@webex/plugin-meetings 3.0.0 → 3.1.0

package/dist/media/MediaConnectionAwaiter.js.map

@@ -0,0 +1 @@
+{"version":3,"names":["_common","require","_internalMediaCore","_loggerProxy","_interopRequireDefault","_constants","MediaConnectionAwaiter","exports","default","_ref","webrtcMediaConnection","_classCallCheck2","_defineProperty2","defer","Defer","retried","onTimeoutCallback","onTimeout","bind","connectionStateCallback","connectionStateListenerCallback","iceGatheringStateCallback","iceGatheringStateListenerCallback","_createClass2","key","value","isConnected","getConnectionState","ConnectionState","Connected","isIceGatheringCompleted","getIceGatheringState","clearCallbacks","off","Event","ICE_GATHERING_STATE_CHANGED","CONNECTION_STATE_CHANGED","LoggerProxy","logger","log","concat","clearTimeout","timer","resolve","iceGatheringState","setTimeout","ICE_AND_DTLS_CONNECTION_TIMEOUT","warn","reject","waitForMediaConnectionConnected","_promise","on","promise"],"sources":["MediaConnectionAwaiter.ts"],"sourcesContent":["import {Defer} from '@webex/common';\nimport {ConnectionState, Event} from '@webex/internal-media-core';\nimport LoggerProxy from '../common/logs/logger-proxy';\nimport {ICE_AND_DTLS_CONNECTION_TIMEOUT} from '../constants';\n\nexport interface MediaConnectionAwaiterProps {\n  webrtcMediaConnection: any;\n}\n\n/**\n * @class MediaConnectionAwaiter\n */\nexport default class MediaConnectionAwaiter {\n  private webrtcMediaConnection: any;\n  private timer: any;\n  private defer: Defer;\n  private retried: boolean;\n  private onTimeoutCallback: () => void;\n  private connectionStateCallback: () => void;\n  private iceGatheringStateCallback: () => void;\n\n  /**\n   * @param {MediaConnectionAwaiterProps} mediaConnectionAwaiterProps\n   */\n  constructor({webrtcMediaConnection}: MediaConnectionAwaiterProps) {\n    this.webrtcMediaConnection = webrtcMediaConnection;\n    this.defer = new Defer();\n    this.retried = false;\n    this.onTimeoutCallback = this.onTimeout.bind(this);\n    this.connectionStateCallback = this.connectionStateListenerCallback.bind(this);\n    this.iceGatheringStateCallback = this.iceGatheringStateListenerCallback.bind(this);\n  }\n\n  /**\n   * Returns true if the connection is connected, false otherwise.\n   *\n   * @returns {boolean}\n   */\n  private isConnected(): boolean {\n    return this.webrtcMediaConnection.getConnectionState() === ConnectionState.Connected;\n  }\n\n  /**\n   * Returns true if the ICE Gathering is completed, false otherwise.\n   *\n   * @returns {boolean}\n   */\n  private isIceGatheringCompleted(): boolean {\n    return this.webrtcMediaConnection.getIceGatheringState() === 'complete';\n  }\n\n  /**\n   * Clears the callbacks.\n   *\n   * @returns {void}\n   */\n  private clearCallbacks(): void {\n    this.webrtcMediaConnection.off(\n      Event.ICE_GATHERING_STATE_CHANGED,\n      this.iceGatheringStateCallback\n    );\n    this.webrtcMediaConnection.off(Event.CONNECTION_STATE_CHANGED, this.connectionStateCallback);\n  }\n\n  /**\n   * Listener for connection state change.\n   *\n   * @returns {void}\n   */\n  connectionStateListenerCallback(): void {\n    LoggerProxy.logger.log(\n      `Media:MediaConnectionAwaiter#connectionStateListenerCallback --> connection state: ${this.webrtcMediaConnection.getConnectionState()}`\n    );\n\n    if (!this.isConnected()) {\n      return;\n    }\n\n    clearTimeout(this.timer);\n\n    this.clearCallbacks();\n\n    this.defer.resolve();\n  }\n\n  /**\n   * Listener for ICE gathering state change.\n   *\n   * @returns {void}\n   */\n  iceGatheringStateListenerCallback(): void {\n    const iceGatheringState = this.webrtcMediaConnection.getIceGatheringState();\n\n    LoggerProxy.logger.log(\n      `Media:MediaConnectionAwaiter#iceGatheringStateListenerCallback --> ICE gathering state change -> ${iceGatheringState}`\n    );\n\n    if (!this.isIceGatheringCompleted()) {\n      return;\n    }\n\n    if (this.isConnected()) {\n      return;\n    }\n\n    clearTimeout(this.timer);\n\n    this.timer = setTimeout(this.onTimeoutCallback, ICE_AND_DTLS_CONNECTION_TIMEOUT);\n  }\n\n  /**\n   * Function called when the timeout is reached.\n   *\n   * @returns {void}\n   */\n  onTimeout(): void {\n    if (this.isConnected()) {\n      this.clearCallbacks();\n\n      this.defer.resolve();\n\n      return;\n    }\n\n    if (!this.isIceGatheringCompleted()) {\n      if (!this.retried) {\n        LoggerProxy.logger.warn(\n          'Media:MediaConnectionAwaiter#onTimeout --> ICE gathering did not complete within the timeout for the first time, retrying once'\n        );\n\n        // retry once if ICE gathering is not completed\n        this.retried = true;\n        clearTimeout(this.timer);\n        this.timer = setTimeout(this.onTimeoutCallback, ICE_AND_DTLS_CONNECTION_TIMEOUT);\n\n        return;\n      }\n\n      LoggerProxy.logger.warn(\n        'Media:MediaConnectionAwaiter#onTimeout --> ICE gathering did not complete within the timeout for the second time, rejecting'\n      );\n    } else {\n      LoggerProxy.logger.warn(\n        'Media:MediaConnectionAwaiter#onTimeout --> ICE gathering completed, but connection state is not connected, rejecting'\n      );\n    }\n\n    this.clearCallbacks();\n\n    this.defer.reject();\n  }\n\n  /**\n   * Waits for the webrtc media connection to be connected.\n   *\n   * @returns {Promise}\n   */\n  waitForMediaConnectionConnected(): Promise<void> {\n    if (this.isConnected()) {\n      return Promise.resolve();\n    }\n\n    this.webrtcMediaConnection.on(Event.CONNECTION_STATE_CHANGED, this.connectionStateCallback);\n\n    this.webrtcMediaConnection.on(\n      Event.ICE_GATHERING_STATE_CHANGED,\n      this.iceGatheringStateCallback\n    );\n\n    this.timer = setTimeout(this.onTimeoutCallback, ICE_AND_DTLS_CONNECTION_TIMEOUT);\n\n    return this.defer.promise;\n  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package/dist/media/index.js

@@ -120,7 +120,10 @@ Media.createMediaConnection = function (isMultistream, debugId, webex, meetingId
     turnServerInfo = options.turnServerInfo,
     bundlePolicy = options.bundlePolicy;
   var iceServers = [];
-  if (turnServerInfo) {
+
+  // we might not have any TURN server if TURN discovery failed or wasn't done or
+  // we might get an empty TURN url if we land on a video mesh node
+  if (turnServerInfo !== null && turnServerInfo !== void 0 && turnServerInfo.url) {
     iceServers.push({
       urls: turnServerInfo.url,
       username: turnServerInfo.username || '',

package/dist/media/index.js.map

@@ -1 +1 @@
-{"version":3,"names":["_internalMediaCore","require","_loggerProxy","_interopRequireDefault","_constants","_config","_config2","_browserDetection","_rtcMetrics","ownKeys","e","r","t","_Object$keys","_Object$getOwnPropertySymbols","o","filter","_Object$getOwnPropertyDescriptor","enumerable","push","apply","_objectSpread","arguments","length","Object","forEach","_defineProperty2","default","_Object$getOwnPropertyDescriptors","_Object$defineProperties","_Object$defineProperty","_BrowserDetection","BrowserDetection","isBrowser","Media","getLocalMedia","options","config","sendAudio","sendVideo","sendShare","sharePreferences","isSharing","getMedia","getDisplayMedia","_promise","resolve","undefined","getDirection","forceSendRecv","receive","send","createMediaConnection","isMultistream","debugId","webex","meetingId","correlationId","_audioStream$outputSt","_videoStream$outputSt","_shareVideoStream$out","_shareAudioStream$out","mediaProperties","remoteQualityLevel","enableRtx","enableExtmap","turnServerInfo","bundlePolicy","iceServers","urls","url","username","credential","password","rtcMetrics","RtcMetrics","MultistreamRoapMediaConnection","data","addMetrics","closeMetrics","sendMetricsInQueue","Error","mediaDirection","audioStream","videoStream","shareVideoStream","shareAudioStream","RoapMediaConnection","