@webex/plugin-meetings 3.12.0-next.92 → 3.12.0-next.93

package/dist/aiEnableRequest/index.js

@@ -191,7 +191,7 @@ var AIEnableRequest = _webexCore.WebexPlugin.extend({
       method: _constants.HTTP_VERBS.PUT
     });
   },
-  version: "3.12.0-next.92"
+  version: "3.12.0-next.93"
 });
 var _default = exports.default = AIEnableRequest;
 //# sourceMappingURL=index.js.map

package/dist/breakouts/breakout.js

@@ -214,7 +214,7 @@ var Breakout = _webexCore.WebexPlugin.extend({
       sessionId: this.sessionId
     });
   },
-  version: "3.12.0-next.92"
+  version: "3.12.0-next.93"
 });
 var _default = exports.default = Breakout;
 //# sourceMappingURL=breakout.js.map

package/dist/breakouts/index.js

@@ -1133,7 +1133,7 @@ var Breakouts = _webexCore.WebexPlugin.extend({
       this.trigger(_constants.BREAKOUTS.EVENTS.ASK_RETURN_TO_MAIN);
     }
   },
-  version: "3.12.0-next.92"
+  version: "3.12.0-next.93"
 });
 var _default = exports.default = Breakouts;
 //# sourceMappingURL=index.js.map

package/dist/common/errors/webex-errors.js

@@ -6,8 +6,9 @@ var _interopRequireDefault = require("@babel/runtime-corejs2/helpers/interopRequ
 _Object$defineProperty(exports, "__esModule", {
   value: true
 });
-exports.UserNotJoinedError = exports.UserInLobbyError = exports.SpaceIDDeprecatedError = exports.SdpResponseTimeoutError = exports.NoMediaEstablishedYetError = exports.MeetingNotActiveError = exports.IceGatheringFailed = exports.AddMediaFailed = void 0;
+exports.UserNotJoinedError = exports.UserInLobbyError = exports.SpaceIDDeprecatedError = exports.SdpResponseTimeoutError = exports.NoMediaEstablishedYetError = exports.MeetingNotActiveError = exports.MediaConnectionTimedOutError = exports.IceGatheringFailed = exports.AddMediaFailed = void 0;
 exports.createMeetingsError = createMeetingsError;
+var _wrapNativeSuper2 = _interopRequireDefault(require("@babel/runtime-corejs2/helpers/wrapNativeSuper"));
 var _createClass2 = _interopRequireDefault(require("@babel/runtime-corejs2/helpers/createClass"));
 var _classCallCheck2 = _interopRequireDefault(require("@babel/runtime-corejs2/helpers/classCallCheck"));
 var _possibleConstructorReturn2 = _interopRequireDefault(require("@babel/runtime-corejs2/helpers/possibleConstructorReturn"));
@@ -158,22 +159,63 @@ WebExMeetingsErrors[IceGatheringFailed.CODE] = IceGatheringFailed;
 var AddMediaFailed = exports.AddMediaFailed = /*#__PURE__*/function (_WebexMeetingsError7) {
   /**
    * Creates a new AddMediaFailed error
-   * @param {Error} [cause] - The underlying error that caused the media addition to fail
+   * @param {Object} options
+   * @param {Error} [options.cause] - The underlying error that caused the media addition to fail
+   * @param {string} [options.connectionType] - The ICE connection type at the time of failure (e.g. 'UDP', 'TURN-TLS')
+   * @param {number} [options.selectedCandidatePairChanges] - Number of times the selected candidate pair changed
+   * @param {number} [options.numTransports] - Number of ICE transports
+   * @param {boolean} [options.iceConnected] - Whether ICE connection was established before the failure
    */
-  function AddMediaFailed(cause) {
+  function AddMediaFailed() {
     var _this;
+    var options = arguments.length > 0 && arguments[0] !== undefined ? arguments[0] : {};
     (0, _classCallCheck2.default)(this, AddMediaFailed);
     _this = _callSuper(this, AddMediaFailed, [AddMediaFailed.CODE, 'Failed to add media']);
     (0, _defineProperty2.default)(_this, "cause", void 0);
-    _this.cause = cause;
+    (0, _defineProperty2.default)(_this, "connectionType", void 0);
+    (0, _defineProperty2.default)(_this, "selectedCandidatePairChanges", void 0);
+    (0, _defineProperty2.default)(_this, "numTransports", void 0);
+    (0, _defineProperty2.default)(_this, "iceConnected", void 0);
+    _this.cause = options.cause;
+    _this.connectionType = options.connectionType;
+    _this.selectedCandidatePairChanges = options.selectedCandidatePairChanges;
+    _this.numTransports = options.numTransports;
+    _this.iceConnected = options.iceConnected;
     return _this;
   }
+
+  /**
+   * Returns true if the failure was due to a DTLS handshake failure
+   * (ICE connected successfully but the overall media connection failed).
+   */
   (0, _inherits2.default)(AddMediaFailed, _WebexMeetingsError7);
-  return (0, _createClass2.default)(AddMediaFailed);
+  return (0, _createClass2.default)(AddMediaFailed, [{
+    key: "isDtlsHandshakeFailure",
+    get: function get() {
+      return this.iceConnected === true;
+    }
+  }]);
 }(_webexMeetingsError.default);
 (0, _defineProperty2.default)(AddMediaFailed, "CODE", 30203);
 WebExMeetingsErrors[AddMediaFailed.CODE] = AddMediaFailed;
 
