          <t>Calling this method may do things such as generating new
          ICE credentials, but it does not change the PeerConnection
          state, trigger candidate gathering, or or cause a media state
          change. Specifically, the answer is not applied, and does not
          become the current local description, until
          setLocalDescription is called.</t>
        </section>
        <section title="SessionDescriptionType"
        anchor="sec.sessiondescriptiontype"> anchor="sec.sessiondescriptiontype" numbered="true" toc="default">
          <name>SessionDescriptionType</name>
          <t>Session description objects (RTCSessionDescription) may be
          of type "offer", "pranswer", "answer" "answer", or "rollback". These
          types provide information as to how the description parameter
          should be parsed, parsed and how the media state should be
          changed.</t>
          <t>"offer" indicates that a description should be parsed as
          an offer; said description may include many possible media
          configurations. A description used as an "offer" may be
          applied anytime any time the PeerConnection is in a stable state, state or
          applied as an update to a previously supplied but unanswered
          "offer".</t>
          <t>"pranswer" indicates that a description should be parsed
          as an answer, but not a final answer, and so should not
          result in the freeing of allocated resources. It may result
          in the start of media transmission, if the answer does not
          specify an inactive media direction. A description used as a
          "pranswer" may be applied as a response to an "offer", "offer" or as an
          update to a previously sent "pranswer".</t>
          <t>"answer" indicates that a description should be parsed as
          an answer, the offer-answer offer/answer exchange should be considered
          complete, and any resources (decoders, candidates) that are
          no longer needed can be released. A description used as an
          "answer" may be applied as a response to an "offer", "offer" or as an
          update to a previously sent "pranswer".</t>
          <t>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, 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, answers and only apply an answer as
          final when it receives one that meets its criteria (e.g. (e.g., a
          live user instead of voicemail).</t>
          <t>"rollback" is a special session description type implying
          that the state machine should be rolled back to the previous
          stable state, as described in
          <xref target="sec.rollback" />. format="default"/>. The contents MUST <bcp14>MUST</bcp14> be
          empty.</t>
          <section title="Use anchor="sec.use-of-provisional-answer" numbered="true" toc="default">
            <name>Use of Provisional Answers"
          anchor="sec.use-of-provisional-answer"> Answers</name>
            <t>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, 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.

<!-- [rfced] Section 4.1.8.1:  We had trouble with this sentence.
If neither suggestion below is correct, please clarify.

Original:
 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.</t> time.

Suggestion #1:
 While
 this could also be done with a "sendonly" pranswer, if 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.

Suggestion #2:
 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. -->

</t>
            <t>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

<!-- [rfced] Section 4.1.8.1:  As it appears that "ICE and moving the audio DTLS
handshaking have" means "ICE handshaking and video DTLS handshaking have,"
we updated this sentence accordingly.  Please let us know if this is
incorrect (i.e., if the text refers to be two-way
            media. A detailed one handshaking process, in
which case "have" should be "has").

Original:
 Upon acceptance by the user, the
 application will plug in the tracks it has acquired, which, because
 ICE and DTLS handshaking have likely completed by this point, can
 start transmitting immediately.

Currently:
 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
            <xref target="sec.warmup-example"></xref>.</t> target="sec.warmup-example" format="default"/>.</t>
            <t>Of course, some applications may not be able to perform
            this double offer-answer 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.</t>
          </section>
          <section title="Rollback" anchor="sec.rollback"> anchor="sec.rollback" numbered="true" toc="default">
            <name>Rollback</name>
            <t>In certain situations situations, it may be desirable to "undo" a
            change made to setLocalDescription or setRemoteDescription.
            Consider a case where a call is ongoing, 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, parameters and sends a reject
            message back to the offerer. Now, the offerer, and possibly
            the answerer as well, need 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.</t>
          </section>
        </section>
        <section title="setLocalDescription"
        anchor="sec.setlocaldescription"> anchor="sec.setlocaldescription" numbered="true" toc="default">
          <name>setLocalDescription</name>
          <t>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.</t>
          <t>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 a (1)&nbsp;a final answer, at which point the PeerConnection can
          fully adopt the pending local description, description or a (2)&nbsp;a rollback,
          which results in a revert to the current local
          description.</t>
          <t>This API indirectly controls the candidate gathering
          process. When a local description is supplied, 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.</t>
          <t>If setRemoteDescription was previously called with an
          offer, and setLocalDescription is called with an answer
          (provisional or final), and the media directions are
          compatible, and media is available to send, this will result
          in the starting of media transmission.</t>
        </section>
        <section title="setRemoteDescription"
        anchor="sec.setremotedescription">

          <t>The setRemoteDescription method instructs transmission.

<!-- [rfced] Sections 4.1.9 and 4.1.10:  We had trouble following the
          PeerConnection to apply
purpose of all of the supplied session description as "and"s in these sentences.  Are four conditions
set in these sentences, or fewer?

Original:
 If setRemoteDescription was previously called with an offer, and
 setLocalDescription is called with an answer (provisional or final),
 and the desired remote configuration. As media directions are compatible, and media is available to
 send, this will result in setLocalDescription, the type field starting of the description indicates how it should be
          processed.</t>

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

          <t>If transmission.
...
 If setLocalDescription was previously called with an offer, and
 setRemoteDescription is called with an answer (provisional or final),
 and the media directions are compatible, and media is available to
 send, this will result in the starting of media transmission.</t>
        </section>
        <section title="currentLocalDescription"
        anchor="sec.currentlocaldescription">

          <t>The currentLocalDescription method returns the current
          negotiated local description - i.e., the local description
          from transmission.

Possibly:
 If (1) setRemoteDescription was previously called with an offer,
 (2) setLocalDescription is called with an answer (provisional or
 final), (3) the last successful offer/answer exchange - in media directions are compatible, and (4) media is
 available to send, media transmission can start.
...
 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, media transmission can start. -->

</t>
        </section>
        <section anchor="sec.setremotedescription" numbered="true" toc="default">
          <name>setRemoteDescription</name>
          <t>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.</t>
          <t>This API changes the local media state; among other
          things, it sets up local resources for sending and encoding
          media.

<!-- [rfced] Section 4.1.10:  Please confirm that "local media state"
and "local resources" (as opposed to remote) are correct in the
context of setRemoteDescription.  (We ask because we see identical
wording in Section 4.1.9 ("setLocalDescription").)

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

</t>
          <t>If setLocalDescription was previously called with an
          offer, and setRemoteDescription is called with an answer
          (provisional or final), and the media directions are
          compatible, and media is available to send, this will result
          in the starting of media transmission.</t>
        </section>
        <section anchor="sec.currentlocaldescription" numbered="true" toc="default">
          <name>currentLocalDescription</name>
          <t>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.</t>
          <t>A null object will be returned if an offer/answer exchange
          has not yet been completed.</t>
        </section>
        <section title="pendingLocalDescription"
        anchor="sec.pendinglocaldescription"> anchor="sec.pendinglocaldescription" numbered="true" toc="default">
          <name>pendingLocalDescription</name>
          <t>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.</t>
          <t>A null object will be returned if the state of the
          PeerConnection is "stable" or "have-remote-offer".</t>
        </section>
        <section title="currentRemoteDescription"
        anchor="sec.currentremotedescription"> anchor="sec.currentremotedescription" numbered="true" toc="default">
          <name>currentRemoteDescription</name>
          <t>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.</t>
          <t>A null object will be returned if an offer/answer exchange
          has not yet been completed.</t>
        </section>
        <section title="pendingRemoteDescription"
        anchor="sec.pendingremotedescription"> anchor="sec.pendingremotedescription" numbered="true" toc="default">
          <name>pendingRemoteDescription</name>
          <t>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.</t>
          <t>A null object will be returned if the state of the
          PeerConnection is "stable" or "have-local-offer".</t>
        </section>
        <section title="canTrickleIceCandidates"
        anchor="sec.cantrickle"> anchor="sec.cantrickle" numbered="true" toc="default">
          <name>canTrickleIceCandidates</name>
          <t>The canTrickleIceCandidates property indicates whether the
          remote side supports receiving trickled candidates. There are
          three potential values:
          <list style="hanging">
            <t hangText="null:">No
          </t>
          <dl newline="false" spacing="normal">
            <dt>null:</dt>
            <dd>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.</t>
            <t hangText="true:">SDP
            called.</dd>
            <dt>true:</dt>
            <dd>SDP has been received from the other
            side indicating that it can support trickle.</t>
            <t hangText="false:">SDP trickle.</dd>
            <dt>false:</dt>
            <dd>SDP has been received from the other
            side indicating that it cannot support trickle.</t>
          </list></t> trickle.</dd>
          </dl>
          <t>As described in
          <xref target="sec.ice-candidate-trickling" />, format="default"/>, JSEP
          implementations always provide candidates to the application
          individually, consistent with what is needed for Trickle ICE.
          However, applications can use the canTrickleIceCandidates
          property to determine whether their peer can actually do
          Trickle ICE, i.e., whether it is safe to send an initial
          offer or answer followed later by candidates as they are
          gathered. As "true" is the only value that definitively
          indicates remote Trickle ICE support, an application which 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.</t>
        </section>
        <section title="setConfiguration"
        anchor="sec.setconfiguration"> anchor="sec.setconfiguration" numbered="true" toc="default">
          <name>setConfiguration</name>
          <t>The setConfiguration method allows the global
          configuration of the PeerConnection, which was initially set
          by constructor parameters, to be changed during the session.
          The effects of this method call depend on when it is invoked,
          and they will differ, depending on which specific parameters are
          changed:

<!-- [rfced] Section 4.1.16:  Should "The effects of this method call"
be "The effects of calling this method," and should "This call" be
"Calling this method"?

Original:
 The effects of this method call
 depend on when it is invoked, and differ depending on which specific
 parameters are
          changed:</t>

          <t>
            <list style="symbols">

              <t>Any changes changed:
...
 This call may result in a change to the state of the ICE Agent. -->

</t>
          <ul spacing="normal">
            <li>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
              <xref target="sec.ice-gather-overview" /> format="default"/> 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, 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.</t>

              <t>Any
              servers.</li>
            <li>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.</t>

              <t>The candidates.</li>
            <li>The ICE candidate pool size MUST NOT <bcp14>MUST NOT</bcp14> 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.</t>

              <t>The discarded.</li>
            <li>The bundle and RTCP-multiplexing policies MUST NOT <bcp14>MUST NOT</bcp14> be
              changed after the construction of the PeerConnection.</t>
            </list>
          </t> PeerConnection.</li>
          </ul>
          <t>This call may result in a change to the state of the ICE
          Agent.</t>
          agent.</t>
        </section>
        <section title="addIceCandidate" anchor="sec.addicecandidate"> anchor="sec.addicecandidate" numbered="true" toc="default">
          <name>addIceCandidate</name>
          <t>The addIceCandidate method provides an update to the ICE
          agent via an IceCandidate object
          <xref
          (<xref target="sec.ice-candidate-format" />. format="default"/>). If the
          IceCandidate's candidate field is filled in, 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
          <xref target="I-D.ietf-ice-trickle" />.</t> target="RFC8838" format="default"/>.</t>
          <t>In either case, the m= "m=" section index, MID, and ufrag
          fields from the supplied IceCandidate are used to determine
          which m= "m=" section and ICE candidate generation the
          IceCandidate belongs to, as described in
          <xref target="sec.ice-candidate-format" /> format="default"/> above. In the case
          of an end-of-candidates indication, the absence of both the
          m=
          "m=" section index and MID fields is interpreted to mean that
          the indication applies to all m= "m=" sections in the specified
          ICE candidate generation. However, if both fields are absent
          for a new remote candidate, this MUST <bcp14>MUST</bcp14> be treated as an
          invalid condition, as specified below.</t>
          <t>If any IceCandidate fields contain invalid values, values or an
          error occurs during the processing of the IceCandidate
          object, the supplied IceCandidate MUST <bcp14>MUST</bcp14> be ignored and an
          error MUST <bcp14>MUST</bcp14> be returned.</t>
          <t>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.</t>
        </section>
      </section>
      <section title="RtpTransceiver" anchor="sec.transceiver">
        <section title="stop" anchor="sec.transceiver-stop"> anchor="sec.transceiver" numbered="true" toc="default">
        <name>RtpTransceiver</name>
        <section anchor="sec.transceiver-stop" numbered="true" toc="default">
          <name>stop</name>
          <t>The stop method stops an RtpTransceiver. This will cause
          future calls to createOffer to generate a zero port for the
          associated m= "m=" section. See below for more details.</t>
        </section>
        <section title="stopped" anchor="sec.transceiver-stopped"> anchor="sec.transceiver-stopped" numbered="true" toc="default">
          <name>stopped</name>
          <t>The stopped property indicates whether the transceiver has
          been stopped, either by a call to stopTransceiver or by
          applying an answer that rejects the associated "m=" section.

<!-- [rfced] Section 4.2.2:  Will "stopTransceiver" be clear to
readers?  We could not find this string elsewhere in this document,
anywhere else in this cluster of documents, in any published RFC, or
in google searches.

Original:
 The stopped property indicates whether the transceiver has been
 stopped, either by a call to stopTransceiver or by applying an answer
 that rejects the associated m= section. -->

 In
          either of these cases, it is set to "true", "true" and otherwise
          will be set to "false".</t>
          <t>A stopped RtpTransceiver does not send any outgoing RTP or
          RTCP or process any incoming RTP or RTCP. It cannot be
          restarted.</t>
        </section>
        <section title="setDirection"
        anchor="sec.transceiver-set-direction"> anchor="sec.transceiver-set-direction" numbered="true" toc="default">
          <name>setDirection</name>
          <t>The setDirection method sets the direction of a
          transceiver, which affects the direction property of the
          associated m= "m=" section on future calls to createOffer and
          createAnswer. The permitted values for direction are
          "recvonly", "sendrecv", "sendonly", and "inactive", mirroring
          the identically-named identically named directional attributes defined in
          <xref target="RFC4566"></xref>, Section 6.</t> target="RFC4566" sectionFormat="comma" section="6"/>.</t>
          <t>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
          <xref target="sec.generating-an-answer" /> format="default"/> below.</t>
          <t>Note that while setDirection sets the direction property
          of the transceiver immediately (
          <xref (<xref target="sec.transceiver-direction" />), format="default"/>), 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.</t>
        </section>
        <section title="direction" anchor="sec.transceiver-direction"> anchor="sec.transceiver-direction" numbered="true" toc="default">
          <name>direction</name>
          <t>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.</t>
        </section>
        <section title="currentDirection"
        anchor="sec.transceiver-current-direction"> anchor="sec.transceiver-current-direction" numbered="true" toc="default">
          <name>currentDirection</name>
          <t>The currentDirection property indicates the last
          negotiated direction for the transceiver's associated m= "m="
          section. More specifically, it indicates the
          directional attribute <xref target="RFC3264"></xref> target="RFC3264" format="default"/> 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.

<!-- [rfced] Section 4.2.5:  We see "direction attribute" but not
"directional attribute" in RFC 3264.  Will this text be clear to
readers?  (We ask because we also see "A direction attribute,
determined by applying the rules regarding the offered direction
specified in [RFC3264], Section 6.1" in Section 5.3.1 of this
document.)

Original:
 More specifically, it
 indicates the [RFC3264] directional attribute 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
          directional attribute for the associated m= "m=" section in a
          remote answer is "recvonly", currentDirection is set to
          "sendonly".</t>
          <t>If an answer that references this transceiver has not yet
          been applied, applied or if the transceiver is stopped,
          currentDirection is set to null.</t> "null".</t>
        </section>
        <section title="setCodecPreferences"
        anchor="sec.transceiver-set-codec-preferences"> anchor="sec.transceiver-set-codec-preferences" numbered="true" toc="default">
          <name>setCodecPreferences</name>
          <t>The setCodecPreferences method sets the codec preferences
          of a transceiver, which in turn affect the presence and order
          of codecs of the associated m= "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.</t>
          <t>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= "m=" section will be excluded. The codec
          preferences cannot add media formats that would otherwise not
          be present.</t>
          <t>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= "m=" section will follow the
          specified preferences.</t>
        </section>
      </section>
    </section>
    <section title="SDP anchor="sec.sdp-interaction-procedure" numbered="true" toc="default">
      <name>SDP Interaction Procedures"
    anchor="sec.sdp-interaction-procedure"> Procedures</name>
      <t>This section describes the specific procedures to be followed
      when creating and parsing SDP objects.</t>
      <section title="Requirements Overview"
      anchor="sec.requirements-overview"> anchor="sec.requirements-overview" numbered="true" toc="default">
        <name>Requirements Overview</name>
        <t>JSEP implementations must comply with the specifications
        listed below that govern the creation and processing of offers
        and answers.</t>
        <section title="Usage Requirements"
        anchor="sec.usage-requirements"> anchor="sec.usage-requirements" numbered="true" toc="default">
          <name>Usage Requirements</name>
          <t>All session descriptions handled by JSEP implementations,
          both local and remote, MUST <bcp14>MUST</bcp14> indicate support for the
          following specifications. If any of these are absent, this
          omission MUST <bcp14>MUST</bcp14> be treated as an error.
          <list style="symbols">

            <t>ICE,
          </t>
          <ul spacing="normal">
            <li>ICE, as specified in
            <xref target="RFC8445"></xref>, MUST target="RFC8445" format="default"/>, <bcp14>MUST</bcp14> be used. Note that the
            remote endpoint may use a Lite lite implementation;
            implementations MUST <bcp14>MUST</bcp14> properly handle remote endpoints which that
            do ICE-Lite.</t>

            <t>DTLS ICE-lite.</li>
            <li>DTLS
            <xref target="RFC6347" /> format="default"/> or DTLS-SRTP
            <xref target="RFC5763"></xref>, MUST target="RFC5763" format="default"/> <bcp14>MUST</bcp14> be used, as
            appropriate for the media type, as specified in
            <xref target="I-D.ietf-rtcweb-security-arch" /></t>
          </list></t> target="RFC8827" format="default"/>.</li>
          </ul>
          <t>The SDES SRTP keying mechanism from
          <xref target="RFC4568" /> format="default"/> <bcp14>MUST NOT</bcp14> be used, as discussed in
          <xref target="RFC8827" format="default"/>.

<!-- [rfced] Section 5.1.1:  Does SDES refer to "source description" or
"security description"?  Neither [RFC4568] nor RFC 8827
[I-D.ietf-rtcweb-security-arch] mention "SDES" or "source description" (as
used in RFC 8852 and other documents in this cluster).

Original:
 The SDES SRTP keying mechanism from [RFC4568] MUST NOT be used, as
 discussed in
          <xref target="I-D.ietf-rtcweb-security-arch" />.</t> [I-D.ietf-rtcweb-security-arch]. -->

</t>
        </section>
        <section title="Profile anchor="sec.profile-names" numbered="true" toc="default">
          <name>Profile Names and Interoperability"
        anchor="sec.profile-names"> Interoperability</name>
          <t>For media m= "m=" sections, JSEP implementations MUST <bcp14>MUST</bcp14> support
          the "UDP/TLS/RTP/SAVPF" profile specified in
          <xref target="RFC5764"></xref> target="RFC5764" format="default"/> as well as the "TCP/DTLS/RTP/SAVPF"
          profile specified in <xref target="RFC7850"></xref>, target="RFC7850" format="default"/> and MUST <bcp14>MUST</bcp14> indicate
          one of these profiles for each media m= "m=" line they produce in an offer.
          For data m= "m=" sections, implementations MUST <bcp14>MUST</bcp14> support the
          "UDP/DTLS/SCTP" profile as well as the "TCP/DTLS/SCTP" profile, profile and
          MUST
          <bcp14>MUST</bcp14> indicate one of these profiles for each data m= "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
          <xref target="I-D.ietf-mmusic-ice-sip-sdp"></xref>, Section 3.2.1.2.
          </t> target="RFC8839" sectionFormat="comma" section="4.2.1.2"/>.</t>
          <t>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 <bcp14>MUST</bcp14> follow the following rules when
          processing the media m= "m=" sections in a received offer:</t>

          <t>
            <list style="symbols">
          <ul spacing="normal">
            <li>
              <t>Any profile in the offer matching one of the following
              MUST
              <bcp14>MUST</bcp14> be accepted:
              <list style="symbols">

                <t>"RTP/AVP" (Defined
              </t>
              <ul spacing="normal">
                <li>"RTP/AVP" (defined in
                <xref target="RFC4566"></xref>, Section 8.2.2)</t>

                <t>"RTP/AVPF" (Defined target="RFC4566" sectionFormat="comma" section="8.2.2"/>)</li>
                <li>"RTP/AVPF" (defined in
                <xref target="RFC4585"></xref>, Section 9)</t>

                <t>"RTP/SAVP" (Defined in
                <xref target="RFC3711"></xref>, Section 12)</t>

                <t>"RTP/SAVPF" (Defined in
                <xref target="RFC5124"></xref>, Section 6)</t>

                <t>"TCP/DTLS/RTP/SAVP" (Defined in
                <xref target="RFC7850"></xref>, Section 3.4)</t>

                <t>"TCP/DTLS/RTP/SAVPF" (Defined in
                <xref target="RFC7850"></xref>, Section 3.5)</t>

                <t>"UDP/TLS/RTP/SAVP" (Defined target="RFC4585" sectionFormat="comma" section="9"/>)</li>
                <li>"RTP/SAVP" (defined in
                <xref target="RFC3711" sectionFormat="comma" section="12"/>)</li>
                <li>"RTP/SAVPF" (defined in
                <xref target="RFC5124" sectionFormat="comma" section="6"/>)</li>
                <li>"TCP/DTLS/RTP/SAVP" (defined in
                <xref target="RFC7850" sectionFormat="comma" section="3.4"/>)</li>
                <li>"TCP/DTLS/RTP/SAVPF" (defined in
                <xref target="RFC7850" sectionFormat="comma" section="3.5"/>)</li>
                <li>"UDP/TLS/RTP/SAVP" (defined in
                <xref target="RFC5764"></xref>, Section 9)</t>

                <t>"UDP/TLS/RTP/SAVPF" (Defined target="RFC5764" sectionFormat="comma" section="9"/>)</li>
                <li>"UDP/TLS/RTP/SAVPF" (defined in
                <xref target="RFC5764"></xref>, Section 9)</t>
              </list></t>

              <t>The target="RFC5764" sectionFormat="comma" section="9"/>)</li>
              </ul>
            </li>
            <li>The profile in any "m=" line in any generated answer
              MUST
              <bcp14>MUST</bcp14> exactly match the profile provided in the offer.</t>

              <t>Because offer.</li>
            <li>Because DTLS-SRTP is REQUIRED, <bcp14>REQUIRED</bcp14>, the choice of SAVP or
              AVP has no effect; support for DTLS-SRTP is determined by
              the presence of one or more "a=fingerprint" attribute. attributes.
              Note that lack of an "a=fingerprint" attribute will lead
              to negotiation failure.</t>

              <t>The failure.</li>
            <li>The use of AVPF or AVP simply controls the timing
              rules used for RTCP feedback. If AVPF is provided, 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 AVP-compatible mode and use
              a value of "trr-int=4000".</t>

              <t>For "trr-int=4000".</li>
            <li>For data m= "m=" sections, implementations MUST <bcp14>MUST</bcp14> support
              receiving the "UDP/DTLS/SCTP", "TCP/DTLS/SCTP", or
              "DTLS/SCTP" (for backwards compatibility) profiles.</t>
            </list>
          </t> profiles.</li>
          </ul>
          <t>Note that re-offers by JSEP implementations MUST <bcp14>MUST</bcp14> 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.</t>
        </section>
      </section>
      <section anchor="sec-create-offer" title="Constructing numbered="true" toc="default">
        <name>Constructing an Offer"> Offer</name>
        <t>When createOffer is called, a new SDP description must be
        created that includes the functionality specified in
        <xref target="I-D.ietf-rtcweb-rtp-usage"></xref>. target="RFC8834" format="default"/>. The exact
        details of this process are explained below.</t>
        <section title="Initial Offers" anchor="sec.initial-offers"> anchor="sec.initial-offers" numbered="true" toc="default">
          <name>Initial Offers</name>
          <t>When createOffer is called for the first time, the result
          is known as the initial offer.</t>
          <t>The first step in generating an initial offer is to
          generate session-level attributes, as specified in
          <xref target="RFC4566"></xref>, Section 5. target="RFC4566" sectionFormat="comma" section="5"/>. Specifically:
          <list style="symbols">

            <t>The
          </t>
          <ul spacing="normal">
            <li>The first SDP line MUST <bcp14>MUST</bcp14> be "v=0", as specified in
            <xref target="RFC4566"></xref>, target="RFC4566" sectionFormat="comma" section="5.1"/>.</li>
            <li>The second SDP line <bcp14>MUST</bcp14> be an "o=" line, as specified
            in
            <xref target="RFC4566" sectionFormat="comma" section="5.2"/>.

<!-- [rfced] Sections 5.2.1 and subsequent:  Several instances of
", as specified in [RFC..." are confusing as written.  Please see the
items below, and let us know if we may either remove the leading
commas or rephrase in these instances (e.g., use "defined" instead of
"specified").  (For example, in the current text it looks like
[RFC4566], Section 5.1</t>

            <t>The 5.1 sets the requirement, but we don't see a
requirement listed there - only a definition.)

Examples from original:
 o  The first SDP line MUST be "v=0", as specified in [RFC4566],
    Section 5.1

 o  The second SDP line MUST be an "o=" line, as specified in
            <xref target="RFC4566"></xref>,
    [RFC4566], Section 5.2.
...
 o  The third SDP line MUST be a "s=" line, as specified in [RFC4566],
    Section 5.3;
...
 o  A "t=" line MUST be added, as specified in [RFC4566], Section 5.9;
...
 The m= line MUST be followed immediately by a "c=" line, as specified
 in [RFC4566], Section 5.7.
...
*  If RTCP mux is indicated, prepare to demux RTP and RTCP from
   the RTP ICE component, as specified in [RFC5761],
   Section 5.1.3.
...
 If media is already being transmitted,
 the same SSRC SHOULD be used unless the clockrate of the new
 codec is different, in which case a new SSRC MUST be chosen, as
 specified in [RFC7160], Section 3.1. -->

 The value of
            the &lt;username&gt; field SHOULD <bcp14>SHOULD</bcp14> be "-". The sess-id MUST <bcp14>MUST</bcp14>
            be representable by a 64-bit signed integer, and the
           value MUST <bcp14>MUST</bcp14> be less than (2**63)-1. It is RECOMMENDED that the
            sess-id be constructed by generating a 64-bit quantity with (2**63)-1
	    ((2<sup>63</sup>)-1).

<!-- [rfced] Section 5.2.1:  Equations and other math-related items:
Per <https://www.rfc-editor.org/materials/FAQ-xml2rfcv3.html>,
xml2rfc v3 provides the highest bit set ability to zero use superscript (among other
features previously not available).  Please note that superscript
will display in the .html and .pdf files but not in the .txt file
(where it appears as "(2^(63))-1").  We added the "superscript"
version of "(2**63)-1" in parentheses here, to show you how it would
appear.  Please let us know if you would like to use the superscript
feature.

Also, if there are other items in this document that could use
similar treatment, please let us know.

Original:
 The sess-id MUST be representable by a 64-bit signed
 integer, and the value MUST be less than (2**63)-1.

Currently:
  .xml (to be updated per your preference):
    ... less than (2**63)-1 ((2<sup>63</sup>)-1). -->

 It is <bcp14>RECOMMENDED</bcp14> 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
            &lt;nettype&gt; &lt;addrtype&gt; &lt;unicast-address&gt;
            tuple SHOULD <bcp14>SHOULD</bcp14> 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
            <xref target="I-D.ietf-rtcweb-ip-handling" />. target="RFC8828" format="default"/>. As mentioned in
            <xref target="RFC4566"></xref>, target="RFC4566" format="default"/>, the entire o= "o=" line needs to
            be unique, but selecting a random number for
            &lt;sess-id&gt; is sufficient to accomplish this.</t>

            <t>The this.</li>
            <li>The third SDP line MUST <bcp14>MUST</bcp14> be a "s=" line, as specified in
            <xref target="RFC4566"></xref>, Section 5.3; target="RFC4566" sectionFormat="comma" section="5.3"/>; to match the
            "o=" line, a single dash SHOULD <bcp14>SHOULD</bcp14> be used as the session
            name, e.g. e.g., "s=-". Note that this differs from the advice in

            <xref target="RFC4566" /> format="default"/>, which proposes a single space, but
            as both "o=" and "s=" are meaningless in JSEP, having the
            same meaningless value seems clearer.</t>

            <t>Session clearer.</li>
            <li>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
            <bcp14>SHOULD NOT</bcp14> be included.</t>

            <t>Encryption included.</li>
            <li>Encryption Keys ("k=") lines do not provide sufficient
            security and MUST NOT <bcp14>MUST NOT</bcp14> be included.</t>

            <t>A included.</li>
            <li>A "t=" line MUST <bcp14>MUST</bcp14> be added, as specified in
            <xref target="RFC4566"></xref>, Section 5.9; target="RFC4566" sectionFormat="comma" section="5.9"/>; both
            &lt;start-time&gt; and &lt;stop-time&gt; SHOULD <bcp14>SHOULD</bcp14> be set to
            zero, e.g. e.g., "t=0 0".</t>

            <t>An 0".</li>
            <li>An "a=ice-options" line with the "trickle" and "ice2"
            options MUST <bcp14>MUST</bcp14> be added, as specified in <xref
            target="I-D.ietf-ice-trickle"></xref>, Section 3
	    target="RFC8840" sectionFormat="comma" section="4.1.1"/> and
            <xref target="RFC8445"></xref>, target="RFC8445" sectionFormat="comma" section="10"/>.

<!-- [rfced] Section 10.</t>

            <t>If WebRTC identity is being used, an "a=identity" 5.2.1:  We found this RFC Editor Note on
<https://datatracker.ietf.org/doc/draft-ietf-rtcweb-jsep/writeup/>:

"OLD:
   o  An "a=ice-options" line with the "trickle" option MUST be added,
      as described specified in
            <xref target="I-D.ietf-rtcweb-security-arch" />, [I-D.ietf-ice-trickle], Section
            5.</t>
          </list></t>

          <t>The next step is to generate m= sections, 4.

NEW:
   o  An "a=ice-options" line with the "trickle" option MUST be added,
      as specified in
          <xref target="RFC4566" />, [I-D.ietf-mmusic-trickle-ice-sip], Section 5.14. An m= section is
          generated for each RtpTransceiver 4.1.1."

Please note that has been added to the
          PeerConnection, excluding any stopped RtpTransceivers; this
          is done "OLD" text does not match what we found in the order the RtpTransceivers were added to
provided draft:

 o  An "a=ice-options" line with the
          PeerConnection. If there are no such RtpTransceivers, no m=
          sections are generated; more can "trickle" and "ice2" options MUST
    be added later, added, as discussed specified in
          <xref target="RFC3264" />, [I-D.ietf-ice-trickle], Section 5.</t>

          <t>For each m= section generated for an RtpTransceiver,
          establish a mapping between the transceiver 3 and
    [RFC8445], Section 10.

We updated as follows.  Is the index of "and [RFC8445], Section 10" still
applicable?

Currently:
 *  An "a=ice-options" line with the generated m= section.</t>

          <t>Each m= section, provided it "trickle" and "ice2" options MUST
    be added, as specified in [RFC8840], Section 4.1.1 and [RFC8445],
    Section 10. -->

</li>
            <li>If WebRTC identity is being used, an "a=identity" line, as described in
            <xref target="RFC8827" sectionFormat="comma" section="5"/>, needs
	    to be included.

<!-- [rfced] Section 5.2.1:  This is the only bullet item in this
list that (1) is a sentence fragment, unlike the rest and (2) does
not contain "RFC 2119" language.  We have updated as follows, but
please let us know if further changes are needed.

Original:
 o  If WebRTC identity is being used, an "a=identity" line as
    described in [I-D.ietf-rtcweb-security-arch], Section 5.

Currently:
 *  If WebRTC identity is being used, an "a=identity" line, as
    described in [RFC8827], Section 5, needs to be included. -->

</li>
          </ul>
          <t>The next step is to generate "m=" sections, as specified in
          <xref target="RFC4566" sectionFormat="comma" 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
          <xref target="RFC3264" sectionFormat="comma" section="5"/>.</t>
          <t>For each "m=" section generated for an RtpTransceiver,
          establish a mapping between the transceiver and the index of
          the generated "m=" section.</t>
          <t>Each "m=" section, provided it is not marked as bundle-only,
          MUST
          <bcp14>MUST</bcp14> generate a unique set of ICE credentials and gather its
          own unique set of ICE candidates. Bundle-only m= "m=" sections
          MUST NOT
          <bcp14>MUST NOT</bcp14> contain any ICE credentials and MUST NOT <bcp14>MUST NOT</bcp14> gather any
          candidates.</t>
          <t>For DTLS, all m= "m=" sections MUST <bcp14>MUST</bcp14> use any and all the certificate(s) certificates
          that have been specified for the PeerConnection; as a result,
          they MUST <bcp14>MUST</bcp14> all have the same
          <xref target="RFC8122"></xref> fingerprint value(s), value or values
          <xref target="RFC8122" format="default"/>, or these
          value(s) MUST
          values <bcp14>MUST</bcp14> be session-level attributes.</t>
          <t>Each m= "m=" section should be generated as specified in
          <xref target="RFC4566"></xref>, Section 5.14. target="RFC4566" sectionFormat="comma" section="5.14"/>. For the m= "m=" line
          itself, the following rules MUST <bcp14>MUST</bcp14> be followed:
          <list style="symbols">

            <t>If
          </t>
          <ul spacing="normal">
            <li>If the m= "m=" section is marked as bundle-only, then the
            port value MUST <bcp14>MUST</bcp14> be set to 0. Otherwise, the port value is
            set to the port of the default ICE candidate for this m= "m="
            section, but given that no candidates are available yet,
            the "dummy" port value of 9 (Discard) MUST <bcp14>MUST</bcp14> be used, as
            indicated in
            <xref target="I-D.ietf-ice-trickle"></xref>, Section
            5.1.</t>

            <t>To target="RFC8840" sectionFormat="comma" section="4.1.1"/>.</li>
<!-- This instance of ", as specified in" is OK. -->
            <li>To properly indicate use of DTLS, the &lt;proto&gt;
            field MUST <bcp14>MUST</bcp14> be set to "UDP/TLS/RTP/SAVPF", as specified in
            <xref target="RFC5764" />, Section 8.</t>

            <t>If sectionFormat="comma" section="8"/>.</li>
            <li>If codec preferences have been set for the associated
            transceiver, media formats MUST <bcp14>MUST</bcp14> be generated in the
            corresponding order, order and MUST <bcp14>MUST</bcp14> exclude any codecs not
            present in the codec preferences.</t>

            <t>Unless preferences.</li>
            <li>Unless excluded by the above restrictions, the media
            formats MUST <bcp14>MUST</bcp14> include the mandatory audio/video codecs as
            specified in
            <xref target="RFC7874"></xref>, Section 3, target="RFC7874" sectionFormat="comma" section="3"/> and
            <xref target="RFC7742"></xref>, Section 5.</t>
          </list></t>

          <t>The m= line MUST be followed immediately by a "c=" line, target="RFC7742" sectionFormat="comma" section="5"/>.

