· 4 years ago · Aug 05, 2021, 10:42 AM
1/* eslint-disable no-unused-vars, no-var */
2
3var jitsiConnectionOptions = {
4 // Connection
5 //
6 serviceUrl: `https://${process.env.MIX_JITSI_DEPLOYMENT_DOMAIN}/http-bind`,
7
8 hosts: {
9 // XMPP domain.
10 domain: process.env.MIX_JITSI_DEPLOYMENT_DOMAIN,
11
12 // When using authentication, domain for guest users.
13 // anonymousdomain: 'guest.example.com',
14
15 // Domain for authenticated users. Defaults to <domain>.
16 // authdomain: 'meet.mazegeek.com',
17
18 // Call control component (Jigasi).
19 // call_control: 'callcontrol.meet.mazegeek.com',
20
21 // Focus component domain. Defaults to focus.<domain>.
22 // focus: 'focus.meet.mazegeek.com',
23
24 // XMPP MUC domain. FIXME: use XEP-0030 to discover it.
25 muc: `conference.${process.env.MIX_JITSI_DEPLOYMENT_DOMAIN}`
26 },
27
28 // BOSH URL. FIXME: use XEP-0156 to discover it.
29 bosh: `https://${process.env.MIX_JITSI_DEPLOYMENT_DOMAIN}/http-bind`,
30
31 // Websocket URL
32 // websocket: 'wss://meet.mazegeek.com/xmpp-websocket',
33
34 // The name of client node advertised in XEP-0115 'c' stanza
35 clientNode: 'https://jitsi.org/jitsimeet',
36
37 // The real JID of focus participant - can be overridden here
38 // Do not change username - FIXME: Make focus username configurable
39 // https://github.com/jitsi/jitsi-meet/issues/7376
40 // focusUserJid: 'focus@auth.meet.mazegeek.com',
41
42
43 // Testing / experimental features.
44 //
45
46 testing: {
47 // Disables the End to End Encryption feature. Useful for debugging
48 // issues related to insertable streams.
49 // disableE2EE: false,
50
51 // P2P test mode disables automatic switching to P2P when there are 2
52 // participants in the conference.
53 p2pTestMode: false
54
55 // Enables the test specific features consumed by jitsi-meet-torture
56 // testMode: false
57
58 // Disables the auto-play behavior of *all* newly created video element.
59 // This is useful when the client runs on a host with limited resources.
60 // noAutoPlayVideo: false
61
62 // Enable / disable 500 Kbps bitrate cap on desktop tracks. When enabled,
63 // simulcast is turned off for the desktop share. If presenter is turned
64 // on while screensharing is in progress, the max bitrate is automatically
65 // adjusted to 2.5 Mbps. This takes a value between 0 and 1 which determines
66 // the probability for this to be enabled.
67 // capScreenshareBitrate: 1 // 0 to disable
68
69 // Enable callstats only for a percentage of users.
70 // This takes a value between 0 and 100. A hash value is calculated using the
71 // conference name and callstats is enabled if the hash value is less than the
72 // the threshold specified here.
73 // callStatsThreshold: 5 // enable callstats for 5% of the conferences.
74 },
75
76 // Disables ICE/UDP by filtering out local and remote UDP candidates in
77 // signalling.
78 // webrtcIceUdpDisable: false,
79
80 // Disables ICE/TCP by filtering out local and remote TCP candidates in
81 // signalling.
82 // webrtcIceTcpDisable: false,
83
84
85 // Media
86 //
87
88 // Audio
89
90 // Disable measuring of audio levels.
91 // disableAudioLevels: false,
92 // audioLevelsInterval: 200,
93
94 // Enabling this will run the lib-jitsi-meet no audio detection module which
95 // will notify the user if the current selected microphone has no audio
96 // input and will suggest another valid device if one is present.
97 enableNoAudioDetection: true,
98
99 // Enabling this will run the lib-jitsi-meet noise detection module which will
100 // notify the user if there is noise, other than voice, coming from the current
101 // selected microphone. The purpose it to let the user know that the input could
102 // be potentially unpleasant for other meeting participants.
103 enableNoisyMicDetection: true,
104
105 // Start the conference in audio only mode (no video is being received nor
106 // sent).
107 // startAudioOnly: false,
108
109 // Every participant after the Nth will start audio muted.
