<?xmlversion="1.0" encoding="us-ascii"?>version='1.0' encoding='utf-8'?> <!DOCTYPE rfc SYSTEM"rfc2629.dtd"> <?rfc toc="yes" ?> <?rfc symrefs="yes" ?> <?rfc iprnotified="no" ?> <?rfc strict="yes" ?> <?rfc compact="yes" ?> <?rfc sortrefs="yes" ?> <?rfc colonspace="yes" ?> <?rfc rfcedstyle="no" ?> <?rfc docmapping="yes" ?> <?rfc tocdepth="4"?>"rfc2629-xhtml.ent"> <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 EstablishmentProtocol</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 6thSt 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. EvelynAve</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 RTCPeerConnectionAPI,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. --> <sectiontitle="Introduction" anchor="sec.introduction">anchor="sec.introduction" numbered="true" toc="default"> <name>Introduction</name> <t>This document describes how the W3CWEBRTCWeb Real-Time Communication (WebRTC) RTCPeerConnection interface <xreftarget="W3C.webrtc"></xref>target="W3C.webrtc" format="default"/> is used to control the setup,managementmanagement, and teardown of a multimedia session.</t> <sectiontitle="Generalanchor="sec.general-design-of-jsep" numbered="true" toc="default"> <name>General Design ofJSEP" anchor="sec.general-design-of-jsep">JSEP</name> <t>WebRTC call setup has been designed to focus on controlling the media plane, leavingsignaling planesignaling-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 thenecessarytransport 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 theICEInteractive Connectivity Establishment (ICE) statemachine.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 thoughNote, though, that in many cases it will actually be between a JSEP endpoint and some kind of server, such as a gateway orMCU.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 <xreftarget="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, <xreftarget="I-D.ietf-mmusic-trickle-ice-sip" />target="RFC8840" format="default"/> allows SIP to be used withtrickleTrickle ICE <xreftarget="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 ahigher levelhigher-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> <sectiontitle="Otheranchor="sec.other-approaches-consider" numbered="true" toc="default"> <name>Other ApproachesConsidered" 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 ofinteractions withthe mediasubsystem thancontrol 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 basicsession 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 getCapabilitiesAPIAPI, 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> <sectiontitle="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 14 <xref target="RFC2119"/> <xreftarget="RFC2119"></xref>.</t>target="RFC8174"/> when, and only when, they appear in all capitals, as shown here.</t> </section> <sectiontitle="Semanticsanchor="sec.semantics-and-syntax" numbered="true" toc="default"> <name>Semantics andSyntax" anchor="sec.semantics-and-syntax">Syntax</name> <sectiontitle="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 <xreftarget="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> <figureanchor="fig-sigModel" title="JSEPanchor="fig-sigModel"> <name>JSEP SignalingModel"> <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> <sectiontitle="Sessionanchor="sec.session-descriptions-and-state-machine" numbered="true" toc="default"> <name>Session Descriptions and StateMachine" 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 aredeclarativedeclarative, and the remote sideMUST<bcp14>MUST</bcp14> either accept them or reject them altogether. An example of such a parameter is the DTLS fingerprints <xreftarget="RFC8122"></xref>,target="RFC8122" format="default"/>, which are calculated based on the local certificate(s)offered,offered and are not subject tonegotiation.</t> <t>In addition, various RFCs put different conditions on the format of offers versus answers. For example, an offer may propose an arbitrary numbernegotiation. <!-- [rfced] Section 3.2: We do not see any mention ofm= sections (i.e., media descriptions as describedDTLS 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 knownRFC 8122; we onlyafter an offer and ansee "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 remoteside.</t> <t>Toside.</li> <li>To know if a session description is an offer or ananswer.</t> <t>Toanswer.</li> <li>To allow the offer to be specified independently of theanswer.</t> </list>JSEPanswer.</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 forand 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 mayeitherbe 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,TURNTraversal 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 <xreftarget="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 issent,sent and is thencancelledcanceled 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 provisionalanswers,answers and then finally end theoffer-answeroffer/answer exchange by sending a final answer. The state machine for this is as follows:</t><t><figureanchor="fig-state-machine" title="JSEPanchor="fig-state-machine"> <name>JSEP StateMachine"> <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 statetransitionstransitions, there is no other difference between the handling of provisional ("pranswer") and final ("answer") answers.</t> </section> <sectiontitle="Sessionanchor="sec.session-description-forma" numbered="true" toc="default"> <name>Session DescriptionFormat" anchor="sec.session-description-forma">Format</name> <t>JSEP's session descriptions useSDPSession Description Protocol (SDP) syntax for their internal representation. While this format is not optimal for manipulation from JavaScript, it is widelyaccepted,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 fromSDP,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> <sectiontitle="Sessionanchor="sec.session-description-ctrl" numbered="true" toc="default"> <name>Session DescriptionControl" anchor="sec.session-description-ctrl">Control</name> <t>In order to give the application control over various common session parameters, JSEP provides control surfaceswhichthat 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 subsequentcreateOffer/AnswercreateOffer/createAnswer calls.</t> <sectiontitle="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 onem="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 withm="m=" sections, although there may be more RtpTransceivers thanm="m=" sections when RtpTransceivers are created but not yet associated witha m=an "m=" section, or if RtpTransceivers have been stopped and disassociated fromm="m=" sections. An RtpTransceiver is said to be associated with anm="m=" section if itsmidmedia identification (mid) property is non-null;otherwiseotherwise, it is said to be disassociated. The associatedm="m=" section is determined using a mapping between transceivers andm="m=" section indices, formed when creating an offer or applying a remote offer.</t> <t>An RtpTransceiver is never associated with more than onem="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 wherea m=an "m=" section has been rejected, as discussed in <xref target="sec.subsequent-offers"/>format="default"/> below, thatm="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 newm="m=" sections.</t> </section> <sectiontitle="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 anm="m=" section. This includes encoding the attached MediaStreamTrack, sending RTP media packets, and generating/processingRTCPthe RTP Control Protocol (RTCP) for the outgoing RTP streams(s).</t> </section> <sectiontitle="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 anm="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> <sectiontitle="ICE" anchor="sec.ice"> <section title="ICEanchor="sec.ice" numbered="true" toc="default"> <name>ICE</name> <section anchor="sec.ice-gather-overview" numbered="true" toc="default"> <name>ICE GatheringOverview" 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 recycledm="m=" section to the local sessiondescription,description or by new ICE credentials in the description, indicating an ICE restart. Use of new ICE credentials can be triggered explicitly by theapplication,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 alocal-locally orremote-initiatedremotely 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/restartingm="m=" sections; otherm="m=" sections continue to use their existing candidates. Also, if anm="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 thatm="m=" section if and only if its MID item is a BUNDLE-tag, as described in <xreftarget="I-D.ietf-mmusic-sdp-bundle-negotiation" />.</t>target="RFC8843" format="default"/>.</t> </section> <sectiontitle="ICEanchor="sec.ice-candidate-trickling" numbered="true" toc="default"> <name>ICE CandidateTrickling" 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 <xreftarget="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 getthenotified 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, atthisthat 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> <sectiontitle="ICEanchor="sec.ice-candidate-format" numbered="true" toc="default"> <name>ICE CandidateFormat" 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 <xreftarget="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 <xreftarget="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 whichm="m=" section it is associated with, which can be identified in one of twoways,ways: either bya m=an "m=" sectionindex,index or by a MID. Them="m=" section index is a zero-based index, with index N referring to the N+1thm="m=" section in the session description referenced by this IceCandidate. The MID is a "media stream identification" value, as defined in <xreftarget="RFC5888"></xref>, Section 4,target="RFC5888" sectionFormat="comma" section="4"/>, which provides a more robust way to identify them="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, itMUST<bcp14>MUST</bcp14> be used for identification; otherwise, them="m=" section index is used instead.</t> <t>When creating an IceCandidate object, JSEP implementationsMUST<bcp14>MUST</bcp14> populate each of the candidate, ufrag,m="m=" section index, and MID fields. ImplementationsMUST<bcp14>MUST</bcp14> also be prepared to receive objects with some fields missing, as mentioned above.</t> </section> </section> <sectiontitle="ICEanchor="sec.ice-candidate-policy" numbered="true" toc="default"> <name>ICE CandidatePolicy" 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 <xreftarget="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 implementationMUST<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 implementationMUST 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> <sectiontitle="ICEanchor="sec.ice-candidate-pool" numbered="true" toc="default"> <name>ICE CandidatePool" 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 componentswhichthat 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, theySHOULD<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 componentsis 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 thoughNote, 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 isusing.</t>using. ("STUN" stands for "Session Traversal Utilities for NAT".)</t> </section> <sectiontitle="ICE Versions">numbered="true" toc="default"> <name>ICE Versions</name> <t>While this specification formally relies on <xreftarget="RFC8445"></xref>,target="RFC8445" format="default"/>, at the time of its publication, the majority of WebRTC implementations support the version of ICE described in <xreftarget="RFC5245"></xref>.target="RFC5245" format="default"/>. Theuse of the"ice2" attribute defined in <xreftarget="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. ImplementationsMUST<bcp14>MUST</bcp14> be able to accept remote descriptions that do not have the "ice2" attribute.</t> </section> </section> <section anchor="sec.imageattr"title="Videonumbered="true" toc="default"> <name>Video SizeNegotiation">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 implementationMUST<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-squarepixels,pixels butSHOULD<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="Creatingnumbered="true" toc="default"> <name>Creating an imageattrAttribute">Attribute</name> <t>The receiver will first intersect any known local limits (e.g., hardware decodercapababilities,capabilities, local policy) to determine the absolute minimum and maximum sizes it can receive. If there are no known local limits, the "a=imageattr" attributeSHOULD<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" attributeMUST<bcp14>MUST</bcp14> be omitted, and them="m=" sectionMUST<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 maximumx="x=" andy="y=" values.</t> <t>The rules here express a single set of preferences, and therefore, the "a=imageattr"q="q=" value is not important. ItSHOULD<bcp14>SHOULD</bcp14> be set to1.0.</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 (*), isRECOMMENDED.<bcp14>RECOMMENDED</bcp14>. However, when the supported video codecs have different limitations, specific "a=imageattr" attributesMUST<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 to720p,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="Interpretingnumbered="true" toc="default"> <name>Interpreting imageattrAttributes">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 canuse,use but gives an indication of the preferred values.</t> <t>This specification prescribes behavior that is morespecific behavior.specific. When a MediaStreamTrack, which is producing video of a certain resolution (the "track resolution"), is attached toaan 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, itMUST<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 rulesMUST<bcp14>MUST</bcp14> be followed as long as the attributes remain present in the remote description, including cases in which the track changes itsresolution,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 certainresolution,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 whenneeded,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 correspondingm="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 containingm="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 withm="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 encodingMUST NOT<bcp14>MUST NOT</bcp14> be transmitted, and an errorSHOULD<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 resolutionMUST<bcp14>MUST</bcp14> be used for the checks. This is required regardless of whether the receiver supports performing receive-side rotation (e.g., throughCVOCoordination of Video Orientation (CVO) <xref target="TS26.114"/>),format="default"/>), as it significantly simplifies the matchinglogic.</t> <t>Iflogic.</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 attributeMUST NOT<bcp14>MUST NOT</bcp14> beused.</t> <t>Ifused.</li> <li>If the encoder resolution exceeds the maximum size permitted by theattribute,attribute and the encoder is allowed to adjust its resolution, the encoderSHOULD<bcp14>SHOULD</bcp14> apply downscaling in order to satisfy the limits. DownscalingMUST 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 is1280x720,1280x720 and the attribute specified a maximum of 640x480, the expected output resolution would be 640x360. If downscaling cannot be applied, the attributeMUST NOT<bcp14>MUST NOT</bcp14> beused.</t> <t>Ifused.</li> <li>If the encoder resolution is less than the minimum size permitted by the attribute, the attributeMUST NOT<bcp14>MUST NOT</bcp14> be used; the encoderMUST NOT<bcp14>MUST NOT</bcp14> apply upscaling. JSEP implementationsSHOULD<bcp14>SHOULD</bcp14> avoid this situation by allowing receipt of arbitrarily small resolutions, perhaps via fallback to a softwaredecoder.</t> <t>Ifdecoder.</li> <li>If the encoder resolution is within the maximum and minimum sizes, no action isneeded.</t> </list></t>needed.</li> </ul> </section> </section> <sectiontitle="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 singlem="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 correspondingm="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 perm="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 itsininitial 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 fromaan RtpSender, the techniques from <xreftarget="I-D.ietf-mmusic-sdp-simulcast" />target="RFC8853" format="default"/> and <xreftarget="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. --> Them="m=" section will also include an "a=rid" attribute for each encoding, as specified in <xreftarget="I-D.ietf-mmusic-rid" />, Section 4;target="RFC8851" sectionFormat="comma" section="4"/>; the use ofRID identifiersRestriction Identifiers (RIDs) allows the individual encodings to be disambiguated even though they are all part of the samem="m=" section.</t> </section> <sectiontitle="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 <xreftarget="RFC3261"></xref>target="RFC3261" format="default"/> defines both a"Parallel Search""parallel search" and"Sequential Search"."sequential search". Although these are primarilysignaling levelsignaling-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 pointofin 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 doare: <list style="symbols"> <t>Startare as follows: </t> <ul spacing="normal"> <li>Start exchanging media with a given remote peer, but keep all the resources reserved in theoffer.</t> <t>Startoffer.</li> <li>Start exchanging media with a given remote peer, and free any resources in the offer that are not beingused.</t> </list></t>used.</li> </ul> <sectiontitle="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 iteitherasaeither (1) a provisional answer, leaving open the possibility of using a different answer in thefuture,future orapply it as a(2) 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> <sectiontitle="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 thecall,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 <xreftarget="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 atime,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 eithertone,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 <xreftarget="RFC3960"></xref> describestarget="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 produceaan SDP offer that could be used by the signaling to perform the UPDATE strategy discussed in <xreftarget="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> <sectiontitle="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 theseconcepts.</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> <sectiontitle="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 theapplication,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 andused.</t> <t></t> <t hangText="relay:">Allused.</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 globallevel.</t> </list></t>level.</dd> </dl> <t>The default ICE candidate policyMUST<bcp14>MUST</bcp14> be set to"all""all", as this is generally the desiredpolicy,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. Becausepre-gatheringpre&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 sizeMUST<bcp14>MUST</bcp14> be zero.</t> <t>The application can specify its preferred policy regarding use of bundle, the multiplexing mechanism defined in <xreftarget="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 singletransport,transport and will offer a single bundle group across allm="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 firstm="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 subsequentm= 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 forforN 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 firstm="m=" section of eachtype.</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-awareendpoints,endpoints but will require separate candidates to be gathered for each mediastream.</t> <t></t> <t hangText="max-bundle:">Onlystream.</dd> <dt>max-bundle:</dt> <dd>Only the firstm="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 anym="m=" sections other than the firstm="m=" section, unless they are in the same bundle group as thatm= section.</t> </list></t>"m=" section.</dd> </dl> <t>As it provides the besttradeofftrade-off between performance and compatibility with legacy endpoints, the default bundle policyMUST<bcp14>MUST</bcp14> be set to "balanced".</t> <t>The application can specify its preferred policy regarding use of RTP/RTCP multiplexing <xreftarget="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-multiplexingendpoints.</t> <t hangText="require:">Theendpoints.</dd> <dt>require:</dt> <dd>The JSEP implementation will only gather RTP candidates and will insert an "a=rtcp-mux-only" indication into any newm="m=" sections in offers it generates. This halves the number of candidates that the offerer needs to gather. Applying a description with anm="m=" section that does not contain an "a=rtcp-mux" attribute will cause an error to bereturned.</t> </list></t>returned.</dd> </dl> <t>The default multiplexing policyMUST<bcp14>MUST</bcp14> be set to "require". ImplementationsMAY<bcp14>MAY</bcp14> choose to reject attempts by the application to set the multiplexing policy to "negotiate".</t> </section> <sectiontitle="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 <xreftarget="RFC5888"></xref>, Section 7. addTracktarget="RFC5888" sectionFormat="comma" section="7"/>. addTrack attempts to minimize the number of transceivers as follows:Ifif 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> <sectiontitle="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 them="m=" section associated with the sender as recvonly (if transceiver.direction is sendrecv) or as inactive (if transceiver.direction is sendonly).</t> </section> <sectiontitle="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 applicationMUST<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 MediaStreamswhichthat the transceiver is associated with (allowingLS"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> <sectiontitle="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 samem="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 argumentswhichthat are used by the PeerConnection (e.g., maxPacketLifetime) but are not reflected in the SDP and do not affect the JSEP state.</t> </section> <sectiontitle="createOffer" anchor="sec.createoffer">anchor="sec.createoffer" numbered="true" toc="default"> <name>createOffer</name> <t>The createOffer method generates a blob of SDP that containsa <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 lastoffer-answeroffer/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> <sectiontitle="createAnswer" anchor="sec.createanswer">anchor="sec.createanswer" numbered="true" toc="default"> <name>createAnswer</name> <t>The createAnswer method generates a blob of SDP that containsa <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, whichMUST<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, ororcause a media state change. Specifically, the answer is not applied, and does not become the current local description, until setLocalDescription is called.