@abrar71/lib-jitsi-meet 0.0.0

package/dist/esm/modules/RTC/TraceablePeerConnection.js

@@ -0,0 +1,2223 @@
+import { getLogger } from '@jitsi/logger';
+import { cloneDeep } from 'lodash-es';
+import transform from 'sdp-transform';
+import { CodecMimeType } from '../../service/RTC/CodecMimeType';
+import { MediaDirection } from '../../service/RTC/MediaDirection';
+import { MediaType } from '../../service/RTC/MediaType';
+import { RTCEvents } from '../../service/RTC/RTCEvents';
+import { SignalingEvents } from '../../service/RTC/SignalingEvents';
+import { getSourceIndexFromSourceName } from '../../service/RTC/SignalingLayer';
+import { SSRC_GROUP_SEMANTICS, VIDEO_QUALITY_LEVELS } from '../../service/RTC/StandardVideoQualitySettings';
+import { VideoType } from '../../service/RTC/VideoType';
+import { AnalyticsEvents } from '../../service/statistics/AnalyticsEvents';
+import RTCStats from '../RTCStats/RTCStats';
+import { RTCStatsEvents } from '../RTCStats/RTCStatsEvents';
+import browser from '../browser';
+import FeatureFlags from '../flags/FeatureFlags';
+import LocalSdpMunger from '../sdp/LocalSdpMunger';
+import RtxModifier from '../sdp/RtxModifier';
+import SDP from '../sdp/SDP';
+import SDPUtil from '../sdp/SDPUtil';
+import SdpSimulcast from '../sdp/SdpSimulcast';
+import { SdpTransformWrap } from '../sdp/SdpTransformUtil';
+import Statistics from '../statistics/statistics';
+import { isValidNumber } from '../util/MathUtil';
+import JitsiRemoteTrack from './JitsiRemoteTrack';
+import RTCUtils from './RTCUtils';
+import { SS_DEFAULT_FRAME_RATE } from './ScreenObtainer';
+import { TPCUtils } from './TPCUtils';
+const logger = getLogger('rtc:TraceablePeerConnection');
+const DEGRADATION_PREFERENCE_CAMERA = 'maintain-framerate';
+const DEGRADATION_PREFERENCE_DESKTOP = 'maintain-resolution';
+/* eslint-disable max-params */
+/**
+ * Creates new instance of 'TraceablePeerConnection'.
+ */
+class TraceablePeerConnection {
+    /**
+     * @param {RTC} rtc the instance of <tt>RTC</tt> service
+     * @param {number} id the peer connection id assigned by the parent RTC module.
+     * @param {SignalingLayer} signalingLayer the signaling layer instance
+     * @param {object} pcConfig The {@code RTCConfiguration} to use for the WebRTC peer connection.
+     * @param {object} constraints WebRTC 'PeerConnection' constraints
+     * @param {boolean} isP2P indicates whether or not the new instance will be used in a peer to peer connection.
+     * @param {object} options <tt>TracablePeerConnection</tt> config options.
+     * @param {Object} options.audioQuality - Quality settings to applied on the outbound audio stream.
+     * @param {boolean} options.capScreenshareBitrate if set to true, lower layers will be disabled for screenshare.
+     * @param {Array<CodecMimeType>} options.codecSettings - codec settings to be applied for video streams.
+     * @param {boolean} options.disableSimulcast if set to 'true' will disable the simulcast.
+     * @param {boolean} options.disableRtx if set to 'true' will disable the RTX.
+     * @param {boolean} options.enableInsertableStreams set to true when the insertable streams constraints is to be
+     * enabled on the PeerConnection.
+     * @param {boolean} options.forceTurnRelay If set to true, the browser will generate only Relay ICE candidates.
+     * @param {boolean} options.startSilent If set to 'true' no audio will be sent or received.
+     * @param {Object} options.videoQuality - Quality settings to applied on the outbound video streams.
+     *
+     * FIXME: initially the purpose of TraceablePeerConnection was to be able to
+     * debug the peer connection. Since many other responsibilities have been added
+     * it would make sense to extract a separate class from it and come up with
+     * a more suitable name.
+     *
+     * @constructor
+     */
+    constructor(rtc, id, signalingLayer, pcConfig, constraints, isP2P, options) {
+        /**
+         * Configures the RTCRtpEncodingParameters of the outbound rtp stream associated with the given track.
+         *
+         * @param {JitsiLocalTracj} localTrack - The local track whose outbound stream needs to be configured.
+         * @returns {Promise} - A promise that resolves when the operation is successful, rejected otherwise.
+         */
+        this._configureSenderEncodings = async (localTrack) => {
+            const mediaType = localTrack.getType();
+            const transceiver = localTrack?.track && localTrack.getOriginalStream()
+                ? this.peerconnection.getTransceivers().find(t => t.sender?.track?.id === localTrack.getTrackId())
+                : this.peerconnection.getTransceivers().find(t => t.receiver?.track?.kind === mediaType);
+            const parameters = transceiver?.sender?.getParameters();
+            // Resolve if the encodings are not available yet. This happens immediately after the track is added to the
+            // peerconnection on chrome in unified-plan. It is ok to ignore and not report the error here since the
+            // action that triggers 'addTrack' (like unmute) will also configure the encodings and set bitrates after that.
+            if (!parameters?.encodings?.length) {
+                return;
+            }
+            parameters.encodings = this.tpcUtils.getStreamEncodings(localTrack);
+            await transceiver.sender.setParameters(parameters);
+        };
+        /**
+         * Enables/disables the streams by changing the active field on RTCRtpEncodingParameters for a given RTCRtpSender.
+         *
+         * @param {RTCRtpSender} sender - the sender associated with a MediaStreamTrack.
+         * @param {boolean} enable - whether the streams needs to be enabled or disabled.
+         * @returns {Promise} - A promise that resolves when the operation is successful, rejected otherwise.
+         */
+        this._enableSenderEncodings = async (sender, enable) => {
+            const parameters = sender.getParameters();
+            if (parameters?.encodings?.length) {
+                for (const encoding of parameters.encodings) {
+                    encoding.active = enable;
+                }
+            }
+            await sender.setParameters(parameters);
+        };
+        /**
+         * Add {@link JitsiLocalTrack} to this TPC.
+         * @param {JitsiLocalTrack} track
+         * @param {boolean} isInitiator indicates if the endpoint is the offerer.
+         * @returns {Promise<void>} - resolved when done.
+         */
+        this.addTrack = async (track, isInitiator = false) => {
+            const rtcId = track.rtcId;
+            if (this.localTracks.has(rtcId)) {
+                throw new Error(`${track} is already in ${this}`);
+            }
+            logger.info(`${this} adding ${track}`);
+            const webrtcStream = track.getOriginalStream();
+            const mediaStreamTrack = track.getTrack();
+            let transceiver;
+            if (isInitiator) {
+                const streams = [];
+                webrtcStream && streams.push(webrtcStream);
+                // Use pc.addTransceiver() for the initiator case when local tracks are getting added
+                // to the peerconnection before a session-initiate is sent over to the peer.
+                const transceiverInit = {
+                    direction: MediaDirection.SENDRECV,
+                    sendEncodings: [],
+                    streams
+                };
+                if (!browser.isFirefox()) {
+                    transceiverInit.sendEncodings = this.tpcUtils.getStreamEncodings(track);
+                }
+                transceiver = this.peerconnection.addTransceiver(mediaStreamTrack, transceiverInit);
+            }
+            else {
+                // Use pc.addTrack() for responder case so that we can re-use the m-lines that were created
+                // when setRemoteDescription was called. pc.addTrack() automatically  attaches to any existing
+                // unused "recv-only" transceiver.
+                const sender = this.peerconnection.addTrack(mediaStreamTrack);
+                // Find the corresponding transceiver that the track was attached to.
+                transceiver = this.peerconnection.getTransceivers().find(t => t.sender === sender);
+            }
+            if (transceiver?.mid) {
+                this.localTrackTransceiverMids.set(track.rtcId, transceiver.mid.toString());
+            }
+            if (track) {
+                this.localTracks.set(rtcId, track);
+                if (track.isAudioTrack()) {
+                    this._hasHadAudioTrack = true;
+                }
+                else {
+                    this._hasHadVideoTrack = true;
+                }
+            }
+            // On Firefox, the encodings have to be configured on the sender only after the transceiver is created.
+            if (browser.isFirefox() && webrtcStream && this.doesTrueSimulcast(track)) {
+                await this._setEncodings(track);
+            }
+        };
+        /**
+         * Indicates whether or not this peer connection instance is actively
+         * sending/receiving audio media. When set to <tt>false</tt> the SDP audio
+         * media direction will be adjusted to 'inactive' in order to suspend
+         * the transmission.
+         * @type {boolean}
+         * @internal
+         */
+        this.audioTransferActive = !(options.startSilent === true);
+        /**
+         * The DTMF sender instance used to send DTMF tones.
+         *
+         * @type {RTCDTMFSender|undefined}
+         * @private
+         */
+        this._dtmfSender = undefined;
+        /**
+         * @typedef {Object} TouchToneRequest
+         * @property {string} tones - The DTMF tones string as defined by
+         * {@code RTCDTMFSender.insertDTMF}, 'tones' argument.
+         * @property {number} duration - The amount of time in milliseconds that
+         * each DTMF should last.
+         * @property {string} interToneGap - The length of time in miliseconds to
+         * wait between tones.
+         */
+        /**
+         * TouchToneRequests which are waiting to be played. This queue is filled
+         * if there are touch tones currently being played.
+         *
+         * @type {Array<TouchToneRequest>}
+         * @private
+         */
+        this._dtmfTonesQueue = [];
+        /**
+         * Indicates whether or not this peer connection instance is actively
+         * sending/receiving video media. When set to <tt>false</tt> the SDP video
+         * media direction will be adjusted to 'inactive' in order to suspend
+         * the transmission.
+         * @type {boolean}
+         * @internal
+         */
+        this.videoTransferActive = true;
+        /**
+         * The parent instance of RTC service which created this
+         * <tt>TracablePeerConnection</tt>.
+         * @type {RTC}
+         */
+        this.rtc = rtc;
+        /**
+         * The peer connection identifier assigned by the RTC module.
+         * @type {number}
+         */
+        this.id = id;
+        /**
+         * RTCStats identifier for this peerconnection.
+         */
+        this._pcId = `PC_${this.id}`;
+        /**
+         * Indicates whether or not this instance is used in a peer to peer
+         * connection.
+         * @type {boolean}
+         */
+        this.isP2P = isP2P;
+        /**
+         * A map that holds remote tracks signaled on the peerconnection indexed by their SSRC.
+         * @type {Map<number, JitsiRemoteTrack>}
+         */
+        this.remoteTracksBySsrc = new Map();
+        /**
+         * The map holds remote tracks associated with this peer connection.
+         * It maps user's JID to media type and a set of
+         * remote tracks.
+         * @type {Map<string, Map<MediaType, Set<JitsiRemoteTrack>>>}
+         */
+        this.remoteTracks = new Map();
+        /**
+         * A map which stores local tracks mapped by {@link JitsiLocalTrack.rtcId}
+         * @type {Map<number, JitsiLocalTrack>}
+         */
+        this.localTracks = new Map();
+        /**
+         * @typedef {Object} TPCGroupInfo
+         * @property {string} semantics the SSRC groups semantics
+         * @property {Array<number>} ssrcs group's SSRCs in order where the first
+         * one is group's primary SSRC, the second one is secondary (RTX) and so
+         * on...
+         */
+        /**
+         * @typedef {Object} TPCSSRCInfo
+         * @property {Array<number>} ssrcs an array which holds all track's SSRCs
+         * @property {Array<TPCGroupInfo>} groups an array stores all track's SSRC
+         * groups
+         */
+        /**
+         * Holds the info about local track's SSRCs mapped per their
+         * {@link JitsiLocalTrack.rtcId}
+         * @type {Map<number, TPCSSRCInfo>}
+         */
+        this.localSSRCs = new Map();
+        /**
+         * The set of remote SSRCs seen so far.
+         * Distinguishes new SSRCs from those that have been remapped.
+         * @type {Set<number>}
+         */
+        this.remoteSSRCs = new Set();
+        /**
+         * Mapping of source-names and their associated SSRCs that have been signaled by the JVB.
+         * @type {Map<string, number>}
+         */
+        this.remoteSources = new Map();
+        /**
+         * The local ICE username fragment for this session.
+         */
+        this._localUfrag = null;
+        /**
+         * The remote ICE username fragment for this session.