<!-- [rfced] Sections 5.2.1, 5.3.1, and 5.11:  As it appears that
each instance of "Section" in these sentences refers to the section
number of the cited RFC, as opposed to a section in this document,
we removed the commas after each first-listed section number, as
follows (particularly for any readers of the text file).  Please let
us know if anything is incorrect.

Original:
 o  Unless excluded by the above restrictions, the media formats MUST
    include the mandatory audio/video codecs as specified in
          <xref target="RFC4566"></xref>,
    [RFC7874], Section 5.7. 3, and [RFC7742], Section 5.
...
 o  For each media format on the m= line, "a=rtpmap" and "a=fmtp"
    lines, as specified in [RFC4566], Section 6, and [RFC3264],
    Section 5.1.
...
 o  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.
...
 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.

Currently:
 *  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.
...
 *  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 media format on the "m=" line, "a=rtpmap" and "a=fmtp"
    lines, as specified in [RFC4566], Section 6 and [RFC3264],
    Section 6.1.
...
 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. -->
</li>
          </ul>

          <t>The "m=" line <bcp14>MUST</bcp14> be followed immediately by a "c=" line,
          as specified in
          <xref target="RFC4566" sectionFormat="comma" section="5.7"/>. Again, as no
          candidates are available yet, the "c=" line must contain the
          "dummy" value "IN IP4 0.0.0.0", as defined in
          <xref target="I-D.ietf-ice-trickle"></xref>, Section 5.1.</t> target="RFC8840" sectionFormat="comma" section="4.1.1"/>.</t>
          <t>
          <xref target="I-D.ietf-mmusic-sdp-mux-attributes" /> target="RFC8859" format="default"/> groups
          SDP attributes into different categories. To avoid
          unnecessary duplication when bundling, attributes of category
          IDENTICAL or TRANSPORT MUST NOT <bcp14>MUST NOT</bcp14> be repeated in bundled m= "m="
          sections, repeating the guidance from
          <xref target="I-D.ietf-mmusic-sdp-bundle-negotiation" />,
          Section 8.1. This includes m= sections for which bundling has
          been negotiated target="RFC8843"
          sectionFormat="comma" section="8.1"/>.

<!-- [rfced] Sections 5.2.1 and is still desired, as well as m= sections
          marked as bundle-only.</t>

          <t>The following 5.2.2:  Please confirm that these
citations for RFC 8843 [I-D.ietf-mmusic-sdp-bundle-negotiation], Section 8.1
are correct; we could not see a relationship.

Original:
 To avoid unnecessary duplication when
 bundling, attributes of category IDENTICAL or TRANSPORT MUST NOT be
 repeated in bundled m= sections, repeating the guidance from
 [I-D.ietf-mmusic-sdp-bundle-negotiation], Section 8.1.
...
 Instead,
 JSEP implementations MUST simply omit parameters in the IDENTICAL
 and TRANSPORT categories for bundled m= sections, as described in
 [I-D.ietf-mmusic-sdp-bundle-negotiation], Section 8.1. -->

 This includes "m=" sections for which bundling has
          been negotiated and is still desired, as well as "m=" sections
          marked as bundle-only.</t>
          <t>The following attributes, which are of a category other
          than IDENTICAL or TRANSPORT, MUST <bcp14>MUST</bcp14> be included in each m= "m="
          section:</t>

          <t>
            <list style="symbols">

              <t>An
          <ul spacing="normal">
            <li>An "a=mid" line, as specified in
              <xref target="RFC5888"></xref>, target="RFC5888" sectionFormat="comma" section="4"/>. All MID values
              <bcp14>MUST</bcp14> be generated in a fashion that does not leak user
              information, e.g., randomly or using a per-PeerConnection
              counter, and <bcp14>SHOULD</bcp14> be 3 bytes or less, to allow them to
              efficiently fit into the RTP header extension defined in
              <xref target="RFC8843" sectionFormat="comma" section="14"/>.

<!-- [rfced] Section 4. 5.2.1:  Please confirm that this citation is
correct; we could not see a relationship between Section 14 of
RFC 8843 [I-D.ietf-mmusic-sdp-bundle-negotiation] and the RTP header extension.

Original:
 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
              <xref target="I-D.ietf-mmusic-sdp-bundle-negotiation">
              </xref>,
 [I-D.ietf-mmusic-sdp-bundle-negotiation], Section 14. -->

 Note that this does not set the
              RtpTransceiver mid property, as that only occurs when the
              description is applied. The generated MID value can be
              considered a "proposed" MID at this point.</t>

              <t>A point.</li>
            <li>A direction attribute which that is the same as that of the
              associated transceiver.</t>

              <t>For transceiver.</li>
            <li>For each media format on the m= "m=" line, "a=rtpmap" and "a=fmtp" lines, as specified in
              <xref target="RFC4566"></xref>, Section 6, target="RFC4566" sectionFormat="comma" section="6"/> and
              <xref target="RFC3264"></xref>, Section 5.1.</t>

              <t>For target="RFC3264" sectionFormat="comma" section="5.1"/>.</li>
            <li>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
              <xref target="RFC4588"></xref>, Section 8.1.</t>

              <t>For target="RFC4588" sectionFormat="comma" section="8.1"/>.</li>
            <li>For each supported FEC Forward Error Correction (FEC) mechanism, "a=rtpmap" and
              "a=fmtp" lines, as specified in
              <xref target="RFC4566"></xref>, Section 6. target="RFC4566" sectionFormat="comma" section="6"/>. The FEC
              mechanisms that MUST <bcp14>MUST</bcp14> be supported are specified in
              <xref target="I-D.ietf-rtcweb-fec"></xref>, Section 6, target="RFC8854" sectionFormat="comma" section="6"/>,
              and specific usage for each media type is outlined in
              Sections 4 <xref target="sec.interface" format="counter"/> and 5.</t>

              <t>If this m= section <xref target="sec.sdp-interaction-procedure"
	      format="counter"/>.

<!-- [rfced] Sections 5.2.1 and 5.3.1:  Please confirm that Section 6
of RFC 8854 [I-D.ietf-rtcweb-fec] is for media with configurable
              durations the correct section to cite here; we
could not see a relationship.

Also, should "Sections 4 and 5" be "Sections 4 and 5 of media per packet, e.g., audio, an
              "a=maxptime" line, indicating
RFC 8854 [I-D.ietf-rtcweb-fec]"?  The hyperlinks in the maximum amount .html file steer to
Sections 4 and 5 of
              media, specified this document, and we could not find relevant
text in milliseconds, those sections.

Original:
 The FEC mechanisms that can
 MUST be
              encapsulated in each supported are specified in [I-D.ietf-rtcweb-fec],
 Section 6, and specific usage for each media type is outlined in
 Sections 4 and 5.
...
 The FEC mechanisms that
 MUST be supported are specified in [I-D.ietf-rtcweb-fec],
 Section 6, and specific usage for each media type is outlined in
 Sections 4 and 5. -->

</li>
            <li>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
              <xref target="RFC4566"></xref>, Section 6. target="RFC4566" sectionFormat="comma" section="6"/>. This value is
              set to the smallest of the maximum duration values across
              all the codecs included in the m= section.</t>

              <t>If "m=" section.</li>
            <li>If this m= "m=" section is for video media, media and there are
              known limitations on the size of images which that can be
              decoded, an "a=imageattr" line, as specified in
              <xref target="sec.imageattr"></xref>.</t>

              <t>For target="sec.imageattr" format="default"/>.</li>
            <li>For each supported RTP header extension, an "a=extmap"
              line, as specified in
              <xref target="RFC5285"></xref>, target="RFC5285" sectionFormat="comma" section="5"/>.

<!-- [rfced] Sections 5.2.1 and subsequent:  RFC 5285 has been
obsoleted by RFC 8285.  Should RFC 8285 be cited throughout the
document and listed as a Normative Reference instead of RFC 5285?
If yes, please review all textual citations (eight, by our count),
and let us know if any of the section numbers (Sections 5, 6, and
7 of RFC 5285 are cited) also need to be updated.

Original:
 o  For each supported RTP header extension, an "a=extmap" line, as
    specified in [RFC5285], Section 5.
...
 o  For each supported RTP header extension that is present in the
    offer, an "a=extmap" line, as specified in [RFC5285], Section 5.
...

... etc. -->

 The list of
              header extensions that SHOULD/MUST <bcp14>SHOULD</bcp14>/<bcp14>MUST</bcp14> be supported is
              specified in
              <xref target="I-D.ietf-rtcweb-rtp-usage"></xref>, Section
              5.2. target="RFC8834" sectionFormat="comma" section="5.2"/>. Any header extensions that require encryption MUST <bcp14>MUST</bcp14>
              be specified as indicated in
              <xref target="RFC6904"></xref>, Section 4.</t>

              <t>For target="RFC6904" sectionFormat="comma" section="4"/>.</li>
            <li>For each supported RTCP feedback mechanism, an
              "a=rtcp-fb" line, as specified in
              <xref target="RFC4585"></xref>, Section 4.2. target="RFC4585" sectionFormat="comma" section="4.2"/>. The list of
              RTCP feedback mechanisms that SHOULD/MUST <bcp14>SHOULD</bcp14>/<bcp14>MUST</bcp14> be supported is
              specified in
              <xref target="I-D.ietf-rtcweb-rtp-usage"></xref>, Section
              5.1.</t> target="RFC8834" sectionFormat="comma" section="5.1"/>.</li>
            <li>
              <t>If the RtpTransceiver has a sendrecv or sendonly
              direction:
              <list style="symbols">

                <t>For
              </t>
              <ul spacing="normal">
                <li>For each MediaStream that was associated with the
                transceiver when it was created via addTrack or
                addTransceiver, an "a=msid" line, as specified in
                <xref target="I-D.ietf-mmusic-msid"></xref>, Section 2, target="RFC8830" sectionFormat="comma" section="2"/>,
                but omitting the "appdata" field.</t>
              </list></t>

              <t>If field.</li>
              </ul>
            </li>
            <li>If the RtpTransceiver has a sendrecv or sendonly
              direction, and the application has specified RID values
              or has specified more than one encoding in the
              RtpSenders's parameters, an "a=rid" line for each
              encoding specified. The "a=rid" line is specified in
              <xref target="I-D.ietf-mmusic-rid"></xref>, target="RFC8851" format="default"/>, and its
              direction MUST <bcp14>MUST</bcp14> be "send". If the application has chosen a
              RID value, it MUST <bcp14>MUST</bcp14> be used as the rid-identifier;
              otherwise
              otherwise, a RID value MUST <bcp14>MUST</bcp14> be generated by the
              implementation. RID values MUST <bcp14>MUST</bcp14> be generated in a fashion
              that does not leak user information, e.g., randomly or
              using a per-PeerConnection counter, and SHOULD <bcp14>SHOULD</bcp14> be 3 bytes
              or less, to allow them to efficiently fit into the RTP
              header extension defined in
              <xref target="I-D.ietf-avtext-rid"></xref>, target="RFC8852" sectionFormat="comma" section="3"/>.

<!-- [rfced] Section 5.2.1:  Section 3 of RFC 8852 [I-D.ietf-avtext-rid] lists
several header extensions.  Should "extension" in this sentence be
"extensions," or should one of the extension types be specified here?

Original:
 RID values MUST be generated in a fashion that
 does not leak user information, e.g., randomly or using a per-
 PeerConnection counter, and SHOULD be 3 bytes or less, to allow
 them to efficiently fit into the RTP header extension defined in
 [I-D.ietf-avtext-rid], Section 3. -->

 If
              no encodings have been specified, or only one encoding is
              specified but without a RID value, then no "a=rid" lines
              are generated.</t>

              <t>If generated.</li>
            <li>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
              <xref target="I-D.ietf-mmusic-sdp-simulcast"></xref>,
              Section 6.2. target="RFC8853" sectionFormat="comma"
              section="6.2"/>. The list of RIDs MUST <bcp14>MUST</bcp14>
	      include all of the RID identifiers
              used in the "a=rid" lines for this m=
              section.</t>

              <t>If "m="
              section.</li>
            <li>If the bundle policy for this PeerConnection is set to
              "max-bundle", and this is not the first "m=" section, or
              the bundle policy is set to "balanced", and this is not
              the first "m=" section for this media type, an
              "a=bundle-only" line.

<!-- [rfced] Section 5.2.1:  We had trouble sorting out the "and"
... "or" relationships here.  If the suggested text is not correct,
please clarify.

Original:
 o  If the bundle policy for this PeerConnection is set to "max-
    bundle", and this is not the first m= section, or the bundle
    policy is set to "balanced", and this is not the first m= section
    for this media type, an "a=bundle-only" line.</t>
            </list>
          </t> line.

Suggested:
 o  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. -->

</li>
          </ul>
          <t>The following attributes, which are of category IDENTICAL
          or TRANSPORT, MUST <bcp14>MUST</bcp14> appear only in "m=" sections which that either
          have a unique address or which are associated with the
          bundle-tag.
          BUNDLE-tag. (In initial offers, this means those "m="
          sections which that do not contain an "a=bundle-only"
          attribute.)</t>

          <t>
            <list style="symbols">

              <t>"a=ice-ufrag"
          <ul spacing="normal">
            <li>"a=ice-ufrag" and "a=ice-pwd" lines, as specified in
              <xref target="I-D.ietf-mmusic-ice-sip-sdp"></xref>,
              Section 4.4.</t>

              <t>For target="RFC8839" sectionFormat="comma" section="5.4"/>.</li>
            <li>For each desired digest algorithm, one or more
              "a=fingerprint" lines for each of the endpoint's
              certificates, as specified in
              <xref target="RFC8122"></xref>, Section 5.</t>

              <t>An target="RFC8122" sectionFormat="comma" section="5"/>.</li>
            <li>An "a=setup" line, as specified in
              <xref target="RFC4145"></xref>, Section 4, target="RFC4145" sectionFormat="comma" section="4"/> and clarified
              for use in DTLS-SRTP scenarios in
              <xref target="RFC5763"></xref>, Section 5. target="RFC5763" sectionFormat="comma" section="5"/>. The role value
              in the offer MUST <bcp14>MUST</bcp14> be "actpass".</t>

              <t>An "actpass".</li>
            <li>An "a=tls-id" line, as specified in
              <xref target="I-D.ietf-mmusic-dtls-sdp" />, Section
              5.2.</t>

              <t>An target="RFC8842" sectionFormat="comma" section="5.2"/>.</li>
            <li>An "a=rtcp" line, as specified in
              <xref target="RFC3605"></xref>, target="RFC3605" sectionFormat="comma" section="2.1"/>, containing
              the dummy value "9 IN IP4 0.0.0.0", because no candidates
              have yet been gathered.

<!-- [rfced] Sections 5.2.1 and 5.3.1:  Please confirm that
"9 IN IP4 0.0.0.0" (and not "IN IP4 0.0.0.0") is correct in these
two items.

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

              <t>An gathered.
...
 Otherwise, an "a=rtcp" line, as
 specified in [RFC3605], Section 2.1, containing the dummy value "9
 IN IP4 0.0.0.0" (because no candidates have yet been gathered). -->

</li>
            <li>An "a=rtcp-mux" line, as specified in
              <xref target="RFC5761"></xref>, Section 5.1.3.</t>

              <t>If target="RFC5761" sectionFormat="comma" section="5.1.3"/>.</li>
            <li>If the RTP/RTCP multiplexing policy is "require", an
              "a=rtcp-mux-only" line, as specified in
              <xref target="I-D.ietf-mmusic-mux-exclusive" />, Section
              4.</t>

              <t>An target="RFC8858" sectionFormat="comma" section="4"/>.</li>
            <li>An "a=rtcp-rsize" line, as specified in
              <xref target="RFC5506"></xref>, Section 5.</t>
            </list>
          </t> target="RFC5506" sectionFormat="comma" section="5"/>.</li>
          </ul>
          <t>Lastly, if a data channel has been created, a m= an "m=" section
          MUST
          <bcp14>MUST</bcp14> be generated for data. The &lt;media&gt; field MUST <bcp14>MUST</bcp14> be
          set to "application" "application", and the &lt;proto&gt; field MUST <bcp14>MUST</bcp14> be set
          to "UDP/DTLS/SCTP"
          <xref target="I-D.ietf-mmusic-sctp-sdp"></xref>. target="RFC8841" format="default"/>. The "fmt"
          value MUST <bcp14>MUST</bcp14> be set to "webrtc-datachannel" as specified in
          <xref target="I-D.ietf-mmusic-sctp-sdp"></xref>, target="RFC8841" sectionFormat="comma" section="4.1"/>.

<!-- [rfced] Section 5.2.1:  We could not find any mention of
"webrtc-datachannel" in Section 4.1 of RFC 8841 [I-D.ietf-mmusic-sctp-sdp].
Please confirm that this citation is correct and will be clear to
readers.

Original:
 The "fmt" value MUST be set to "webrtc-
 datachannel" as specified in [I-D.ietf-mmusic-sctp-sdp], Section
          4.1.</t> 4.1. -->

</t>
          <t>Within the data m= "m=" section, an "a=mid" line MUST <bcp14>MUST</bcp14> be
          generated and included as described above, along with an
          "a=sctp-port" line referencing the SCTP port number, as
          defined in
          <xref target="I-D.ietf-mmusic-sctp-sdp"></xref>, Section 5.1, target="RFC8841" sectionFormat="comma" section="5.1"/>;
          and, if appropriate, an "a=max-message-size" line, as defined
          in
          <xref target="I-D.ietf-mmusic-sctp-sdp"></xref>, Section
          6.1.</t> target="RFC8841" sectionFormat="comma" section="6.1"/>.</t>
          <t>As discussed above, the following attributes of category
          IDENTICAL or TRANSPORT are included only if the data m= "m="
          section either has a unique address or is associated with the
          bundle-tag
          BUNDLE-tag (e.g., if it is the only m= "m=" section):
          <list style="symbols">

            <t>"a=ice-ufrag"</t>

            <t>"a=ice-pwd"</t>

            <t>"a=fingerprint"</t>

            <t>"a=setup"</t>

            <t>"a=tls-id"</t>
          </list></t>
          </t>
          <ul spacing="normal">
            <li>"a=ice-ufrag"</li>
            <li>"a=ice-pwd"</li>
            <li>"a=fingerprint"</li>
            <li>"a=setup"</li>
            <li>"a=tls-id"</li>
          </ul>
          <t>Once all m= "m=" sections have been generated, a session-level
          "a=group" attribute MUST <bcp14>MUST</bcp14> be added as specified in
          <xref target="RFC5888"></xref>. target="RFC5888" format="default"/>. This attribute MUST <bcp14>MUST</bcp14> have
          semantics "BUNDLE", "BUNDLE" and MUST <bcp14>MUST</bcp14> include the mid identifiers of
          each m= "m=" section. The effect of this is that the JSEP
          implementation offers all m= "m=" sections as one bundle group.
          However, whether the m= "m=" sections are bundle-only or not
          depends on the bundle policy.</t>
          <t>The next step is to generate session-level lip sync groups
          as defined in
          <xref target="RFC5888" />, Section 7. sectionFormat="comma" 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 <bcp14>MUST</bcp14> be added that contains
          the mid values for each RtpTransceiver.</t>
          <t>Attributes which that SDP permits to either be at either the session
          level or the media level SHOULD <bcp14>SHOULD</bcp14> 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 <bcp14>MAY</bcp14> choose to aggregate attributes at the
          session level level, and JSEP implementations MUST <bcp14>MUST</bcp14> be prepared to
          receive attributes in either location.</t>
          <t>Attributes other than the ones specified above MAY <bcp14>MAY</bcp14> be
          included, except for the following attributes attributes, which are
          specifically incompatible with the requirements of
          <xref target="I-D.ietf-rtcweb-rtp-usage"></xref>, target="RFC8834" format="default"/> and MUST
          NOT <bcp14>MUST
          NOT</bcp14> be included:
          <list style="symbols">

            <t>"a=crypto"</t>

            <t>"a=key-mgmt"</t>

            <t>"a=ice-lite"</t>
          </list></t>
          </t>
          <ul spacing="normal">
            <li>"a=crypto"</li>
            <li>"a=key-mgmt"</li>
            <li>"a=ice-lite"</li>
          </ul>
          <t>Note that when bundle is used, any additional attributes
          that are added MUST <bcp14>MUST</bcp14> follow the advice in
          <xref target="I-D.ietf-mmusic-sdp-mux-attributes"></xref> target="RFC8859" format="default"/> on
          how those attributes interact with bundle.</t>
          <t>Note that these requirements are in some cases stricter
          than those of SDP. Implementations MUST <bcp14>MUST</bcp14> be prepared to accept
          compliant SDP even if it would not conform to the
          requirements for generating SDP in this specification.</t>
        </section>
        <section title="Subsequent Offers"
        anchor="sec.subsequent-offers"> anchor="sec.subsequent-offers" numbered="true" toc="default">
          <name>Subsequent Offers</name>
          <t>When createOffer is called a second (or later) time, time or is
          called after a local description has already been installed,
          the processing is somewhat different than for an initial
          offer.</t>
          <t>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
          should be followed, subject to the following restriction:
          <list style="symbols">

            <t>The
          </t>
          <ul spacing="normal">
            <li>The fields of the "o=" line MUST <bcp14>MUST</bcp14> stay the same except
            for the &lt;session-version&gt; field, which MUST <bcp14>MUST</bcp14> increment
            by one on each call to createOffer if the offer might
            differ from the output of the previous call to createOffer;
            implementations MAY <bcp14>MAY</bcp14> opt to increment
            &lt;session-version&gt; on every call. The value of the
            generated &lt;session-version&gt; is independent of the
            &lt;session-version&gt; 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.</t>
          </list></t>
            N+2.</li>
          </ul>
          <t>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 &lt;session-version&gt; 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
          &lt;session-version&gt;.</t>
          <t>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:
          <list style="symbols">

            <t>The
          </t>
          <ul spacing="normal">
            <li>The "s=" and "t=" lines MUST <bcp14>MUST</bcp14> stay the same.</t>

            <t>If same.</li>
            <li>If any RtpTransceiver has been added, added and there exists
            an m= "m=" section with a zero port in the current local
            description or the current remote description, that m= "m="
            section MUST <bcp14>MUST</bcp14> be recycled by generating an m= "m=" section for
            the added RtpTransceiver as if the m= "m=" section were being
            added to the session description (including a new MID
            value),
            value) and placing it at the same index as the m= "m=" section
            with a zero port.</t>

            <t>If port.</li>
            <li>If an RtpTransceiver is stopped and is not associated
            with an m= "m=" section, an m= "m=" section MUST NOT <bcp14>MUST NOT</bcp14> be generated for
            it. This prevents adding back RtpTransceivers whose m= "m="
            sections were recycled and used for a new RtpTransceiver in
            a previous offer/ answer exchange, as described above.</t>

            <t>If above.</li>
            <li>If an RtpTransceiver has been stopped and is associated
            with an m= "m=" section, and the m= "m=" section is not being
            recycled as described above, an m= "m=" section MUST <bcp14>MUST</bcp14> be
            generated for it with the port set to zero and all "a=msid"
            lines removed.</t>

            <t>For removed.</li>
            <li>For RtpTransceivers that are not stopped, the "a=msid"
            line(s) MUST
            line or lines <bcp14>MUST</bcp14> 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 <bcp14>MUST</bcp14>
            be generated according to the same rules as for an initial
            offer.</t>

            <t>Each
            offer.</li>
            <li>Each "m=" and c=" "c=" line MUST <bcp14>MUST</bcp14> be filled in with the port,
            relevant RTP profile, and address of the default candidate for the
            m=
            "m=" section, as described in
            <xref target="I-D.ietf-mmusic-ice-sip-sdp"></xref>,
            Section 3.2.1.2, target="RFC8839" sectionFormat="comma" section="4.2.1.2"/> and clarified in
            <xref target="sec.profile-names"/>. target="sec.profile-names" format="default"/>.
            If no RTP candidates have yet been gathered, dummy
            values MUST <bcp14>MUST</bcp14> still be used, as described above.</t>

            <t>Each above.</li>
            <li>Each "a=mid" line MUST <bcp14>MUST</bcp14> stay the same.</t>

            <t>Each same.</li>
            <li>Each "a=ice-ufrag" and "a=ice-pwd" line <bcp14>MUST</bcp14> 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
            (<xref target="sec.icerestart" format="default"/>) was specified.

<!-- [rfced] Section 5.2.2:  As it appears that "either changes to"
means "changes to either," we updated this sentence accordingly.
Please let us know if this is incorrect.

Original (the parentheses around "Section 5.2.3.1" have been fixed):
 o  Each "a=ice-ufrag" and "a=ice-pwd" line MUST stay the same, unless
    the ICE configuration has changed (either changes to the supported
    STUN/TURN servers, or the ICE candidate policy), or the
    "IceRestart" option (
            <xref target="sec.icerestart" /> Section 5.2.3.1 was specified.

Currently:
 *  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= "m="
            section is bundled into another m= "m=" section, it still MUST
            NOT <bcp14>MUST
            NOT</bcp14> contain any ICE credentials.</t>

            <t>If credentials.</li>
            <li>If the m= "m=" section is not bundled into another m= "m="
            section, its "a=rtcp" attribute line MUST <bcp14>MUST</bcp14> be filled in with
            the port and address of the default RTCP candidate, as
            indicated in
            <xref target="RFC5761"></xref>, Section 5.1.3. target="RFC5761" sectionFormat="comma" section="5.1.3"/>. If no RTCP
            candidates have yet been gathered, dummy values <bcp14>MUST</bcp14> be
            used, as described in <xref target="sec.initial-offers"/> above.

<!-- [rfced] Sections 5.2.2, 5.3.1, and 5.3.2:  We changed "the
initial offer section" and "the initial offers section" to
"Section 5.2.1," and we changed "the initial answer section" to
"Section 5.3.1."  Please let us know if these changes are incorrect.

Original:
 If no RTCP candidates have yet been gathered,
 dummy values MUST be used, as described in the initial offer
 section above.</t>

            <t>If above.
...
 The process here is identical to that
 indicated in the m= initial offers section is not bundled above, except that the
 "a=ice-options" line, with the "trickle" option as specified in
 [I-D.ietf-ice-trickle], Section 3, and the "ice2" option as specified
 in [RFC8445], Section 10, is only included if such an option was
 present in the offer.
...
 If no RTCP candidates have yet been gathered, dummy values MUST be
 used, as described in the initial answer section above.

Currently:
 If no RTCP candidates have yet been gathered,
 dummy values MUST be used, as described in Section 5.2.1 above.
...
 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.
...
 If no RTCP candidates have yet been gathered, dummy values MUST be
 used, as described in Section 5.3.1 above. -->

</li>

            <li>If the "m=" section is not bundled into another m= "m="
            section, for each candidate that has been gathered during
            the most recent gathering phase (see
            <xref target="sec.ice-gather-overview"></xref>), target="sec.ice-gather-overview" format="default"/>), an
            "a=candidate" line MUST <bcp14>MUST</bcp14> be added, as defined in
            <xref target="I-D.ietf-mmusic-ice-sip-sdp"></xref>, Section 4.1. target="RFC8839" sectionFormat="comma" section="5.1"/>.
            If candidate gathering for the section has completed, an
            "a=end-of-candidates" attribute MUST <bcp14>MUST</bcp14> be added, as described
            in
            <xref target="I-D.ietf-ice-trickle"></xref>, Section 9.3. target="RFC8840" sectionFormat="comma" section="8.2"/>.
            If the m= "m=" section is bundled into another m= "m=" section, both
            "a=candidate" and "a=end-of-candidates" MUST <bcp14>MUST</bcp14> be
            omitted.</t>

            <t>For RtpTransceivers that are still present, the "a=rid"
            lines MUST stay
            omitted.

<!-- [rfced] Section 5.2.2:  We found this RFC Editor Note on
<https://datatracker.ietf.org/doc/draft-ietf-rtcweb-jsep/writeup/>:

"OLD:
   o  If the same.</t>

            <t>For RtpTransceivers m= section is not bundled into another m= section, for each
      candidate that are still present, any
            "a=simulcast" line MUST stay the same.</t>
          </list></t>

          <t>If the previous offer was applied using
          setLocalDescription, and a corresponding answer from the
          remote side has been applied using setRemoteDescription,
          meaning gathered during the PeerConnection is most recent gathering
      phase (see Section 3.5.1), an "a=candidate" line MUST be added, as
      defined in [RFC5245], Section 4.3., paragraph 3.  If candidate
      gathering for the "have-remote-pranswer"
          or "stable" states, section has completed, an offer is generated based on the
          negotiated session descriptions by following the steps
          mentioned for the "have-local-offer" state above.</t>

          <t>In addition, for each existing, non-recycled, non-rejected
          m= section "a=end-of-candidates"
      attribute MUST be added, as described in [I-D.ietf-ice-trickle],
      Section 9.3.  If 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:
          <list style="symbols">

            <t>The is bundled into another m= line and corresponding "a=rtpmap"
      section, both "a=candidate" and "a=fmtp"
            lines "a=end-of-candidates" MUST only include media formats which have not been
            excluded by the codec preferences of be
      omitted.

NEW:
   o  If the associated
            transceiver, and MUST include all currently available
            formats. Media formats m= section is not bundled into another m= section, for each
      candidate that were previously offered but are
            no longer available (e.g., a shared hardware codec) MAY be
            excluded.</t>

            <t>Unless codec preferences have has been set for the
            associated transceiver, the media formats on gathered during the m= most recent gathering
      phase (see Section 3.5.1), an "a=candidate" line MUST be generated added, as
      defined in [RFC5245], Section 4.3., paragraph 3.  If candidate
      gathering for the same order section has completed, an "a=end-of-candidates"
      attribute MUST be added, as described in
      [I-D.ietf-mmusic-trickle-ice-sip], Section 8.2.  If the most recent
            answer. Any media formats m= section is
      bundled into another m= section, both "a=candidate" and
      "a=end-of-candidates" MUST be omitted."

Please note that were the "OLD" text does not present match what we found in the most
            recent answer MUST be added after all existing formats.</t>

            <t>The RTP header extensions MUST only include those
provided draft (i.e., "[RFC5245], Section 4.3., paragraph 3" versus
"[I-D.ietf-mmusic-ice-sip-sdp], Section 4.1"):

 o  If the m= section is not bundled into another m= section, for each
    candidate that
            are present in has been gathered during the most recent answer.</t>

            <t>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.</t>

            <t>The "a=rtcp" gathering
    phase (see Section 3.5.1), an "a=candidate" line MUST NOT be added if added, as
    defined in [I-D.ietf-mmusic-ice-sip-sdp], Section 4.1.  If
    candidate gathering for the most recent
            answer included section has completed, an "a=rtcp-mux" line.</t>

            <t>The "a=rtcp-mux" line "a=end-of-
    candidates" attribute MUST be the same added, as that described in
    [I-D.ietf-ice-trickle], Section 9.3.  If the
            most recent answer.</t>

            <t>The "a=rtcp-mux-only" line MUST NOT be added.</t>

            <t>The "a=rtcp-rsize" line m= section is bundled
    into another m= section, both "a=candidate" and "a=end-of-
    candidates" MUST NOT be added unless present
            in omitted.

Please review, and let us know if further changes are needed.
(Note:  "[RFC8839]" and "[RFC8840]" are the RFC numbers assigned to
[I-D.ietf-mmusic-ice-sip-sdp] and [I-D.ietf-mmusic-trickle-ice-sip],
respectively.)

Currently:
 *  If the "m=" section is not bundled into another "m=" section, for each
    candidate that has been gathered during the most recent answer.</t>

            <t>An "a=bundle-only" gathering
    phase (see Section 3.5.1), an "a=candidate" line MUST NOT be added, as indicated
    defined in
            <xref target="I-D.ietf-mmusic-sdp-bundle-negotiation" />, [RFC8839], Section 6. Instead, JSEP implementations MUST simply omit
            parameters in the IDENTICAL and TRANSPORT categories 5.1.  If candidate gathering for
            bundled m= sections, the
    section has completed, an "a=end-of-candidates" attribute MUST be
    added, as described in
            <xref target="I-D.ietf-mmusic-sdp-bundle-negotiation" />, [RFC8840], Section 8.1.</t>

            <t>Note that if media m= sections are 8.2.  If the "m=" section
    is bundled into a data
            m= another "m=" section, then certain TRANSPORT both "a=candidate" and IDENTICAL attributes
            may appear in the data m= section even if they would
            otherwise only "a=end-
    of-candidates" MUST be appropriate for a media m= section (e.g.,
            "a=rtcp-mux"). This cannot happen in initial offers because
            in omitted. -->

</li>
            <li>For RtpTransceivers that are still present, the initial offer JSEP implementations always list media
            m= sections (if any) before "a=rid"
            lines <bcp14>MUST</bcp14> stay 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, same.</li>
            <li>For RtpTransceivers that are still present, any "a=bundle-only" m= sections will be bundled
            into
            "a=simulcast" line <bcp14>MUST</bcp14> stay the same.</li>
          </ul>
          <t>If the previous offer was applied using
          setLocalDescription, and a preceding non-bundle-only media m= section.</t>
          </list></t>

          <t>The "a=group:BUNDLE" attribute MUST include corresponding answer from the MID
          identifiers specified
          remote side has been applied using setRemoteDescription,
          meaning the PeerConnection is in the bundle group "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.</t>
          <t>In addition, for each existing, non-recycled, non-rejected
          "m=" section in the most recent
          answer, minus any m= sections 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:
          </t>
          <ul spacing="normal">
            <li>The "m=" line and corresponding "a=rtpmap" and "a=fmtp"
            lines <bcp14>MUST</bcp14> only include media formats that have not been marked as
          rejected, plus any newly added or re-enabled m= sections. In
          other words,
            excluded by the bundle attribute must contain codec preferences of the associated
            transceiver and also <bcp14>MUST</bcp14> include all m=
          sections currently available
            formats. Media formats that were previously bundled, as long as they are
          still alive, as well as any new m= sections.</t>

          <t>"a=group:LS" attributes offered but are
            no longer available (e.g., a shared hardware codec) <bcp14>MAY</bcp14> be
            excluded.</li>
            <li>Unless codec preferences have been set for the
            associated transceiver, the media formats on the "m=" line
            <bcp14>MUST</bcp14> be generated in the same way order as
          for initial offers, with in the additional stipulation that any
          lip sync groups most recent
            answer. Any media formats that were not present in the most
            recent answer
          MUST continue to exist and MUST contain any previously <bcp14>MUST</bcp14> be added after all existing MID identifiers, as long as the identified m=
          sections still exist and formats.</li>
            <li>The RTP header extensions <bcp14>MUST</bcp14> only include those that
            are not rejected, and present in the group
          still contains at least two MID identifiers. This ensures most recent answer.</li>
            <li>The RTCP feedback mechanisms <bcp14>MUST</bcp14> only include those
            that any synchronized "recvonly" m= sections continue to be
          synchronized are present in the new offer.</t>
        </section>
        <section title="Options Handling"
        anchor="sec.options-handling1">

          <t>The createOffer method takes as a parameter an
          RTCOfferOptions object. Special processing is performed when
          generating a SDP description if most recent answer, except for the following options
            case of format-specific mechanisms that are
          present.</t>
          <section title="IceRestart" anchor="sec.icerestart">

            <t>If the "IceRestart" option is specified, with referencing a value of
            "true",
            newly added media format.</li>
            <li>The "a=rtcp" line <bcp14>MUST NOT</bcp14> be added if the offer MUST indicate most recent
            answer included an ICE restart by
            generating new ICE ufrag and pwd attributes, "a=rtcp-mux" line.</li>
            <li>The "a=rtcp-mux" line <bcp14>MUST</bcp14> be the same as specified that in
            <xref target="I-D.ietf-mmusic-ice-sip-sdp"></xref>,
            Section 3.4.1.1.1. If this
            option is specified on an initial offer, it has no effect
            (since a new ICE ufrag and pwd are already generated).
            Similarly, if the ICE configuration has changed, this
            option has no effect, since new ufrag and pwd attributes
            will
            most recent answer.</li>
            <li>The "a=rtcp-mux-only" line <bcp14>MUST NOT</bcp14> be generated automatically. This option is primarily
            useful for reestablishing connectivity added.</li>
            <li>The "a=rtcp-rsize" line <bcp14>MUST NOT</bcp14> be added unless present
            in cases where
            failures are detected by the application.</t>
          </section>
          <section title="VoiceActivityDetection"
          anchor="sec.voiceactivitydetection1">

            <t>Silence suppression, also known most recent answer.</li>
            <li>An "a=bundle-only" line <bcp14>MUST NOT</bcp14> be added, as discontinuous
            transmission ("DTX"), can reduce the bandwidth used for
            audio by switching to a special encoding when voice
            activity is indicated
            in
            <xref target="RFC8843" sectionFormat="comma" section="6"/>.