skipInactiveTransceivers","requireH264","sdpMunging","convertPort9to0","addContentSlides","bandwidthLimits","audio","StaticConfig","meetings","bandwidth","video","startBitrate","periodicKeyframes","disableExtmap","disableRtx","localTracks","outputStream","getTracks","screenShareVideo","screenShareAudio","direction","receiveAudio","receiveVideo","receiveShare","customResolution","screenResolution","customShareFrameRate","screenFrameRate","hasSharePreferences","hasCustomConstraints","shareConstraints","hasHighFrameRate","highFrameRate","_Config$meetings","Config","resolution","videoShareFrameRate","aspectRatio","cursor","MEDIA_TRACK_CONSTRAINT","CURSOR","AWLAYS","frameRate","height","idealHeight","width","idealWidth","mediaConfig","navigator","mediaDevices","then","stream","getVideoTracks","applyConstraints","getDisplayMediaParams","defaultWidth","ideal","max","maxWidth","defaultHeight","maxHeight","deviceId","facingMode","fake","process","env","NODE_ENV","getUserMedia","catch","err","logPath","LoggerProxy","logger","error","concat","constraint","toggleStream","stopStream","stop","readyState","mediaSetting","audioVideo","localStream","shareStream","_default","exports"],"sources":["index.ts"],"sourcesContent":["/*!\n * Copyright (c) 2015-2020 Cisco Systems, Inc. See LICENSE file.\n */\n/* globals navigator */\n\nimport {\n  RoapMediaConnection,\n  MultistreamRoapMediaConnection,\n  type MultistreamConnectionConfig,\n} from '@webex/internal-media-core';\nimport {\n  LocalStream,\n  LocalCameraStream,\n  LocalDisplayStream,\n  LocalSystemAudioStream,\n  LocalMicrophoneStream,\n} from '@webex/media-helpers';\nimport LoggerProxy from '../common/logs/logger-proxy';\nimport {MEDIA_TRACK_CONSTRAINT} from '../constants';\nimport Config from '../config';\nimport StaticConfig from '../common/config';\nimport BrowserDetection from '../common/browser-detection';\nimport RtcMetrics from '../rtcMetrics';\n\nconst {isBrowser} = BrowserDetection();\n\nexport type BundlePolicy = MultistreamConnectionConfig['bundlePolicy'];\n\n/**\n * MediaDirection\n * @typedef {Object} MediaDirection\n * @property {boolean} sendAudio\n * @property {boolean} receiveAudio\n * @property {boolean} sendVideo\n * @property {boolean} receiveVideo\n * @property {boolean} sendShare\n * @property {boolean} receiveShare\n */\n\n/**\n * SendOptions\n * @typedef {Object} SendOptions\n * @property sendAudio\n * @property sendVideo\n * @property sendShare\n * @property isSharing\n * @property {Object} sharePreferences\n */\n/**\n *\n * @public\n * @export\n * Mimic browser APIs as \"the ultimate browser\".\n * Handles the quirks of each browser.\n * Extends and enhances adapter.js, i.e., the \"media\" file from the web client.\n */\nconst Media: any = {};\n\n/**\n * make a browser call to get the media\n * @param {SendOptions} options\n * @param {Object} config SDK Configuration for meetings plugin\n * @returns {Promise}\n */\nMedia.getLocalMedia = (options: any, config: object) => {\n  const {sendAudio, sendVideo, sendShare, sharePreferences, isSharing} = options;\n\n  if (sendAudio || sendVideo) {\n    return Media.getMedia(sendAudio, sendVideo, config);\n  }\n\n  if (sendShare && !isSharing) {\n    return Media.getDisplayMedia(\n      {\n        sendAudio: false,\n        sendShare: true,\n        sharePreferences,\n      },\n      config\n    );\n  }\n\n  return Promise.resolve(undefined);\n};\n\nMedia.getDirection = (forceSendRecv: boolean, receive: boolean, send: boolean) => {\n  if (!receive && !send) {\n    return 'inactive';\n  }\n\n  if (forceSendRecv || (receive && send)) {\n    return 'sendrecv';\n  }\n\n  if (receive) {\n    return 'recvonly';\n  }\n\n  return 'sendonly';\n};\n\n/**\n * creates a webrtc media connection with provided tracks and mediaDirection configuration\n *\n * @param {boolean} isMultistream\n * @param {string} debugId string useful for debugging (will appear in media connection logs)\n * @param {object} webex main `webex` object.\n * @param {string} meetingId id for the meeting using this connection\n * @param {string} correlationId id used in requests to correlate to this session\n * @param {Object} options\n * @param {Object} [options.mediaProperties] contains mediaDirection and local tracks:\n *                                 audioTrack, videoTrack, shareVideoTrack, and shareAudioTrack\n * @param {string} [options.remoteQualityLevel] LOW|MEDIUM|HIGH applicable only to non-multistream connections\n * @param {boolean} [options.enableRtx] applicable only to non-multistream connections\n * @param {boolean} [options.enableExtmap] applicable only to non-multistream connections\n * @param {Object} [options.turnServerInfo]\n * @param {BundlePolicy} [options.bundlePolicy]\n * @returns {RoapMediaConnection | MultistreamRoapMediaConnection}\n */\nMedia.createMediaConnection = (\n  isMultistream: boolean,\n  debugId: string,\n  webex: object,\n  meetingId: string,\n  correlationId: string,\n  options: {\n    mediaProperties: {\n      mediaDirection?: {\n        receiveAudio: boolean;\n        receiveVideo: boolean;\n        receiveShare: boolean;\n        sendAudio: boolean;\n        sendVideo: boolean;\n        sendShare: boolean;\n      };\n      audioStream?: LocalMicrophoneStream;\n      videoStream?: LocalCameraStream;\n      shareVideoStream?: LocalDisplayStream;\n      shareAudioStream?: LocalSystemAudioStream;\n    };\n    remoteQualityLevel?: 'LOW' | 'MEDIUM' | 'HIGH';\n    enableRtx?: boolean;\n    enableExtmap?: boolean;\n    turnServerInfo?: {\n      url: string;\n      username: string;\n      password: string;\n    };\n    bundlePolicy?: BundlePolicy;\n  }\n) => {\n  const {\n    mediaProperties,\n    remoteQualityLevel,\n    enableRtx,\n    enableExtmap,\n    turnServerInfo,\n    bundlePolicy,\n  } = options;\n\n  const iceServers = [];\n\n  if (turnServerInfo) {\n    iceServers.push({\n      urls: turnServerInfo.url,\n      username: turnServerInfo.username || '',\n      credential: turnServerInfo.password || '',\n    });\n  }\n\n  if (isMultistream) {\n    const config: MultistreamConnectionConfig = {\n      iceServers,\n    };\n\n    if (bundlePolicy) {\n      config.bundlePolicy = bundlePolicy;\n    }\n\n    const rtcMetrics = new RtcMetrics(webex, meetingId, correlationId);\n\n    return new MultistreamRoapMediaConnection(\n      config,\n      meetingId,\n      /* the rtc metrics objects callbacks */\n      (data) => rtcMetrics.addMetrics(data),\n      () => rtcMetrics.closeMetrics(),\n      () => rtcMetrics.sendMetricsInQueue()\n    );\n  }\n\n  if (!mediaProperties) {\n    throw new Error('mediaProperties have to be provided for non-multistream media connections');\n  }\n\n  const {mediaDirection, audioStream, videoStream, shareVideoStream, shareAudioStream} =\n    mediaProperties;\n\n  return new RoapMediaConnection(\n    {\n      iceServers,\n      skipInactiveTransceivers: false,\n      requireH264: true,\n      sdpMunging: {\n        convertPort9to0: false,\n        addContentSlides: true,\n        bandwidthLimits: {\n          audio: StaticConfig.meetings.bandwidth.audio,\n          video: StaticConfig.meetings.bandwidth.video,\n        },\n        startBitrate: StaticConfig.meetings.bandwidth.startBitrate,\n        periodicKeyframes: 20, // it's always been hardcoded in SDK so for now keeping it that way\n        disableExtmap: !enableExtmap,\n        disableRtx: !enableRtx, // see https://bugs.chromium.org/p/chromium/issues/detail?id=1020642 why we might want to remove RTX from SDP\n      },\n    },\n    {\n      // TODO: RoapMediaConnection is not ready to use stream classes yet, so we pass the raw MediaStreamTrack for now SPARK-460530\n      localTracks: {\n        audio: audioStream?.outputStream?.getTracks()[0],\n        video: videoStream?.outputStream?.getTracks()[0],\n        screenShareVideo: shareVideoStream?.outputStream?.getTracks()[0],\n        