+         */
+        this._remoteUfrag = null;
+        /**
+         * The DTLS transport object for the PeerConnection.
+         * Note: this assume only one shared transport exists because we bundled
+         *       all streams on the same underlying transport.
+         */
+        this._dtlsTransport = null;
+        /**
+         * The signaling layer which operates this peer connection.
+         * @type {SignalingLayer}
+         */
+        this._signalingLayer = signalingLayer;
+        // SignalingLayer listeners
+        this._peerVideoTypeChanged = this._peerVideoTypeChanged.bind(this);
+        this._signalingLayer.on(SignalingEvents.PEER_VIDEO_TYPE_CHANGED, this._peerVideoTypeChanged);
+        this._peerMutedChanged = this._peerMutedChanged.bind(this);
+        this._signalingLayer.on(SignalingEvents.PEER_MUTED_CHANGED, this._peerMutedChanged);
+        this.options = options;
+        // Setup SignalingLayer listeners for source-name based events.
+        this._signalingLayer.on(SignalingEvents.SOURCE_MUTED_CHANGED, (sourceName, isMuted) => this._sourceMutedChanged(sourceName, isMuted));
+        this._signalingLayer.on(SignalingEvents.SOURCE_VIDEO_TYPE_CHANGED, (sourceName, videoType) => this._sourceVideoTypeChanged(sourceName, videoType));
+        // Make sure constraints is properly formatted in order to provide information about whether or not this
+        // connection is P2P to rtcstats.
+        const safeConstraints = constraints || {};
+        safeConstraints.optional = safeConstraints.optional || [];
+        // The `optional` parameter needs to be of type array, otherwise chrome will throw an error.
+        // Firefox and Safari just ignore it.
+        if (Array.isArray(safeConstraints.optional)) {
+            safeConstraints.optional.push({ rtcStatsSFUP2P: this.isP2P });
+        }
+        else {
+            logger.warn('Optional param is not an array, rtcstats p2p data is omitted.');
+        }
+        // RTCPeerConnection constructor only accepts RTCConfiguration
+        // @ts-ignore
+        this.peerconnection = new RTCPeerConnection(pcConfig, safeConstraints);
+        this.tpcUtils = new TPCUtils(this, {
+            audioQuality: options.audioQuality,
+            isP2P: this.isP2P,
+            videoQuality: options.videoQuality
+        });
+        this.updateLog = [];
+        this.stats = {};
+        this.statsinterval = null;
+        /**
+         * Flag used to indicate if low fps screenshare is desired.
+         */
+        this._capScreenshareBitrate = this.options.capScreenshareBitrate;
+        /**
+         * Codec preferences set for the peerconnection through config.js.
+         */
+        this.codecSettings = this.options.codecSettings;
+        /**
+         * Flag used to indicate if RTCRtpTransceiver#setCodecPreferences is to be used instead of SDP
+         * munging for codec selection.
+         */
+        browser.supportsCodecPreferences()
+            && logger.info('Using RTCRtpTransceiver#setCodecPreferences for codec selection');
+        /**
+         * Flag used to indicate if the codecs are configured using the codec selection API without having the need to
+         * trigger a renegotiation for the change to be effective.
+         */
+        this._usesCodecSelectionAPI = this.options.usesCodecSelectionAPI;
+        /**
+         * Indicates whether an audio track has ever been added to the peer connection.
+         */
+        this._hasHadAudioTrack = false;
+        /**
+         * Indicates whether a video track has ever been added to the peer connection.
+         */
+        this._hasHadVideoTrack = false;
+        /**
+         * @type {number} The max number of stats to keep in this.stats. Limit to
+         * 300 values, i.e. 5 minutes; set to 0 to disable
+         */
+        this.maxstats = options.maxstats;
+        this.simulcast = new SdpSimulcast();
+        /**
+         * Munges local SDP provided to the Jingle Session in order to prevent from
+         * sending SSRC updates on attach/detach and mute/unmute (for video).
+         * @type {LocalSdpMunger}
+         */
+        this.localSdpMunger = new LocalSdpMunger(this, this.rtc.getLocalEndpointId());
+        /**
+         * TracablePeerConnection uses RTC's eventEmitter
+         * @type {EventEmitter}
+         */
+        this.eventEmitter = rtc.eventEmitter;
+        this.rtxModifier = new RtxModifier();
+        /**
+         * The height constraints to be applied on the sender per local video source (source name as the key).
+         * @type {Map<string, number>}
+         */
+        this._senderMaxHeights = new Map();
+        /**
+         * Holds the RTCRtpTransceiver mids that the local tracks are attached to, mapped per their
+         * {@link JitsiLocalTrack.rtcId}.
+         * @type {Map<string, string>}
+         */
+        this.localTrackTransceiverMids = new Map();
+        /**
+         * Holds the SSRC map for the local tracks mapped by their source names.
+         *
+         * @type {Map<string, TPCSourceInfo>}
+         * @property {string} msid - The track's MSID.
+         * @property {Array<string>} ssrcs - The SSRCs associated with the track.
+         * @property {Array<TPCGroupInfo>} groups - The SSRC groups associated with the track.
+         */
+        this._localSsrcMap = null;
+        /**
+         * Holds the SSRC map for the remote tracks mapped by their source names.
+         *
+         * @type {Map<string, TPCSourceInfo>}
+         * @property {string} mediaType - The media type of the track.
+         * @property {string} msid - The track's MSID.
+         * @property {Array<TPCGroupInfo>} groups - The SSRC groups associated with the track.
+         * @property {Array<string>} ssrcList - The SSRCs associated with the track.
+         * @property {VideoType} videoType - The videoType of the track (undefined for audio tracks).
+         */
+        this._remoteSsrcMap = new Map();
+        // override as desired
+        this.trace = (what, info) => {
+            logger.trace(what, info);
+            this.updateLog.push({
+                time: new Date(),
+                type: what,
+                value: info || ''
+            });
+        };
+        this.onicecandidate = null;
+        this.peerconnection.onicecandidate = event => {
+            this.trace('onicecandidate', JSON.stringify(event.candidate, null, ' '));
+            if (this.onicecandidate !== null) {
+                this.onicecandidate(event);
+            }
+        };
+        this.onTrack = evt => {
+            const stream = evt.streams[0];
+            this._remoteTrackAdded(stream, evt.track, evt.transceiver);
+            stream.addEventListener('removetrack', e => {
+                this._remoteTrackRemoved(stream, e.track);
+            });
+        };
+        this.peerconnection.addEventListener('track', this.onTrack);
+        this.onsignalingstatechange = null;
+        this.peerconnection.onsignalingstatechange = event => {
+            this.trace('onsignalingstatechange', this.signalingState);
+            if (this.onsignalingstatechange !== null) {
+                this.onsignalingstatechange(event);
+            }
+        };
+        this.oniceconnectionstatechange = null;
+        this.peerconnection.oniceconnectionstatechange = event => {
+            this.trace('oniceconnectionstatechange', this.iceConnectionState);
+            if (this.oniceconnectionstatechange !== null) {
+                this.oniceconnectionstatechange(event);
+            }
+        };
+        this.onnegotiationneeded = null;
+        this.peerconnection.onnegotiationneeded = event => {
+            this.trace('onnegotiationneeded');
+            if (this.onnegotiationneeded !== null) {
+                this.onnegotiationneeded(event);
+            }
+        };
+        this.onconnectionstatechange = null;
+        this.peerconnection.onconnectionstatechange = event => {
+            this.trace('onconnectionstatechange', this.connectionState);
+            if (this.onconnectionstatechange !== null) {
+                this.onconnectionstatechange(event);
+            }
+        };
+        this.ondatachannel = null;
+        this.peerconnection.ondatachannel = event => {
+            this.trace('ondatachannel');
+            if (this.ondatachannel !== null) {
+                this.ondatachannel(event);
+            }
+        };
+        if (this.maxstats) {
+            this.statsinterval = window.setInterval(() => {
+                this.getStats().then((stats) => {
+                    if (typeof stats?.result === 'function') {
+                        const results = stats.result();
+                        for (let i = 0; i < results.length; ++i) {
+                            const res = results[i];
+                            res.names().forEach(name => {
+                                this._processStat(res, name, res.stat(name));
+                            });
+                        }
+                    }
+                    else {
+                        stats.forEach(r => this._processStat(r, '', r));
+                    }
+                });
+            }, 1000);
+        }
+        this._lastVideoSenderUpdatePromise = Promise.resolve();
+        logger.info(`Create new ${this}`);
+    }
+    /**
+     * Handles remote track mute / unmute events.
+     * @param {string} endpointId the track owner's identifier (MUC nickname)
+     * @param {MediaType} mediaType "audio" or "video"
+     * @param {boolean} isMuted the new mute state
+     * @private
+     */
+    _peerMutedChanged(endpointId, mediaType, isMuted) {
+        // Check if endpointId is a value to avoid doing action on all remote tracks
+        if (!endpointId) {
+            logger.error(`${this} On peerMuteChanged - no endpoint ID`);
+            return;
+        }
+        const track = this.getRemoteTracks(endpointId, mediaType);
+        if (track.length) {
+            // NOTE 1 track per media type is assumed
+            track[0].setMute(isMuted);
+        }
+    }
+    /**
+     * Handles remote source videoType changed events.
+     *
+     * @param {string} sourceName - The name of the remote source.
+     * @param {boolean} isMuted - The new value.
+     */
+    _sourceVideoTypeChanged(sourceName, videoType) {
+        const track = this.getRemoteTracks().slice().reverse().find(t => t.getSourceName() === sourceName);
+        if (!track) {
+            return;
+        }
+        track._setVideoType(videoType);
+    }
+    /**
+     * Returns the list of RTCRtpReceivers created for the source of the given media type associated with
+     * the set of remote endpoints specified.
+     * @param {Array<string>} endpoints list of the endpoints
+     * @param {string} mediaType 'audio' or 'video'
+     * @returns {Array<RTCRtpReceiver>} list of receivers created by the peerconnection.
+     */
+    _getReceiversByEndpointIds(endpoints, mediaType) {
+        let remoteTracks = [];
+        let receivers = [];
+        for (const endpoint of endpoints) {
+            remoteTracks = remoteTracks.concat(this.getRemoteTracks(endpoint, mediaType));
+        }
+        // Get the ids of the MediaStreamTracks associated with each of these remote tracks.
+        const remoteTrackIds = remoteTracks.map(remote => remote.track?.id);
+        receivers = this.peerconnection.getReceivers()
+            .filter(receiver => receiver.track
+            && receiver.track.kind === mediaType
+            && remoteTrackIds.find(trackId => trackId === receiver.track.id));
+        return receivers;
+    }
+    /**
+     * Handles {@link SignalingEvents.PEER_VIDEO_TYPE_CHANGED}
+     * @param {string} endpointId the video owner's ID (MUC nickname)
+     * @param {VideoType} videoType the new value
+     * @private
+     */
+    _peerVideoTypeChanged(endpointId, videoType) {
+        // Check if endpointId has a value to avoid action on random track
+        if (!endpointId) {
+            logger.error(`${this} No endpointID on peerVideoTypeChanged`);
+            return;
+        }
+        const videoTrack = this.getRemoteTracks(endpointId, MediaType.VIDEO);
+        if (videoTrack.length) {
+            // NOTE 1 track per media type is assumed
+            videoTrack[0]._setVideoType(videoType);
+        }
+    }
+    /**
+     * Adjusts the media direction on the remote description based on availability of local and remote sources in a p2p
+     * media connection.
+     *
+     * @param {RTCSessionDescription} remoteDescription the WebRTC session description
+     *  instance for the remote description.
+     * @returns the transformed remoteDescription.
+     * @private
+     */
+    _adjustRemoteMediaDirection(remoteDescription) {
+        const transformer = new SdpTransformWrap(remoteDescription?.sdp);
+        [MediaType.AUDIO, MediaType.VIDEO].forEach(mediaType => {
+            const media = transformer.selectMedia(mediaType);
+            const localSources = this.getLocalTracks(mediaType).length;
+            const remoteSources = this.getRemoteTracks(null, mediaType).length;
+            media.forEach((mLine, idx) => {
+                if (localSources && localSources === remoteSources) {
+                    mLine.direction = MediaDirection.SENDRECV;
+                }
+                else if (!localSources && !remoteSources) {
+                    mLine.direction = MediaDirection.INACTIVE;
+                }
+                else if (!localSources) {
+                    mLine.direction = MediaDirection.SENDONLY;
+                }
+                else if (!remoteSources) {
+                    mLine.direction = MediaDirection.RECVONLY;
+                    // When there are 2 local sources and 1 remote source,
+                    // the first m-line should be set to 'sendrecv' while
+                    // the second one needs to be set to 'recvonly'.