110 // startAudioMuted: 10,
111
112 // Start calls with audio muted. Unlike the option above, this one is only
113 // applied locally. FIXME: having these 2 options is confusing.
114 // startWithAudioMuted: false,
115
116 // Enabling it (with #params) will disable local audio output of remote
117 // participants and to enable it back a reload is needed.
118 // startSilent: false
119
120 // Sets the preferred target bitrate for the Opus audio codec by setting its
121 // 'maxaveragebitrate' parameter. Currently not available in p2p mode.
122 // Valid values are in the range 6000 to 510000
123 // opusMaxAverageBitrate: 20000,
124
125 // Enables redundancy for Opus
126 // enableOpusRed: false
127
128 // Video
129
130 // Sets the preferred resolution (height) for local video. Defaults to 720.
131 // resolution: 720,
132
133 // How many participants while in the tile view mode, before the receiving video quality is reduced from HD to SD.
134 // Use -1 to disable.
135 // maxFullResolutionParticipants: 2,
136
137 // w3c spec-compliant video constraints to use for video capture. Currently
138 // used by browsers that return true from lib-jitsi-meet's
139 // util#browser#usesNewGumFlow. The constraints are independent from
140 // this config's resolution value. Defaults to requesting an ideal
141 // resolution of 720p.
142 // constraints: {
143 // video: {
144 // height: {
145 // ideal: 720,
146 // max: 720,
147 // min: 240
148 // }
149 // }
150 // },
151
152 // Enable / disable simulcast support.
153 // disableSimulcast: false,
154
155 // Enable / disable layer suspension. If enabled, endpoints whose HD
156 // layers are not in use will be suspended (no longer sent) until they
157 // are requested again.
158 // enableLayerSuspension: false,
159
160 // Every participant after the Nth will start video muted.
161 // startVideoMuted: 10,
162
163 // Start calls with video muted. Unlike the option above, this one is only
164 // applied locally. FIXME: having these 2 options is confusing.
165 // startWithVideoMuted: false,
166
167 // If set to true, prefer to use the H.264 video codec (if supported).
168 // Note that it's not recommended to do this because simulcast is not
169 // supported when using H.264. For 1-to-1 calls this setting is enabled by
170 // default and can be toggled in the p2p section.
171 // This option has been deprecated, use preferredCodec under videoQuality section instead.
172 // preferH264: true,
173
174 // If set to true, disable H.264 video codec by stripping it out of the
175 // SDP.
176 // disableH264: false,
177
178 // Desktop sharing
179
180 // Optional desktop sharing frame rate options. Default value: min:5, max:5.
181 // desktopSharingFrameRate: {
182 // min: 5,
183 // max: 5
184 // },
185
186 // Try to start calls with screen-sharing instead of camera video.
187 // startScreenSharing: false,
188
189 // Recording
190
191 // Whether to enable file recording or not.
192 // fileRecordingsEnabled: false,
193 // Enable the dropbox integration.
194 // dropbox: {
195 // appKey: '<APP_KEY>' // Specify your app key here.
196 // // A URL to redirect the user to, after authenticating
197 // // by default uses:
198 // // 'https://meet.mazegeek.com/static/oauth.html'
199 // redirectURI:
200 // 'https://meet.mazegeek.com/subfolder/static/oauth.html'
201 // },
202 // When integrations like dropbox are enabled only that will be shown,
203 // by enabling fileRecordingsServiceEnabled, we show both the integrations
204 // and the generic recording service (its configuration and storage type
205 // depends on jibri configuration)
206 // fileRecordingsServiceEnabled: false,
207 // Whether to show the possibility to share file recording with other people
208 // (e.g. meeting participants), based on the actual implementation
209 // on the backend.
210 // fileRecordingsServiceSharingEnabled: false,
211
212 // Whether to enable live streaming or not.
213 // liveStreamingEnabled: false,
214
215 // Transcription (in interface_config,
216 // subtitles and buttons can be configured)
217 // transcribingEnabled: false,
218
219 // Enables automatic turning on captions when recording is started
220 // autoCaptionOnRecord: false,
221
222 // Misc
223
224 // Default value for the channel "last N" attribute. -1 for unlimited.
225 channelLastN: -1,
226
227 // Provides a way to use different "last N" values based on the number of participants in the conference.
228 // The keys in an Object represent number of participants and the values are "last N" to be used when number of
229 // participants gets to or above the number.