+/**
+ * @class MediaConnectionTimedOutError
+ * @classdesc Internal error thrown when the media connection times out waiting to connect.
+ */
+var MediaConnectionTimedOutError = exports.MediaConnectionTimedOutError = /*#__PURE__*/function (_Error) {
+  function MediaConnectionTimedOutError(message, iceConnected) {
+    var _this2;
+    (0, _classCallCheck2.default)(this, MediaConnectionTimedOutError);
+    _this2 = _callSuper(this, MediaConnectionTimedOutError, [message]);
+    (0, _defineProperty2.default)(_this2, "iceConnected", void 0);
+    _this2.name = 'MediaConnectionTimedOutError';
+    _this2.iceConnected = iceConnected;
+    return _this2;
+  }
+  (0, _inherits2.default)(MediaConnectionTimedOutError, _Error);
+  return (0, _createClass2.default)(MediaConnectionTimedOutError);
+}(/*#__PURE__*/(0, _wrapNativeSuper2.default)(Error));
 /**
  * @class SdpResponseTimeoutError
  * @classdesc Raised whenever we timeout waiting for remote SDP answer

package/dist/common/errors/webex-errors.js.map

@@ -1 +1 @@
-{"version":3,"names":["_webexMeetingsError","_interopRequireDefault","require","_callSuper","t","o","e","_getPrototypeOf2","default","_possibleConstructorReturn2","_isNativeReflectConstruct","_Reflect$construct","constructor","apply","Boolean","prototype","valueOf","call","WebExMeetingsErrors","createMeetingsError","code","message","WebexMeetingsError","MeetingNotActiveError","exports","_WebexMeetingsError","_classCallCheck2","CODE","_inherits2","_createClass2","_defineProperty2","UserNotJoinedError","_WebexMeetingsError2","NoMediaEstablishedYetError","_WebexMeetingsError3","UserInLobbyError","_WebexMeetingsError4","SpaceIDDeprecatedError","_WebexMeetingsError5","IceGatheringFailed","_WebexMeetingsError6","AddMediaFailed","_WebexMeetingsError7","cause","_this","SdpResponseTimeoutError","_WebexMeetingsError8"],"sources":["webex-errors.ts"],"sourcesContent":["// The document would auto generate the doc for errors\n\n// 1) Error codes for Oauth, login, mercury should be separated out (Code range)\n// 2) Errors from the actual locus server or other server in case, we can use the same error code as locus and redirect it\n// 3) Any errors we generate from the SDK can be split into various categories\n// Parameter validation, user action, connection, media specific(They can have different range)\n\n// TODO: all the errors from the server need to be captured\n\n// add a way to log and send metrics if needed\nimport WebexMeetingsError from './webex-meetings-error';\n\nconst WebExMeetingsErrors = {};\n\n/**\n * Create a {@link WebexMeetingsError} for a given code and message.\n * @private\n * @param {number} [code] - Error code\n * @param {string} [message] - Error message\n * @returns {WebexMeetingsError}\n */\nexport function createMeetingsError(code?: number, message?: string) {\n  code = typeof code === 'number' ? code : 0;\n  message = typeof message === 'string' && message ? message : 'Unknown error';\n\n  return WebExMeetingsErrors[code]\n    ? new WebExMeetingsErrors[code]()\n    : new WebexMeetingsError(code, message);\n}\n\n/**\n * @class MeetingNotActiveError\n * @classdesc Raised whenever Meeting has already ended and user tries to do a action.\n * @extends WebexMeetingsError\n * @property {number} code - 30101\n * @property {string} message - 'Meeting has already Ended or not Active'\n */\nclass MeetingNotActiveError extends WebexMeetingsError {\n  static CODE = 30101;\n\n  constructor() {\n    super(MeetingNotActiveError.CODE, 'Meeting has already Ended or not Active');\n  }\n}\n\nexport {MeetingNotActiveError};\nWebExMeetingsErrors[MeetingNotActiveError.CODE] = MeetingNotActiveError;\n\n/**\n * @class UserNotJoinedError\n * @classdesc Raised whenever the user has already left the meeting and user tries to do a action.\n * @extends WebexMeetingsError\n * @property {number} code - 30102\n * @property {string} message - 'User has already left the meeting'\n */\nclass UserNotJoinedError extends WebexMeetingsError {\n  static CODE = 30102;\n\n  constructor() {\n    super(\n      UserNotJoinedError.CODE,\n      'User currently not in meeting. Please join a meeting before adding media.'\n    );\n  }\n}\n\nexport {UserNotJoinedError};\nWebExMeetingsErrors[UserNotJoinedError.CODE] = UserNotJoinedError;\n\n/**\n * @class NoMediaEstablishedYetError\n * @classdesc Raised whenever the user has not established media yet.\n * @extends WebexMeetingsError\n * @property {number} code - 30103\n * @property {string} message - error message\n */\nclass NoMediaEstablishedYetError extends WebexMeetingsError {\n  static CODE = 30103;\n\n  // eslint-disable-next-line require-jsdoc\n  constructor() {\n    super(\n      NoMediaEstablishedYetError.CODE,\n      'Webrtc media connection is missing, call addMedia() first'\n    );\n  }\n}\n\nexport {NoMediaEstablishedYetError};\nWebExMeetingsErrors[NoMediaEstablishedYetError.CODE] = NoMediaEstablishedYetError;\n\n/**\n * @class UserInLobbyError\n * @classdesc Raised whenever the user is in lobby and not joined yet.\n * @extends WebexMeetingsError\n * @property {number} code - 30104\n * @property {string} message - 'user is still in the lobby or not joined'\n */\nclass UserInLobbyError extends WebexMeetingsError {\n  static CODE = 30104;\n\n  constructor() {\n    super(UserInLobbyError.CODE, 'user is still in the lobby or not joined');\n  }\n}\n\nexport {UserInLobbyError};\nWebExMeetingsErrors[UserInLobbyError.CODE] = UserInLobbyError;\n\n/**\n * @class SpaceIDDeprecatedError\n * @classdesc Raised whenever the user passes Space ID as destination for create meeting.\n * @extends WebexMeetingsError\n * @property {number} code - 30105\n * @property {string} message - Using the space ID as a destination is no longer supported. Please refer to the [migration guide](https://github.com/webex/webex-js-sdk/wiki/Migration-to-Unified-Space-Meetings) to migrate to use the meeting ID or SIP address.'\n */\nclass SpaceIDDeprecatedError extends WebexMeetingsError {\n  static CODE = 30105;\n\n  constructor() {\n    super(\n      SpaceIDDeprecatedError.CODE,\n      'Using the space ID as a destination is no longer supported. Please refer to the [migration guide](https://github.com/webex/webex-js-sdk/wiki/Migration-to-Unified-Space-Meetings) to migrate to use the meeting ID or SIP address.'\n    );\n  }\n}\n\nexport {SpaceIDDeprecatedError};\nWebExMeetingsErrors[SpaceIDDeprecatedError.CODE] = SpaceIDDeprecatedError;\n\n/**\n * @class IceGatheringFailed\n * @classdesc Raised whenever ice gathering fails.