+                }
+                else if (localSources > remoteSources) {
+                    mLine.direction = idx ? MediaDirection.RECVONLY : MediaDirection.SENDRECV;
+                    // When there are 2 remote sources and 1 local source, the first m-line
+                    // should be set to 'sendrecv' while
+                    // the second one needs to be set to 'sendonly'.
+                }
+                else {
+                    mLine.direction = idx ? MediaDirection.SENDONLY : MediaDirection.SENDRECV;
+                }
+            });
+        });
+        return new RTCSessionDescription({
+            sdp: transformer.toRawSDP(),
+            type: remoteDescription.type
+        });
+    }
+    /**
+     * Returns the codec to be used for screenshare based on the supported codecs and the preferred codec requested
+     * through config.js setting.
+     *
+     * @param {CodecMimeType} defaultCodec - the preferred codec for video tracks.
+     * @returns {CodecMimeType}
+     */
+    _getPreferredCodecForScreenshare(defaultCodec) {
+        // Use the same codec for both camera and screenshare if the client doesn't support the codec selection API.
+        if (!this.usesCodecSelectionAPI()) {
+            return defaultCodec;
+        }
+        const { screenshareCodec } = this.codecSettings;
+        if (screenshareCodec && this.codecSettings.codecList.find(c => c === screenshareCodec)) {
+            return screenshareCodec;
+        }
+        // Default to AV1 for screenshare if its supported and is not overriden through config.js.
+        if (this.codecSettings.codecList.find(c => c === CodecMimeType.AV1)) {
+            return CodecMimeType.AV1;
+        }
+        return defaultCodec;
+    }
+    /**
+     * Sets up the _dtlsTransport object and initializes callbacks for it.
+     */
+    _initializeDtlsTransport() {
+        // We are assuming here that we only have one bundled transport here
+        if (!this.peerconnection.getSenders || this._dtlsTransport) {
+            return;
+        }
+        const senders = this.peerconnection.getSenders();
+        if (senders.length !== 0 && senders[0].transport) {
+            this._dtlsTransport = senders[0].transport;
+            this._dtlsTransport.onerror = error => {
+                logger.error(`${this} DtlsTransport error: ${error}`);
+            };
+            this._dtlsTransport.onstatechange = () => {
+                this.trace('dtlsTransport.onstatechange', this._dtlsTransport.state);
+            };
+        }
+    }
+    /**
+     * Set the simulcast stream encoding properties on the RTCRtpSender.
+     *
+     * @param {JitsiLocalTrack} localTrack - the current track in use for which the encodings are to be set.
+     * @returns {Promise<void>} - resolved when done.
+     */
+    _setEncodings(localTrack) {
+        if (localTrack.getType() === MediaType.VIDEO) {
+            return this._updateVideoSenderParameters(() => this._configureSenderEncodings(localTrack));
+        }
+        return this._configureSenderEncodings(localTrack);
+    }
+    /**
+     * Munges the provided description to update the codec order, set the max bitrates (for VP9 K-SVC), set stereo flag
+     * and update the DD Header extensions for AV1.
+     *
+     * @param {RTCSessionDescription} description - The description to be munged.
+     * @returns {RTCSessionDescription} - The munged description.
+     */
+    _mungeDescription(description) {
+        this.trace('RTCSessionDescription::preTransform', TraceablePeerConnection.dumpSDP(description));
+        let mungedSdp = transform.parse(description?.sdp);
+        mungedSdp = this.tpcUtils.mungeOpus(mungedSdp);
+        mungedSdp = this.tpcUtils.mungeCodecOrder(mungedSdp);
+        mungedSdp = this.tpcUtils.setMaxBitrates(mungedSdp, true);
+        const mungedDescription = new RTCSessionDescription({
+            sdp: transform.write(mungedSdp),
+            type: description.type
+        });
+        this.trace('RTCSessionDescription::postTransform', TraceablePeerConnection.dumpSDP(mungedDescription));
+        return mungedDescription;
+    }
+    /**
+     * Returns a wrapped-up promise so that the setParameters() call on the RTCRtpSender
+     * for video sources are chained.
+     * This is needed on Chrome as it resets the transaction id after
+     * executing setParameters() and can affect the next on
+     * the fly updates if they are not chained.
+     * https://chromium.googlesource.com/external/webrtc/+/master/pc/rtp_sender.cc#340
+     * @param {Function} nextFunction - The function to be called when the last video sender update promise is settled.
+     * @returns {Promise}
+     */
+    _updateVideoSenderParameters(nextFunction) {
+        const nextPromise = this._lastVideoSenderUpdatePromise
+            .finally(nextFunction);
+        this._lastVideoSenderUpdatePromise = nextPromise;
+        return nextPromise;
+    }
+    /**
+     * Configures the video stream with resolution / degradation / maximum bitrates
+     *
+     * @param {number} frameHeight - The max frame height to be imposed on the outgoing video stream.
+     * @param {JitsiLocalTrack} - The local track for which the sender constraints have to be applied.
+     * @param {preferredCodec} - The video codec that needs to be configured on
+     * the sender associated with the video source.
+     * @returns {Promise} promise that will be resolved when the operation is successful and rejected otherwise.
+     */
+    _updateVideoSenderEncodings(frameHeight, localVideoTrack, preferredCodec) {
+        const videoSender = this.findSenderForTrack(localVideoTrack.getTrack());
+        const videoType = localVideoTrack.getVideoType();
+        const isScreensharingTrack = videoType === VideoType.DESKTOP;
+        if (!videoSender) {
+            return Promise.resolve();
+        }
+        const parameters = videoSender.getParameters();
+        if (!parameters?.encodings?.length) {
+            return Promise.resolve();
+        }
+        const isSharingLowFpsScreen = isScreensharingTrack && this._capScreenshareBitrate;
+        // Set the degradation preference.
+        const preference = isSharingLowFpsScreen
+            ? DEGRADATION_PREFERENCE_DESKTOP // Prefer resolution for low fps share.
+            : DEGRADATION_PREFERENCE_CAMERA; // Prefer frame-rate for high fps share and camera.
+        parameters.degradationPreference = preference;
+        // Calculate the encodings active state based on the resolution requested by the bridge.
+        const codecForCamera = preferredCodec ?? this.tpcUtils.getConfiguredVideoCodec(localVideoTrack);
+        const codec = isScreensharingTrack ? this._getPreferredCodecForScreenshare(codecForCamera) : codecForCamera;
+        const activeState = this.tpcUtils.calculateEncodingsActiveState(localVideoTrack, codec, frameHeight);
+        let bitrates = this.tpcUtils.calculateEncodingsBitrates(localVideoTrack, codec, frameHeight);
+        const scalabilityModes = this.tpcUtils.calculateEncodingsScalabilityMode(localVideoTrack, codec, frameHeight);
+        let scaleFactors = this.tpcUtils.calculateEncodingsScaleFactor(localVideoTrack, codec, frameHeight);
+        let needsUpdate = false;
+        // Do not configure 'scaleResolutionDownBy' and 'maxBitrate' for encoders running in VP9 legacy K-SVC mode since
+        // the browser sends only the lowest resolution layer when those
+        // are configured. Those fields need to be reset in
+        // case they were set when the endpoint was encoding video using the other codecs before switching over to VP9
+        // K-SVC codec.
+        if (codec === CodecMimeType.VP9
+            && browser.supportsSVC()
+            && this.isSpatialScalabilityOn()
+            && !this.tpcUtils.codecSettings[codec].scalabilityModeEnabled) {
+            scaleFactors = scaleFactors.map(() => undefined);
+            bitrates = bitrates.map(() => undefined);
+        }
+        for (const idx in parameters.encodings) {
+            if (parameters.encodings.hasOwnProperty(idx)) {
+                const encoding = parameters.encodings[idx];
+                const { active = undefined, codec: currentCodec = undefined, maxBitrate = undefined, scalabilityMode = undefined, scaleResolutionDownBy = undefined } = encoding;
+                if (active !== activeState[idx]) {
+                    encoding.active = activeState[idx];
+                    needsUpdate = true;
+                }
+                // Firefox doesn't follow the spec and lets application specify the degradation preference on the
+                // encodings.
+                browser.isFirefox() && (encoding.degradationPreference = preference);
+                if (scaleResolutionDownBy !== scaleFactors[idx]) {
+                    encoding.scaleResolutionDownBy = scaleFactors[idx];
+                    needsUpdate = true;
+                }
+                if (maxBitrate !== bitrates[idx]) {
+                    encoding.maxBitrate = bitrates[idx];
+                    needsUpdate = true;
+                }
+                // Configure scalability mode when its supported and enabled.
+                if (scalabilityModes) {
+                    if (scalabilityMode !== scalabilityModes[idx]) {
+                        encoding.scalabilityMode = scalabilityModes[idx];
+                        needsUpdate = true;
+                    }
+                }
+                else {
+                    encoding.scalabilityMode = undefined;
+                }
+                const expectedPattern = `${MediaType.VIDEO}/${codec.toUpperCase()}`;
+                // Configure the codec here if its supported.
+                if (this.usesCodecSelectionAPI() && currentCodec?.mimeType !== expectedPattern) {
+                    const matchingCodec = parameters.codecs.find(pt => pt.mimeType === expectedPattern);
+                    encoding.codec = matchingCodec;
+                    needsUpdate = true;
+                    Statistics.sendAnalytics(AnalyticsEvents.VIDEO_CODEC_CHANGED, {
+                        value: codec,
+                        videoType
+                    });
+                }
+            }
+        }
+        if (!needsUpdate) {
+            return Promise.resolve();
+        }
+        logger.info(`${this} setting max height=${frameHeight},encodings=${JSON.stringify(parameters.encodings)}`);
+        return videoSender.setParameters(parameters).then(() => {
+            localVideoTrack.maxEnabledResolution = frameHeight;
+            this.eventEmitter.emit(RTCEvents.LOCAL_TRACK_MAX_ENABLED_RESOLUTION_CHANGED, localVideoTrack);
+        });
+    }
+    /**
+     * Callback ivoked by {@code this._dtmfSender} when it has finished playing
+     * a single tone.
+     *
+     * @param {Object} event - The tonechange event which indicates what characters
+     * are left to be played for the current tone.
+     * @private
+     * @returns {void}
+     */
+    _onToneChange(event) {
+        // An empty event.tone indicates the current tones have finished playing.
+        // Automatically start playing any queued tones on finish.
+        if (this._dtmfSender && event.tone === '' && this._dtmfTonesQueue.length) {
+            const { tones, duration, interToneGap } = this._dtmfTonesQueue.shift();
+            this._dtmfSender.insertDTMF(tones, duration, interToneGap);
+        }
+    }
+    /**
+     * Internal method to create an SDP offer or answer for the peer connection.
+     * Handles codec preferences, SDP munging for simulcast and RTX, and source information extraction.
+     * @private
+     */
+    _createOfferOrAnswer(isOffer, constraints) {
+        const logName = isOffer ? 'Offer' : 'Answer';
+        this.trace(`create${logName}`, JSON.stringify(constraints, null, ' '));
+        const handleSuccess = (resultSdp, resolveFn, rejectFn) => {
+            try {
+                this.trace(`create${logName}OnSuccess::preTransform`, TraceablePeerConnection.dumpSDP(resultSdp));
+                // Munge local description to add 3 SSRCs for video tracks when spatial scalability is enabled.
+                if (this.isSpatialScalabilityOn() && browser.usesSdpMungingForSimulcast()) {
+                    // eslint-disable-next-line no-param-reassign
+                    resultSdp = this.simulcast.mungeLocalDescription(resultSdp);
+                    this.trace(`create${logName} OnSuccess::postTransform (simulcast)`, TraceablePeerConnection.dumpSDP(resultSdp));
+                }
+                if (!this.options.disableRtx && browser.usesSdpMungingForSimulcast()) {
+                    // eslint-disable-next-line no-param-reassign
+                    resultSdp = new RTCSessionDescription({
+                        sdp: this.rtxModifier.modifyRtxSsrcs(resultSdp?.sdp),
+                        type: resultSdp.type
+                    });
+                    this.trace(`create${logName}`
+                        + 'OnSuccess::postTransform (rtx modifier)', TraceablePeerConnection.dumpSDP(resultSdp));
+                }
+                if (resultSdp?.sdp) {
+                    this._processAndExtractSourceInfo(resultSdp.sdp);
+                }
+                resolveFn(resultSdp);
+            }
+            catch (e) {
+                this.trace(`create${logName}OnError`, e);
+                this.trace(`create${logName}OnError`, TraceablePeerConnection.dumpSDP(resultSdp));
+                logger.error(`${this} create${logName}OnError`, e, TraceablePeerConnection.dumpSDP(resultSdp));
+                rejectFn(e);
+            }
+        };
+        const handleFailure = (err, rejectFn) => {
+            this.trace(`create${logName}OnFailure`, err);
+            rejectFn(err);
+        };
+        // Set the codec preference before creating an offer or answer so that the generated SDP will have
+        // the correct preference order.