230 //
231 // For the given example mapping, "last N" will be set to 20 as long as there are at least 5, but less than
232 // 29 participants in the call and it will be lowered to 15 when the 30th participant joins. The 'channelLastN'
233 // will be used as default until the first threshold is reached.
234 //
235 // lastNLimits: {
236 // 5: 20,
237 // 30: 15,
238 // 50: 10,
239 // 70: 5,
240 // 90: 2
241 // },
242
243 // Specify the settings for video quality optimizations on the client.
244 // videoQuality: {
245 // // Provides a way to prevent a video codec from being negotiated on the JVB connection. The codec specified
246 // // here will be removed from the list of codecs present in the SDP answer generated by the client. If the
247 // // same codec is specified for both the disabled and preferred option, the disable settings will prevail.
248 // // Note that 'VP8' cannot be disabled since it's a mandatory codec, the setting will be ignored in this case.
249 // disabledCodec: 'H264',
250 //
251 // // Provides a way to set a preferred video codec for the JVB connection. If 'H264' is specified here,
252 // // simulcast will be automatically disabled since JVB doesn't support H264 simulcast yet. This will only
253 // // rearrange the the preference order of the codecs in the SDP answer generated by the browser only if the
254 // // preferred codec specified here is present. Please ensure that the JVB offers the specified codec for this
255 // // to take effect.
256 // preferredCodec: 'VP8',
257 //
258 // // Provides a way to configure the maximum bitrates that will be enforced on the simulcast streams for
259 // // video tracks. The keys in the object represent the type of the stream (LD, SD or HD) and the values
260 // // are the max.bitrates to be set on that particular type of stream. The actual send may vary based on
261 // // the available bandwidth calculated by the browser, but it will be capped by the values specified here.
262 // // This is currently not implemented on app based clients on mobile.
263 // maxBitratesVideo: {
264 // low: 200000,
265 // standard: 500000,
266 // high: 1500000
267 // },
268 //
269 // // The options can be used to override default thresholds of video thumbnail heights corresponding to
270 // // the video quality levels used in the application. At the time of this writing the allowed levels are:
271 // // 'low' - for the low quality level (180p at the time of this writing)
272 // // 'standard' - for the medium quality level (360p)
273 // // 'high' - for the high quality level (720p)
274 // // The keys should be positive numbers which represent the minimal thumbnail height for the quality level.
275 // //
276 // // With the default config value below the application will use 'low' quality until the thumbnails are
277 // // at least 360 pixels tall. If the thumbnail height reaches 720 pixels then the application will switch to
278 // // the high quality.
279 // minHeightForQualityLvl: {
280 // 360: 'standard,
281 // 720: 'high'
282 // }
283 // },
284
285 // // Options for the recording limit notification.
286 // recordingLimit: {
287 //
288 // // The recording limit in minutes. Note: This number appears in the notification text
289 // // but doesn't enforce the actual recording time limit. This should be configured in
290 // // jibri!
291 // limit: 60,
292 //
293 // // The name of the app with unlimited recordings.
294 // appName: 'Unlimited recordings APP',
295 //
296 // // The URL of the app with unlimited recordings.
297 // appURL: 'https://unlimited.recordings.app.com/'
298 // },
299
300 // Disables or enables RTX (RFC 4588) (defaults to false).
301 // disableRtx: false,
302
303 // Disables or enables TCC (the default is in Jicofo and set to true)
304 // (draft-holmer-rmcat-transport-wide-cc-extensions-01). This setting
305 // affects congestion control, it practically enables send-side bandwidth
306 // estimations.
307 // enableTcc: true,
308
309 // Disables or enables REMB (the default is in Jicofo and set to false)
310 // (draft-alvestrand-rmcat-remb-03). This setting affects congestion
311 // control, it practically enables recv-side bandwidth estimations. When
312 // both TCC and REMB are enabled, TCC takes precedence. When both are
313 // disabled, then bandwidth estimations are disabled.
314 // enableRemb: false,
315
316 // Enables ICE restart logic in LJM and displays the page reload overlay on
317 // ICE failure. Current disabled by default because it's causing issues with
318 // signaling when Octo is enabled. Also when we do an "ICE restart"(which is
319 // not a real ICE restart), the client maintains the TCC sequence number
320 // counter, but the bridge resets it. The bridge sends media packets with
321 // TCC sequence numbers starting from 0.