</t> </section> <sectiontitle="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 beparsed,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 appliedanytimeany time the PeerConnection is in a stablestate,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, theoffer-answeroffer/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 orfinal,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 provisionalanswers,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 contentsMUST<bcp14>MUST</bcp14> be empty.</t> <sectiontitle="Useanchor="sec.use-of-provisional-answer" numbered="true" toc="default"> <name>Use of ProvisionalAnswers" 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 thecaller,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 thistime.</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 andmoving the audioDTLS handshaking have" means "ICE handshaking andvideoDTLS handshaking have," we updated this sentence accordingly. Please let us know if this is incorrect (i.e., if the text refers tobe two-way media. A detailedone 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 <xreftarget="sec.warmup-example"></xref>.</t>target="sec.warmup-example" format="default"/>.</t> <t>Of course, some applications may not be able to perform this doubleoffer-answeroffer/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> <sectiontitle="Rollback" anchor="sec.rollback">anchor="sec.rollback" numbered="true" toc="default"> <name>Rollback</name> <t>In certainsituationssituations, it may be desirable to "undo" a change made to setLocalDescription or setRemoteDescription. Consider a case where a call isongoing,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 newparameters,parameters and sends a reject message back to the offerer. Now, the offerer, and possibly the answerer as well,needneeds 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> <sectiontitle="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 eithera(1) a final answer, at which point the PeerConnection can fully adopt the pending localdescription,description ora(2) 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 issupplied,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 mediatransmission.</t> </section> <section title="setRemoteDescription" anchor="sec.setremotedescription"> <t>The setRemoteDescription method instructstransmission. <!-- [rfced] Sections 4.1.9 and 4.1.10: We had trouble following thePeerConnection to applypurpose of all of thesupplied 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 thedesired remote configuration. Asmedia directions are compatible, and media is available to send, this will result insetLocalDescription,thetype fieldstarting ofthe description indicates how it should be processed.</t> <t>This API changes the localmediastate; among other things, it sets up local resources for sending and encoding media.</t> <t>Iftransmission. ... 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 mediatransmission.</t> </section> <section title="currentLocalDescription" anchor="sec.currentlocaldescription"> <t>The currentLocalDescription method returns the current negotiated local description - i.e., the local description fromtransmission. Possibly: If (1) setRemoteDescription was previously called with an offer, (2) setLocalDescription is called with an answer (provisional or final), (3) thelast successful offer/answer exchange - inmedia 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> <sectiontitle="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> <sectiontitle="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> <sectiontitle="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> <sectiontitle="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() iscalled.</t> <t hangText="true:">SDPcalled.</dd> <dt>true:</dt> <dd>SDP has been received from the other side indicating that it can supporttrickle.</t> <t hangText="false:">SDPtrickle.</dd> <dt>false:</dt> <dd>SDP has been received from the other side indicating that it cannot supporttrickle.</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 applicationwhichthat 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> <sectiontitle="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 arechanged:</t> <t> <list style="symbols"> <t>Any changeschanged: ... 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 nonzerosize,size and a local description has not yet been applied, any existing candidates will be discarded, and new candidates will be gathered from the newservers.</t> <t>Anyservers.</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 pooledcandidates.</t> <t>Thecandidates.</li> <li>The ICE candidate pool sizeMUST 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 arediscarded.</t> <t>Thediscarded.</li> <li>The bundle and RTCP-multiplexing policiesMUST NOT<bcp14>MUST NOT</bcp14> be changed after the construction of thePeerConnection.</t> </list> </t>PeerConnection.</li> </ul> <t>This call may result in a change to the state of the ICEAgent.</t>agent.</t> </section> <sectiontitle="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 <xreftarget="I-D.ietf-ice-trickle" />.</t>target="RFC8838" format="default"/>.</t> <t>In either case, them="m=" section index, MID, and ufrag fields from the supplied IceCandidate are used to determine whichm="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 them="m=" section index and MID fields is interpreted to mean that the indication applies to allm="m=" sections in the specified ICE candidate generation. However, if both fields are absent for a new remote candidate, thisMUST<bcp14>MUST</bcp14> be treated as an invalid condition, as specified below.</t> <t>If any IceCandidate fields contain invalidvalues,values or an error occurs during the processing of the IceCandidate object, the supplied IceCandidateMUST<bcp14>MUST</bcp14> be ignored and an errorMUST<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> <sectiontitle="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 associatedm="m=" section. See below for more details.</t> </section> <sectiontitle="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> <sectiontitle="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 associatedm="m=" section on future calls to createOffer and createAnswer. The permitted values for direction are "recvonly", "sendrecv", "sendonly", and "inactive", mirroring theidentically-namedidentically named directional attributes defined in <xreftarget="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> <sectiontitle="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> <sectiontitle="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 associatedm="m=" section. More specifically, it indicates the directional attribute <xreftarget="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 associatedm="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 beenapplied,applied or if the transceiver is stopped, currentDirection is set tonull.</t>"null".</t> </section> <sectiontitle="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 associatedm="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 associatedm="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 associatedm="m=" section will follow the specified preferences.</t> </section> </section> </section> <sectiontitle="SDPanchor="sec.sdp-interaction-procedure" numbered="true" toc="default"> <name>SDP InteractionProcedures" anchor="sec.sdp-interaction-procedure">Procedures</name> <t>This section describes the specific procedures to be followed when creating and parsing SDP objects.</t> <sectiontitle="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> <sectiontitle="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 omissionMUST<bcp14>MUST</bcp14> be treated as an error.<list style="symbols"> <t>ICE,</t> <ul spacing="normal"> <li>ICE, as specified in <xreftarget="RFC8445"></xref>, MUSTtarget="RFC8445" format="default"/>, <bcp14>MUST</bcp14> be used. Note that the remote endpoint may use aLitelite implementation; implementationsMUST<bcp14>MUST</bcp14> properly handle remote endpointswhichthat doICE-Lite.</t> <t>DTLSICE-lite.</li> <li>DTLS <xref target="RFC6347"/>format="default"/> or DTLS-SRTP <xreftarget="RFC5763"></xref>, MUSTtarget="RFC5763" format="default"/> <bcp14>MUST</bcp14> be used, as appropriate for the media type, as specified in <xreftarget="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> <sectiontitle="Profileanchor="sec.profile-names" numbered="true" toc="default"> <name>Profile Names andInteroperability" anchor="sec.profile-names">Interoperability</name> <t>For mediam="m=" sections, JSEP implementationsMUST<bcp14>MUST</bcp14> support the "UDP/TLS/RTP/SAVPF" profile specified in <xreftarget="RFC5764"></xref>target="RFC5764" format="default"/> as well as the "TCP/DTLS/RTP/SAVPF" profile specified in <xreftarget="RFC7850"></xref>,target="RFC7850" format="default"/> andMUST<bcp14>MUST</bcp14> indicate one of these profiles for each mediam="m=" line they produce in an offer. For datam="m=" sections, implementationsMUST<bcp14>MUST</bcp14> support the "UDP/DTLS/SCTP" profile as well as the "TCP/DTLS/SCTP"profile,profile andMUST<bcp14>MUST</bcp14> indicate one of these profiles for each datam="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 <xreftarget="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 implementationsMUST<bcp14>MUST</bcp14> follow the following rules when processing the mediam="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 followingMUST<bcp14>MUST</bcp14> be accepted:<list style="symbols"> <t>"RTP/AVP" (Defined</t> <ul spacing="normal"> <li>"RTP/AVP" (defined in <xreftarget="RFC4566"></xref>, Section 8.2.2)</t> <t>"RTP/AVPF" (Definedtarget="RFC4566" sectionFormat="comma" section="8.2.2"/>)</li> <li>"RTP/AVPF" (defined in <xreftarget="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" (Definedtarget="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 <xreftarget="RFC5764"></xref>, Section 9)</t> <t>"UDP/TLS/RTP/SAVPF" (Definedtarget="RFC5764" sectionFormat="comma" section="9"/>)</li> <li>"UDP/TLS/RTP/SAVPF" (defined in <xreftarget="RFC5764"></xref>, Section 9)</t> </list></t> <t>Thetarget="RFC5764" sectionFormat="comma" section="9"/>)</li> </ul> </li> <li>The profile in any "m=" line in any generated answerMUST<bcp14>MUST</bcp14> exactly match the profile provided in theoffer.</t> <t>Becauseoffer.</li> <li>Because DTLS-SRTP isREQUIRED,<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 negotiationfailure.</t> <t>Thefailure.</li> <li>The use of AVPF or AVP simply controls the timing rules used for RTCP feedback. If AVPF isprovided,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 anAVP compatibleAVP-compatible mode and use a value of"trr-int=4000".</t> <t>For"trr-int=4000".</li> <li>For datam="m=" sections, implementationsMUST<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 implementationsMUST<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="Constructingnumbered="true" toc="default"> <name>Constructing anOffer">Offer</name> <t>When createOffer is called, a new SDP description must be created that includes the functionality specified in <xreftarget="I-D.ietf-rtcweb-rtp-usage"></xref>.target="RFC8834" format="default"/>. The exact details of this process are explained below.</t> <sectiontitle="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 <xreftarget="RFC4566"></xref>, Section 5.target="RFC4566" sectionFormat="comma" section="5"/>. Specifically:<list style="symbols"> <t>The</t> <ul spacing="normal"> <li>The first SDP lineMUST<bcp14>MUST</bcp14> be "v=0", as specified in <xreftarget="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], Section5.1</t> <t>The5.1 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 <username> fieldSHOULD<bcp14>SHOULD</bcp14> be "-". The sess-idMUST<bcp14>MUST</bcp14> be representable by a 64-bit signed integer, and the valueMUST<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 thehighest bit setability tozerouse 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 <nettype> <addrtype> <unicast-address> tupleSHOULD<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 <xreftarget="I-D.ietf-rtcweb-ip-handling" />.target="RFC8828" format="default"/>. As mentioned in <xreftarget="RFC4566"></xref>,target="RFC4566" format="default"/>, the entireo="o=" line needs to be unique, but selecting a random number for <sess-id> is sufficient to accomplishthis.</t> <t>Thethis.</li> <li>The third SDP lineMUST<bcp14>MUST</bcp14> be a "s=" line, as specified in <xreftarget="RFC4566"></xref>, Section 5.3;target="RFC4566" sectionFormat="comma" section="5.3"/>; to match the "o=" line, a single dashSHOULD<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 seemsclearer.</t> <t>Sessionclearer.</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 andSHOULD NOT<bcp14>SHOULD NOT</bcp14> beincluded.</t> <t>Encryptionincluded.</li> <li>Encryption Keys ("k=") lines do not provide sufficient security andMUST NOT<bcp14>MUST NOT</bcp14> beincluded.</t> <t>Aincluded.</li> <li>A "t=" lineMUST<bcp14>MUST</bcp14> be added, as specified in <xreftarget="RFC4566"></xref>, Section 5.9;target="RFC4566" sectionFormat="comma" section="5.9"/>; both <start-time> and <stop-time>SHOULD<bcp14>SHOULD</bcp14> be set to zero,e.g.e.g., "t=00".</t> <t>An0".</li> <li>An "a=ice-options" line with the "trickle" and "ice2" optionsMUST<bcp14>MUST</bcp14> be added, as specified in <xreftarget="I-D.ietf-ice-trickle"></xref>, Section 3target="RFC8840" sectionFormat="comma" section="4.1.1"/> and <xreftarget="RFC8445"></xref>,target="RFC8445" sectionFormat="comma" section="10"/>. <!-- [rfced] Section10.</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, asdescribedspecified in<xref target="I-D.ietf-rtcweb-security-arch" />,[I-D.ietf-ice-trickle], Section5.</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], Section5.14. An m= section is generated for each RtpTransceiver4.1.1." Please note thathas been added tothePeerConnection, excluding any stopped RtpTransceivers; this is done"OLD" text does not match what we found in theorder the RtpTransceivers were added toprovided draft: o An "a=ice-options" line with thePeerConnection. If there are no such RtpTransceivers, no m= sections are generated; more can"trickle" and "ice2" options MUST beadded later,added, asdiscussedspecified in<xref target="RFC3264" />,[I-D.ietf-ice-trickle], Section5.</t> <t>For each m= section generated for an RtpTransceiver, establish a mapping between the transceiver3 and [RFC8445], Section 10. We updated as follows. Is theindex of"and [RFC8445], Section 10" still applicable? Currently: * An "a=ice-options" line with thegenerated 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-onlym="m=" sectionsMUST NOT<bcp14>MUST NOT</bcp14> contain any ICE credentials andMUST NOT<bcp14>MUST NOT</bcp14> gather any candidates.</t> <t>For DTLS, allm="m=" sectionsMUST<bcp14>MUST</bcp14> use any and allthe certificate(s)certificates that have been specified for the PeerConnection; as a result, theyMUST<bcp14>MUST</bcp14> all have the same<xref target="RFC8122"></xref>fingerprintvalue(s),value or values <xref target="RFC8122" format="default"/>, or thesevalue(s) MUSTvalues <bcp14>MUST</bcp14> be session-level attributes.</t> <t>Eachm="m=" section should be generated as specified in <xreftarget="RFC4566"></xref>, Section 5.14.target="RFC4566" sectionFormat="comma" section="5.14"/>. For them="m=" line itself, the following rulesMUST<bcp14>MUST</bcp14> be followed:<list style="symbols"> <t>If</t> <ul spacing="normal"> <li>If them="m=" section is marked as bundle-only, then the port valueMUST<bcp14>MUST</bcp14> be set to 0. Otherwise, the port value is set to the port of the default ICE candidate for thism="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 <xreftarget="I-D.ietf-ice-trickle"></xref>, Section 5.1.</t> <t>Totarget="RFC8840" sectionFormat="comma" section="4.1.1"/>.</li> <!-- This instance of ", as specified in" is OK. --> <li>To properly indicate use of DTLS, the <proto> fieldMUST<bcp14>MUST</bcp14> be set to "UDP/TLS/RTP/SAVPF", as specified in <xref target="RFC5764"/>, Section 8.</t> <t>IfsectionFormat="comma" section="8"/>.</li> <li>If codec preferences have been set for the associated transceiver, media formatsMUST<bcp14>MUST</bcp14> be generated in the correspondingorder,order andMUST<bcp14>MUST</bcp14> exclude any codecs not present in the codecpreferences.</t> <t>Unlesspreferences.</li> <li>Unless excluded by the above restrictions, the media formatsMUST<bcp14>MUST</bcp14> include the mandatory audio/video codecs as specified in <xreftarget="RFC7874"></xref>, Section 3,target="RFC7874" sectionFormat="comma" section="3"/> and <xreftarget="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], Section5.7.3, 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 <xreftarget="I-D.ietf-ice-trickle"></xref>, Section 5.1.</t>target="RFC8840" sectionFormat="comma" section="4.1.1"/>.</t> <t> <xreftarget="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 TRANSPORTMUST NOT<bcp14>MUST NOT</bcp14> be repeated in bundledm="m=" sections, repeating the guidance from <xreftarget="I-D.ietf-mmusic-sdp-bundle-negotiation" />, Section 8.1. This includes m= sections for which bundling has been negotiatedtarget="RFC8843" sectionFormat="comma" section="8.1"/>. <!-- [rfced] Sections 5.2.1 andis still desired, as well as m= sections marked as bundle-only.</t> <t>The following5.2.2: 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 eachm="m=" section:</t><t> <list style="symbols"> <t>An<ul spacing="normal"> <li>An "a=mid" line, as specified in <xreftarget="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] Section4.5.2.1: 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 thispoint.</t> <t>Apoint.</li> <li>A direction attributewhichthat is the same as that of the associatedtransceiver.</t> <t>Fortransceiver.</li> <li>For each media format on them="m=" line, "a=rtpmap" and "a=fmtp" lines, as specified in <xreftarget="RFC4566"></xref>, Section 6,target="RFC4566" sectionFormat="comma" section="6"/> and <xreftarget="RFC3264"></xref>, Section 5.1.</t> <t>Fortarget="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 <xreftarget="RFC4588"></xref>, Section 8.1.</t> <t>Fortarget="RFC4588" sectionFormat="comma" section="8.1"/>.</li> <li>For each supportedFECForward Error Correction (FEC) mechanism, "a=rtpmap" and "a=fmtp" lines, as specified in <xreftarget="RFC4566"></xref>, Section 6.target="RFC4566" sectionFormat="comma" section="6"/>. The FEC mechanisms thatMUST<bcp14>MUST</bcp14> be supported are specified in <xreftarget="I-D.ietf-rtcweb-fec"></xref>, Section 6,target="RFC8854" sectionFormat="comma" section="6"/>, and specific usage for each media type is outlined in Sections4<xref target="sec.interface" format="counter"/> and5.</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] isfor media with configurable durationsthe correct section to cite here; we could not see a relationship. Also, should "Sections 4 and 5" be "Sections 4 and 5 ofmedia per packet, e.g., audio, an "a=maxptime" line, indicatingRFC 8854 [I-D.ietf-rtcweb-fec]"? The hyperlinks in themaximum amount.html file steer to Sections 4 and 5 ofmedia, specifiedthis document, and we could not find relevant text inmilliseconds,those sections. Original: The FEC mechanisms thatcanMUST beencapsulated in eachsupported 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 <xreftarget="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 them= section.</t> <t>If"m=" section.</li> <li>If thism="m=" section is for videomedia,media and there are known limitations on the size of imageswhichthat can be decoded, an "a=imageattr" line, as specified in <xreftarget="sec.imageattr"></xref>.</t> <t>Fortarget="sec.imageattr" format="default"/>.</li> <li>For each supported RTP header extension, an "a=extmap" line, as specified in <xreftarget="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 thatSHOULD/MUST<bcp14>SHOULD</bcp14>/<bcp14>MUST</bcp14> be supported is specified in <xreftarget="I-D.ietf-rtcweb-rtp-usage"></xref>, Section 5.2.target="RFC8834" sectionFormat="comma" section="5.2"/>. Any header extensions that require encryptionMUST<bcp14>MUST</bcp14> be specified as indicated in <xreftarget="RFC6904"></xref>, Section 4.</t> <t>Fortarget="RFC6904" sectionFormat="comma" section="4"/>.</li> <li>For each supported RTCP feedback mechanism, an "a=rtcp-fb" line, as specified in <xreftarget="RFC4585"></xref>, Section 4.2.target="RFC4585" sectionFormat="comma" section="4.2"/>. The list of RTCP feedback mechanisms thatSHOULD/MUST<bcp14>SHOULD</bcp14>/<bcp14>MUST</bcp14> be supported is specified in <xreftarget="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 <xreftarget="I-D.ietf-mmusic-msid"></xref>, Section 2,target="RFC8830" sectionFormat="comma" section="2"/>, but omitting the "appdata"field.</t> </list></t> <t>Iffield.</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 <xreftarget="I-D.ietf-mmusic-rid"></xref>,target="RFC8851" format="default"/>, and its directionMUST<bcp14>MUST</bcp14> be "send". If the application has chosen a RID value, itMUST<bcp14>MUST</bcp14> be used as the rid-identifier;otherwiseotherwise, a RID valueMUST<bcp14>MUST</bcp14> be generated by the implementation. RID valuesMUST<bcp14>MUST</bcp14> be generated in a fashion that does not leak user information, e.g., randomly or using a per-PeerConnection counter, andSHOULD<bcp14>SHOULD</bcp14> be 3 bytes or less, to allow them to efficiently fit into the RTP header extension defined in <xreftarget="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 aregenerated.</t> <t>Ifgenerated.</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 <xreftarget="I-D.ietf-mmusic-sdp-simulcast"></xref>, Section 6.2.target="RFC8853" sectionFormat="comma" section="6.2"/>. The list of RIDsMUST<bcp14>MUST</bcp14> include all of the RID identifiers used in the "a=rid" lines for thism= 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=" sectionswhichthat either have a unique address orwhichare associated with thebundle-tag.BUNDLE-tag. (In initial offers, this means those "m=" sectionswhichthat 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 <xreftarget="I-D.ietf-mmusic-ice-sip-sdp"></xref>, Section 4.4.</t> <t>Fortarget="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 <xreftarget="RFC8122"></xref>, Section 5.</t> <t>Antarget="RFC8122" sectionFormat="comma" section="5"/>.