+        if (browser.supportsCodecPreferences() && this.codecSettings) {
+            const { codecList, mediaType } = this.codecSettings;
+            const transceivers = this.peerconnection.getTransceivers()
+                .filter(t => t.receiver && t.receiver?.track?.kind === mediaType);
+            let capabilities = RTCRtpReceiver.getCapabilities(mediaType)?.codecs;
+            if (transceivers.length && capabilities) {
+                // Rearrange the codec list as per the preference order.
+                for (const codec of codecList.slice().reverse()) {
+                    // Move the desired codecs (all variations of it as well) to the beginning of the list
+                    /* eslint-disable-next-line arrow-body-style */
+                    capabilities.sort(caps => {
+                        return caps.mimeType.toLowerCase() === `${mediaType}/${codec}` ? -1 : 1;
+                    });
+                }
+                // Disable ulpfec and RED on the p2p peerconnection.
+                if (this.isP2P && mediaType === MediaType.VIDEO) {
+                    capabilities = capabilities
+                        .filter(caps => caps.mimeType.toLowerCase() !== `${MediaType.VIDEO}/${CodecMimeType.ULPFEC}`
+                        && caps.mimeType.toLowerCase() !== `${MediaType.VIDEO}/${CodecMimeType.RED}`);
+                }
+                // Apply codec preference to all the transceivers associated with the given media type.
+                for (const transceiver of transceivers) {
+                    transceiver.setCodecPreferences(capabilities);
+                }
+            }
+        }
+        return new Promise((resolve, reject) => {
+            let oaPromise;
+            if (isOffer) {
+                oaPromise = this.peerconnection.createOffer(constraints);
+            }
+            else {
+                oaPromise = this.peerconnection.createAnswer(constraints);
+            }
+            oaPromise
+                .then(sdp => handleSuccess(sdp, resolve, reject), error => handleFailure(error, reject));
+        });
+    }
+    /**
+     * Extract primary SSRC from given {@link ITPCSSRCInfo} object.
+     * @param {ITPCSSRCInfo} ssrcObj
+     * @return {Nullable<number>} the primary SSRC or <tt>null</tt>
+     */
+    _extractPrimarySSRC(ssrcObj) {
+        if (ssrcObj?.groups?.length) {
+            return ssrcObj.groups[0].ssrcs[0];
+        }
+        else if (ssrcObj?.ssrcs?.length) {
+            return ssrcObj.ssrcs[0];
+        }
+        return null;
+    }
+    /**
+     * Sends a stats entry to the rtcstats server about the mute state change for local track.
+     *
+     * @param {JitsiLocalTrack} track - The local track.
+     * @param {boolean} muted - muted state.
+     */
+    _sendRtcStatsEvent(track, muted) {
+        const mediaType = track.getType();
+        if (mediaType === MediaType.AUDIO) {
+            RTCStats.sendStatsEntry(RTCStatsEvents.AUDIO_MUTE_CHANGED_EVENT, this._pcId, muted);
+        }
+        else if (track.getVideoType() === VideoType.DESKTOP) {
+            RTCStats.sendStatsEntry(RTCStatsEvents.SCREENSHARE_MUTE_CHANGED_EVENT, this._pcId, muted);
+        }
+        else {
+            RTCStats.sendStatsEntry(RTCStatsEvents.VIDEO_MUTE_CHANGED_EVENT, this._pcId, muted);
+        }
+    }
+    /**
+     * Handles remote source mute and unmute changed events.
+     *
+     * @param {string} sourceName - The name of the remote source.
+     * @param {boolean} isMuted - The new mute state.
+     * @internal
+     */
+    _sourceMutedChanged(sourceName, isMuted) {
+        const track = this.getRemoteTracks().slice().reverse().find(t => t.getSourceName() === sourceName);
+        if (!track) {
+            logger.debug(`Remote track not found for source=${sourceName}, mute update failed!`);
+            return;
+        }
+        track.setMute(isMuted);
+    }
+    /* eslint-enable max-params */
+    /**
+     * Process stat and adds it to the array of stats we store.
+     * @param report the current stats report.
+     * @param name the name of the report, if available
+     * @param statValue the value to add.
+     * @private
+     */
+    _processStat(report, name, statValue) {
+        const id = `${report.id}-${name}`;
+        let s = this.stats[id];
+        const now = new Date();
+        if (!s) {
+            this.stats[id] = s = {
+                endTime: now,
+                startTime: now,
+                times: [],
+                values: []
+            };
+        }
+        s.values.push(statValue);
+        s.times.push(now.getTime());
+        if (s.values.length > this.maxstats) {
+            s.values.shift();
+            s.times.shift();
+        }
+        s.endTime = now;
+    }
+    /**
+     * Forwards the {@link peerconnection.iceConnectionState} state except that it
+     * will convert "completed" into "connected" where both mean that the ICE has
+     * succeeded and is up and running. We never see "completed" state for
+     * the JVB connection, but it started appearing for the P2P one. This method
+     * allows to adapt old logic to this new situation.
+     * @return {RTCIceConnectionState}
+     */
+    getConnectionState() {
+        const state = this.peerconnection.iceConnectionState;
+        if (state === 'completed') {
+            return 'connected';
+        }
+        return state;
+    }
+    /**
+     * Obtains the media direction for given {@link MediaType} that needs to be set on a p2p peerconnection's remote SDP
+     * after a source-add or source-remove action. The method takes into account whether or not there are any
+     * local tracks for the given media type.
+     * @param {MediaType} mediaType - The media type for which the direction is to be calculated.
+     * @param {boolean} isAddOperation whether the direction is to be calculated after a source-add action.
+     * @return {string} one of the SDP direction constants ('sendrecv, 'recvonly' etc.)
+     * which should be used when setting
+     * local description on the peerconnection.
+     * @internal
+     */
+    getDesiredMediaDirection(mediaType, isAddOperation = false) {
+        return this.tpcUtils.getDesiredMediaDirection(mediaType, isAddOperation);
+    }
+    /**
+     * Tells whether or not this TPC instance has spatial scalability enabled.
+     * @return {boolean} <tt>true</tt> if spatial scalability is enabled and active or
+     * <tt>false</tt> if it's turned off.
+     */
+    isSpatialScalabilityOn() {
+        const h264SimulcastEnabled = this.tpcUtils.codecSettings[CodecMimeType.H264].scalabilityModeEnabled;
+        return !this.options.disableSimulcast
+            && (this.codecSettings.codecList[0] !== CodecMimeType.H264 || h264SimulcastEnabled);
+    }
+    /**
+     * Obtains audio levels of the remote audio tracks by getting the source information on the RTCRtpReceivers.
+     * The information relevant to the ssrc is updated each time a RTP packet constaining the ssrc is received.
+     * @param {Array<string>} speakerList list of endpoint ids for which audio levels are to be gathered.
+     * @returns {Object} containing ssrc and audio level information as a key-value pair.
+     */
+    getAudioLevels(speakerList = []) {
+        const audioLevels = {};
+        const audioReceivers = speakerList.length
+            ? this._getReceiversByEndpointIds(speakerList, MediaType.AUDIO)
+            : this.peerconnection.getReceivers()
+                .filter(receiver => receiver.track
+                && receiver.track.kind === MediaType.AUDIO && receiver.track.enabled);
+        audioReceivers.forEach(remote => {
+            const ssrc = remote.getSynchronizationSources();
+            if (ssrc?.length) {
+                // As per spec, this audiolevel is a value between 0..1 (linear), where 1.0
+                // represents 0 dBov, 0 represents silence, and 0.5 represents approximately
+                // 6 dBSPL change in the sound pressure level from 0 dBov.
+                // https://www.w3.org/TR/webrtc/#dom-rtcrtpcontributingsource-audiolevel
+                audioLevels[ssrc[0].source] = ssrc[0].audioLevel;
+            }
+        });
+        return audioLevels;
+    }
+    /**
+     * Checks if the browser is currently doing true simulcast where in three
+     * different media streams are being sent to the
+     * bridge. Currently this happens always for VP8 and only if simulcast is enabled for VP9/AV1/H264.
+     *
+     * @param {JitsiLocalTrack} localTrack - The local video track.
+     * @returns {boolean}
+     */
+    doesTrueSimulcast(localTrack) {
+        const currentCodec = this.tpcUtils.getConfiguredVideoCodec(localTrack);
+        return this.isSpatialScalabilityOn() && this.tpcUtils.isRunningInSimulcastMode(currentCodec);
+    }
+    /**
+     * Returns the SSRCs associated with a given local video track.
+     *
+     * @param {JitsiLocalTrack} localTrack
+     * @returns
+     */
+    getLocalVideoSSRCs(localTrack) {
+        const ssrcs = [];
+        if (!localTrack?.isVideoTrack()) {
+            return ssrcs;
+        }
+        const ssrcGroup = this.isSpatialScalabilityOn() ? SSRC_GROUP_SEMANTICS.SIM : SSRC_GROUP_SEMANTICS.FID;
+        return this.localSSRCs.get(localTrack.rtcId)
+            ?.groups?.find(group => group.semantics === ssrcGroup)?.ssrcs || ssrcs;
+    }
+    /**
+     * Obtains local tracks for given {@link MediaType}. If the <tt>mediaType</tt>
+     * argument is omitted the list of all local tracks will be returned.
+     * @param {MediaType} [mediaType]
+     * @return {Array<JitsiLocalTrack>}
+     */
+    getLocalTracks(mediaType = undefined) {
+        let tracks = Array.from(this.localTracks.values());
+        if (mediaType !== undefined) {
+            tracks = tracks.filter(track => track.getType() === mediaType);
+        }
+        return tracks;
+    }
+    /**
+     * Retrieves the local video tracks.
+     *
+     * @returns {Array<JitsiLocalTrack>} - local video tracks.
+     */
+    getLocalVideoTracks() {
+        return this.getLocalTracks(MediaType.VIDEO);
+    }
+    /**
+     * Obtains all remote tracks currently known to this PeerConnection instance.
+     *
+     * @param {optional<string>} [endpointId] - The track owner's identifier (MUC nickname)
+     * @param {optional<MediaType>} [mediaType] - The remote tracks will be filtered by their media type if this argument is
+     * specified.
+     * @return {Array<JitsiRemoteTrack>}
+     */
+    getRemoteTracks(endpointId = undefined, mediaType = undefined) {
+        let remoteTracks = [];
+        if (FeatureFlags.isSsrcRewritingSupported()) {
+            for (const remoteTrack of this.remoteTracksBySsrc.values()) {
+                const owner = remoteTrack.getParticipantId();
+                if (owner && (!endpointId || owner === endpointId)) {
+                    if (!mediaType || remoteTrack.getType() === mediaType) {
+                        remoteTracks.push(remoteTrack);
+                    }
+                }
+            }
+            return remoteTracks;
+        }
+        const endpoints = endpointId ? [endpointId] : this.remoteTracks.keys();
+        for (const endpoint of endpoints) {
+            const endpointTracksByMediaType = this.remoteTracks.get(endpoint);
+            if (endpointTracksByMediaType) {
+                for (const trackMediaType of endpointTracksByMediaType.keys()) {
+                    // per media type filtering
+                    if (!mediaType || mediaType === trackMediaType) {
+                        remoteTracks = remoteTracks.concat(Array.from(endpointTracksByMediaType.get(trackMediaType)));
+                    }
+                }
+            }
+        }
+        return remoteTracks;
+    }
+    /**
+     * Returns the remote sourceInfo for a given source name.
+     *
+     * @param {string} sourceName - The source name.
+     * @returns {TPCSourceInfo}
+     */
+    getRemoteSourceInfoBySourceName(sourceName) {
+        return cloneDeep(this._remoteSsrcMap.get(sourceName));
+    }
+    /**
+     * Returns a map of source names and their associated SSRCs for the remote participant.
+     *
+     * @param {string} id Endpoint id of the remote participant.
+     * @returns {Map<string, TPCSourceInfo>} The map of source names and their associated SSRCs.
