@webex/plugin-meetings 3.3.1-next.45 → 3.3.1-next.47

package/dist/breakouts/breakout.js

@@ -209,7 +209,7 @@ var Breakout = _webexCore.WebexPlugin.extend({
       sessionId: this.sessionId
     });
   },
-  version: "3.3.1-next.45"
+  version: "3.3.1-next.47"
 });
 var _default = exports.default = Breakout;
 //# sourceMappingURL=breakout.js.map

package/dist/breakouts/index.js

@@ -1046,7 +1046,7 @@ var Breakouts = _webexCore.WebexPlugin.extend({
       this.trigger(_constants.BREAKOUTS.EVENTS.ASK_RETURN_TO_MAIN);
     }
   },
-  version: "3.3.1-next.45"
+  version: "3.3.1-next.47"
 });
 var _default = exports.default = Breakouts;
 //# sourceMappingURL=index.js.map

package/dist/interpretation/index.js

@@ -373,7 +373,7 @@ var SimultaneousInterpretation = _webexCore.WebexPlugin.extend({
       throw error;
     });
   },
-  version: "3.3.1-next.45"
+  version: "3.3.1-next.47"
 });
 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: "3.3.1-next.45"
+  version: "3.3.1-next.47"
 });
 var _default = exports.default = SILanguage;
 //# sourceMappingURL=siLanguage.js.map

package/dist/media/index.js

@@ -124,6 +124,18 @@ Media.createMediaConnection = function (isMultistream, debugId, webex, meetingId
   // 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) {
+    if (!isBrowser('firefox')) {
+      var bareTurnServer = turnServerInfo.url;
+      bareTurnServer = bareTurnServer.replace('turns:', 'turn:');
+      bareTurnServer = bareTurnServer.replace('443', '5004');
+      iceServers.push({
+        urls: bareTurnServer,
+        username: turnServerInfo.username || '',
+        credential: turnServerInfo.password || ''
+      });
+    }
+
+    // TURN-TLS server
     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","url","urls","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  // 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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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    if (!isBrowser('firefox')) {\n      let bareTurnServer = turnServerInfo.url;\n      bareTurnServer = bareTurnServer.replace('turns:', 'turn:');\n      bareTurnServer = bareTurnServer.replace('443', '5004');\n\n      iceServers.push({\n        urls: bareTurnServer,\n        username: turnServerInfo.username || '',\n        credential: turnServerInfo.password || '',\n      });\n    }\n\n    // TURN-TLS server\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/webinar/index.js

@@ -62,7 +62,7 @@ var Webinar = _webexCore.WebexPlugin.extend({
   updateCanManageWebcast: function updateCanManageWebcast(canManageWebcast) {
     this.set('canManageWebcast', canManageWebcast);
   },
-  version: "3.3.1-next.45"
+  version: "3.3.1-next.47"
 });
 var _default = exports.default = Webinar;
 //# sourceMappingURL=index.js.map

package/package.json

@@ -43,7 +43,7 @@
     "@webex/eslint-config-legacy": "0.0.0",
     "@webex/jest-config-legacy": "0.0.0",
     "@webex/legacy-tools": "0.0.0",
-    "@webex/plugin-meetings": "3.3.1-next.45",
+    "@webex/plugin-meetings": "3.3.1-next.47",
     "@webex/plugin-rooms": "3.3.1-next.13",
     "@webex/test-helper-chai": "3.3.1-next.12",
     "@webex/test-helper-mocha": "3.3.1-next.12",
@@ -62,7 +62,7 @@
   },
   "dependencies": {
     "@webex/common": "3.3.1-next.12",
-    "@webex/internal-media-core": "2.9.2",
+    "@webex/internal-media-core": "2.9.3",
     "@webex/internal-plugin-conversation": "3.3.1-next.13",
     "@webex/internal-plugin-device": "3.3.1-next.12",
     "@webex/internal-plugin-llm": "3.3.1-next.14",
@@ -70,8 +70,8 @@
     "@webex/internal-plugin-metrics": "3.3.1-next.12",
     "@webex/internal-plugin-support": "3.3.1-next.13",
     "@webex/internal-plugin-user": "3.3.1-next.12",
-    "@webex/internal-plugin-voicea": "3.3.1-next.45",
-    "@webex/media-helpers": "3.3.1-next.22",
+    "@webex/internal-plugin-voicea": "3.3.1-next.47",
+    "@webex/media-helpers": "3.3.1-next.23",
     "@webex/plugin-people": "3.3.1-next.13",
     "@webex/plugin-rooms": "3.3.1-next.13",
     "@webex/web-capabilities": "^1.4.0",
@@ -91,5 +91,5 @@
   "//": [
     "TODO: upgrade jwt-decode when moving to node 18"
   ],
-  "version": "3.3.1-next.45"
+  "version": "3.3.1-next.47"
 }