<!-- [rfced] Section 5.2.2:  We could not detected, at the cost of some fidelity.</t>

            <t>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 find any indication in
            <xref target="RFC3389"></xref>,
RFC 8843 [I-D.ietf-mmusic-sdp-bundle-negotiation], 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 6 that codec an
"a=bundle-only" line MUST NOT be specified added.  Will this be clear to indicate that silence
            suppression for received audio is desired. For example,
            when using the Opus codec
            <xref target="RFC6716" />, the "usedtx=1" parameter,
            specified in
            <xref target="RFC7587" />, would
readers?

Original:
 o  An "a=bundle-only" line MUST NOT be used added, as indicated in the offer.</t>

            <t>If the "VoiceActivityDetection" option is specified,
            with a value of "false", the JSEP implementation
    [I-D.ietf-mmusic-sdp-bundle-negotiation], Section 6.

Possibly:
 *  An "a=bundle-only" line, as described in [RFC8843], Section 6,
    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 added. -->

 Instead, JSEP implementations <bcp14>MUST</bcp14> simply omit
            parameters in the offer. In addition, the
            implementation MUST NOT use silence suppression IDENTICAL and TRANSPORT categories for
            bundled "m=" sections, as described in
            <xref target="RFC8843" sectionFormat="comma" section="8.1"/>.</li>
            <li>Note that if media
            it generates, regardless of whether the "CN" codecs or
            related fmtp parameters "m=" sections are bundled into a data
            "m=" section, then certain TRANSPORT and IDENTICAL attributes
            may appear in the peer's description.
            The impact of these rules is that silence suppression 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 depends on mutual agreement of both sides, which
            ensures consistent handling regardless implementations always list media
            "m=" sections (if any) before the data "m=" section (if any),
            and at least one of which codec is
            used.</t>

            <t>The "VoiceActivityDetection" option does those media "m=" sections will not have
            the "a=bundle-only" attribute. Therefore, in initial
            offers, any
            impact on "a=bundle-only" "m=" sections will be bundled
            into a preceding non-bundle-only media "m=" section.</li>
          </ul>
          <t>The "a=group:BUNDLE" attribute <bcp14>MUST</bcp14> include the setting of MID
          identifiers specified in the "vad" value bundle group in the signaling
            of 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 client to mixer audio level header extension
            described in
            <xref target="RFC6464"></xref>, Section 4.</t>
          </section>
        </section>
      </section>
      <section title="Generating an Answer"
      anchor="sec.generating-an-answer">

        <t>When createAnswer is called, a new SDP description bundle attribute must be
        created contain all "m="
          sections that is compatible with the supplied remote description were previously bundled, as long as they are
          still alive, as well as the requirements specified in
        <xref target="I-D.ietf-rtcweb-rtp-usage"></xref>. The exact
        details of this process any new "m=" sections.</t>
          <t>"a=group:LS" attributes are explained below.</t>
        <section title="Initial Answers" anchor="sec.initial-answers">

          <t>When createAnswer is called generated in the same way as
          for initial offers, with the first time after a
          remote description has been provided, additional stipulation that any
          lip sync groups that were present in the result is known most recent answer
          <bcp14>MUST</bcp14> continue to exist and <bcp14>MUST</bcp14> contain any previously
          existing MID identifiers, as long as the initial answer. If no remote description has been
          installed, an answer cannot be generated, and an error MUST
          be returned.</t>

          <t>Note that the remote description SDP may not have been
          created by a JSEP endpoint identified "m="
          sections still exist and may not conform to all the
          requirements listed in
          <xref target="sec-create-offer"></xref>. 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, rejected, and
          MUST cause the affected m= group
          still contains at least two MID identifiers. This ensures
          that any synchronized "recvonly" "m=" sections continue to be marked as
          rejected.</t>

          <t>The first step
          synchronized in generating the new offer.</t>
        </section>
        <section anchor="sec.options-handling1" numbered="true" toc="default">
          <name>Options Handling</name>
          <t>The createOffer method takes as a parameter an initial answer is to
          generate session-level attributes. The process here
          RTCOfferOptions object. Special processing is
          identical to that indicated in performed when
          generating an SDP description if the initial offers section
          above, except that following options are
          present.</t>
          <section anchor="sec.icerestart" numbered="true" toc="default">
            <name>IceRestart</name>
            <t>If the "a=ice-options" line, "IceRestart" option is specified, with a value of
            "true", the
          "trickle" option as specified in
          <xref target="I-D.ietf-ice-trickle"></xref>, Section 3, offer <bcp14>MUST</bcp14> indicate an ICE restart by
            generating new ICE ufrag and the "ice2" option pwd attributes, as specified
            in
            <xref target="RFC8445"></xref>, Section 10, is
          only included if such an target="RFC8839" sectionFormat="comma" section="4.4.3.1.1"/>. If this
            option was present in the offer.</t>

          <t>The next step is to generate session-level lip sync
          groups, as defined in
          <xref target="RFC5888" />, Section 7. For each group of type
          "LS" present in the specified on an initial offer, select the local RtpTransceivers
          that it has no effect
            (since a new ICE ufrag and pwd are referenced by already generated).
            Similarly, if the MID values in the specified group, ICE configuration has changed, this
            option has no effect, since new ufrag and determine which of them either reference a common local
          MediaStream (specified pwd attributes
            will be generated automatically. This option is primarily
            useful for reestablishing connectivity in cases where
            failures are detected by the calls to
          addTrack/addTransceiver application.</t>
          </section>
          <section anchor="sec.voiceactivitydetection1" numbered="true" toc="default">
            <name>VoiceActivityDetection</name>
            <t>Silence suppression, also known as discontinuous
            transmission ("DTX"), can reduce the bandwidth used for
            audio by switching to create them), or have no
          MediaStream to reference because they were a special encoding when voice
            activity is not created by
          addTrack/addTransceiver. If detected, at least two such RtpTransceivers
          exist, a group of type "LS" with the mid values cost of these
          RtpTransceivers MUST be added. Otherwise the offered "LS"
          group MUST be ignored and no corresponding group generated in some fidelity.</t>
            <t>If the answer.</t>

          <t>As "VoiceActivityDetection" option is specified,
            with a simple example, consider value of "true", the following offer of a
          single <bcp14>MUST</bcp14> indicate support for
            silence suppression in the audio and single video track contained it receives by including
            comfort noise ("CN") codecs for each offered audio codec,
            as specified in
            <xref target="RFC3389" sectionFormat="comma" 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 same
          MediaStream. SDP lines not relevant appropriate fmtp parameters for
            that codec <bcp14>MUST</bcp14> be specified to this example have been
          removed indicate that silence
            suppression for clarity. As explained received audio is desired. For example,
            when using the Opus codec
            <xref target="RFC6716" format="default"/>, the "usedtx=1" parameter,
            specified in
            <xref target="sec-create-offer" />, target="RFC7587" format="default"/>, would be used in the offer.</t>
            <t>If the "VoiceActivityDetection" option is specified,
            with a group value of type "LS" has
          been added that references each track's RtpTransceiver.</t>

          <t>
            <figure>
              <artwork>
                <![CDATA[
          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
                ]]>
</artwork>
            </figure>
          </t>

          <t>If "false", the answerer uses a single MediaStream when it adds its
          tracks, both of its transceivers will reference this stream,
          and so JSEP implementation <bcp14>MUST NOT</bcp14>
            emit "CN" codecs. For codecs that have their own internal
            silence suppression support, the subsequent answer will contain a "LS" group
          identical appropriate fmtp
            parameters for that codec <bcp14>MUST</bcp14> 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, as shown below:</t>

          <t>
            <figure>
              <artwork>
                <![CDATA[
          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
                ]]>
</artwork>
            </figure>
          </t>

          <t>However, if offer. In addition, the answerer groups its tracks into separate
          MediaStreams, its transceivers will reference different
          streams, and so
            implementation <bcp14>MUST NOT</bcp14> use silence suppression for media
            it generates, regardless of whether the subsequent answer will not contain a "LS"
          group.</t>

          <t>
            <figure>
              <artwork>
                <![CDATA[
          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
                ]]>
</artwork>
            </figure>
          </t>

          <t>Finally, if "CN" codecs or
            related fmtp parameters appear in the answerer 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.</t>
            <t>The "VoiceActivityDetection" option does not add any tracks, its
          transceivers will not reference have any MediaStreams, causing
            impact on the
          preferences setting of the offerer to be maintained, and so "vad" value in the
          subsequent answer will contain an identical "LS" group.</t>

          <t>
            <figure>
              <artwork>
                <![CDATA[
          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
                ]]>
</artwork>
            </figure>
          </t>

          <t>The
          <xref target="sec.detailed-example" /> example later in this
          document shows a more involved case signaling
            of "LS" group
          generation.</t>

          <t>The next step the client-to-mixer audio level header extension
            described in
            <xref target="RFC6464" sectionFormat="comma" section="4"/>.</t>
          </section>
        </section>
      </section>
      <section anchor="sec.generating-an-answer" numbered="true" toc="default">
        <name>Generating an Answer</name>
        <t>When createAnswer is to generate m= sections for each m=
          section called, a new SDP description must be
        created that is present in compatible with the supplied remote offer, description
        as well as the requirements specified in
        <xref target="RFC3264"></xref>, Section 6. For the purposes target="RFC8834" format="default"/>. The exact
        details of this discussion, any session-level attributes in the offer
          that process are also valid explained below.</t>
        <section anchor="sec.initial-answers" numbered="true" toc="default">
          <name>Initial Answers</name>
          <t>When createAnswer is called for the first time after a
          remote description has been provided, the result is known as media-level attributes are considered
          to
          the initial answer. If no remote description has been
          installed, an answer cannot be present in each m= section. Each offered m= section
          will have generated, and an associated RtpTransceiver, as described in
          <xref target="sec.applying-a-remote-desc" />. If there are
          more RtpTransceivers than there are m= sections, error <bcp14>MUST</bcp14>
          be returned.</t>
          <t>Note that the
          unmatched RtpTransceivers will need remote description SDP may not have been
          created by a JSEP endpoint and may not conform to be associated all the
          requirements listed in
          <xref target="sec-create-offer" format="default"/>. For many cases, this
          is not a
          subsequent offer.</t>

          <t>For each offered m= section, problem. However, if any of the following
          conditions mandatory SDP attributes
          are true, the corresponding m= section in the
          answer MUST missing or functionality listed as mandatory-to-use
          above is not present, this <bcp14>MUST</bcp14> be marked treated as rejected by setting an error and
          <bcp14>MUST</bcp14> cause the port in the
          m= line affected "m=" sections to zero, be marked as
          rejected.</t>

          <t>The first step in generating an initial answer is to
          generate session-level attributes. The process here is
          identical to that indicated in <xref target="RFC3264"></xref>, Section 6, and further
          processing for this m= section can be skipped:
          <list style="symbols">

            <t>The associated RtpTransceiver has been stopped.</t>

            <t>None of target="sec.initial-offers"/> above, except that the offered media formats are supported and, if
            applicable, allowed by codec preferences.</t>

            <t>The bundle policy is "max-bundle", and this is not "a=ice-options" line, with the
            first m= section or
          "trickle" option as specified in
          <xref target="RFC8840" sectionFormat="comma" section="4.1.3"/>
          and the same bundle group "ice2" option as specified in
          <xref target="RFC8445" sectionFormat="comma" section="10"/>, is
          only included if such an option was present in the offer.

<!-- [rfced] Section 5.3.1:  We found this RFC Editor Note on
<https://datatracker.ietf.org/doc/draft-ietf-rtcweb-jsep/writeup/>:

"OLD:
   The first
            m= section.</t>

            <t>The bundle policy step in generating an initial answer is "balanced", and this to generate
   session-level attributes.  The process here is not identical to that
   indicated in the
            first m= initial offers section for this media type or in above, except that the same bundle
            group
   "a=ice-options" line, with the "trickle" option as specified in
   [I-D.ietf-ice-trickle], Section 4, is only included if such an option
   was present in the offer.

NEW:
   The first m= section for this media type.</t>

            <t>This m= section step in generating an initial answer is to generate
   session-level attributes.  The process here is identical to that
   indicated in a bundle group, and the group's
            offerer tagged m= initial offers section is being rejected due to one of above, except that the above reasons. This requires all m= sections in
   "a=ice-options" line, with the
            bundle group to be rejected, "trickle" option as specified in
            <xref target="I-D.ietf-mmusic-sdp-bundle-negotiation" />,
   [I-D.ietf-mmusic-trickle-ice-sip], Section 7.3.3.</t>
          </list></t>

          <t>Otherwise, each m= section in the answer should then be
          generated as specified 4.1.3, is only included
   if such an option was present in
          <xref target="RFC3264"></xref>, Section 6.1. For the m= line
          itself, offer."

Please note that the following rules must be followed:
          <list style="symbols">

            <t>The port value would normally be set to "OLD" text does not match what we found in the port of
provided draft:

 The first step in generating an initial answer is to generate
 session-level attributes.  The process here is identical to that
 indicated in the
            default ICE candidate for this m= section, but given initial offers section above, except that
            no candidates are available yet, the "dummy" port value of
            9 (Discard) MUST be used,
 "a=ice-options" line, with the "trickle" option as indicated specified in
            <xref target="I-D.ietf-ice-trickle"></xref>,
 [I-D.ietf-ice-trickle], Section
            5.1.</t>

            <t>The &lt;proto&gt; field MUST be set to exactly match the
            &lt;proto&gt; field for 3, and the corresponding m= line "ice2" option as specified
 in the
            offer.</t>

            <t>If codec preferences have been set for the associated
            transceiver, media formats MUST be generated [RFC8445], Section 10, is only included if such an option was
 present in the
            corresponding order, regardless offer.

We updated as follows.  As noted previously, (1) for ease of what was offered, the
reader and
            MUST exclude any codecs not present (2) to create a usable hyperlink in the codec
            preferences.</t>

            <t>Otherwise, .html file, we
changed "the initial offers section above" to "Section 5.2.1 above."

Currently:
 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 media formats on "a=ice-options"
 line, with the m= line MUST be
            generated "trickle" option as specified in [RFC8840],
 Section 4.1.3 and the same order "ice2" option as those offered specified in [RFC8445],
 Section 10, is only included if such an option was present 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.</t>

            <t>In either case, the media formats in the answer MUST
            include at least one format that
 offer. -->

</t>
          <t>The next step is to generate session-level lip sync
          groups, as defined in
          <xref target="RFC5888" sectionFormat="comma" section="7"/>. For each group of type
          "LS" present in the offer,
            but MAY include formats select the local RtpTransceivers
          that are locally supported but not
            present in referenced by the offer, as mentioned MID values in
            <xref target="RFC3264" />, Section 6.1. If no common format
            exists, the m= section is rejected as described above.</t>
          </list></t>

          <t>The m= line MUST be followed immediately by a "c=" line,
          as specified group,
          and determine which of them either reference a common local
          MediaStream (specified in
          <xref target="RFC4566"></xref>, Section 5.7. Again, as no
          candidates are available yet, the "c=" line must contain the
          "dummy" value "IN IP4 0.0.0.0", as defined in
          <xref target="I-D.ietf-ice-trickle"></xref>, Section 5.1.</t>

          <t>If the offer supports bundle, all m= sections 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 <bcp14>MUST</bcp14> be
          bundled must use added. Otherwise, the same ICE credentials and candidates; all
          m= sections not being bundled must use unique ICE credentials offered "LS"
          group <bcp14>MUST</bcp14> be ignored and candidates. Each m= section MUST contain no corresponding group generated in
          the answer.</t>
          <t>As a simple example, consider the following
          attributes (which are offer of attribute types other than IDENTICAL
          and TRANSPORT):
          <list style="symbols">

            <t>If a
          single audio and only if present in the offer, an "a=mid" line, as
            specified in
            <xref target="RFC5888"></xref>, Section 9.1. The "mid"
            value MUST match that specified single video track contained in the offer.</t>

            <t>A direction attribute, determined by applying the rules
            regarding the offered direction specified same
          MediaStream. SDP lines not relevant to this example have been
          removed for clarity. As explained in
          <xref target="RFC3264" />, Section 6.1, and then
            intersecting with the direction target="sec-create-offer" format="default"/>, a group of type "LS" has
          been added that references each track's RtpTransceiver.</t>
          <sourcecode name="" type="sdp"><![CDATA[
          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 ]]></sourcecode>
          <t>If the associated
            RtpTransceiver. For example, in the case where an m=
            section is offered as "sendonly", answerer uses a single MediaStream when it adds its
          tracks, both of its transceivers will reference this stream,
          and so the local transceiver
            is set to "sendrecv", the result in the subsequent answer is will contain a
            "recvonly" direction.</t>

            <t>For each media format on the m= line, "a=rtpmap" and
            "a=fmtp" lines, as specified in
            <xref target="RFC4566"></xref>, Section 6, and
            <xref target="RFC3264"></xref>, Section 6.1.</t>

            <t>If "rtx" is present "LS" group
          identical to that 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 as shown below:</t>
          <sourcecode name="" type="sdp"><![CDATA[
          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 ]]></sourcecode>
          <t>However, if the
            primary codec answerer groups its tracks into separate
          MediaStreams, its transceivers will reference different
          streams, and an "a=fmtp" line that references so the
            payload type of subsequent answer will not contain a "LS"
          group.</t>
          <sourcecode name="" type="sdp"><![CDATA[
          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 ]]></sourcecode>
          <t>Finally, if the primary codec, as specified 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.</t>
          <sourcecode name="" type="sdp"><![CDATA[
          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 ]]></sourcecode>
          <t>The example in <xref target="RFC4588"></xref>, Section 8.1.</t>

            <t>For target="sec.detailed-example" format="default"/> shows a more involved case of "LS" group
          generation.</t>
          <t>The next step is to generate "m=" sections for each supported FEC mechanism, "a=rtpmap" and
            "a=fmtp" lines, "m="
          section that is present in the remote offer, as specified in
          <xref target="RFC4566"></xref>, Section 6. The FEC
            mechanisms target="RFC3264" sectionFormat="comma" section="6"/>. For the purposes
          of this discussion, any session-level attributes in the offer
          that MUST be supported are specified also valid as media-level attributes are considered
          to be present in
            <xref target="I-D.ietf-rtcweb-fec"></xref>, Section 6, and
            specific usage for each media type is outlined in Sections
            4 and 5.</t>

            <t>If this m= "m=" section. Each offered "m=" section is for media with configurable
            durations of media per packet, e.g., audio,
          will have an "a=maxptime"
            line, associated RtpTransceiver, as described in
          <xref target="sec-create-offer" />.</t>

            <t>If this m= section is for video media, and target="sec.applying-a-remote-desc" format="default"/>. If there are
            known limitations on
          more RtpTransceivers than there are "m=" sections, the size of images which can
          unmatched RtpTransceivers will need to be
            decoded, an "a=imageattr" line, as specified associated in
            <xref target="sec.imageattr"></xref>.</t> a
          subsequent offer.</t>
          <t>For each supported RTP header extension that is present offered "m=" section, if any of the following
          conditions are true, the corresponding "m=" section in the offer, an "a=extmap" line,
          answer <bcp14>MUST</bcp14> be marked as specified rejected by setting the port in the
          "m=" line to zero, as indicated in
          <xref target="RFC5285"></xref>, Section 5. The list of
            header extensions that SHOULD/MUST target="RFC3264" sectionFormat="comma" section="6"/>, and further
          processing for this "m=" section can be skipped:
          </t>
          <ul spacing="normal">
            <li>The associated RtpTransceiver has been stopped.</li>
            <li>None of the offered media formats are supported is
            specified in
            <xref target="I-D.ietf-rtcweb-rtp-usage"></xref>, Section
            5.2. Any header extensions that require encryption MUST be
            specified as indicated in
            <xref target="RFC6904"></xref>, and, if
            applicable, allowed by codec preferences.

<!-- [rfced] Section 4.</t>

            <t>For each supported RTCP feedback mechanism 5.3.1:  Does this text mean that is
            present in the offer, an "a=rtcp-fb" line, as specified in
            <xref target="RFC4585"></xref>, Section 4.2. The list of
            RTCP feedback mechanisms that SHOULD/MUST offered
media formats are allowed (in which case it should say "they are
allowed") or are not allowed (in which case "and, if applicable"
should be "or, if applicable")?

Original:
 o  None of the offered media formats are supported and, if
    applicable, allowed by codec preferences. -->

</li>
            <li>The bundle policy is
            specified in
            <xref target="I-D.ietf-rtcweb-rtp-usage"></xref>, Section
            5.1.</t>

            <t>If "max-bundle", and this is not the RtpTransceiver has a sendrecv
            first "m=" section or sendonly
            direction:
            <list style="symbols">

              <t>For each MediaStream that was associated with in the
              transceiver when it was created via addTrack or
              addTransceiver, an "a=msid" line, same bundle group as specified in
              <xref target="I-D.ietf-mmusic-msid"></xref>, Section 2,
              but omitting the "appdata" field.</t>
            </list></t>
          </list></t>

          <t>Each m= section which first
            "m=" section.</li>
            <li>The bundle policy is "balanced", and this is not bundled into another m=
          section, MUST contain the following attributes (which are of
          category IDENTICAL
            first "m=" section for this media type or TRANSPORT):</t>

          <t>
            <list style="symbols">

              <t>"a=ice-ufrag" and "a=ice-pwd" lines, in the same bundle
            group as specified the first "m=" section for this media type.</li>
            <li>This "m=" section is in
              <xref target="I-D.ietf-mmusic-ice-sip-sdp"></xref>,
              Section 4.4.</t>

              <t>For each desired digest algorithm, a bundle group, and the group's
            offerer tagged "m=" section is being rejected due to one or more
              "a=fingerprint" lines for each of
            the endpoint's
              certificates, above reasons. This requires all "m=" sections in the
            bundle group to be rejected, as specified in
            <xref target="RFC8122"></xref>, Section 5.</t>

              <t>An "a=setup" line, as specified in
              <xref target="RFC4145"></xref>, Section 4, and clarified
              for use in DTLS-SRTP scenarios in
              <xref target="RFC5763"></xref>, Section 5. The role value target="RFC8843" sectionFormat="comma" section="7.3.3"/>.</li>
          </ul>
          <t>Otherwise, each "m=" section in the answer MUST should then be "active" or "passive". When
          generated as specified in
          <xref target="RFC3264" sectionFormat="comma" section="6.1"/>. For the
              offer contains "m=" line
          itself, the "actpass" value, as will always following rules must be followed:

<!-- [rfced] Section 5.3.1: Should "must be followed:" here be "MUST be
followed:"? (We ask because (1) we see "For the
              case with JSEP endpoints, the answerer SHOULD use m= line itself, the
              "active" role. Offers from non-JSEP endpoints MAY send
              other values for "a=setup", following
rules MUST be followed:" in which case Section 5.2.1 and (2) all of the answer MUST
              use rules
listed below this sentence include a value consistent with "MUST.")

Original:
 For the value in m= line itself, the offer.</t>

              <t>An "a=tls-id" line, as specified in
              <xref target="I-D.ietf-mmusic-dtls-sdp" />, Section
              5.3.</t>

              <t>If present in
 following rules must be followed: -->

          </t>
          <ul spacing="normal">
            <li>The port value would normally be set to the offer, an "a=rtcp-mux" line, as
              specified in
              <xref target="RFC5761"></xref>, Section 5.1.3. Otherwise,
              an "a=rtcp" line, as specified in
              <xref target="RFC3605"></xref>, Section 2.1, containing port of the dummy value "9 IN IP4 0.0.0.0" (because
            default ICE candidate for this "m=" section, but given that
            no candidates
              have yet been gathered).</t>

              <t>If present in are available yet, the offer, an "a=rtcp-rsize" line, "dummy" port value of
            9 (Discard) <bcp14>MUST</bcp14> be used, as
              specified indicated in
            <xref target="RFC5506"></xref>, Section 5.</t>
            </list>
          </t>

          <t>If a data channel m= section has been offered, a m=
          section MUST also be generated for data. The &lt;media&gt;
          field MUST be set to "application" and the target="RFC8840" sectionFormat="comma" section="4.1.1"/>.</li>
            <li>The &lt;proto&gt; and
          &lt;fmt&gt; fields MUST field <bcp14>MUST</bcp14> be set to exactly match the fields in
            &lt;proto&gt; field for the offer.</t>

          <t>Within corresponding "m=" line in the data m= section, an "a=mid" line MUST
            offer.</li>
            <li>If codec preferences have been set for the associated
            transceiver, media formats <bcp14>MUST</bcp14> be generated and included as described above, along with an
          "a=sctp-port" line referencing the SCTP port number, as
          defined in
          <xref target="I-D.ietf-mmusic-sctp-sdp"></xref>, Section 5.1,
          and, if appropriate, an "a=max-message-size" line, as defined
          in
          <xref target="I-D.ietf-mmusic-sctp-sdp"></xref>, Section
          6.1.</t>

          <t>As discussed above, the following attributes
            corresponding order, regardless of category
          IDENTICAL or TRANSPORT are included only if the data m=
          section is not bundled into another m= section:
          <list style="symbols">

            <t>"a=ice-ufrag"</t>

            <t>"a=ice-pwd"</t>

            <t>"a=fingerprint"</t>

            <t>"a=setup"</t>

            <t>"a=tls-id"</t>
          </list></t>

          <t>Note that if media m= sections are bundled into a data m=
          section, then certain TRANSPORT what was offered, and IDENTICAL attributes may
          also appear
            <bcp14>MUST</bcp14> exclude any codecs not present in the data m= section even if they would
          otherwise only be appropriate for a codec
            preferences.</li>
            <li>Otherwise, the media m= section (e.g.,
          "a=rtcp-mux").</t>

          <t>If "a=group" attributes with semantics of "BUNDLE" are
          offered, corresponding session-level "a=group" attributes
          MUST formats on the "m=" line <bcp14>MUST</bcp14> be added as specified
            generated in
          <xref target="RFC5888"></xref>. These attributes MUST have
          semantics "BUNDLE", and MUST include the all mid identifiers
          from the same order as those offered bundle groups in the current
            remote description, excluding any currently unsupported
            formats. Any currently available media formats that have are not been rejected.
          Note that regardless of the presence of "a=bundle-only"
            present in the offer, no m= sections current remote description <bcp14>MUST</bcp14> be added
            after all existing formats.</li>
            <li>In either case, the media formats in the answer should have an
          "a=bundle-only" line.</t>

          <t>Attributes that are common between all m= sections MAY be
          moved to session-level, if explicitly defined to be valid <bcp14>MUST</bcp14>
            include at
          session-level.</t>

          <t>The attributes prohibited least one format that is present in the creation of offers offer
            but <bcp14>MAY</bcp14> include formats that are
          also prohibited in the creation of answers.</t>
        </section>
        <section title="Subsequent Answers"
        anchor="sec.subsequent-answers">

          <t>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.</t>

          <t>If the previous answer was locally supported but not applied using
          setLocalDescription, meaning the PeerConnection is still
            present in the "have-remote-offer" state, the steps for generating an
          initial answer should be followed, subject to offer, as mentioned in
            <xref target="RFC3264" sectionFormat="comma" section="6.1"/>. If no common format
            exists, the following
          restriction:
          <list style="symbols"> "m=" section is rejected as described above.</li>
          </ul>

          <t>The fields of the "o=" "m=" line MUST stay the same except
            for <bcp14>MUST</bcp14> be followed immediately by a "c=" line,
          as specified in
          <xref target="RFC4566" sectionFormat="comma" section="5.7"/>. Again, as no
          candidates are available yet, the &lt;session-version&gt; field, which MUST increment
            if "c=" line must contain the session description changes
          "dummy" value "IN IP4 0.0.0.0", as defined in any way from the
            previously generated answer.</t>
          </list></t>
          <xref target="RFC8840" sectionFormat="comma" section="4.1.3"/>.</t>
          <t>If any session description was previously supplied to
          setLocalDescription, an answer is generated by following the
          steps in offer supports bundle, all "m=" sections to be
          bundled must use the "have-remote-offer" state above, along with
          these exceptions:
          <list style="symbols">

            <t>The "s=" same ICE credentials and "t=" lines MUST stay the same.</t>

            <t>Each candidates; all
          "m=" sections not being bundled must use unique ICE credentials
          and c=" line MUST be filled in with candidates. Each "m=" section <bcp14>MUST</bcp14> contain the port
            and address following
          attributes (which are of attribute types other than IDENTICAL
          or TRANSPORT):
          </t>
          <ul spacing="normal">
            <li>If and only if present in the default candidate for the m= section, offer, an "a=mid" line, as
            described
            specified in
            <xref target="I-D.ietf-mmusic-ice-sip-sdp"></xref>,
            Section 3.2.1.2. Note target="RFC5888" sectionFormat="comma" section="9.1"/>. The "mid"
            value <bcp14>MUST</bcp14> match that specified in certain cases, the m= line protocol
            may not match that of offer.</li>
            <li>A direction attribute, determined by applying the default candidate, because rules
            regarding the m= line
            protocol value MUST match what was supplied offered direction specified in the offer, as
            described above.</t>

            <t>Each "a=ice-ufrag"
            <xref target="RFC3264" sectionFormat="comma" section="6.1"/>, and "a=ice-pwd" line MUST stay then
            intersecting with the
            same, unless direction of the m= section is restarting, in which case
            new ICE credentials must be created as specified associated
            RtpTransceiver. For example, in
            <xref target="I-D.ietf-mmusic-ice-sip-sdp"></xref>,
            Section 3.4.1.1.1. If the m= case where an "m="
            section is bundled into another m= section, it still MUST
            NOT contain any ICE credentials.</t>

            <t>Each "a=tls-id" line MUST stay offered as "sendonly" and the same unless local transceiver
            is set to "sendrecv", the
            offerer's "a=tls-id" line changed, result in which case the answer is a new
            "a=tls-id" value MUST be created,
            "recvonly" direction.</li>
            <li>For each media format on the "m=" line, "a=rtpmap" and "a=fmtp" lines, as described specified in
            <xref target="I-D.ietf-mmusic-dtls-sdp" />, Section
            5.2.</t>

            <t>Each "a=setup" line MUST use an "active" or "passive"
            role value consistent with the existing DTLS association,
            if the association target="RFC4566" sectionFormat="comma" section="6"/> and
            <xref target="RFC3264" sectionFormat="comma" section="6.1"/>.</li>
            <li>If "rtx" is being continued by present in the offerer.</t>

            <t>RTCP multiplexing must offer, for each primary codec
            where RTP retransmission should be used, and an "a=rtcp-mux" a corresponding
            "a=rtpmap" line
            inserted if and only if indicating "rtx" with the m= section previously used RTCP
            multiplexing.</t>

            <t>If clock rate of the m= section is not bundled into another m= section
            primary codec and RTCP multiplexing is not active, an "a=rtcp" attribute "a=fmtp" line MUST be filled in with that references the port and address
            payload type of the
            default RTCP candidate. If no RTCP candidates have yet been
            gathered, dummy values MUST be used, primary codec, as described specified in the
            initial answer section above.</t>

            <t>If the m= section is not bundled into another m=
            section, for
            <xref target="RFC4588" sectionFormat="comma" section="8.1"/>.</li>
            <li>For each candidate that has been gathered during
            the most recent gathering phase (see supported FEC mechanism, "a=rtpmap" and
            "a=fmtp" lines, as specified in
            <xref target="sec.ice-gather-overview"></xref>), an
            "a=candidate" line MUST target="RFC4566" sectionFormat="comma" section="6"/>. The FEC
            mechanisms that <bcp14>MUST</bcp14> be added, as defined supported are specified in
            <xref target="I-D.ietf-mmusic-ice-sip-sdp"></xref>,
            Section 4.1.
            If candidate gathering target="RFC8854" sectionFormat="comma" section="6"/>, and
            specific usage for the each media type is outlined in Sections
<xref target="sec.interface" format="counter"/> and <xref
target="sec.sdp-interaction-procedure" format="counter"/>.</li>
            <li>If this "m=" section has completed, is for media with configurable
            durations of media per packet, e.g., audio, an
            "a=end-of-candidates" attribute MUST be added, "a=maxptime"
            line, as described in
            <xref target="I-D.ietf-ice-trickle"></xref>, Section 9.3.
            If the m= target="sec-create-offer" format="default"/>.</li>
            <li>If this "m=" section is bundled into another m= section, both
            "a=candidate" for video media and "a=end-of-candidates" MUST be
            omitted.</t>