+     */
+    getRemoteSourceInfoByParticipant(id) {
+        const removeSsrcInfo = new Map();
+        const remoteTracks = this.getRemoteTracks(id);
+        if (!remoteTracks?.length) {
+            return removeSsrcInfo;
+        }
+        const primarySsrcs = remoteTracks.map(track => track.getSsrc());
+        for (const [sourceName, sourceInfo] of this._remoteSsrcMap) {
+            if (sourceInfo.ssrcList?.some(ssrc => primarySsrcs.includes(Number(ssrc)))) {
+                removeSsrcInfo.set(sourceName, sourceInfo);
+            }
+        }
+        return removeSsrcInfo;
+    }
+    /**
+     * Returns the target bitrates configured for the local video source.
+     *
+     * @param {JitsiLocalTrack} - The local video track.
+     * @returns {Object}
+     */
+    getTargetVideoBitrates(localTrack) {
+        const currentCodec = this.tpcUtils.getConfiguredVideoCodec(localTrack);
+        return this.tpcUtils.codecSettings[currentCodec].maxBitratesVideo;
+    }
+    /**
+     * Tries to find {@link JitsiTrack} for given SSRC number. It will search both local and remote tracks bound to this
+     * instance.
+     * @param {number} ssrc
+     * @return {Nullable<JitsiRemoteTrack | JitsiLocalTrack>}
+     */
+    getTrackBySSRC(ssrc) {
+        if (typeof ssrc !== 'number') {
+            throw new Error(`SSRC ${ssrc} is not a number`);
+        }
+        for (const localTrack of this.localTracks.values()) {
+            const { ssrcs } = this.localSSRCs.get(localTrack.rtcId) ?? { ssrcs: [] };
+            if (ssrcs.find(localSsrc => Number(localSsrc) === ssrc)) {
+                return localTrack;
+            }
+        }
+        if (FeatureFlags.isSsrcRewritingSupported()) {
+            return this.remoteTracksBySsrc.get(ssrc);
+        }
+        for (const remoteTrack of this.getRemoteTracks()) {
+            if (remoteTrack.getSsrc() === ssrc) {
+                return remoteTrack;
+            }
+        }
+        return null;
+    }
+    /**
+     * Tries to find SSRC number for given {@link JitsiTrack} id. It will search
+     * both local and remote tracks bound to this instance.
+     * @param {string} id
+     * @return {Nullable<number>}
+     */
+    getSsrcByTrackId(id) {
+        const findTrackById = track => track.getTrack().id === id;
+        const localTrack = this.getLocalTracks().find(findTrackById);
+        if (localTrack) {
+            return this.getLocalSSRC(localTrack);
+        }
+        const remoteTrack = this.getRemoteTracks().find(findTrackById);
+        if (remoteTrack) {
+            return remoteTrack.getSsrc();
+        }
+        return null;
+    }
+    /**
+     * Called on "track added" and "stream added" PeerConnection events (because we
+     * handle streams on per track basis). Finds the owner and the SSRC for
+     * the track and passes that to ChatRoom for further processing.
+     * @param {MediaStream} stream the WebRTC MediaStream instance which is
+     * the parent of the track
+     * @param {MediaStreamTrack} track the WebRTC MediaStreamTrack added for remote
+     * participant.
+     * @param {RTCRtpTransceiver} transceiver the WebRTC transceiver that is created
+     * for the remote participant in unified plan.
+     */
+    _remoteTrackAdded(stream, track, transceiver = null) {
+        const streamId = stream.id;
+        const mediaType = track.kind;
+        // Do not create remote tracks for 'mixed' JVB SSRCs (used by JVB for RTCP termination).
+        if (!this.isP2P && !RTCUtils.isUserStreamById(streamId)) {
+            return;
+        }
+        logger.info(`${this} Received track event for remote stream[id=${streamId},type=${mediaType}]`);
+        // look up an associated JID for a stream id
+        if (!mediaType) {
+            logger.error(`MediaType undefined for remote track, stream id: ${streamId}, track creation failed!`);
+            return;
+        }
+        const remoteSDP = new SDP(this.remoteDescription?.sdp);
+        let mediaLine;
+        // Find the matching mline using 'mid' or the 'msid' attr of the stream.
+        if (transceiver?.mid) {
+            const mid = transceiver.mid;
+            // @ts-ignore
+            mediaLine = remoteSDP.media.find(mls => SDPUtil.findLine(mls, `a=mid:${mid}`));
+        }
+        else {
+            mediaLine = remoteSDP.media.find(mls => {
+                // @ts-ignore
+                const msid = SDPUtil.findLine(mls, 'a=msid:');
+                return typeof msid === 'string' && streamId === msid.substring(7).split(' ')[0];
+            });
+        }
+        if (!mediaLine) {
+            logger.error(`Matching media line not found in remote SDP for remote stream[id=${streamId},type=${mediaType}],`
+                + 'track creation failed!');
+            return;
+        }
+        // @ts-ignore
+        let ssrcLines = SDPUtil.findLines(mediaLine, 'a=ssrc:');
+        ssrcLines = ssrcLines.filter(line => line.indexOf(`msid:${streamId}`) !== -1);
+        if (!ssrcLines.length) {
+            logger.error(`No SSRC lines found in remote SDP for remote stream[msid=${streamId},type=${mediaType}]`
+                + 'track creation failed!');
+            return;
+        }
+        // FIXME the length of ssrcLines[0] not verified, but it will fail
+        // with global error handler anyway
+        const ssrcStr = ssrcLines[0].substring(7).split(' ')[0];
+        const trackSsrc = Number(ssrcStr);
+        const ownerEndpointId = this._signalingLayer.getSSRCOwner(trackSsrc);
+        if (!isValidNumber(trackSsrc) || trackSsrc < 0) {
+            logger.error(`Invalid SSRC for remote stream[ssrc=${trackSsrc},id=${streamId},type=${mediaType}]`
+                + 'track creation failed!');
+            return;
+        }
+        if (!ownerEndpointId) {
+            logger.error(`No SSRC owner known for remote stream[ssrc=${trackSsrc},id=${streamId},type=${mediaType}]`
+                + 'track creation failed!');
+            return;
+        }
+        const sourceName = this._signalingLayer.getTrackSourceName(trackSsrc);
+        const peerMediaInfo = this._signalingLayer.getPeerMediaInfo(ownerEndpointId, mediaType, sourceName);
+        const trackDetails = {
+            mediaType,
+            muted: peerMediaInfo?.muted ?? true,
+            ssrc: trackSsrc,
+            stream,
+            track,
+            videoType: peerMediaInfo?.videoType
+        };
+        if (this._remoteSsrcMap.has(sourceName) && mediaType === MediaType.VIDEO) {
+            trackDetails.videoType = this._remoteSsrcMap.get(sourceName).videoType;
+        }
+        this._createRemoteTrack(ownerEndpointId, sourceName, trackDetails);
+    }
+    /**
+     * Initializes a new JitsiRemoteTrack instance with the data provided by the signaling layer and SDP.
+     *
+     * @param {string} ownerEndpointId - The owner's endpoint ID (MUC nickname)
+     * @param {String} sourceName - The track's source name
+     * @param {Object} trackDetails - The track's details.
+     * @param {MediaType} trackDetails.mediaType - media type, 'audio' or 'video'.
+     * @param {boolean} trackDetails.muted - The initial muted status.
+     * @param {number} trackDetails.ssrc - The track's main SSRC number.
+     * @param {MediaStream} trackDetails.stream - The WebRTC stream instance.
+     * @param {MediaStreamTrack} trackDetails.track - The WebRTC track instance.
+     * @param {VideoType} trackDetails.videoType - The track's type of the video (if applicable).
+     */
+    _createRemoteTrack(ownerEndpointId, sourceName, trackDetails) {
+        const { mediaType, muted, ssrc, stream, track, videoType } = trackDetails;
+        logger.info(`${this} creating remote track[endpoint=${ownerEndpointId},ssrc=${ssrc},`
+            + `type=${mediaType},sourceName=${sourceName}]`);
+        let remoteTracksMap;
+        let userTracksByMediaType;
+        if (FeatureFlags.isSsrcRewritingSupported()) {
+            const existingTrack = this.remoteTracksBySsrc.get(ssrc);
+            if (existingTrack) {
+                logger.info(`${this} ignored duplicated track event for SSRC[ssrc=${ssrc},type=${mediaType}]`);
+                return;
+            }
+        }
+        else {
+            remoteTracksMap = this.remoteTracks.get(ownerEndpointId);
+            if (!remoteTracksMap) {
+                remoteTracksMap = new Map();
+                remoteTracksMap.set(MediaType.AUDIO, new Set());
+                remoteTracksMap.set(MediaType.VIDEO, new Set());
+                this.remoteTracks.set(ownerEndpointId, remoteTracksMap);
+            }
+            userTracksByMediaType = remoteTracksMap.get(mediaType);
+            if (userTracksByMediaType?.size
+                && Array.from(userTracksByMediaType).find((jitsiTrack) => jitsiTrack.getTrack() === track)) {
+                // Ignore duplicated event which can originate either from 'onStreamAdded' or 'onTrackAdded'.
+                logger.info(`${this} ignored duplicated track event for track[endpoint=${ownerEndpointId},`
+                    + `type=${mediaType}]`);
+                return;
+            }
+        }
+        const remoteTrack = new JitsiRemoteTrack(this.rtc, this.rtc.conference, ownerEndpointId, stream, track, mediaType, videoType, ssrc, muted, this.isP2P, sourceName);
+        if (FeatureFlags.isSsrcRewritingSupported()) {
+            this.remoteTracksBySsrc.set(ssrc, remoteTrack);
+        }
+        else {
+            userTracksByMediaType.add(remoteTrack);
+        }
+        this.eventEmitter.emit(RTCEvents.REMOTE_TRACK_ADDED, remoteTrack, this);
+    }
+    /**
+     * Handles remote media track removal.
+     *
+     * @param {MediaStream} stream - WebRTC MediaStream instance which is the parent of the track.
+     * @param {MediaStreamTrack} track - WebRTC MediaStreamTrack which has been removed from the PeerConnection.
+     * @returns {void}
+     */
+    _remoteTrackRemoved(stream, track) {
+        const streamId = stream.id;
+        const trackId = track?.id;
+        // Ignore stream removed events for JVB "mixed" sources (used for RTCP termination).
+        if (!RTCUtils.isUserStreamById(streamId)) {
+            return;
+        }
+        if (!streamId) {
+            logger.error(`${this} remote track removal failed - no stream ID`);
+            return;
+        }
+        if (!trackId) {
+            logger.error(`${this} remote track removal failed - no track ID`);
+            return;
+        }
+        const toBeRemoved = this.getRemoteTracks().find(remoteTrack => remoteTrack.getStreamId() === streamId && remoteTrack.getTrackId() === trackId);
+        if (!toBeRemoved) {
+            logger.error(`${this} remote track removal failed - track not found`);
+            return;
+        }
+        this._removeRemoteTrack(toBeRemoved);
+    }
+    /**
+     * Removes and disposes given <tt>JitsiRemoteTrack</tt> instance. Emits {@link RTCEvents.REMOTE_TRACK_REMOVED}.
+     *
+     * @param {JitsiRemoteTrack} toBeRemoved - The remote track to be removed.
+     * @returns {void}
+     */
+    _removeRemoteTrack(toBeRemoved) {
+        logger.info(`${this} Removing remote track stream[id=${toBeRemoved.getStreamId()},`
+            + `trackId=${toBeRemoved.getTrackId()}]`);
+        toBeRemoved.dispose();
+        const participantId = toBeRemoved.getParticipantId();
+        if (FeatureFlags.isSsrcRewritingSupported() && !participantId) {
+            return;
+        }
+        else if (!FeatureFlags.isSsrcRewritingSupported()) {
+            const userTracksByMediaType = this.remoteTracks.get(participantId);
+            if (!userTracksByMediaType) {
+                logger.error(`${this} removeRemoteTrack: no remote tracks map for endpoint=${participantId}`);
+            }
+            else if (!userTracksByMediaType.get(toBeRemoved.getType())?.delete(toBeRemoved)) {
+                logger.error(`${this} Failed to remove ${toBeRemoved} - type mapping messed up ?`);
+            }
+        }
+        this.eventEmitter.emit(RTCEvents.REMOTE_TRACK_REMOVED, toBeRemoved);
+    }
+    /**
+     * Processes the local SDP and creates an SSRC map for every local track.
+     *
+     * @param {string} localSDP - SDP from the local description.