\n * @extends WebexMeetingsError\n * @property {number} code - 30202\n * @property {string} message - 'user failed ice gathering check network/firewall'\n */\nclass IceGatheringFailed extends WebexMeetingsError {\n  static CODE = 30202;\n\n  constructor() {\n    super(IceGatheringFailed.CODE, 'iceConnection: gethering ice candidate failed');\n  }\n}\n\nexport {IceGatheringFailed};\nWebExMeetingsErrors[IceGatheringFailed.CODE] = IceGatheringFailed;\n\n/**\n * @class AddMediaFailed\n * @classdesc Raised whenever we fail to successfully add media to a meeting\n * @extends WebexMeetingsError\n * @property {number} code - 30203\n * @property {string} message - 'Failed to add media'\n * @property {Error} [cause] - The underlying error that caused the failure\n */\nclass AddMediaFailed extends WebexMeetingsError {\n  static CODE = 30203;\n  cause?: Error;\n\n  /**\n   * Creates a new AddMediaFailed error\n   * @param {Error} [cause] - The underlying error that caused the media addition to fail\n   */\n  constructor(cause?: Error) {\n    super(AddMediaFailed.CODE, 'Failed to add media');\n    this.cause = cause;\n  }\n}\nexport {AddMediaFailed};\nWebExMeetingsErrors[AddMediaFailed.CODE] = AddMediaFailed;\n\n/**\n * @class SdpResponseTimeoutError\n * @classdesc Raised whenever we timeout waiting for remote SDP answer\n * @extends WebexMeetingsError\n * @property {number} code - 30204\n * @property {string} message - 'Timed out waiting for REMOTE SDP ANSWER'\n */\nclass SdpResponseTimeoutError extends WebexMeetingsError {\n  static CODE = 30204;\n\n  // eslint-disable-next-line require-jsdoc\n  constructor() {\n    super(SdpResponseTimeoutError.CODE, 'Timed out waiting for REMOTE SDP ANSWER');\n  }\n}\n\nexport {SdpResponseTimeoutError};\nWebExMeetingsErrors[SdpResponseTimeoutError.CODE] = 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Please join a meeting before adding media.'\n    );\n  }\n}\n\nexport {UserNotJoinedError};\nWebExMeetingsErrors[UserNotJoinedError.CODE] = UserNotJoinedError;\n\n/**\n * @class NoMediaEstablishedYetError\n * @classdesc Raised whenever the user has not established media yet.\n * @extends WebexMeetingsError\n * @property {number} code - 30103\n * @property {string} message - error message\n */\nclass NoMediaEstablishedYetError extends WebexMeetingsError {\n  static CODE = 30103;\n\n  // eslint-disable-next-line require-jsdoc\n  constructor() {\n    super(\n      NoMediaEstablishedYetError.CODE,\n      'Webrtc media connection is missing, call addMedia() first'\n    );\n  }\n}\n\nexport {NoMediaEstablishedYetError};\nWebExMeetingsErrors[NoMediaEstablishedYetError.CODE] = NoMediaEstablishedYetError;\n\n/**\n * @class UserInLobbyError\n * @classdesc Raised whenever the user is in lobby and not joined yet.\n * @extends WebexMeetingsError\n * @property {number} code - 30104\n * @property {string} message - 'user is still in the lobby or not joined'\n */\nclass UserInLobbyError extends WebexMeetingsError {\n  static CODE = 30104;\n\n  constructor() {\n    super(UserInLobbyError.CODE, 'user is still in the lobby or not joined');\n  }\n}\n\nexport {UserInLobbyError};\nWebExMeetingsErrors[UserInLobbyError.CODE] = UserInLobbyError;\n\n/**\n * @class SpaceIDDeprecatedError\n * @classdesc Raised whenever the user passes Space ID as destination for create meeting.\n * @extends WebexMeetingsError\n * @property {number} code - 30105\n * @property {string} message - Using the space ID as a destination is no longer supported. Please refer to the [migration guide](https://github.com/webex/webex-js-sdk/wiki/Migration-to-Unified-Space-Meetings) to migrate to use the meeting ID or SIP address.'\n */\nclass SpaceIDDeprecatedError extends WebexMeetingsError {\n  static CODE = 30105;\n\n  constructor() {\n    super(\n      SpaceIDDeprecatedError.CODE,\n      'Using the space ID as a destination is no longer supported. Please refer to the [migration guide](https://github.com/webex/webex-js-sdk/wiki/Migration-to-Unified-Space-Meetings) to migrate to use the meeting ID or SIP address.'\n    );\n  }\n}\n\nexport {SpaceIDDeprecatedError};\nWebExMeetingsErrors[SpaceIDDeprecatedError.CODE] = SpaceIDDeprecatedError;\n\n/**\n * @class IceGatheringFailed\n * @classdesc Raised whenever ice gathering fails.\n * @extends WebexMeetingsError\n * @property {number} code - 30202\n * @property {string} message - 'user failed ice gathering check network/firewall'\n */\nclass IceGatheringFailed extends WebexMeetingsError {\n  static CODE = 30202;\n\n  constructor() {\n    super(IceGatheringFailed.CODE, 'iceConnection: gethering ice candidate failed');\n  }\n}\n\nexport {IceGatheringFailed};\nWebExMeetingsErrors[IceGatheringFailed.CODE] = IceGatheringFailed;\n\n/**\n * @class AddMediaFailed\n * @classdesc Raised whenever we fail to successfully add media to a meeting\n * @extends WebexMeetingsError\n * @property {number} code - 30203\n * @property {string} message - 'Failed to add media'\n * @property {Error} [cause] - The underlying error that caused the failure\n */\nclass AddMediaFailed extends WebexMeetingsError {\n  static CODE = 30203;\n  cause?: Error;\n  connectionType?: string;\n  selectedCandidatePairChanges?: number;\n  numTransports?: number;\n  iceConnected?: boolean;\n\n  /**\n   * Creates a new AddMediaFailed error\n   * @param {Object} options\n   * @param {Error} [options.cause] - The underlying error that caused the media addition to fail\n   * @param {string} [options.connectionType] - The ICE connection type at the time of failure (e.g. 'UDP', 'TURN-TLS')\n   * @param {number} [options.selectedCandidatePairChanges] - Number of times the selected candidate pair changed\n   * @param {number} [options.numTransports] - Number of ICE transports\n   * @param {boolean} [options.iceConnected] - Whether ICE connection was established before the failure\n   */\n  constructor(\n    options: {\n      cause?: Error;\n      connectionType?: string;\n      selectedCandidatePairChanges?: number;\n      numTransports?: number;\n      iceConnected?: boolean;\n    } = {}\n  ) {\n    super(AddMediaFailed.CODE, 'Failed to add media');\n    this.cause = options.cause;\n    this.connectionType = options.connectionType;\n    this.selectedCandidatePairChanges = options.selectedCandidatePairChanges;\n    this.numTransports = options.numTransports;\n    this.iceConnected = options.iceConnected;\n  }\n\n  /**\n   * Returns true if the failure was due to a DTLS handshake failure\n   * (ICE connected successfully but the overall media connection failed).\n   */\n  get isDtlsHandshakeFailure(): boolean {\n    return this.iceConnected === true;\n  }\n}\nexport {AddMediaFailed};\nWebExMeetingsErrors[AddMediaFailed.CODE] = AddMediaFailed;\n\n/**\n * @class MediaConnectionTimedOutError\n * @classdesc Internal error thrown when the media connection times out waiting to connect.\n 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package/dist/interpretation/index.js