</li> <li>An "a=setup" line, as specified in <xreftarget="RFC4145"></xref>, Section 4,target="RFC4145" sectionFormat="comma" section="4"/> and clarified for use in DTLS-SRTP scenarios in <xreftarget="RFC5763"></xref>, Section 5.target="RFC5763" sectionFormat="comma" section="5"/>. The role value in the offerMUST<bcp14>MUST</bcp14> be"actpass".</t> <t>An"actpass".</li> <li>An "a=tls-id" line, as specified in <xreftarget="I-D.ietf-mmusic-dtls-sdp" />, Section 5.2.</t> <t>Antarget="RFC8842" sectionFormat="comma" section="5.2"/>.</li> <li>An "a=rtcp" line, as specified in <xreftarget="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 beengathered.</t> <t>Angathered. ... 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 <xreftarget="RFC5761"></xref>, Section 5.1.3.</t> <t>Iftarget="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 <xreftarget="I-D.ietf-mmusic-mux-exclusive" />, Section 4.</t> <t>Antarget="RFC8858" sectionFormat="comma" section="4"/>.</li> <li>An "a=rtcp-rsize" line, as specified in <xreftarget="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=" sectionMUST<bcp14>MUST</bcp14> be generated for data. The <media> fieldMUST<bcp14>MUST</bcp14> be set to"application""application", and the <proto> fieldMUST<bcp14>MUST</bcp14> be set to "UDP/DTLS/SCTP" <xreftarget="I-D.ietf-mmusic-sctp-sdp"></xref>.target="RFC8841" format="default"/>. The "fmt" valueMUST<bcp14>MUST</bcp14> be set to "webrtc-datachannel" as specified in <xreftarget="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], Section4.1.</t>4.1. --> </t> <t>Within the datam="m=" section, an "a=mid" lineMUST<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 <xreftarget="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 <xreftarget="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 datam="m=" section either has a unique address or is associated with thebundle-tagBUNDLE-tag (e.g., if it is the onlym="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 allm="m=" sections have been generated, a session-level "a=group" attributeMUST<bcp14>MUST</bcp14> be added as specified in <xreftarget="RFC5888"></xref>.target="RFC5888" format="default"/>. This attributeMUST<bcp14>MUST</bcp14> have semantics"BUNDLE","BUNDLE" andMUST<bcp14>MUST</bcp14> include the mid identifiers of eachm="m=" section. The effect of this is that the JSEP implementation offers allm="m=" sections as one bundle group. However, whether them="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>Attributeswhichthat SDP permits toeitherbe at either the session level or the media levelSHOULD<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, implementationsMAY<bcp14>MAY</bcp14> choose to aggregate attributes at the sessionlevellevel, and JSEP implementationsMUST<bcp14>MUST</bcp14> be prepared to receive attributes in either location.</t> <t>Attributes other than the ones specified aboveMAY<bcp14>MAY</bcp14> be included, except for the followingattributesattributes, which are specifically incompatible with the requirements of <xreftarget="I-D.ietf-rtcweb-rtp-usage"></xref>,target="RFC8834" format="default"/> andMUST 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 addedMUST<bcp14>MUST</bcp14> follow the advice in <xreftarget="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. ImplementationsMUST<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> <sectiontitle="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=" lineMUST<bcp14>MUST</bcp14> stay the same except for the <session-version> field, whichMUST<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; implementationsMAY<bcp14>MAY</bcp14> opt to increment <session-version> on every call. The value of the generated <session-version> is independent of the <session-version> of the current local description; in particular, in the case where the current version is N, an offer is created and applied with version N+1, and then that offer is rolled back so that the current version is again N, the next generated offer will still have versionN+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 <session-version> will not have changed. There are no known scenarios in which this causes problems, but if this is a concern, the solution is simply to use createOffer to ensure a unique <session-version>.</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=" linesMUST<bcp14>MUST</bcp14> stay thesame.</t> <t>Ifsame.</li> <li>If any RtpTransceiver has beenadded,added and there exists anm="m=" section with a zero port in the current local description or the current remote description, thatm="m=" sectionMUST<bcp14>MUST</bcp14> be recycled by generating anm="m=" section for the added RtpTransceiver as if them="m=" section were being added to the session description (including a new MIDvalue),value) and placing it at the same index as them="m=" section with a zeroport.</t> <t>Ifport.</li> <li>If an RtpTransceiver is stopped and is not associated with anm="m=" section, anm="m=" sectionMUST NOT<bcp14>MUST NOT</bcp14> be generated for it. This prevents adding back RtpTransceivers whosem="m=" sections were recycled and used for a new RtpTransceiver in a previous offer/ answer exchange, as describedabove.</t> <t>Ifabove.</li> <li>If an RtpTransceiver has been stopped and is associated with anm="m=" section, and them="m=" section is not being recycled as described above, anm="m=" sectionMUST<bcp14>MUST</bcp14> be generated for it with the port set to zero and all "a=msid" linesremoved.</t> <t>Forremoved.</li> <li>For RtpTransceivers that are not stopped, the "a=msid"line(s) MUSTline 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 initialoffer.</t> <t>Eachoffer.</li> <li>Each "m=" andc=""c=" lineMUST<bcp14>MUST</bcp14> be filled in with the port, relevant RTP profile, and address of the default candidate for them="m=" section, as described in <xreftarget="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 <xreftarget="sec.profile-names"/>.target="sec.profile-names" format="default"/>. If no RTP candidates have yet been gathered, dummy valuesMUST<bcp14>MUST</bcp14> still be used, as describedabove.</t> <t>Eachabove.</li> <li>Each "a=mid" lineMUST<bcp14>MUST</bcp14> stay thesame.</t> <t>Eachsame.</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 them="m=" section is bundled into anotherm="m=" section, it stillMUST NOT<bcp14>MUST NOT</bcp14> contain any ICEcredentials.</t> <t>Ifcredentials.</li> <li>If them="m=" section is not bundled into anotherm="m=" section, its "a=rtcp" attribute lineMUST<bcp14>MUST</bcp14> be filled in with the port and address of the default RTCP candidate, as indicated in <xreftarget="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 sectionabove.</t> <t>Ifabove. ... The process here is identical to that indicated in them=initial offers sectionis not bundledabove, 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 anotherm="m=" section, for each candidate that has been gathered during the most recent gathering phase (see <xreftarget="sec.ice-gather-overview"></xref>),target="sec.ice-gather-overview" format="default"/>), an "a=candidate" lineMUST<bcp14>MUST</bcp14> be added, as defined in <xreftarget="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" attributeMUST<bcp14>MUST</bcp14> be added, as described in <xreftarget="I-D.ietf-ice-trickle"></xref>, Section 9.3.target="RFC8840" sectionFormat="comma" section="8.2"/>. If them="m=" section is bundled into anotherm="m=" section, both "a=candidate" and "a=end-of-candidates"MUST<bcp14>MUST</bcp14> beomitted.</t> <t>For RtpTransceivers that are still present, the "a=rid" lines MUST stayomitted. <!-- [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 thesame.</t> <t>For RtpTransceiversm= section is not bundled into another m= section, for each candidate thatare 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 sidehas beenapplied using setRemoteDescription, meaninggathered during thePeerConnection ismost 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, anoffer 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 thenew offer, the following adjustments are made based on the contents of the correspondingm= sectionin the current local or remote description, as appropriate: <list style="symbols"> <t>Theis bundled into another m=line and corresponding "a=rtpmap"section, both "a=candidate" and"a=fmtp" lines"a=end-of-candidates" MUSTonly include media formats which have not been excluded by the codec preferences ofbe omitted. NEW: o If theassociated transceiver, and MUST include all currently available formats. Media formatsm= section is not bundled into another m= section, for each candidate thatwere previously offered but are no longer available (e.g., a shared hardware codec) MAY be excluded.</t> <t>Unless codec preferences havehas beenset for the associated transceiver, the media formats ongathered during them=most recent gathering phase (see Section 3.5.1), an "a=candidate" line MUST begeneratedadded, as defined in [RFC5245], Section 4.3., paragraph 3. If candidate gathering for thesame ordersection 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 themost recent answer. Any media formatsm= section is bundled into another m= section, both "a=candidate" and "a=end-of-candidates" MUST be omitted." Please note thatwerethe "OLD" text does notpresentmatch what we found in themost recent answer MUST be added after all existing formats.</t> <t>The RTP header extensions MUST only include thoseprovided 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 thatare present inhas been gathered during the most recentanswer.</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 MUSTNOTbeadded ifadded, as defined in [I-D.ietf-mmusic-ice-sip-sdp], Section 4.1. If candidate gathering for themost recent answer includedsection has completed, an"a=rtcp-mux" line.</t> <t>The "a=rtcp-mux" line"a=end-of- candidates" attribute MUST bethe sameadded, asthatdescribed in [I-D.ietf-ice-trickle], Section 9.3. If themost recent answer.</t> <t>The "a=rtcp-mux-only" line MUST NOT be added.</t> <t>The "a=rtcp-rsize" linem= section is bundled into another m= section, both "a=candidate" and "a=end-of- candidates" MUSTNOTbeadded unless present inomitted. 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 recentanswer.</t> <t>An "a=bundle-only"gathering phase (see Section 3.5.1), an "a=candidate" line MUSTNOTbe added, asindicateddefined in<xref target="I-D.ietf-mmusic-sdp-bundle-negotiation" />,[RFC8839], Section6. Instead, JSEP implementations MUST simply omit parameters in the IDENTICAL and TRANSPORT categories5.1. If candidate gathering forbundled 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], Section8.1.</t> <t>Note that if media m= sections are8.2. If the "m=" section is bundled intoa data m=another "m=" section,then certain TRANSPORTboth "a=candidate" andIDENTICAL attributes may appear in the data m= section even if they would otherwise only"a=end- of-candidates" MUST beappropriate for a media m= section (e.g., "a=rtcp-mux"). This cannot happen in initial offers because inomitted. --> </li> <li>For RtpTransceivers that are still present, theinitial offer JSEP implementations always list media m= sections (if any) before"a=rid" lines <bcp14>MUST</bcp14> stay thedata 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 apreceding non-bundle-only media m= section.</t> </list></t> <t>The "a=group:BUNDLE" attribute MUST includecorresponding answer from theMID identifiers specifiedremote side has been applied using setRemoteDescription, meaning the PeerConnection is in thebundle 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 themost recent answer, minus any m= sectionsnew 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 beenmarked as rejected, plus any newly added or re-enabled m= sections. In other words,excluded by thebundle attribute must containcodec preferences of the associated transceiver and also <bcp14>MUST</bcp14> include allm= sectionscurrently available formats. Media formats that were previouslybundled, as long as they are still alive, as well as any new m= sections.</t> <t>"a=group:LS" attributesoffered 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 samewayorder asfor initial offers, within theadditional stipulation that any lip sync groupsmost recent answer. Any media formats that were not present in the most recent answerMUST continue to exist and MUST contain any previously<bcp14>MUST</bcp14> be added after all existingMID identifiers, as long as the identified m= sections still exist andformats.</li> <li>The RTP header extensions <bcp14>MUST</bcp14> only include those that arenot rejected, andpresent in thegroup still contains at least two MID identifiers. This ensuresmost recent answer.</li> <li>The RTCP feedback mechanisms <bcp14>MUST</bcp14> only include those thatany synchronized "recvonly" m= sections continue to be synchronizedare present in thenew 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 ifmost recent answer, except for thefollowing optionscase of format-specific mechanisms that arepresent.</t> <section title="IceRestart" anchor="sec.icerestart"> <t>If the "IceRestart" option is specified, withreferencing avalue of "true",newly added media format.</li> <li>The "a=rtcp" line <bcp14>MUST NOT</bcp14> be added if theoffer MUST indicatemost recent answer included anICE 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 asspecifiedthat 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, iftheICE configuration has changed, this option has no effect, since new ufrag and pwd attributes willmost recent answer.</li> <li>The "a=rtcp-mux-only" line <bcp14>MUST NOT</bcp14> begenerated automatically. This option is primarily useful for reestablishing connectivityadded.</li> <li>The "a=rtcp-rsize" line <bcp14>MUST NOT</bcp14> be added unless present incases where failures are detected bytheapplication.</t> </section> <section title="VoiceActivityDetection" anchor="sec.voiceactivitydetection1"> <t>Silence suppression, also knownmost recent answer.</li> <li>An "a=bundle-only" line <bcp14>MUST NOT</bcp14> be added, asdiscontinuous transmission ("DTX"), can reduce the bandwidth used for audio by switching to a special encoding when voice activity isindicated in <xref target="RFC8843" sectionFormat="comma" section="6"/>. <!-- [rfced] Section 5.2.2: We could notdetected, 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 specifiedfind any indication in<xref target="RFC3389"></xref>,RFC 8843 [I-D.ietf-mmusic-sdp-bundle-negotiation], Section5.1, 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 for6 thatcodecan "a=bundle-only" line MUST NOT bespecifiedadded. Will this be clear toindicate 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" />, wouldreaders? Original: o An "a=bundle-only" line MUST NOT beusedadded, as indicated inthe 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 NOTemit "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 wouldbespecifiedadded. --> Instead, JSEP implementations <bcp14>MUST</bcp14> simply omit parameters in theoffer. In addition, the implementation MUST NOT use silence suppressionIDENTICAL and TRANSPORT categories for bundled "m=" sections, as described in <xref target="RFC8843" sectionFormat="comma" section="8.1"/>.</li> <li>Note that if mediait 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 thepeer's description. The impact of these rules is that silence suppressiondata "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 JSEPdepends on mutual agreement of both sides, which ensures consistent handling regardlessimplementations always list media "m=" sections (if any) before the data "m=" section (if any), and at least one ofwhich codec is used.</t> <t>The "VoiceActivityDetection" option doesthose media "m=" sections will not have the "a=bundle-only" attribute. Therefore, in initial offers, anyimpact 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 thesetting ofMID identifiers specified in the"vad" valuebundle group in thesignaling ofmost recent answer, minus any "m=" sections that have been marked as rejected, plus any newly added or re-enabled "m=" sections. In other words, theclient 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 descriptionbundle attribute mustbe createdcontain all "m=" sections thatis compatible with the supplied remote descriptionwere previously bundled, as long as they are still alive, as well asthe requirements specified in <xref target="I-D.ietf-rtcweb-rtp-usage"></xref>. The exact details of this processany new "m=" sections.</t> <t>"a=group:LS" attributes areexplained below.</t> <section title="Initial Answers" anchor="sec.initial-answers"> <t>When createAnswer is calledgenerated in the same way as for initial offers, with thefirst time after a remote description has been provided,additional stipulation that any lip sync groups that were present in theresult is knownmost recent answer <bcp14>MUST</bcp14> continue to exist and <bcp14>MUST</bcp14> contain any previously existing MID identifiers, as long as theinitial 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 endpointidentified "m=" sections still exist andmay 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 attributesaremissing, or functionality listed as mandatory-to-use above isnotpresent, this MUST be treated as an error,rejected, andMUST causetheaffected m=group still contains at least two MID identifiers. This ensures that any synchronized "recvonly" "m=" sections continue to bemarked as rejected.</t> <t>The first stepsynchronized ingeneratingthe 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 aninitial answer is to generate session-level attributes. The process hereRTCOfferOptions object. Special processing isidentical to that indicated inperformed when generating an SDP description if theinitial offers section above, except thatfollowing 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 andthe "ice2" optionpwd attributes, as specified in <xreftarget="RFC8445"></xref>, Section 10, is only included if such antarget="RFC8839" sectionFormat="comma" section="4.4.3.1.1"/>. If this optionwas present in the offer.</t> <t>The next stepisto generate session-level lip sync groups, as defined in <xref target="RFC5888" />, Section 7. For each group of type "LS" present in thespecified on an initial offer,select the local RtpTransceivers thatit has no effect (since a new ICE ufrag and pwd arereferenced byalready generated). Similarly, if theMID values in the specified group,ICE configuration has changed, this option has no effect, since new ufrag anddetermine which of them either reference a common local MediaStream (specifiedpwd attributes will be generated automatically. This option is primarily useful for reestablishing connectivity in cases where failures are detected by thecalls to addTrack/addTransceiverapplication.</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 tocreate them), or have no MediaStream to reference because they werea special encoding when voice activity is notcreated by addTrack/addTransceiver. Ifdetected, atleast two such RtpTransceivers exist, a group of type "LS" withthemid valuescost ofthese RtpTransceivers MUST be added. Otherwise the offered "LS" group MUST be ignored and no corresponding group generated insome fidelity.</t> <t>If theanswer.</t> <t>As"VoiceActivityDetection" option is specified, with asimple example, considervalue of "true", thefollowingofferof a single<bcp14>MUST</bcp14> indicate support for silence suppression in the audioand single video track containedit 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, thesame MediaStream. SDP lines not relevantappropriate fmtp parameters for that codec <bcp14>MUST</bcp14> be specified tothis example have been removedindicate that silence suppression forclarity. As explainedreceived audio is desired. For example, when using the Opus codec <xref target="RFC6716" format="default"/>, the "usedtx=1" parameter, specified in <xreftarget="sec-create-offer" />,target="RFC7587" format="default"/>, would be used in the offer.</t> <t>If the "VoiceActivityDetection" option is specified, with agroupvalue oftype "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", theanswerer uses a single MediaStream when it adds its tracks, both of its transceivers will reference this stream, and soJSEP implementation <bcp14>MUST NOT</bcp14> emit "CN" codecs. For codecs that have their own internal silence suppression support, thesubsequent answer will contain a "LS" group identicalappropriate 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 theoffer, 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, ifoffer. In addition, theanswerer groups its tracks into separate MediaStreams, its transceivers will reference different streams, and soimplementation <bcp14>MUST NOT</bcp14> use silence suppression for media it generates, regardless of whether thesubsequent 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 theanswererpeer'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 notadd any tracks, its transceivers will not referencehave anyMediaStreams, causingimpact on thepreferencessetting of theofferer to be maintained, and so"vad" value in thesubsequent 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 casesignaling of"LS" group generation.</t> <t>The next stepthe 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 isto generate m= sections for each m= sectioncalled, a new SDP description must be created that ispresent incompatible with the supplied remoteoffer,description as well as the requirements specified in <xreftarget="RFC3264"></xref>, Section 6. For the purposestarget="RFC8834" format="default"/>. The exact details of thisdiscussion, any session-level attributes in the offer thatprocess arealso validexplained 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 asmedia-level attributes are considered tothe initial answer. If no remote description has been installed, an answer cannot bepresent in each m= section. Each offered m= section will havegenerated, and anassociated 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 theunmatched RtpTransceivers will needremote description SDP may not have been created by a JSEP endpoint and may not conform tobe associatedall the requirements listed in <xref target="sec-create-offer" format="default"/>. For many cases, this is not asubsequent offer.</t> <t>For each offered m= section,problem. However, if anyof the following conditionsmandatory SDP attributes aretrue, the corresponding m= section in the answer MUSTmissing or functionality listed as mandatory-to-use above is not present, this <bcp14>MUST</bcp14> bemarkedtreated asrejected by settingan error and <bcp14>MUST</bcp14> cause theport in the m= lineaffected "m=" sections tozero,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 <xreftarget="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 oftarget="sec.initial-offers"/> above, except that theoffered 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 thefirst m= section or"trickle" option as specified in <xref target="RFC8840" sectionFormat="comma" section="4.1.3"/> and thesame 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 firstm= section.