screenShareAudio: shareAudioStream?.outputStream?.getTracks()[0], // TODO: add type for screenShareAudio in internal-media-core SPARK-446923\n      } as unknown,\n      direction: {\n        audio: Media.getDirection(true, mediaDirection.receiveAudio, mediaDirection.sendAudio),\n        video: Media.getDirection(true, mediaDirection.receiveVideo, mediaDirection.sendVideo),\n        screenShareVideo: Media.getDirection(\n          false,\n          mediaDirection.receiveShare,\n          mediaDirection.sendShare\n        ),\n      },\n      remoteQualityLevel,\n    },\n    debugId\n  );\n};\n\n/**\n * generates share streams\n * @param {Object} options parameter\n * @param {Boolean} options.sendAudio send audio from the display share\n * @param {Boolean} options.sendShare send video from the display share\n * @param {Object} options.sharePreferences\n * @param {MediaTrackConstraints} options.sharePreferences.shareConstraints constraints to apply to video\n *   @see {@link https://developer.mozilla.org/en-US/docs/Web/API/MediaTrackConstraints}\n * @param {Boolean} options.sharePreferences.highFrameRate if shareConstraints isn't provided, set default values based off of this boolean\n * @param {Object} config SDK Configuration for meetings plugin\n * @returns {Promise.<MediaStream>}\n */\nMedia.getDisplayMedia = (\n  options: {\n    sendAudio: boolean;\n    sendShare: boolean;\n    sharePreferences: {\n      shareConstraints: MediaTrackConstraints;\n      highFrameRate: any;\n    };\n  },\n  config: any = {}\n) => {\n  // SDK screen share resolution settings from Webex.init\n  const customResolution = config.screenResolution || {};\n  // user defined screen share frame rate\n  const customShareFrameRate = config.screenFrameRate || null;\n  // user defined share preferences\n  const hasSharePreferences = options.sharePreferences;\n  const hasCustomConstraints = hasSharePreferences && hasSharePreferences.shareConstraints;\n  const hasHighFrameRate = hasSharePreferences && hasSharePreferences.highFrameRate;\n  const {screenResolution, resolution, videoShareFrameRate, screenFrameRate, aspectRatio} =\n    Config.meetings;\n\n  let shareConstraints: any = {\n    cursor: MEDIA_TRACK_CONSTRAINT.CURSOR.AWLAYS,\n    aspectRatio,\n  };\n\n  if (hasCustomConstraints) {\n    shareConstraints = hasSharePreferences.shareConstraints;\n  } else if (hasHighFrameRate) {\n    shareConstraints = {\n      ...shareConstraints,\n      frameRate: videoShareFrameRate,\n      height: resolution.idealHeight,\n      width: resolution.idealWidth,\n      ...config.resolution,\n    };\n  } else {\n    shareConstraints = {\n      ...shareConstraints,\n      frameRate: customShareFrameRate || screenFrameRate,\n      height: customResolution.idealHeight || screenResolution.idealHeight,\n      width: customResolution.idealWidth || screenResolution.idealWidth,\n      ...config.screenResolution,\n    };\n  }\n\n  // chrome and webkit based browsers (edge, safari) automatically adjust everything\n  // and we have noticed higher quality with those browser types\n  // firefox specifically has some issues with resolution and frame rate decision making\n  // so we are making it optional and configurable (with defaults) for firefox\n  // to have higher quality, and for developers to control the values\n  // eventually we may have to add the same functionality to chrome, OR conversely, get to with firefox\n\n  if (isBrowser('firefox')) {\n    const mediaConfig: any = {\n      audio: options.sendAudio,\n      video: options.sendShare,\n    };\n\n    return navigator.mediaDevices\n      .getDisplayMedia({audio: options.sendAudio, video: mediaConfig})\n      .then((stream) => {\n        if (options.sendShare && stream.getVideoTracks().length > 0) {\n          // Firefox has a bug with the spec where changing in the height and width only happens\n          // after we get the inital tracks\n          // https://bugzilla.mozilla.org/show_bug.cgi?id=1321221\n          stream.getVideoTracks()[0].applyConstraints(shareConstraints);\n        }\n\n        return stream;\n      });\n  }\n\n  const getDisplayMediaParams: any = {video: options.sendShare ? shareConstraints : false};\n\n  // safari doesn't support sending screen share audio\n  // https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/getDisplayMedia\n  if (options.sendAudio && isBrowser('safari')) {\n    getDisplayMediaParams.audio = options.sendAudio;\n  }\n\n  return navigator.mediaDevices.getDisplayMedia(getDisplayMediaParams);\n};\n\n/**\n * generates audio and video using constraints (often called after getSupportedDevices)\n * @param {Object|Boolean} audio gum constraints\n * @param {Object|Boolean} video gum constraints\n * @param {Object} config SDK Configuration for meetings plugin\n * @returns {Object} {streams}\n */\nMedia.getMedia = (audio: any | boolean, video: any | boolean, config: any) => {\n  const defaultWidth = {ideal: config.resolution.idealWidth, max: config.resolution.maxWidth};\n  const defaultHeight = {ideal: config.resolution.idealHeight, max: config.resolution.maxHeight};\n  const mediaConfig = {\n    audio,\n    // TODO: Remove temporary workaround once Firefox fixes low constraint issues\n    // eslint-disable-next-line no-nested-ternary\n    video: video\n      ? isBrowser('firefox') && video.width && video.width.max === 320\n        ? {\n            deviceId: video.deviceId ? video.deviceId : undefined,\n            width: 320,\n            height: 180,\n            frameRate: video.frameRate ? video.frameRate : undefined,\n            facingMode: video.facingMode ? video.facingMode : undefined,\n          }\n        : {\n            deviceId: video.deviceId ? video.deviceId : undefined,\n            width: video.width ? video.width : defaultWidth,\n            height: video.height ? video.height : defaultHeight,\n            frameRate: video.frameRate ? video.frameRate : undefined,\n            facingMode: video.facingMode ? video.facingMode : undefined,\n          }\n      : false,\n    fake: process.env.NODE_ENV === 'test', // Special case to get fake media for Firefox browser for testing\n  };\n\n  return navigator.mediaDevices.getUserMedia(mediaConfig).catch((err) => {\n    const logPath = 'Media:index#getMedia --> navigator.mediaDevices.getUserMedia';\n\n    LoggerProxy.logger.error(`${logPath} failed - ${err} (${err.constraint})`);\n    throw err;\n  });\n};\n\n/**\n *\n * Toggle a specific stream\n * noop as of now, does nothing\n * @returns {null}\n */\nMedia.toggleStream = () => {};\n\n/**\n * Stop input stream\n * @param {LocalStream} stream A local stream\n * @returns {null}\n */\nMedia.stopStream = (stream: LocalStream) => {\n  if (!stream) {\n    return Promise.resolve();\n  }\n\n  return Promise.resolve().then(() => {\n    try {\n      stream.stop();\n    } catch (e) {\n      LoggerProxy.logger.error(\n        `Media:index#stopStream --> Unable to stop the stream with ready state => ${stream.readyState}, error: ${e}`\n      );\n    }\n  });\n};\n\n/**\n * generates streams for audio video and share\n * @param {object} mediaSetting parameter\n * @param {Object} mediaSetting.sendAudio sendAudio: {Boolean} sendAudio constraints\n * @param {Object} mediaSetting.sendVideo sendVideo: {Boolean} sendVideo constraints\n * @param {Object} mediaSetting.sendShare sendShare: {Boolean} sendShare constraints\n * @param {Object} mediaSetting.isSharing isSharing: {Boolean} isSharing constraints\n * @param {Object} audioVideo parameter\n * @param {Object} audioVideo.audio {deviceId: {String}}\n * @param {Object} audioVideo.video {deviceId: {String}}\n * @param {Object} sharePreferences parameter\n * @param {Object} sharePreferences.shareConstraints parameter\n * @param {Boolean} sharePreferences.highFrameRate parameter\n * @param {Object} config SDK Config\n * @returns {Array} [localStream, shareStream]\n */\nMedia.getUserMedia = (\n  mediaSetting: {\n    sendAudio: object;\n    sendVideo: object;\n    sendShare: object;\n    isSharing: object;\n  },\n  audioVideo: {\n    audio: object;\n    video: object;\n  },\n  sharePreferences: {\n    shareConstraints: object;\n    highFrameRate: boolean;\n  },\n  config: object\n) =>\n  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See LICENSE file.