+     * @returns {void}
+     */
+    _processAndExtractSourceInfo(localSDP) {
+        /**
+         * @type {Map<string, TPCSourceInfo>} The map of source names and their associated SSRCs.
+         */
+        const ssrcMap = new Map();
+        if (!localSDP || typeof localSDP !== 'string') {
+            throw new Error('Local SDP must be a valid string, aborting!!');
+        }
+        const session = transform.parse(localSDP);
+        const media = session.media.filter(mline => mline.direction === MediaDirection.SENDONLY
+            || mline.direction === MediaDirection.SENDRECV);
+        if (!media.length) {
+            this._localSsrcMap = ssrcMap;
+            return;
+        }
+        for (const localTrack of this.localTracks.values()) {
+            const sourceName = localTrack.getSourceName();
+            const trackIndex = getSourceIndexFromSourceName(sourceName);
+            const mediaType = localTrack.getType();
+            const mLines = media.filter(m => m.type === mediaType);
+            const ssrcGroups = mLines[trackIndex].ssrcGroups;
+            let ssrcs = mLines[trackIndex].ssrcs;
+            if (ssrcs?.length) {
+                // Filter the ssrcs with 'cname' attribute.
+                ssrcs = ssrcs.filter(s => s.attribute === 'cname');
+                const msid = `${this.rtc.getLocalEndpointId()}-${mediaType}-${trackIndex}`;
+                const ssrcInfo = {
+                    groups: [],
+                    msid,
+                    ssrcs: []
+                };
+                ssrcs.forEach(ssrc => ssrcInfo.ssrcs.push(ssrc.id));
+                if (ssrcGroups?.length) {
+                    for (const group of ssrcGroups) {
+                        const parsedGroup = {
+                            semantics: group.semantics,
+                            ssrcs: group.ssrcs.split(' ').map(ssrcStr => parseInt(ssrcStr, 10))
+                        };
+                        ssrcInfo.groups.push(parsedGroup);
+                    }
+                    const simGroup = ssrcGroups.find(group => group.semantics === SSRC_GROUP_SEMANTICS.SIM);
+                    // Add a SIM group if its missing in the description (happens on Firefox).
+                    if (this.isSpatialScalabilityOn() && !simGroup) {
+                        const groupSsrcs = ssrcGroups.map(group => group.ssrcs[0]);
+                        ssrcInfo.groups.push({
+                            semantics: SSRC_GROUP_SEMANTICS.SIM,
+                            ssrcs: groupSsrcs
+                        });
+                    }
+                }
+                ssrcMap.set(sourceName, ssrcInfo);
+                const oldSsrcInfo = this.localSSRCs.get(localTrack.rtcId);
+                const oldSsrc = this._extractPrimarySSRC(oldSsrcInfo);
+                const newSsrc = this._extractPrimarySSRC(ssrcInfo);
+                if (oldSsrc !== newSsrc) {
+                    oldSsrc && logger.debug(`${this} Overwriting SSRC for track=${localTrack}] with ssrc=${newSsrc}`);
+                    this.localSSRCs.set(localTrack.rtcId, ssrcInfo);
+                    localTrack.setSsrc(newSsrc);
+                }
+            }
+        }
+        this._localSsrcMap = ssrcMap;
+    }
+    /**
+     * @param {JitsiLocalTrack} localTrack
+     * @returns {Nullable<number>}
+     */
+    getLocalSSRC(localTrack) {
+        const ssrcInfo = this._getSSRC(localTrack.rtcId.toString());
+        return ssrcInfo?.ssrcs[0];
+    }
+    /**
+     * Gets the signaling state of the peer connection.
+     */
+    get signalingState() {
+        return this.peerconnection.signalingState;
+    }
+    /**
+     * Gets the ICE connection state of the peer connection.
+     */
+    get iceConnectionState() {
+        return this.peerconnection.iceConnectionState;
+    }
+    /**
+     * Gets the connection state of the peer connection.
+     */
+    get connectionState() {
+        return this.peerconnection.connectionState;
+    }
+    /**
+     * Gets the local description of the peer connection, with optional transformations for
+     * simulcast and stream identifiers.
+     */
+    get localDescription() {
+        let desc = this.peerconnection.localDescription;
+        if (!desc) {
+            logger.debug(`${this} getLocalDescription no localDescription found`);
+            // @ts-ignore
+            return {};
+        }
+        this.trace('getLocalDescription::preTransform', TraceablePeerConnection.dumpSDP(desc));
+        if (!this.isP2P) {
+            desc = this.tpcUtils.injectSsrcGroupForSimulcast(desc);
+            this.trace('getLocalDescription::postTransform (inject ssrc group)', TraceablePeerConnection.dumpSDP(desc));
+        }
+        desc = this.localSdpMunger.transformStreamIdentifiers(desc, this._localSsrcMap);
+        return desc;
+    }
+    /**
+     * Gets the remote description of the peer connection, with optional adjustments for media direction in P2P mode.
+     */
+    get remoteDescription() {
+        let desc = this.peerconnection.remoteDescription;
+        if (!desc) {
+            logger.debug(`${this} getRemoteDescription no remoteDescription found`);
+            // @ts-ignore
+            return {};
+        }
+        this.trace('getRemoteDescription::preTransform', TraceablePeerConnection.dumpSDP(desc));
+        if (this.isP2P) {
+            desc = this._adjustRemoteMediaDirection(desc);
+        }
+        return desc;
+    }
+    /**
+     * Retrieves the SSRC (Synchronization Source) identifier associated with the given RTC ID.
+     * @private
+     */
+    _getSSRC(rtcId) {
+        return this.localSSRCs.get(Number(rtcId));
+    }
+    /**
+     * Checks if low fps screensharing is in progress.
+     *
+     * @private
+     * @returns {boolean} Returns true if 5 fps screensharing is in progress, false otherwise.
+     */
+    isSharingLowFpsScreen() {
+        return this._isSharingScreen() && this._capScreenshareBitrate;
+    }
+    /**
+     * Checks if screensharing is in progress.
+     *
+     * @returns {boolean}  Returns true if a desktop track has been added to the peerconnection, false otherwise.
+     */
+    _isSharingScreen() {
+        const tracks = this.getLocalVideoTracks();
+        return Boolean(tracks.find(track => track.videoType === VideoType.DESKTOP));
+    }
+    /**
+     * Adds local track to the RTCPeerConnection.
+     *
+     * @param {JitsiLocalTrack} track the track to be added to the pc.
+     * @return {Promise<boolean>} Promise that resolves to true if the underlying PeerConnection's state has changed and
+     * renegotiation is required, false if no renegotiation is needed or Promise is rejected when something goes wrong.
+     */
+    addTrackToPc(track) {
+        logger.info(`${this} Adding track=${track} to PC`);
+        if (!this._assertTrackBelongs('addTrackToPc', track)) {
+            // Abort
+            return Promise.reject('Track not found on the peerconnection');
+        }
+        const webRtcStream = track.getOriginalStream();
+        if (!webRtcStream) {
+            logger.error(`${this} Unable to add track=${track} to PC - no WebRTC stream`);
+            return Promise.reject('Stream not found');
+        }
+        return this.replaceTrack(null, track, true /* isMuteOperation */).then(() => {
+            if (track) {
+                if (track.isAudioTrack()) {
+                    this._hasHadAudioTrack = true;
+                }
+                else {
+                    this._hasHadVideoTrack = true;
+                }
+            }
+            return false;
+        });
+    }
+    /**
+     * This method when called will check if given <tt>localTrack</tt> belongs to
+     * this TPC (that it has been previously added using {@link addTrack}). If the
+     * track does not belong an error message will be logged.
+     * @param {string} methodName the method name that will be logged in an error
+     * message
+     * @param {JitsiLocalTrack} localTrack
+     * @return {boolean} <tt>true</tt> if given local track belongs to this TPC or
+     * <tt>false</tt> otherwise.
+     * @private
+     */
+    _assertTrackBelongs(methodName, localTrack) {
+        const doesBelong = this.localTracks.has(localTrack?.rtcId);
+        if (!doesBelong) {
+            logger.error(`${this} ${methodName}: track=${localTrack} does not belong to pc`);
+        }
+        return doesBelong;
+    }
+    /**
+     * Returns the codecs in the current order of preference as configured on the peerconnection.
+     *
+     * @param {RTCSessionDescription} - The local description to be used.
+     * @returns {Array}
+     */
+    getConfiguredVideoCodecs(description) {
+        return this.tpcUtils.getConfiguredVideoCodecs(description?.sdp);
+    }
+    /**
+     * Enables or disables simulcast for screenshare based on the frame rate requested for desktop track capture.
+     *
+     * @param {number} maxFps framerate to be used for desktop track capture.
+     */
+    setDesktopSharingFrameRate(maxFps) {
+        const lowFps = maxFps <= SS_DEFAULT_FRAME_RATE;
+        this._capScreenshareBitrate = this.isSpatialScalabilityOn() && lowFps;
+    }
+    /**
+     * Sets the codec preference on the peerconnection. The codec preference goes into effect when
+     * the next renegotiation happens for older clients that do not support the codec selection API.
+     *
+     * @param {Array<CodecMimeType>} codecList - Preferred codecs for video.
+     * @param {CodecMimeType} screenshareCodec - The preferred codec for screenshare.
+     * @returns {boolean} - Returns true if the codec settings were updated, false otherwise.
+     */
+    setVideoCodecs(codecList, screenshareCodec) {
+        let updated = false;
+        if (!this.codecSettings || !codecList?.length) {
+            return updated;
+        }
+        this.codecSettings.codecList = codecList;
+        if (screenshareCodec) {
+            this.codecSettings.screenshareCodec = screenshareCodec;
+        }
+        if (!this.usesCodecSelectionAPI()) {
+            return updated;
+        }
+        for (const track of this.getLocalVideoTracks()) {
+            const currentCodec = this.tpcUtils.getConfiguredVideoCodec(track);
+            if (screenshareCodec && track.getVideoType() === VideoType.DESKTOP && screenshareCodec !== currentCodec) {
+                this.configureVideoSenderEncodings(track, screenshareCodec);
+                updated = true;
+            }
+            else if (currentCodec !== codecList[0]) {
+                this.configureVideoSenderEncodings(track);
+                updated = true;
+            }
+        }
+        updated && RTCStats.sendStatsEntry(RTCStatsEvents.CODEC_CHANGED_EVENT, this._pcId, {
+            camera: codecList[0],
+            screenshare: screenshareCodec
+        });
+        return updated;
+    }
+    /**
+     * Remove local track from this TPC.
+     * @param {JitsiLocalTrack} localTrack the track to be removed from this TPC.
+     *
+     * FIXME It should probably remove a boolean just like {@link removeTrackFromPc}
+     *       The same applies to addTrack.
+     */
+    removeTrack(localTrack) {
+        const webRtcStream = localTrack.getOriginalStream();
+        this.trace('removeStream', localTrack.rtcId.toString(), webRtcStream ? webRtcStream.id : undefined);
+        if (!this._assertTrackBelongs('removeStream', localTrack)) {
+            // Abort - nothing to be done here
+            return;
+        }
+        this.localTracks.delete(localTrack.rtcId);
+        this.localSSRCs.delete(localTrack.rtcId);
+        if (webRtcStream) {
+            // @ts-ignore
+            this.peerconnection.removeStream(webRtcStream);
+        }
+    }
+    /**
+     * Returns the receiver corresponding to the given MediaStreamTrack.
+     *
+     * @param {MediaSreamTrack} track - The media stream track used for the search.
+     * @returns {Optional<RTCRtpReceiver>} - The found receiver or undefined if no receiver
+     * was found.
+     */
+    findReceiverForTrack(track) {
+        return this.peerconnection.getReceivers().find(r => r.track === track);
+    }
+    /**
+     * Returns the sender corresponding to the given MediaStreamTrack.
+     *
+     * @param {MediaSreamTrack} track - The media stream track used for the search.
+     * @returns {Optional<RTCRtpSender>} - The found sender or undefined if no sender
+     * was found.
+     */
+    findSenderForTrack(track) {
+        return this.peerconnection.getSenders().find(s => s.track === track);
+    }
+    /**
+     * Processes the local description SDP and caches the mids of the mlines associated with the given tracks.
+     *
+     * @param {Array<JitsiLocalTrack>} localTracks - local tracks that are added to the peerconnection.