</t> <t>The bundle policystep in generating an initial answer is"balanced", and thisto generate session-level attributes. The process here isnotidentical to that indicated in thefirst m=initial offers sectionfor this media type or inabove, except that thesame 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 firstm= section for this media type.</t> <t>This m= sectionstep in generating an initial answer is to generate session-level attributes. The process here is identical to that indicated ina bundle group, andthegroup's offerer tagged m=initial offers sectionis being rejected due to one ofabove, except that theabove reasons. This requires all m= sections in"a=ice-options" line, with thebundle 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], Section7.3.3.</t> </list></t> <t>Otherwise, each m= section in the answer should then be generated as specified4.1.3, is only included if such an option was present in<xref target="RFC3264"></xref>, Section 6.1. Forthem= line itself,offer." Please note that thefollowing 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 theport ofprovided draft: The first step in generating an initial answer is to generate session-level attributes. The process here is identical to that indicated in thedefault ICE candidate for this m= section, but giveninitial offers section above, except thatno candidates are available yet,the"dummy" port value of 9 (Discard) MUST be used,"a=ice-options" line, with the "trickle" option asindicatedspecified in<xref target="I-D.ietf-ice-trickle"></xref>,[I-D.ietf-ice-trickle], Section5.1.</t> <t>The <proto> field MUST be set to exactly match the <proto> field for3, and thecorresponding m= line"ice2" option as specified inthe 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 thecorresponding order, regardlessoffer. We updated as follows. As noted previously, (1) for ease ofwhat was offered,the reader andMUST exclude any codecs not present(2) to create a usable hyperlink in thecodec 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 themedia formats on"a=ice-options" line, with them= line MUST be generated"trickle" option as specified in [RFC8840], Section 4.1.3 and thesame order"ice2" option asthose offeredspecified in [RFC8445], Section 10, is only included if such an option was present in thecurrent 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 thatoffer. --> </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 formatsselect the local RtpTransceivers that arelocally supported but not present inreferenced by theoffer, as mentionedMID values in<xref target="RFC3264" />, Section 6.1. If no common format exists,them= section is rejected as described above.</t> </list></t> <t>The m= line MUST be followed immediately by a "c=" line, asspecified 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>Iftheoffer supports bundle, all m= sectionscalls 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> bebundled must useadded. Otherwise, thesame ICE credentials and candidates; all m= sections not being bundled must use unique ICE credentialsoffered "LS" group <bcp14>MUST</bcp14> be ignored andcandidates. Each m= section MUST containno corresponding group generated in the answer.</t> <t>As a simple example, consider the followingattributes (which areoffer ofattribute types other than IDENTICAL and TRANSPORT): <list style="symbols"> <t>Ifa single audio andonly 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 specifiedsingle video track contained in theoffer.</t> <t>A direction attribute, determined by applying the rules regarding the offered direction specifiedsame MediaStream. SDP lines not relevant to this example have been removed for clarity. As explained in <xreftarget="RFC3264" />, Section 6.1, and then intersecting with the directiontarget="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 theassociated 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 thelocal transceiver is set to "sendrecv", the result in thesubsequent answeriswill 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 ofas 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 theprimary codecanswerer groups its tracks into separate MediaStreams, its transceivers will reference different streams, andan "a=fmtp" line that referencesso thepayload type ofsubsequent 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 theprimary codec, as specifiedanswerer 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 <xreftarget="RFC4588"></xref>, Section 8.1.</t> <t>Fortarget="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 eachsupported FEC mechanism, "a=rtpmap" and "a=fmtp" lines,"m=" section that is present in the remote offer, as specified in <xreftarget="RFC4566"></xref>, Section 6. The FEC mechanismstarget="RFC3264" sectionFormat="comma" section="6"/>. For the purposes of this discussion, any session-level attributes in the offer thatMUST be supportedarespecifiedalso valid as media-level attributes are considered to be present in<xref target="I-D.ietf-rtcweb-fec"></xref>, Section 6, and specific usage foreachmedia type is outlined in Sections 4 and 5.</t> <t>If this m="m=" section. Each offered "m=" sectionis for media with configurable durations of media per packet, e.g., audio,will have an"a=maxptime" line,associated RtpTransceiver, as described in <xreftarget="sec-create-offer" />.</t> <t>If this m= section is for video media, andtarget="sec.applying-a-remote-desc" format="default"/>. If there areknown limitations onmore RtpTransceivers than there are "m=" sections, thesize of images which canunmatched RtpTransceivers will need to bedecoded, an "a=imageattr" line, as specifiedassociated in<xref target="sec.imageattr"></xref>.</t>a subsequent offer.</t> <t>For eachsupported RTP header extension that is presentoffered "m=" section, if any of the following conditions are true, the corresponding "m=" section in theoffer, an "a=extmap" line,answer <bcp14>MUST</bcp14> be marked asspecifiedrejected by setting the port in the "m=" line to zero, as indicated in <xreftarget="RFC5285"></xref>, Section 5. The list of header extensions that SHOULD/MUSTtarget="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 supportedis 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] Section4.</t> <t>For each supported RTCP feedback mechanism5.3.1: Does this text mean thatis present intheoffer, an "a=rtcp-fb" line, as specified in <xref target="RFC4585"></xref>, Section 4.2. The list of RTCP feedback mechanisms that SHOULD/MUSToffered 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 isspecified in <xref target="I-D.ietf-rtcweb-rtp-usage"></xref>, Section 5.1.</t> <t>If"max-bundle", and this is not theRtpTransceiver has a sendrecvfirst "m=" section orsendonly direction: <list style="symbols"> <t>For each MediaStream that was associated within thetransceiver when it was created via addTrack or addTransceiver, an "a=msid" line,same bundle group asspecified in <xref target="I-D.ietf-mmusic-msid"></xref>, Section 2, but omittingthe"appdata" field.</t> </list></t> </list></t> <t>Each m= section whichfirst "m=" section.</li> <li>The bundle policy is "balanced", and this is notbundled into another m= section, MUST containthefollowing attributes (which are of category IDENTICALfirst "m=" section for this media type orTRANSPORT):</t> <t> <list style="symbols"> <t>"a=ice-ufrag" and "a=ice-pwd" lines,in the same bundle group asspecifiedthe 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 oneor more "a=fingerprint" lines for eachof theendpoint's certificates,above reasons. This requires all "m=" sections in the bundle group to be rejected, as specified in <xreftarget="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 valuetarget="RFC8843" sectionFormat="comma" section="7.3.3"/>.</li> </ul> <t>Otherwise, each "m=" section in the answerMUSTshould then be"active" or "passive". Whengenerated as specified in <xref target="RFC3264" sectionFormat="comma" section="6.1"/>. For theoffer contains"m=" line itself, the"actpass" value, as will alwaysfollowing 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 thecase with JSEP endpoints, the answerer SHOULD usem= line itself, the"active" role. Offers from non-JSEP endpoints MAY send other values for "a=setup",following rules MUST be followed:" inwhich caseSection 5.2.1 and (2) all of theanswer MUST userules listed below this sentence include avalue consistent with"MUST.") Original: For thevalue inm= line itself, theoffer.</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 infollowing rules must be followed: --> </t> <ul spacing="normal"> <li>The port value would normally be set to theoffer, 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, containingport of thedummy value "9 IN IP4 0.0.0.0" (becausedefault ICE candidate for this "m=" section, but given that no candidateshave yet been gathered).</t> <t>If present inare available yet, theoffer, an "a=rtcp-rsize" line,"dummy" port value of 9 (Discard) <bcp14>MUST</bcp14> be used, asspecifiedindicated in <xreftarget="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 <media> field MUST be set to "application" and thetarget="RFC8840" sectionFormat="comma" section="4.1.1"/>.</li> <li>The <proto>and <fmt> fields MUSTfield <bcp14>MUST</bcp14> be set to exactly match thefields in<proto> field for theoffer.</t> <t>Withincorresponding "m=" line in thedata m= section, an "a=mid" line MUSToffer.</li> <li>If codec preferences have been set for the associated transceiver, media formats <bcp14>MUST</bcp14> be generatedand included as described above, along with an "a=sctp-port" line referencing the SCTP port number, as definedin<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,thefollowing attributescorresponding order, regardless ofcategory 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 TRANSPORTwhat was offered, andIDENTICAL attributes may also appear<bcp14>MUST</bcp14> exclude any codecs not present in thedata m= section even if they would otherwise only be appropriate for acodec preferences.</li> <li>Otherwise, the mediam= section (e.g., "a=rtcp-mux").</t> <t>If "a=group" attributes with semantics of "BUNDLE" are offered, corresponding session-level "a=group" attributes MUSTformats on the "m=" line <bcp14>MUST</bcp14> beadded as specifiedgenerated in<xref target="RFC5888"></xref>. These attributes MUST have semantics "BUNDLE", and MUST include the all mid identifiers fromthe same order as those offeredbundle groupsin the current remote description, excluding any currently unsupported formats. Any currently available media formats thathaveare notbeen rejected. Note that regardless of the presence of "a=bundle-only"present in theoffer, no m= sectionscurrent remote description <bcp14>MUST</bcp14> be added after all existing formats.</li> <li>In either case, the media formats in the answershould 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 atsession-level.</t> <t>The attributes prohibitedleast one format that is present in thecreation of offersoffer but <bcp14>MAY</bcp14> include formats that arealso 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 waslocally supported but notapplied using setLocalDescription, meaning the PeerConnection is stillpresent in the"have-remote-offer" state, the steps for generating an initial answer should be followed, subject tooffer, as mentioned in <xref target="RFC3264" sectionFormat="comma" section="6.1"/>. If no common format exists, thefollowing restriction: <list style="symbols">"m=" section is rejected as described above.</li> </ul> <t>Thefields of the "o=""m=" lineMUST 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<session-version> field, which MUST increment if"c=" line must contain thesession description changes"dummy" value "IN IP4 0.0.0.0", as defined inany way from the previously generated answer.</t> </list></t><xref target="RFC8840" sectionFormat="comma" section="4.1.3"/>.</t> <t>Ifany session description was previously supplied to setLocalDescription, an answer is generated by followingthesteps inoffer 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>Eachcandidates; all "m=" sections not being bundled must use unique ICE credentials andc=" line MUST be filled in withcandidates. Each "m=" section <bcp14>MUST</bcp14> contain theport and addressfollowing attributes (which are of attribute types other than IDENTICAL or TRANSPORT): </t> <ul spacing="normal"> <li>If and only if present in thedefault candidate for the m= section,offer, an "a=mid" line, asdescribedspecified in <xreftarget="I-D.ietf-mmusic-ice-sip-sdp"></xref>, Section 3.2.1.2. Notetarget="RFC5888" sectionFormat="comma" section="9.1"/>. The "mid" value <bcp14>MUST</bcp14> match that specified incertain cases,them= line protocol may not match that ofoffer.</li> <li>A direction attribute, determined by applying thedefault candidate, becauserules regarding them= line protocol value MUST match what was suppliedoffered direction specified inthe offer, as described above.</t> <t>Each "a=ice-ufrag"<xref target="RFC3264" sectionFormat="comma" section="6.1"/>, and"a=ice-pwd" line MUST staythen intersecting with thesame, unlessdirection of them= section is restarting, in which case new ICE credentials must be created as specifiedassociated RtpTransceiver. For example, in<xref target="I-D.ietf-mmusic-ice-sip-sdp"></xref>, Section 3.4.1.1.1. Ifthem=case where an "m=" section isbundled into another m= section, it still MUST NOT contain any ICE credentials.</t> <t>Each "a=tls-id" line MUST stayoffered as "sendonly" and thesame unlesslocal transceiver is set to "sendrecv", theofferer's "a=tls-id" line changed,result inwhich casethe answer is anew "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, asdescribedspecified in <xreftarget="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 associationtarget="RFC4566" sectionFormat="comma" section="6"/> and <xref target="RFC3264" sectionFormat="comma" section="6.1"/>.</li> <li>If "rtx" isbeing continued bypresent in theofferer.</t> <t>RTCP multiplexing mustoffer, for each primary codec where RTP retransmission should be used,and an "a=rtcp-mux"a corresponding "a=rtpmap" lineinserted if and only ifindicating "rtx" with them= section previously used RTCP multiplexing.</t> <t>Ifclock rate of them= section is not bundled into another m= sectionprimary codec andRTCP multiplexing is not active,an"a=rtcp" attribute"a=fmtp" lineMUST be filled in withthat references theport and addresspayload type of thedefault RTCP candidate. If no RTCP candidates have yet been gathered, dummy values MUST be used,primary codec, asdescribedspecified inthe 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 eachcandidate that has been gathered during the most recent gathering phase (seesupported FEC mechanism, "a=rtpmap" and "a=fmtp" lines, as specified in <xreftarget="sec.ice-gather-overview"></xref>), an "a=candidate" line MUSTtarget="RFC4566" sectionFormat="comma" section="6"/>. The FEC mechanisms that <bcp14>MUST</bcp14> beadded, as definedsupported are specified in <xreftarget="I-D.ietf-mmusic-ice-sip-sdp"></xref>, Section 4.1. If candidate gatheringtarget="RFC8854" sectionFormat="comma" section="6"/>, and specific usage fortheeach 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=" sectionhas 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 <xreftarget="I-D.ietf-ice-trickle"></xref>, Section 9.3. If the m=target="sec-create-offer" format="default"/>.</li> <li>If this "m=" section isbundled into another m= section, both "a=candidate"for video media and"a=end-of-candidates" MUST be omitted.</t> <t>For RtpTransceivers thatthere arenot stopped, the "a=msid" line(s) MUST stayknown limitations on thesame, regardlesssize ofchanges to the transceiver's direction or track. If no "a=msid" lineimages 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 thecurrent description, "a=msid" line(s) MUST be generated according to the same rules as foroffer, aninitial answer.</t> </list></t> </section> <section title="Options Handling" anchor="sec.options-handling2"> <t>The createAnswer method takes"a=extmap" line, asa parameter an RTCAnswerOptions object.specified in <xref target="RFC5285" sectionFormat="comma" section="5"/>. Thesetlist ofparameters for RTCAnswerOptions is different than those supported in RTCOfferOptions; the IceRestart option is unnecessary, as ICE credentials will automaticallyheader extensions that <bcp14>SHOULD</bcp14>/<bcp14>MUST</bcp14> bechanged for all m= sections where the offerer chose to perform ICE restart.</t> <t>The following options aresupportedin RTCAnswerOptions.</t> <section title="VoiceActivityDetection" anchor="sec.voiceactivitydetection2"> <t>Silence suppression in the answerishandledspecified in <xref target="RFC8834" sectionFormat="comma" section="5.2"/>. Any header extensions that require encryption <bcp14>MUST</bcp14> be specified asdescribedindicated in <xreftarget="sec.voiceactivitydetection1"></xref>, with one exception: if support for silence suppression was not indicatedtarget="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 generatedan "a=rtcp-fb" line, asif 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 resultspecified inan answer with CN codecs and "usedtx=0").<xref target="RFC4585" sectionFormat="comma" section="4.2"/>. Theimpactlist ofthis rule is that an answerer will not try to use silence suppression with any endpointRTCP feedback mechanisms thatdoes 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> bechanged before passing it to setLocalDescription. If precise control over the SDPsupported isneeded,specified in <xref target="RFC8834" sectionFormat="comma" section="5.1"/>.</li> <li> <t>If theaforementioned createOffer/createAnswer options orRtpTransceiverAPIs MUST be used.</t> <t>After calling setLocalDescription with an offerhas a sendrecv oranswer, the application MAY modifysendonly direction: </t> <ul spacing="normal"> <li>For each MediaStream that was associated with theSDP to reduce its capabilities before sendingtransceiver when itto the far side, as longwas created via addTrack or addTransceiver, an "a=msid" line, asit followsspecified in <xref target="RFC8830" sectionFormat="comma" section="2"/>, but omitting therules above"appdata" field.</li> </ul> </li> </ul> <t>Each "m=" section thatdefine a valid JSEP offeris not bundled into another "m=" section <bcp14>MUST</bcp14> contain the following attributes (which are of category IDENTICAL oranswer. Likewise, an application that has received an offerTRANSPORT):</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 oranswer from a peer MAY modifymore "a=fingerprint" lines for each of thereceived SDP, subject toendpoint'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 thesame constraints, before calling setRemoteDescription.</t> <t>As always,answer <bcp14>MUST</bcp14> be "active" or "passive". When theapplication is solely responsible for what it sends tooffer contains theother party, and all incoming SDP"actpass" value, as will always beprocessed bythe case with JSEPimplementation toendpoints, theextent 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 cominganswerer <bcp14>SHOULD</bcp14> use the "active" role. Offers fromanynon-JSEP endpoints <bcp14>MAY</bcp14> send otherimplementation 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 thefollowing steps MUST be performed: <list style="symbols"> <t>Ifanswer <bcp14>MUST</bcp14> use a value consistent with thedescription is of type "rollback", followvalue in theprocessing definedoffer.</li> <li>An "a=tls-id" line, as specified in <xreftarget="sec.processing-a-rollback" /> and skip the processing describedtarget="RFC8842" sectionFormat="comma" section="5.3"/>.</li> <li>If present in therest of this section.</t> <t>Otherwise, the type of the SessionDescription is checked againstoffer, 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 thecurrent state ofdummy value "9 IN IP4 0.0.0.0" (because no candidates have yet been gathered).</li> <li>If present in thePeerConnection: <list style="symbols">offer, an "a=rtcp-rsize" line, as specified in <xref target="RFC5506" sectionFormat="comma" section="5"/>.</li> </ul> <t>Ifthe 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 MUSTa data channel "m=" section has been offered, an "m=" section <bcp14>MUST</bcp14> also beeither "have-remote-offer" or "have-local-pranswer".</t> </list></t> <t>If the type is not correctgenerated for data. The <media> field <bcp14>MUST</bcp14> be set to "application", and thecurrent state, processing MUST stop<proto> andan error MUST<fmt> fields <bcp14>MUST</bcp14> bereturned.</t> <t>The SessionDescription is then checked to ensure that its contents are identicalset tothose generatedexactly match the fields in thelast call to createOffer/createAnswer,offer.</t> <t>Within the data "m=" section, an "a=mid" line <bcp14>MUST</bcp14> be generated andthus have not been altered,included asdiscussed in <xref target="sec.modifying-sdp" />; otherwise, processing MUST stop anddescribed above, along with anerror MUST be returned.</t> <t>Next,"a=sctp-port" line referencing theSessionDescription is parsed into a data structure,SCTP port number, asdescribeddefined in <xreftarget="sec.parsing-a-desc" /> below.</t> <t>Finally, the parsed SessionDescription is appliedtarget="RFC8841" sectionFormat="comma" section="5.1"/>; and, if appropriate, an "a=max-message-size" line, asdescribeddefined in <xreftarget="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 followingsteps MUST be performed: <list style="symbols"> <t>If the description isattributes oftype "rollback", followcategory IDENTICAL or TRANSPORT are included only if theprocessing 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 andskip the processing describedIDENTICAL attributes may also appear in therest of this section.</t> <t>Otherwise, the typedata "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 theSessionDescription is checked against the current stateoffered bundle groups that have not been rejected. Note that regardless of thePeerConnection: <list style="symbols"> <t>Ifpresence of "a=bundle-only" in thetype is "offer",offer, no "m=" sections in thePeerConnection state MUSTanswer should have an "a=bundle-only" line.</t> <t>Attributes that are common between all "m=" sections <bcp14>MAY</bcp14> beeither "stable" or "have-remote-offer".</t> <t>If the type is "pranswer" or "answer",moved to thePeerConnection state MUSTsession level if explicitly defined to beeither "have-local-offer" or "have-remote-pranswer".