\n */\n/* globals navigator */\n\nimport {\n  RoapMediaConnection,\n  MultistreamRoapMediaConnection,\n  type MultistreamConnectionConfig,\n} from '@webex/internal-media-core';\nimport {\n  LocalStream,\n  LocalCameraStream,\n  LocalDisplayStream,\n  LocalSystemAudioStream,\n  LocalMicrophoneStream,\n} from '@webex/media-helpers';\nimport LoggerProxy from '../common/logs/logger-proxy';\nimport {MEDIA_TRACK_CONSTRAINT} from '../constants';\nimport Config from '../config';\nimport StaticConfig from '../common/config';\nimport BrowserDetection from '../common/browser-detection';\nimport RtcMetrics from '../rtcMetrics';\n\nconst {isBrowser} = BrowserDetection();\n\nexport type BundlePolicy = MultistreamConnectionConfig['bundlePolicy'];\n\n/**\n * MediaDirection\n * @typedef {Object} MediaDirection\n * @property {boolean} sendAudio\n * @property {boolean} receiveAudio\n * @property {boolean} sendVideo\n * @property {boolean} receiveVideo\n * @property {boolean} sendShare\n * @property {boolean} receiveShare\n */\n\n/**\n * SendOptions\n * @typedef {Object} SendOptions\n * @property sendAudio\n * @property sendVideo\n * @property sendShare\n * @property isSharing\n * @property {Object} sharePreferences\n */\n/**\n *\n * @public\n * @export\n * Mimic browser APIs as \"the ultimate browser\".\n * Handles the quirks of each browser.\n * Extends and enhances adapter.js, i.e., the \"media\" file from the web client.\n */\nconst Media: any = {};\n\n/**\n * make a browser call to get the media\n * @param {SendOptions} options\n * @param {Object} config SDK Configuration for meetings plugin\n * @returns {Promise}\n */\nMedia.getLocalMedia = (options: any, config: object) => {\n  const {sendAudio, sendVideo, sendShare, sharePreferences, isSharing} = options;\n\n  if (sendAudio || sendVideo) {\n    return Media.getMedia(sendAudio, sendVideo, config);\n  }\n\n  if (sendShare && !isSharing) {\n    return Media.getDisplayMedia(\n      {\n        sendAudio: false,\n        sendShare: true,\n        sharePreferences,\n      },\n      config\n    );\n  }\n\n  return Promise.resolve(undefined);\n};\n\nMedia.getDirection = (forceSendRecv: boolean, receive: boolean, send: boolean) => {\n  if (!receive && !send) {\n    return 'inactive';\n  }\n\n  if (forceSendRecv || (receive && send)) {\n    return 'sendrecv';\n  }\n\n  if (receive) {\n    return 'recvonly';\n  }\n\n  return 'sendonly';\n};\n\n/**\n * creates a webrtc media connection with provided tracks and mediaDirection configuration\n *\n * @param {boolean} isMultistream\n * @param {string} debugId string useful for debugging (will appear in media connection logs)\n * @param {object} webex main `webex` object.\n * @param {string} meetingId id for the meeting using this connection\n * @param {string} correlationId id used in requests to correlate to this session\n * @param {Object} options\n * @param {Object} [options.mediaProperties] contains mediaDirection and local tracks:\n *                                 audioTrack, videoTrack, shareVideoTrack, and shareAudioTrack\n * @param {string} [options.remoteQualityLevel] LOW|MEDIUM|HIGH applicable only to non-multistream connections\n * @param {boolean} [options.enableRtx] applicable only to non-multistream connections\n * @param {boolean} [options.enableExtmap] applicable only to non-multistream connections\n * @param {Object} [options.turnServerInfo]\n * @param {BundlePolicy} [options.bundlePolicy]\n * @returns {RoapMediaConnection | MultistreamRoapMediaConnection}\n */\nMedia.createMediaConnection = (\n  isMultistream: boolean,\n  debugId: string,\n  webex: object,\n  meetingId: string,\n  correlationId: string,\n  options: {\n    mediaProperties: {\n      mediaDirection?: {\n        receiveAudio: boolean;\n        receiveVideo: boolean;\n        receiveShare: boolean;\n        sendAudio: boolean;\n        sendVideo: boolean;\n        sendShare: boolean;\n      };\n      audioStream?: LocalMicrophoneStream;\n      videoStream?: LocalCameraStream;\n      shareVideoStream?: LocalDisplayStream;\n      shareAudioStream?: LocalSystemAudioStream;\n    };\n    remoteQualityLevel?: 'LOW' | 'MEDIUM' | 'HIGH';\n    enableRtx?: boolean;\n    enableExtmap?: boolean;\n    turnServerInfo?: {\n      url: string;\n      username: string;\n      password: string;\n    };\n    bundlePolicy?: BundlePolicy;\n  }\n) => {\n  const {\n    mediaProperties,\n    remoteQualityLevel,\n    enableRtx,\n    enableExtmap,\n    turnServerInfo,\n    bundlePolicy,\n  } = options;\n\n  const iceServers = [];\n\n  // we might not have any TURN server if TURN discovery failed or wasn't done or\n  // we might get an empty TURN url if we land on a video mesh node\n  if (turnServerInfo?.url) {\n    iceServers.push({\n      urls: turnServerInfo.url,\n      username: turnServerInfo.username || '',\n      credential: turnServerInfo.password || '',\n    });\n  }\n\n  if (isMultistream) {\n    const config: MultistreamConnectionConfig = {\n      iceServers,\n    };\n\n    if (bundlePolicy) {\n      config.bundlePolicy = bundlePolicy;\n    }\n\n    const rtcMetrics = new RtcMetrics(webex, meetingId, correlationId);\n\n    return new MultistreamRoapMediaConnection(\n      config,\n      meetingId,\n      /* the rtc metrics objects callbacks */\n      (data) => rtcMetrics.addMetrics(data),\n      () => rtcMetrics.closeMetrics(),\n      () => rtcMetrics.sendMetricsInQueue()\n    );\n  }\n\n  if (!mediaProperties) {\n    throw new Error('mediaProperties have to be provided for non-multistream media connections');\n  }\n\n  const {mediaDirection, audioStream, videoStream, shareVideoStream, shareAudioStream} =\n    mediaProperties;\n\n  return new RoapMediaConnection(\n    {\n      iceServers,\n      skipInactiveTransceivers: false,\n      requireH264: true,\n      sdpMunging: {\n        convertPort9to0: false,\n        addContentSlides: true,\n        bandwidthLimits: {\n          audio: StaticConfig.meetings.bandwidth.audio,\n          video: StaticConfig.meetings.bandwidth.video,\n        },\n        startBitrate: StaticConfig.meetings.bandwidth.startBitrate,\n        periodicKeyframes: 20, // it's always been hardcoded in SDK so for now keeping it that way\n        disableExtmap: !enableExtmap,\n        disableRtx: !enableRtx, // see https://bugs.chromium.org/p/chromium/issues/detail?id=1020642 why we might want to remove RTX from SDP\n      },\n    },\n    {\n      // TODO: RoapMediaConnection is not ready to use stream classes yet, so we pass the raw MediaStreamTrack for now SPARK-460530\n      localTracks: {\n        audio: audioStream?.outputStream?.getTracks()[0],\n        video: videoStream?.outputStream?.getTracks()[0],\n        screenShareVideo: shareVideoStream?.outputStream?.getTracks()[0],\n        screenShareAudio: shareAudioStream?.outputStream?.getTracks()[0], // TODO: add type for screenShareAudio in internal-media-core SPARK-446923\n      } as unknown,\n      direction: {\n        audio: Media.getDirection(true, mediaDirection.receiveAudio, mediaDirection.sendAudio),\n        video: Media.getDirection(true, mediaDirection.receiveVideo, mediaDirection.sendVideo),\n        screenShareVideo: Media.getDirection(\n          false,\n          mediaDirection.receiveShare,\n          mediaDirection.sendShare\n        ),\n      },\n      remoteQualityLevel,\n    },\n    debugId\n  );\n};\n\n/**\n * generates share streams\n * @param {Object} options parameter\n * @param {Boolean} options.sendAudio send audio from the display share\n * @param {Boolean} options.sendShare send video from the display share\n * @param {Object} options.sharePreferences\n * @param {MediaTrackConstraints} options.sharePreferences.shareConstraints constraints to apply to video\n *   @see {@link https://developer.mozilla.org/en-US/docs/Web/API/MediaTrackConstraints}\n * @param {Boolean} options.sharePreferences.highFrameRate if shareConstraints isn't provided, set default values based off of this boolean\n * @param {Object} config SDK Configuration for meetings plugin\n * @returns {Promise.