package/src/media/index.ts

@@ -163,6 +163,19 @@ Media.createMediaConnection = (
   // 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?.url) {
+    if (!isBrowser('firefox')) {
+      let bareTurnServer = turnServerInfo.url;
+      bareTurnServer = bareTurnServer.replace('turns:', 'turn:');
+      bareTurnServer = bareTurnServer.replace('443', '5004');
+
+      iceServers.push({
+        urls: bareTurnServer,
+        username: turnServerInfo.username || '',
+        credential: turnServerInfo.password || '',
+      });
+    }
+
+    // TURN-TLS server
     iceServers.push({
       urls: turnServerInfo.url,
       username: turnServerInfo.username || '',

package/test/unit/spec/media/index.ts

@@ -16,39 +16,39 @@ describe('createMediaConnection', () => {
     id: 'roap media connection',
   };
   const fakeTrack = {
-    id: 'any fake track'
-  }
+    id: 'any fake track',
+  };
   const fakeAudioStream = {
     outputStream: {
       getTracks: () => {
         return [fakeTrack];
-      }
-    }
+      },
+    },
   };
   const fakeVideoStream = {
     outputStream: {
       getTracks: () => {
         return [fakeTrack];
-      }
-    }
+      },
+    },
   };
   const fakeShareVideoStream = {
     outputStream: {
       getTracks: () => {
         return [fakeTrack];
-      }
-    }
+      },
+    },
   };
   const fakeShareAudioStream = {
     outputStream: {
       getTracks: () => {
         return [fakeTrack];
-      }
-    }
+      },
+    },
   };
   afterEach(() => {
     sinon.restore();
-    clock.uninstall()
+    clock.uninstall();
   });
 
   it('creates a RoapMediaConnection when multistream is disabled', () => {
@@ -80,7 +80,7 @@ describe('createMediaConnection', () => {
       enableRtx: ENABLE_RTX,
       enableExtmap: ENABLE_EXTMAP,
       turnServerInfo: {
-        url: 'turn server url',
+        url: 'turns:turn-server-url:443?transport=tcp',
         username: 'turn username',
         password: 'turn password',
       },
@@ -91,7 +91,12 @@ describe('createMediaConnection', () => {
       {
         iceServers: [
           {
-            urls: 'turn server url',
+            urls: 'turn:turn-server-url:5004?transport=tcp',
+            username: 'turn username',
+            credential: 'turn password',
+          },
+          {
+            urls: 'turns:turn-server-url:443?transport=tcp',
             username: 'turn username',
             credential: 'turn password',
           },
@@ -146,7 +151,7 @@ describe('createMediaConnection', () => {
         },
       },
       turnServerInfo: {
-        url: 'turn server url',
+        url: 'turns:turn-server-url:443?transport=tcp',
         username: 'turn username',
         password: 'turn password',
       },
@@ -158,7 +163,12 @@ describe('createMediaConnection', () => {
       {
         iceServers: [
           {
-            urls: 'turn server url',
+            urls: 'turn:turn-server-url:5004?transport=tcp',
+            username: 'turn username',
+            credential: 'turn password',
+          },
+          {
+            urls: 'turns:turn-server-url:443?transport=tcp',
             username: 'turn username',
             credential: 'turn password',
           },
@@ -171,7 +181,10 @@ describe('createMediaConnection', () => {
 