@@ -381,7 +381,7 @@ var SimultaneousInterpretation = _webexCore.WebexPlugin.extend({
       throw error;
     });
   },
-  version: "3.12.0-next.92"
+  version: "3.12.0-next.93"
 });
 var _default = exports.default = SimultaneousInterpretation;
 //# sourceMappingURL=index.js.map

package/dist/interpretation/siLanguage.js

@@ -18,7 +18,7 @@ var SILanguage = _webexCore.WebexPlugin.extend({
     languageCode: 'number',
     languageName: 'string'
   },
-  version: "3.12.0-next.92"
+  version: "3.12.0-next.93"
 });
 var _default = exports.default = SILanguage;
 //# sourceMappingURL=siLanguage.js.map

package/dist/media/MediaConnectionAwaiter.js

@@ -73,6 +73,18 @@ var MediaConnectionAwaiter = exports.default = /*#__PURE__*/function () {
       return this.webrtcMediaConnection.getConnectionState() === _internalMediaCore.ConnectionState.Failed;
     }
 
+    /**
+     * Returns true if ICE connection state indicates connectivity.
+     *
+     * @returns {boolean}
+     */
+  }, {
+    key: "isIceConnected",
+    value: function isIceConnected() {
+      var state = this.webrtcMediaConnection.getIceConnectionState();
+      return state === 'connected' || state === 'completed';
+    }
+
     /**
      * Returns true if the ICE Gathering is completed, false otherwise.
      *
@@ -145,7 +157,7 @@ var MediaConnectionAwaiter = exports.default = /*#__PURE__*/function () {
     value: function iceConnectionStateHandler() {
       var iceConnectionState = this.webrtcMediaConnection.getIceConnectionState();
       _loggerProxy.default.logger.log("Media:MediaConnectionAwaiter#iceConnectionStateHandler --> ICE connection state change -> ".concat(iceConnectionState));
-      if (iceConnectionState === 'connected' && !this.iceConnected) {
+      if (this.isIceConnected() && !this.iceConnected) {
         this.iceConnected = true;
       }
       this.connectionStateChange();
@@ -253,7 +265,7 @@ var MediaConnectionAwaiter = exports.default = /*#__PURE__*/function () {
     /**
      * Waits for the webrtc media connection to be connected.
      *
-     * @returns {Promise}
+     * @returns {Promise<void>} In case of failure, the promise is rejected with FailureResult
      */
   }, {
     key: "waitForMediaConnectionConnected",
@@ -263,6 +275,7 @@ var MediaConnectionAwaiter = exports.default = /*#__PURE__*/function () {
         return _promise.default.resolve();
       }
       _loggerProxy.default.logger.log('Media:MediaConnectionAwaiter#waitForMediaConnectionConnected --> Waiting for media connection to be connected');
+      this.iceConnected = this.isIceConnected();
       this.webrtcMediaConnection.on(_internalMediaCore.MediaConnectionEventNames.PEER_CONNECTION_STATE_CHANGED, this.peerConnectionStateCallback);
       this.webrtcMediaConnection.on(_internalMediaCore.MediaConnectionEventNames.ICE_CONNECTION_STATE_CHANGED, this.iceConnectionStateCallback);
       this.webrtcMediaConnection.on(_internalMediaCore.MediaConnectionEventNames.ICE_GATHERING_STATE_CHANGED, this.iceGatheringStateCallback);