<MediaStream>}\n */\nMedia.getDisplayMedia = (\n  options: {\n    sendAudio: boolean;\n    sendShare: boolean;\n    sharePreferences: {\n      shareConstraints: MediaTrackConstraints;\n      highFrameRate: any;\n    };\n  },\n  config: any = {}\n) => {\n  // SDK screen share resolution settings from Webex.init\n  const customResolution = config.screenResolution || {};\n  // user defined screen share frame rate\n  const customShareFrameRate = config.screenFrameRate || null;\n  // user defined share preferences\n  const hasSharePreferences = options.sharePreferences;\n  const hasCustomConstraints = hasSharePreferences && hasSharePreferences.shareConstraints;\n  const hasHighFrameRate = hasSharePreferences && hasSharePreferences.highFrameRate;\n  const {screenResolution, resolution, videoShareFrameRate, screenFrameRate, aspectRatio} =\n    Config.meetings;\n\n  let shareConstraints: any = {\n    cursor: MEDIA_TRACK_CONSTRAINT.CURSOR.AWLAYS,\n    aspectRatio,\n  };\n\n  if (hasCustomConstraints) {\n    shareConstraints = hasSharePreferences.shareConstraints;\n  } else if (hasHighFrameRate) {\n    shareConstraints = {\n      ...shareConstraints,\n      frameRate: videoShareFrameRate,\n      height: resolution.idealHeight,\n      width: resolution.idealWidth,\n      ...config.resolution,\n    };\n  } else {\n    shareConstraints = {\n      ...shareConstraints,\n      frameRate: customShareFrameRate || screenFrameRate,\n      height: customResolution.idealHeight || screenResolution.idealHeight,\n      width: customResolution.idealWidth || screenResolution.idealWidth,\n      ...config.screenResolution,\n    };\n  }\n\n  // chrome and webkit based browsers (edge, safari) automatically adjust everything\n  // and we have noticed higher quality with those browser types\n  // firefox specifically has some issues with resolution and frame rate decision making\n  // so we are making it optional and configurable (with defaults) for firefox\n  // to have higher quality, and for developers to control the values\n  // eventually we may have to add the same functionality to chrome, OR conversely, get to with firefox\n\n  if (isBrowser('firefox')) {\n    const mediaConfig: any = {\n      audio: options.sendAudio,\n      video: options.sendShare,\n    };\n\n    return navigator.mediaDevices\n      .getDisplayMedia({audio: options.sendAudio, video: mediaConfig})\n      .then((stream) => {\n        if (options.sendShare && stream.getVideoTracks().length > 0) {\n          // Firefox has a bug with the spec where changing in the height and width only happens\n          // after we get the inital tracks\n          // https://bugzilla.mozilla.org/show_bug.cgi?id=1321221\n          stream.getVideoTracks()[0].applyConstraints(shareConstraints);\n        }\n\n        return stream;\n      });\n  }\n\n  const getDisplayMediaParams: any = {video: options.sendShare ? shareConstraints : false};\n\n  // safari doesn't support sending screen share audio\n  // https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/getDisplayMedia\n  if (options.sendAudio && isBrowser('safari')) {\n    getDisplayMediaParams.audio = options.sendAudio;\n  }\n\n  return navigator.mediaDevices.getDisplayMedia(getDisplayMediaParams);\n};\n\n/**\n * generates audio and video using constraints (often called after getSupportedDevices)\n * @param {Object|Boolean} audio gum constraints\n * @param {Object|Boolean} video gum constraints\n * @param {Object} config SDK Configuration for meetings plugin\n * @returns {Object} {streams}\n */\nMedia.getMedia = (audio: any | boolean, video: any | boolean, config: any) => {\n  const defaultWidth = {ideal: config.resolution.idealWidth, max: config.resolution.maxWidth};\n  const defaultHeight = {ideal: config.resolution.idealHeight, max: config.resolution.maxHeight};\n  const mediaConfig = {\n    audio,\n    // TODO: Remove temporary workaround once Firefox fixes low constraint issues\n    // eslint-disable-next-line no-nested-ternary\n    video: video\n      ? isBrowser('firefox') && video.width && video.width.max === 320\n        ? {\n            deviceId: video.deviceId ? video.deviceId : undefined,\n            width: 320,\n            height: 180,\n            frameRate: video.frameRate ? video.frameRate : undefined,\n            facingMode: video.facingMode ? video.facingMode : undefined,\n          }\n        : {\n            deviceId: video.deviceId ? video.deviceId : undefined,\n            width: video.width ? video.width : defaultWidth,\n            height: video.height ? video.height : defaultHeight,\n            frameRate: video.frameRate ? video.frameRate : undefined,\n            facingMode: video.facingMode ? video.facingMode : undefined,\n          }\n      : false,\n    fake: process.env.NODE_ENV === 'test', // Special case to get fake media for Firefox browser for testing\n  };\n\n  return navigator.mediaDevices.getUserMedia(mediaConfig).catch((err) => {\n    const logPath = 'Media:index#getMedia --> navigator.mediaDevices.getUserMedia';\n\n    LoggerProxy.logger.error(`${logPath} failed - ${err} (${err.constraint})`);\n    throw err;\n  });\n};\n\n/**\n *\n * Toggle a specific stream\n * noop as of now, does nothing\n * @returns {null}\n */\nMedia.toggleStream = () => {};\n\n/**\n * Stop input stream\n * @param {LocalStream} stream A local stream\n * @returns {null}\n */\nMedia.stopStream = (stream: LocalStream) => {\n  if (!stream) {\n    return Promise.resolve();\n  }\n\n  return Promise.resolve().then(() => {\n    try {\n      stream.stop();\n    } catch (e) {\n      LoggerProxy.logger.error(\n        `Media:index#stopStream --> Unable to stop the stream with ready state => ${stream.readyState}, error: ${e}`\n      );\n    }\n  });\n};\n\n/**\n * generates streams for audio video and share\n * @param {object} mediaSetting parameter\n * @param {Object} mediaSetting.sendAudio sendAudio: {Boolean} sendAudio constraints\n * @param {Object} mediaSetting.sendVideo sendVideo: {Boolean} sendVideo constraints\n * @param {Object} mediaSetting.sendShare sendShare: {Boolean} sendShare constraints\n * @param {Object} mediaSetting.isSharing isSharing: {Boolean} isSharing constraints\n * @param {Object} audioVideo parameter\n * @param {Object} audioVideo.audio {deviceId: {String}}\n * @param {Object} audioVideo.video {deviceId: {String}}\n * @param {Object} sharePreferences parameter\n * @param {Object} sharePreferences.shareConstraints parameter\n * @param {Boolean} sharePreferences.highFrameRate parameter\n * @param {Object} config SDK Config\n * @returns {Array} [localStream, shareStream]\n */\nMedia.getUserMedia = (\n  mediaSetting: {\n    sendAudio: object;\n    sendVideo: object;\n    sendShare: object;\n    isSharing: object;\n  },\n  audioVideo: {\n    audio: object;\n    video: object;\n  },\n  sharePreferences: {\n    shareConstraints: object;\n    highFrameRate: boolean;\n  },\n  config: object\n) =>\n  Media.getLocalMedia(\n    {\n      sendAudio: mediaSetting.sendAudio ? audioVideo.audio || mediaSetting.sendAudio : false,\n      sendVideo: mediaSetting.sendVideo ? audioVideo.video || mediaSetting.sendVideo : false,\n    },\n    config\n  ).then((localStream) =>\n    Media.getLocalMedia(\n      {\n        sendShare: mediaSetting.sendShare,\n        isSharing: mediaSetting.isSharing,\n        sharePreferences,\n      },\n      config\n    ).then((shareStream) => [localStream, shareStream])\n  );\n\nexport default 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package/dist/media/properties.js