</t> </list></t> <t>If the type is not correct forvalid at thecurrent state, processing MUST stopsession level. <!-- [rfced] Section 5.3.1: Per "the session level" used elsewhere in this document andan 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 stopthe Change Log, we changed "to session-level" andan error MUST be returned.</t> <t>Finally,"at session-level" to "to theparsed 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 ifsession level" and "at thePeerConnection is insession level." Please let us know anystate except for "stable". This meansobjections. Original: Attributes thatboth offers and provisional answers can be rolled back. Rollback can onlyare common between all m= sections MAY beusedmoved tocancel proposed changes; there is no support for rolling back from a stable statesession-level, if explicitly defined toa previous stable state. If a rollback is attempted in the "stable" state, processing MUST stop and an error MUSTbereturned. Note that this impliesvalid at session-level. Currently: Attributes thatonceare common between all "m=" sections MAY be moved to theanswerer has performed setLocalDescription with his answer, this cannotsession level if explicitly defined to berolled back.</t>valid at the session level. --> </t> <t>Theeffectattributes prohibited in the creation ofrollback MUST beoffers are also prohibited in thesame regardlesscreation ofwhether setLocalDescriptionanswers.</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 orsetRemoteDescriptioniscalled.</t> <t>In order to process rollback,called after aJSEP implementation abandons the current offer/answer transaction, sets the signaling state to "stable", and sets the pendinglocaland/or remotedescription(see <xref target="sec.pendinglocaldescription" /> and <xref target="sec.pendingremotedescription" />) to null. Any resources or candidates that were allocated byhas already been installed, theabandoned local description are discarded; any media that is receivedprocessing isprocessed according tosomewhat different than for an initial answer.</t> <t>If the previouslocal and remote descriptions.</t> <t>A rollback disassociates any RtpTransceivers that were associated with m= sections byanswer was not applied using setLocalDescription, meaning theapplicationPeerConnection 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 therolled-back"o=" line <bcp14>MUST</bcp14> stay the same except for the <session-version> 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 werechanges in any way from the previouslyassociated will no longer be associated withgenerated answer.</li> </ul> <t>If anym= section;session description was previously supplied to setLocalDescription, an answer is generated by following the steps insuch cases,thevalue 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 themapping between the transceiversame.</li> <li>Each "m=" andits 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> bestopped and removed fromfilled in with thePeerConnection. However, a RtpTransceiver MUST NOT be removed if a track was attached toport and address of theRtpTransceiver viadefault candidate for theaddTrack method. This is so"m=" section, as described in <xref target="RFC8839" sectionFormat="comma" section="4.2.1.2"/>. Note thatan applicationin certain cases, the "m=" line protocol maycall addTrack, then call setRemoteDescription with an offer, then roll backnot match that of the default candidate, because the "m=" line protocol value <bcp14>MUST</bcp14> match what was supplied in the offer,then call createOfferas described above.</li> <li>Each "a=ice-ufrag" andhave a m= section for"a=ice-pwd" line <bcp14>MUST</bcp14> stay theadded track appear insame, unless thegenerated 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 thesession 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 akey-value expression,new "a=tls-id" value <bcp14>MUST</bcp14> be created, as described in <xreftarget="RFC4566" />, Section 5. The SDPtarget="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 isread, line-by-line, and converted to a data structure that containsbeing continued by thedeserialized 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 cannotofferer.</li> <li>RTCP multiplexing must beparsed as described, the parser MUST stop withused, and anerror"a=rtcp-mux" line inserted if andreject the session description, evenonly if thevalue"m=" section previously used RTCP multiplexing.</li> <li>If the "m=" section isto be discarded. This ensures that implementations donotaccidentally 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 andwith a specific syntax, as defined in <xref target="RFC4566" />, Section 5. Note that while the specificRTCP multiplexing is not active, an "a=rtcp" attribute linetypes (e.g. "v=", "c=") MUST occur<bcp14>MUST</bcp14> be filled in with thedefined order, linesport and address of thesame 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 theydefault RTCP candidate. If no RTCP candidates have yet beenchecked. <list> <t>The "c=" line MUSTgathered, dummy values <bcp14>MUST</bcp14> bechecked for syntax but its value is only used for ICE mismatch detection,used, asdefineddescribed in <xreftarget="RFC8445" />, Section 5.4. Note that JSEP implementations should never encounter this condition because ICEtarget="sec.initial-answers"/> above.</li> <li>If the "m=" section isrequirednot bundled into another "m=" section, forWebRTC.</t> <t>The "i=", "u=", "e=", "p=", "t=", "r=", "z=", and "k=" lines are not used by this specification; they MUSTeach 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> bechecked for syntax but their values are not used.</t> </list></t> <t>The remaining non-attribute lines are processedadded, asfollows: <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, asspecifieddescribed in <xreftarget="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] Section5.1.</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 beparsedadded, asspecifieddefined in<xref target="RFC4566" />,[RFC5245], Section5.2.</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 beparsedadded, asspecifieddescribed in<xref target="RFC4566" />,[I-D.ietf-ice-trickle], Section5.8, and9.3. If thebwtypem= section is bundled into another m= section, both "a=candidate" andbandwidth values stored.</t> </list></t> <t>Finally,"a=end-of-candidates" MUST be omitted. NEW: o If theattribute lines are processed. Specific processingm= 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 beappliedadded, as defined in [RFC5245], Section 4.3., paragraph 3. If candidate gathering for thefollowing session-levelsection has completed, an "a=end-of-candidates" attribute("a=") lines: <list style="symbols"> <t>Any "a=group" lines are parsedMUST be added, asspecifieddescribed in<xref target="RFC5888" />,[I-D.ietf-mmusic-trickle-ice-sip], Section5, and8.2. If thegroup's semantics and mids are stored.</t> <t>If present, a single "a=ice-lite" linem= section isparsed as specified in <xref target="I-D.ietf-mmusic-ice-sip-sdp" />, Section 4.3,bundled into another m= section, both "a=candidate" anda value indicating"a=end-of-candidates" MUST be omitted." Please note that thepresence 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], Section4.4, and4.3., paragraph 3" versus "[I-D.ietf-mmusic-ice-sip-sdp], Section 4.1"): o If theufrag valuem= section isstored.</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" lineis parsedMUST be added, asspecifieddefined in<xref target="I-D.ietf-mmusic-ice-sip-sdp" />,[I-D.ietf-mmusic-ice-sip-sdp], Section4.4, and4.1. If candidate gathering for thepassword value is stored.</t> <t>If present, a single "a=ice-options" line is parsedsection has completed, an "a=end-of- candidates" attribute MUST be added, asspecifieddescribed in<xref target="I-D.ietf-mmusic-ice-sip-sdp" />,[I-D.ietf-ice-trickle], Section4.6, and9.3. If theset of specified optionsm= section isstored.</t> <t>Any "a=fingerprint" linesbundled into another m= section, both "a=candidate" and "a=end-of- candidates" MUST be omitted. Please review, and let us know if further changes areparsed as specified in <xref target="RFC8122" />, Section 5,needed. (As noted previously, "[RFC8839]" and "[RFC8840]" are theset of fingerprintRFC numbers assigned for [I-D.ietf-mmusic-ice-sip-sdp] andalgorithm values[I-D.ietf-mmusic-trickle-ice-sip], respectively.) Currently: * If the "m=" section isstored.</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" lineis parsedMUST be added, asspecifieddefined in<xref target="RFC4145" />,[RFC8839], Section4, and5.1. If candidate gathering for thesetup value is stored.</t> <t>If present, a single "a=tls-id" line is parsedsection has completed, an "a=end-of-candidates" attribute MUST be added, asspecifieddescribed in<xref target="I-D.ietf-mmusic-dtls-sdp" />[RFC8840], Section5, and8.2. If thetls-id value"m=" section isstored.</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" andtheir values are stored.</t> </list></t> <t>Other attributes"a=end- of-candidates" MUST be omitted. --> </li> <li>For RtpTransceivers that are notrelevant 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 withstopped, thelines 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 thesession-level lines,same, regardless of changes to themedia section lines MUST occurtransceiver's direction or track. If no "a=msid" line is present in thespecific order and withcurrent description, "a=msid" line(s) <bcp14>MUST</bcp14> be generated according to thespecific 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 parsedcreateAnswer method takes asdescribeda 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>FollowingRTCOfferOptions; the"m=" line, specific processing MUSTIceRestart option is unnecessary, as ICE credentials will automatically beappliedchanged for all "m=" sections where the offerer chose to perform ICE restart.</t> <t>The followingnon-attribute lines: <list style="symbols"> <t>As withoptions are supported in RTCAnswerOptions.</t> <section anchor="sec.voiceactivitydetection2" numbered="true" toc="default"> <name>VoiceActivityDetection</name> <t>Silence suppression in the"c=" line atanswer is handled as described in <xref target="sec.voiceactivitydetection1" format="default"/>, with one exception: if support for silence suppression was not indicated in thesession level,offer, the"c=" line MUSTVoiceActivityDetection parameter has no effect, and the answer should beparsed accordinggenerated as if VoiceActivityDetection was set to<xref target="RFC4566" />, Section 5.7, but its value"false". This isnot used.</t> <t>The "b=" line,done on a per-codec basis (e.g., ifpresent, MUST be parsed as specified in <xref target="RFC4566" />, Section 5.8, andthebwtype and bandwidth values stored.</t> </list></t> <t>Specific processing MUST also be appliedofferer somehow offered support forthe following attribute lines: <list style="symbols"> <t>If present, a single "a=ice-ufrag" line is parsed as specifiedCN 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 andthe ufrag value is stored.</t> <t>If present, a single "a=ice-pwd" line"usedtx=0"). The impact of this rule isparsed as specified in <xref target="I-D.ietf-mmusic-ice-sip-sdp" />, Section 4.4, andthat 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 thepassword value is stored.</t> <t>If present, a single "a=ice-options" lineSDP isparsed as specified in <xref target="I-D.ietf-mmusic-ice-sip-sdp" />, Section 4.6, andneeded, theset of specifiedaforementioned createOffer/createAnswer optionsis stored.</t> <t>Any "a=candidate" attributes MUSTor RtpTransceiver APIs <bcp14>MUST</bcp14> beparsedused.</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, asspecified 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 parsedlong asspecified 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 asingle "a=end-of-candidates" attribute MUST be parsed as specified in <xref target="I-D.ietf-ice-trickle" />, Section 8.2, and its presencevalid JSEP offer orabsence flagged and stored.</t> <t>Any "a=fingerprint" lines are parsed as specified in <xref target="RFC8122" />, Section 5, andanswer. Likewise, an application that has received an offer or answer from a peer <bcp14>MAY</bcp14> modify theset of fingerprint and algorithm valuesreceived SDP, subject to the same constraints, before calling setRemoteDescription.</t> <t>As always, the application isstored.</t> </list></t> <t>Ifsolely 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 thefollowing attribute lines MUSTJSEP implementation to the extent of its capabilities. It is an error to assume that all SDP is well formed; however, one should beprocessed: <list style="symbols"> <t>The "m=" fmt value MUSTable to assume that any implementation of this specification will beparsedable to process, asspecified 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 asingle "a=ptime" line MUSTLocal Description</name> <t>When a SessionDescription is supplied to setLocalDescription, the following steps <bcp14>MUST</bcp14> beparsed as describedperformed: </t> <ul spacing="normal"> <li>If the description is of type "rollback", follow the processing defined in <xreftarget="RFC4566" />, Section 6,target="sec.processing-a-rollback" format="default"/> andits value stored.</t> <t>If present, a single "a=maxptime" line MUST be parsed asskip 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") MUSTthe 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> beparsed as described in <xref target="RFC4566" />, Section 6, and its value stored.</t> <t>Any "a=ssrc" attributes MUSTeither "stable" or "have-local-offer".</li> <li>If the type is "pranswer" or "answer", the PeerConnection state <bcp14>MUST</bcp14> beparsed 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 andtheir values stored.</t> <t>Any "a=extmap" attributes MUSTan error <bcp14>MUST</bcp14> beparsed as specifiedreturned.</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, andtheir values stored.</t> <t>Any "a=rtcp-fb" attributes MUST be parsedthus have not been altered, asspecifieddiscussed in <xreftarget="RFC4585" />, Section 4.2.,target="sec.modifying-sdp" format="default"/>; otherwise, processing <bcp14>MUST</bcp14> stop andtheir values stored.</t> <t>If present, a single "a=rtcp-mux" attribute MUSTan error <bcp14>MUST</bcp14> be returned.</li> <li>Next, the SessionDescription is parsed into a data structure, asspecifieddescribed in <xreftarget="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 betarget="sec.parsing-a-desc" format="default"/> below.</li> <li>Finally, the parsed SessionDescription is applied asspecifieddescribed in <xreftarget="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 asingle "a=rtcp-rsize" attribute MUSTRemote Description</name> <t>When a SessionDescription is supplied to setRemoteDescription, the following steps <bcp14>MUST</bcp14> beparsed as specifiedperformed: </t> <ul spacing="normal"> <li>If the description is of type "rollback", follow the processing defined in <xreftarget="RFC5506" />, Section 5, and its presence or absence flaggedtarget="sec.processing-a-rollback" format="default"/> andstored.</t> <t>If present, a single "a=rtcp" attribute MUST be parsed as specifiedskip the processing described in<xref target="RFC3605" />, Section 2.1, but its value is ignored, asthe rest of thisinformationsection.</li> <li> <t>Otherwise, the type of the SessionDescription issuperfluous when using ICE.</t> <t>If present, "a=msid" attributes MUSTchecked against the current state of the PeerConnection: </t> <ul spacing="normal"> <li>If the type is "offer", the PeerConnection state <bcp14>MUST</bcp14> beparsed 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 valueeither "stable" or "have-remote-offer".</li> <li>If the type isgenerated for a "default" MediaStream for"pranswer" or "answer", thesession, ifPeerConnection state <bcp14>MUST</bcp14> be either "have-local-offer" or "have-remote-pranswer".</li> </ul> </li> <li>If the type is notalready present,correct for the current state, processing <bcp14>MUST</bcp14> stop andthis value is stored.</t> <t>Any "a=imageattr" attributes MUSTan error <bcp14>MUST</bcp14> be returned.</li> <li>Next, the SessionDescription is parsed into a data structure, asspecifieddescribed in <xreftarget="RFC6236" />, Section 3,target="sec.parsing-a-desc" format="default"/> below. If parsing fails for any reason, processing <bcp14>MUST</bcp14> stop andtheir values stored.</t> <t>Any "a=rid" lines MUSTan error <bcp14>MUST</bcp14> be returned.</li> <li>Finally, the parsed SessionDescription is applied asspecifieddescribed in <xreftarget="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 asingle "a=simulcast" line MUSTRollback</name> <t>A rollback may beparsed 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 specifiedPeerConnection is in<xref target="I-D.ietf-mmusic-sctp-sdp" />, Section 4.3,any state except for "stable". This means that both offers andthe application protocol value stored.</t> <t>An "a=sctp-port" attribute MUSTprovisional answers can bepresent, and it MUSTrolled back. Rollback can only beparsed 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 relevantused toJSEP 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 descriptioncancel proposed changes; there isthen evaluated to ensure internal consistency as well as properno support formandatory features. Specifically,rolling back from a stable state to a previous stable state. If a rollback is attempted in thefollowing 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 themandatory-to-use features enumerated in <xref target="sec.usage-requirements" /> MUSTanswerer has performed setLocalDescription with its answer, this cannot bepresent. These values MAY eitherrolled back.</t> <t>The effect of rollback <bcp14>MUST</bcp14> bepresent atthemedia level,same regardless of whether setLocalDescription orinherited 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 thissetRemoteDescription isa re-offer or a responsecalled.</t> <t>In order to process rollback, are-offer andJSEP implementation abandons thetls-id value is different from that presently in use,current offer/answer transaction, sets theDTLS connection is not being continuedsignaling state to "stable", and sets the pending local and/or remote descriptionMUST be part of an ICE restart, together with new ufrag(see Sections <xref target="sec.pendinglocaldescription" format="counter"/> andpassword 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 therules specified in <xref target="RFC5763" />, Section 5previous local andMUST be consistentremote descriptions.</t> <t>A rollback disassociates any RtpTransceivers that were associated with "m=" sections by theselected roleapplication of thecurrent DTLS connection, if one existsrolled-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 isbeing 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 sectionis also checkedtoensure prohibited features arecite here. We easily found relevant text in Section 5.10 but notused.</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 thatsection MUST also contain an "a=rtcp-mux" attribute.</t> <t>If anwere associated with m=section was present in the previous answer,sections by thestate of RTP/RTCP multiplexing MUST match what was previously negotiated.</t> </list></t> <t>If this session description isapplication oftype "pranswer" or "answer",thefollowing additional checks are applied: <list style="symbols"> <t>Therolled-back session descriptionmust follow the rules defined in <xref target="RFC3264" />,(see Section6, including the requirement5.10 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, thenumbervalue ofm= sections MUST exactly matchthenumber of m= sections inRtpTransceiver's mid property <bcp14>MUST</bcp14> be set to "null", and theassociated offer.</t> <t>For each m= section,mapping between themedia typetransceiver andprotocol values MUST exactly match the media typeits "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 andprotocol values inremoved from thecorresponding m= section inPeerConnection. However, an RtpTransceiver <bcp14>MUST NOT</bcp14> be removed if a track was attached to theassociated offer.</t> </list></t> <t>If any ofRtpTransceiver via thepreceding checks failed, processing MUST stopaddTrack 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 anerror MUST be returned.</t> </section>"m=" section for the added track appear in the generated offer.</t> </section> <sectiontitle="Applyinganchor="sec.parsing-a-desc" numbered="true" toc="default"> <name>Parsing aLocal Description" anchor="sec.applying-a-local-desc">Session Description</name> <t>Thefollowing steps are performed atSDP contained in themedia engine level to applysession description object consists of alocal description. If an errorsequence of text lines, each containing a key-value expression, as described in <xref target="RFC4566" sectionFormat="comma" section="5"/>. The SDP isreturned, the session MUST be restoredread, line by line, and converted to a data structure that contains thestate it was in before performing these steps.</t> <t>First, m= sectionsdeserialized information. However, SDP allows many types of lines, not all of which areprocessed.relevant to JSEP applications. For eachm= section,line, thefollowing 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= sectionimplementation will first ensure that it isnew, begin gathering candidates for it, as definedsyntactically correct according to its defining ABNF, check that it conforms to the semantics used in <xreftarget="RFC8445" />, Section 5.1.1, unless it is definitively being bundled (either this is an offertarget="RFC4566" format="default"/> andthe m= section is marked bundle-only, or it is an answer<xref target="RFC3264" format="default"/>, andthe m= section is bundled into into another m= section.)</t> <t>Or, if the ICE ufragthen either parse andpassword values have changed, triggerstore or discard theICE agent to start an ICE restartprovided value, as describedin <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>Ifthereany line isno RtpTransceiver associated with this m= section, find one and associate it with this m= section according tonot well formed or cannot be parsed as described, thefollowing steps. Note that this situation will only occur when applyingparser <bcp14>MUST</bcp14> stop with anoffer. <list style="symbols"> <t>Find the RtpTransceiver that corresponds to this m= section, using the mapping between transceiverserror andm= section indices established when creatingreject theoffer.</t> <t>Setsession description, even if the valueof this RtpTransceiver's mid property to the MID of the m= section.