   [
     {testCase: 'turnServerInfo is undefined', turnServerInfo: undefined},
-    {testCase: 'turnServerInfo.url is empty string', turnServerInfo: {url: '', username: 'turn username', password: 'turn password'}},
+    {
+      testCase: 'turnServerInfo.url is empty string',
+      turnServerInfo: {url: '', username: 'turn username', password: 'turn password'},
+    },
   ].forEach(({testCase, turnServerInfo}) => {
     it(`passes empty ICE servers array to MultistreamRoapMediaConnection if ${testCase} (multistream enabled)`, () => {
       const multistreamRoapMediaConnectionConstructorStub = sinon
@@ -198,7 +211,7 @@ describe('createMediaConnection', () => {
           iceServers: [],
         },
         'meeting id'
-        );
+      );
     });
   });
 
@@ -232,7 +245,10 @@ describe('createMediaConnection', () => {
 
   [
     {testCase: 'turnServerInfo is undefined', turnServerInfo: undefined},
-    {testCase: 'turnServerInfo.url is empty string', turnServerInfo: {url: '', username: 'turn username', password: 'turn password'}},
+    {
+      testCase: 'turnServerInfo.url is empty string',
+      turnServerInfo: {url: '', username: 'turn username', password: 'turn password'},
+    },
   ].forEach(({testCase, turnServerInfo}) => {
     it(`passes empty ICE servers array to RoapMediaConnection if ${testCase} (multistream disabled)`, () => {
       const roapMediaConnectionConstructorStub = sinon

package/test/unit/spec/meeting/index.js

@@ -625,7 +625,9 @@ describe('plugin-meetings', () => {
 
         beforeEach(() => {
           meeting.join = sinon.stub().returns(Promise.resolve(fakeJoinResult));
-          addMediaInternalStub = sinon.stub(meeting, 'addMediaInternal').returns(Promise.resolve(test4));
+          addMediaInternalStub = sinon
+            .stub(meeting, 'addMediaInternal')
+            .returns(Promise.resolve(test4));
 
           webex.meetings.reachability.getReachabilityResults.resolves(fakeReachabilityResults);
 
@@ -656,12 +658,21 @@ describe('plugin-meetings', () => {
             meeting,
             fakeJoinResult
           );
-          assert.calledOnceWithExactly(meeting.addMediaInternal, sinon.match.any, fakeTurnServerInfo, false, mediaOptions);
+          assert.calledOnceWithExactly(
+            meeting.addMediaInternal,
+            sinon.match.any,
+            fakeTurnServerInfo,
+            false,
+            mediaOptions
+          );
 
           assert.deepEqual(result, {join: fakeJoinResult, media: test4});
 
           // resets joinWithMediaRetryInfo
-          assert.deepEqual(meeting.joinWithMediaRetryInfo, {isRetry: false, prevJoinResponse: undefined});
+          assert.deepEqual(meeting.joinWithMediaRetryInfo, {
+            isRetry: false,
+            prevJoinResponse: undefined,
+          });
         });
 
         it("should not call handleTurnDiscoveryHttpResponse if we don't send a TURN discovery request with join", async () => {
@@ -681,7 +692,13 @@ describe('plugin-meetings', () => {
           assert.calledOnceWithExactly(generateTurnDiscoveryRequestMessageStub, meeting, true);
           assert.notCalled(handleTurnDiscoveryHttpResponseStub);
           assert.notCalled(abortTurnDiscoveryStub);
-          assert.calledOnceWithExactly(meeting.addMediaInternal, sinon.match.any, undefined, false, mediaOptions);
+          assert.calledOnceWithExactly(
+            meeting.addMediaInternal,
+            sinon.match.any,
+            undefined,
+            false,
+            mediaOptions
+          );
 