            <t>For RtpTransceivers that there are not stopped, the "a=msid"
            line(s) MUST stay
            known limitations on the same, regardless size of changes to the
            transceiver's direction or track. If no "a=msid" line images that can be
            decoded, an "a=imageattr" line, as specified in
            <xref target="sec.imageattr" format="default"/>.</li>
            <li>For each supported RTP header extension that is present
            in the current description, "a=msid" line(s) MUST
            be generated according to the same rules as for offer, an initial
            answer.</t>
          </list></t>
        </section>
        <section title="Options Handling"
        anchor="sec.options-handling2">

          <t>The createAnswer method takes "a=extmap" line, as a parameter an
          RTCAnswerOptions object. specified in
            <xref target="RFC5285" sectionFormat="comma" section="5"/>. The set list of parameters for
          RTCAnswerOptions is different than those supported in
          RTCOfferOptions; the IceRestart option is unnecessary, as ICE
          credentials will automatically
            header extensions that <bcp14>SHOULD</bcp14>/<bcp14>MUST</bcp14> be changed for all m= sections
          where the offerer chose to perform ICE restart.</t>

          <t>The following options are supported in
          RTCAnswerOptions.</t>
          <section title="VoiceActivityDetection"
          anchor="sec.voiceactivitydetection2">

            <t>Silence suppression in the answer is handled
            specified in
            <xref target="RFC8834" sectionFormat="comma" section="5.2"/>. Any header extensions that require encryption <bcp14>MUST</bcp14> be
            specified as
            described indicated in
            <xref target="sec.voiceactivitydetection1"></xref>, with
            one exception: if support for silence suppression was not
            indicated target="RFC6904" sectionFormat="comma" section="4"/>.</li>
            <li>For each supported RTCP feedback mechanism that is
            present in the offer, the VoiceActivityDetection
            parameter has no effect, and the answer should be generated an "a=rtcp-fb" line, 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 specified in an answer
            with CN codecs and "usedtx=0").
            <xref target="RFC4585" sectionFormat="comma" section="4.2"/>. The impact list of this rule is
            that an answerer will not try to use silence suppression
            with any endpoint
            RTCP feedback mechanisms that does not offer it, making silence
            suppression support bilateral even with non-JSEP
            endpoints.</t>
          </section>
        </section>
      </section>
      <section title="Modifying an Offer or Answer"
      anchor="sec.modifying-sdp">

        <t>The SDP returned from createOffer or createAnswer MUST NOT <bcp14>SHOULD</bcp14>/<bcp14>MUST</bcp14> be changed before passing it to setLocalDescription. If precise
        control over the SDP supported is needed,
            specified in
            <xref target="RFC8834" sectionFormat="comma" section="5.1"/>.</li>
            <li>
              <t>If the aforementioned
        createOffer/createAnswer options or RtpTransceiver APIs MUST be
        used.</t>

        <t>After calling setLocalDescription with an offer has a sendrecv or answer,
        the application MAY modify sendonly
            direction:
              </t>
              <ul spacing="normal">
                <li>For each MediaStream that was associated with the SDP to reduce its capabilities
        before sending
              transceiver when it to the far side, as long was created via addTrack or
              addTransceiver, an "a=msid" line, as it follows specified in
              <xref target="RFC8830" sectionFormat="comma" section="2"/>,
              but omitting the
        rules above "appdata" field.</li>
              </ul>
            </li>
          </ul>
          <t>Each "m=" section that define a valid JSEP offer is not bundled into another "m="
          section <bcp14>MUST</bcp14> contain the following attributes (which are of
          category IDENTICAL or answer. Likewise,
        an application that has received an offer TRANSPORT):</t>
          <ul spacing="normal">
            <li>"a=ice-ufrag" and "a=ice-pwd" lines, as specified in
              <xref target="RFC8839" sectionFormat="comma" section="5.4"/>.</li>
            <li>For each desired digest algorithm, one or answer from a peer
        MAY modify more
              "a=fingerprint" lines for each of the received SDP, subject to endpoint's
              certificates, as specified in
              <xref target="RFC8122" sectionFormat="comma" section="5"/>.</li>
            <li>An "a=setup" line, as specified in
              <xref target="RFC4145" sectionFormat="comma" section="4"/> and clarified
              for use in DTLS-SRTP scenarios in
              <xref target="RFC5763" sectionFormat="comma" section="5"/>. The role value
              in the same constraints,
        before calling setRemoteDescription.</t>

        <t>As always, answer <bcp14>MUST</bcp14> be "active" or "passive". When the application is solely responsible for what it
        sends to
              offer contains the other party, and all incoming SDP "actpass" value, as will always be
        processed by the
              case with JSEP implementation to endpoints, 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 answerer <bcp14>SHOULD</bcp14> use the
              "active" role. Offers from any non-JSEP endpoints <bcp14>MAY</bcp14> send
              other implementation of this specification.</t>
      </section>
      <section title="Processing a Local Description"
      anchor="sec.processing-a-local-desc">

        <t>When a SessionDescription is supplied to
        setLocalDescription, values for "a=setup", in which case the following steps MUST be performed:
        <list style="symbols">

          <t>If answer <bcp14>MUST</bcp14>
              use a value consistent with the description is of type "rollback", follow value in the
          processing defined offer.</li>
            <li>An "a=tls-id" line, as specified in
              <xref target="sec.processing-a-rollback" /> and skip the
          processing described target="RFC8842" sectionFormat="comma" section="5.3"/>.</li>
            <li>If present in the rest of this section.</t>

          <t>Otherwise, the type of the SessionDescription is checked
          against offer, an "a=rtcp-mux" line, as
              specified in
              <xref target="RFC5761" sectionFormat="comma" section="5.1.3"/>. Otherwise,
              an "a=rtcp" line, as specified in
              <xref target="RFC3605" sectionFormat="comma" section="2.1"/>, containing
              the current state of dummy value "9 IN IP4 0.0.0.0" (because no candidates
              have yet been gathered).</li>
            <li>If present in the PeerConnection:
          <list style="symbols"> offer, an "a=rtcp-rsize" line, as
              specified in
              <xref target="RFC5506" sectionFormat="comma" section="5"/>.</li>
          </ul>
          <t>If the type is "offer", the PeerConnection state MUST be
            either "stable" or "have-local-offer".</t>

            <t>If the type is "pranswer" or "answer", the
            PeerConnection state MUST a data channel "m=" section has been offered, an "m="
          section <bcp14>MUST</bcp14> also be either "have-remote-offer" or
            "have-local-pranswer".</t>
          </list></t>

          <t>If the type is not correct generated for data. The &lt;media&gt;
          field <bcp14>MUST</bcp14> be set to "application", and the current state,
          processing MUST stop &lt;proto&gt; and an error MUST
          &lt;fmt&gt; fields <bcp14>MUST</bcp14> be returned.</t>

          <t>The SessionDescription is then checked to ensure that its
          contents are identical set to those generated exactly match the fields in
          the last call to
          createOffer/createAnswer, offer.</t>
          <t>Within the data "m=" section, an "a=mid" line <bcp14>MUST</bcp14> be
          generated and thus have not been altered, included as
          discussed in
          <xref target="sec.modifying-sdp" />; otherwise, processing
          MUST stop and described above, along with an error MUST be returned.</t>

          <t>Next,
          "a=sctp-port" line referencing the SessionDescription is parsed into a data
          structure, SCTP port number, as described
          defined in
          <xref target="sec.parsing-a-desc" /> below.</t>

          <t>Finally, the parsed SessionDescription is applied target="RFC8841" sectionFormat="comma" section="5.1"/>;
          and, if appropriate, an "a=max-message-size" line, as
          described defined
          in
          <xref target="sec.applying-a-local-desc" /> below.</t>
        </list></t>
      </section>
      <section title="Processing a Remote Description"
      anchor="sec.processing-a-remote-desc">

        <t>When a SessionDescription is supplied to
        setRemoteDescription, target="RFC8841" sectionFormat="comma" section="6.1"/>.</t>
          <t>As discussed above, the following steps MUST be performed:
        <list style="symbols">

          <t>If the description is attributes of type "rollback", follow category
          IDENTICAL or TRANSPORT are included only if the
          processing defined in
          <xref target="sec.processing-a-rollback" /> data "m="
          section is not bundled into another "m=" section:
          </t>
          <ul spacing="normal">
            <li>"a=ice-ufrag"</li>
            <li>"a=ice-pwd"</li>
            <li>"a=fingerprint"</li>
            <li>"a=setup"</li>
            <li>"a=tls-id"</li>
          </ul>
          <t>Note that if media "m=" sections are bundled into a data "m="
          section, then certain TRANSPORT and skip the
          processing described IDENTICAL attributes may
          also appear in the rest of this section.</t>

          <t>Otherwise, the type data "m=" section even if they would
          otherwise only be appropriate for a media "m=" section (e.g.,
          "a=rtcp-mux").</t>
          <t>If "a=group" attributes with semantics of "BUNDLE" are
          offered, corresponding session-level "a=group" attributes
          <bcp14>MUST</bcp14> be added as specified in
          <xref target="RFC5888" format="default"/>. These attributes <bcp14>MUST</bcp14> have
          semantics "BUNDLE" and <bcp14>MUST</bcp14> include all mid identifiers
          from the SessionDescription is checked
          against the current state offered bundle groups that have not been rejected.
          Note that regardless of the PeerConnection:
          <list style="symbols">

            <t>If presence of "a=bundle-only" in
          the type is "offer", offer, no "m=" sections in the PeerConnection state MUST answer should have an
          "a=bundle-only" line.</t>
          <t>Attributes that are common between all "m=" sections <bcp14>MAY</bcp14> be
            either "stable" or "have-remote-offer".</t>

            <t>If the type is "pranswer" or "answer",
          moved to the
            PeerConnection state MUST session level if explicitly defined to be either "have-local-offer" or
            "have-remote-pranswer".</t>
          </list></t>

          <t>If the type is not correct for valid at
          the current state,
          processing MUST stop session level.

<!-- [rfced] Section 5.3.1: Per "the session level" used elsewhere in this
document and an error MUST be returned.</t>

          <t>Next, the SessionDescription is parsed into a data
          structure, as described "ice-options are now at session level" in
          <xref target="sec.parsing-a-desc" /> below. If parsing fails
          for any reason, processing MUST stop the Change Log, we
changed "to session-level" and an error MUST be
          returned.</t>

          <t>Finally, "at session-level" to "to the parsed SessionDescription is applied as
          described in
          <xref target="sec.applying-a-remote-desc" /> below.</t>
        </list></t>
      </section>
      <section title="Processing a Rollback"
      anchor="sec.processing-a-rollback">

        <t>A rollback may be performed if session level"
and "at the PeerConnection is in session level."  Please let us know any
        state except for "stable". This means objections.

Original:
 Attributes that both offers and
        provisional answers can be rolled back. Rollback can only are common between all m= sections MAY be
        used moved to cancel proposed changes; there is no support for
        rolling back from a stable state
 session-level, if explicitly defined 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 valid at session-level.

Currently:
 Attributes that
        once are common between all "m=" sections MAY be moved to
 the answerer has performed setLocalDescription with his
        answer, this cannot session level if explicitly defined to be rolled back.</t> valid at the session
 level. -->

</t>
          <t>The effect attributes prohibited in the creation of rollback MUST be offers are
          also prohibited in the same regardless creation of
        whether setLocalDescription answers.</t>
        </section>
        <section anchor="sec.subsequent-answers" numbered="true" toc="default">
          <name>Subsequent Answers</name>
          <t>When createAnswer is called a second (or later) time or setRemoteDescription
          is
        called.</t>

        <t>In order to process rollback, called after 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
        <xref target="sec.pendinglocaldescription" /> and
        <xref target="sec.pendingremotedescription" />) to null. Any
        resources or candidates that were allocated by has already been
          installed, the abandoned
        local description are discarded; any media that is received processing is
        processed according to somewhat different than for an
          initial answer.</t>
          <t>If the previous local and remote
        descriptions.</t>

        <t>A rollback disassociates any RtpTransceivers that were
        associated with m= sections by answer was not applied using
          setLocalDescription, meaning the application PeerConnection is still in
          the "have-remote-offer" state, the steps for generating an
          initial answer should be followed, subject to the following
          restriction:
          </t>
          <ul spacing="normal">
            <li>The fields of the
        rolled-back "o=" line <bcp14>MUST</bcp14> stay the same except
            for the &lt;session-version&gt; field, which <bcp14>MUST</bcp14> increment
            if the session description (see
        <xref target="sec.applying-a-remote-desc" /> and
        <xref target="sec.applying-a-local-desc" />). This means that
        some RtpTransceivers that were changes in any way from the
            previously associated will no
        longer be associated with generated answer.</li>
          </ul>
          <t>If any m= section; session description was previously supplied to
          setLocalDescription, an answer is generated by following the
          steps in such cases, the
        value of the RtpTransceiver's mid property MUST be set to null, "have-remote-offer" state above, along with
          these exceptions:
          </t>
          <ul spacing="normal">
            <li>The "s=" and "t=" lines <bcp14>MUST</bcp14> stay the mapping between the transceiver same.</li>
            <li>Each "m=" and its m= section
        index MUST be discarded. RtpTransceivers that were created by
        applying a remote offer that was subsequently rolled back MUST "c=" line <bcp14>MUST</bcp14> be stopped and removed from filled in with the PeerConnection. However, a
        RtpTransceiver MUST NOT be removed if a track was attached to port
            and address of the RtpTransceiver via default candidate for the addTrack method. This is so "m=" section, as
            described in
            <xref target="RFC8839" sectionFormat="comma" section="4.2.1.2"/>. Note that an
        application in certain cases, the "m=" line protocol
            may call addTrack, then call setRemoteDescription
        with an offer, then roll back not match that of the default candidate, because the "m=" line
            protocol value <bcp14>MUST</bcp14> match what was supplied in the offer, then call createOffer as
            described above.</li>
            <li>Each "a=ice-ufrag" and have a m= section for "a=ice-pwd" line <bcp14>MUST</bcp14> stay the added track appear in
            same, unless the
        generated offer.</t>
      </section>
      <section title="Parsing a Session Description"
      anchor="sec.parsing-a-desc">

        <t>The SDP contained "m=" section is restarting, in which case
            new ICE credentials must be created as specified in
            <xref target="RFC8839" sectionFormat="comma" section="4.4.1.1.1"/>. If the session description object consists
        of a sequence of text lines, each containing "m="
            section is bundled into another "m=" section, it still <bcp14>MUST
            NOT</bcp14> contain any ICE credentials.</li>
            <li>Each "a=tls-id" line <bcp14>MUST</bcp14> stay the same, unless the
            offerer's "a=tls-id" line changed, in which case a key-value
        expression, new
            "a=tls-id" value <bcp14>MUST</bcp14> be created, as described in
            <xref target="RFC4566" />, Section 5. The SDP target="RFC8842" sectionFormat="comma" section="5.2"/>.</li>
            <li>Each "a=setup" line <bcp14>MUST</bcp14> use an "active" or "passive"
            role value consistent with the existing DTLS association,
            if the association is read,
        line-by-line, and converted to a data structure that contains being continued by the deserialized information. However, SDP allows many types of
        lines, not all of which are relevant to JSEP applications. For
        each line, the implementation will first ensure it is
        syntactically correct according to its defining ABNF, check
        that it conforms to
        <xref target="RFC4566" /> and
        <xref target="RFC3264" /> semantics, and then either parse and
        store or discard the provided value, as described below.</t>

        <t>If any line is not well-formed, or cannot offerer.</li>
            <li>RTCP multiplexing must be parsed as
        described, the parser MUST stop with used, and an error "a=rtcp-mux" line
            inserted if and reject the
        session description, even only if the value "m=" section previously used RTCP
            multiplexing.</li>
            <li>If the "m=" section is to be discarded. This
        ensures that implementations do not accidentally misinterpret
        ambiguous SDP.</t>
        <section title="Session-Level Parsing"
        anchor="sec.session-level-parse">

          <t>First, the session-level lines are checked and parsed.
          These lines MUST occur in a specific order, bundled into another "m=" section
            and with a
          specific syntax, as defined in
          <xref target="RFC4566" />, Section 5. Note that while the
          specific RTCP multiplexing is not active, an "a=rtcp" attribute
            line types (e.g. "v=", "c=") MUST occur <bcp14>MUST</bcp14> be filled in with the
          defined order, lines port and address of the same type (typically "a=") can
          occur in any order.</t>

          <t>The following non-attribute lines are not meaningful in
          the JSEP context and MAY be discarded once they
            default RTCP candidate. If no RTCP candidates have yet been
          checked.
          <list>

            <t>The "c=" line MUST
            gathered, dummy values <bcp14>MUST</bcp14> be checked for syntax but its value
            is only used for ICE mismatch detection, used, as defined described in
            <xref target="RFC8445" />, Section 5.4. Note that JSEP
            implementations should never encounter this condition
            because ICE target="sec.initial-answers"/> above.</li>

            <li>If the "m=" section is required not bundled into another "m="
            section, for WebRTC.</t>

            <t>The "i=", "u=", "e=", "p=", "t=", "r=", "z=", and "k="
            lines are not used by this specification; they MUST each candidate that has been gathered during
            the most recent gathering phase (see
            <xref target="sec.ice-gather-overview" format="default"/>), an
            "a=candidate" line <bcp14>MUST</bcp14> be
            checked for syntax but their values are not used.</t>
          </list></t>

          <t>The remaining non-attribute lines are processed added, as
          follows:
          <list>

            <t>The "v=" line MUST have a version of 0, defined in
            <xref target="RFC8839" sectionFormat="comma" section="5.1"/>.
            If candidate gathering for the section has completed, an
            "a=end-of-candidates" attribute <bcp14>MUST</bcp14> be added, as specified described
            in
            <xref target="RFC4566" />, target="RFC8840" sectionFormat="comma" section="8.2"/>.
            If the "m=" section is bundled into another "m=" section, both
            "a=candidate" and "a=end-of-candidates" <bcp14>MUST</bcp14> be
            omitted.

<!-- [rfced] Section 5.1.</t>

            <t>The "o=" 5.3.2:  We found this RFC Editor Note on
<https://datatracker.ietf.org/doc/draft-ietf-rtcweb-jsep/writeup/>:

"OLD:
 o  If the m= section is not bundled into another m= section, for each
    candidate that has been gathered during the most recent gathering
    phase (see Section 3.5.1), an "a=candidate" line MUST be parsed added, as specified
    defined in
            <xref target="RFC4566" />, [RFC5245], Section 5.2.</t>

            <t>The "b=" line, if present, 4.3., paragraph 3.  If candidate
    gathering for the section has completed, an "a=end-of-candidates"
    attribute MUST be parsed added, as specified described in
            <xref target="RFC4566" />, [I-D.ietf-ice-trickle],
    Section 5.8, and 9.3.  If the bwtype m= section is bundled into another m=
    section, both "a=candidate" and
            bandwidth values stored.</t>
          </list></t>

          <t>Finally, "a=end-of-candidates" MUST be
    omitted.

NEW:
 o  If the attribute lines are processed. Specific
          processing 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 applied added, as
    defined in [RFC5245], Section 4.3., paragraph 3.  If candidate
    gathering for the following session-level section has completed, an "a=end-of-candidates"
    attribute ("a=") lines:
          <list style="symbols">

            <t>Any "a=group" lines are parsed MUST be added, as specified described in
            <xref target="RFC5888" />,
    [I-D.ietf-mmusic-trickle-ice-sip], Section 5, and 8.2.  If the group's
            semantics and mids are stored.</t>

            <t>If present, a single "a=ice-lite" line m= section is parsed as
            specified in
            <xref target="I-D.ietf-mmusic-ice-sip-sdp" />,
            Section 4.3,
    bundled into another m= section, both "a=candidate" and a value
            indicating
    "a=end-of-candidates" MUST be omitted."

Please note that the presence of ice-lite is stored.</t>

            <t>If present, a single "a=ice-ufrag" line is parsed as
            specified "OLD" text does not match what we found in
            <xref target="I-D.ietf-mmusic-ice-sip-sdp" />, the
provided draft (i.e., "[RFC5245], Section 4.4, and 4.3., paragraph 3" versus
"[I-D.ietf-mmusic-ice-sip-sdp], Section 4.1"):

 o  If the ufrag value m= section is stored.</t>

            <t>If present, a single "a=ice-pwd" 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 is parsed MUST be added, as
            specified
    defined in
            <xref target="I-D.ietf-mmusic-ice-sip-sdp" />, [I-D.ietf-mmusic-ice-sip-sdp], Section 4.4, and 4.1.  If
    candidate gathering for the password value is stored.</t>

            <t>If present, a single "a=ice-options" line is parsed section has completed, an "a=end-of-
    candidates" attribute MUST be added, as
            specified described in
            <xref target="I-D.ietf-mmusic-ice-sip-sdp" />,
    [I-D.ietf-ice-trickle], Section 4.6, and 9.3.  If the set of specified options m= section is stored.</t>

            <t>Any "a=fingerprint" lines bundled
    into another m= section, both "a=candidate" and "a=end-of-
    candidates" MUST be omitted.

Please review, and let us know if further changes are parsed as specified in
            <xref target="RFC8122" />, Section 5, needed.  (As
noted previously, "[RFC8839]" and "[RFC8840]" are the set of
            fingerprint RFC numbers assigned
for [I-D.ietf-mmusic-ice-sip-sdp] and algorithm values
[I-D.ietf-mmusic-trickle-ice-sip], respectively.)

Currently:
 *  If the "m=" section is stored.</t>

            <t>If present, a single "a=setup" 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 is parsed MUST be added, as
            specified
    defined in
            <xref target="RFC4145" />, [RFC8839], Section 4, and 5.1.  If candidate gathering for the setup value
            is stored.</t>

            <t>If present, a single "a=tls-id" line is parsed
    section has completed, an "a=end-of-candidates" attribute MUST be
    added, as
            specified described in
            <xref target="I-D.ietf-mmusic-dtls-sdp" /> [RFC8840], Section 5, and 8.2.  If the tls-id value "m=" section
    is stored.</t>

            <t>Any "a=identity" lines are parsed and the identity
            values stored for subsequent verification, as specified
            <xref target="I-D.ietf-rtcweb-security-arch" />, Section
            5.</t>

            <t>Any "a=extmap" lines are parsed as specified in
            <xref target="RFC5285" />, Section 5, bundled into another "m=" section, both "a=candidate" and their values are
            stored.</t>
          </list></t>

          <t>Other attributes "a=end-
    of-candidates" MUST be omitted. -->

</li>
            <li>For RtpTransceivers that are not relevant to JSEP may also be
          present, and implementations SHOULD process any that they
          recognize. As required by
          <xref target="RFC4566"></xref>, Section 5.13, unknown
          attribute lines MUST be ignored.</t>

          <t>Once all the session-level lines have been parsed,
          processing continues with stopped, the lines in m= sections.</t>
        </section>
        <section title="Media Section Parsing"
        anchor="sec.media-level-parse">

          <t>Like "a=msid"
            line(s) <bcp14>MUST</bcp14> stay the session-level lines, same, regardless of changes to the media section lines MUST
          occur
            transceiver's direction or track. If no "a=msid" line is
            present in the specific order and with current description, "a=msid" line(s) <bcp14>MUST</bcp14>
            be generated according to the specific syntax
          defined in
          <xref target="RFC4566" />, Section 5.</t> same rules as for an initial
            answer.</li>
          </ul>
        </section>
        <section anchor="sec.options-handling2" numbered="true" toc="default">
          <name>Options Handling</name>
          <t>The "m=" line itself MUST be parsed createAnswer method takes as described a parameter an
          RTCAnswerOptions object. The set of parameters for
          RTCAnswerOptions is different than those supported in
          <xref target="RFC4566" />, Section 5.14, and the media, port,
          proto, and fmt values stored.</t>

          <t>Following
          RTCOfferOptions; the "m=" line, specific processing MUST IceRestart option is unnecessary, as ICE
          credentials will automatically be
          applied changed for all "m=" sections
          where the offerer chose to perform ICE restart.</t>
          <t>The following non-attribute lines:
          <list style="symbols">

            <t>As with options are supported in
          RTCAnswerOptions.</t>
          <section anchor="sec.voiceactivitydetection2" numbered="true" toc="default">
            <name>VoiceActivityDetection</name>
            <t>Silence suppression in the "c=" line at answer is handled as
            described in
            <xref target="sec.voiceactivitydetection1" format="default"/>, with
            one exception: if support for silence suppression was not
            indicated in the session level, offer, the "c="
            line MUST VoiceActivityDetection
            parameter has no effect, and the answer should be parsed according generated
            as if VoiceActivityDetection was set to
            <xref target="RFC4566" />, Section 5.7, but its value "false". This is
            not used.</t>

            <t>The "b=" line, done
            on a per-codec basis (e.g., if present, MUST be parsed as specified
            in
            <xref target="RFC4566" />, Section 5.8, and the bwtype and
            bandwidth values stored.</t>
          </list></t>

          <t>Specific processing MUST also be applied offerer somehow offered
            support for the following
          attribute lines:
          <list style="symbols">

            <t>If present, a single "a=ice-ufrag" line is parsed as
            specified CN but set "usedtx=0" for Opus, setting
            VoiceActivityDetection to "true" would result in
            <xref target="I-D.ietf-mmusic-ice-sip-sdp" />,
            Section 4.4, an answer
            with CN codecs and the ufrag value is stored.</t>

            <t>If present, a single "a=ice-pwd" line "usedtx=0"). The impact of this rule is parsed as
            specified in
            <xref target="I-D.ietf-mmusic-ice-sip-sdp" />,
            Section 4.4, and
            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.</t>
          </section>
        </section>
      </section>
      <section anchor="sec.modifying-sdp" numbered="true" toc="default">
        <name>Modifying an Offer or Answer</name>
        <t>The SDP returned from createOffer or createAnswer <bcp14>MUST NOT</bcp14>
        be changed before passing it to setLocalDescription. If precise
        control over the password value is stored.</t>

            <t>If present, a single "a=ice-options" line SDP is parsed as
            specified in
            <xref target="I-D.ietf-mmusic-ice-sip-sdp" />, Section 4.6,
            and needed, the set of specified aforementioned
        createOffer/createAnswer options is stored.</t>

            <t>Any "a=candidate" attributes MUST or RtpTransceiver APIs <bcp14>MUST</bcp14> be parsed
        used.</t>
        <t>After calling setLocalDescription with an offer or answer,
        the application <bcp14>MAY</bcp14> modify the SDP to reduce its capabilities
        before sending it to the far side, as specified
            in
            <xref target="I-D.ietf-mmusic-ice-sip-sdp" />,
            Section 4.1, and their values stored.</t>

            <t>Any "a=remote-candidates" attributes MUST be parsed long as
            specified in
            <xref target="I-D.ietf-mmusic-ice-sip-sdp" />,
            Section 4.2, but their values are ignored.</t>

            <t>If present, it follows the
        rules above that define a single "a=end-of-candidates" attribute
            MUST be parsed as specified in
            <xref target="I-D.ietf-ice-trickle" />, Section 8.2, and
            its presence valid JSEP offer or absence flagged and stored.</t>

            <t>Any "a=fingerprint" lines are parsed as specified in
            <xref target="RFC8122" />, Section 5, and answer. Likewise,
        an application that has received an offer or answer from a peer
        <bcp14>MAY</bcp14> modify the set of
            fingerprint and algorithm values received SDP, subject to the same constraints,
        before calling setRemoteDescription.</t>
        <t>As always, the application is stored.</t>
          </list></t>

          <t>If solely responsible for what it
        sends to the "m=" proto value indicates use of RTP, as described
          in
          <xref target="sec.profile-names" /> above, other party, and all incoming SDP will be
        processed by the following
          attribute lines MUST JSEP implementation to the extent of its
        capabilities. It is an error to assume that all SDP is
        well formed; however, one should be processed:
          <list style="symbols">

            <t>The "m=" fmt value MUST able to assume that any
        implementation of this specification will be parsed able to process,
        as specified in
            <xref target="RFC4566" />, Section 5.14, and the individual
            values stored.</t>

            <t>Any "a=rtpmap" a remote offer or "a=fmtp" lines MUST be parsed as
            specified in
            <xref target="RFC4566" />, Section 6, and their values
            stored.</t>

            <t>If present, answer, unmodified SDP coming from any
        other implementation of this specification.</t>
      </section>
      <section anchor="sec.processing-a-local-desc" numbered="true" toc="default">
        <name>Processing a single "a=ptime" line MUST Local Description</name>
        <t>When a SessionDescription is supplied to
        setLocalDescription, the following steps <bcp14>MUST</bcp14> be parsed as
            described performed:
        </t>
        <ul spacing="normal">
          <li>If the description is of type "rollback", follow the
          processing defined in
          <xref target="RFC4566" />, Section 6, target="sec.processing-a-rollback" format="default"/> and its value
            stored.</t>

            <t>If present, a single "a=maxptime" line MUST be parsed as skip the
          processing described in
            <xref target="RFC4566" />, Section 6, and its value
            stored.</t>

            <t>If present, a single direction attribute line (e.g.
            "a=sendrecv") MUST the rest of this section.</li>
          <li>
            <t>Otherwise, the type of the SessionDescription is checked
          against the current state of the PeerConnection:
            </t>
            <ul spacing="normal">
              <li>If the type is "offer", the PeerConnection state <bcp14>MUST</bcp14> be parsed as described in
            <xref target="RFC4566" />, Section 6, and its value
            stored.</t>

            <t>Any "a=ssrc" attributes MUST
            either "stable" or "have-local-offer".</li>
              <li>If the type is "pranswer" or "answer", the
            PeerConnection state <bcp14>MUST</bcp14> be parsed as specified in
            <xref target="RFC5576" />, Section 4.1, either "have-remote-offer" or
            "have-local-pranswer".</li>
            </ul>
          </li>
          <li>If the type is not correct for the current state,
          processing <bcp14>MUST</bcp14> stop and their values
            stored.</t>

            <t>Any "a=extmap" attributes MUST an error <bcp14>MUST</bcp14> be parsed as specified returned.</li>
          <li>The SessionDescription is then checked to ensure that its
          contents are identical to those generated in

            <xref target="RFC5285" />, Section 5, the last call to
          createOffer/createAnswer, and their values
            stored.</t>

            <t>Any "a=rtcp-fb" attributes MUST be parsed thus have not been altered, as specified
          discussed in
          <xref target="RFC4585" />, Section 4.2., target="sec.modifying-sdp" format="default"/>; otherwise, processing
          <bcp14>MUST</bcp14> stop and their values
            stored.</t>

            <t>If present, a single "a=rtcp-mux" attribute MUST an error <bcp14>MUST</bcp14> be returned.</li>
          <li>Next, the SessionDescription is parsed into a data
          structure, as specified described in
          <xref target="RFC5761"></xref>, Section 5.1.3, and its
            presence or absence flagged and stored.</t>

            <t>If present, a single "a=rtcp-mux-only" attribute MUST be target="sec.parsing-a-desc" format="default"/> below.</li>
          <li>Finally, the parsed SessionDescription is applied as specified
          described in
          <xref target="I-D.ietf-mmusic-mux-exclusive" />, Section 3,
            and its presence or absence flagged and stored.</t>

            <t>If present, target="sec.applying-a-local-desc" format="default"/> below.</li>
        </ul>
      </section>
      <section anchor="sec.processing-a-remote-desc" numbered="true" toc="default">
        <name>Processing a single "a=rtcp-rsize" attribute MUST Remote Description</name>
        <t>When a SessionDescription is supplied to
        setRemoteDescription, the following steps <bcp14>MUST</bcp14> be
            parsed as specified performed:
        </t>
        <ul spacing="normal">
          <li>If the description is of type "rollback", follow the
          processing defined in
          <xref target="RFC5506" />, Section 5, and its presence or
            absence flagged target="sec.processing-a-rollback" format="default"/> and stored.</t>

            <t>If present, a single "a=rtcp" attribute MUST be parsed
            as specified skip the
          processing described in
            <xref target="RFC3605" />, Section 2.1, but its value is
            ignored, as the rest of this information section.</li>
          <li>
            <t>Otherwise, the type of the SessionDescription is superfluous when using
            ICE.</t>

            <t>If present, "a=msid" attributes MUST checked
          against the current state of the PeerConnection:
            </t>
            <ul spacing="normal">
              <li>If the type is "offer", the PeerConnection state <bcp14>MUST</bcp14> be parsed as
            specified in
            <xref target="I-D.ietf-mmusic-msid" />, Section 3.2, and
            their values stored, ignoring any "appdata" field. If no "a=msid"
            attributes are present, a random msid-id value
            either "stable" or "have-remote-offer".</li>
              <li>If the type is generated for a
            "default" MediaStream for "pranswer" or "answer", the session, if
            PeerConnection state <bcp14>MUST</bcp14> be either "have-local-offer" or
            "have-remote-pranswer".</li>
            </ul>
          </li>
          <li>If the type is not already present, correct for the current state,
          processing <bcp14>MUST</bcp14> stop and
            this value is stored.</t>

            <t>Any "a=imageattr" attributes MUST an error <bcp14>MUST</bcp14> be returned.</li>
          <li>Next, the SessionDescription is parsed into a data
          structure, as specified described in
          <xref target="RFC6236" />, Section 3, target="sec.parsing-a-desc" format="default"/> below. If parsing fails
          for any reason, processing <bcp14>MUST</bcp14> stop and their values
            stored.</t>

            <t>Any "a=rid" lines MUST an error <bcp14>MUST</bcp14> be
          returned.</li>
          <li>Finally, the parsed SessionDescription is applied as specified
          described in
          <xref target="I-D.ietf-mmusic-rid"></xref>, Section 10, and
            their values stored.</t>

            <t>If present, target="sec.applying-a-remote-desc" format="default"/> below.</li>
        </ul>
      </section>
      <section anchor="sec.processing-a-rollback" numbered="true" toc="default">
        <name>Processing a single "a=simulcast" line MUST Rollback</name>
        <t>A rollback may be parsed
            as specified in
            <xref target="I-D.ietf-mmusic-sdp-simulcast"></xref>, and
            its values stored.</t>
          </list></t>

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

            <t>The "m=" fmt value MUST be parsed as specified PeerConnection is in
            <xref target="I-D.ietf-mmusic-sctp-sdp" />, Section 4.3, any
        state except for "stable". This means that both offers and the application protocol value stored.</t>

            <t>An "a=sctp-port" attribute MUST
        provisional answers can be present, and it MUST rolled back. Rollback can only be parsed as specified in
            <xref target="I-D.ietf-mmusic-sctp-sdp" />, Section 5.2,
            and the value stored.</t>

            <t>If present, a single "a=max-message-size" attribute MUST
            be parsed as specified in
            <xref target="I-D.ietf-mmusic-sctp-sdp" />, Section 6, and
            the value stored. Otherwise, use the specified default.</t>
          </list></t>

          <t>Other attributes that are not relevant
        used to JSEP may also be
          present, and implementations SHOULD process any that they
          recognize. As required by
          <xref target="RFC4566"></xref>, Section 5.13, unknown
          attribute lines MUST be ignored.</t>
        </section>
        <section title="Semantics Verification">

          <t>Assuming parsing completes successfully, the parsed
          description cancel proposed changes; there is then evaluated to ensure internal consistency
          as well as proper no support for mandatory features.
          Specifically,
        rolling back from a stable state to a previous stable state. If
        a rollback is attempted in the following checks are performed:
          <list style="symbols">

            <t>For each m= section, valid values for each of "stable" state, processing <bcp14>MUST</bcp14>
        stop and an error <bcp14>MUST</bcp14> be returned. Note that this implies that
        once the
            mandatory-to-use features enumerated in
            <xref target="sec.usage-requirements" /> MUST answerer has performed setLocalDescription with its
        answer, this cannot be present.
            These values MAY either rolled back.</t>
        <t>The effect of rollback <bcp14>MUST</bcp14> be present at the media level, same regardless of
        whether setLocalDescription or
            inherited from the session level.
            <list style="symbols">

              <t>ICE ufrag and password values, which MUST comply with
              the size limits specified in
              <xref target="I-D.ietf-mmusic-ice-sip-sdp" />, Section 4.4.</t>

              <t>tls-id value, which MUST be set according to
              <xref target="I-D.ietf-mmusic-dtls-sdp" />, Section 5. If
              this setRemoteDescription is a re-offer or a response
        called.</t>
        <t>In order to process rollback, a re-offer and JSEP implementation abandons
        the
              tls-id value is different from that presently in use, current offer/answer transaction, sets the
              DTLS connection is not being continued signaling state
        to "stable", and sets the pending local and/or remote
        description MUST be part of an ICE restart, together with
              new ufrag (see Sections
        <xref target="sec.pendinglocaldescription" format="counter"/> and password values.</t>

              <t>DTLS setup value, which MUST be set
        <xref target="sec.pendingremotedescription" format="counter"/>) 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
              rules specified in
              <xref target="RFC5763" />, Section 5 previous local and MUST be
              consistent remote
        descriptions.</t>
        <t>A rollback disassociates any RtpTransceivers that were
        associated with "m=" sections by the selected role application of the current DTLS
              connection, if one exists
        rolled-back session description (see Sections
        <xref target="sec.applying-a-remote-desc" format="counter"/> and
        <xref target="sec.applying-a-local-desc" format="counter"/>).