322 // enableIceRestart: false,
323
324 // Defines the minimum number of participants to start a call (the default
325 // is set in Jicofo and set to 2).
326 // minParticipants: 2,
327
328 // Use the TURN servers discovered via XEP-0215 for the jitsi-videobridge
329 // connection
330 useStunTurn: true,
331
332 // Use TURN/UDP servers for the jitsi-videobridge connection (by default
333 // we filter out TURN/UDP because it is usually not needed since the
334 // bridge itself is reachable via UDP)
335 // useTurnUdp: false
336
337 // Enables / disables a data communication channel with the Videobridge.
338 // Values can be 'datachannel', 'websocket', true (treat it as
339 // 'datachannel'), undefined (treat it as 'datachannel') and false (don't
340 // open any channel).
341 // openBridgeChannel: true,
342
343
344 // UI
345 //
346
347 // Hides lobby button
348 // hideLobbyButton: false,
349
350 // Require users to always specify a display name.
351 // requireDisplayName: true,
352
353 // Whether to use a welcome page or not. In case it's false a random room
354 // will be joined when no room is specified.
355 enableWelcomePage: true,
356
357 // Enabling the close page will ignore the welcome page redirection when
358 // a call is hangup.
359 // enableClosePage: false,
360
361 // Disable hiding of remote thumbnails when in a 1-on-1 conference call.
362 // disable1On1Mode: false,
363
364 // Default language for the user interface.
365 // defaultLanguage: 'en',
366
367 // If true all users without a token will be considered guests and all users
368 // with token will be considered non-guests. Only guests will be allowed to
369 // edit their profile.
370 enableUserRolesBasedOnToken: false,
371
372 // Whether or not some features are checked based on token.
373 // enableFeaturesBasedOnToken: false,
374
375 // Enable lock room for all moderators, even when userRolesBasedOnToken is enabled and participants are guests.
376 // lockRoomGuestEnabled: false,
377
378 // When enabled the password used for locking a room is restricted to up to the number of digits specified
379 // roomPasswordNumberOfDigits: 10,
380 // default: roomPasswordNumberOfDigits: false,
381
382 // Message to show the users. Example: 'The service will be down for
383 // maintenance at 01:00 AM GMT,
384 // noticeMessage: '',
385
386 // Enables calendar integration, depends on googleApiApplicationClientID
387 // and microsoftApiApplicationClientID
388 // enableCalendarIntegration: false,
389
390 // When 'true', it shows an intermediate page before joining, where the user can configure their devices.
391 // prejoinPageEnabled: false,
392
393 // If true, shows the unsafe room name warning label when a room name is
394 // deemed unsafe (due to the simplicity in the name) and a password is not
395 // set or the lobby is not enabled.
396 // enableInsecureRoomNameWarning: false,
397
398 // Whether to automatically copy invitation URL after creating a room.
399 // Document should be focused for this option to work
400 // enableAutomaticUrlCopy: false,
401
402 // Stats
403 //
404
405 // Whether to enable stats collection or not in the TraceablePeerConnection.
406 // This can be useful for debugging purposes (post-processing/analysis of
407 // the webrtc stats) as it is done in the jitsi-meet-torture bandwidth
408 // estimation tests.
409 // gatherStats: false,
410
411 // The interval at which PeerConnection.getStats() is called. Defaults to 10000
412 // pcStatsInterval: 10000,
413
414 // To enable sending statistics to callstats.io you must provide the
415 // Application ID and Secret.
416 // callStatsID: '',
417 // callStatsSecret: '',
418
419 // Enables sending participants' display names to callstats
420 // enableDisplayNameInStats: false,
421
422 // Enables sending participants' emails (if available) to callstats and other analytics
423 // enableEmailInStats: false,
424
425 // Privacy
426 //
427
428 // If third party requests are disabled, no other server will be contacted.
429 // This means avatars will be locally generated and callstats integration
430 // will not function.
431 // disableThirdPartyRequests: false,
432
433
434 // Peer-To-Peer mode: used (if enabled) when there are just 2 participants.
435 //
436
437 p2p: {
438 // Enables peer to peer mode. When enabled the system will try to
439 // establish a direct connection when there are exactly 2 participants
440 // in the room. If that succeeds the conference will stop sending data
441 // through the JVB and use the peer to peer connection instead. When a
442 // 3rd participant joins the conference will be moved back to the JVB
443 // connection.