@@ -7,15 +7,14 @@ _Object$defineProperty(exports, "__esModule", {
 });
 exports.default = void 0;
 var _regenerator = _interopRequireDefault(require("@babel/runtime-corejs2/regenerator"));
-var _promise = _interopRequireDefault(require("@babel/runtime-corejs2/core-js/promise"));
 var _stringify = _interopRequireDefault(require("@babel/runtime-corejs2/core-js/json/stringify"));
 var _asyncToGenerator2 = _interopRequireDefault(require("@babel/runtime-corejs2/helpers/asyncToGenerator"));
 var _classCallCheck2 = _interopRequireDefault(require("@babel/runtime-corejs2/helpers/classCallCheck"));
 var _createClass2 = _interopRequireDefault(require("@babel/runtime-corejs2/helpers/createClass"));
 var _defineProperty2 = _interopRequireDefault(require("@babel/runtime-corejs2/helpers/defineProperty"));
-var _internalMediaCore = require("@webex/internal-media-core");
 var _constants = require("../constants");
 var _loggerProxy = _interopRequireDefault(require("../common/logs/logger-proxy"));
+var _MediaConnectionAwaiter = _interopRequireDefault(require("./MediaConnectionAwaiter"));
 /**
  * @class MediaProperties
  */
@@ -198,29 +197,10 @@ var MediaProperties = exports.default = /*#__PURE__*/function () {
   }, {
     key: "waitForMediaConnectionConnected",
     value: function waitForMediaConnectionConnected() {
-      var _this = this;
-      var isConnected = function isConnected() {
-        return _this.webrtcMediaConnection.getConnectionState() === _internalMediaCore.ConnectionState.Connected;
-      };
-      if (isConnected()) {
-        return _promise.default.resolve();
-      }
-      return new _promise.default(function (resolve, reject) {
-        var timer;
-        var connectionStateListener = function connectionStateListener() {
-          _loggerProxy.default.logger.log("Media:properties#waitForMediaConnectionConnected --> connection state: ".concat(_this.webrtcMediaConnection.getConnectionState()));
-          if (isConnected()) {
-            clearTimeout(timer);
-            _this.webrtcMediaConnection.off(_internalMediaCore.Event.CONNECTION_STATE_CHANGED, connectionStateListener);
-            resolve();
-          }
-        };
-        timer = setTimeout(function () {
-          _this.webrtcMediaConnection.off(_internalMediaCore.Event.CONNECTION_STATE_CHANGED, connectionStateListener);
-          reject();
-        }, _constants.ICE_AND_DTLS_CONNECTION_TIMEOUT);
-        _this.webrtcMediaConnection.on(_internalMediaCore.Event.CONNECTION_STATE_CHANGED, connectionStateListener);
+      var mediaConnectionAwaiter = new _MediaConnectionAwaiter.default({
+        webrtcMediaConnection: this.webrtcMediaConnection
       });
+      return mediaConnectionAwaiter.waitForMediaConnectionConnected();
     }
 
     /**

package/dist/media/properties.js.map

@@ -1 +1 @@
-{"version":3,"names":["_internalMediaCore","require","_constants","_loggerProxy","_interopRequireDefault","MediaProperties","exports","default","_classCallCheck2","_defineProperty2","MEETINGS","webrtcMediaConnection","mediaDirection","receiveAudio","receiveVideo","receiveShare","sendAudio","sendVideo","sendShare","videoStream","audioStream","shareVideoStream","shareAudioStream","remoteShareStream","undefined","remoteAudioStream","remoteVideoStream","remoteQualityLevel","QUALITY_LEVELS","HIGH","mediaSettings","videoDeviceId","_createClass2","key","value","getVideoDeviceId","setMediaDirection","setMediaSettings","type","values","setMediaPeerConnection","mediaPeerConnection","setLocalVideoStream","setLocalAudioStream","setLocalShareVideoStream","setLocalShareAudioStream","setRemoteQualityLevel","setRemoteShareStream","setRemoteAudioStream","setRemoteVideoStream","setVideoDeviceId","deviceId","unsetPeerConnection","unsetRemoteMedia","unsetRemoteShareStream","unsetRemoteStreams","hasLocalShareStream","waitForMediaConnectionConnected","_this","isConnected","getConnectionState","ConnectionState","Connected","_promise","resolve","reject","timer","connectionStateListener","LoggerProxy","logger","log","concat","clearTimeout","off","Event","CONNECTION_STATE_CHANGED","setTimeout","ICE_AND_DTLS_CONNECTION_TIMEOUT","on","_getCurrentConnectionType","_asyncToGenerator2","_regenerator","mark","_callee","allStatsReports","statsResult","successfulCandidatePairs","foundConnectionType","candidatePairStates","wrap","_callee$","_context","prev","next","getStats","sent","forEach","report","push","t0","warn","filter","_report$state","state","toLowerCase","some","pair","localCandidate","find","id","localCandidateId","connectionType","relayProtocol","toUpperCase","_localCandidate$proto","protocol","candidateType","map","_stringify","abrupt","stop","getCurrentConnectionType","apply","arguments"],"sources":["properties.ts"],"sourcesContent":["import {ConnectionState, Event} from '@webex/internal-media-core';\n\nimport {\n  LocalCameraStream,\n  LocalMicrophoneStream,\n  LocalDisplayStream,\n  LocalSystemAudioStream,\n  RemoteStream,\n} from '@webex/media-helpers';\n\nimport {MEETINGS, ICE_AND_DTLS_CONNECTION_TIMEOUT, QUALITY_LEVELS} from '../constants';\nimport LoggerProxy from '../common/logs/logger-proxy';\n\nexport type MediaDirection = {\n  sendAudio: boolean;\n  sendVideo: boolean;\n  sendShare: boolean;\n  receiveAudio: boolean;\n  receiveVideo: boolean;\n  receiveShare: boolean;\n};\n\n/**\n * @class MediaProperties\n */\nexport default class MediaProperties {\n  audioStream?: LocalMicrophoneStream;\n  mediaDirection: MediaDirection;\n  mediaSettings: any;\n  webrtcMediaConnection: any;\n  remoteAudioStream: RemoteStream;\n  remoteQualityLevel: any;\n  remoteShareStream: RemoteStream;\n  remoteVideoStream: RemoteStream;\n  shareVideoStream?: LocalDisplayStream;\n  shareAudioStream?: LocalSystemAudioStream;\n  videoDeviceId: any;\n  videoStream?: LocalCameraStream;\n  namespace = MEETINGS;\n\n  /**\n   * @param {Object} [options] -- to auto construct\n   * @returns {MediaProperties}\n   */\n  constructor() {\n    this.webrtcMediaConnection = null;\n    this.mediaDirection = {\n      receiveAudio: false,\n      receiveVideo: false,\n      receiveShare: false,\n      sendAudio: false,\n      sendVideo: false,\n      sendShare: false,\n    };\n    this.videoStream = null;\n    this.audioStream = null;\n    this.shareVideoStream = null;\n    this.shareAudioStream = null;\n    this.remoteShareStream = undefined;\n    this.remoteAudioStream = undefined;\n    this.remoteVideoStream = undefined;\n    this.remoteQualityLevel = QUALITY_LEVELS.HIGH;\n    this.mediaSettings = {};\n    this.videoDeviceId = null;\n  }\n\n  /**\n   * Retrieves the preferred video input device\n   * @returns {Object|null}\n   */\n  getVideoDeviceId() {\n    return this.videoDeviceId || null;\n  }\n\n  setMediaDirection(mediaDirection) {\n    this.mediaDirection = mediaDirection;\n  }\n\n  setMediaSettings(type, values) {\n    this.mediaSettings[type] = values;\n  }\n\n  setMediaPeerConnection(mediaPeerConnection) {\n    this.webrtcMediaConnection = mediaPeerConnection;\n  }\n\n  setLocalVideoStream(videoStream?: LocalCameraStream) {\n    this.videoStream = videoStream;\n  }\n\n  setLocalAudioStream(audioStream?: LocalMicrophoneStream) {\n    this.audioStream = audioStream;\n  }\n\n  setLocalShareVideoStream(shareVideoStream?: LocalDisplayStream) {\n    this.shareVideoStream = shareVideoStream;\n  }\n\n  setLocalShareAudioStream(shareAudioStream?: LocalSystemAudioStream) {\n    this.shareAudioStream = shareAudioStream;\n  }\n\n  setRemoteQualityLevel(remoteQualityLevel) {\n    this.remoteQualityLevel = remoteQualityLevel;\n  }\n\n  setRemoteShareStream(remoteShareStream: RemoteStream) {\n    this.remoteShareStream = remoteShareStream;\n  }\n\n  /**\n   * Sets the remote audio stream\n   * @param {RemoteStream} remoteAudioStream