           assert.deepEqual(result, {join: fakeJoinResult, media: test4});
           assert.equal(meeting.turnServerUsed, false);
@@ -711,7 +728,13 @@ describe('plugin-meetings', () => {
             fakeJoinResult
           );
           assert.calledOnceWithExactly(abortTurnDiscoveryStub);
-          assert.calledOnceWithExactly(meeting.addMediaInternal, sinon.match.any, undefined, false, mediaOptions);
+          assert.calledOnceWithExactly(
+            meeting.addMediaInternal,
+            sinon.match.any,
+            undefined,
+            false,
+            mediaOptions
+          );
 
           assert.deepEqual(result, {join: fakeJoinResult, media: test4});
         });
@@ -758,12 +781,20 @@ describe('plugin-meetings', () => {
           );
 
           // resets joinWithMediaRetryInfo
-          assert.deepEqual(meeting.joinWithMediaRetryInfo, {isRetry: false, prevJoinResponse: undefined});
+          assert.deepEqual(meeting.joinWithMediaRetryInfo, {
+            isRetry: false,
+            prevJoinResponse: undefined,
+          });
         });
 
         it('should resolve if join() fails the first time but succeeds the second time', async () => {
           const error = new Error('fake');
-          meeting.join = sinon.stub().onFirstCall().returns(Promise.reject(error)).onSecondCall().returns(Promise.resolve(fakeJoinResult));
+          meeting.join = sinon
+            .stub()
+            .onFirstCall()
+            .returns(Promise.reject(error))
+            .onSecondCall()
+            .returns(Promise.resolve(fakeJoinResult));
           const leaveStub = sinon.stub(meeting, 'leave').resolves();
 
           const result = await meeting.joinWithMedia({
@@ -795,7 +826,10 @@ describe('plugin-meetings', () => {
           assert.deepEqual(result, {join: fakeJoinResult, media: test4});
 
           // resets joinWithMediaRetryInfo
-          assert.deepEqual(meeting.joinWithMediaRetryInfo, {isRetry: false, prevJoinResponse: undefined});
+          assert.deepEqual(meeting.joinWithMediaRetryInfo, {
+            isRetry: false,
+            prevJoinResponse: undefined,
+          });
         });
 
         it('should fail if called with allowMediaInLobby:false', async () => {
@@ -828,7 +862,6 @@ describe('plugin-meetings', () => {
             reason: 'joinWithMedia failure',
           });
 
-
           // Behavioral metric is sent on both calls of joinWithMedia
           assert.calledTwice(Metrics.sendBehavioralMetric);
           assert.calledWith(
@@ -1068,12 +1101,15 @@ describe('plugin-meetings', () => {
           const addMediaError = new Error('fake addMedia error');
           addMediaError.name = 'SdpOfferCreationError';
 
-          meeting.addMediaInternal.rejects(addMediaError)
+          meeting.addMediaInternal.rejects(addMediaError);
 
-          await assert.isRejected(meeting.joinWithMedia({
-            joinOptions,
-            mediaOptions,
-          }), addMediaError);
+          await assert.isRejected(
+            meeting.joinWithMedia({
+              joinOptions,
+              mediaOptions,
+            }),
+            addMediaError
+          );
 
           // check that only 1 attempt was done
           assert.calledOnce(meeting.join);
@@ -1339,10 +1375,7 @@ describe('plugin-meetings', () => {
 
         it('should trigger meeting:caption-received event', () => {
           meeting.voiceaListenerCallbacks[VOICEAEVENTS.NEW_CAPTION]({});
-          assert.calledWith(
-            meeting.trigger,
-            EVENT_TRIGGERS.MEETING_CAPTION_RECEIVED
-          );
+          assert.calledWith(meeting.trigger, EVENT_TRIGGERS.MEETING_CAPTION_RECEIVED);
         });
 
         it('should trigger meeting:receiveTranscription:started event', () => {
@@ -1355,10 +1388,7 @@ describe('plugin-meetings', () => {
 
         it('should trigger meeting:caption-received event', () => {
           meeting.voiceaListenerCallbacks[VOICEAEVENTS.NEW_CAPTION]({});
-          assert.calledWith(
-            meeting.trigger,
-            EVENT_TRIGGERS.MEETING_CAPTION_RECEIVED
-          );
+          assert.calledWith(meeting.trigger, EVENT_TRIGGERS.MEETING_CAPTION_RECEIVED);
         });
       });
 