<!-- [rfced] Section 5.7:  Please confirm that Section 5.9 is being continued.</t>

              <t>DTLS fingerprint values, where at least one
              fingerprint MUST be present.</t>
            </list></t>

            <t>All RID values referenced in an "a=simulcast" line MUST
            exist as "a=rid" lines.</t>

            <t>Each m= the
correct section is also checked to ensure prohibited
            features are cite here.  We easily found relevant text in
Section 5.10 but not used.</t>

            <t>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, in Section 5.9.

Original:
 A rollback disassociates any RtpTransceivers that
            section MUST also contain an "a=rtcp-mux" attribute.</t>

            <t>If an were associated
 with m= section was present in the previous answer, sections by the
            state of RTP/RTCP multiplexing MUST match what was
            previously negotiated.</t>
          </list></t>

          <t>If this session description is application of type "pranswer" or
          "answer", the following additional checks are applied:
          <list style="symbols">

            <t>The rolled-back session
 description must follow the rules defined in

            <xref target="RFC3264" />, (see Section 6, including the
            requirement 5.10 and Section 5.9). -->

 This means that
        some RtpTransceivers that were previously associated will no
        longer be associated with any "m=" section; in such cases, the number
        value of m= sections MUST exactly
            match the number of m= sections in RtpTransceiver's mid property <bcp14>MUST</bcp14> be set to "null",
        and the associated
            offer.</t>

            <t>For each m= section, mapping between the media type transceiver and protocol values
            MUST exactly match the media type its "m=" section
        index <bcp14>MUST</bcp14> be discarded. RtpTransceivers that were created by
        applying a remote offer that was subsequently rolled back <bcp14>MUST</bcp14>
        be stopped and protocol values in removed from the corresponding m= section in PeerConnection. However, an
        RtpTransceiver <bcp14>MUST NOT</bcp14> be removed if a track was attached to
        the associated offer.</t>
          </list></t>

          <t>If any of RtpTransceiver via the preceding checks failed, processing MUST
          stop 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 error MUST be returned.</t>
        </section> "m=" section for the added track appear in the
        generated offer.</t>
      </section>
      <section title="Applying anchor="sec.parsing-a-desc" numbered="true" toc="default">
        <name>Parsing a Local Description"
      anchor="sec.applying-a-local-desc"> Session Description</name>
        <t>The following steps are performed at SDP contained in the media engine level
        to apply session description object consists
        of a local description. If an error sequence of text lines, each containing a key-value
        expression, as described in
        <xref target="RFC4566" sectionFormat="comma" section="5"/>. The SDP is returned, the
        session MUST be restored read,
        line by line, and converted to a data structure that contains
        the state it was in before
        performing these steps.</t>

        <t>First, m= sections deserialized information. However, SDP allows many types of
        lines, not all of which are processed. relevant to JSEP applications. For
        each m= section, line, 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.
        <list style="symbols">

          <t>If this m= section implementation will first ensure that it is new, begin gathering candidates for
          it, as defined
        syntactically correct according to its defining ABNF, check
        that it conforms to the semantics used in
        <xref target="RFC8445" />, Section 5.1.1, unless it is
          definitively being bundled (either this is an offer target="RFC4566" format="default"/> and the
          m= section is marked bundle-only, or it is an answer
        <xref target="RFC3264" format="default"/>, and the
          m= section is bundled into into another m= section.)</t>

          <t>Or, if the ICE ufrag then either parse and password values have changed,
          trigger
        store or discard the ICE agent to start an ICE restart provided value, as described in
          <xref target="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.</t>

          <t>If the m= section proto value indicates use of RTP:
          <list style="symbols"> below.</t>
        <t>If there any line is no RtpTransceiver associated with this m=
            section, find one and associate it with this m= section
            according to not well formed or cannot be parsed as
        described, the following steps. Note that this situation
            will only occur when applying parser <bcp14>MUST</bcp14> stop with an offer.
            <list style="symbols">

              <t>Find the RtpTransceiver that corresponds to this m=
              section, using the mapping between transceivers error and m=
              section indices established when creating reject the offer.</t>

              <t>Set
        session description, even if the value of this RtpTransceiver's mid property to
              the MID of the m= section.</t>
            </list></t>

            <t>If RTCP mux is indicated, prepare to demux RTP and RTCP
            from be discarded. This
        ensures that implementations do not accidentally misinterpret
        ambiguous SDP.</t>
        <section anchor="sec.session-level-parse" numbered="true" toc="default">
          <name>Session-Level Parsing</name>
          <t>First, the RTP ICE component, as specified session-level lines are checked and parsed.
          These lines <bcp14>MUST</bcp14> occur in
            <xref target="RFC5761" />, Section 5.1.3.</t>

            <t>For each specified RTP header extension, establish a
            mapping between the extension ID specific order, and URI, with a
          specific syntax, as described defined in
          <xref target="RFC5285" />, Section 6.</t>

            <t>If target="RFC4566" sectionFormat="comma" section="5"/>. Note that while the MID header extension is supported, prepare to
            demux RTP streams intended for this m= section based on the
            MID header extension, as described
          specific line types (e.g., "v=", "c=") <bcp14>MUST</bcp14> occur in
            <xref target="I-D.ietf-mmusic-sdp-bundle-negotiation" />,
            Section 15.</t>

            <t>For each specified media format, establish a mapping
            between the payload
          defined order, lines of the same type and (typically "a=") can
          occur in any order.</t>
          <t>The following non-attribute lines are not meaningful in
          the actual media format, JSEP context and <bcp14>MAY</bcp14> be discarded once they have been
          checked.
          </t>
          <ul spacing="normal">
            <li>The "c=" line <bcp14>MUST</bcp14> be checked for syntax, but its value
            is only used for ICE mismatch detection, as
            described defined in
            <xref target="RFC3264" />, Section 6.1. In addition,
            prepare to demux RTP streams intended target="RFC8445" sectionFormat="comma" section="5.4"/>. Note that JSEP
            implementations should never encounter this condition
            because ICE is required for WebRTC.</li>
            <li>The "i=", "u=", "e=", "p=", "t=", "r=", "z=", and "k="
            lines are not used by this specification; they <bcp14>MUST</bcp14> be
            checked for syntax, but their values are not used.

<!-- [rfced] Section 5.8.1:  We found this m= section
            based on the media formats supported sentence confusing; should
"t=" remain in this list of lines not used by this m= section, specification?

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

We ask because we see (for example):

o  A "t=" line MUST be added, as
            described specified in
            <xref target="I-D.ietf-mmusic-sdp-bundle-negotiation" />, [RFC4566], Section 10.2.</t>

            <t>For each specified "rtx" media format, establish a
            mapping between the RTX payload type 5.9;
    both <start-time> and its associated
            primary payload type, as described in
            <xref target="RFC4588" />, Sections 8.6 <stop-time> SHOULD be set to zero, e.g. "t=0
    0".
...
 o  The "s=" and 8.7.</t>

            <t>If "t=" lines MUST stay the directional attribute is of type "sendrecv" or
            "recvonly", enable receipt and decoding of media.</t>
          </list></t>
        </list></t>

        <t>Finally, if this description is of type "pranswer" or
        "answer", follow the processing defined in
        <xref target="sec.applying-an-answer" /> below.</t>
      </section>
      <section title="Applying a Remote Description"
      anchor="sec.applying-a-remote-desc"> same.
...
 t=0 0 -->

</li>
          </ul>
          <t>The following steps remaining non-attribute lines are performed to apply processed as
          follows:
          </t>
          <ul spacing="normal">
            <li>The "v=" line <bcp14>MUST</bcp14> have a remote
        description. If an error is returned, the session MUST version of 0, as specified in
            <xref target="RFC4566" sectionFormat="comma" section="5.1"/>.</li>
            <li>The "o=" line <bcp14>MUST</bcp14> be
        restored to the state it was parsed as specified in before performing these
        steps.</t>

        <t>If the answer contains any "a=ice-options" attributes where
        "trickle" is listed
            <xref target="RFC4566" sectionFormat="comma" section="5.2"/>.</li>
            <li>The "b=" line, if present, <bcp14>MUST</bcp14> be parsed as an attribute, update specified
            in
            <xref target="RFC4566" sectionFormat="comma" section="5.8"/>, and the PeerConnection
        canTrickle property to be true. Otherwise, set this property to
        false.</t>

        <t>The following steps MUST be performed for attributes at bwtype and
            bandwidth values stored.</li>
          </ul>
          <t>Finally, the
        session level; if any parameters attribute lines are out of bounds, or cannot
        be applied, processed. Specific
          processing MUST stop and an error MUST <bcp14>MUST</bcp14> be returned.

        <list style="symbols">

          <t>For any specified "CT" bandwidth value, set this as the
          limit applied for the maximum total bitrate for all m= sections, following session-level
          attribute ("a=") lines:
          </t>
          <ul spacing="normal">
            <li>Any "a=group" lines are parsed as specified in
            <xref target="RFC4566"></xref>, Section 5.8. Within this
          overall limit, the implementation can dynamically decide how
          to best allocate target="RFC5888" sectionFormat="comma" section="5"/>, and the available bandwidth between m= sections,
          respecting any specific limits that have been specified for
          individual m= sections.</t>

          <t>For any specified "RR" or "RS" bandwidth values, handle group's
            semantics and mids are stored.</li>
            <li>If present, a single "a=ice-lite" line is parsed as
            specified in
            <xref target="RFC3556"></xref>, Section 2.</t>

          <t>Any "AS" bandwidth target="RFC8839" sectionFormat="comma" section="5.3"/>, and a value MUST be ignored, as
            indicating the meaning presence of this construct at the session level ice-lite is not well
          defined.</t>
        </list></t>

        <t>For each m= section, the following steps MUST be performed;
        if any parameters are out of bounds, or cannot be applied,
        processing MUST stop stored.</li>
            <li>If present, a single "a=ice-ufrag" line is parsed as
            specified in
            <xref target="RFC8839" sectionFormat="comma" section="5.4"/>, and an error MUST be returned.
        <list style="symbols">

          <t>If the ICE ufrag or password changed from the previous
          remote description:
          <list style="symbols">

            <t>If the description value is of type "offer", the
            implementation MUST note that an ICE restart stored.</li>
            <li>If present, a single "a=ice-pwd" line is needed, parsed as
            described
            specified in
            <xref target="I-D.ietf-mmusic-ice-sip-sdp" />,
            Section 3.4.1.1.1</t>

            <t>If target="RFC8839" sectionFormat="comma" section="5.4"/>, and the description password value is of type "answer" or "pranswer",
            then check to see if the current local description stored.</li>
            <li>If present, a single "a=ice-options" line is an
            ICE restart, parsed as
            specified in
            <xref target="RFC8839" sectionFormat="comma" section="5.6"/>, and if not, generate an error. If the
            PeerConnection state set of specified options 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 check list state for the m= section. Finally, signal
            the ICE agent to begin checks.</t>
          </list></t>

          <t>If the current local description indicates an ICE restart,
          and either the ICE ufrag or password has not changed from the
          previous remote description, stored.</li>
            <li>Any "a=fingerprint" lines are parsed as prescribed by specified in
            <xref target="RFC8445" />, Section 9, generate an
          error.</t>

          <t>Configure the ICE components associated with this media
          section to use target="RFC8122" sectionFormat="comma" section="5"/>, and the supplied ICE remote ufrag set of
            fingerprint and password for
          their connectivity checks.</t>

          <t>Pair any supplied ICE candidates with any gathered local
          candidates, algorithm values is stored.</li>
            <li>If present, a single "a=setup" line is parsed as described
            specified in
            <xref target="RFC8445" />, Section 6.1.2, target="RFC4145" sectionFormat="comma" section="4"/>, and start
          connectivity checks with the appropriate credentials.</t>

          <t>If an "a=end-of-candidates" attribute setup value
            is stored.</li>
            <li>If present, process
          the end-of-candidates indication a single "a=tls-id" line is parsed as described
            specified in <xref target="I-D.ietf-ice-trickle" />, Section 11.</t>

          <t>If target="RFC8842" sectionFormat="comma" section="5"/>, and
            the m= section proto tls-id value indicates use of RTP:
          <list style="symbols">

            <t>If the m= section is being recycled (see
            <xref target="sec.subsequent-offers"></xref>), dissociate
            the currently associated RtpTransceiver by setting its mid
            property to null, stored.</li>
            <li>Any "a=identity" lines are parsed and discard the mapping between the
            transceiver identity
            values stored for subsequent verification, as specified in
            <xref target="RFC8827" sectionFormat="comma" section="5"/>.</li>
            <li>Any "a=extmap" lines are parsed as specified in
            <xref target="RFC5285" sectionFormat="comma" section="5"/>, and its m= section index.</t>

            <t>If the m= section is their values are
            stored.</li>
          </ul>
          <t>Other attributes that are not associated with relevant to JSEP may also be
          present, and implementations <bcp14>SHOULD</bcp14> process any
            RtpTransceiver (possibly because it was dissociated in that they
          recognize. As required by
          <xref target="RFC4566" sectionFormat="comma" section="5.13"/>, unknown
          attribute lines <bcp14>MUST</bcp14> be ignored.</t>
          <t>Once all the
            previous step), either find an RtpTransceiver or create one
            according to session-level lines have been parsed,
          processing continues with the following steps:
            <list style="symbols">

              <t>If lines in "m=" sections.</t>
        </section>
        <section anchor="sec.media-level-parse" numbered="true" toc="default">
          <name>Media Section Parsing</name>
          <t>Like the m= section is sendrecv or recvonly, and there
              are RtpTransceivers of session-level lines, the same type that were added to media section lines <bcp14>MUST</bcp14>
          occur in the PeerConnection by addTrack specific order and are not associated with any m= section and are not stopped, find the first
              (according to the canonical order described specific syntax
          defined in
          <xref target="sec.initial-offers" />) such
              RtpTransceiver.</t>

              <t>If no RtpTransceiver was found target="RFC4566" sectionFormat="comma" section="5"/>.</t>
          <t>The "m=" line itself <bcp14>MUST</bcp14> be parsed as described in
          <xref target="RFC4566" sectionFormat="comma" section="5.14"/>, and the previous step,
              create one with a recvonly direction.</t>

              <t>Associate media, port,
          proto, and fmt values stored.</t>
          <t>Following the found or created RtpTransceiver "m=" line, specific processing <bcp14>MUST</bcp14> be
          applied for the following non-attribute lines:
          </t>
          <ul spacing="normal">
            <li>As with the
              m= section by setting "c=" line at the value of session level, the RtpTransceiver's
              mid property "c="
            line <bcp14>MUST</bcp14> be parsed according to the MID of the m= section,
            <xref target="RFC4566" sectionFormat="comma" section="5.7"/>, but its value is
            not used.</li>
            <li>The "b=" line, if present, <bcp14>MUST</bcp14> be parsed as specified
            in
            <xref target="RFC4566" sectionFormat="comma" section="5.8"/>, and establish
              a mapping between the transceiver bwtype 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
            bandwidth values stored.</li>
          </ul>
          <t>Specific processing <bcp14>MUST</bcp14> also be applied for the RtpTransceiver mid property, following the guidance for "a=mid" mentioned
          attribute lines:
          </t>
          <ul spacing="normal">
            <li>If present, a single "a=ice-ufrag" line is parsed as
            specified in
            <xref target="sec.initial-offers" />.</t>
            </list></t>

            <t>For each specified media format that is also supported
            by target="RFC8839" sectionFormat="comma" section="5.4"/>, and the local implementation, establish ufrag value is stored.</li>
            <li>If present, a mapping between
            the single "a=ice-pwd" line is parsed as
            specified payload type in
            <xref target="RFC8839" sectionFormat="comma" section="5.4"/>, and the media format, password value is stored.</li>
            <li>If present, a single "a=ice-options" line is parsed as
            described
            specified in
            <xref target="RFC3264" />, Section 6.1. Specifically, this
            means that the implementation records target="RFC8839" sectionFormat="comma" section="5.6"/>,
            and the payload type to
            be used in outgoing RTP packets when sending each set of specified
            media format, as well as the relative preference for each
            format that options is indicated stored.</li>
            <li>Any "a=candidate" attributes <bcp14>MUST</bcp14> be parsed as specified
            in
            <xref target="RFC8839" sectionFormat="comma" section="5.1"/>, and their ordering. If any
            indicated media format is not supported by the local
            implementation, it MUST values stored.</li>
            <li>Any "a=remote-candidates" attributes <bcp14>MUST</bcp14> be ignored.</t>

            <t>For each parsed as
            specified "rtx" media format, establish in
            <xref target="RFC8839" sectionFormat="comma" section="5.2"/>, but their values are ignored.</li>

            <li>If present, a
            mapping between the RTX payload type single "a=end-of-candidates" attribute
            <bcp14>MUST</bcp14> be parsed as specified in
            <xref target="RFC8840" sectionFormat="comma" section="8.1"/>, and
            its associated
            primary payload type, presence or absence flagged and stored.</li>
            <li>Any "a=fingerprint" lines are parsed as described specified in
            <xref target="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 which are not supported by target="RFC8122" sectionFormat="comma" section="5"/>, and the local media
            implementation, set of
            fingerprint and algorithm values is stored.</li>
          </ul>
          <t>If the "m=" proto value indicates use of RTP, as described
          in which case,
          <xref target="sec.profile-names" format="default"/> above, the RTX payload type MUST
            also following
          attribute lines <bcp14>MUST</bcp14> be ignored.</t>

            <t>For each processed:
          </t>
          <ul spacing="normal">
            <li>The "m=" fmt value <bcp14>MUST</bcp14> be parsed as specified fmtp parameter that is supported by
            the local implementation, enable them on in
            <xref target="RFC4566" sectionFormat="comma" section="5.14"/>, and the associated
            media formats.</t>

            <t>For each individual
            values stored.</li>
            <li>Any "a=rtpmap" or "a=fmtp" lines <bcp14>MUST</bcp14> be parsed as
            specified SSRC that is signaled in the m=
            section, prepare to demux RTP streams intended for this m=
            section using that SSRC,
            <xref target="RFC4566" sectionFormat="comma" section="6"/>, and their values
            stored.</li>
            <li>If present, a single "a=ptime" line <bcp14>MUST</bcp14> be parsed as
            described in
            <xref target="I-D.ietf-mmusic-sdp-bundle-negotiation" />,
            Section 10.2.</t>

            <t>For each specified RTP header extension that is also
            supported by the local implementation, establish a mapping
            between the extension ID target="RFC4566" sectionFormat="comma" section="6"/>, and URI, its value
            stored.</li>
            <li>If present, a single "a=maxptime" line <bcp14>MUST</bcp14> be parsed as
            described in
            <xref target="RFC5285" />, Section 5. Specifically, this
            means that the implementation records the extension ID to target="RFC4566" sectionFormat="comma" section="6"/>, and its value
            stored.</li>
            <li>If present, a single direction attribute line (e.g.,
            "a=sendrecv") <bcp14>MUST</bcp14> be used parsed as described in outgoing RTP packets when sending each
            <xref target="RFC4566" sectionFormat="comma" section="6"/>, and its value
            stored.</li>
            <li>Any "a=ssrc" attributes <bcp14>MUST</bcp14> be parsed as specified
            header extension. If any indicated RTP header extension is
            not supported by the local implementation, it MUST in
            <xref target="RFC5576" sectionFormat="comma" section="4.1"/>, and their values
            stored.</li>
            <li>Any "a=extmap" attributes <bcp14>MUST</bcp14> be
            ignored.</t>

            <t>For each parsed as specified RTCP feedback mechanism that is
            supported by the local implementation, enable them on the
            associated media formats.</t>

            <t>For any specified "TIAS" bandwidth value, set this value in

            <xref target="RFC5285" sectionFormat="comma" section="5"/>, and their values
            stored.</li>
            <li>Any "a=rtcp-fb" attributes <bcp14>MUST</bcp14> be parsed as specified
            in
            <xref target="RFC4585" sectionFormat="comma" section="4.2"/>, and their values
            stored.</li>
            <li>If present, a constraint on the maximum RTP bitrate to single "a=rtcp-mux" attribute <bcp14>MUST</bcp14> be used when
            sending media,
            parsed as specified in
            <xref target="RFC3890"></xref>. If a "TIAS" value is not target="RFC5761" sectionFormat="comma" section="5.1.3"/>, and its
            presence or absence flagged and stored.</li>
            <li>If present, but an "AS" value is specified, generate a "TIAS"
            value using this formula:
            <list style="format">

              <t>TIAS = AS * 1000 * 0.95 - (50 * 40 * 8)</t>
            </list>The 50 is based on 50 packets per second, the 40 is
            based on an estimate of total header size, the 1000 changes
            the unit from kbps to bps (as required by TIAS), and the
            0.95 is to allocate 5% to RTCP. "TIAS" is used single "a=rtcp-mux-only" attribute <bcp14>MUST</bcp14> be
            parsed as specified in
            preference to "AS" because it provides more accurate
            control of bandwidth.</t>

            <t>For any "RR"
            <xref target="RFC8858" sectionFormat="comma" section="3"/>,
            and its presence or "RS" bandwidth values, handle absence flagged and stored.</li>
            <li>If present, a single "a=rtcp-rsize" attribute <bcp14>MUST</bcp14> be
            parsed as specified in
            <xref target="RFC3556"></xref>, Section 2.</t>

            <t>Any target="RFC5506" sectionFormat="comma" section="5"/>, and its presence or
            absence flagged and stored.</li>
            <li>If present, a single "a=rtcp" attribute <bcp14>MUST</bcp14> be parsed
            as specified "CT" bandwidth in
            <xref target="RFC3605" sectionFormat="comma" section="2.1"/>, but its value MUST be is
            ignored, as
            the meaning of this construct at the media level is not
            well defined.</t>

            <t>If the m= section information is of type audio:
            <list style="symbols">

              <t>For each specified "CN" media format, configure
              silence suppression for all supported media formats with
              the same clockrate, superfluous when using
            ICE.</li>
            <li>If present, "a=msid" attributes <bcp14>MUST</bcp14> be parsed as described
            specified in
            <xref target="RFC3389" />, Section 5, except for formats
              that have target="RFC8830" sectionFormat="comma" section="3.2"/>, and
            their own internal silence suppression
              mechanisms. Silence suppression values stored, ignoring any "appdata" field. If no "a=msid"
            attributes are present, a random msid-id value is generated for such formats (e.g.,
              Opus) a
            "default" MediaStream for the session, if not already present, and
            this value is controlled via fmtp parameters, stored.</li>
            <li>Any "a=imageattr" attributes <bcp14>MUST</bcp14> be parsed as discussed specified
            in
            <xref target="sec.voiceactivitydetection1" />.</t>

              <t>For each specified "telephone-event" media format,
              enable DTMF transmission for all supported media formats
              with the same clockrate, target="RFC6236" sectionFormat="comma" section="3"/>, and their values
            stored.</li>
            <li>Any "a=rid" lines <bcp14>MUST</bcp14> be parsed as described specified in
            <xref target="RFC4733" />, Section 2.5.1.2. If there are
              any supported media formats that do not have target="RFC8851" sectionFormat="comma" section="10"/>, and
            their values stored.</li>
            <li>If present, a
              corresponding telephone-event format, disable DTMF
              transmission for those formats.</t>

              <t>For any single "a=simulcast" line <bcp14>MUST</bcp14> be parsed
            as specified "ptime" value, configure in
            <xref target="RFC8853" format="default"/>, and
            its values stored.</li>
          </ul>
          <t>Otherwise, if the
              available media formats to "m=" proto value indicates use of SCTP,
          the following attribute lines <bcp14>MUST</bcp14> be processed:
          </t>
          <ul spacing="normal">
            <li>The "m=" fmt value <bcp14>MUST</bcp14> be parsed as specified packet size
              when sending. If in
            <xref target="RFC8841" sectionFormat="comma" section="4.3"/>,
            and the application protocol value stored.</li>
            <li>An "a=sctp-port" attribute <bcp14>MUST</bcp14> be present, and it <bcp14>MUST</bcp14>
            be parsed as specified size is not supported for
              a media format, use in
            <xref target="RFC8841" sectionFormat="comma" section="5.2"/>,
            and the next closest value instead.</t>
            </list></t>
          </list></t>
        </list></t>

        <t>Finally, if this description is of type "pranswer" or
        "answer", follow the processing defined stored.</li>
            <li>If present, a single "a=max-message-size" attribute <bcp14>MUST</bcp14>
            be parsed as specified in
            <xref target="sec.applying-an-answer" /> below.</t>
      </section>
      <section title="Applying an Answer"
      anchor="sec.applying-an-answer">

        <t>In addition to target="RFC8841" sectionFormat="comma" section="6"/>, and
            the steps mentioned above for processing a
        local or remote description, value stored. Otherwise, use the following steps specified default.</li>
          </ul>
          <t>Other attributes that are performed
        when processing a description of type "pranswer" or
        "answer".</t>

        <t>For each m= section, the following steps MUST not relevant to JSEP may also be performed:
        <list style="symbols">

          <t>If
          present, and implementations <bcp14>SHOULD</bcp14> process any that they
          recognize. As required by
          <xref target="RFC4566" sectionFormat="comma" section="5.13"/>, unknown
          attribute lines <bcp14>MUST</bcp14> be ignored.</t>
        </section>
        <section numbered="true" toc="default">
          <name>Semantics Verification</name>
          <t>Assuming that parsing completes successfully, the m= section has been rejected (i.e. port parsed
          description is set then evaluated to
          zero in the answer), stop any reception or transmission of
          media ensure internal consistency
          as well as proper support for this section, and, unless a non-rejected m= section
          is bundled with this m= mandatory features.
          Specifically, the following checks are performed:
          </t>
          <ul spacing="normal">
            <li>
              <t>For each "m=" section, discard any associated ICE
          components, as described valid values for each of the
            mandatory-to-use features enumerated in
            <xref target="I-D.ietf-mmusic-ice-sip-sdp" />, Section 3.4.3.1.</t>

          <t>If target="sec.usage-requirements" format="default"/> <bcp14>MUST</bcp14> be present.
            These values <bcp14>MAY</bcp14> be either present at the remote DTLS fingerprint has been changed media level or
            inherited from the
          tls-id 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 session level.
              </t>
              <ul spacing="normal">
                <li>ICE ufrag and password values, which <bcp14>MUST</bcp14> comply with
              the answer does not indicate an ICE restart or, size limits specified in
          the case of "have-remote-pranswer", new ICE credentials, an
          error MUST
              <xref target="RFC8839" sectionFormat="comma" section="5.4"/>.</li>
                <li>A tls-id value, which <bcp14>MUST</bcp14> be generated. set according to
              <xref target="RFC8842" sectionFormat="comma" section="5"/>. If an ICE restart
	       this is performed
          without a change in tls-id re-offer or fingerprint, then a response to a re-offer and the
	       tls-id value is different from that presently in use, the same
	       DTLS connection is not being 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 as long as the offer contained remote
	       description <bcp14>MUST</bcp14> be part of an ICE restart. Thus, JSEP implementations which process restart, together with
	       new ufrag and password values.</li>
		 <li>A DTLS
          data prior to receiving an answer MUST setup value, which <bcp14>MUST</bcp14> be prepared to receive
          either a ClientHello or data from the previous DTLS
          connection.</t>

          <t>If no valid DTLS connection exists, prepare set according to start a
          DTLS connection, using the
	       rules specified roles in
	       <xref target="RFC5763" sectionFormat="comma" section="5"/> and fingerprints,
          on any underlying ICE components, once they are active.</t>

          <t>If <bcp14>MUST</bcp14> be
	       consistent with the m= section proto value indicates use selected role of RTP:
          <list style="symbols">

            <t>If the m= current DTLS
	       connection, if one exists and is being continued.</li>
		 <li>DTLS fingerprint values, where at least one
	       fingerprint <bcp14>MUST</bcp14> be present.</li>
	       </ul>
	     </li>
	     <li>All RID values referenced in an "a=simulcast" line <bcp14>MUST</bcp14>
	     exist as "a=rid" lines.</li>
	     <li>Each "m=" section references RTCP feedback mechanisms is also checked to ensure that were prohibited
	     features are not present in used.</li>
	     <li>If the corresponding m= RTP/RTCP multiplexing policy is "require", each
	     "m=" section <bcp14>MUST</bcp14> contain an "a=rtcp-mux" attribute. If an "m="
	     section contains an "a=rtcp-mux-only" attribute, that
	     section <bcp14>MUST</bcp14> also contain an "a=rtcp-mux" attribute.</li>
	     <li>If an "m=" section was present in the offer, previous answer, the
	     state of RTP/RTCP multiplexing <bcp14>MUST</bcp14> match what was
	     previously negotiated.</li>
	   </ul>
	   <t>If this indicates a negotiation problem and MUST
            result in an error. However, new media formats and new RTP
            header extension values session description is of type "pranswer" or
	   "answer", the following additional checks are permitted in applied:
	   </t>
	   <ul spacing="normal">
	     <li>The session description must follow the answer, as
            described rules defined in

	     <xref target="RFC3264" />, Section 7, and
            <xref target="RFC5285" />, Section 6.</t>

            <t>If sectionFormat="comma" section="6"/>, including the m= section has RTCP mux enabled, discard
	     requirement that the RTCP
            ICE component, if one exists, and begin or continue muxing
            RTCP over number of "m=" sections <bcp14>MUST</bcp14> exactly
	     match the RTP ICE component, as specified number of "m=" sections in
            <xref target="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.</t>

            <t>If associated
	     offer.</li>
	     <li>For each "m=" section, the m= section has reduced-size RTCP enabled,
            configure media type and protocol values
	     <bcp14>MUST</bcp14> exactly match the RTCP transmission for this m= section to use
            reduced-size RTCP, as specified media type and protocol values in
            <xref target="RFC5506" />.</t>

            <t>If
	     the directional attribute corresponding "m=" section in the answer indicates
            that associated offer.</li>
	   </ul>
	   <t>If any of the JSEP implementation should preceding checks failed, processing <bcp14>MUST</bcp14>
	   stop and an error <bcp14>MUST</bcp14> be sending returned.</t>
	 </section>
       </section>
       <section anchor="sec.applying-a-local-desc" numbered="true" toc="default">
	 <name>Applying a Local Description</name>
	 <t>The following steps are performed at the media
            ("sendonly" for engine level
	 to apply a local answers, "recvonly" for remote
            answers, or "sendrecv" for either type of answer), choose description. If an error is returned, the media format
	 session <bcp14>MUST</bcp14> be restored to send as the most preferred media format
            from state it was in before
	 performing these steps.</t>
	 <t>First, "m=" sections are processed. For each "m=" section, the remote description that
	 following steps <bcp14>MUST</bcp14> be performed; if any parameters are out of
	 bounds or cannot be applied, processing <bcp14>MUST</bcp14> stop and an error
	 <bcp14>MUST</bcp14> be returned.
	 </t>
	 <ul spacing="normal">
	   <li>If this "m=" section is also locally supported, new, begin gathering candidates for
	   it, as discussed defined in
	   <xref target="RFC3264" />, Sections 6.1 target="RFC8445" sectionFormat="comma" section="5.1.1"/>, unless it is
	   definitively being bundled (either (1) this is an offer and 7, the
	   "m=" section is marked bundle-only or (2)&nbsp;it is an answer and start
            transmitting RTP media using that format once the
            underlying transport layers
	   "m=" section is bundled into another "m=" section).</li>
	   <li>Or, if the ICE ufrag and password values have been established. If an
            SSRC has not already been chosen changed,
	   trigger the ICE agent to start an ICE restart as described in
	   <xref target="RFC8445" sectionFormat="comma" section="9"/>, and begin
	   gathering new candidates for this outgoing RTP
            stream, choose a random one. the "m=" section. If this
	   description is an answer, also start checks on that media
	   section.</li>
	   <li>
	     <t>If the "m=" section proto value indicates use of RTP:
	     </t>
	     <ul spacing="normal">
	       <li>
		 <t>If there is already being
            transmitted, no RtpTransceiver associated with this "m="
	     section, find one and associate it with this "m=" section
	     according to the same SSRC SHOULD be used unless following steps. Note that this situation
	     will only occur when applying an offer.
		 </t>
		 <ul spacing="normal">
		   <li>Find the
            clockrate RtpTransceiver that corresponds to this "m="
	       section, using the mapping between transceivers and "m="
	       section indices established when creating the offer.</li>
		   <li>Set the value of this RtpTransceiver's mid property to
	       the new codec MID of the "m=" section.</li>
		 </ul>
	       </li>
	       <li>If RTCP mux is different, in which case a
            new SSRC MUST be chosen, indicated, prepare to demux RTP and RTCP
	     from the RTP ICE component, as specified in
	     <xref target="RFC7160" />, Section 3.1.</t>

            <t>The payload type target="RFC5761" sectionFormat="comma" section="5.1.3"/>.</li>
	       <li>For each specified RTP header extension, establish a
	     mapping from between the remote description extension ID and URI, as described in
	     <xref target="RFC5285" sectionFormat="comma" section="6"/>.</li>
	       <li>If the MID header extension is
            used supported, prepare to determine payload types
	     demux RTP streams intended for this "m=" section based on the outgoing RTP
            streams, including
	     MID header extension, as described in
	     <xref target="RFC8843" sectionFormat="comma" section="15"/>.</li>
	       <li>For each specified media format, establish a mapping
	     between the payload type for and the send actual media
            format chosen above. Any RTP header extensions that were
            negotiated should be included format, as
	     described in the outgoing
	     <xref target="RFC3264" sectionFormat="comma" section="6.1"/>. In addition,
	     prepare to demux RTP streams,
            using the extension mapping from the remote description; if
            the RID header extension has been negotiated, and RID
            values are specified, include the RID header extension in streams intended for this "m=" section
	     based on the outgoing RTP streams, media formats supported by this "m=" section, as indicated
	     described in
	     <xref target="I-D.ietf-mmusic-rid"></xref>, target="RFC8843" sectionFormat="comma" section="9.2"/>.