package/dist/media/MediaConnectionAwaiter.js.map

@@ -1 +1 @@
-{"version":3,"names":["_common","require","_internalMediaCore","_loggerProxy","_interopRequireDefault","_constants","_constants2","_metrics","MediaConnectionAwaiter","exports","default","_ref","webrtcMediaConnection","correlationId","_classCallCheck2","_defineProperty2","defer","Defer","retried","iceConnected","onTimeoutCallback","onTimeout","bind","peerConnectionStateCallback","peerConnectionStateHandler","iceConnectionStateCallback","iceConnectionStateHandler","iceGatheringStateCallback","iceGatheringStateHandler","_createClass2","key","value","isConnected","getConnectionState","ConnectionState","Connected","isFailed","Failed","isIceGatheringCompleted","getIceGatheringState","clearCallbacks","off","MediaConnectionEventNames","ICE_GATHERING_STATE_CHANGED","PEER_CONNECTION_STATE_CHANGED","ICE_CONNECTION_STATE_CHANGED","connectionStateChange","LoggerProxy","logger","log","concat","warn","reject","clearTimeout","timer","resolve","peerConnectionState","getPeerConnectionState","iceConnectionState","getIceConnectionState","iceGatheringState","setTimeout","ICE_AND_DTLS_CONNECTION_TIMEOUT","_sendMetric","_asyncToGenerator2","_regenerator","mark","_callee","_transportReports$","_transportReports$2","_transportReports$3","_transportReports$4","_this$webrtcMediaConn","_this$webrtcMediaConn2","stats","transportReports","wrap","_context","prev","next","getStats","sent","_from","values","filter","report","type","Metrics","sendBehavioralMetric","BEHAVIORAL_METRICS","MEDIA_STILL_NOT_CONNECTED","correlation_id","numTransports","length","dtlsState","iceState","packetsSent","packetsReceived","dataChannelState","multistreamConnection","dataChannel","readyState","stop","sendMetric","apply","arguments","waitForMediaConnectionConnected","_promise","on","promise"],"sources":["MediaConnectionAwaiter.ts"],"sourcesContent":["import {Defer} from '@webex/common';\nimport {ConnectionState, MediaConnectionEventNames} from '@webex/internal-media-core';\nimport LoggerProxy from '../common/logs/logger-proxy';\nimport {ICE_AND_DTLS_CONNECTION_TIMEOUT} from '../constants';\nimport BEHAVIORAL_METRICS from '../metrics/constants';\nimport Metrics from '../metrics';\n\nexport interface MediaConnectionAwaiterProps {\n  webrtcMediaConnection: any;\n  correlationId: string;\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 iceConnected: boolean;\n  private correlationId: string;\n  private onTimeoutCallback: () => void;\n  private peerConnectionStateCallback: () => void;\n  private iceConnectionStateCallback: () => void;\n  private iceGatheringStateCallback: () => void;\n\n  /**\n   * @param {MediaConnectionAwaiterProps} mediaConnectionAwaiterProps\n   */\n  constructor({webrtcMediaConnection, correlationId}: MediaConnectionAwaiterProps) {\n    this.webrtcMediaConnection = webrtcMediaConnection;\n    this.defer = new Defer();\n    this.retried = false;\n    this.iceConnected = false;\n    this.correlationId = correlationId;\n    this.onTimeoutCallback = this.onTimeout.bind(this);\n    this.peerConnectionStateCallback = this.peerConnectionStateHandler.bind(this);\n    this.iceConnectionStateCallback = this.iceConnectionStateHandler.bind(this);\n    this.iceGatheringStateCallback = this.iceGatheringStateHandler.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 connection is in an unrecoverable \"failed\" state\n   *\n   * @returns {boolean}\n   */\n  private isFailed(): boolean {\n    return this.webrtcMediaConnection.getConnectionState() === ConnectionState.Failed;\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      MediaConnectionEventNames.ICE_GATHERING_STATE_CHANGED,\n      this.iceGatheringStateCallback\n    );\n    this.webrtcMediaConnection.off(\n      MediaConnectionEventNames.PEER_CONNECTION_STATE_CHANGED,\n      this.peerConnectionStateCallback\n    );\n    this.webrtcMediaConnection.off(\n      MediaConnectionEventNames.ICE_CONNECTION_STATE_CHANGED,\n      this.iceConnectionStateCallback\n    );\n  }\n\n  /**\n   * On connection state change.\n   *\n   * @returns {void}\n   */\n  connectionStateChange(): void {\n    LoggerProxy.logger.log(\n      `Media:MediaConnectionAwaiter#connectionStateChange --> connection state: ${this.webrtcMediaConnection.getConnectionState()}`\n    );\n\n    if (this.isFailed()) {\n      LoggerProxy.logger.warn(\n        'Media:MediaConnectionAwaiter#connectionStateChange --> ICE failed, rejecting'\n      );\n      this.clearCallbacks();\n\n      this.defer.reject({\n        iceConnected: this.iceConnected,\n      });\n    }\n\n    if (!this.isConnected()) {\n      return;\n    }\n\n    clearTimeout(this.timer);\n\n    this.clearCallbacks();\n\n    LoggerProxy.logger.warn('Media:MediaConnectionAwaiter#connectionStateChange --> Resolving');\n\n    this.defer.resolve();\n  }\n\n  /**\n   * Listener for peer connection state change.