@@ -1513,11 +1543,7 @@ describe('plugin-meetings', () => {
 
         it('turns off llm online, emits transcription connected events', () => {
           meeting.handleLLMOnline();
-          assert.calledOnceWithExactly(
-            webex.internal.llm.off,
-            'online',
-            meeting.handleLLMOnline
-          );
+          assert.calledOnceWithExactly(webex.internal.llm.off, 'online', meeting.handleLLMOnline);
           assert.calledWith(
             TriggerProxy.trigger,
             sinon.match.instanceOf(Meeting),
@@ -1579,11 +1605,7 @@ describe('plugin-meetings', () => {
             assert.calledOnce(MeetingUtil.joinMeeting);
             assert.calledOnce(meeting.setLocus);
             assert.equal(result, joinMeetingResult);
-            assert.calledWith(
-              webex.internal.llm.on,
-              'online',
-              meeting.handleLLMOnline
-            );
+            assert.calledWith(webex.internal.llm.on, 'online', meeting.handleLLMOnline);
           });
 
           [true, false].forEach((enableMultistream) => {
@@ -1906,7 +1928,9 @@ describe('plugin-meetings', () => {
           };
           meeting.mediaProperties.setMediaDirection = sinon.stub().returns(true);
           meeting.mediaProperties.waitForMediaConnectionConnected = sinon.stub().resolves();
-          meeting.mediaProperties.getCurrentConnectionInfo = sinon.stub().resolves({connectionType: 'udp', selectedCandidatePairChanges: 2, numTransports: 1});
+          meeting.mediaProperties.getCurrentConnectionInfo = sinon
+            .stub()
+            .resolves({connectionType: 'udp', selectedCandidatePairChanges: 2, numTransports: 1});
           meeting.audio = muteStateStub;
           meeting.video = muteStateStub;
           sinon.stub(Media, 'createMediaConnection').returns(fakeMediaConnection);
@@ -1920,7 +1944,12 @@ describe('plugin-meetings', () => {
           // normally the first Roap message we send is creating confluence, so mock LocusMediaRequest.isConfluenceCreated()
           // to return false the first time it's called and true the 2nd time, to simulate how it would happen for real
           meeting.locusMediaRequest = {
-            isConfluenceCreated: sinon.stub().onFirstCall().returns(false).onSecondCall().returns(true)
+            isConfluenceCreated: sinon
+              .stub()
+              .onFirstCall()
+              .returns(false)
+              .onSecondCall()
+              .returns(true),
           };
         });
 
@@ -2485,7 +2514,6 @@ describe('plugin-meetings', () => {
         it('should reject if waitForMediaConnectionConnected() rejects after turn server retry', async () => {
           const FAKE_ERROR = {fatal: true};
           const getErrorPayloadForClientErrorCodeStub =
-
             (webex.internal.newMetrics.callDiagnosticMetrics.getErrorPayloadForClientErrorCode =
               sinon.stub().returns(FAKE_ERROR));
           webex.meetings.reachability = {
@@ -2690,9 +2718,16 @@ describe('plugin-meetings', () => {
         it('should resolve if waitForMediaConnectionConnected() rejects the first time but resolves the second time', async () => {
           const FAKE_ERROR = {fatal: true};
           webex.meetings.reachability = {
-            isWebexMediaBackendUnreachable: sinon.stub().onCall(0).rejects().onCall(1).resolves(true).onCall(2).resolves(false),
+            isWebexMediaBackendUnreachable: sinon
+              .stub()
+              .onCall(0)
+              .rejects()
+              .onCall(1)
+              .resolves(true)
+              .onCall(2)
+              .resolves(false),
             getReachabilityMetrics: sinon.stub().resolves({}),
-          }
+          };
           const getErrorPayloadForClientErrorCodeStub =
             (webex.internal.newMetrics.callDiagnosticMetrics.getErrorPayloadForClientErrorCode =
               sinon.stub().returns(FAKE_ERROR));
@@ -3445,9 +3480,9 @@ describe('plugin-meetings', () => {
                 .returns(clientErrorCode);
 