</t> </list></t> <t>If RTCP muxisindicated, preparetodemux RTP and RTCP frombe 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, theRTP ICE component, as specifiedsession-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, establishamapping between the extension IDspecific order, andURI,with a specific syntax, asdescribeddefined in <xreftarget="RFC5285" />, Section 6.</t> <t>Iftarget="RFC4566" sectionFormat="comma" section="5"/>. Note that while theMID header extension is supported, prepare to demux RTP streams intended for this m= section based on the MID header extension, as describedspecific 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 betweenthepayloaddefined order, lines of the same typeand(typically "a=") can occur in any order.</t> <t>The following non-attribute lines are not meaningful in theactual 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, asdescribeddefined in <xreftarget="RFC3264" />, Section 6.1. In addition, prepare to demux RTP streams intendedtarget="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 thism= section based on the media formats supportedsentence confusing; should "t=" remain in this list of lines not used by thism= 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, asdescribedspecified in<xref target="I-D.ietf-mmusic-sdp-bundle-negotiation" />,[RFC4566], Section10.2.</t> <t>For each specified "rtx" media format, establish a mapping between the RTX payload type5.9; both <start-time> andits 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=" and8.7.</t> <t>If"t=" lines MUST stay thedirectional 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>Thefollowing stepsremaining non-attribute lines areperformed to applyprocessed as follows: </t> <ul spacing="normal"> <li>The "v=" line <bcp14>MUST</bcp14> have aremote description. If an error is returned, the session MUSTversion of 0, as specified in <xref target="RFC4566" sectionFormat="comma" section="5.1"/>.</li> <li>The "o=" line <bcp14>MUST</bcp14> berestored to the state it wasparsed as specified inbefore 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 asan attribute, updatespecified in <xref target="RFC4566" sectionFormat="comma" section="5.8"/>, and thePeerConnection canTrickle property to be true. Otherwise, set this property to false.</t> <t>The following steps MUST be performed for attributes atbwtype and bandwidth values stored.</li> </ul> <t>Finally, thesession level; if any parametersattribute lines areout of bounds, or cannot be applied,processed. Specific processingMUST stop and an error MUST<bcp14>MUST</bcp14> bereturned. <list style="symbols"> <t>For any specified "CT" bandwidth value, set this as the limitapplied for themaximum 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 <xreftarget="RFC4566"></xref>, Section 5.8. Within this overall limit, the implementation can dynamically decide how to best allocatetarget="RFC5888" sectionFormat="comma" section="5"/>, and theavailable 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, handlegroup's semantics and mids are stored.</li> <li>If present, a single "a=ice-lite" line is parsed as specified in <xreftarget="RFC3556"></xref>, Section 2.</t> <t>Any "AS" bandwidthtarget="RFC8839" sectionFormat="comma" section="5.3"/>, and a valueMUST be ignored, asindicating themeaningpresence ofthis construct at the session levelice-lite isnot 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 stopstored.</li> <li>If present, a single "a=ice-ufrag" line is parsed as specified in <xref target="RFC8839" sectionFormat="comma" section="5.4"/>, andan error MUST be returned. <list style="symbols"> <t>IftheICEufragor password changed from the previous remote description: <list style="symbols"> <t>If the descriptionvalue isof type "offer", the implementation MUST note that an ICE restartstored.</li> <li>If present, a single "a=ice-pwd" line isneeded,parsed asdescribedspecified in <xreftarget="I-D.ietf-mmusic-ice-sip-sdp" />, Section 3.4.1.1.1</t> <t>Iftarget="RFC8839" sectionFormat="comma" section="5.4"/>, and thedescriptionpassword value isof type "answer" or "pranswer", then check to see if the current local descriptionstored.</li> <li>If present, a single "a=ice-options" line isan ICE restart,parsed as specified in <xref target="RFC8839" sectionFormat="comma" section="5.6"/>, andif not, generate an error. IfthePeerConnection stateset 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 asprescribed byspecified in <xreftarget="RFC8445" />, Section 9, generate an error.</t> <t>Configure the ICE components associated with this media section to usetarget="RFC8122" sectionFormat="comma" section="5"/>, and thesupplied ICE remote ufragset of fingerprint andpassword 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 asdescribedspecified in <xreftarget="RFC8445" />, Section 6.1.2,target="RFC4145" sectionFormat="comma" section="4"/>, andstart connectivity checks withtheappropriate credentials.</t> <t>If an "a=end-of-candidates" attributesetup value is stored.</li> <li>If present,process the end-of-candidates indicationa single "a=tls-id" line is parsed asdescribedspecified in <xreftarget="I-D.ietf-ice-trickle" />, Section 11.</t> <t>Iftarget="RFC8842" sectionFormat="comma" section="5"/>, and them= section prototls-id valueindicates use of RTP: <list style="symbols"> <t>If the m= sectionisbeing 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 anddiscard the mapping betweenthetransceiveridentity 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"/>, andits m= section index.</t> <t>If the m= section istheir values are stored.</li> </ul> <t>Other attributes that are notassociated withrelevant to JSEP may also be present, and implementations <bcp14>SHOULD</bcp14> process anyRtpTransceiver (possibly because it was dissociated inthat 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 theprevious step), either find an RtpTransceiver or create one according tosession-level lines have been parsed, processing continues with thefollowing steps: <list style="symbols"> <t>Iflines in "m=" sections.</t> </section> <section anchor="sec.media-level-parse" numbered="true" toc="default"> <name>Media Section Parsing</name> <t>Like them= section is sendrecv or recvonly, and there are RtpTransceivers ofsession-level lines, thesame type that were added tomedia section lines <bcp14>MUST</bcp14> occur in thePeerConnection by addTrackspecific order andare not associatedwithany m= section and are not stopped, find the first (according tothecanonical order describedspecific syntax defined in <xreftarget="sec.initial-offers" />) such RtpTransceiver.</t> <t>If no RtpTransceiver was foundtarget="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 theprevious step, create one with a recvonly direction.</t> <t>Associatemedia, port, proto, and fmt values stored.</t> <t>Following thefound 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 them= section by setting"c=" line at thevalue ofsession level, theRtpTransceiver's mid property"c=" line <bcp14>MUST</bcp14> be parsed according tothe 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"/>, andestablish a mapping betweenthetransceiverbwtype andthe 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 valuebandwidth values stored.</li> </ul> <t>Specific processing <bcp14>MUST</bcp14> also be applied for theRtpTransceiver mid property,followingthe guidance for "a=mid" mentionedattribute lines: </t> <ul spacing="normal"> <li>If present, a single "a=ice-ufrag" line is parsed as specified in <xreftarget="sec.initial-offers" />.</t> </list></t> <t>For each specified media format that is also supported bytarget="RFC8839" sectionFormat="comma" section="5.4"/>, and thelocal implementation, establishufrag value is stored.</li> <li>If present, amapping between thesingle "a=ice-pwd" line is parsed as specifiedpayload typein <xref target="RFC8839" sectionFormat="comma" section="5.4"/>, and themedia format,password value is stored.</li> <li>If present, a single "a=ice-options" line is parsed asdescribedspecified in <xreftarget="RFC3264" />, Section 6.1. Specifically, this means that the implementation recordstarget="RFC8839" sectionFormat="comma" section="5.6"/>, and thepayload type to be used in outgoing RTP packets when sending eachset of specifiedmedia format, as well as the relative preference for each format thatoptions isindicatedstored.</li> <li>Any "a=candidate" attributes <bcp14>MUST</bcp14> be parsed as specified in <xref target="RFC8839" sectionFormat="comma" section="5.1"/>, and theirordering. If any indicated media format is not supported by the local implementation, it MUSTvalues stored.</li> <li>Any "a=remote-candidates" attributes <bcp14>MUST</bcp14> beignored.</t> <t>For eachparsed as specified"rtx" media format, establishin <xref target="RFC8839" sectionFormat="comma" section="5.2"/>, but their values are ignored.</li> <li>If present, amapping between the RTX payload typesingle "a=end-of-candidates" attribute <bcp14>MUST</bcp14> be parsed as specified in <xref target="RFC8840" sectionFormat="comma" section="8.1"/>, and itsassociated primary payload type,presence or absence flagged and stored.</li> <li>Any "a=fingerprint" lines are parsed asdescribedspecified in <xreftarget="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 bytarget="RFC8122" sectionFormat="comma" section="5"/>, and thelocal media implementation,set of fingerprint and algorithm values is stored.</li> </ul> <t>If the "m=" proto value indicates use of RTP, as described inwhich case,<xref target="sec.profile-names" format="default"/> above, theRTX payload type MUST alsofollowing attribute lines <bcp14>MUST</bcp14> beignored.</t> <t>For eachprocessed: </t> <ul spacing="normal"> <li>The "m=" fmt value <bcp14>MUST</bcp14> be parsed as specifiedfmtp parameter that is supported by the local implementation, enable them onin <xref target="RFC4566" sectionFormat="comma" section="5.14"/>, and theassociated media formats.</t> <t>For eachindividual values stored.</li> <li>Any "a=rtpmap" or "a=fmtp" lines <bcp14>MUST</bcp14> be parsed as specifiedSSRC that is signaledinthe 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 <xreftarget="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 IDtarget="RFC4566" sectionFormat="comma" section="6"/>, andURI,its value stored.</li> <li>If present, a single "a=maxptime" line <bcp14>MUST</bcp14> be parsed as described in <xreftarget="RFC5285" />, Section 5. Specifically, this means that the implementation records the extension ID totarget="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> beusedparsed as described inoutgoing 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 specifiedheader extension. If any indicated RTP header extension is not supported by the local implementation, it MUSTin <xref target="RFC5576" sectionFormat="comma" section="4.1"/>, and their values stored.</li> <li>Any "a=extmap" attributes <bcp14>MUST</bcp14> beignored.</t> <t>For eachparsed as specifiedRTCP 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 valuein <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, aconstraint on the maximum RTP bitrate tosingle "a=rtcp-mux" attribute <bcp14>MUST</bcp14> beused when sending media,parsed as specified in <xreftarget="RFC3890"></xref>. If a "TIAS" value is nottarget="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, generatea"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 usedsingle "a=rtcp-mux-only" attribute <bcp14>MUST</bcp14> be parsed as specified inpreference 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, handleabsence flagged and stored.</li> <li>If present, a single "a=rtcp-rsize" attribute <bcp14>MUST</bcp14> be parsed as specified in <xreftarget="RFC3556"></xref>, Section 2.</t> <t>Anytarget="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" bandwidthin <xref target="RFC3605" sectionFormat="comma" section="2.1"/>, but its valueMUST beis ignored, asthe meaning ofthisconstruct at the media level is not well defined.</t> <t>If the m= sectioninformation isof 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 asdescribedspecified in <xreftarget="RFC3389" />, Section 5, except for formats that havetarget="RFC8830" sectionFormat="comma" section="3.2"/>, and theirown internal silence suppression mechanisms. Silence suppressionvalues stored, ignoring any "appdata" field. If no "a=msid" attributes are present, a random msid-id value is generated forsuch formats (e.g., Opus)a "default" MediaStream for the session, if not already present, and this value iscontrolled via fmtp parameters,stored.</li> <li>Any "a=imageattr" attributes <bcp14>MUST</bcp14> be parsed asdiscussedspecified in <xreftarget="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 asdescribedspecified in <xreftarget="RFC4733" />, Section 2.5.1.2. If there are any supported media formats that do not havetarget="RFC8851" sectionFormat="comma" section="10"/>, and their values stored.</li> <li>If present, acorresponding telephone-event format, disable DTMF transmission for those formats.</t> <t>For anysingle "a=simulcast" line <bcp14>MUST</bcp14> be parsed as specified"ptime" value, configurein <xref target="RFC8853" format="default"/>, and its values stored.</li> </ul> <t>Otherwise, if theavailable 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 specifiedpacket size when sending. Ifin <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 specifiedsize is not supported for a media format, usein <xref target="RFC8841" sectionFormat="comma" section="5.2"/>, and thenext closestvalueinstead.</t> </list></t> </list></t> </list></t> <t>Finally, if this description is of type "pranswer" or "answer", follow the processing definedstored.</li> <li>If present, a single "a=max-message-size" attribute <bcp14>MUST</bcp14> be parsed as specified in <xreftarget="sec.applying-an-answer" /> below.</t> </section> <section title="Applying an Answer" anchor="sec.applying-an-answer"> <t>In addition totarget="RFC8841" sectionFormat="comma" section="6"/>, and thesteps mentioned above for processing a local or remote description,value stored. Otherwise, use thefollowing stepsspecified default.</li> </ul> <t>Other attributes that areperformed when processing a description of type "pranswer" or "answer".</t> <t>For each m= section, the following steps MUSTnot relevant to JSEP may also beperformed: <list style="symbols"> <t>Ifpresent, 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, them= section has been rejected (i.e. portparsed description issetthen evaluated tozero in the answer), stop any reception or transmission of mediaensure internal consistency as well as proper support forthis 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 describedvalid values for each of the mandatory-to-use features enumerated in <xreftarget="I-D.ietf-mmusic-ice-sip-sdp" />, Section 3.4.3.1.</t> <t>Iftarget="sec.usage-requirements" format="default"/> <bcp14>MUST</bcp14> be present. These values <bcp14>MAY</bcp14> be either present at theremote DTLS fingerprint has been changedmedia level or inherited from thetls-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 butsession level. </t> <ul spacing="normal"> <li>ICE ufrag and password values, which <bcp14>MUST</bcp14> comply with theanswer does not indicate an ICE restart or,size limits specified inthe 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> begenerated.set according to <xref target="RFC8842" sectionFormat="comma" section="5"/>. Ifan ICE restartthis isperformed withoutachange in tls-idre-offer orfingerprint, thena response to a re-offer and the tls-id value is different from that presently in use, thesameDTLS connection is not being continuedover the new ICE channel. Note that although JSEP requires that answerers change the tls-id value ifandonly if the offerer does, non-JSEP answerers are permitted to change the tls-id as long astheoffer containedremote description <bcp14>MUST</bcp14> be part of an ICErestart. Thus, JSEP implementations which processrestart, together with new ufrag and password values.</li> <li>A DTLSdata prior to receiving an answer MUSTsetup value, which <bcp14>MUST</bcp14> beprepared to receive either a ClientHello or data from the previous DTLS connection.</t> <t>If no valid DTLS connection exists, prepareset according tostart a DTLS connection, usingthe rules specifiedrolesin <xref target="RFC5763" sectionFormat="comma" section="5"/> andfingerprints, on any underlying ICE components, once they are active.</t> <t>If<bcp14>MUST</bcp14> be consistent with them= section proto value indicates useselected role ofRTP: <list style="symbols"> <t>Ifthem=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=" sectionreferences RTCP feedback mechanismsis also checked to ensure thatwereprohibited features are notpresent inused.</li> <li>If thecorresponding 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 theoffer,previous answer, the state of RTP/RTCP multiplexing <bcp14>MUST</bcp14> match what was previously negotiated.</li> </ul> <t>If thisindicates a negotiation problem and MUST result in an error. However, new media formats and new RTP header extension valuessession description is of type "pranswer" or "answer", the following additional checks arepermitted inapplied: </t> <ul spacing="normal"> <li>The session description must follow theanswer, as describedrules defined in <xref target="RFC3264"/>, Section 7, and <xref target="RFC5285" />, Section 6.</t> <t>IfsectionFormat="comma" section="6"/>, including them= section has RTCP mux enabled, discardrequirement that theRTCP ICE component, if one exists, and begin or continue muxing RTCP overnumber of "m=" sections <bcp14>MUST</bcp14> exactly match theRTP ICE component, as specifiednumber of "m=" sections in<xref target="RFC5761" />, Section 5.1.3. Otherwise, prepare to transmit RTCP overtheRTCP ICE component; if no RTCP ICE component exists, because RTCP mux was previously enabled, this MUST result in an error.</t> <t>Ifassociated offer.</li> <li>For each "m=" section, them= section has reduced-size RTCP enabled, configuremedia type and protocol values <bcp14>MUST</bcp14> exactly match theRTCP transmission for this m= section to use reduced-size RTCP, as specifiedmedia type and protocol values in<xref target="RFC5506" />.</t> <t>Ifthedirectional attributecorresponding "m=" section in theanswer indicates thatassociated offer.</li> </ul> <t>If any of theJSEP implementation shouldpreceding checks failed, processing <bcp14>MUST</bcp14> stop and an error <bcp14>MUST</bcp14> besendingreturned.</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" forengine level to apply a localanswers, "recvonly" for remote answers, or "sendrecv" for either type of answer), choosedescription. If an error is returned, themedia formatsession <bcp14>MUST</bcp14> be restored tosend asthemost preferred media format fromstate it was in before performing these steps.</t> <t>First, "m=" sections are processed. For each "m=" section, theremote description thatfollowing 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 isalso locally supported,new, begin gathering candidates for it, asdiscusseddefined in <xreftarget="RFC3264" />, Sections 6.1target="RFC8445" sectionFormat="comma" section="5.1.1"/>, unless it is definitively being bundled (either (1) this is an offer and7,the "m=" section is marked bundle-only or (2) it is an answer andstart transmitting RTP media using that format oncetheunderlying transport layers"m=" section is bundled into another "m=" section).</li> <li>Or, if the ICE ufrag and password values havebeen established. If an SSRC has not already been chosenchanged, trigger the ICE agent to start an ICE restart as described in <xref target="RFC8445" sectionFormat="comma" section="9"/>, and begin gathering new candidates forthis 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 isalready being transmitted,no RtpTransceiver associated with this "m=" section, find one and associate it with this "m=" section according to thesame SSRC SHOULD be used unlessfollowing steps. Note that this situation will only occur when applying an offer. </t> <ul spacing="normal"> <li>Find theclockrateRtpTransceiver 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 thenew codecMID of the "m=" section.</li> </ul> </li> <li>If RTCP mux isdifferent, in which case a new SSRC MUST be chosen,indicated, prepare to demux RTP and RTCP from the RTP ICE component, as specified in <xreftarget="RFC7160" />, Section 3.1.</t> <t>The payload typetarget="RFC5761" sectionFormat="comma" section="5.1.3"/>.</li> <li>For each specified RTP header extension, establish a mappingfrombetween theremote descriptionextension ID and URI, as described in <xref target="RFC5285" sectionFormat="comma" section="6"/>.</li> <li>If the MID header extension isusedsupported, prepare todetermine payload typesdemux RTP streams intended for this "m=" section based on theoutgoing RTP streams, includingMID 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 typeforand thesendactual mediaformat chosen above. Any RTP header extensions that were negotiated should be includedformat, as described inthe outgoing<xref target="RFC3264" sectionFormat="comma" section="6.1"/>. In addition, prepare to demux RTPstreams, 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 instreams intended for this "m=" section based on theoutgoing RTP streams,media formats supported by this "m=" section, asindicateddescribed in <xreftarget="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 Section4.</t> <t>If the m= section is of type audio,10.2. We have updated these to refer to 7.2 andsilence suppression was configured9.2, respectively. Please review. Original: In addition, prepare to demux RTP streams intended for this m= section based on thesendmediaformatformats supported by this m= section, asa result of processing the remote description, anddescribed in [I-D.ietf-mmusic-sdp-bundle-negotiation], Section 10.2. ... * For each specified SSRC that isalso enabledsignaled in the m= section, prepare to demux RTP streams intended for this m= section using thatformatSSRC, as described in [I-D.ietf-mmusic-sdp-bundle-negotiation], Section 10.2. ... If thelocal description, use silence suppressionanswer contains valid bundle groups, discard any ICE components foroutgoing media, in accordance withtheguidance in <xref target="sec.voiceactivitydetection1" />. If these conditions are not met, silence suppression MUST NOTm= sections that will beused for outgoing media.</t> <t>If simulcast has been negotiated, sendbundled onto thenumber of Source RTP Streamsprimary ICE components in each bundle, and begin muxing these m= sections accordingly, asspecifieddescribed in<xref target="I-D.ietf-mmusic-sdp-simulcast"></xref>,[I-D.ietf-mmusic-sdp-bundle-negotiation], Section6.2.2.