<!-- [rfced] Sections 5.9, 5.10, 5.11, and 6:

or a Section 4.</t>

            <t>If the m= section is of type audio, 10.2.  We have updated these to refer to 7.2 and silence
            suppression was configured 9.2,
respectively.  Please review.

 Original:
  In addition, prepare to demux RTP
  streams intended for this m= section based on the send media format formats
  supported by this m= section, as a
            result of processing the remote description, and described in
  [I-D.ietf-mmusic-sdp-bundle-negotiation], Section 10.2.
 ...
  *  For each specified SSRC that is also
            enabled signaled in the m= section,
     prepare to demux RTP streams intended for this m= section using
     that format SSRC, as described in
     [I-D.ietf-mmusic-sdp-bundle-negotiation], Section 10.2.
 ...
  If the local description, use
            silence suppression answer contains valid bundle groups, discard any ICE
  components for outgoing media, in accordance with the guidance in
            <xref target="sec.voiceactivitydetection1" />. If these
            conditions are not met, silence suppression MUST NOT m= sections that will be
            used for outgoing media.</t>

            <t>If simulcast has been negotiated, send bundled onto the number of
            Source RTP Streams primary
  ICE components in each bundle, and begin muxing these m= sections
  accordingly, as specified described in
            <xref target="I-D.ietf-mmusic-sdp-simulcast"></xref>,
  [I-D.ietf-mmusic-sdp-bundle-negotiation], Section 6.2.2.</t>

            <t>If 8.2.
 ...
  When bundling, associating incoming RTP/RTCP with the send media format chosen above has a
            corresponding proper m=
  section is defined in [I-D.ietf-mmusic-sdp-bundle-negotiation],
  Section 10.2. -->

 </li>
	       <li>For each specified "rtx" media format, or a FEC mechanism has
            been negotiated, establish a Redundancy RTP Stream with a
            random SSRC for each Source RTP Stream, and start or
            continue transmitting RTX/FEC packets as needed.</t>

            <t>If
	     mapping between the send media format chosen above has a
            corresponding "red" media format of the same clockrate,
            allow redundant encoding using the specified format for
            resiliency purposes, RTX payload type and its associated
	     primary payload type, as discussed described in
	     Sections <xref target="RFC4588" section="8.6"
	     sectionFormat="bare"/> and <xref target="RFC4588" section="8.7"
	     sectionFormat="bare"/> of <xref target="I-D.ietf-rtcweb-fec" />, target="RFC4588"/>.

<!-- [rfced] Sections 5.9 and 5.11:  We had to change "[RFC4588],
Sections 8.6 and 8.7" to "Sections 8.6 and 8.7 of [RFC4588]"
(in 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.</t>

            <t>Enable the RTCP feedback mechanisms referenced 5.9) and "[RFC3264], Sections 6.1 and 7" to "Sections 6.1
and 7 of [RFC3264]" (in Section 5.11) in order to get the
            media section for all Source RTP Streams using hyperlinks
to work properly in the .html/.pdf files.  Please let us know any concerns.

Original:
 *  For each specified "rtx" media formats. Specifically, begin or continue
            sending format, establish a mapping
    between the requested feedback types RTX payload type and reacting to
            received feedback, its associated primary payload
    type, as specified described in
            <xref target="RFC4585" />, Section 4.2. When sending RTCP
            feedback, follow the rules [RFC4588], Sections 8.6 and recommendations from
            <xref target="RFC8108"></xref> Section 5.4.1, to select
            which SSRC to use.</t>

            <t>If 8.7.
...
 *  If the directional attribute in the answer indicates that the
    JSEP implementation should not be sending media
            ("recvonly" ("sendonly" for
    local answers, "sendonly" "recvonly" for remote answers, or "inactive" "sendrecv" for
    either type of answer) stop 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 [RFC3264], Sections 6.1
    and 7, and start transmitting all RTP media, but continue sending RTCP, media using that format once
    the underlying transport layers have been established.

Currently:
 -  For each specified "rtx" media format, establish a mapping
    between the RTX payload type and its associated primary payload
    type, as described in
            <xref target="RFC3264" />, Section 5.1.</t>
          </list></t>

          <t>If Sections 8.6 and 8.7 of [RFC4588].
...
 -  If the m= section proto value directional attribute in the answer indicates use of SCTP:
          <list style="symbols">

            <t>If an SCTP association exists, and that the
    JSEP implementation should be sending media ("sendonly" for
    local answers, "recvonly" for remote SCTP port
            has changed, discard the existing SCTP association. This
            includes the case when answers, or "sendrecv" for
    either type of answer), choose the PeerConnection state is
            "have-remote-pranswer".</t>

            <t>If no valid SCTP association exists, prepare media format to initiate
            a SCTP association over the associated ICE component and
            DTLS connection, using the local SCTP port value from the
            local description, and send as the remote SCTP port value
    most preferred media format from the remote description, description that is
    also locally supported, as described discussed in
            <xref target="I-D.ietf-mmusic-sctp-sdp" />, Section
            10.2.</t>
          </list></t>
        </list></t>

        <t>If the answer contains valid bundle groups, discard any ICE
        components for the m= sections Sections 6.1 and 7 of
    [RFC3264], and start transmitting RTP media using that will be bundled onto format
    once the
        primary ICE components in each bundle, and begin muxing these
        m= sections accordingly, as described in
        <xref target="I-D.ietf-mmusic-sdp-bundle-negotiation" />,
        Section 8.2.</t>

        <t>If underlying transport layers have been established. -->

 </li>
	       <li>If the directional attribute is of type "sendrecv" or
	     "recvonly", enable receipt and decoding of media.</li>
	     </ul>
	   </li>
	 </ul>
	 <t>Finally, if this description is of type "pranswer" or
	 "answer", and there are still
        remaining candidates in the ICE candidate pool, discard
        them.</t>
      </section>
    </section>
    <section title="Processing RTP/RTCP" anchor="sec.rtp.demux">

      <t>When bundling, associating incoming RTP/RTCP with follow the proper
      m= section is processing defined in
	 <xref target="I-D.ietf-mmusic-sdp-bundle-negotiation" />, Section
      10.2. When not bundling, the proper m= section target="sec.applying-an-answer" format="default"/> below.</t>
       </section>
       <section anchor="sec.applying-a-remote-desc" numbered="true" toc="default">
	 <name>Applying a Remote Description</name>
	 <t>The following steps are performed to apply a remote
	 description. If an error is clear from the
      ICE component over which returned, the RTP/RTCP is received.</t>

      <t>Once session <bcp14>MUST</bcp14> be
	 restored to the proper m= section(s) are known, RTP/RTCP state it was in before performing these
	 steps.</t>
	 <t>If the answer contains any "a=ice-options" attributes where
	 "trickle" is delivered
      to listed as an attribute, update the RtpTransceiver(s) associated with PeerConnection
	 canTrickle property to be "true". Otherwise, set this property to
	 "false".</t>
	 <t>The following steps <bcp14>MUST</bcp14> be performed for attributes at the m= section(s) and
      further processing
	 session level; if any parameters are out of bounds or cannot
	 be applied, processing <bcp14>MUST</bcp14> stop and an error <bcp14>MUST</bcp14> be returned.

	 </t>
	 <ul spacing="normal">
	   <li>For any specified "CT" bandwidth value, set this value as the RTP/RTCP is done at
	   limit for the RtpTransceiver
      level. This includes using RID maximum total bitrate for all "m=" sections, as
	   specified in
	   <xref target="I-D.ietf-mmusic-rid" /> target="RFC4566" sectionFormat="comma" section="5.8"/>. Within this
	   overall limit, the implementation can dynamically decide how
	   to distinguish best allocate the available bandwidth between
      multiple Encoded Streams, as well as determine which Source RTP
      stream should be repaired by a given Redundancy RTP stream.</t>
    </section>
    <section title="Examples" anchor="sec.examples">

      <t>Note "m=" sections,
	   respecting any specific limits that this example section shows several SDP fragments. To
      format have been specified for
	   individual "m=" sections.</li>
	   <li>For any specified "RR" or "RS" bandwidth values, handle as
	   specified in 72 columns, some
	   <xref target="RFC3556" sectionFormat="comma" section="2"/>.</li>
	   <li>Any "AS" bandwidth value <bcp14>MUST</bcp14> be ignored, as the meaning
	   of this construct at the lines in SDP have been split
      into multiple lines, where leading whitespace indicates that a
      line session level is not well
	   defined.

<!-- [rfced] Section 5.10:  For ease of the reader, we suggest adding
a continuation citation that provides the definition of "AS."  Please let us know
if we may update as follows.

Original:
 o  Any "AS" bandwidth value MUST be ignored, as the previous line. In addition, some
      blank lines have been added to improve readability but are not
      valid in SDP.</t>

      <t>More examples meaning of SDP for WebRTC call flows, including examples
      with IPv6 addresses, can this
    construct at the session level is not well defined.

Suggested:
 *  Any "AS" bandwidth value ([RFC4566], Section 5.8) MUST be found in
      <xref target="I-D.ietf-rtcweb-sdp"></xref>.</t>
      <section title="Simple Example" anchor="sec.simple-examples">

        <t>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
    ignored, as the following section
        provides a more detailed example meaning of what could happen in a JSEP
        session.</t>

        <t>The code flow below shows Alice's endpoint initiating this construct at the session to Bob's endpoint. The messages from the JavaScript
        application in Alice's browser to level is
    not well defined. -->

 </li>
	 </ul>
	 <t>For each "m=" section, the JavaScript in Bob's
        browser, abbreviated as AliceJS and BobJS respectively, following steps <bcp14>MUST</bcp14> be performed;
	 if any parameters are
        assumed to flow over some signaling protocol via a web server.
        The JavaScript on both Alice's side out of bounds or cannot be applied,
	 processing <bcp14>MUST</bcp14> stop and Bob's side waits for
        all candidates before sending an error <bcp14>MUST</bcp14> be returned.
	 </t>
	 <ul spacing="normal">
	   <li>
	     <t>If the offer ICE ufrag or answer, so password changed from the
        offers and answers are complete; trickle previous
	   remote description:
	     </t>
	     <ul spacing="normal">
	       <li>If the description is of type "offer", the
	     implementation <bcp14>MUST</bcp14> note that an ICE restart is not used. The
        user agents (JSEP implementations) in Alice and Bob's browsers,
        abbreviated needed, as AliceUA and BobUA respectively, are using
	     described in
	     <xref target="RFC8839" sectionFormat="comma" section="4.4.1.1.1"/>.</li>
	       <li>If the
        default bundle policy description is of "balanced", and type "answer" or "pranswer",
	     then check to see if the default RTCP mux
        policy of "require".</t>

        <t>
          <figure>
            <artwork>
              <![CDATA[
//                  set up current local media description is an
	     ICE restart, and if not, generate an error. If the
	     PeerConnection state
AliceJS->AliceUA:   create new PeerConnection
AliceJS->AliceUA:   addTrack with two tracks: audio is "have-remote-pranswer" and video
AliceJS->AliceUA:   createOffer the ICE
	     ufrag or password changed from the previous provisional
	     answer, then signal the ICE agent to get offer
AliceJS->AliceUA:   setLocalDescription with offer
AliceUA->AliceJS:   multiple onicecandidate events with candidates

//                  wait discard any previous
	     ICE check list state for the "m=" section. Finally, signal
	     the ICE gathering agent to complete
AliceUA->AliceJS:   onicecandidate event with null candidate
AliceJS->AliceUA:   get |offer-A1| begin checks.</li>
	     </ul>
	   </li>
	   <li>If the current local description indicates an ICE restart
	   and either the ICE ufrag or password has not changed from pendingLocalDescription

//                  |offer-A1| the
	   previous remote description, as prescribed by
	   <xref target="RFC8445" sectionFormat="comma" section="9"/>, generate an
	   error.

<!-- [rfced] Section 5.10:  We found this sentence confusing, as we
could not tell what "as prescribed by [RFC8445], Section 9" and
"generate an error" refer to.  Please confirm that the citation 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
correct and video tracks

//                  Bob accepts call
BobJS->BobUA:       addTrack with will be clear to readers.

Original:
 o  If the current local tracks
BobJS->BobUA:       createAnswer
BobJS->BobUA:       setLocalDescription with answer
BobUA->BobJS:       multiple onicecandidate events description indicates an ICE restart, and
    either the ICE ufrag or password has not changed from the previous
    remote description, as prescribed by [RFC8445], Section 9,
    generate an error. -->

 </li>
	   <li>Configure the ICE components associated with candidates

//                  wait this media
	   section to use the supplied ICE remote ufrag and password for
	   their connectivity checks.</li>
	   <li>Pair any supplied ICE gathering to complete
BobUA->BobJS:       onicecandidate event candidates with null candidate
BobJS->BobUA:       get |answer-A1| from currentLocalDescription

//                  |answer-A1| is sent over signaling protocol to Alice
BobJS->WebServer:   signaling any gathered local
	   candidates, as described in
	   <xref target="RFC8445" sectionFormat="comma" section="6.1.2"/>, and start
	   connectivity checks with |answer-A1|
WebServer->AliceJS: signaling the appropriate credentials.</li>

	   <li>If an "a=end-of-candidates" attribute is present, process
	   the end-of-candidates indication as described in
	   <xref target="RFC8838" sectionFormat="comma" section="14"/>.</li>
	    <li>
	      <t>If the "m=" section proto value indicates use of RTP:
	      </t>
	      <ul spacing="normal">
		<li>If the "m=" section is being recycled (see
	      <xref target="sec.subsequent-offers" format="default"/>), dissociate
	      the currently associated RtpTransceiver by setting its mid
	      property to "null", and discard the mapping between the
	      transceiver and its "m=" section index.</li>
		<li>
		  <t>If the "m=" section is not associated with |answer-A1|

//                  |answer-A1| arrives at Alice
AliceJS->AliceUA:   setRemoteDescription any
	      RtpTransceiver (possibly because it was dissociated in the
	      previous step), either find an RtpTransceiver or create one
	      according to the following steps:
		  </t>
		  <ul spacing="normal">
		    <li>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 |answer-A1|
AliceUA->AliceJS:   ontrack events for audio any "m=" section and video tracks

//                  media flows
BobUA->AliceUA:     media sent from Bob are not stopped, find the first
		(according to Alice
AliceUA->BobUA:     media sent from Alice the canonical order described in
		<xref target="sec.initial-offers" format="default"/>) such
		RtpTransceiver.</li>
		    <li>If no RtpTransceiver was found in the previous step,
		create one with a recvonly direction.</li>
		    <li>Associate the found or created RtpTransceiver with the
		"m=" section by setting the value of the RtpTransceiver's
		mid property to Bob
]]>
</artwork>
          </figure>
        </t>

        <t>The SDP for |offer-A1| looks like:</t>

        <t>
          <figure>
            <artwork alt="offer-A1">
              <![CDATA[
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
=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
]]>
</artwork>
          </figure>
        </t>

        <t>The SDP for |answer-A1| looks like:</t>

        <t>
          <figure>
            <artwork alt="answer-A1">
              <![CDATA[
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
=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
]]>
</artwork>
          </figure>
        </t>
      </section>
      <section title="Detailed Example" anchor="sec.detailed-example">

        <t>This section shows a more involved example the MID of the "m=" section, and establish
		a session mapping 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", the transceiver and the index of the "m="
		section. If the "m=" section does not include a single TURN server. Initially, both Alice
        and Bob establish an audio channel and MID (i.e.,
		the remote endpoint does not support the MID extension),
		generate a data channel. Later,
        Bob adds two video flows, one value for his video feed, and one the RtpTransceiver mid property,
		following the guidance for
        screensharing, both supporting FEC, "a=mid" mentioned in
		<xref target="sec.initial-offers" format="default"/>.</li>
		  </ul>
		</li>
		<li>For each specified media format that is also supported
	      by the local implementation, establish a mapping between
	      the specified payload type and with the video feed
        configured media format, as
	      described in
	      <xref target="RFC3264" sectionFormat="comma" 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 simulcast. Alice accepts these video flows, but
        does each
	      format that is indicated in their ordering. If any
	      indicated media format is not add video flows of her own, so they are handled as
        recvonly. Alice also specifies a maximum video decoder
        resolution.</t>

        <t>
          <figure>
            <artwork>
              <![CDATA[
//                  set up supported by the local
	      implementation, it <bcp14>MUST</bcp14> be ignored.</li>
		<li>For each specified "rtx" media state
AliceJS->AliceUA:   create new PeerConnection
AliceJS->AliceUA:   addTrack with format, establish a
	      mapping between the RTX payload type and its associated
	      primary payload type, as described in
	      <xref target="RFC4588" sectionFormat="comma" section="4"/>. If any referenced
	      primary payload types are not present, this <bcp14>MUST</bcp14> result in
	      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 error. Note that RTX payload types may refer 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 with audio track from Alice

//                  candidates primary
	      payload types that 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 not supported by the local audio
BobJS->BobUA:       createDataChannel to get data channel
BobJS->BobUA:       createAnswer to get |answer-B1|
BobJS->BobUA:       setLocalDescription with |answer-B1|

//                  |answer-B1| media
	      implementation, in which case the RTX payload type <bcp14>MUST</bcp14>
	      also be ignored.</li>
		<li>For each specified fmtp parameter that is sent supported by
	      the local implementation, enable them on the associated
	      media formats.</li>
		<li>For each specified Synchronization Source (SSRC) that is signaled in the "m="
	      section, prepare 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 demux RTP streams intended for this "m="
	      section using that SSRC, as described in
	      <xref target="RFC8843" sectionFormat="comma" section="9.2"/>.</li>
		<li>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
	      <xref target="RFC5285" sectionFormat="comma" section="5"/>. Specifically, this
	      means that the implementation records the extension ID 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

//
	      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 <bcp14>MUST</bcp14> be
	      ignored.</li>
		<li>For each specified RTCP feedback mechanism that is
	      supported by the local implementation, enable them on the
	      associated media formats.</li>
		<li>
		  <t>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
	      <xref target="RFC3890" format="default"/>. If a "TIAS" value is flowing between endpoints
BobUA->AliceUA:     audio+data sent not
	      present but an "AS" value is specified, generate a "TIAS"
	      value using this formula:
		  </t>
		  <ul empty="true">
		    <li>TIAS = AS * 1000 * 0.95 - (50 * 40 * 8)</li>
		  </ul>
		  <t>
	      The "50" is based on 50 packets per second, the "40" is
	      based on an estimate of total header size, the "1000" changes
	      the unit from Bob kbps to Alice
AliceUA->BobUA:     audio+data sent from Alice bps (as required by TIAS), and the
	      "0.95" is to Bob

//                  some time later Bob adds two video streams
//                  note, no candidates exchanged, because allocate 5% to RTCP.

<!-- [rfced] Section 5.10:  For ease of bundle
BobJS->BobUA:       addTrack with first video stream
BobJS->BobUA:       addTrack with second video stream
BobJS->BobUA:       createOffer the reader, should the "8"
also be explained (e.g., possibly "bytes to get |offer-B2|
BobJS->BobUA:       setLocalDescription with |offer-B2|

//                  |offer-B2| bits")?  We ask because
explanations are provided for the other four numbers.

Original:
 TIAS = AS * 1000 * 0.95 - (50 * 40 * 8)

 The 50 is sent based on 50 packets per second, the 40 is based on an
 estimate of total header size, the 1000 changes the unit from
 kbps to 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 bps (as required by TIAS), and the 0.95 is to get |answer-B2|
AliceJS->AliceUA:   setLocalDescription with |answer-B2|

//                  |answer-B2| allocate
 5% to RTCP.-->

   "TIAS" is sent over signaling protocol used in
	      preference to Bob
AliceJS->WebServer: signaling with |answer-B2|
WebServer->BobJS:   signaling with |answer-B2|
BobJS->BobUA:       setRemoteDescription with |answer-B2|

// "AS" because it provides more accurate
	      control of bandwidth.</t>
		</li>
		<li>For any "RR" or "RS" bandwidth values, handle as
	      specified in
	      <xref target="RFC3556" sectionFormat="comma" section="2"/>.</li>
		<li>Any specified "CT" bandwidth value <bcp14>MUST</bcp14> be ignored, as
	      the meaning of this construct at the media level is flowing between endpoints
BobUA->AliceUA:     audio+video+data sent from Bob to Alice
AliceUA->BobUA:     audio+video+data sent from Alice to Bob
]]>
</artwork>
          </figure> not
	      well defined.</li>
		<li>
		  <t>If the "m=" section is of type "audio":
		  </t>

        <t>The SDP
		  <ul spacing="normal">
		    <li>For each specified "CN" media format, configure
		silence suppression for |offer-B1| looks like:</t>

        <t>
          <figure>
            <artwork alt="offer-B1">
              <![CDATA[
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
]]>
</artwork>
          </figure>
        </t>

        <t>|offer-B1-candidate-1| looks like:</t>

        <t>
          <figure>
            <artwork alt="offer-B1-candidate-1">
              <![CDATA[
ufrag ATEn
index 0
mid   a1
attr  candidate:1 1 udp 2113929471 203.0.113.100 10100 typ host
]]>
</artwork>
          </figure>
        </t>

        <t>|offer-B1-candidate-2| looks like:</t>

        <t>
          <figure>
            <artwork alt="offer-B1-candidate-2">
              <![CDATA[
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
]]>
</artwork>
          </figure> all supported media formats with
		the same clock rate, as described in
		<xref target="RFC3389" sectionFormat="comma" 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
		<xref target="sec.voiceactivitydetection1" format="default"/>.</li>
		    <li>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
		<xref target="RFC4733" sectionFormat="comma" 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.</li>
		    <li>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.</li>
		  </ul>
		</li>
	      </ul>
	    </li>
	  </ul>
	  <t>Finally, if this description is of type "pranswer" or
	  "answer", follow the processing defined in
	  <xref target="sec.applying-an-answer" format="default"/> below.</t>
	</section>
	<section anchor="sec.applying-an-answer" numbered="true" toc="default">
	  <name>Applying an Answer</name>
	  <t>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".</t>
	  <t>For each "m=" section, the following steps <bcp14>MUST</bcp14> be performed:
	  </t>

        <t>|offer-B1-candidate-3| looks like:</t>

        <t>
          <figure>
            <artwork alt="offer-B1-candidate-3">
              <![CDATA[
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
]]>
</artwork>
          </figure>
	  <ul spacing="normal">
	    <li>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
	    <xref target="RFC8839" sectionFormat="comma" section="4.4.3.1"/>.</li>
	     <li>If the remote DTLS fingerprint has been changed or the
	     tls-id 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 <bcp14>MUST</bcp14> be generated. If an ICE restart is performed
	     without a change in tls-id 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 as long as the offer contained
	     an ICE restart. Thus, JSEP implementations that process DTLS
	     data prior to receiving an answer <bcp14>MUST</bcp14> be prepared to receive
	     either a ClientHello or data from the previous DTLS
	     connection.</li>
	     <li>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.</li>
	     <li>
	       <t>If the "m=" section proto value indicates use of RTP:
	       </t>

        <t>The SDP
	       <ul spacing="normal">
		 <li>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 <bcp14>MUST</bcp14>
	       result in an error. However, new media formats and new RTP
	       header extension values are permitted in the answer, as
	       described in
	       <xref target="RFC3264" sectionFormat="comma" section="7"/> and
	       <xref target="RFC5285" sectionFormat="comma" section="6"/>.</li>
		 <li>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
	       <xref target="RFC5761" sectionFormat="comma" 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 <bcp14>MUST</bcp14> result in an error.</li>
		 <li>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
	       <xref target="RFC5506" format="default"/>.</li>
		 <li>If the directional 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 <xref target="RFC3264" section="6.1"
   sectionFormat="bare"/> and <xref target="RFC3264" section="7" sectionFormat="bare"/> of <xref target="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 random one. If media is already being
	       transmitted, the same SSRC <bcp14>SHOULD</bcp14> be used unless the
	       clock rate of the new codec is different, in which case a
	       new SSRC <bcp14>MUST</bcp14> be chosen, as specified in
	       <xref target="RFC7160" sectionFormat="comma" section="3.1"/>.</li>
		 <li>The payload type mapping from the remote description is
	       used to determine payload types for the outgoing RTP
	       streams, including the payload type for the send media
	       format chosen above. Any RTP header extensions that were
	       negotiated should be included in the outgoing RTP streams,
	       using the extension mapping from the remote description; if
	       the RID header extension has been negotiated and RID
	       values are specified, include the RID header extension in
	       the outgoing RTP streams, as indicated in
	       <xref target="RFC8851" sectionFormat="comma" section="4"/>.</li>
		 <li>If (1) the "m=" section is of type "audio" and (2)&nbsp;silence
	       suppression was configured for the send media format as a
	       result of processing the remote description and is also
	       enabled for that format in the local description, use
	       silence suppression for outgoing media, in accordance with
	       the guidance in
	       <xref target="sec.voiceactivitydetection1" format="default"/>.

<!-- [rfced] Section 5.11:  As it appears that "is also enabled"
refers to silence suppression, we updated this sentence as follows.
Please let us know if this is incorrect.

Original:
 *  If the m= section is of type audio, and silence suppression was
    configured for the send media format as a result of processing
    the remote description, and is 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.

Currently:
 -  If (1) the "m=" section is of type "audio" and (2) silence
    suppression was configured for the send media format as a
    result of processing the remote description and is 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 <bcp14>MUST NOT</bcp14> be
	       used for outgoing media.</li>
		 <li>If simulcast has been negotiated, send the number of
	       Source RTP Streams as specified in
	       <xref target="RFC8853" sectionFormat="comma" section="6.2.2"/>.

<!-- [rfced] Section 5.11:  Please (1) confirm that Section 6.2.2 of
RFC 8853 [I-D.ietf-mmusic-sdp-simulcast] (with version -14 being the latest
for this draft) is the correct section to cite here and (2) clarify
the meaning of "send the number of Source RTP Streams as specified in"
(perhaps "... the specified number of ..." or "... the appropriate
number of ..."?).

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

   </li>
		 <li>If the send media format chosen above has a
	       corresponding "rtx" media format or a FEC mechanism has
	       been negotiated, establish a redundancy RTP stream with a
	       random SSRC for each Source RTP Stream, and start or
	       continue transmitting RTX/FEC packets as needed.</li>
		 <li>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 |answer-B1| looks like:</t>

        <t>
          <figure>
            <artwork alt="answer-B1">
              <![CDATA[
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
]]>
</artwork>
          </figure>
        </t>

        <t>|answer-B1-candidate-1| looks like:</t>

        <t>
          <figure>
            <artwork alt="answer-B1-candidate-1">
              <![CDATA[
ufrag 7sFv
index 0
mid   a1
attr  candidate:1 1 udp 2113929471 203.0.113.200 10200 typ host
]]>
</artwork>
          </figure>
        </t>

        <t>|answer-B1-candidate-2| looks like:</t>

        <t>
          <figure>
            <artwork alt="answer-B1-candidate-2">
              <![CDATA[
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
]]>
</artwork>
          </figure>
        </t>

        <t>|answer-B1-candidate-3| looks like:</t>

        <t>
          <figure>
            <artwork alt="answer-B1-candidate-3">
              <![CDATA[
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
]]>
</artwork>
          </figure>
	       resiliency purposes, as discussed in
	       <xref target="RFC8854" sectionFormat="comma" 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.</li>
		 <li>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
	       <xref target="RFC4585" sectionFormat="comma" section="4.2"/>. When sending RTCP
	       feedback, follow the rules and recommendations from
   <xref target="RFC8108" sectionFormat="comma" section="5.4.1"/> to select
	       which SSRC to use.</li>
		 <li>If the directional 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
	       <xref target="RFC3264" sectionFormat="comma" section="5.1"/>.</li>
	       </ul>
	     </li>
	     <li>
	       <t>If the "m=" section proto value indicates use of SCTP:
	       </t>

        <t>The
	       <ul spacing="normal">
		 <li>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".</li>
		 <li>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
	       <xref target="RFC8841" sectionFormat="comma" section="10.2"/>.</li>
	       </ul>
	     </li>
	   </ul>
	   <t>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
	   <xref target="RFC8843" sectionFormat="comma" section="7.2"/>.</t>
	   <t>If the description is of type "answer" and there are still
	   remaining candidates in the ICE candidate pool, discard
	   them.</t>
	 </section>
       </section>
       <section anchor="sec.rtp.demux" numbered="true" toc="default">
	 <name>Processing RTP/RTCP</name>
	 <t>When bundling, associating incoming RTP/RTCP with the proper
	 "m=" section is defined in
	 <xref target="RFC8843" sectionFormat="comma" section="9.2"/>. When not bundling, the proper "m=" section is clear from the
	 ICE component over which the RTP/RTCP is received.</t>
	 <t>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 RID
	 <xref target="RFC8851" format="default"/> to distinguish between
	 multiple encoded streams, as well as to determine which Source RTP
	 stream should be repaired by a given redundancy RTP stream.</t>
       </section>
       <section anchor="sec.examples" numbered="true" toc="default">
	 <name>Examples</name>
	 <t>Note that this example section shows several SDP for |offer-B2| 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 shown below. a continuation of the previous line. In addition addition, some
	 blank lines have been added to the
        new m= sections improve readability but are not
	 valid in SDP.</t>
	 <t>More examples of SDP for video, both WebRTC call flows, including examples
	 with IPv6 addresses, can be found in
	 <xref target="I-D.ietf-rtcweb-sdp" format="default"/>.</t>
	 <section anchor="sec.simple-examples" numbered="true" toc="default">
	   <name>Simple Example</name>
	   <t>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 which what could happen in a JSEP
	   session.</t>
	   <t>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 offering FEC,
	   assumed to flow over some signaling protocol via a web server.
	   The JavaScript on both Alice's side and
        one 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 using the
	   default bundle policy of which "balanced", and the default RTCP mux
	   policy of "require".

<!-- [rfced] Section 7.1:  We had trouble following the meaning of
this sentence.  If the suggested text is offering simulcast, note not correct, please clarify.

Original:
 The user agents (JSEP implementations) in Alice and Bob's
 browsers, abbreviated as AliceUA and BobUA respectively, are using
 the increment default bundle policy of the
        version number in the o= line, changes to the c= line,
        indicating the local candidate that was selected, "balanced", and the
        inclusion default RTCP mux
 policy of gathered candidates as a=candidate lines.</t>

        <t>
          <figure>
            <artwork alt="offer-B2">
              <![CDATA[
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
=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
=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
]]>
</artwork>
          </figure>
        </t>

        <t>The SDP for |answer-B2| is shown below. In addition to "require".

Suggested:
 The user agents (JSEP implementations) in Alice's and Bob's
 browsers, abbreviated as "AliceUA" and "BobUA", respectively, are
 using the
        acceptance default bundle policy of "balanced" (AliceUA) and the video m= sections, the use
 default RTCP mux policy of a=recvonly "require" (BobUA). -->

   </t>
<!-- [rfced] Throughout: <artwork> has been converted to
        indicate one-way video, <sourcecode> as
applicable and we set the use of a=imageattr "type" attribute.  Please review and let us know
corrections are needed.
-->

	   <sourcecode name="" type="pseudocode"><![CDATA[
//                  set up local media state
AliceJS->AliceUA:   create new PeerConnection
AliceJS->AliceUA:   addTrack with two tracks: audio and video
AliceJS->AliceUA:   createOffer to limit the
        received resolution, note the use of setup:passive get offer
AliceJS->AliceUA:   setLocalDescription with offer
AliceUA->AliceJS:   multiple onicecandidate events with candidates

//                  wait for ICE gathering to maintain
        the existing DTLS roles.</t>

        <t>
          <figure>
            <artwork alt="answer-B2">
              <![CDATA[ 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 ]]></sourcecode>

	   <t>The SDP for |offer-A1| looks like:</t>
	   <sourcecode name="offer-A1" type="sdp"><![CDATA[
v=0
o=- 4962303333179871723 2 4962303333179871722 1 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 10100 UDP/TLS/RTP/SAVPF 96 0 8 97 98
c=IN IP4 192.0.2.100 203.0.113.100
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:57017fee-b6c1-4162-929c-a25110252400
a=ice-ufrag:ATEn
a=ice-pwd:AtSK0WpNtpUjkY4+86js7ZQl
a=msid:47017fee-b6c1-4162-929c-a25110252400
a=ice-ufrag:ETEn
a=ice-pwd:OtSK0WpNtpUjkY4+86js7ZQl
a=fingerprint:sha-256
              29:E2:1C:3B:4B:9F:81:E6:B8:5C:F4:A5:A8:D8:73:04:
              19:E2:1C:3B:4B:9F:81:E6:B8:5C:F4:A5:A8:D8:73:04:
              BB:05:2F:70:9F:04:A9:0E:05:E9:26:33:E8:70:88:A2
a=setup:passive
a=tls-id:17f0f4ba8a5f1213faca591b58ba52a7
a=setup:actpass
a=tls-id:91bbf309c0990a6bec11e38ba2933cee
a=rtcp:10101 IN IP4 203.0.113.100
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 2 udp 1845494015 198.51.100.100 11100 typ srflx
            raddr 2113929470 203.0.113.100 rport 10100
a=candidate:1 1 udp 255 192.0.2.100 12100 10101 typ relay
            raddr 198.51.100.100 rport 11100 host
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 10102 UDP/TLS/RTP/SAVPF 100 101 102 103
c=IN IP4 192.0.2.100 203.0.113.100
a=mid:v1
a=recvonly
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
=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
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 ]]></sourcecode>

<!-- [rfced] Sections 7.1, 7.2, and 7.3:  Please confirm that the
"=" signs in these ten "=rtpmap:103" entries do not need to be
preceded by an "a" line identifier.  We ask because these are the
only "=rtpmap:" SDP lines that begin with "=" instead of "a=".