               meeting.meetingState = 'ACTIVE';
-              meeting.mediaProperties.waitForMediaConnectionConnected.rejects(
-                {iceConnected: false}
-              );
+              meeting.mediaProperties.waitForMediaConnectionConnected.rejects({
+                iceConnected: false,
+              });
 
               let errorThrown = false;
 
@@ -3562,12 +3597,18 @@ describe('plugin-meetings', () => {
             meeting.meetingState = 'ACTIVE';
             meeting.selfUrl = 'selfUrl';
             meeting.mediaProperties.waitForMediaConnectionConnected = sinon.stub().resolves();
-            meeting.mediaProperties.getCurrentConnectionInfo = sinon.stub().resolves({connectionType: 'udp', selectedCandidatePairChanges: 2, numTransports: 1});
+            meeting.mediaProperties.getCurrentConnectionInfo = sinon
+              .stub()
+              .resolves({connectionType: 'udp', selectedCandidatePairChanges: 2, numTransports: 1});
             meeting.setMercuryListener = sinon.stub();
             meeting.locusInfo.onFullLocus = sinon.stub();
             meeting.webex.meetings.geoHintInfo = {regionCode: 'EU', countryCode: 'UK'};
             meeting.roap.doTurnDiscovery = sinon.stub().resolves({
-              turnServerInfo: {url: 'turn-url', username: 'turn user', password: 'turn password'},
+              turnServerInfo: {
+                url: 'turns:turn-server-url:443?transport=tcp',
+                username: 'turn user',
+                password: 'turn password',
+              },
               turnDiscoverySkippedReason: 'reachability',
             });
             meeting.deferSDPAnswer = new Defer();
@@ -3580,7 +3621,18 @@ describe('plugin-meetings', () => {
             // setup things that are expected to be the same across all the tests and are actually irrelevant for these tests
             expectedDebugId = `MC-${meeting.id.substring(0, 4)}`;
             expectedMediaConnectionConfig = {
-              iceServers: [{urls: 'turn-url', username: 'turn user', credential: 'turn password'}],
+              iceServers: [
+                {
+                  urls: 'turn:turn-server-url:5004?transport=tcp',
+                  username: 'turn user',
+                  credential: 'turn password',
+                },
+                {
+                  urls: 'turns:turn-server-url:443?transport=tcp',
+                  username: 'turn user',
+                  credential: 'turn password',
+                },
+              ],
               skipInactiveTransceivers: false,
               requireH264: true,
               sdpMunging: {
@@ -3665,13 +3717,11 @@ describe('plugin-meetings', () => {
             // that's being tested in these tests)
             meeting.webex.meetings.registered = true;
             meeting.webex.internal.device.config = {};
-            sinon
-              .stub(MeetingUtil, 'joinMeeting')
-              .resolves({
-                id: 'fake locus from mocked join request',
-                locusUrl: 'fake locus url',
-                mediaId: 'fake media id',
-              });
+            sinon.stub(MeetingUtil, 'joinMeeting').resolves({
+              id: 'fake locus from mocked join request',
+              locusUrl: 'fake locus url',
+              mediaId: 'fake media id',
+            });
             await meeting.join({enableMultistream: isMultistream});
           });
 