\n   *\n   * @returns {void}\n   */\n  peerConnectionStateHandler(): void {\n    const peerConnectionState = this.webrtcMediaConnection.getPeerConnectionState();\n\n    LoggerProxy.logger.log(\n      `Media:MediaConnectionAwaiter#peerConnectionStateHandler --> Peer connection state change -> ${peerConnectionState}`\n    );\n\n    this.connectionStateChange();\n  }\n\n  /**\n   * Listener for ICE connection state change.\n   *\n   * @returns {void}\n   */\n  iceConnectionStateHandler(): void {\n    const iceConnectionState = this.webrtcMediaConnection.getIceConnectionState();\n\n    LoggerProxy.logger.log(\n      `Media:MediaConnectionAwaiter#iceConnectionStateHandler --> ICE connection state change -> ${iceConnectionState}`\n    );\n\n    if (iceConnectionState === 'connected' && !this.iceConnected) {\n      this.iceConnected = true;\n    }\n\n    this.connectionStateChange();\n  }\n\n  /**\n   * Listener for ICE gathering state change.\n   *\n   * @returns {void}\n   */\n  iceGatheringStateHandler(): void {\n    const iceGatheringState = this.webrtcMediaConnection.getIceGatheringState();\n\n    LoggerProxy.logger.log(\n      `Media:MediaConnectionAwaiter#iceGatheringStateHandler --> 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   * sends a metric with some additional info that might help debugging\n   * issues where browser doesn't update the RTCPeerConnection's iceConnectionState or connectionState\n   *\n   * @returns {void}\n   */\n  async sendMetric() {\n    const stats = await this.webrtcMediaConnection.getStats();\n\n    // in theory we can end up with more than one transport report in the stats,\n    // but for the purpose of this metric it's fine to just use the first one\n    const transportReports = Array.from(\n      stats.values().filter((report) => report.type === 'transport')\n    ) as Record<string, number | string>[];\n\n    Metrics.sendBehavioralMetric(BEHAVIORAL_METRICS.MEDIA_STILL_NOT_CONNECTED, {\n      correlation_id: this.correlationId,\n      numTransports: transportReports.length,\n      dtlsState: transportReports[0]?.dtlsState,\n      iceState: transportReports[0]?.iceState,\n      packetsSent: transportReports[0]?.packetsSent,\n      packetsReceived: transportReports[0]?.packetsReceived,\n      dataChannelState: this.webrtcMediaConnection.multistreamConnection?.dataChannel?.readyState,\n    });\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    this.sendMetric();\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      iceConnected: this.iceConnected,\n    });\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      LoggerProxy.logger.log(\n        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{Defer} from '@webex/common';\nimport {ConnectionState, MediaConnectionEventNames} from '@webex/internal-media-core';\nimport LoggerProxy from '../common/logs/logger-proxy';\nimport {ICE_AND_DTLS_CONNECTION_TIMEOUT} from '../constants';\nimport BEHAVIORAL_METRICS from '../metrics/constants';\nimport Metrics from '../metrics';\n\nexport interface MediaConnectionAwaiterProps {\n  webrtcMediaConnection: any;\n  correlationId: string;\n}\n\nexport interface FailureResult {\n  iceConnected: boolean;\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 iceConnected: boolean;\n  private correlationId: string;\n  private onTimeoutCallback: () => void;\n  private peerConnectionStateCallback: () => void;\n  private iceConnectionStateCallback: () => void;\n  private iceGatheringStateCallback: () => void;\n\n  /**\n   * @param {MediaConnectionAwaiterProps} mediaConnectionAwaiterProps\n   */\n  constructor({webrtcMediaConnection, correlationId}: MediaConnectionAwaiterProps) {\n    this.webrtcMediaConnection = webrtcMediaConnection;\n    this.defer = new Defer();\n    this.retried = false;\n    this.iceConnected = false;\n    this.correlationId = correlationId;\n    this.onTimeoutCallback = this.onTimeout.bind(this);\n    this.peerConnectionStateCallback = this.peerConnectionStateHandler.bind(this);\n    this.iceConnectionStateCallback = this.iceConnectionStateHandler.bind(this);\n    this.iceGatheringStateCallback = this.iceGatheringStateHandler.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 connection is in an unrecoverable \"failed\" state\n   *\n   * @returns {boolean}\n   */\n  private isFailed(): boolean {\n    return this.webrtcMediaConnection.getConnectionState() === ConnectionState.Failed;\n  }\n\n  /**\n   * Returns true if ICE connection state indicates connectivity.