Example from original:
 =rtpmap:103 rtx/90000 -->

	   <t>The SDP for |answer-A1| looks like:</t>
	   <sourcecode name="answer-A1" type="sdp"><![CDATA[
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 12100 10200 UDP/TLS/RTP/SAVPF 100 101 102 103
c=IN IP4 192.0.2.100
a=mid:v2
a=recvonly 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
=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
]]>
</artwork>
          </figure>
        </t>
a=msid:61317484-2ed4-49d7-9eb7-1414322a7aae ]]></sourcecode>
	 </section>
	 <section title="Early Transport Warmup Example"
      anchor="sec.warmup-example"> anchor="sec.detailed-example" numbered="true" toc="default">
	   <name>Detailed Example</name>
	   <t>This section shows a more involved example demonstrates 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 early warmup technique
        described in
        <xref target="sec.use-of-provisional-answer" />. Here, Alice's
        endpoint sends 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.</t>

	   <sourcecode name="" type="pseudocode"><![CDATA[
//                  set up local media state
AliceJS->AliceUA:   create new PeerConnection
AliceJS->AliceUA:   addTrack with an offer audio track
AliceJS->AliceUA:   createDataChannel to Bob's endpoint get data channel
AliceJS->AliceUA:   createOffer to get |offer-B1|
AliceJS->AliceUA:   setLocalDescription with |offer-B1|

//                  |offer-B1| is sent over signaling protocol to start an
        audio/video call. Bob immediately responds
AliceJS->WebServer: signaling with an answer that |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 the audio/video m= sections, but marks them as sendonly
        (from his perspective), meaning that 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 will not yet send
        media. This allows the JSEP implementation
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 start negotiating
        ICE and DTLS immediately. Bob's endpoint then prompts him Alice
AliceUA->BobUA:     audio+data sent from Alice to
        answer the call, and when he does, his endpoint sends a second
        offer which enables the audio and Bob

//                  some time later, Bob adds two video m= sections, and
        thereby bidirectional media transmission. The advantage streams
//                  note: no candidates exchanged, because of such
        a flow is that as soon as the bundle
BobJS->BobUA:       addTrack with first answer is received, the
        implementation can proceed video stream
BobJS->BobUA:       addTrack 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 immediately 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.</t>

        <t>This example also makes use of the "relay" ICE candidate
        policy described in
        <xref target="sec.ice-candidate-policy" /> to minimize the ICE
        gathering and checking needed.</t>

        <t>
          <figure>
            <artwork>
              <![CDATA[ video stream
BobJS->BobUA:       createOffer to get |offer-B2|
BobJS->BobUA:       setLocalDescription with |offer-B2|

//                  set up local media state
AliceJS->AliceUA:   create new PeerConnection                  |offer-B2| is sent to Alice
BobJS->WebServer:   signaling with "relay" ICE policy |offer-B2|
WebServer->AliceJS: signaling with |offer-B2|
AliceJS->AliceUA:   addTrack   setRemoteDescription with two tracks: audio and |offer-B2|
AliceUA->AliceJS:   ontrack event with first video track
AliceUA->AliceJS:   ontrack event with second video track
AliceJS->AliceUA:   createOffer   createAnswer to get |offer-C1| |answer-B2|
AliceJS->AliceUA:   setLocalDescription with |offer-C1| |answer-B2|

//                  |offer-C1|                  |answer-B2| is sent over signaling protocol
//                  to Bob
AliceJS->WebServer: signaling with |offer-C1| |answer-B2|
WebServer->BobJS:   signaling with |offer-C1|

//                  |offer-C1| arrives at Bob
BobJS->BobUA:       create new PeerConnection with "relay" ICE policy |answer-B2|
BobJS->BobUA:       setRemoteDescription with |offer-C1| |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 ]]></sourcecode>

<!-- [rfced] Section 7.2:  We changed "ontrack with" to "ontrack event
with" per the other three "ontrack event with" items.  Please let us
know if this is incorrect.

Original:
 BobUA->BobJS:       ontrack events for with audio and video

//                  a track from Alice

Currently:
 BobUA->BobJS:       ontrack event with audio track from Alice -->

	   <t>The SDP for |offer-B1| looks like:</t>
	   <sourcecode name="offer-B1" type="sdp"><![CDATA[
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 ]]></sourcecode>

	   <t>|offer-B1-candidate-1| looks like:</t>

	   <sourcecode name="offer-B1-candidate-1" type="sdp"><![CDATA[
ufrag ATEn
index 0
mid   a1
attr  candidate:1 1 udp 2113929471 203.0.113.100 10100 typ host ]]></sourcecode>
	   <t>|offer-B1-candidate-2| looks like:</t>
	   <sourcecode name="offer-B1-candidate-2" type="sdp"><![CDATA[
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 ]]></sourcecode>
	   <t>|offer-B1-candidate-3| looks like:</t>
	   <sourcecode name="offer-B1-candidate-3" type="sdp"><![CDATA[
ufrag ATEn
index 0
mid   a1
attr  candidate:1 1 udp 255 192.0.2.100 12100 typ 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 warmup 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
                raddr 198.51.100.100 rport 11100 ]]></sourcecode>
	   <t>The SDP 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 |answer-B1| looks like:</t>
	   <sourcecode name="answer-B1" type="sdp"><![CDATA[
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 ]]></sourcecode>

	   <t>|answer-B1-candidate-1| looks like:</t>

	   <sourcecode name="answer-B1-candidate-1" type="sdp"><![CDATA[
ufrag 7sFv
index 0
mid   a1
attr  candidate:1 1 udp 2113929471 203.0.113.200 10200 typ host ]]></sourcecode>
	   <t>|answer-B1-candidate-2| looks like:</t>
	   <sourcecode name="answer-B1-candidate-2" type="sdp"><![CDATA[
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 ]]></sourcecode>
	   <t>|answer-B1-candidate-3| looks like:</t>
	   <sourcecode name="answer-B1-candidate-3" type="sdp"><![CDATA[
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 ]]></sourcecode>
	   <t>The SDP for |offer-B2| is ringing

//                  Bob accepts call, starts media, and sends shown below. In addition to the
	   new offer
BobJS->BobUA:       transceiver.setTrack with audio "m=" sections for video, both of which are offering FEC and video tracks
BobUA->AliceUA:     media sent from Bob
	   one of which is offering simulcast, note the increment of the
	   version number in the "o=" line; changes to Alice
BobJS->BobUA:       transceiver.setDirection(sendrecv) the "c=" line,
	   indicating the local candidate that was selected; and the
	   inclusion of gathered candidates as a=candidate lines.</t>
	   <sourcecode name="offer-B2" type="sdp"><![CDATA[
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
=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
=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 ]]></sourcecode>
	   <t>The SDP for both
                    transceivers
BobJS->BobUA:       createOffer
BobJS->BobUA:       setLocalDescription with offer

//                  |offer-C2| |answer-B2| is sent over signaling protocol shown below. In addition 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 the
	   acceptance of the video "m=" sections, the use of a=recvonly to Bob

//                  |answer-C2| is sent over signaling protocol
	   indicate one-way video, and the use of a=imageattr to Bob
AliceJS->WebServer: signaling with |answer-C2|
WebServer->BobJS:   signaling with |answer-C2|
BobJS->BobUA:       setRemoteDescription with |answer-C2|
]]>
</artwork>
          </figure>
        </t>

        <t>The SDP for |offer-C1| looks like:</t>

        <t>
          <figure>
            <artwork alt="offer-C1">
              <![CDATA[ limit the
	   received resolution, note the use of setup:passive to maintain
	   the existing DTLS roles.</t>
	   <sourcecode name="answer-B2" type="sdp"><![CDATA[
v=0
o=- 1070771854436052752 1 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 9 12100 UDP/TLS/RTP/SAVPF 96 0 8 97 98
c=IN IP4 0.0.0.0 192.0.2.100
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=msid:57017fee-b6c1-4162-929c-a25110252400
a=ice-ufrag:ATEn
a=ice-pwd:AtSK0WpNtpUjkY4+86js7ZQl
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
              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 0 12100 UDP/TLS/RTP/SAVPF 100 101 102 103
c=IN IP4 0.0.0.0 192.0.2.100
a=mid:v1
a=sendrecv
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
=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
=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
a=msid:bbce3ba6-abfc-ac63-d00a-e15b286f8fce
a=bundle-only
]]>
</artwork>
          </figure>
        </t>

        <t>|offer-C1-candidate-1| looks like:</t>

        <t>
          <figure>
            <artwork alt="offer-C1-candidate-1">
              <![CDATA[
ufrag 4ZcD
index 0
mid   a1
attr  candidate:1 1 udp 255 192.0.2.100 12100 typ ]]></sourcecode>
	 </section>
	 <section anchor="sec.warmup-example" numbered="true" toc="default">
	   <name>Early Transport Warmup Example</name>
	   <t>This example demonstrates the early-warmup technique
	   described in
	   <xref target="sec.use-of-provisional-answer" format="default"/>. 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.</t>
	   <t>This example also makes use of the "relay" ICE candidate
	   policy described in
	   <xref target="sec.ice-candidate-policy" format="default"/> to minimize the ICE
	   gathering and checking needed.</t>

	   <sourcecode name="" type="pseudocode"><![CDATA[
//                  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
                raddr 0.0.0.0 rport 0
]]>
</artwork>
          </figure>
        </t>

        <t>The SDP 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| looks like:</t>

        <t>
          <figure>
            <artwork alt="answer-C1">
              <![CDATA[
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
=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
]]>
</artwork>
          </figure>
        </t>

        <t>|answer-C1-candidate-1| looks like:</t>

        <t>
          <figure>
            <artwork alt="answer-C1-candidate-1">
              <![CDATA[
ufrag TpaA
index 0
mid   a1
attr  candidate:1 1 udp 255 192.0.2.200 12200 typ 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
                raddr 0.0.0.0 rport 0
]]>
</artwork>
          </figure>
        </t>

        <t>The SDP 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| looks like:</t>

        <t>
          <figure>
            <artwork alt="offer-C2">
              <![CDATA[
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
=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
]]>
</artwork>
          </figure>
        </t> 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| ]]></sourcecode>
	   <t>The SDP for |answer-C2| |offer-C1| looks like:</t>

        <t>
          <figure>
            <artwork alt="answer-C2">
              <![CDATA[
	   <sourcecode name="offer-C1" type="sdp"><![CDATA[
v=0
o=- 1070771854436052752 2 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 12100 9 UDP/TLS/RTP/SAVPF 96 0 8 97 98
c=IN IP4 192.0.2.100 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:passive
a=setup:actpass
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 0 UDP/TLS/RTP/SAVPF 100 101 102 103
c=IN IP4 192.0.2.100 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
=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
]]>
</artwork>
          </figure>
        </t>
      </section>
    </section>
    <section title="Security Considerations"
    anchor="sec.security-considerations">
a=bundle-only ]]></sourcecode>
	   <t>|offer-C1-candidate-1| looks like:</t>
	   <sourcecode name="offer-C1-candidate-1" type="sdp"><![CDATA[
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 ]]></sourcecode>
	   <t>The IETF has published separate documents
      <xref target="I-D.ietf-rtcweb-security-arch" />
      <xref target="I-D.ietf-rtcweb-security" /> describing the security
      architecture for WebRTC as a whole. The remainder of this section
      describes security considerations for this document.</t>

      <t>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
      <xref target="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 which is passed to setLocalDescription(). While correct
      API usage requires that the application pass in SDP which was
      derived from createOffer() or createAnswer(), there is no
      guarantee that applications do so. The JSEP implementation MUST
      be prepared for the JavaScript to pass in bogus data instead.</t>

      <t>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 |answer-C1| looks like:</t>
	   <sourcecode name="answer-C1" type="sdp"><![CDATA[
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
=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 ]]></sourcecode>
	   <t>|answer-C1-candidate-1| looks like:</t>
	   <sourcecode name="answer-C1-candidate-1" type="sdp"><![CDATA[
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 ]]></sourcecode>
	   <t>The SDP or suppressing trickled candidates
      does not have the expected behavior: implementations will still
      perform checks from those candidates even if they are not sent to
      the other side. Thus, for instance, it is not possible to prevent
      the remote peer from learning your public IP address by removing
      server reflexive candidates. Applications which wish to conceal
      their public IP address should instead configure the ICE agent to
      use only |offer-C2| looks like:</t>
	   <sourcecode name="offer-C2" type="sdp"><![CDATA[
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 candidates.</t>
    </section>
    <section title="IANA Considerations"
    anchor="sec.iana-considerations">

      <t>This document requires no actions from IANA.</t>
    </section>
    <section title="Acknowledgements" anchor="sec.acknowledgements">

      <t>Harald Alvestrand, Taylor Brandstetter, Suhas Nandakumar, and
      Peter Thatcher provided significant text
            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
=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 ]]></sourcecode>
	   <t>The SDP for this draft. Bernard
      Aboba, Adam Bergkvist, Dan Burnett, Ben Campbell, Alissa Cooper,
      Richard Ejzak, Stefan Hakansson, 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 proposal.</t>
    </section>
  </middle>
  <back>
    <references title="Normative References">
      <?rfc include='reference.I-D.ietf-avtext-rid'?>
      <?rfc include='reference.I-D.ietf-ice-trickle'?>
      <?rfc include='reference.I-D.ietf-mmusic-dtls-sdp'?>
      <?rfc include='reference.I-D.ietf-mmusic-ice-sip-sdp'?>
      <?rfc include='reference.I-D.ietf-mmusic-msid'?>
      <?rfc include='reference.I-D.ietf-mmusic-mux-exclusive'?>
      <?rfc include='reference.I-D.ietf-mmusic-rid'?>
      <?rfc include='reference.I-D.ietf-mmusic-sctp-sdp'?>
      <?rfc include='reference.I-D.ietf-mmusic-sdp-bundle-negotiation'?>
      <?rfc include='reference.I-D.ietf-mmusic-sdp-mux-attributes'?>
      <?rfc include='reference.I-D.ietf-mmusic-sdp-simulcast'?>
      <?rfc include='reference.I-D.ietf-rtcweb-fec'?>
      <?rfc include='reference.I-D.ietf-rtcweb-rtp-usage'?>
      <?rfc include='reference.I-D.ietf-rtcweb-security'?>
      <?rfc include='reference.I-D.ietf-rtcweb-security-arch'?>
      <?rfc include='reference.RFC.2119.xml'?>
      <?rfc include='reference.RFC.3261.xml'?>
      <?rfc include='reference.RFC.3264.xml'?>
      <?rfc include='reference.RFC.3552.xml'?>
      <?rfc include='reference.RFC.3605.xml'?>
      <?rfc include='reference.RFC.3890.xml'?>
      <?rfc include='reference.RFC.4145.xml'?>
      <?rfc include='reference.RFC.4566.xml'?>
      <?rfc include='reference.RFC.4585.xml'?>
      <?rfc include='reference.RFC.5124.xml'?>
      <?rfc include='reference.RFC.5285.xml'?>
      <?rfc include='reference.RFC.5761.xml'?>
      <?rfc include='reference.RFC.5888.xml'?>
      <?rfc include='reference.RFC.6236.xml'?>
      <?rfc include='reference.RFC.6347.xml'?>
      <?rfc include='reference.RFC.6716.xml'?>
      <?rfc include='reference.RFC.6904.xml'?>
      <?rfc include='reference.RFC.7160.xml'?>
      <?rfc include='reference.RFC.7587.xml'?>
      <?rfc include='reference.RFC.7742.xml'?>
      <?rfc include='reference.RFC.7850.xml'?>
      <?rfc include='reference.RFC.7874.xml'?>
      <?rfc include='reference.RFC.8108.xml'?>
      <?rfc include='reference.RFC.8122.xml'?>
      <?rfc include='reference.RFC.8445.xml'?>
      <?rfc include='reference.RFC.3711.xml'?>
    </references>
    <references title="Informative References">
      <?rfc include='reference.I-D.ietf-rtcweb-ip-handling'?>
      <?rfc include='reference.I-D.ietf-mmusic-trickle-ice-sip'?>
      <?rfc include='reference.I-D.ietf-rtcweb-sdp'?>
      <?rfc include='reference.RFC.3389.xml'?>
      <?rfc include='reference.RFC.3960.xml'?>
      <?rfc include='reference.RFC.4568.xml'?>
      <?rfc include='reference.RFC.4588.xml'?>
      <?rfc include='reference.RFC.4733.xml'?>
      <?rfc include='reference.RFC.5245.xml'?>
      <?rfc include='reference.RFC.5506.xml'?>
      <?rfc include='reference.RFC.5576.xml'?>
      <?rfc include='reference.RFC.5763.xml'?>
      <?rfc include='reference.RFC.5764.xml'?>
      <?rfc include='reference.RFC.6464.xml'?>
      <?rfc include='reference.RFC.6544.xml'?>
      <?rfc include='reference.RFC.3556.xml'?>
      <reference anchor="W3C.webrtc"
      target="https://www.w3.org/TR/2017/WD-webrtc-20170515/">
        <front>
          <title>WebRTC 1.0: Real-time Communication Between
          Browsers</title>
          <author fullname="Adam Bergkvist" initials="A."
          surname="Bergkvist">
            <organization>Ericsson</organization>
          </author>
          <author fullname="Daniel C. Burnett" initials="D."
          surname="Burnett">
            <organization></organization>
          </author>
          <author fullname="Cullen Jennings" initials="C."
          surname="Jennings">
            <organization>Cisco</organization>
          </author>
          <author fullname="Anant Narayanan" initials="A."
          surname="Narayanan">
            <organization>Mozilla</organization>
          </author>
          <author fullname="Bernard Aboba" initials="B."
          surname="Aboba">
            <organization>Microsoft Corporation</organization>
          </author>
          <author fullname="Taylor Brandstetter" initials="T."
          surname="Brandstetter">
            <organization>Google</organization>
          </author>
          <date day="15" month="May" year="2017" />
        </front>
        <seriesInfo name="World Wide Web Consortium WD"
        value="WD-webrtc-20170515" />
        <format target="https://www.w3.org/TR/2017/WD-webrtc-20170515/"
        type="HTML" />
      </reference>
      <reference anchor="TS26.114"
      target="http://www.3gpp.org/DynaReport/26114.htm">
        <front>
          <title>3rd Generation Partnership Project; Technical
          Specification Group Services and System Aspects; IP
          Multimedia Subsystem (IMS); Multimedia Telephony; Media
          handling and interaction (Release 12)</title>
          <author>
            <organization>3GPP TS 26.114 V12.8.0</organization>
          </author>
          <date year="2014" month="December" />
        </front>
      </reference>
    </references> |answer-C2| looks like:</t>
	   <sourcecode name="answer-C2" type="sdp"><![CDATA[
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
=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 ]]></sourcecode>
	 </section>
       </section>
       <section title="Appendix A" anchor="sec.appendix-a">

      <t>For the syntax validation performed in anchor="sec.security-considerations" numbered="true" toc="default">
	 <name>Security Considerations</name>
	 <t>The IETF has published separate documents
	 <xref target="sec.parsing-a-desc" />, target="RFC8827" format="default"/>
	   <xref target="RFC8826" format="default"/> describing the following list security
	 architecture for WebRTC as a whole. The remainder of ABNF
      definitions is used:</t>
      <texttable anchor="sdp-abnf" title="SDP ABNF References">
        <ttcol align='left'>Attribute</ttcol>
        <ttcol align='left'>Reference</ttcol>
        <c>ptime</c>
        <c>
        <xref target="RFC4566" /> Section 9</c>
        <c>maxptime</c>
        <c>
        <xref target="RFC4566" /> Section 9</c>
        <c>rtpmap</c>
        <c>
        <xref target="RFC4566" /> Section 9</c>
        <c>recvonly</c>
        <c>
        <xref target="RFC4566" /> Section 9</c>
        <c>sendrecv</c>
        <c>
        <xref target="RFC4566" /> Section 9</c>
        <c>sendonly</c>
        <c>
        <xref target="RFC4566" /> Section 9</c>
        <c>inactive</c>
        <c>
        <xref target="RFC4566" /> Section 9</c>
        <c>framerate</c>
        <c>
        <xref target="RFC4566" /> Section 9</c>
        <c>fmtp</c>
        <c>
        <xref target="RFC4566" /> Section 9</c>
        <c>quality</c>
        <c>
        <xref target="RFC4566" /> Section 9</c>
        <c>rtcp</c>
        <c>
        <xref target="RFC3605" /> Section 2.1</c>
        <c>setup</c>
        <c>
        <xref target="RFC4145" /> Sections 3, 4, and 5</c>
        <c>connection</c>
        <c>
        <xref target="RFC4145" /> Sections 3, 4, and 5</c>
        <c>fingerprint</c>
        <c>
        <xref target="RFC8122" /> Section 5</c>
        <c>rtcp-fb</c>
        <c>
        <xref target="RFC4585" /> Section 4.2</c>
        <c>extmap</c>
        <c>
        <xref target="RFC5285" /> Section 7</c>
        <c>mid</c>
        <c>
        <xref target="RFC5888" /> Sections 4 and 5</c>
        <c>group</c>
        <c>
        <xref target="RFC5888" /> Sections 4 and 5</c>
        <c>imageattr</c>
        <c>
        <xref target="RFC6236" /> Section 3.1</c>
        <c>extmap (encrypt option)</c>
        <c>
        <xref target="RFC6904" /> Section 4</c>
        <c>candidate</c>
        <c>
        <xref target="I-D.ietf-mmusic-ice-sip-sdp" /> Section 4.1</c>
        <c>remote-candidates</c>
        <c>
        <xref target="I-D.ietf-mmusic-ice-sip-sdp" /> Section 4.2</c>
        <c>ice-lite</c>
        <c>
        <xref target="I-D.ietf-mmusic-ice-sip-sdp" /> Section 4.3</c>
        <c>ice-ufrag</c>
        <c>
        <xref target="I-D.ietf-mmusic-ice-sip-sdp" /> Section 4.4</c>
        <c>ice-pwd</c>
        <c>
        <xref target="I-D.ietf-mmusic-ice-sip-sdp" /> Section 4.4</c>
        <c>ice-options</c>
        <c>
        <xref target="I-D.ietf-mmusic-ice-sip-sdp" /> Section 4.6</c>
        <c>msid</c>
        <c>
        <xref target="I-D.ietf-mmusic-msid" /> Section 2</c>
        <c>rid</c>
        <c>
        <xref target="I-D.ietf-mmusic-rid" /> Section 10</c>
        <c>simulcast</c>
        <c>
        <xref target="I-D.ietf-mmusic-sdp-simulcast" /> Section 6.1</c>
        <c>tls-id</c>
        <c>
        <xref target="I-D.ietf-mmusic-dtls-sdp" /> Section 4</c>
      </texttable>
    </section>
    <section title="Change log" anchor="sec.change-log">

      <t>Note to RFC Editor: Please remove this section before
      publication.</t>

      <t>Changes in draft-26:</t>

      <t>
        <list style="symbols">
          <t>Update guidance on generation of
	 describes security considerations for this document.</t>
	 <t>While formally the m= proto value JSEP interface is an API, it is better to be
          consistent
	 think of it as an Internet protocol, with ice-sip-sdp.</t>
        </list>
      </t>

      <t>Changes in draft-25:</t>

      <t>
        <list style="symbols">

          <t>Remove MSID track ID the application
	 JavaScript being untrustworthy from offers and answers.</t>

          <t>Add note about rejecting all m= sections the perspective of the JSEP
	 implementation. Thus, the threat model of
	 <xref target="RFC3552" format="default"/> applies. In particular, JavaScript can
	 call the API in a BUNDLE group.</t>

          <t>Update ICE references to RFC 8445 any order and mention ice2.</t>
        </list>
      </t>

      <t>Changes in draft-24:</t>

      <t>
        <list style="symbols">

          <t>Clarify that rounding with any inputs, including
	 malicious ones. This is permitted particularly relevant when trying to maintain
          aspect ratio.</t>

          <t>Update tls-id handling to match what we consider
	 the SDP that is specified in
          dtls-sdp.</t>
        </list>
      </t>

      <t>Changes in draft-23:</t>

      <t>
        <list style="symbols">

          <t>Clarify rollback handling, and treat it similarly passed to other
          setLocal/setRemote usages.</t>

          <t>Adopt a first-fit policy for handling multiple remote
          a=imageattr attributes.</t>

          <t>Clarify setLocalDescription(). While correct
	 API usage requires that a session description with zero m= sections
          is legal.</t>
        </list>
      </t>

      <t>Changes the application pass in draft-22:</t>

      <t>
        <list style="symbols">

          <t>Clarify currentDirection versus direction.</t>

          <t>Correct session-id text so that it aligns with RFC
          3264.</t>

          <t>Clarify SDP that generated ICE candidate objects must have all
          four fields.</t>

          <t>Make rollback work was
	 derived from any state besides stable and
          regardless of whether setLocalDescription createOffer() or
          setRemoteDescription createAnswer(), there is used.</t>

          <t>Allow modifying SDP before sending or after receiving
          either offers or answers (previously this was forbidden for
          answers).</t>

          <t>Provide rationale for several design choices.</t>
        </list>
      </t>

      <t>Changes in draft-21:</t>

      <t>
        <list style="symbols">

          <t>Change dtls-id to tls-id to match MMUSIC draft.</t>

          <t>Replace regular expression for proto field with a list and
          clarify no
	 guarantee that applications do so. The JSEP implementation <bcp14>MUST</bcp14>
	 be prepared for the answer must exactly match JavaScript to pass in bogus data instead.</t>
	 <t>Conversely, the offer.</t>

          <t>Remove text about how application programmer needs to error check on setLocal because
          local descriptions cannot be changed.</t>

          <t>Rework silence suppression support to always require aware that
          both sides agree to silence suppression
	 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 none 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 used.</t>

          <t>Remove instructions not possible to parse "a=ssrc-group".</t>

          <t>Allow prevent
	 the addition of new codecs in answers and in
          subsequent offers.</t>

          <t>Clarify imageattr processing. Replace remote peer from learning your public IP address by removing
	 server-reflexive candidates. Applications that wish to conceal
	 their public IP address should instead configure the ICE agent to
	 use only relay candidates.</t>
       </section>
       <section anchor="sec.iana-considerations" numbered="true" toc="default">
	 <name>IANA Considerations</name>
	 <t>This document has no IANA actions.</t>
       </section>
     </middle>
     <back>
<!-- draft-ietf-rtcweb-sdp ("Publication Requested") -->
   <displayreference target="I-D.ietf-rtcweb-sdp" to="SDP4WebRTC"/>
    <references>
	 <name>References</name>
	 <references>
	   <name>Normative References</name>

<!--draft-ietf-mmusic-trickle-ice-sip-18: 8840  -->
	   <reference anchor="RFC8840" target="https://www.rfc-editor.org/info/rfc8840">
	     <front>
	       <title>A Session Initiation Protocol (SIP) Usage for Incremental
	       Provisioning of [x=0,y=0]
          with direction indicators.</t>

          <t>Document when early media can occur.</t>

          <t>Fix ICE default port handling when bundle-only is
          used.</t>

          <t>Forbid duplicating IDENTICAL/TRANSPORT attributes when you
          are bundling.</t>

          <t>Clarify Candidates for the number Interactive Connectivity
	       Establishment (Trickle ICE)</title>

	       <author initials="E" surname="Ivov" fullname="Emil Ivov">
		 <organization/>
	       </author>
	       <author initials="T" surname="Stach" fullname="Thomas Stach">
		 <organization/>
	       </author>
	       <author initials="E" surname="Marocco" fullname="Enrico Marocco">
		 <organization/>
	       </author>
	       <author initials="C" surname="Holmberg" fullname="Christer Holmberg">
		 <organization/>
	       </author>
	       <date month="July" year="2018"/>
	     </front>
	     <seriesInfo name="DOI" value="10.17487/RFC8840"/>
	     <seriesInfo name="RFC" value="8840"/>
	   </reference>

<!-- draft-ietf-avtext-rid-09; 8852 -->
<reference anchor="RFC8852" target="https://www.rfc-editor.org/info/rfc8852">
  <front>
    <title>RTP Stream Identifier Source Description (SDES)</title>
    <author initials="A.B." surname="Roach" fullname="Adam Roach"/>
    <author initials="S" surname="Nandakumar" fullname="Suhas Nandakumar"/>
    <author initials="P" surname="Thatcher" fullname="Peter Thatcher"/>
    <date month="July" year="2020"/>
  </front>
    <seriesInfo name="DOI" value="10.17487/RFC8852"/>
    <seriesInfo name="RFC" value="8852"/>
</reference>

<!-- draft-ietf-ice-trickle: RFC 8838 -->
<reference anchor="RFC8838" target="https://www.rfc-editor.org/info/rfc8838">
  <front>
    <title>Trickle ICE: Incremental Provisioning of components to gather when bundle is
          involved.</t>

          <t>Explicitly state that PTs and SSRCs are to be used Candidates for
          demuxing.</t>

          <t>Update guidance on "a=setup" line. This should now match the MMUSIC draft.</t>

          <t>Update guidance on certificate/digest matching to conform
          to RFC8122.</t>

          <t>Update examples.</t>
        </list>
      </t>

      <t>Changes in draft-20:</t>

      <t>
        <list style="symbols">

          <t>Remove Appendix-B.</t>
        </list>
      </t>

      <t>Changes in draft-19:</t>

      <t>
        <list style="symbols">

          <t>Examples are now machine-generated
    Interactive Connectivity Establishment (ICE) Protocol</title>

    <author initials="E" surname="Ivov" fullname="Emil Ivov">
	 <organization />
    </author>

    <author initials="J" surname="Uberti" fullname="Justin Uberti">
	 <organization />
    </author>

    <author initials="P" surname="Saint-Andre" fullname="Peter Saint-Andre">
	 <organization />
    </author>

    <date month="July" year="2020" />
  </front>
  <seriesInfo name="RFC" value="8838" />
  <seriesInfo name="DOI" value="10.17487/RFC8838"/>
</reference>

<!-- draft-ietf-mmusic-dtls-sdp   RFC-to-be 8842 -->
<reference anchor="RFC8842" target="https://www.rfc-editor.org/info/rfc8842">

  <front>
    <title>Session Description Protocol (SDP) Offer/Answer Considerations for correctness, and
          use IETF-approved example IP addresses.</t>

          <t>Add early transport warmup example, and add missing
          attributes to existing examples.</t>

          <t>Only send "a=rtcp-mux-only"
Datagram Transport Layer Security (DTLS) and "a=bundle-only" on new m=
          sections.</t>

          <t>Update references.</t>

          <t>Add coverage Transport Layer Security (TLS)</title>

    <author initials="C." surname="Holmberg" fullname="Christer Holmberg">
	<organization />
    </author>

    <author initials="R." surname="Shpount" fullname="Roman Shpount">
	<organization />
    </author>

    <date month="July" year="2020" />
  </front>
  <seriesInfo name="RFC" value="8842" />
  <seriesInfo name="DOI" value="10.17487/RFC8842"/>

</reference>

<!-- draft-ietf-mmusic-ice-sip-sdp-39  RFC-to-be 8839 -->

<reference anchor='RFC8839' target="https://www.rfc-editor.org/info/rfc8839">
<front>
<title>Session Description Protocol (SDP) Offer/Answer Procedures for Interactive Connectivity Establishment (ICE)</title>

<author initials='M' surname='Petit-Huguenin' fullname='Marc Petit-Huguenin'>
    <organization />
</author>

<author initials='S' surname='Nandakumar' fullname='Suhas Nandakumar'>
    <organization />
</author>

<author initials='C' surname='Holmberg' fullname='Christer Holmberg'>
    <organization />
</author>

<author initials='A' surname='Keränen' fullname='Ari Keränen'>
    <organization />
</author>

<author initials='R' surname='Shpount' fullname='Roman Shpount'>
    <organization />
</author>

<date month="July" year="2020"/>

</front>
<seriesInfo name="RFC" value="8839"/>
<seriesInfo name="DOI" value="10.17487/RFC8839"/>

</reference>

<!-- draft-ietf-mmusic-msid: 8830 -->
<reference anchor="RFC8830" target="https://www.rfc-editor.org/info/rfc8830">
  <front>
    <title>WebRTC MediaStream Identification in the Session Description Protocol</title>
    <author initials="H" surname="Alvestrand" fullname="Harald Alvestrand">
      <organization />
    </author>
    <date month="July" year="2020" />
  </front>
  <seriesInfo name="RFC" value="8830" />
  <seriesInfo name="DOI" value="10.17487/RFC8830"/>
</reference>

<!--draft-ietf-mmusic-mux-exclusive-12; part of a=identity.</t>

          <t>Explain the lipsync group algorithm more thoroughly.</t>

          <t>Remove unnecessary list C238; RFC 8858-->
<reference anchor='RFC8858' target="https://www.rfc-editor.org/info/rfc8858">
<front>
<title>Indicating Exclusive Support of MTI specs.</t>

          <t>Allow codecs which weren't offered to appear in answers RTP and which weren't selected to appear RTP Control Protocol (RTCP)
       Multiplexing Using the Session Description Protocol (SDP)</title>
<author initials='C.' surname='Holmberg' fullname='Christer Holmberg'>
    <organization />
</author>

<date month="July" year='2020' />
</front>
<seriesInfo name='RFC' value='8858' />
<seriesInfo name="DOI" value="10.17487/RFC8858"/>
</reference>

<!-- draft-ietf-mmusic-rid: 8851 -->
<reference anchor="RFC8851" target="https://www.rfc-editor.org/info/rfc8851">
  <front>
    <title>RTP Payload Format Restrictions</title>
    <author initials="A.B." surname="Roach" fullname="Adam Roach" role="editor">
      <organization/>
    </author>
    <date month="April" year="2020"/>
  </front>
    <seriesInfo name="DOI" value="10.17487/RFC8851"/>
    <seriesInfo name="RFC" value="8851"/>
</reference>

<!-- draft-ietf-mmusic-sctp-sdp: 8841 -->

<reference anchor="RFC8841" target="https://www.rfc-editor.org/info/rfc8841">

  <front>
    <title>Session Description Protocol (SDP) Offer/Answer Procedures for
    Stream Control Transmission Protocol (SCTP) over Datagram Transport Layer
    Security (DTLS) Transport</title>

    <author initials="C." surname="Holmberg" fullname="Christer Holmberg">
      <organization />
    </author>

    <author initials="R." surname="Shpount" fullname="Roman Shpount">
      <organization />
    </author>