+     * @returns {void}
+     */
+    processLocalSdpForTransceiverInfo(localTracks) {
+        const localSdp = this.localDescription?.sdp;
+        if (!localSdp) {
+            return;
+        }
+        [MediaType.AUDIO, MediaType.VIDEO].forEach(mediaType => {
+            const tracks = localTracks.filter(t => t.getType() === mediaType);
+            const parsedSdp = transform.parse(localSdp);
+            const mLines = parsedSdp.media.filter(mline => mline.type === mediaType);
+            tracks.forEach((track, idx) => {
+                if (!this.localTrackTransceiverMids.has(track.rtcId)) {
+                    this.localTrackTransceiverMids.set(track.rtcId, mLines[idx].mid.toString());
+                }
+            });
+        });
+    }
+    /**
+     * Replaces <tt>oldTrack</tt> with <tt>newTrack</tt> from the peer connection.
+     * Either <tt>oldTrack</tt> or <tt>newTrack</tt> can be null; replacing a valid
+     * <tt>oldTrack</tt> with a null <tt>newTrack</tt> effectively just removes
+     * <tt>oldTrack</tt>
+     *
+     * @param {Nullable<JitsiLocalTrack>} oldTrack - The current track in use to be replaced on the pc.
+     * @param {Nullable<JitsiLocalTrack>} newTrack - The new track to be used.
+     * @param {boolean} isMuteOperation - Whether the operation is a mute/unmute operation.
+     * @returns {Promise<boolean>} - If the promise resolves with true, renegotiation will be needed.
+     * Otherwise no renegotiation is needed.
+     */
+    replaceTrack(oldTrack, newTrack, isMuteOperation = false) {
+        if (!(oldTrack || newTrack)) {
+            logger.info(`${this} replaceTrack called with no new track and no old track`);
+            return Promise.resolve(false);
+        }
+        logger.info(`${this} TPC.replaceTrack old=${oldTrack}, new=${newTrack}`);
+        let transceiver;
+        const mediaType = newTrack?.getType() ?? oldTrack?.getType();
+        const localTracks = this.getLocalTracks(mediaType);
+        const track = newTrack?.getTrack() ?? null;
+        const isNewLocalSource = localTracks?.length
+            && !oldTrack
+            && newTrack
+            && !localTracks.find(t => t === newTrack);
+        // If old track exists, replace the track on the corresponding sender.
+        if (oldTrack && !oldTrack.isMuted()) {
+            transceiver = this.peerconnection.getTransceivers().find(t => t.sender.track === oldTrack.getTrack());
+            // Find the first recvonly transceiver when more than one track of the same media type is being added to the pc.
+            // As part of the track addition, a new m-line was added to the remote description with direction set to
+            // recvonly.
+        }
+        else if (isNewLocalSource) {
+            transceiver = this.peerconnection.getTransceivers().find(t => t.receiver.track.kind === mediaType
+                && t.direction === MediaDirection.RECVONLY
+                // Re-use any existing recvonly transceiver (if available) for p2p case.
+                && ((this.isP2P && t.currentDirection === MediaDirection.RECVONLY)
+                    // @ts-ignore
+                    || (t.currentDirection === MediaDirection.INACTIVE && !t.stopped)));
+            // For mute/unmute operations, find the transceiver based on the track index in the source name if present,
+            // otherwise it is assumed to be the first local track that was added to the peerconnection.
+        }
+        else {
+            transceiver = this.peerconnection.getTransceivers().find(t => t.receiver.track.kind === mediaType);
+            const sourceName = newTrack?.getSourceName() ?? oldTrack?.getSourceName();
+            if (sourceName) {
+                const trackIndex = getSourceIndexFromSourceName(sourceName);
+                if (this.isP2P) {
+                    transceiver = this.peerconnection.getTransceivers()
+                        .filter(t => t.receiver.track.kind === mediaType)[trackIndex];
+                }
+                else if (oldTrack) {
+                    const transceiverMid = this.localTrackTransceiverMids.get(oldTrack.rtcId);
+                    transceiver = this.peerconnection.getTransceivers().find(t => t.mid === transceiverMid);
+                }
+                else if (trackIndex) {
+                    transceiver = this.peerconnection.getTransceivers()
+                        .filter(t => t.receiver.track.kind === mediaType
+                        && t.direction !== MediaDirection.RECVONLY)[trackIndex];
+                }
+            }
+        }
+        if (!transceiver) {
+            return Promise.reject(new Error(`Replace track failed - no transceiver for old: ${oldTrack}, new: ${newTrack}`));
+        }
+        return transceiver.sender.replaceTrack(track)
+            .then(() => {
+            if (isMuteOperation) {
+                // Send RTCStats events for mute operations.
+                this._sendRtcStatsEvent((oldTrack ?? newTrack), (oldTrack && !newTrack));
+                return Promise.resolve();
+            }
+            if (oldTrack) {
+                this.localTracks.delete(oldTrack.rtcId);
+                this.localTrackTransceiverMids.delete(oldTrack.rtcId);
+            }
+            if (newTrack) {
+                if (newTrack.isAudioTrack()) {
+                    this._hasHadAudioTrack = true;
+                }
+                else {
+                    this._hasHadVideoTrack = true;
+                }
+                this.localTrackTransceiverMids.set(newTrack.rtcId, transceiver?.mid?.toString());
+                this.localTracks.set(newTrack.rtcId, newTrack);
+                // Send RTCStats event when the track is added for the first time.
+                this._sendRtcStatsEvent(newTrack, false);
+            }
+            // Update the local SSRC cache for the case when one track gets replaced with another and no
+            // renegotiation is triggered as a result of this.
+            if (oldTrack && newTrack) {
+                const oldTrackSSRC = this.localSSRCs.get(oldTrack.rtcId);
+                if (oldTrackSSRC) {
+                    this.localSSRCs.delete(oldTrack.rtcId);
+                    this.localSSRCs.set(newTrack.rtcId, oldTrackSSRC);
+                    const oldSsrcNum = this._extractPrimarySSRC(oldTrackSSRC);
+                    newTrack.setSsrc(oldSsrcNum);
+                }
+                // When a screenshare is stopped and started again, replace track will be called.
+                if (oldTrack.getVideoType() === VideoType.DESKTOP) {
+                    RTCStats.sendStatsEntry(RTCStatsEvents.SCREENSHARE_MUTE_CHANGED_EVENT, this._pcId, false);
+                }
+            }
+            // In the scenario where we remove the oldTrack (oldTrack is not null and newTrack is null) on FF
+            // if we change the direction to RECVONLY, create answer will generate SDP with only 1 receive
+            // only ssrc instead of keeping all 6 ssrcs that we currently have. Stopping the screen sharing
+            // and then starting it again will trigger 2 rounds of source-remove and source-add replacing
+            // the 6 ssrcs for the screen sharing with 1 receive only ssrc and then removing the receive
+            // only ssrc and adding the same 6 ssrcs. On the remote participant's side the same ssrcs will
+            // be reused on a new m-line and if the remote participant is FF due to
+            // https://bugzilla.mozilla.org/show_bug.cgi?id=1768729 the video stream won't be rendered.
+            // That's why we need keep the direction to SENDRECV for FF.
+            //
+            // NOTE: If we return to the approach of not removing the track for FF and instead using the
+            // enabled property for muting the track, we may need to change the direction to
+            // RECVONLY if FF still sends the media even though the enabled flag is set to false.
+            transceiver.direction
+                = newTrack || browser.isFirefox() ? MediaDirection.SENDRECV : MediaDirection.RECVONLY;
+            // Configure simulcast encodings on Firefox when a track is added to the
+            // peerconnection for the first time.
+            const configureEncodingsPromise = browser.isFirefox() && !oldTrack && newTrack && this.doesTrueSimulcast(newTrack)
+                ? this._setEncodings(newTrack)
+                : Promise.resolve();
+            return configureEncodingsPromise.then(() => this.isP2P);
+        });
+    }
+    /**
+     * Removes local track from the RTCPeerConnection.
+     *
+     * @param {JitsiLocalTrack} localTrack the local track to be removed.
+     * @return {Promise<boolean>} Promise that resolves to true if the underlying PeerConnection's state has changed and
+     * renegotiation is required, false if no renegotiation is needed or Promise is rejected when something goes wrong.
+     */
+    removeTrackFromPc(localTrack) {
+        const webRtcStream = localTrack.getOriginalStream();
+        this.trace('removeTrack', `${localTrack.rtcId} ${webRtcStream ? webRtcStream.id : 'null'}`);
+        if (!this._assertTrackBelongs('removeTrack', localTrack)) {
+            // Abort - nothing to be done here
+            return Promise.reject('Track not found in the peerconnection');
+        }
+        return this.replaceTrack(localTrack, null, true /* isMuteOperation */).then(() => false);
+    }
+    /**
+     * Updates the remote source map with the given source map for adding or removing sources.
+     *
+     * @param {Map<string, TPCSourceInfo>} sourceMap - The map of source names to their corresponding SSRCs.
+     * @param {boolean} isAdd - Whether the sources are being added or removed.
+     * @returns {void}
+     */
+    updateRemoteSources(sourceMap, isAdd) {
+        for (const [sourceName, ssrcInfo] of sourceMap) {
+            if (isAdd) {
+                this._remoteSsrcMap.set(sourceName, ssrcInfo);
+            }
+            else {
+                this._remoteSsrcMap.delete(sourceName);
+            }
+        }
+    }
+    /**
+     * Returns true if the codec selection API is used for switching between codecs for the video sources.
+     *
+     * @returns {boolean}
+     */
+    usesCodecSelectionAPI() {
+        // Browser throws an error when H.264 is set on the encodings. Therefore, munge the SDP when H.264 needs to be
+        // selected.
+        // TODO: Remove this check when the above issue is fixed.
+        return this._usesCodecSelectionAPI && this.codecSettings.codecList[0] !== CodecMimeType.H264;
+    }
+    /**
+     * Creates a new data channel on the peer connection with the specified label and options.
+     */
+    createDataChannel(label, opts) {
+        this.trace('createDataChannel', label, opts);
+        return this.peerconnection.createDataChannel(label, opts);
+    }
+    /**
+     * Returns the expected send resolution for a local video track based on what encodings are currently active.
+     *
+     * @param {JitsiLocalTrack} localTrack - The local video track.
+     * @returns {number}
+     */
+    calculateExpectedSendResolution(localTrack) {
+        const captureResolution = localTrack.getCaptureResolution();
+        let result = Math.min(localTrack.maxEnabledResolution, captureResolution);
+        if (localTrack.getVideoType() === VideoType.CAMERA) {
+            // Find the closest matching resolution based on the current codec, simulcast config and the requested
+            // resolution by the bridge or the peer.
+            if (this.doesTrueSimulcast(localTrack)) {
+                const sender = this.findSenderForTrack(localTrack.getTrack());
+                if (!sender) {
+                    return result;
+                }
+                const { encodings } = sender.getParameters();
+                result = encodings.reduce((maxValue, encoding) => {
+                    if (encoding.active) {
+                        // eslint-disable-next-line no-param-reassign
+                        maxValue = Math.max(maxValue, Math.floor(captureResolution / encoding.scaleResolutionDownBy));
+                    }
+                    return maxValue;
+                }, 0);
+            }
+        }
+        return result;
+    }
+    /**
+     * Configures the stream encodings for the audio tracks that are added to the peerconnection.
+     *
+     * @param {Nullable<JitsiLocalTrack>} localAudioTrack - The local audio track.
+     * @returns {Promise} promise that will be resolved when the operation is successful and rejected otherwise.
+     */
+    configureAudioSenderEncodings(localAudioTrack = null) {
+        if (localAudioTrack) {
+            return this._setEncodings(localAudioTrack);
+        }
+        const promises = [];
+        for (const track of this.getLocalTracks(MediaType.AUDIO)) {
+            promises.push(this._setEncodings(track));
+        }
+        return Promise.allSettled(promises);
+    }
+    /**
+     * Configures the stream encodings depending on the video type,
+     * scalability mode and the bitrate settings for the codec
+     * that is currently selected.
+     *
+     * @param {Nullable<JitsiLocalTrack>} - The local track for which the sender encodings have to configured.
+     * @param {CodecMimeType} - The preferred codec for the video track.
+     * @returns {Promise} promise that will be resolved when the operation is successful and rejected otherwise.