</t> <t>If8.2. ... When bundling, associating incoming RTP/RTCP with thesend media format chosen above has a correspondingproper 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 aRedundancy RTP Stream with a random SSRC for each Source RTP Stream, and start or continue transmitting RTX/FEC packets as needed.</t> <t>Ifmapping between thesend 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, asdiscusseddescribed in Sections <xref target="RFC4588" section="8.6" sectionFormat="bare"/> and <xref target="RFC4588" section="8.7" sectionFormat="bare"/> of <xreftarget="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 Section3.2. 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 referenced5.9) and "[RFC3264], Sections 6.1 and 7" to "Sections 6.1 and 7 of [RFC3264]" (in Section 5.11) in order to get themedia section for all Source RTP Streams usinghyperlinks to work properly in the .html/.pdf files. Please let us know any concerns. Original: * For each specified "rtx" mediaformats. Specifically, begin or continue sendingformat, establish a mapping between therequested feedback typesRTX payload type andreacting to received feedback,its associated primary payload type, asspecifieddescribed in<xref target="RFC4585" />, Section 4.2. When sending RTCP feedback, follow the rules[RFC4588], Sections 8.6 andrecommendations from <xref target="RFC8108"></xref> Section 5.4.1, to select which SSRC to use.</t> <t>If8.7. ... * If the directional attribute in the answer indicates that the JSEP implementation shouldnotbe sending media("recvonly"("sendonly" for local answers,"sendonly""recvonly" for remote answers, or"inactive""sendrecv" for either type ofanswer) stopanswer), 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 transmittingallRTPmedia, 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>IfSections 8.6 and 8.7 of [RFC4588]. ... - If them= section proto valuedirectional attribute in the answer indicatesuse of SCTP: <list style="symbols"> <t>If an SCTP association exists, andthat the JSEP implementation should be sending media ("sendonly" for local answers, "recvonly" for remoteSCTP port has changed, discard the existing SCTP association. This includes the case whenanswers, or "sendrecv" for either type of answer), choose thePeerConnection state is "have-remote-pranswer".</t> <t>If no valid SCTP association exists, preparemedia format toinitiate a SCTP association over the associated ICE component and DTLS connection, using the local SCTP port value from the local description, andsend as theremote SCTP port valuemost preferred media format from the remotedescription,description that is also locally supported, asdescribeddiscussed 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= sectionsSections 6.1 and 7 of [RFC3264], and start transmitting RTP media using thatwill be bundled ontoformat once theprimary 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>Ifunderlying 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 withfollow theproper m= section isprocessing defined in <xreftarget="I-D.ietf-mmusic-sdp-bundle-negotiation" />, Section 10.2. When not bundling, the proper m= sectiontarget="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 isclear from the ICE component over whichreturned, theRTP/RTCP is received.</t> <t>Oncesession <bcp14>MUST</bcp14> be restored to theproper m= section(s) are known, RTP/RTCPstate it was in before performing these steps.</t> <t>If the answer contains any "a=ice-options" attributes where "trickle" isdelivered tolisted as an attribute, update theRtpTransceiver(s) associated withPeerConnection canTrickle property to be "true". Otherwise, set this property to "false".</t> <t>The following steps <bcp14>MUST</bcp14> be performed for attributes at them= section(s) and further processingsession 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 theRTP/RTCP is done atlimit for theRtpTransceiver level. This includes using RIDmaximum total bitrate for all "m=" sections, as specified in <xreftarget="I-D.ietf-mmusic-rid" />target="RFC4566" sectionFormat="comma" section="5.8"/>. Within this overall limit, the implementation can dynamically decide how todistinguishbest allocate the available bandwidth betweenmultiple 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 thatthis example section shows several SDP fragments. To formathave been specified for individual "m=" sections.</li> <li>For any specified "RR" or "RS" bandwidth values, handle as specified in72 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 thelines in SDP have been split into multiple lines, where leading whitespace indicates that a linesession level is not well defined. <!-- [rfced] Section 5.10: For ease of the reader, we suggest adding acontinuationcitation 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 theprevious line. In addition, some blank lines have been added to improve readability but are not valid in SDP.</t> <t>More examplesmeaning ofSDP for WebRTC call flows, including examples with IPv6 addresses, canthis construct at the session level is not well defined. Suggested: * Any "AS" bandwidth value ([RFC4566], Section 5.8) MUST befound 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 inignored, as thefollowing section provides a more detailed examplemeaning ofwhat could happen in a JSEP session.</t> <t>The code flow below shows Alice's endpoint initiatingthis construct at the sessionto Bob's endpoint. The messages from the JavaScript application in Alice's browser tolevel is not well defined. --> </li> </ul> <t>For each "m=" section, theJavaScript in Bob's browser, abbreviated as AliceJS and BobJS respectively,following steps <bcp14>MUST</bcp14> be performed; if any parameters areassumed to flow over some signaling protocol via a web server. The JavaScript on both Alice's sideout of bounds or cannot be applied, processing <bcp14>MUST</bcp14> stop andBob's side waits for all candidates before sendingan error <bcp14>MUST</bcp14> be returned. </t> <ul spacing="normal"> <li> <t>If theofferICE ufrag oranswer, sopassword changed from theoffers and answers are complete; trickleprevious remote description: </t> <ul spacing="normal"> <li>If the description is of type "offer", the implementation <bcp14>MUST</bcp14> note that an ICE restart isnot used. The user agents (JSEP implementations) in Alice and Bob's browsers, abbreviatedneeded, asAliceUA and BobUA respectively, are usingdescribed in <xref target="RFC8839" sectionFormat="comma" section="4.4.1.1.1"/>.</li> <li>If thedefault bundle policydescription is of"balanced", andtype "answer" or "pranswer", then check to see if thedefault RTCP mux policy of "require".</t> <t> <figure> <artwork> <![CDATA[ // set upcurrent localmediadescription is an ICE restart, and if not, generate an error. If the PeerConnection stateAliceJS->AliceUA: create new PeerConnection AliceJS->AliceUA: addTrack with two tracks: audiois "have-remote-pranswer" andvideo AliceJS->AliceUA: createOfferthe ICE ufrag or password changed from the previous provisional answer, then signal the ICE agent toget offer AliceJS->AliceUA: setLocalDescription with offer AliceUA->AliceJS: multiple onicecandidate events with candidates // waitdiscard any previous ICE check list state for the "m=" section. Finally, signal the ICEgatheringagent tocomplete AliceUA->AliceJS: onicecandidate event with null candidate AliceJS->AliceUA: get |offer-A1|begin checks. <!-- [rfced] Other documents in this cluster spell "checklist" as one word. May we change "check list" in this document to "checklist"? --> </li> </ul> </li> <li>If the current local description indicates an ICE restart and either the ICE ufrag or password has not changed frompendingLocalDescription // |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 issent 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 audiocorrect andvideo tracks // Bob accepts call BobJS->BobUA: addTrack withwill be clear to readers. Original: o If the current localtracks BobJS->BobUA: createAnswer BobJS->BobUA: setLocalDescription with answer BobUA->BobJS: multiple onicecandidate eventsdescription 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 withcandidates // waitthis media section to use the supplied ICE remote ufrag and password for their connectivity checks.</li> <li>Pair any supplied ICEgathering to complete BobUA->BobJS: onicecandidate eventcandidates withnull candidate BobJS->BobUA: get |answer-A1| from currentLocalDescription // |answer-A1|any gathered local candidates, as described in <xref target="RFC8445" sectionFormat="comma" section="6.1.2"/>, and start connectivity checks with the appropriate credentials.</li> <li>If an "a=end-of-candidates" attribute issent over signaling protocolpresent, 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 toAlice BobJS->WebServer: signaling with |answer-A1| WebServer->AliceJS: signaling"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: setRemoteDescriptionany 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 audioany "m=" section andvideo tracks // media flows BobUA->AliceUA: media sent from Bobare not stopped, find the first (according toAlice AliceUA->BobUA: media sent from Alicethe 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 toBob ]]> </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=audiothe MID of the "m=" section, and establish a mapping between the transceiver and the index of the "m=" section. If the "m=" section does not include a MID (i.e., the remote endpoint does not support the MID extension), generate a value for the RtpTransceiver mid property, following the guidance for "a=mid" mentioned in <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 the 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 each format that is indicated in their ordering. If any indicated media format is not supported by the local implementation, it <bcp14>MUST</bcp14> be ignored.</li> <li>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="RFC4588" sectionFormat="comma" section="4"/>. If any referenced primary payload types are not present, this <bcp14>MUST</bcp14> result in an error. Note that RTX payload types may refer to primary payload types that are not supported by the local media implementation, in which case the RTX payload type <bcp14>MUST</bcp14> also be ignored.</li> <li>For each specified fmtp parameter that is 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 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 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 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 kbps to bps (as required by TIAS), and the "0.95" is to allocate 5% to RTCP. <!-- [rfced] Section 5.10: For ease of the reader, should the "8" also be explained (e.g., possibly "bytes to bits")? We ask because explanations are provided for the other four numbers. Original: TIAS = AS * 1000 * 0.95 - (50 * 40 * 8) The 50 is based on 50 packets per second, the 40 is based on an estimate of total header size, the 1000 changes the unit from kbps to bps (as required by TIAS), and the 0.95 is to allocate 5% to RTCP.--> "TIAS" is used in preference to "AS" because it provides more accurate control of bandwidth.</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 not well defined.</li> <li> <t>If the "m=" section is of type "audio": </t> <ul spacing="normal"> <li>For each specified "CN" media format, configure silence suppression for 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> <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> <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) 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 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> <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 fragments. To accommodate RFC line-length restrictions, some of the SDP lines have been split into multiple lines, where leading whitespace indicates that a line is a continuation of the previous line. In addition, some blank lines have been added to improve readability but are not valid in SDP.</t> <t>More examples of SDP for 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 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 assumed to flow over some signaling protocol via a web server. The JavaScript on both Alice's side and Bob's side waits for all candidates before sending the offer or answer, so the offers and answers are complete; Trickle ICE is not used. The user agents (JSEP implementations) in Alice's and Bob's browsers, abbreviated as "AliceUA" and "BobUA", respectively, are using the default bundle policy of "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 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 default bundle policy of "balanced", and the default RTCP mux policy of "require". Suggested: 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 "balanced" (AliceUA) and the default RTCP mux policy of "require" (BobUA). --> </t> <!-- [rfced] Throughout: <artwork> has been converted to <sourcecode> as applicable and we set the "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 get offer AliceJS->AliceUA: setLocalDescription with offer AliceUA->AliceJS: multiple onicecandidate events with candidates // wait for ICE gathering to complete AliceUA->AliceJS: onicecandidate event with null candidate AliceJS->AliceUA: get |offer-A1| from pendingLocalDescription // |offer-A1| is sent over signaling protocol to Bob AliceJS->WebServer: signaling with |offer-A1| WebServer->BobJS: signaling with |offer-A1| // |offer-A1| arrives at Bob BobJS->BobUA: create a PeerConnection BobJS->BobUA: setRemoteDescription with |offer-A1| BobUA->BobJS: ontrack events for audio and video tracks // Bob accepts call BobJS->BobUA: addTrack with local tracks BobJS->BobUA: createAnswer BobJS->BobUA: setLocalDescription with answer BobUA->BobJS: multiple onicecandidate events with candidates // wait for ICE gathering to complete BobUA->BobJS: onicecandidate event with null candidate BobJS->BobUA: get |answer-A1| from currentLocalDescription // |answer-A1| is sent over signaling protocol // to Alice BobJS->WebServer: signaling with |answer-A1| WebServer->AliceJS: signaling with |answer-A1| // |answer-A1| arrives at Alice AliceJS->AliceUA: setRemoteDescription with |answer-A1| AliceUA->AliceJS: ontrack events for audio and video tracks // media flows BobUA->AliceUA: media sent from Bob to Alice AliceUA->BobUA: media sent from Alice to Bob ]]></sourcecode> <t>The SDP for |offer-A1| looks like:</t> <sourcecode name="offer-A1" type="sdp"><![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>]]></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><t> <figure> <artwork alt="answer-A1"> <![CDATA[<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 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>]]></sourcecode> </section> <sectiontitle="Detailed Example" anchor="sec.detailed-example">anchor="sec.detailed-example" numbered="true" toc="default"> <name>Detailed Example</name> <t>This section shows a more involved example of a session between two JSEP endpoints. Trickle ICE is used in full trickle mode, with a bundle policy of "max-bundle", an RTCP mux policy of "require", and a single TURN server. Initially, both Alice and Bob establish an audio channel and a data channel. Later, Bob adds two videoflows,flows -- one for his videofeed,feed and one forscreensharing,screen sharing, both supportingFEC, andFEC -- with the video feed configured for simulcast. Alice accepts these videoflows,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><t> <figure> <artwork> <![CDATA[<sourcecode name="" type="pseudocode"><![CDATA[ // set up local media state AliceJS->AliceUA: create new PeerConnection AliceJS->AliceUA: addTrack with an audio track AliceJS->AliceUA: createDataChannel to get data channel AliceJS->AliceUA: createOffer to get |offer-B1| AliceJS->AliceUA: setLocalDescription with |offer-B1| // |offer-B1| is sent over signaling protocol to Bob AliceJS->WebServer: signaling with |offer-B1| WebServer->BobJS: signaling with |offer-B1| // |offer-B1| arrives at Bob BobJS->BobUA: create a PeerConnection BobJS->BobUA: setRemoteDescription with |offer-B1| BobUA->BobJS: ontrack event with audio track from Alice // candidates are sent to Bob AliceUA->AliceJS: onicecandidate (host) |offer-B1-candidate-1| AliceJS->WebServer: signaling with |offer-B1-candidate-1| AliceUA->AliceJS: onicecandidate (srflx) |offer-B1-candidate-2| AliceJS->WebServer: signaling with |offer-B1-candidate-2| AliceUA->AliceJS: onicecandidate (relay) |offer-B1-candidate-3| AliceJS->WebServer: signaling with |offer-B1-candidate-3| WebServer->BobJS: signaling with |offer-B1-candidate-1| BobJS->BobUA: addIceCandidate with |offer-B1-candidate-1| WebServer->BobJS: signaling with |offer-B1-candidate-2| BobJS->BobUA: addIceCandidate with |offer-B1-candidate-2| WebServer->BobJS: signaling with |offer-B1-candidate-3| BobJS->BobUA: addIceCandidate with |offer-B1-candidate-3| // Bob accepts call BobJS->BobUA: addTrack with local audio BobJS->BobUA: createDataChannel to get data channel BobJS->BobUA: createAnswer to get |answer-B1| BobJS->BobUA: setLocalDescription with |answer-B1| // |answer-B1| is sent to Alice BobJS->WebServer: signaling with |answer-B1| WebServer->AliceJS: signaling with |answer-B1| AliceJS->AliceUA: setRemoteDescription with |answer-B1| AliceUA->AliceJS: ontrack event with audio track from Bob // candidates are sent to Alice BobUA->BobJS: onicecandidate (host) |answer-B1-candidate-1| BobJS->WebServer: signaling with |answer-B1-candidate-1| BobUA->BobJS: onicecandidate (srflx) |answer-B1-candidate-2| BobJS->WebServer: signaling with |answer-B1-candidate-2| BobUA->BobJS: onicecandidate (relay) |answer-B1-candidate-3| BobJS->WebServer: signaling with |answer-B1-candidate-3| WebServer->AliceJS: signaling with |answer-B1-candidate-1| AliceJS->AliceUA: addIceCandidate with |answer-B1-candidate-1| WebServer->AliceJS: signaling with |answer-B1-candidate-2| AliceJS->AliceUA: addIceCandidate with |answer-B1-candidate-2| WebServer->AliceJS: signaling with |answer-B1-candidate-3| AliceJS->AliceUA: addIceCandidate with |answer-B1-candidate-3| // data channel opens BobUA->BobJS: ondatachannel event AliceUA->AliceJS: ondatachannel event BobUA->BobJS: onopen AliceUA->AliceJS: onopen // media is flowing between endpoints BobUA->AliceUA: audio+data sent from Bob to Alice AliceUA->BobUA: audio+data sent from Alice to Bob // some timelaterlater, Bob adds two video streams //note,note: no candidates exchanged, because of bundle BobJS->BobUA: addTrack with first video stream BobJS->BobUA: addTrack with second video stream BobJS->BobUA: createOffer to get |offer-B2| BobJS->BobUA: setLocalDescription with |offer-B2| // |offer-B2| is sent to Alice BobJS->WebServer: signaling with |offer-B2| WebServer->AliceJS: signaling with |offer-B2| AliceJS->AliceUA: setRemoteDescription with |offer-B2| AliceUA->AliceJS: ontrack event with first video track AliceUA->AliceJS: ontrack event with second video track AliceJS->AliceUA: createAnswer to get |answer-B2| AliceJS->AliceUA: setLocalDescription with |answer-B2| // |answer-B2| is sent over signaling protocol // to Bob AliceJS->WebServer: signaling with |answer-B2| WebServer->BobJS: signaling with |answer-B2| BobJS->BobUA: setRemoteDescription with |answer-B2| // media is flowing between endpoints BobUA->AliceUA: audio+video+data sent from Bob to Alice AliceUA->BobUA: audio+video+data sent from Alice to Bob]]> </artwork> </figure> </t>]]></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 with audio track from Alice Currently: BobUA->BobJS: ontrack event with audio track from Alice --> <t>The SDP for |offer-B1| looks like:</t><t> <figure> <artwork alt="offer-B1"> <![CDATA[<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]]> </artwork> </figure> </t>]]></sourcecode> <t>|offer-B1-candidate-1| looks like:</t><t> <figure> <artwork alt="offer-B1-candidate-1"> <![CDATA[<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]]> </artwork> </figure> </t>]]></sourcecode> <t>|offer-B1-candidate-2| looks like:</t><t> <figure> <artwork alt="offer-B1-candidate-2"> <![CDATA[<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]]> </artwork> </figure> </t>]]></sourcecode> <t>|offer-B1-candidate-3| looks like:</t><t> <figure> <artwork alt="offer-B1-candidate-3"> <![CDATA[<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 raddr 198.51.100.100 rport 11100]]> </artwork> </figure> </t>]]></sourcecode> <t>The SDP for |answer-B1| looks like:</t><t> <figure> <artwork alt="answer-B1"> <![CDATA[<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]]> </artwork> </figure> </t>]]></sourcecode> <t>|answer-B1-candidate-1| looks like:</t><t> <figure> <artwork alt="answer-B1-candidate-1"> <![CDATA[<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]]> </artwork> </figure> </t>]]></sourcecode> <t>|answer-B1-candidate-2| looks like:</t><t> <figure> <artwork alt="answer-B1-candidate-2"> <![CDATA[<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]]> </artwork> </figure> </t>]]></sourcecode> <t>|answer-B1-candidate-3| looks like:</t><t> <figure> <artwork alt="answer-B1-candidate-3"> <![CDATA[<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]]> </artwork> </figure> </t>]]></sourcecode> <t>The SDP for |offer-B2| is shown below. In addition to the newm="m=" sections for video, both of which are offeringFEC,FEC and one of which is offering simulcast, note the increment of the version number in theo= line,"o=" line; changes to thec="c=" line, indicating the local candidate that wasselected,selected; and the inclusion of gathered candidates as a=candidate lines.</t><t> <figure> <artwork alt="offer-B2"> <![CDATA[<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]]> </artwork> </figure> </t>]]></sourcecode> <t>The SDP for |answer-B2| is shown below. In addition to the acceptance of the videom="m=" sections, the use of a=recvonly to indicate one-way video, and the use of a=imageattr to limit the received resolution, note the use of setup:passive to maintain the existing DTLS roles.