    <author initials="S." surname="Loreto" fullname="Salvatore Loreto">
      <organization />
    </author>

    <author initials="G." surname="Camarillo" fullname="Gonzalo Camarillo">
      <organization />
    </author>

    <date month="July" year="2020" />
  </front>
  <seriesInfo name="RFC" value="8841" />
  <seriesInfo name="DOI" value="10.17487/RFC8841"/>

</reference>

<!-- draft-ietf-mmusic-sdp-bundle-negotiation (RFC 8843) -->
    <reference anchor="RFC8843" target="https://www.rfc-editor.org/info/rfc8843">
      <front>
        <title>Negotiating Media Multiplexing Using the Session Description Protocol (SDP)</title>
        <author initials="C" surname="Holmberg" fullname="Christer Holmberg">
          <organization/>
        </author>
        <author initials="H" surname="Alvestrand" fullname="Harald Alvestrand">
          <organization/>
        </author>
        <author initials="C" surname="Jennings" fullname="Cullen Jennings">
          <organization/>
        </author>
        <date month="July" year="2020"/>
      </front>
        <seriesInfo name="RFC" value="8843"/>
        <seriesInfo name="DOI" value="10.17487/RFC8843"/>
    </reference>

<!-- draft-ietf-mmusic-sdp-mux-attributes-17 (RFC 8859) -->
<reference anchor="RFC8859" target="https://www.rfc-editor.org/info/rfc8859">
  <front>
    <title>A Framework for Session Description Protocol (SDP)
    Attributes When Multiplexing</title>
    <author initials="S" surname="Nandakumar" fullname="Suhas Nandakumar">
      <organization/>
    </author>
    <date month="July" year="2020"/>
  </front>
    <seriesInfo name="DOI" value="10.17487/RFC8859"/>
    <seriesInfo name="RFC" value="8859"/>

</reference>

<!-- draft-ietf-mmusic-sdp-simulcast: 8853 -->
<reference anchor="RFC8853" target="https://www.rfc-editor.org/info/rfc8853">
  <front>
    <title>Using Simulcast in subsequent
          offers.</t>

          <t>Codec preferences now are applied on both initial Session Description Protocol (SDP) and
          subsequent offers RTP
    Sessions</title>

    <author initials="B" surname="Burman" fullname="Bo Burman">
      <organization/>
    </author>
    <author initials="M" surname="Westerlund" fullname="Magnus Westerlund">
      <organization/>
    </author>
    <author initials="S" surname="Nandakumar" fullname="Suhas Nandakumar">
      <organization/>
    </author>
    <author initials="M" surname="Zanaty" fullname="Mo Zanaty">
      <organization/>
    </author>
    <date month="July" year="2020"/>
  </front>
    <seriesInfo name="DOI" value="10.17487/RFC8853"/>
    <seriesInfo name="RFC" value="8853"/>
</reference>

<!-- draft-ietf-rtcweb-fec: 8854 -->
<reference anchor="RFC8854" target="https://www.rfc-editor.org/info/rfc8854">
  <front>
    <title>WebRTC Forward Error Correction Requirements</title>
    <author initials="J." surname="Uberti" fullname="Justin Uberti">
      <organization/>
    </author>
    <date month="July" year="2020"/>
  </front>
    <seriesInfo name="RFC" value="8854"/>
    <seriesInfo name="DOI" value="10.17487/RFC8854"/>
</reference>

<!-- draft-ietf-rtcweb-rtp-usage; RFC 8834 -->
<reference anchor="RFC8834" target="https://www.rfc-editor.org/info/rfc8834">
  <front>
    <title>Media Transport and answers.</t>

          <t>Clarify a=msid handling for recvonly m= sections.</t>

          <t>Clarify behavior Use of attributes for bundle-only data
          channels.</t>

          <t>Allow media attributes to appear RTP in data m= sections when
          all the media m= sections are bundle-only.</t>

          <t>Use consistent terminology WebRTC</title>
    <author initials="C." surname="Perkins" fullname="Colin Perkins">
      <organization />
    </author>
    <author initials="M." surname="Westerlund" fullname="Magnus Westerlund">
      <organization />
    </author>
    <author initials="J." surname="Ott" fullname="Jörg Ott">
      <organization />
    </author>
    <date month="July" year="2020" />
  </front>
  <seriesInfo name="RFC" value="8834" />
  <seriesInfo name="DOI" value="10.17487/RFC8834"/>
</reference>

<!--draft-ietf-rtcweb-security: RFC 8826 -->
<reference anchor="RFC8826" target="https://www.rfc-editor.org/info/rfc8826">
<front>
<title>Security Considerations for JSEP implementations.</t>

          <t>Describe how to handle failed API calls.</t>

          <t>Some cleanup on routing rules.</t>
        </list>
      </t>

      <t>Changes in draft-18:</t>

      <t>
        <list style="symbols">

          <t>Update demux algorithm and move it WebRTC</title>
<author initials='E.' surname='Rescorla' fullname='Eric Rescorla'>
  <organization/>
</author>
<date month='July' year='2020'/>
</front>
<seriesInfo name="RFC" value="8826"/>
<seriesInfo name="DOI" value="10.17487/RFC8826"/>
</reference>

<!--draft-ietf-rtcweb-security-arch: 8827 -->
<reference anchor="RFC8827" target="https://www.rfc-editor.org/info/rfc8827">
<front>
<title>WebRTC Security Architecture</title>
<author initials='E.' surname='Rescorla' fullname='Eric Rescorla'>
  <organization/>
</author>
<date month='May' year='2020'/>
</front>
<seriesInfo name="RFC" value="8827"/>
<seriesInfo name="DOI" value="10.17487/RFC8827"/>
</reference>

<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.2119.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.8174.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.3261.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.3264.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.3552.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.3605.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.3890.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.4145.xml"/>

<!-- [rfced] Throughout: RFC 4566, which will be obsoleted by RFC-to-be 8866
(soon to an appendix in
          preparation for merging it into BUNDLE.</t>

          <t>Clarify why we can't handle an incoming offer be released to send
          simulcast.</t>

          <t>Expand IceCandidate object text.</t>

          <t>Further document use AUTH48) as part of ICE candidate pool.</t>

          <t>Document removeTrack.</t>

          <t>Update requirements to only accept this cluster, is heavily cited in
this document.  Should the last generated
          offer/answer as an argument text instead refer to setLocalDescription.</t>

          <t>Allow round pixels.</t>

          <t>Fix code around default timing when AVPF RFC 8866?
-->

<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.4566.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.4585.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.5124.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.5285.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.5761.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.5888.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.6236.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.6347.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.6716.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.6904.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.7160.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.7587.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.7742.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.7850.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.7874.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.8108.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.8122.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.8445.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.3711.xml"/>
      </references>
      <references>
        <name>Informative References</name>

<!-- draft-ietf-rtcweb-ip-handling: 8828 -->
<reference anchor="RFC8828" target="https://www.rfc-editor.org/info/rfc8828">
  <front>
    <title>WebRTC IP Address Handling Requirements</title>
    <author initials="J" surname="Uberti" fullname="Justin Uberti">
      <organization />
    </author>

    <date month="June" year="2020" />
  </front>
  <seriesInfo name="RFC" value="8828" />
  <seriesInfo name="DOI" value="10.17487/RFC8828"/>
</reference>

<!-- 7/6/2020: IESG eval -->
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml3/reference.I-D.ietf-rtcweb-sdp.xml"/>

<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.3389.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.4568.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.4588.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.4733.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.5245.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.5506.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.5576.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.5763.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.5764.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.6120.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.6464.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.6544.xml"/>

<!-- [rfced] Informative References:  RFC 6544 is not
          specified.</t>

          <t>Clean up terminology around m= line cited anywhere
in this document.  Please let us know where it should be cited.

Original:
 [RFC6544]  Rosenberg, J., Keranen, A., Lowekamp, B., and m=section.</t>

          <t>Provide a more realistic example A. Roach,
            "TCP Candidates with Interactive Connectivity
            Establishment (ICE)", RFC 6544, DOI 10.17487/RFC6544,
            March 2012, <https://www.rfc-editor.org/info/rfc6544>. -->

<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.3556.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.3960.xml"/>

        <reference anchor="W3C.webrtc" target="https://www.w3.org/TR/2017/WD-webrtc-20170515/">
          <front>
            <title>WebRTC 1.0: Real-time Communication Between
          Browsers</title>
            <author fullname="Adam Bergkvist" initials="A." surname="Bergkvist">
              <organization>Ericsson</organization>
            </author>
            <author fullname="Daniel C. Burnett" initials="D." surname="Burnett">
              <organization/>
            </author>
            <author fullname="Cullen Jennings" initials="C." surname="Jennings">
              <organization>Cisco</organization>
            </author>
            <author fullname="Anant Narayanan" initials="A." surname="Narayanan">
              <organization>Mozilla</organization>
            </author>
            <author fullname="Bernard Aboba" initials="B." surname="Aboba">
              <organization>Microsoft Corporation</organization>
            </author>
            <author fullname="Taylor Brandstetter" initials="T." surname="Brandstetter">
              <organization>Google</organization>
            </author>
            <date month="May" year="2017"/>
          </front>
            <refcontent>World Wide Web Consortium WD WD-webrtc-20170515</refcontent>
        </reference>

<!-- [rfced] Informative References:  The provided URL for minimum decoder
          capabilities.</t>

          <t>Document behavior when rtcp-mux policy
[W3C.webrtc] steers to a page with a red window that says
"This version is require but
          rtcp-mux attribute not provided.</t>

          <t>Expanded discussion outdated!"  Because the latest version
(September 2018) also discusses the RTCPeerConnection interface,
may we update this listing as suggested below?

Also, if you agree to this update, should Jan-Ivar Bruaroey be
added to the second sentence of RtpSender the Acknowledgements section
(after Adam Bergkvist)?

Original:
 [W3C.webrtc]
            Bergkvist, A., Burnett, D., Jennings, C., Narayanan, A.,
            Aboba, B., and T. Brandstetter, "WebRTC 1.0: Real-time
            Communication Between Browsers", World Wide Web Consortium
            WD WD-webrtc-20170515, May 2017,
            <https://www.w3.org/TR/2017/WD-webrtc-20170515/>.

Suggested:
 [W3C.webrtc]
            Bergkvist, A., Burnett, D., Jennings, C., Narayanan, A.,
            Aboba, B., Brandstetter, T., and J-I. Bruaroey, "WebRTC
            1.0: Real-time Communication Between Browsers", World
            Wide Web Consortium Candidate Recommendation, September
            2018, <https://www.w3.org/TR/webrtc/>. -->

        <reference anchor="TS26.114" target="https://www.3gpp.org/DynaReport/26114.htm">
          <front>
            <title>3rd Generation Partnership Project; Technical
          Specification Group Services and RtpReceiver.</t>

          <t>Add RtpTransceiver.currentDirection System Aspects; IP
          Multimedia Subsystem (IMS); Multimedia Telephony; Media
          handling and document
          setDirection.</t>

          <t>Require imageattr x=0, y=0 to indicate interaction (Release 12)</title>
          <seriesInfo name="3GPP TS" value="26.114 V12.8.0"/>
            <author>
              <organization>3GPP</organization>
            </author>
            <date year="2014" month="December"/>
          </front>
        </reference>

<!-- [rfced] Informative References:  We see on
<https://www.3gpp.org/DynaReport/26114.htm> that there are no
          valid resolutions.</t>

          <t>Require a privacy-preserving MID/RID construction.</t>

          <t>Require support for RFC 3556 bandwidth modifiers.</t>

          <t>Update maxptime description.</t>

          <t>Note that endpoints may encounter extra codecs newer version
dated September 2019 is available.  The new version also discusses
CVO (as mentioned in answers Section 3.6.2 of this document).  May we update
this listing as suggested below?

Original:
 This is required regardless of whether the receiver
 supports performing receive-side rotation (e.g., through CVO
 [TS26.114]), as it significantly simplifies the matching logic.
...
 [TS26.114]
            3GPP TS 26.114 V12.8.0, "3rd Generation Partnership
            Project; Technical Specification Group Services and subsequent offers from non-JSEP peers.</t>

          <t>Update references.</t>
        </list>
      </t>

      <t>Changes in draft-17:</t>

      <t>
        <list style="symbols">

          <t>Split createOffer System
            Aspects; IP Multimedia Subsystem (IMS); Multimedia
            Telephony; Media handling and createAnswer sections to clearly
          indicate attributes which always appear interaction (Release 12)",
            December 2014, <http://www.3gpp.org/DynaReport/26114.htm>.

Suggested:
 [TS26.114]
            3GPP, "3rd Generation Partnership Project; Technical
            Specification Group Services and which only appear
          when not bundled into another m= section.</t>

          <t>Add descriptions of RtpTransceiver methods.</t>

          <t>Describe how to process RTCP feedback attributes.</t>

          <t>Clarify transceiver directions System Aspects; IP
            Multimedia Subsystem (IMS); Multimedia Telephony; Media
            handling and their interaction with
          3264.</t>

          <t>Describe setCodecPreferences.</t>

          <t>Update RTP demux algorithm. Include RTCP.</t>

          <t>Update requirements for when a=rtcp is included, limiting
          to cases where it is needed for backward compatibility.</t>

          <t>Clarify SAR handling.</t>

          <t>Updated addTrack matching algorithm.</t>

          <t>Remove a=ssrc requirements.</t>

          <t>Handle a=setup (Release 16)", 3GPP TS 26.114
            V16.3.0, September 2019,
            <http://www.3gpp.org/DynaReport/26114.htm>. -->

      </references>
    </references>
    <section anchor="sec.appendix-a" numbered="true" toc="default">
      <name>ABNF Definitions</name>
      <t>For the syntax validation performed in reoffers.</t>

          <t>Discuss how RTX/FEC should be handled.</t>

          <t>Discuss how telephone-event should be handled.</t>

          <t>Discuss how CN/DTX should be handled.</t>

          <t>Add missing references to
      <xref target="sec.parsing-a-desc" format="default"/>, the following list of ABNF table.</t>
        </list>
      </t>

      <t>Changes in draft-16:</t>

      <t>
        <list style="symbols">

          <t>Update addIceCandidate to indicate ICE generation
      definitions is used:</t>
      <table anchor="sdp-abnf" align="center">
        <name>SDP ABNF References</name>
        <thead>
          <tr>
            <th align="left">Attribute</th>
            <th align="left">Reference</th>
          </tr>
        </thead>
        <tbody>
          <tr>
            <td align="left">ptime</td>
            <td align="left">
              <xref target="RFC4566" sectionFormat="of" section="9"/></td>
          </tr>
          <tr>
            <td align="left">maxptime</td>
            <td align="left">
              <xref target="RFC4566" sectionFormat="of" section="9"/></td>
          </tr>
          <tr>
            <td align="left">rtpmap</td>
            <td align="left">
              <xref target="RFC4566" sectionFormat="of" section="9"/></td>
          </tr>
          <tr>
            <td align="left">recvonly</td>
            <td align="left">
              <xref target="RFC4566" sectionFormat="of" section="9"/></td>
          </tr>
          <tr>
            <td align="left">sendrecv</td>
            <td align="left">
              <xref target="RFC4566" sectionFormat="of" section="9"/></td>
          </tr>
          <tr>
            <td align="left">sendonly</td>
            <td align="left">
              <xref target="RFC4566" sectionFormat="of" section="9"/></td>
          </tr>
          <tr>
            <td align="left">inactive</td>
            <td align="left">
              <xref target="RFC4566" sectionFormat="of" section="9"/></td>
          </tr>
          <tr>
            <td align="left">framerate</td>
            <td align="left">
              <xref target="RFC4566" sectionFormat="of" section="9"/></td>
          </tr>
          <tr>
            <td align="left">fmtp</td>
            <td align="left">
              <xref target="RFC4566" sectionFormat="of" section="9"/></td>
          </tr>
          <tr>
            <td align="left">quality</td>
            <td align="left">
              <xref target="RFC4566" sectionFormat="of" section="9"/></td>
          </tr>
          <tr>
            <td align="left">rtcp</td>
            <td align="left">
              <xref target="RFC3605" sectionFormat="of" section="2.1"/></td>
          </tr>
          <tr>
            <td align="left">setup</td>
            <td align="left">
              Sections <xref target="RFC4145" section="3"
	      sectionFormat="bare"/>,
              <xref target="RFC4145" section="4" sectionFormat="bare"/>, and
              <xref target="RFC4145" section="5" sectionFormat="bare"/> of
              <xref target="RFC4145"/></td>
          </tr>
          <tr>
            <td align="left">connection</td>
            <td align="left">
              Sections <xref target="RFC4145" section="3"
	      sectionFormat="bare"/>,
              <xref target="RFC4145" section="4" sectionFormat="bare"/>, and
          allow per-m= section end-of-candidates.</t>

          <t>Update fingerprint handling to use
          draft-ietf-mmusic-4572-update.</t>

          <t>Update text around SDP processing
              <xref target="RFC4145" section="5" sectionFormat="bare"/> of RTP header extensions
              <xref target="RFC4145"/></td>
          </tr>
          <tr>
            <td align="left">fingerprint</td>
            <td align="left">
              <xref target="RFC8122" sectionFormat="of" section="5"/></td>
          </tr>
          <tr>
            <td align="left">rtcp-fb</td>
            <td align="left">
              <xref target="RFC4585" sectionFormat="of" section="4.2"/></td>
          </tr>
          <tr>
            <td align="left">extmap</td>
            <td align="left">
              <xref target="RFC5285" sectionFormat="of" section="7"/></td>
          </tr>
          <tr>
            <td align="left">mid</td>
            <td align="left">
              Sections <xref target="RFC5888" section="4"
	      sectionFormat="bare"/> and payload formats.</t>

          <t>Add sections on simulcast, addTransceiver,
              <xref target="RFC5888" section="5" sectionFormat="bare"/> of
              <xref target="RFC5888"/></td>
          </tr>
          <tr>
            <td align="left">group</td>
            <td align="left">
             Sections <xref target="RFC5888" section="4"
	     sectionFormat="bare"/> and
          createDataChannel.</t>

          <t>Clarify text to ensure that
             <xref target="RFC5888" section="5" sectionFormat="bare"/> of
             <xref target="RFC5888"/></td>
          </tr>
          <tr>
            <td align="left">imageattr</td>
            <td align="left">
              <xref target="RFC6236" sectionFormat="of" section="3.1"/></td>
          </tr>
          <tr>
            <td align="left">extmap (encrypt option)</td>
            <td align="left">
              <xref target="RFC6904" sectionFormat="of" section="4"/></td>
          </tr>
          <tr>
            <td align="left">candidate</td>
            <td align="left">
              <xref target="RFC8839" sectionFormat="of" section="5.1"/></td>
          </tr>
          <tr>
            <td align="left">remote-candidates</td>
            <td align="left">
              <xref target="RFC8839" sectionFormat="of" section="5.2"/></td>
          </tr>
          <tr>
            <td align="left">ice-lite</td>
            <td align="left">
              <xref target="RFC8839" sectionFormat="of" section="5.3"/></td>
          </tr>
          <tr>
            <td align="left">ice-ufrag</td>
            <td align="left">
              <xref target="RFC8839" sectionFormat="of" section="5.4"/></td>
          </tr>
          <tr>
            <td align="left">ice-pwd</td>
            <td align="left">
              <xref target="RFC8839" sectionFormat="of" section="5.4"/></td>
          </tr>
          <tr>
            <td align="left">ice-options</td>
            <td align="left">
              <xref target="RFC8839" sectionFormat="of" section="5.6"/></td>
          </tr>
          <tr>
            <td align="left">msid</td>
            <td align="left">
              <xref target="RFC8830" sectionFormat="of" section="3"/></td>
          </tr>
          <tr>
            <td align="left">rid</td>
            <td align="left">
              <xref target="RFC8851" sectionFormat="of" section="10"/></td>
          </tr>
          <tr>
            <td align="left">simulcast</td>
            <td align="left">
              <xref target="RFC8853" sectionFormat="of" section="6.1"/></td>
          </tr>
          <tr>
            <td align="left">tls-id</td>
            <td align="left">
              <xref target="RFC8842" sectionFormat="of" section="4"/></td>
          </tr>
        </tbody>
      </table>

<!-- [rfced] Appendix A:  Please review the session ID is a positive
          63 bit integer.</t>

          <t>Clarify SDP processing for direction indication.</t>

          <t>Describe SDP processing for rtcp-mux-only.</t>

          <t>Specify how SDP session version in o= line.</t>

          <t>Require that when doing an re-offer, following, and let us
know any concerns.

a) We changed the capabilities title of
          the new session are mostly required Appendix A from "Appendix A" to be a subset of the
          previously negotiated session.</t>

          <t>Clarified ICE restart interaction with bundle-only.</t>

          <t>Remove support for changing SDP before calling
          setLocalDescription.</t>

          <t>Specify algorithm for demuxing RTP based on MID, PT, and
          SSRC.</t>

          <t>Clarify rules for rejecting m= lines when bundle policy
"ABNF Definitions."  If this is
          balanced or max-bundle.</t>
        </list>
      </t>

      <t>Changes in draft-15:</t>

      <t>
        <list style="symbols">

          <t>Clarify text around codecs offered in subsequent
          transactions to refer to what's been negotiated.</t>

          <t>Rewrite LS handling text to indicate edge cases and that
          we're living with them.</t>

          <t>Require that answerer reject m= lines when there are no
          codecs not correct, please provide an
appropriately descriptive title.

b) Please review the citations listed in common.</t>

          <t>Enforce max-bundle on offer processing.</t>

          <t>Fix TIAS formula to handle bits vs. kilobits.</t>

          <t>Describe addTrack algorithm.</t>

          <t>Clean up references.</t>
        </list>
      </t>

      <t>Changes Table 1.  In many cases,
we could not see the relevant parameters listed in draft-14:</t>

      <t>
        <list style="symbols">

          <t>Added discussion the cited
sections.  For example, we do not see

* any of RtpTransceivers + RtpSenders +
          RtpReceivers, and how they interact with
          createOffer/createAnswer.</t>

          <t>Removed obsolete OfferToReceiveX options.</t>

          <t>Explained how addIceCandidate can be used for
          end-of-candidates.</t>
        </list>
      </t>

      <t>Changes in draft-13:</t>

      <t>
        <list style="symbols">

          <t>Clarified which SDP lines can be ignored.</t>

          <t>Clarified how to handle various received attributes.</t>

          <t>Revised how the attributes should be generated for bundled m=
          lines.</t>

          <t>Remove unused references.</t>

          <t>Remove text advocating use (ptime, maxptime, rtpmap, recvonly, ...)
  listed in Section 9 of unilateral PTs.</t>

          <t>Trigger an ICE restart even if [RFC4566].
* the setup attribute mentioned in Section 3 of [RFC4145].
* the ICE candidate policy is
          being made more strict.</t>

          <t>Remove connection attribute mentioned in Sections 3 or 4 of [RFC4145].
* the 'public' ICE candidate policy.</t>

          <t>Move open issues into GitHub issues.</t>

          <t>Split local/remote description accessors into
          current/pending.</t>

          <t>Clarify a=imageattr handling.</t>

          <t>Add more detail on VoiceActivityDetection handling.</t>

          <t>Reference draft-shieh-rtcweb-ip-handling.</t>

          <t>Make it clear when an ICE restart should occur.</t>

          <t>Resolve changes needed "mid" attribute listed in references.</t>

          <t>Remove MSID semantics.</t>

          <t>ice-options are now at session level.</t>

          <t>Default RTCP mux policy is now 'require'.</t>
        </list>
      </t>

      <t>Changes Section 5 of [RFC5888].
* the "group" attribute listed in draft-12:</t>

      <t>
        <list style="symbols">

          <t>Filled Section 4 of [RFC5888].

c) We do not see the attributes "framerate," "quality," or
"connection" listed anywhere else in sections on applying local and remote
          descriptions.</t>

          <t>Discussed downscaling and upscaling this document.  Do they need to fulfill imageattr
          requirements.</t>

          <t>Updated what SDP can
be modified by the application.</t>

          <t>Updated to latest datachannel SDP.</t>

          <t>Allowed multiple fingerprint lines.</t>

          <t>Switched back included here?

d) Please note that in order to IPv4 for dummy candidates.</t>

          <t>Added additional clarity on ICE default candidates.</t>
        </list>
      </t>

      <t>Changes render workable hyperlinks in draft-11:</t>

      <t>
        <list style="symbols">

          <t>Clarified handling the
.html/.pdf files we changed the original citation format (for example,
"[RFC5285] Section 7") to this style (for example, "Section 7 of RTP CNAMEs.</t>

          <t>Updated what SDP lines
[RFC5285]").

e) As noted previously, it appears that most of the listed section
numbers for RFC 8839 [I-D.ietf-mmusic-ice-sip-sdp] are "off by one" (i.e.,
that "4.1" should be processed or ignored.</t>

          <t>Specified how a=imageattr "5.1," "4.2" should be used.</t>
        </list>
      </t>

      <t>Changes in draft-10:</t>

      <t>
        <list style="symbols">

          <t>Described video size negotiation with imageattr.</t>

          <t>Clarified rejection of sections "5.2," etc.  We have updated the
section references accordingly; please review and let us know if any
corrections are needed.

f) Please confirm that do Section 6.1 of RFC 8853 [I-D.ietf-mmusic-sdp-simulcast]
is the correct section to cite here.
(We ask because we could not have
          mux-only.</t>

          <t>Add handling see a relationship, other than the
mention of LS groups</t>
        </list>
      </t>

      <t>Changes "simulcast" mostly in draft-09:</t>

      <t>
        <list style="symbols">

          <t>Don't return null terms of simulcast streams, and also
because we see ABNF for the "a=simulcast" attribute in Section 5.1 of
RFC 8853 [I-D.ietf-mmusic-sdp-simulcast].)

Original:
 Appendix A.  Appendix A
...
 | simulcast              | [I-D.ietf-mmusic-sdp-simulcast] Section  |
 |                        | 6.1

Currently:
 Appendix A.  ABNF Definitions
...
 | simulcast               | Section 6.1 of [RFC8853]          | -->

    </section>
    <section anchor="sec.acknowledgements" numbered="false" toc="default">
      <name>Acknowledgements</name>
      <t><contact fullname="Harald Alvestrand"/>, <contact fullname="Taylor
      Brandstetter"/>, <contact fullname="Suhas Nandakumar"/>, and
      <contact fullname="Peter Thatcher"/> provided significant text for this
      document. <contact fullname="Bernard Aboba"/>, <contact fullname="Adam
      Bergkvist"/>, <contact fullname="Dan Burnett"/>, <contact fullname="Ben
      Campbell"/>, <contact fullname="Alissa Cooper"/>,
      <contact fullname="Richard Ejzak"/>, <contact fullname="Stefan
      Håkansson"/>, <contact fullname="Ted Hardie"/>, <contact fullname="Christer Holmberg"/>,
      <contact fullname="Andrew Hutton"/>, <contact fullname="Randell
      Jesup"/>, <contact fullname="Matthew Kaufman"/>, <contact fullname="Anant Narayanan"/>,
      <contact fullname="Adam Roach"/>, <contact fullname="Robert Sparks"/>,
      <contact fullname="Neil Stratford"/>, <contact fullname="Martin
      Thomson"/>, <contact fullname="Sean
      Turner"/>, and <contact fullname="Magnus Westerlund"/> all provided valuable feedback on
      this document.</t>
    </section>
  </back>

<!-- [rfced] Please let us know if any changes are needed for {local,remote}Description after
          close().</t>

          <t>Changed TCP/TLS to UDP/DTLS the
following:

a) The following terms were used inconsistently in RTP profile names.</t>

          <t>Separate out bundle and mux policy.</t>

          <t>Added specific references this document.
We chose to FEC mechanisms.</t>

          <t>Added canTrickle mechanism.</t>

          <t>Added section on subsequent answers and, answer
          options.</t>

          <t>Added text defining set{Local,Remote}Description
          behavior.</t>
        </list>
      </t>

      <t>Changes in draft-08:
      <list style="symbols">

        <t>Added new example section and removed old examples in
        appendix.</t>

        <t>Fixed &lt;proto&gt; field handling.</t>

        <t>Added text describing a=rtcp attribute.</t>

        <t>Reworked handling of OfferToReceiveAudio and
        OfferToReceiveVideo per discussion at IETF 90.</t>

        <t>Reworked trickle ICE handling and its impact on use the latter forms.  Please let us know any objections.

 a m= / an "m=" (per published RFCs and c=
        lines per discussion at interim.</t>

        <t>Added max-bundle-and-rtcp-mux policy.</t>

        <t>Added description of maxptime handling.</t>

        <t>Updated ICE candidate pool default author feedback for other
   documents in this cluster

 q= value / "q=" value  (We also changed 'x= and y= values' to 0.</t>

        <t>Resolved open issues around AppID/receiver-ID.</t>

        <t>Reworked
   "x=" and expanded how changes "y=" values')

 a RTP / an RTP

 to true / to "true"  (per the ICE configuration
        are handled.</t>

        <t>Some reference updates.</t>

        <t>Editorial clarification.</t>
      </list></t>

      <t>Changes in draft-07:
      <list style="symbols">

        <t>Expanded discussion rest of VAD and Opus DTX.</t>

        <t>Added a security considerations section.</t>

        <t>Rewrote the section on modifying SDP cluster)

 to require
        implementations false / to clearly indicate whether any given
        modification is allowed.</t>

        <t>Clarified impact "false"  (per the rest of IceRestart on CreateOffer in local-offer
        state.</t>

        <t>Guidance on whether attributes should be defined at the
        media level cluster)

 bundle-tag / BUNDLE-tag (per other documents in this cluster)

 offer-answer exchange / offer/answer exchange

 clockrate / clock rate (per RFCs 3389 and 7160)

 other than IDENTICAL and TRANSPORT / other than IDENTICAL or the session level.</t>

        <t>Renamed "default" bundle policy TRANSPORT

b) The following terms appear to "balanced".</t>

        <t>Removed default ICE candidate pool size be used inconsistently in this
document.  Please let us know which form is preferred.

 createOffer API / createOffer() API (We see "createAnswer() API" but
  not "createAnswer API"; however, we also see "getCapabilities API"
  and clarify how "W3C RTCPeerConnection API."  In RFC 8826 (draft-ietf-rtcweb-security),
  we see "XMLHttpRequest() API.")

 setLocalDescription API / setLocalDescription() API
   / setRemoteDescription() API
    (We also see "whether setLocalDescription or
     setRemoteDescription is called" and "before
     setRemoteDescription() is called")

 mid identifiers / MID identifiers
  Because RFC 8843 (draft-ietf-mmusic-sdp-bundle-negotiation) and
  RFC 8852 (draft-ietf-avtext-rid) define "MID" as "Media Identification,"
  would it
        works.</t>

        <t>Defined a canonical order for assignment of MSTs be appropriate to m=
        lines.</t>

        <t>Removed discussion use "mid values" (or "MID values"*)
  and "MIDs" instead of rehydration.</t>

        <t>Added Eric Rescorla as a draft editor.</t>

        <t>Cleaned up references.</t>

        <t>Editorial cleanup</t>
      </list></t>

      <t>Changes "mid identifiers" and "MID identifiers"?

  * We see inconsistent capitalization in draft-06:
      <list style="symbols">

        <t>Reworked handling of m= line recycling.</t>

        <t>Added handling this cluster of BUNDLE documents;
    a separate question regarding which form to use will be sent
    separately.

 "RID identifiers" and bundle-only.</t>

        <t>Clarified handling of rollback.</t>

        <t>Added text describing the ICE Candidate Pool "rid-identifier":  RFC 8851 (draft-ietf-mmusic-rid)
   defines "RID" as "restriction identifier."  Should "RID
   identifiers" and its
        behavior.</t>

        <t>Allowed OfferToReceiveX "rid-identifier" be "rid-ids" and "rid-id"
   per RFC 8851 (draft-ietf-mmusic-rid), to create multiple recvonly m=
        sections.</t>
      </list></t>

      <t>Changes in draft-05:
      <list style="symbols">

        <t>Fixed several issues identified in avoid "identifier identifier(s)"?

 the createOffer/Answer
        sections during document review.</t>

        <t>Updated references.</t>
      </list></t>

      <t>Changes in draft-04:
      <list style="symbols">

        <t>Filled in sections on createOffer and createAnswer.</t>

        <t>Added SDP examples.</t>

        <t>Fixed references.</t>
      </list></t>

      <t>Changes in draft-03:
      <list style="symbols">

        <t>Added text describing relationship stable state (1 instance) / the "stable" state (3 instances)
   (Also, should 'the previous stable state' be 'a previous stable
   state' (per Section 5.7) or 'the previous "stable" state'?)

 sess-id / <sess-id> (Section 5.2.1, second bullet item)

 set to W3C specification</t>
      </list></t>

      <t>Changes in draft-02:
      <list style="symbols">
        <!-- A -->

        <t>Converted from nroff</t>
        <!-- B -->

        <t>Removed comparisons zero / set to old approaches abandoned by 0  (May we use the
        working group</t>
        <!-- C -->

        <t>Removed stuff that has moved to W3C specification</t>
        <!-- D -->

        <t>Align SDP handling with W3C draft</t>
        <!-- E -->

        <t>Clarified section on forking.</t>
        <!-- F -->
        <!-- G -->
        <!-- H -->
        <!-- I -->
        <!-- J -->
        <!-- K -->
        <!-- L -->
      </list></t>

      <t>Changes latter, because of
   '"dummy" port value of 9' in draft-01:
      <list style="symbols">

        <t>Added diagrams for architecture Sections 5.2.1 and state machine.</t>

        <t>Added 5.3.1?)

 dummy value / "dummy" value

 fmt value / "fmt" value

 mid value / "mid" value

 tls-id value / "a=tls-id" value

 "IceRestart" option / IceRestart option

 dissociate (Section 5.10) / disassociate (Sections 3.4.1 and 5.7)
   Example:  "A rollback disassociates any RtpTransceivers that were
   associated with m= sections on forking and rehydration.</t>

        <t>Clarified meaning by the application of "pranswer" and "answer".</t>

        <t>Reworked how ICE restarts the rolled-back
   session description (see Section 5.10 and media directions are
        controlled.</t>

        <t>Added list of parameters that can Section 5.9)."

c) Are "canTrickleIceCandidates" (Section 4.1.15) and "canTrickle
property" (Section 5.10) distinct terms?  If so, should this
distinction be changed in a
        description.</t>

        <t>Updated suggested API and examples to match latest
        thinking.</t>

        <t>Suggested API and examples have been moved to an
        appendix.</t>
      </list></t>

      <t>Changes clarified in draft -00:
      <list style="symbols">

        <t>Migrated from draft-uberti-rtcweb-jsep-02.</t>
      </list></t>
    </section>
  </back> the text? -->

</rfc>