+     */
+    configureVideoSenderEncodings(localVideoTrack = null, codec = null) {
+        const preferredCodec = codec ?? this.codecSettings.codecList[0];
+        if (localVideoTrack) {
+            const height = this._senderMaxHeights.get(localVideoTrack.getSourceName())
+                ?? VIDEO_QUALITY_LEVELS[0].height;
+            return this.setSenderVideoConstraints(height, localVideoTrack, preferredCodec);
+        }
+        const promises = [];
+        for (const track of this.getLocalVideoTracks()) {
+            const maxHeight = this._senderMaxHeights.get(track.getSourceName()) ?? VIDEO_QUALITY_LEVELS[0].height;
+            promises.push(this.setSenderVideoConstraints(maxHeight, track, preferredCodec));
+        }
+        return Promise.allSettled(promises);
+    }
+    /**
+     * Sets the local description on the peerconnection.
+     *
+     * @param {RTCSessionDescription} description - The local description to be set.
+     * @returns {Promise<void>} - Resolved when the operation is successful and rejected with an error otherwise.
+     */
+    setLocalDescription(description) {
+        let localDescription = description;
+        localDescription = this._mungeDescription(localDescription);
+        return new Promise((resolve, reject) => {
+            this.peerconnection.setLocalDescription(localDescription)
+                .then(() => {
+                this.trace('setLocalDescriptionOnSuccess');
+                const localUfrag = SDPUtil.getUfrag(localDescription.sdp);
+                if (localUfrag !== this._localUfrag) {
+                    this._localUfrag = localUfrag;
+                    this.eventEmitter.emit(RTCEvents.LOCAL_UFRAG_CHANGED, this, localUfrag);
+                }
+                this._initializeDtlsTransport();
+                resolve();
+            }, err => {
+                this.trace('setLocalDescriptionOnFailure', err);
+                reject(err);
+            });
+        });
+    }
+    /**
+     * Sets the remote description on the peerconnection.
+     *
+     * @param {RTCSessionDescription} description - The remote description to be set.
+     * @returns {Promise<void>} - Resolved when the operation is successful and rejected with an error otherwise.
+     */
+    setRemoteDescription(description) {
+        let remoteDescription = description;
+        if (this.isSpatialScalabilityOn()) {
+            remoteDescription = this.tpcUtils.insertUnifiedPlanSimulcastReceive(remoteDescription);
+            this.trace('setRemoteDescription::postTransform (sim receive)', TraceablePeerConnection.dumpSDP(remoteDescription));
+        }
+        remoteDescription = this.tpcUtils.ensureCorrectOrderOfSsrcs(remoteDescription);
+        this.trace('setRemoteDescription::postTransform (correct ssrc order)', TraceablePeerConnection.dumpSDP(remoteDescription));
+        remoteDescription = this._mungeDescription(remoteDescription);
+        return new Promise((resolve, reject) => {
+            this.peerconnection.setRemoteDescription(remoteDescription)
+                .then(() => {
+                this.trace('setRemoteDescriptionOnSuccess');
+                const remoteUfrag = SDPUtil.getUfrag(remoteDescription.sdp);
+                if (remoteUfrag !== this._remoteUfrag) {
+                    this._remoteUfrag = remoteUfrag;
+                    this.eventEmitter.emit(RTCEvents.REMOTE_UFRAG_CHANGED, this, remoteUfrag);
+                }
+                this._initializeDtlsTransport();
+                resolve();
+            })
+                .catch(err => {
+                this.trace('setRemoteDescriptionOnFailure', err);
+                reject(err);
+            });
+        });
+    }
+    /**
+     * Changes the resolution of the video stream that is sent to the peer based on the resolution requested by the peer
+     * and user preference, sets the degradation preference on the sender
+     * based on the video type, configures the maximum
+     * bitrates on the send stream.
+     *
+     * @param {number} frameHeight - The max frame height to be imposed on the outgoing video stream.
+     * @param {JitsiLocalTrack} - The local track for which the sender constraints have to be applied.
+     * @param {preferredCodec} - The video codec that needs to be configured on the
+     * sender associated with the video source.
+     * @returns {Promise} promise that will be resolved when the operation is successful and rejected otherwise.
+     */
+    setSenderVideoConstraints(frameHeight, localVideoTrack, preferredCodec) {
+        if (frameHeight < 0 || !isValidNumber(frameHeight)) {
+            throw new Error(`Invalid frameHeight: ${frameHeight}`);
+        }
+        if (!localVideoTrack) {
+            throw new Error('Local video track is missing');
+        }
+        const sourceName = localVideoTrack.getSourceName();
+        this._senderMaxHeights.set(sourceName, frameHeight);
+        // Ignore sender constraints if the video track is muted.
+        if (localVideoTrack.isMuted()) {
+            return Promise.resolve();
+        }
+        const codec = preferredCodec ?? this.codecSettings.codecList[0];
+        return this._updateVideoSenderParameters(() => this._updateVideoSenderEncodings(frameHeight, localVideoTrack, codec));
+    }
+    /**
+     * Resumes or suspends media on the peerconnection by setting the active state on RTCRtpEncodingParameters
+     * associated with all the senders that have a track attached to it.
+     *
+     * @param {boolean} enable - whether outgoing media needs to be enabled or disabled.
+     * @param {string} mediaType - media type, 'audio' or 'video', if neither is passed, all outgoing media will either
+     * be enabled or disabled.
+     * @returns {Promise} - A promise that is resolved when the change is succesful on all the senders, rejected
+     * otherwise.
+     */
+    setMediaTransferActive(enable, mediaType) {
+        logger.info(`${this} ${enable ? 'Resuming' : 'Suspending'} media transfer.`);
+        const senders = this.peerconnection.getSenders()
+            .filter(s => Boolean(s.track) && (!mediaType || s.track.kind === mediaType));
+        const promises = [];
+        for (const sender of senders) {
+            if (sender.track.kind === MediaType.VIDEO) {
+                promises.push(this._updateVideoSenderParameters(() => this._enableSenderEncodings(sender, enable)));
+            }
+            else {
+                promises.push(this._enableSenderEncodings(sender, enable));
+            }
+        }
+        return Promise.allSettled(promises)
+            .then(settledResult => {
+            const errors = settledResult
+                .filter(result => result.status === 'rejected')
+                .map(result => result.reason);
+            if (errors.length) {
+                return Promise.reject(new Error('Failed to change encodings on the RTCRtpSenders'
+                    + `${errors.join(' ')}`));
+            }
+            return Promise.resolve();
+        });
+    }
+    /**
+     * Enables/disables outgoing video media transmission on this peer connection. When disabled the stream encoding's
+     * active state is enabled or disabled to send or stop the media.
+     * @param {boolean} active <tt>true</tt> to enable video media transmission
+     * or <tt>false</tt> to disable. If the value
+     * is not a boolean the call will have no effect.
+     * @return {Promise} A promise that is resolved when the change is succesful, rejected otherwise.
+     * @public
+     */
+    setVideoTransferActive(active) {
+        logger.debug(`${this} video transfer active: ${active}`);
+        const changed = this.videoTransferActive !== active;
+        this.videoTransferActive = active;
+        if (changed) {
+            return this.setMediaTransferActive(active, MediaType.VIDEO);
+        }
+        return Promise.resolve();
+    }
+    /**
+     * Sends DTMF tones if possible.
+     *
+     * @param {string} tones - The DTMF tones string as defined by {@code RTCDTMFSender.insertDTMF}, 'tones' argument.
+     * @param {number} duration - The amount of time in milliseconds that each DTMF should last. It's 200ms by default.
+     * @param {number} interToneGap - The length of time in miliseconds to wait between tones. It's 200ms by default.
+     *
+     * @returns {void}
+     */
+    sendTones(tones, duration = 200, interToneGap = 200) {
+        if (!this._dtmfSender) {
+            if (this.peerconnection.getSenders) {
+                const rtpSender = this.peerconnection.getSenders().find(s => s.dtmf);
+                this._dtmfSender = rtpSender?.dtmf;
+                this._dtmfSender && logger.info(`${this} initialized DTMFSender using getSenders`);
+            }
+            if (!this._dtmfSender) {
+                const localAudioTrack = Array.from(this.localTracks.values()).find(t => t.isAudioTrack());
+                // @ts-ignore
+                if (this.peerconnection.createDTMFSender && localAudioTrack) {
+                    // @ts-ignore
+                    this._dtmfSender = this.peerconnection.createDTMFSender(localAudioTrack.getTrack());
+                }
+                this._dtmfSender && logger.info(`${this} initialized DTMFSender using deprecated createDTMFSender`);
+            }
+            if (this._dtmfSender) {
+                this._dtmfSender.ontonechange = this._onToneChange.bind(this);
+            }
+        }
+        if (this._dtmfSender) {
+            if (this._dtmfSender.toneBuffer) {
+                this._dtmfTonesQueue.push({
+                    duration,
+                    interToneGap,
+                    tones
+                });
+                return;
+            }
+            this._dtmfSender.insertDTMF(tones, duration, interToneGap);
+        }
+        else {
+            logger.warn(`${this} sendTones - failed to select DTMFSender`);
+        }
+    }
+    /**
+     * Closes underlying WebRTC PeerConnection instance and removes all remote
+     * tracks by emitting {@link RTCEvents.REMOTE_TRACK_REMOVED} for each one of
+     * them.
+     */
+    close() {
+        this.trace('stop');
+        // Off SignalingEvents
+        this._signalingLayer.off(SignalingEvents.PEER_MUTED_CHANGED, this._peerMutedChanged);
+        this._signalingLayer.off(SignalingEvents.PEER_VIDEO_TYPE_CHANGED, this._peerVideoTypeChanged);
+        this.peerconnection.removeEventListener('track', this.onTrack);
+        if (FeatureFlags.isSsrcRewritingSupported()) {
+            for (const remoteTrack of this.remoteTracksBySsrc.values()) {
+                this._removeRemoteTrack(remoteTrack);
+            }
+            this.remoteTracksBySsrc.clear();
+        }
+        else {
+            for (const peerTracks of this.remoteTracks.values()) {
+                for (const remoteTracks of peerTracks.values()) {
+                    for (const remoteTrack of remoteTracks) {
+                        this._removeRemoteTrack(remoteTrack);
+                    }
+                }
+            }
+            this.remoteTracks.clear();
+        }
+        this._dtmfSender = null;
+        this._dtmfTonesQueue = [];
+        if (!this.rtc._removePeerConnection(this)) {
+            logger.error(`${this} rtc._removePeerConnection returned false`);
+        }
+        if (this.statsinterval !== null) {
+            window.clearInterval(this.statsinterval);
+            this.statsinterval = null;
+        }
+        logger.info(`${this} Closing peerconnection`);
+        this.peerconnection.close();
+    }
+    /**
+     * Creates an SDP answer for the peer connection based on the provided constraints.
+     */
+    createAnswer(constraints) {
+        return this._createOfferOrAnswer(false /* answer */, constraints);
+    }
+    /**
+     * Creates an SDP offer for the peer connection based on the provided constraints.
+     */
+    createOffer(constraints) {
+        return this._createOfferOrAnswer(true /* offer */, constraints);
+    }
+    /**
+     * Track the SSRCs seen so far.
+     * @param {number} ssrc - SSRC.
+     * @return {boolean} - Whether this is a new SSRC.
+     */
+    addRemoteSsrc(ssrc) {
+        const existing = this.remoteSSRCs.has(ssrc);
+        if (!existing) {
+            this.remoteSSRCs.add(ssrc);
+        }
+        return !existing;
+    }
+    /**
+     * Adds an ICE candidate to the peer connection.
+     */
+    addIceCandidate(candidate) {
+        this.trace('addIceCandidate', JSON.stringify({
+            candidate: candidate.candidate,
+            sdpMLineIndex: candidate.sdpMLineIndex,
+            sdpMid: candidate.sdpMid,
+            usernameFragment: candidate.usernameFragment
+        }, null, ' '));
+        return this.peerconnection.addIceCandidate(candidate);
+    }
+    /**
+     * Obtains call-related stats from the peer connection.
+     *
+     * @returns {Promise<Object>} Promise which resolves with data providing statistics about
+     * the peerconnection.
+     */
+    getStats() {
+        return this.peerconnection.getStats();
+    }
+    /**
+     * Creates a text representation of this <tt>TraceablePeerConnection</tt>
+     * instance.
+     * @return {string}
+     */
+    toString() {
+        return `TPC[id=${this.id},type=${this.isP2P ? 'P2P' : 'JVB'}]`;
+    }
+}
+/**
+ * Returns a string representation of a SessionDescription object.
+ */
+TraceablePeerConnection.dumpSDP = function (description) {
+    if (!description?.sdp) {
+        return '';
+    }
+    return `type: ${description.type}\r\n${description.sdp}`;
+};
+export default TraceablePeerConnection;
+//# sourceMappingURL=TraceablePeerConnection.js.map