</t><t> <figure> <artwork alt="answer-B2"> <![CDATA[<sourcecode name="answer-B2" type="sdp"><![CDATA[ v=0 o=- 4962303333179871723 2 IN IP4 0.0.0.0 s=- t=0 0 a=ice-options:trickle ice2 a=group:BUNDLE a1 d1 v1 v2 a=group:LS a1 v1 m=audio 12100 UDP/TLS/RTP/SAVPF 96 0 8 97 98 c=IN IP4 192.0.2.100 a=mid:a1 a=sendrecv a=rtpmap:96 opus/48000/2 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=rtpmap:98 telephone-event/48000 a=fmtp:97 0-15 a=fmtp:98 0-15 a=maxptime:120 a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=msid:57017fee-b6c1-4162-929c-a25110252400 a=ice-ufrag:ATEn a=ice-pwd:AtSK0WpNtpUjkY4+86js7ZQl a=fingerprint:sha-256 29:E2:1C:3B:4B:9F:81:E6:B8:5C:F4:A5:A8:D8:73:04: BB:05:2F:70:9F:04:A9:0E:05:E9:26:33:E8:70:88:A2 a=setup:passive a=tls-id:17f0f4ba8a5f1213faca591b58ba52a7 a=rtcp-mux a=rtcp-mux-only a=rtcp-rsize a=candidate:1 1 udp 2113929471 203.0.113.100 10100 typ host a=candidate:1 1 udp 1845494015 198.51.100.100 11100 typ srflx raddr 203.0.113.100 rport 10100 a=candidate:1 1 udp 255 192.0.2.100 12100 typ relay raddr 198.51.100.100 rport 11100 a=end-of-candidates m=application 12100 UDP/DTLS/SCTP webrtc-datachannel c=IN IP4 192.0.2.100 a=mid:d1 a=sctp-port:5000 a=max-message-size:65536 m=video 12100 UDP/TLS/RTP/SAVPF 100 101 102 103 c=IN IP4 192.0.2.100 a=mid:v1 a=recvonly a=rtpmap:100 VP8/90000 a=rtpmap:101 H264/90000 a=fmtp:101 packetization-mode=1;profile-level-id=42e01f a=rtpmap:102 rtx/90000 a=fmtp:102 apt=100 =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]]> </artwork> </figure> </t>]]></sourcecode> </section> <sectiontitle="Earlyanchor="sec.warmup-example" numbered="true" toc="default"> <name>Early Transport WarmupExample" anchor="sec.warmup-example">Example</name> <t>This example demonstrates theearly warmupearly-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/videom= sections,"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 secondofferoffer, which enables the audio and videom="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 transmittedimmediatelyby the callee immediately upon answering the call, minimizing perceivedpost-dial-delay.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><t> <figure> <artwork> <![CDATA[<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 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 towarmupwarm up the // transport BobJS->BobUA: addTransceiver with null audio and video tracks BobJS->BobUA: transceiver.setDirection(sendonly) for both BobJS->BobUA: createAnswer BobJS->BobUA: setLocalDescription with answer // |answer-C1| is sent over signaling protocol // to Alice BobJS->WebServer: signaling with |answer-C1| WebServer->AliceJS: signaling with |answer-C1| // |answer-C1| (sendonly) arrives at Alice AliceJS->AliceUA: setRemoteDescription with |answer-C1| AliceUA->AliceJS: ontrack events for audio and video // a relay candidate is sent to Alice BobUA->BobJS: onicecandidate (relay) |answer-B1-candidate-1| BobJS->WebServer: signaling with |answer-B1-candidate-1| WebServer->AliceJS: signaling with |answer-B1-candidate-1| AliceJS->AliceUA: addIceCandidate with |answer-B1-candidate-1| // ICE and DTLS establish while call is ringing // Bob accepts call, starts media, and sends // new offer BobJS->BobUA: transceiver.setTrack with audio and video tracks BobUA->AliceUA: media sent from Bob to Alice BobJS->BobUA: transceiver.setDirection(sendrecv) for both transceivers BobJS->BobUA: createOffer BobJS->BobUA: setLocalDescription with offer // |offer-C2| is sent over signaling protocol // to Alice BobJS->WebServer: signaling with |offer-C2| WebServer->AliceJS: signaling with |offer-C2| // |offer-C2| (sendrecv) arrives at Alice AliceJS->AliceUA: setRemoteDescription with |offer-C2| AliceJS->AliceUA: createAnswer AliceJS->AliceUA: setLocalDescription with |answer-C2| AliceUA->BobUA: media sent from Alice to Bob // |answer-C2| is sent over signaling protocol // to Bob AliceJS->WebServer: signaling with |answer-C2| WebServer->BobJS: signaling with |answer-C2| BobJS->BobUA: setRemoteDescription with |answer-C2|]]> </artwork> </figure> </t>]]></sourcecode> <t>The SDP for |offer-C1| looks like:</t><t> <figure> <artwork alt="offer-C1"> <![CDATA[<sourcecode name="offer-C1" type="sdp"><![CDATA[ v=0 o=- 1070771854436052752 1 IN IP4 0.0.0.0 s=- t=0 0 a=ice-options:trickle ice2 a=group:BUNDLE a1 v1 a=group:LS a1 v1 m=audio 9 UDP/TLS/RTP/SAVPF 96 0 8 97 98 c=IN IP4 0.0.0.0 a=mid:a1 a=sendrecv a=rtpmap:96 opus/48000/2 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=rtpmap:98 telephone-event/48000 a=fmtp:97 0-15 a=fmtp:98 0-15 a=maxptime:120 a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=msid:bbce3ba6-abfc-ac63-d00a-e15b286f8fce a=ice-ufrag:4ZcD a=ice-pwd:ZaaG6OG7tCn4J/lehAGz+HHD a=fingerprint:sha-256 C4:68:F8:77:6A:44:F1:98:6D:7C:9F:47:EB:E3:34:A4: 0A:AA:2D:49:08:28:70:2E:1F:AE:18:7D:4E:3E:66:BF a=setup:actpass a=tls-id:9e5b948ade9c3d41de6617b68f769e55 a=rtcp-mux a=rtcp-mux-only a=rtcp-rsize m=video 0 UDP/TLS/RTP/SAVPF 100 101 102 103 c=IN IP4 0.0.0.0 a=mid:v1 a=sendrecv a=rtpmap:100 VP8/90000 a=rtpmap:101 H264/90000 a=fmtp:101 packetization-mode=1;profile-level-id=42e01f a=rtpmap:102 rtx/90000 a=fmtp:102 apt=100 =rtpmap:103 rtx/90000 a=fmtp:103 apt=101 a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=rtcp-fb:100 ccm fir a=rtcp-fb:100 nack a=rtcp-fb:100 nack pli a=msid:bbce3ba6-abfc-ac63-d00a-e15b286f8fce a=bundle-only]]> </artwork> </figure> </t>]]></sourcecode> <t>|offer-C1-candidate-1| looks like:</t><t> <figure> <artwork alt="offer-C1-candidate-1"> <![CDATA[<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]]> </artwork> </figure> </t>]]></sourcecode> <t>The SDP for |answer-C1| looks like:</t><t> <figure> <artwork alt="answer-C1"> <![CDATA[<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]]> </artwork> </figure> </t>]]></sourcecode> <t>|answer-C1-candidate-1| looks like:</t><t> <figure> <artwork alt="answer-C1-candidate-1"> <![CDATA[<sourcecode name="answer-C1-candidate-1" type="sdp"><![CDATA[ ufrag TpaA index 0 mid a1 attr candidate:11 udp 255 192.0.2.200 12200 typ relay raddr 0.0.0.0 rport 0 ]]> </artwork> </figure> </t> <t>The SDP for |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>1 udp 255 192.0.2.200 12200 typ relay raddr 0.0.0.0 rport 0 ]]></sourcecode> <t>The SDP for|answer-C2||offer-C2| looks like:</t><t> <figure> <artwork alt="answer-C2"> <![CDATA[<sourcecode name="offer-C2" type="sdp"><![CDATA[ v=0 o=-10707718544360527526386516489780559513 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=audio1210012200 UDP/TLS/RTP/SAVPF 96 0 8 97 98 c=IN IP4192.0.2.100192.0.2.200 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-levela=msid:bbce3ba6-abfc-ac63-d00a-e15b286f8fce a=ice-ufrag:4ZcD a=ice-pwd:ZaaG6OG7tCn4J/lehAGz+HHDa=msid:751f239e-4ae0-c549-aa3d-890de772998b a=ice-ufrag:TpaA a=ice-pwd:t2Ouhc67y8JcCaYZxUUTgKw/ a=fingerprint:sha-256C4:68: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:9e5b948ade9c3d41de6617b68f769e55A2: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 255192.0.2.100 12100192.0.2.200 12200 typ relay raddr 0.0.0.0 rport 0 a=end-of-candidates m=video1210012200 UDP/TLS/RTP/SAVPF 100 101 102 103 c=IN IP4192.0.2.100192.0.2.200 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 nackpli a=msid:bbce3ba6-abfc-ac63-d00a-e15b286f8fce ]]> </artwork> </figure> </t> </section> </section> <section title="Security Considerations" anchor="sec.security-considerations"> <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 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,pli a=msid:751f239e-4ae0-c549-aa3d-890de772998b ]]></sourcecode> <t>The SDP forinstance, 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|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 relaycandidates.</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 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>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> <sectiontitle="Appendix A" anchor="sec.appendix-a"> <t>For the syntax validation performed inanchor="sec.security-considerations" numbered="true" toc="default"> <name>Security Considerations</name> <t>The IETF has published separate documents <xreftarget="sec.parsing-a-desc" />,target="RFC8827" format="default"/> <xref target="RFC8826" format="default"/> describing thefollowing listsecurity architecture for WebRTC as a whole. The remainder ofABNF 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 removethis sectionbefore publication.</t> <t>Changes in draft-26:</t> <t> <list style="symbols"> <t>Update guidance on generation ofdescribes security considerations for this document.</t> <t>While formally them= proto valueJSEP interface is an API, it is better tobe consistentthink of it as an Internet protocol, withice-sip-sdp.</t> </list> </t> <t>Changes in draft-25:</t> <t> <list style="symbols"> <t>Remove MSID track IDthe application JavaScript being untrustworthy fromoffers and answers.</t> <t>Add note about rejecting all m= sectionsthe perspective of the JSEP implementation. Thus, the threat model of <xref target="RFC3552" format="default"/> applies. In particular, JavaScript can call the API ina BUNDLE group.</t> <t>Update ICE references to RFC 8445any order andmention ice2.</t> </list> </t> <t>Changes in draft-24:</t> <t> <list style="symbols"> <t>Clarify that roundingwith any inputs, including malicious ones. This ispermittedparticularly relevant whentrying to maintain aspect ratio.</t> <t>Update tls-id handling to match whatwe consider the SDP that isspecified in dtls-sdp.</t> </list> </t> <t>Changes in draft-23:</t> <t> <list style="symbols"> <t>Clarify rollback handling, and treat it similarlypassed toother setLocal/setRemote usages.</t> <t>Adopt a first-fit policy for handling multiple remote a=imageattr attributes.</t> <t>ClarifysetLocalDescription(). While correct API usage requires thata session description with zero m= sections is legal.</t> </list> </t> <t>Changesthe application pass indraft-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>ClarifySDP thatgenerated ICE candidate objects must have all four fields.</t> <t>Make rollback workwas derived fromany state besides stable and regardless of whether setLocalDescriptioncreateOffer() orsetRemoteDescriptioncreateAnswer(), there isused.</t> <t>Allow modifying SDP before sending or after receiving either offers or answers (previously this was forbidden for answers).</t> <t>Provide rationaleno guarantee that applications do so. The JSEP implementation <bcp14>MUST</bcp14> be prepared forseveral design choices.</t> </list> </t> <t>Changes in draft-21:</t> <t> <list style="symbols"> <t>Change dtls-idthe JavaScript totls-idpass in bogus data instead.</t> <t>Conversely, the application programmer needs tomatch MMUSIC draft.</t> <t>Replace regular expression for proto field with a list and clarifybe aware that theanswer must exactly matchJavaScript does not have complete control of endpoint behavior. One case that bears particular mention is that editing ICE candidates out of theoffer.</t> <t>Remove text about howSDP or suppressing trickled candidates does not have the expected behavior: implementations will still perform checks from those candidates even if they are not sent toerror check on setLocal because local descriptions cannot be changed.</t> <t>Rework silence suppression supportthe other side. Thus, for instance, it is not possible toalways requireprevent the remote peer from learning your public IP address by removing server-reflexive candidates. Applications thatboth sides agreewish tosilence suppression or none is used.</t> <t>Remove instructionsconceal their public IP address should instead configure the ICE agent toparse "a=ssrc-group".</t> <t>Allowuse 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 Candidates for theadditionInteractive 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 ofnew codecs in answersCandidates for the 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 Datagram Transport Layer Security (DTLS) and 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 insubsequent offers.</t> <t>Clarify imageattr processing. Replace use 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>ClarifythenumberSession 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 ofcomponents to gather when bundle is involved.</t> <t>Explicitly state that PTsC238; RFC 8858--> <reference anchor='RFC8858' target="https://www.rfc-editor.org/info/rfc8858"> <front> <title>Indicating Exclusive Support of RTP andSSRCs are to be usedRTP 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="July" 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 fordemuxing.</t> <t>Update guidance on "a=setup" line. This should now matchStream 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 theMMUSIC 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-generatedSession 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 forcorrectness, and use IETF-approved example IP addresses.</t> <t>Add early transport warmup example,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 Session Description Protocol (SDP) andadd missing attributes to existing examples.</t> <t>Only send "a=rtcp-mux-only"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"a=bundle-only" on new m= sections.</t> <t>Update references.</t> <t>Add coverage of a=identity.</t> <t>Explain the lipsync group algorithm more thoroughly.</t> <t>Remove unnecessary listUse ofMTI specs.</t> <t>Allow codecs which weren't offered to appear in answers and which weren't selected to appearRTP insubsequent offers.</t> <t>Codec preferences now are applied on both initial and subsequent offersWebRTC</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 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='July' 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"/> <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="July" 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 cited anywhere in this document. Please let us know where it should be cited. Original: [RFC6544] Rosenberg, J., Keranen, A., Lowekamp, B., andanswers.</t> <t>Clarify a=msid handling for recvonly m= sections.</t> <t>Clarify behavior of attributesA. 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 forbundle-only data channels.</t> <t>Allow media attributes[W3C.webrtc] steers toappear in data m= sections when alla page with a red window that says "This version is outdated!" Because themedia m= sections are bundle-only.</t> <t>Use consistent terminology 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 to an appendix in preparation for merging it into BUNDLE.</t> <t>Clarify whylatest version (September 2018) also discusses the RTCPeerConnection interface, may wecan't handle an incoming offerupdate this listing as suggested below? Also, if you agree tosend simulcast.</t> <t>Expand IceCandidate object text.</t> <t>Further document use of ICE candidate pool.</t> <t>Document removeTrack.</t> <t>Update requirementsthis update, should Jan-Ivar Bruaroey be added toonly acceptthelast generated offer/answer as an argument to setLocalDescription.</t> <t>Allow round pixels.</t> <t>Fix code around default timing when AVPF is not specified.</t> <t>Clean up terminology around m= line and m=section.</t> <t>Provide a more realistic example for minimum decoder capabilities.</t> <t>Document behavior when rtcp-mux policy is require but rtcp-mux attribute not provided.</t> <t>Expanded discussionsecond sentence ofRtpSenderthe 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 andRtpReceiver.</t> <t>Add RtpTransceiver.currentDirectionSystem Aspects; IP Multimedia Subsystem (IMS); Multimedia Telephony; Media handling anddocument setDirection.</t> <t>Require imageattr x=0, y=0 to indicateinteraction (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> thatthere are no valid resolutions.</t> <t>Requireaprivacy-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 codecsnewer version dated September 2019 is available. The new version also discusses CVO (as mentioned inanswersSection 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 andsubsequent offers from non-JSEP peers.</t> <t>Update references.</t> </list> </t> <t>Changes in draft-17:</t> <t> <list style="symbols"> <t>Split createOfferSystem Aspects; IP Multimedia Subsystem (IMS); Multimedia Telephony; Media handling andcreateAnswer sections to clearly indicate attributes which always appearinteraction (Release 12)", December 2014, <http://www.3gpp.org/DynaReport/26114.htm>. Suggested: [TS26.114] 3GPP, "3rd Generation Partnership Project; Technical Specification Group Services andwhich 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 directionsSystem Aspects; IP Multimedia Subsystem (IMS); Multimedia Telephony; Media handling andtheirinteractionwith 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 inreoffers.</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 ABNFtable.</t> </list> </t> <t>Changes in draft-16:</t> <t> <list style="symbols"> <t>Update addIceCandidate to indicate ICE generationdefinitions 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"/>, andallow 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"/> ofRTP 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"/> andpayload 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"/> andcreateDataChannel.</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 thesession 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 thecapabilitiestitle ofthe new session are mostly requiredAppendix A from "Appendix A" tobe 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 isbalanced 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 codecsnot correct, please provide an appropriately descriptive title. b) Please review the citations listed incommon.</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>ChangesTable 1. In many cases, we could not see the relevant parameters listed indraft-14:</t> <t> <list style="symbols"> <t>Added discussionthe cited sections. For example, we do not see * any ofRtpTransceivers + 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 howthe attributesshould 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 ofunilateral PTs.</t> <t>Trigger an ICE restart even if[RFC4566]. * the setup attribute mentioned in Section 3 of [RFC4145]. * theICE candidate policy is being made more strict.</t> <t>Removeconnection 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 inreferences.</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>ChangesSection 5 of [RFC5888]. * the "group" attribute listed indraft-12:</t> <t> <list style="symbols"> <t>FilledSection 4 of [RFC5888]. c) We do not see the attributes "framerate," "quality," or "connection" listed anywhere else insections on applying local and remote descriptions.</t> <t>Discussed downscaling and upscalingthis document. Do they need tofulfill imageattr requirements.</t> <t>Updated what SDP canbemodified by the application.</t> <t>Updated to latest datachannel SDP.</t> <t>Allowed multiple fingerprint lines.</t> <t>Switched backincluded here? d) Please note that in order toIPv4 for dummy candidates.</t> <t>Added additional clarity on ICE default candidates.</t> </list> </t> <t>Changesrender workable hyperlinks indraft-11:</t> <t> <list style="symbols"> <t>Clarified handlingthe .html/.pdf files we changed the original citation format (for example, "[RFC5285] Section 7") to this style (for example, "Section 7 ofRTP 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 beprocessed or ignored.</t> <t>Specified how a=imageattr"5.1," "4.2" should beused.</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 thatdoSection 6.1 of RFC 8853 [I-D.ietf-mmusic-sdp-simulcast] is the correct section to cite here. (We ask because we could nothave mux-only.</t> <t>Add handlingsee a relationship, other than the mention ofLS groups</t> </list> </t> <t>Changes"simulcast" mostly indraft-09:</t> <t> <list style="symbols"> <t>Don't return nullterms 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/DTLSthe following: a) The following terms were used inconsistently inRTP profile names.</t> <t>Separate out bundle and mux policy.</t> <t>Added specific referencesthis document. We chose toFEC 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 <proto> 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 onuse the latter forms. Please let us know any objections. a m= / an "m=" (per published RFCs andc= linesperdiscussion at interim.</t> <t>Added max-bundle-and-rtcp-mux policy.</t> <t>Added description of maxptime handling.</t> <t>Updated ICE candidate pool defaultauthor feedback for other documents in this cluster q= value / "q=" value (We also changed 'x= and y= values' to0.</t> <t>Resolved open issues around AppID/receiver-ID.</t> <t>Reworked"x=" andexpanded how changes"y=" values') a RTP / an RTP to true / to "true" (per theICE 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 discussionrest ofVAD and Opus DTX.</t> <t>Added a security considerations section.</t> <t>Rewrotethesection on modifying SDPcluster) torequire implementationsfalse / toclearly indicate whether any given modification is allowed.</t> <t>Clarified impact"false" (per the rest ofIceRestart on CreateOffer in local-offer state.</t> <t>Guidance on whether attributes should be defined atthemedia levelcluster) 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 orthe session level.</t> <t>Renamed "default" bundle policyTRANSPORT b) The following terms appear to"balanced".</t> <t>Removed default ICE candidate pool sizebe 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" andclarify 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 itworks.</t> <t>Defined a canonical order for assignment of MSTsbe appropriate tom= lines.</t> <t>Removed discussionuse "mid values" (or "MID values"*) and "MIDs" instead ofrehydration.</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 indraft-06: <list style="symbols"> <t>Reworked handling of m= line recycling.</t> <t>Added handlingthis cluster ofBUNDLEdocuments; a separate question regarding which form to use will be sent separately. "RID identifiers" andbundle-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" andits behavior.</t> <t>Allowed OfferToReceiveX"rid-identifier" be "rid-ids" and "rid-id" per RFC 8851 (draft-ietf-mmusic-rid), tocreate multiple recvonly m= sections.</t> </list></t> <t>Changes in draft-05: <list style="symbols"> <t>Fixed several issues identified inavoid "identifier identifier(s)"? thecreateOffer/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 relationshipstable 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 toW3C specification</t> </list></t> <t>Changes in draft-02: <list style="symbols"> <!-- A --> <t>Converted from nroff</t> <!-- B --> <t>Removed comparisonszero / set toold approaches abandoned by0 (May we use theworking 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>Changeslatter, because of '"dummy" port value of 9' indraft-01: <list style="symbols"> <t>Added diagrams for architectureSections 5.2.1 andstate machine.</t> <t>Added5.3.1?) 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= sectionson forking and rehydration.</t> <t>Clarified meaningby the application of"pranswer" and "answer".</t> <t>Reworked how ICE restartsthe rolled-back session description (see Section 5.10 andmedia directions are controlled.</t> <t>Added list of parameters that canSection 5.9)." c) Are "canTrickleIceCandidates" (Section 4.1.15) and "canTrickle property" (Section 5.10) distinct terms? If so, should this distinction bechanged 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>Changesclarified indraft -00: <list style="symbols"> <t>Migrated from draft-uberti-rtcweb-jsep-02.</t> </list></t> </section> </back>the text? --> </rfc>