@@ -3701,7 +3751,8 @@ describe('plugin-meetings', () => {
 
             for (let idx = 0; idx < roapMediaConnectionToCheck.on.callCount; idx += 1) {
               if (
-                roapMediaConnectionToCheck.on.getCall(idx).args[0] === MediaConnectionEventNames.ROAP_MESSAGE_TO_SEND
+                roapMediaConnectionToCheck.on.getCall(idx).args[0] ===
+                MediaConnectionEventNames.ROAP_MESSAGE_TO_SEND
               ) {
                 return roapMediaConnectionToCheck.on.getCall(idx).args[1];
               }
@@ -4148,13 +4199,13 @@ describe('plugin-meetings', () => {
             await meeting.addMedia({
               localStreams: {microphone: fakeMicrophoneStream},
               audioEnabled: false,
-              videoEnabled: false
+              videoEnabled: false,
             });
             await simulateRoapOffer();
             await simulateRoapOk();
 
             assert.notCalled(handleDeviceLoggingSpy);
-          })
+          });
 
           it('addMedia() works correctly when media is disabled with no streams to publish', async () => {
             await meeting.addMedia({audioEnabled: false});
@@ -6589,14 +6640,14 @@ describe('plugin-meetings', () => {
         beforeEach(() => {
           sandbox = sinon.createSandbox();
           meeting.statsAnalyzer = {
-            stopAnalyzer: sinon.stub().returns(Promise.resolve())
+            stopAnalyzer: sinon.stub().returns(Promise.resolve()),
           };
 
           meeting.reconnectionManager = {
-            cleanUp: sinon.stub()
+            cleanUp: sinon.stub(),
           };
 
-          meeting.cleanupLocalStreams=sinon.stub();
+          meeting.cleanupLocalStreams = sinon.stub();
           meeting.closeRemoteStreams = sinon.stub().returns(Promise.resolve());
           meeting.closePeerConnections = sinon.stub().returns(Promise.resolve());
           meeting.unsetRemoteStreams = sinon.stub();
@@ -6678,7 +6729,6 @@ describe('plugin-meetings', () => {
             'SELF_OBSERVING'
           );
 
-
           // Verify that the event handler behaves as expected
           expect(meeting.statsAnalyzer.stopAnalyzer.calledOnce).to.be.true;
           expect(meeting.closeRemoteStreams.calledOnce).to.be.true;
@@ -6690,11 +6740,13 @@ describe('plugin-meetings', () => {
           expect(meeting.unsetPeerConnections.calledOnce).to.be.true;
           expect(meeting.reconnectionManager.cleanUp.calledOnce).to.be.true;
           expect(meeting.mediaProperties.setMediaDirection.calledOnce).to.be.true;
-          expect(meeting.addMedia.calledOnceWithExactly({
-            audioEnabled: false,
-            videoEnabled: false,
-            shareVideoEnabled: true
-          })).to.be.true;
+          expect(
+            meeting.addMedia.calledOnceWithExactly({
+              audioEnabled: false,
+              videoEnabled: false,
+              shareVideoEnabled: true,
+            })
+          ).to.be.true;
           await testUtils.flushPromises();
           assert.equal(meeting.isMoveToInProgress, false);
         });
@@ -7482,9 +7534,11 @@ describe('plugin-meetings', () => {
           getTracks: () => [{id: 'track', addEventListener: sinon.stub()}],
         };
         const simulateConnectionStateChange = (newState) => {
-          meeting.mediaProperties.webrtcMediaConnection.getConnectionState = sinon.stub().returns(newState);
+          meeting.mediaProperties.webrtcMediaConnection.getConnectionState = sinon
+            .stub()
+            .returns(newState);
           eventListeners[MediaConnectionEventNames.PEER_CONNECTION_STATE_CHANGED]();
-        }
+        };
 
         beforeEach(() => {
           eventListeners = {};
@@ -7506,7 +7560,9 @@ describe('plugin-meetings', () => {
           assert.isFunction(eventListeners[MediaConnectionEventNames.ROAP_FAILURE]);
           assert.isFunction(eventListeners[MediaConnectionEventNames.ROAP_MESSAGE_TO_SEND]);
           assert.isFunction(eventListeners[MediaConnectionEventNames.REMOTE_TRACK_ADDED]);
-          assert.isFunction(eventListeners[MediaConnectionEventNames.PEER_CONNECTION_STATE_CHANGED]);
+          assert.isFunction(
+            eventListeners[MediaConnectionEventNames.PEER_CONNECTION_STATE_CHANGED]
+          );
           assert.isFunction(eventListeners[MediaConnectionEventNames.ICE_CONNECTION_STATE_CHANGED]);
           assert.isFunction(eventListeners[MediaConnectionEventNames.ICE_CANDIDATE]);
           assert.isFunction(eventListeners[MediaConnectionEventNames.ICE_CANDIDATE_ERROR]);
@@ -7569,28 +7625,44 @@ describe('plugin-meetings', () => {
           });
 