444 enabled: true,
445
446 // Use XEP-0215 to fetch STUN and TURN servers.
447 useStunTurn: true,
448
449 // The STUN servers that will be used in the peer to peer connections
450 stunServers: [
451
452 // { urls: 'stun:meet.mazegeek.com:3478' },
453 { urls: 'stun:meet-jit-si-turnrelay.jitsi.net:443' }
454 ]
455
456 // Sets the ICE transport policy for the p2p connection. At the time
457 // of this writing the list of possible values are 'all' and 'relay',
458 // but that is subject to change in the future. The enum is defined in
459 // the WebRTC standard:
460 // https://www.w3.org/TR/webrtc/#rtcicetransportpolicy-enum.
461 // If not set, the effective value is 'all'.
462 // iceTransportPolicy: 'all',
463
464 // If set to true, it will prefer to use H.264 for P2P calls (if H.264
465 // is supported). This setting is deprecated, use preferredCodec instead.
466 // preferH264: true
467
468 // Provides a way to set the video codec preference on the p2p connection. Acceptable
469 // codec values are 'VP8', 'VP9' and 'H264'.
470 // preferredCodec: 'H264',
471
472 // If set to true, disable H.264 video codec by stripping it out of the
473 // SDP. This setting is deprecated, use disabledCodec instead.
474 // disableH264: false,
475
476 // Provides a way to prevent a video codec from being negotiated on the p2p connection.
477 // disabledCodec: '',
478
479 // How long we're going to wait, before going back to P2P after the 3rd
480 // participant has left the conference (to filter out page reload).
481 // backToP2PDelay: 5
482 },
483
484 analytics: {
485 // The Google Analytics Tracking ID:
486 // googleAnalyticsTrackingId: 'your-tracking-id-UA-123456-1'
487
488 // Matomo configuration:
489 // matomoEndpoint: 'https://your-matomo-endpoint/',
490 // matomoSiteID: '42',
491
492 // The Amplitude APP Key:
493 // amplitudeAPPKey: '<APP_KEY>'
494
495 // Configuration for the rtcstats server:
496 // By enabling rtcstats server every time a conference is joined the rtcstats
497 // module connects to the provided rtcstatsEndpoint and sends statistics regarding
498 // PeerConnection states along with getStats metrics polled at the specified
499 // interval.
500 // rtcstatsEnabled: true,
501
502 // In order to enable rtcstats one needs to provide a endpoint url.
503 // rtcstatsEndpoint: wss://rtcstats-server-pilot.jitsi.net/,
504
505 // The interval at which rtcstats will poll getStats, defaults to 1000ms.
506 // If the value is set to 0 getStats won't be polled and the rtcstats client
507 // will only send data related to RTCPeerConnection events.
508 // rtcstatsPolIInterval: 1000
509
510 // Array of script URLs to load as lib-jitsi-meet "analytics handlers".
511 // scriptURLs: [
512 // "libs/analytics-ga.min.js", // google-analytics
513 // "https://example.com/my-custom-analytics.js"
514 // ],
515 },
516
517 // Logs that should go be passed through the 'log' event if a handler is defined for it
518 // apiLogLevels: ['warn', 'log', 'error', 'info', 'debug'],
519
520 // Information about the jitsi-meet instance we are connecting to, including
521 // the user region as seen by the server.
522 deploymentInfo: {
523 // shard: "shard1",
524 // region: "europe",
525 // userRegion: "asia"
526 },
527
528 // Decides whether the start/stop recording audio notifications should play on record.
529 // disableRecordAudioNotification: false,
530
531 // Information for the chrome extension banner
532 // chromeExtensionBanner: {
533 // // The chrome extension to be installed address
534 // url: 'https://chrome.google.com/webstore/detail/jitsi-meetings/kglhbbefdnlheedjiejgomgmfplipfeb',
535
536 // // Extensions info which allows checking if they are installed or not
537 // chromeExtensionsInfo: [
538 // {
539 // id: 'kglhbbefdnlheedjiejgomgmfplipfeb',
540 // path: 'jitsi-logo-48x48.png'
541 // }
542 // ]
543 // },
544
545 // Local Recording
546 //
547
548 // localRecording: {
549 // Enables local recording.
550 // Additionally, 'localrecording' (all lowercase) needs to be added to
551 // TOOLBAR_BUTTONS in interface_config.js for the Local Recording
552 // button to show up on the toolbar.