\n   *\n   * @returns {boolean}\n   */\n  private isIceConnected(): boolean {\n    const state = this.webrtcMediaConnection.getIceConnectionState();\n\n    return state === 'connected' || state === 'completed';\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      MediaConnectionEventNames.ICE_GATHERING_STATE_CHANGED,\n      this.iceGatheringStateCallback\n    );\n    this.webrtcMediaConnection.off(\n      MediaConnectionEventNames.PEER_CONNECTION_STATE_CHANGED,\n      this.peerConnectionStateCallback\n    );\n    this.webrtcMediaConnection.off(\n      MediaConnectionEventNames.ICE_CONNECTION_STATE_CHANGED,\n      this.iceConnectionStateCallback\n    );\n  }\n\n  /**\n   * On connection state change.\n   *\n   * @returns {void}\n   */\n  connectionStateChange(): void {\n    LoggerProxy.logger.log(\n      `Media:MediaConnectionAwaiter#connectionStateChange --> connection state: ${this.webrtcMediaConnection.getConnectionState()}`\n    );\n\n    if (this.isFailed()) {\n      LoggerProxy.logger.warn(\n        'Media:MediaConnectionAwaiter#connectionStateChange --> ICE failed, rejecting'\n      );\n      this.clearCallbacks();\n\n      this.defer.reject({\n        iceConnected: this.iceConnected,\n      } satisfies FailureResult);\n    }\n\n    if (!this.isConnected()) {\n      return;\n    }\n\n    clearTimeout(this.timer);\n\n    this.clearCallbacks();\n\n    LoggerProxy.logger.warn('Media:MediaConnectionAwaiter#connectionStateChange --> Resolving');\n\n    this.defer.resolve();\n  }\n\n  /**\n   * Listener for peer connection state change.\n   *\n   * @returns {void}\n   */\n  peerConnectionStateHandler(): void {\n    const peerConnectionState = this.webrtcMediaConnection.getPeerConnectionState();\n\n    LoggerProxy.logger.log(\n      `Media:MediaConnectionAwaiter#peerConnectionStateHandler --> Peer connection state change -> ${peerConnectionState}`\n    );\n\n    this.connectionStateChange();\n  }\n\n  /**\n   * Listener for ICE connection state change.\n   *\n   * @returns {void}\n   */\n  iceConnectionStateHandler(): void {\n    const iceConnectionState = this.webrtcMediaConnection.getIceConnectionState();\n\n    LoggerProxy.logger.log(\n      `Media:MediaConnectionAwaiter#iceConnectionStateHandler --> ICE connection state change -> ${iceConnectionState}`\n    );\n\n    if (this.isIceConnected() && !this.iceConnected) {\n      this.iceConnected = true;\n    }\n\n    this.connectionStateChange();\n  }\n\n  /**\n   * Listener for ICE gathering state change.\n   *\n   * @returns {void}\n   */\n  iceGatheringStateHandler(): void {\n    const iceGatheringState = this.webrtcMediaConnection.getIceGatheringState();\n\n    LoggerProxy.logger.log(\n      `Media:MediaConnectionAwaiter#iceGatheringStateHandler --> 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   * sends a metric with some additional info that might help debugging\n   * issues where browser doesn't update the RTCPeerConnection's iceConnectionState or connectionState\n   *\n   * @returns {void}\n   */\n  async sendMetric() {\n    const stats = await this.webrtcMediaConnection.getStats();\n\n    // in theory we can end up with more than one transport report in the stats,\n    // but for the purpose of this metric it's fine to just use the first one\n    const transportReports = Array.from(\n      stats.values().filter((report) => report.type === 'transport')\n    ) as Record<string, number | string>[];\n\n    Metrics.sendBehavioralMetric(BEHAVIORAL_METRICS.MEDIA_STILL_NOT_CONNECTED, {\n      correlation_id: this.correlationId,\n      numTransports: transportReports.length,\n      dtlsState: transportReports[0]?.dtlsState,\n      iceState: transportReports[0]?.iceState,\n      packetsSent: transportReports[0]?.packetsSent,\n      packetsReceived: transportReports[0]?.packetsReceived,\n      dataChannelState: this.webrtcMediaConnection.multistreamConnection?.dataChannel?.readyState,\n    });\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    this.sendMetric();\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      iceConnected: this.iceConnected,\n    } satisfies FailureResult);\n  }\n\n  /**\n   * Waits for the webrtc media connection to be connected.\n   *\n   * @returns {Promise<void>} In case of failure, the promise is rejected with FailureResult\n   */\n  waitForMediaConnectionConnected(): Promise<void> {\n    if (this.isConnected()) {\n      LoggerProxy.logger.log(\n        'Media:MediaConnectionAwaiter#waitForMediaConnectionConnected --> Already connected'\n      );\n\n      return Promise.resolve();\n    }\n    LoggerProxy.logger.log(\n      'Media:MediaConnectionAwaiter#waitForMediaConnectionConnected --> Waiting for media connection to be connected'\n    );\n\n    this.iceConnected = this.isIceConnected();\n\n    this.webrtcMediaConnection.on(\n      MediaConnectionEventNames.PEER_CONNECTION_STATE_CHANGED,\n      this.peerConnectionStateCallback\n    );\n\n    this.webrtcMediaConnection.on(\n      MediaConnectionEventNames.ICE_CONNECTION_STATE_CHANGED,\n      this.iceConnectionStateCallback\n    );\n\n    this.webrtcMediaConnection.on(\n      MediaConnectionEventNames.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