           it('should not collect skipped ice candidates error', () => {
-            eventListeners[MediaConnectionEventNames.ICE_CANDIDATE_ERROR]({error: { errorCode: 600, errorText: 'Address not associated with the desired network interface.' }});
+            eventListeners[MediaConnectionEventNames.ICE_CANDIDATE_ERROR]({
+              error: {
+                errorCode: 600,
+                errorText: 'Address not associated with the desired network interface.',
+              },
+            });
 
             assert.equal(meeting.iceCandidateErrors.size, 0);
           });
 
           it('should collect valid ice candidates error', () => {
-            eventListeners[MediaConnectionEventNames.ICE_CANDIDATE_ERROR]({error: { errorCode: 701, errorText: '' }});
+            eventListeners[MediaConnectionEventNames.ICE_CANDIDATE_ERROR]({
+              error: {errorCode: 701, errorText: ''},
+            });
 
             assert.equal(meeting.iceCandidateErrors.size, 1);
             assert.equal(meeting.iceCandidateErrors.has('701_'), true);
           });
 
           it('should increment counter if same valid ice candidates error collected', () => {
-            eventListeners[MediaConnectionEventNames.ICE_CANDIDATE_ERROR]({error: { errorCode: 701, errorText: '' }});
+            eventListeners[MediaConnectionEventNames.ICE_CANDIDATE_ERROR]({
+              error: {errorCode: 701, errorText: ''},
+            });
 
-            eventListeners[MediaConnectionEventNames.ICE_CANDIDATE_ERROR]({error: { errorCode: 701, errorText: 'STUN host lookup received error.' }});
-            eventListeners[MediaConnectionEventNames.ICE_CANDIDATE_ERROR]({error: { errorCode: 701, errorText: 'STUN host lookup received error.' }});
+            eventListeners[MediaConnectionEventNames.ICE_CANDIDATE_ERROR]({
+              error: {errorCode: 701, errorText: 'STUN host lookup received error.'},
+            });
+            eventListeners[MediaConnectionEventNames.ICE_CANDIDATE_ERROR]({
+              error: {errorCode: 701, errorText: 'STUN host lookup received error.'},
+            });
 
             assert.equal(meeting.iceCandidateErrors.size, 2);
             assert.equal(meeting.iceCandidateErrors.has('701_'), true);
             assert.equal(meeting.iceCandidateErrors.get('701_'), 1);
-            assert.equal(meeting.iceCandidateErrors.has('701_stun_host_lookup_received_error'), true);
+            assert.equal(
+              meeting.iceCandidateErrors.has('701_stun_host_lookup_received_error'),
+              true
+            );
             assert.equal(meeting.iceCandidateErrors.get('701_stun_host_lookup_received_error'), 2);
           });
         });
@@ -8263,8 +8335,12 @@ describe('plugin-meetings', () => {
           });
 
           it('registers for audio and video source count changed', () => {
-            assert.isFunction(eventListeners[MediaConnectionEventNames.VIDEO_SOURCES_COUNT_CHANGED]);
-            assert.isFunction(eventListeners[MediaConnectionEventNames.AUDIO_SOURCES_COUNT_CHANGED]);
+            assert.isFunction(
+              eventListeners[MediaConnectionEventNames.VIDEO_SOURCES_COUNT_CHANGED]
+            );
+            assert.isFunction(
+              eventListeners[MediaConnectionEventNames.AUDIO_SOURCES_COUNT_CHANGED]
+            );
           });
 
           it('forwards the VIDEO_SOURCES_COUNT_CHANGED event as "media:remoteVideoSourceCountChanged"', () => {