RemoteStream to save\n   * @returns {void}\n   */\n  setRemoteAudioStream(remoteAudioStream: RemoteStream) {\n    this.remoteAudioStream = remoteAudioStream;\n  }\n\n  /**\n   * Sets the remote video stream\n   * @param {RemoteStream} remoteVideoStream RemoteStream to save\n   * @returns {void}\n   */\n  setRemoteVideoStream(remoteVideoStream: RemoteStream) {\n    this.remoteVideoStream = remoteVideoStream;\n  }\n\n  /**\n   * Stores the preferred video input device\n   * @param {string} deviceId Preferred video input device\n   * @returns {void}\n   */\n  setVideoDeviceId(deviceId: string) {\n    this.videoDeviceId = deviceId;\n  }\n\n  unsetPeerConnection() {\n    this.webrtcMediaConnection = null;\n  }\n\n  /**\n   * Removes both remote audio and video from class instance\n   * @returns {void}\n   */\n  unsetRemoteMedia() {\n    this.remoteAudioStream = null;\n    this.remoteVideoStream = null;\n  }\n\n  unsetRemoteShareStream() {\n    this.remoteShareStream = null;\n  }\n\n  /**\n   * Unsets all remote streams\n   * @returns {void}\n   */\n  unsetRemoteStreams() {\n    this.unsetRemoteMedia();\n    this.unsetRemoteShareStream();\n  }\n\n  /**\n   * Returns if we have at least one local share stream or not.\n   * @returns {Boolean}\n   */\n  hasLocalShareStream() {\n    return !!(this.shareAudioStream || this.shareVideoStream);\n  }\n\n  /**\n   * Waits for the webrtc media connection to be connected.\n   *\n   * @returns {Promise<void>}\n   */\n  waitForMediaConnectionConnected(): Promise<void> {\n    const isConnected = () =>\n      this.webrtcMediaConnection.getConnectionState() === ConnectionState.Connected;\n\n    if (isConnected()) {\n      return Promise.resolve();\n    }\n\n    return new Promise<void>((resolve, reject) => {\n      let timer;\n\n      const connectionStateListener = () => {\n        LoggerProxy.logger.log(\n          `Media:properties#waitForMediaConnectionConnected --> connection state: ${this.webrtcMediaConnection.getConnectionState()}`\n        );\n\n        if (isConnected()) {\n          clearTimeout(timer);\n          this.webrtcMediaConnection.off(Event.CONNECTION_STATE_CHANGED, connectionStateListener);\n          resolve();\n        }\n      };\n\n      timer = setTimeout(() => {\n        this.webrtcMediaConnection.off(Event.CONNECTION_STATE_CHANGED, connectionStateListener);\n        reject();\n      }, ICE_AND_DTLS_CONNECTION_TIMEOUT);\n\n      this.webrtcMediaConnection.on(Event.CONNECTION_STATE_CHANGED, connectionStateListener);\n    });\n  }\n\n  /**\n   * Returns the type of a connection that has been established\n   *\n   * @returns {Promise<'UDP' | 'TCP' | 'TURN-TLS' | 'TURN-TCP' | 'TURN-UDP' | 'unknown'>}\n   */\n  async getCurrentConnectionType() {\n    // we can only get the connection type after ICE connection has been established\n    await this.waitForMediaConnectionConnected();\n\n    const allStatsReports = [];\n\n    try {\n      const statsResult = await this.webrtcMediaConnection.getStats();\n      statsResult.forEach((report) => allStatsReports.push(report));\n    } catch (error) {\n      LoggerProxy.logger.warn(\n        `Media:properties#getCurrentConnectionType --> getStats() failed: ${error}`\n      );\n    }\n\n    const successfulCandidatePairs = allStatsReports.filter(\n      (report) => report.type === 'candidate-pair' && report.state?.toLowerCase() === 'succeeded'\n    );\n\n    let foundConnectionType = 'unknown';\n\n    // all of the successful pairs should have the same connection type, so just return the type for the first one\n    successfulCandidatePairs.some((pair) => {\n      const localCandidate = allStatsReports.find(\n        (report) => report.type === 'local-candidate' && report.id === pair.localCandidateId\n      );\n\n      if (localCandidate === undefined) {\n        LoggerProxy.logger.warn(\n          `Media:properties#getCurrentConnectionType --> failed to find local candidate 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{\n  LocalCameraStream,\n  LocalMicrophoneStream,\n  LocalDisplayStream,\n  LocalSystemAudioStream,\n  RemoteStream,\n} from '@webex/media-helpers';\n\nimport {MEETINGS, QUALITY_LEVELS} from '../constants';\nimport LoggerProxy from '../common/logs/logger-proxy';\nimport MediaConnectionAwaiter from './MediaConnectionAwaiter';\n\nexport type MediaDirection = {\n  sendAudio: boolean;\n  sendVideo: boolean;\n  sendShare: boolean;\n  receiveAudio: boolean;\n  receiveVideo: boolean;\n  receiveShare: boolean;\n};\n\n/**\n * @class MediaProperties\n */\nexport default class MediaProperties {\n  audioStream?: LocalMicrophoneStream;\n  mediaDirection: MediaDirection;\n  mediaSettings: any;\n  webrtcMediaConnection: any;\n  remoteAudioStream: RemoteStream;\n  remoteQualityLevel: any;\n  remoteShareStream: RemoteStream;\n  remoteVideoStream: RemoteStream;\n  shareVideoStream?: LocalDisplayStream;\n  shareAudioStream?: LocalSystemAudioStream;\n  videoDeviceId: any;\n  videoStream?: LocalCameraStream;\n  namespace = MEETINGS;\n\n  /**\n   * @param {Object} [options] -- to auto construct\n   * @returns {MediaProperties}\n   */\n  constructor() {\n    this.webrtcMediaConnection = null;\n    this.mediaDirection = {\n      receiveAudio: false,\n      receiveVideo: false,\n      receiveShare: false,\n      sendAudio: false,\n      sendVideo: false,\n      sendShare: false,\n    };\n    this.videoStream = null;\n    this.audioStream = null;\n    this.shareVideoStream = null;\n    this.shareAudioStream = null;\n    this.remoteShareStream = undefined;\n    this.remoteAudioStream = undefined;\n    this.remoteVideoStream = undefined;\n    this.remoteQualityLevel = QUALITY_LEVELS.HIGH;\n    this.mediaSettings = {};\n    this.videoDeviceId = null;\n  }\n\n  /**\n   * Retrieves the preferred video input device\n   * @returns {Object|null}\n   */\n  getVideoDeviceId() {\n    return this.videoDeviceId || null;\n  }\n\n  setMediaDirection(mediaDirection) {\n    this.mediaDirection = mediaDirection;\n  }\n\n  setMediaSettings(type, values) {\n    this.mediaSettings[type] = values;\n  }\n\n  setMediaPeerConnection(mediaPeerConnection) {\n    this.webrtcMediaConnection = mediaPeerConnection;\n  }\n\n  setLocalVideoStream(videoStream?: LocalCameraStream) {\n    this.videoStream = videoStream;\n  }\n\n  setLocalAudioStream(audioStream?: LocalMicrophoneStream) {\n    this.audioStream = audioStream;\n  }\n\n  setLocalShareVideoStream(shareVideoStream?: LocalDisplayStream) {\n    this.shareVideoStream = shareVideoStream;\n  }\n\n  setLocalShareAudioStream(shareAudioStream?: LocalSystemAudioStream) {\n    this.shareAudioStream = shareAudioStream;\n  }\n\n  setRemoteQualityLevel(remoteQualityLevel) {\n    this.remoteQualityLevel = remoteQualityLevel;\n  }\n\n  setRemoteShareStream(remoteShareStream: RemoteStream) {\n    this.remoteShareStream = remoteShareStream;\n  }\n\n  /**\n   * Sets the remote audio stream\n   * @param {RemoteStream} remoteAudioStream RemoteStream to save\n   * @returns {void}\n   */\n  setRemoteAudioStream(remoteAudioStream: RemoteStream) {\n    this.remoteAudioStream = remoteAudioStream;\n  }\n\n  /**\n   * Sets the remote video stream\n   * @param {RemoteStream} remoteVideoStream RemoteStream to save\n   * @returns {void}\n   */\n  setRemoteVideoStream(remoteVideoStream: RemoteStream) {\n    this.remoteVideoStream = remoteVideoStream;\n  }\n\n  /**\n   * Stores the preferred video input device\n   * @param {string} deviceId Preferred video input device\n   * @returns {void}\n   */\n  setVideoDeviceId(deviceId: string) {\n    this.videoDeviceId = deviceId;\n  }\n\n  unsetPeerConnection() {\n    this.webrtcMediaConnection = null;\n  }\n\n  /**\n   * Removes both remote audio and video from class instance\n   * @returns {void}\n   */\n  unsetRemoteMedia() {\n    this.remoteAudioStream = null;\n    this.remoteVideoStream = null;\n  }\n\n  unsetRemoteShareStream() {\n    this.remoteShareStream = null;\n  }\n\n  /**\n   * Unsets all remote streams\n   * @returns {void}\n   */\n  unsetRemoteStreams() {\n    this.unsetRemoteMedia();\n    this.unsetRemoteShareStream();\n  }\n\n  /**\n   * Returns if we have at least one local share stream or not.