@@ -118,6 +118,7 @@ Media.createMediaConnection = function (isMultistream, debugId, meetingId, optio
     enableExtmap = options.enableExtmap,
     turnServerInfo = options.turnServerInfo,
     bundlePolicy = options.bundlePolicy,
+    iceTransportPolicy = options.iceTransportPolicy,
     iceCandidatesTimeout = options.iceCandidatesTimeout,
     disableAudioMainDtx = options.disableAudioMainDtx,
     enableAudioTwcc = options.enableAudioTwcc,
@@ -126,7 +127,7 @@ Media.createMediaConnection = function (isMultistream, debugId, meetingId, optio
   var iceServers = [];
 
   // we might not have any TURN server if TURN discovery failed or wasn't done or we land on a video mesh node
-  if ((turnServerInfo === null || turnServerInfo === void 0 ? void 0 : turnServerInfo.urls.length) > 0) {
+  if (turnServerInfo && turnServerInfo.urls.length > 0) {
     // TURN-TLS server
     iceServers.push({
       urls: turnServerInfo.urls,
@@ -143,6 +144,9 @@ Media.createMediaConnection = function (isMultistream, debugId, meetingId, optio
     if (bundlePolicy) {
       config.bundlePolicy = bundlePolicy;
     }
+    if (iceTransportPolicy) {
+      config.iceTransportPolicy = iceTransportPolicy;
+    }
     if (disableAudioMainDtx !== undefined) {
       config.disableAudioMainDtx = disableAudioMainDtx;
     }

package/dist/media/index.js.map

@@ -1 +1 @@
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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 {RtcMetrics} from '@webex/internal-plugin-metrics';\nimport {BrowserInfo} from '@webex/web-capabilities';\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 {TurnServerInfo} from '../roap/types';\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 {string} meetingId id for the meeting using this connection\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  meetingId: string,\n  options: {\n    rtcMetrics?: RtcMetrics;\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?: TurnServerInfo;\n    bundlePolicy?: BundlePolicy;\n    iceCandidatesTimeout?: number;\n    disableAudioMainDtx?: boolean;\n    enableAudioTwcc?: boolean;\n    enableAv1SlidesSupport?: boolean;\n    stopIceGatheringAfterFirstRelayCandidate?: boolean;\n  }\n) => {\n  const {\n    rtcMetrics,\n    mediaProperties,\n    remoteQualityLevel,\n    enableRtx,\n    enableExtmap,\n    turnServerInfo,\n    bundlePolicy,\n    iceCandidatesTimeout,\n    disableAudioMainDtx,\n    enableAudioTwcc,\n    enableAv1SlidesSupport,\n    stopIceGatheringAfterFirstRelayCandidate,\n  } = options;\n\n  const iceServers = [];\n\n  // we might not have any TURN server if TURN discovery failed or wasn't done or we land on a video mesh node\n  if (turnServerInfo?.urls.length > 0) {\n    // TURN-TLS server\n    iceServers.push({\n      urls: turnServerInfo.urls,\n      username: turnServerInfo.username || '',\n      credential: turnServerInfo.password || '',\n    });\n  }\n\n  if (isMultistream) {\n    const config: MultistreamConnectionConfig = {\n      iceServers,\n      disableAudioTwcc: !enableAudioTwcc,\n      enableAV1SlidesSupport: !!enableAv1SlidesSupport,\n    };\n\n    if (bundlePolicy) {\n      config.bundlePolicy = bundlePolicy;\n    }\n\n    if (disableAudioMainDtx !== undefined) {\n      config.disableAudioMainDtx = disableAudioMainDtx;\n    }\n\n    if (BrowserInfo.isFirefox()) {\n      config.doFullIce = true;\n\n      config.stopIceGatheringAfterFirstRelayCandidate = stopIceGatheringAfterFirstRelayCandidate;\n    }\n\n    if (BrowserInfo.isEdge() || BrowserInfo.isChrome()) {\n      // we need this for getting inbound audio metadata\n      // but the audioLevel that we use is only available on Chromium based browsers\n      config.enableInboundAudioLevelMonitoring = true;\n    }\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      iceCandidatesTimeout,\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(false, mediaDirection.receiveAudio, mediaDirection.sendAudio),\n        video: Media.getDirection(false, 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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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 {RtcMetrics} from '@webex/internal-plugin-metrics';\nimport {BrowserInfo} from '@webex/web-capabilities';\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 {TurnServerInfo} from '../roap/types';\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 {string} meetingId id for the meeting using this connection\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  meetingId: string,\n  options: {\n    rtcMetrics?: RtcMetrics;\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?: TurnServerInfo;\n    bundlePolicy?: BundlePolicy;\n    iceTransportPolicy?: RTCIceTransportPolicy;\n    iceCandidatesTimeout?: number;\n    disableAudioMainDtx?: boolean;\n    enableAudioTwcc?: boolean;\n    enableAv1SlidesSupport?: boolean;\n    stopIceGatheringAfterFirstRelayCandidate?: boolean;\n  }\n) => {\n  const {\n    rtcMetrics,\n    mediaProperties,\n    remoteQualityLevel,\n    enableRtx,\n    enableExtmap,\n    turnServerInfo,\n    bundlePolicy,\n    iceTransportPolicy,\n    iceCandidatesTimeout,\n    disableAudioMainDtx,\n    enableAudioTwcc,\n    enableAv1SlidesSupport,\n    stopIceGatheringAfterFirstRelayCandidate,\n  } = options;\n\n  const iceServers = [];\n\n  // we might not have any TURN server if TURN discovery failed or wasn't done or we land on a video mesh node\n  if (turnServerInfo && turnServerInfo.urls.length > 0) {\n    // TURN-TLS server\n    iceServers.push({\n      urls: turnServerInfo.urls,\n      username: turnServerInfo.username || '',\n      credential: turnServerInfo.password || '',\n    });\n  }\n\n  if (isMultistream) {\n    const config: MultistreamConnectionConfig = {\n      iceServers,\n      disableAudioTwcc: !enableAudioTwcc,\n      enableAV1SlidesSupport: !!enableAv1SlidesSupport,\n    };\n\n    if (bundlePolicy) {\n      config.bundlePolicy = bundlePolicy;\n    }\n\n    if (iceTransportPolicy) {\n      config.iceTransportPolicy = iceTransportPolicy;\n    }\n\n    if (disableAudioMainDtx !== undefined) {\n      config.disableAudioMainDtx = disableAudioMainDtx;\n    }\n\n    if (BrowserInfo.isFirefox()) {\n      config.doFullIce = true;\n\n      config.stopIceGatheringAfterFirstRelayCandidate = stopIceGatheringAfterFirstRelayCandidate;\n    }\n\n    if (BrowserInfo.isEdge() || BrowserInfo.isChrome()) {\n      // we need this for getting inbound audio metadata\n      // but the audioLevel that we use is only available on Chromium based browsers\n      config.enableInboundAudioLevelMonitoring = true;\n    }\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      iceCandidatesTimeout,\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(false, mediaDirection.receiveAudio, mediaDirection.sendAudio),\n        video: Media.getDirection(false, 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: 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