\n   * @returns {Boolean}\n   */\n  hasLocalShareStream() {\n    return !!(this.shareAudioStream || this.shareVideoStream);\n  }\n\n  /**\n   * Waits for the webrtc media connection to be connected.\n   *\n   * @returns {Promise<void>}\n   */\n  waitForMediaConnectionConnected(): Promise<void> {\n    const mediaConnectionAwaiter = new MediaConnectionAwaiter({\n      webrtcMediaConnection: this.webrtcMediaConnection,\n    });\n\n    return mediaConnectionAwaiter.waitForMediaConnectionConnected();\n  }\n\n  /**\n   * Returns the type of a connection that has been established\n   *\n   * @returns {Promise<'UDP' | 'TCP' | 'TURN-TLS' | 'TURN-TCP' | 'TURN-UDP' | 'unknown'>}\n   */\n  async getCurrentConnectionType() {\n    // we can only get the connection type after ICE connection has been established\n    await this.waitForMediaConnectionConnected();\n\n    const allStatsReports = [];\n\n    try {\n      const statsResult = await this.webrtcMediaConnection.getStats();\n      statsResult.forEach((report) => allStatsReports.push(report));\n    } catch (error) {\n      LoggerProxy.logger.warn(\n        `Media:properties#getCurrentConnectionType --> getStats() failed: ${error}`\n      );\n    }\n\n    const successfulCandidatePairs = allStatsReports.filter(\n      (report) => report.type === 'candidate-pair' && report.state?.toLowerCase() === 'succeeded'\n    );\n\n    let foundConnectionType = 'unknown';\n\n    // all of the successful pairs should have the same connection type, so just return the type for the first one\n    successfulCandidatePairs.some((pair) => {\n      const localCandidate = allStatsReports.find(\n        (report) => report.type === 'local-candidate' && report.id === pair.localCandidateId\n      );\n\n      if (localCandidate === undefined) {\n        LoggerProxy.logger.warn(\n          `Media:properties#getCurrentConnectionType --> failed to find local candidate \"${pair.localCandidateId}\" in getStats() results`\n        );\n\n        return false;\n      }\n\n      let connectionType;\n\n      if (localCandidate.relayProtocol) {\n        connectionType = `TURN-${localCandidate.relayProtocol.toUpperCase()}`;\n      } else {\n        connectionType = localCandidate.protocol?.toUpperCase(); // it will be UDP or TCP\n      }\n\n      if (connectionType) {\n        foundConnectionType = connectionType;\n\n        return true;\n      }\n      LoggerProxy.logger.warn(\n        `Media:properties#getCurrentConnectionType --> missing localCandidate.protocol, candidateType=${localCandidate.candidateType}`\n      );\n\n      return false;\n    });\n\n    if (foundConnectionType === 'unknown') {\n      const candidatePairStates = allStatsReports\n        .filter((report) => report.type === 'candidate-pair')\n        .map((report) => report.state);\n\n      LoggerProxy.logger.warn(\n        `Media:properties#getCurrentConnectionType --> all candidate pair states: ${JSON.stringify(\n          candidatePairStates\n        )}`\n      );\n    }\n\n    return foundConnectionType;\n  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package/dist/meeting/index.d.ts

@@ -1,10 +1,12 @@
 /// <reference types="node" />
 import { StatelessWebexPlugin } from '@webex/webex-core';
 import { ClientEvent, ClientEventLeaveReason } from '@webex/internal-plugin-metrics';
+import { ClientEvent as RawClientEvent } from '@webex/event-dictionary-ts';
+import { MediaType } from '@webex/internal-media-core';
 import { LocalStream, LocalCameraStream, LocalDisplayStream, LocalSystemAudioStream, LocalMicrophoneStream } from '@webex/media-helpers';
 import { StatsAnalyzer } from '../statsAnalyzer';
 import NetworkQualityMonitor from '../networkQualityMonitor';
-import Roap from '../roap/index';
+import Roap, { type TurnServerInfo, type TurnDiscoverySkipReason } from '../roap/index';
 import { type BundlePolicy } from '../media';
 import MediaProperties from '../media/properties';
 import ReconnectionManager from '../reconnection-manager';
@@ -434,8 +436,8 @@ export default class Meeting extends StatelessWebexPlugin {
     resourceUrl: string;
     selfId: string;
     state: any;
-    localAudioStreamMuteStateHandler: (muted: boolean) => void;
-    localVideoStreamMuteStateHandler: (muted: boolean) => void;
+    localAudioStreamMuteStateHandler: () => void;
+    localVideoStreamMuteStateHandler: () => void;
     localOutputTrackChangeHandler: () => void;
     roles: any[];
     environment: string;
@@ -443,7 +445,7 @@ export default class Meeting extends StatelessWebexPlugin {
     allowMediaInLobby: boolean;
     localShareInstanceId: string;
     remoteShareInstanceId: string;
-    turnDiscoverySkippedReason: string;
+    turnDiscoverySkippedReason: TurnDiscoverySkipReason;
     turnServerUsed: boolean;
     areVoiceaEventsSetup: boolean;
     voiceaListenerCallbacks: object;
@@ -1326,6 +1328,14 @@ export default class Meeting extends StatelessWebexPlugin {
      * @returns {Promise<void>}
      */
     private handleWaitForMediaConnectionConnectedError;
+    /**
+     * Performs TURN discovery as a separate call to the Locus /media API
+     *
+     * @param {boolean} isRetry
+     * @param {boolean} isForced
+     * @returns {Promise}
+     */
+    private doTurnDiscovery;
     /**
      * Does TURN discovery, SDP offer/answer exhange, establishes ICE connection and DTLS handshake.
      *
@@ -1333,6 +1343,7 @@ export default class Meeting extends StatelessWebexPlugin {
      * @param {RemoteMediaManagerConfiguration} [remoteMediaManagerConfig]
      * @param {BundlePolicy} [bundlePolicy]
      * @param {boolean} [isForced] - let isForced be true to do turn discovery regardless of reachability results
+     * @param {TurnServerInfo} [turnServerInfo]
      * @returns {Promise<void>}
      */
     private establishMediaConnection;
@@ -1355,11 +1366,12 @@ export default class Meeting extends StatelessWebexPlugin {
      * Creates a media connection to the server. Media connection is required for sending or receiving any audio/video.
      *
      * @param {AddMediaOptions} options
+     * @param {TurnServerInfo} turnServerInfo - TURN server information (used only internally by the SDK)
      * @returns {Promise<void>}
      * @public
      * @memberof Meeting
      */
-    addMedia(options?: AddMediaOptions): Promise<void>;
+    addMedia(options?: AddMediaOptions, turnServerInfo?: TurnServerInfo): Promise<void>;
     /**
      * Informs if the peer connection is in a state that can be updated with updateMedia (audio/video/share)
      * @returns {Boolean}
@@ -1681,9 +1693,9 @@ export default class Meeting extends StatelessWebexPlugin {
     private handleMediaLogging;
     /**
      *
-     * @returns {string} one of 'attendee','host','cohost', returns the user type of the current user
+     * @returns {string} one of 'panelist', 'attendee', 'host', 'cohost', returns the user type of the current user
      */
-    getCurUserType(): "host" | "cohost" | "presenter" | "attendee";
+    getCurUserType(): RawClientEvent['userType'] | null;
     /**
      * End the current meeting for all
      * @returns {Promise}
@@ -1749,6 +1761,13 @@ export default class Meeting extends StatelessWebexPlugin {
      * @returns {Promise<void>}
      */
     private updateTranscodedMediaConnection;
+    /**
+     * set sending named media group which the audio should send to
+     * @param {MediaType} mediaType of the stream
+     * @param {number} languageCode of the stream
+     * @returns {void}
+     */
+    setSendNamedMediaGroup(mediaType: MediaType, languageCode?: number): void;
     /**
      * Publishes a stream.
      *