553 //
554 // enabled: true,
555 //
556
557 // The recording format, can be one of 'ogg', 'flac' or 'wav'.
558 // format: 'flac'
559 //
560
561 // },
562
563 // Options related to end-to-end (participant to participant) ping.
564 // e2eping: {
565 // // The interval in milliseconds at which pings will be sent.
566 // // Defaults to 10000, set to <= 0 to disable.
567 // pingInterval: 10000,
568 //
569 // // The interval in milliseconds at which analytics events
570 // // with the measured RTT will be sent. Defaults to 60000, set
571 // // to <= 0 to disable.
572 // analyticsInterval: 60000,
573 // },
574
575 // If set, will attempt to use the provided video input device label when
576 // triggering a screenshare, instead of proceeding through the normal flow
577 // for obtaining a desktop stream.
578 // NOTE: This option is experimental and is currently intended for internal
579 // use only.
580 // _desktopSharingSourceDevice: 'sample-id-or-label',
581
582 // If true, any checks to handoff to another application will be prevented
583 // and instead the app will continue to display in the current browser.
584 // disableDeepLinking: false,
585
586 // A property to disable the right click context menu for localVideo
587 // the menu has option to flip the locally seen video for local presentations
588 // disableLocalVideoFlip: false,
589
590 // Mainly privacy related settings
591
592 // Disables all invite functions from the app (share, invite, dial out...etc)
593 // disableInviteFunctions: true,
594
595 // Disables storing the room name to the recents list
596 // doNotStoreRoom: true,
597
598 // Deployment specific URLs.
599 // deploymentUrls: {
600 // // If specified a 'Help' button will be displayed in the overflow menu with a link to the specified URL for
601 // // user documentation.
602 // userDocumentationURL: 'https://docs.example.com/video-meetings.html',
603 // // If specified a 'Download our apps' button will be displayed in the overflow menu with a link
604 // // to the specified URL for an app download page.
605 // downloadAppsUrl: 'https://docs.example.com/our-apps.html'
606 // },
607
608 // Options related to the remote participant menu.
609 // remoteVideoMenu: {
610 // // If set to true the 'Kick out' button will be disabled.
611 // disableKick: true
612 // },
613
614 // If set to true all muting operations of remote participants will be disabled.
615 // disableRemoteMute: true,
616
617 /**
618 External API url used to receive branding specific information.
619 If there is no url set or there are missing fields, the defaults are applied.
620 None of the fields are mandatory and the response must have the shape:
621 {
622 // The hex value for the colour used as background
623 backgroundColor: '#fff',
624 // The url for the image used as background
625 backgroundImageUrl: 'https://example.com/background-img.png',
626 // The anchor url used when clicking the logo image
627 logoClickUrl: 'https://example-company.org',
628 // The url used for the image used as logo
629 logoImageUrl: 'https://example.com/logo-img.png'
630 }
631 */
632 // brandingDataUrl: '',
633
634 // The URL of the moderated rooms microservice, if available. If it
635 // is present, a link to the service will be rendered on the welcome page,
636 // otherwise the app doesn't render it.
637 // moderatedRoomServiceUrl: 'https://moderated.meet.mazegeek.com',
638
639 // List of undocumented settings used in jitsi-meet
640 /**
641 _immediateReloadThreshold
642 debug
643 debugAudioLevels
644 deploymentInfo
645 dialInConfCodeUrl
646 dialInNumbersUrl
647 dialOutAuthUrl
648 dialOutCodesUrl
649 disableRemoteControl
650 displayJids
651 etherpad_base
652 externalConnectUrl
653 firefox_fake_device
654 googleApiApplicationClientID
655 iAmRecorder
656 iAmSipGateway
657 microsoftApiApplicationClientID
658 peopleSearchQueryTypes
659 peopleSearchUrl
660 requireDisplayName
661 tokenAuthUrl
662 */
663
664 /**
665 * This property can be used to alter the generated meeting invite links (in combination with a branding domain
666 * which is retrieved internally by jitsi meet) (e.g. https://meet.mazegeek.com/someMeeting
667 * can become https://brandedDomain/roomAlias)
668 */
669 // brandingRoomAlias: null,
670
671 // List of undocumented settings used in lib-jitsi-meet
672 /**
673 _peerConnStatusOutOfLastNTimeout
674 _peerConnStatusRtcMuteTimeout
675 abTesting
676 avgRtpStatsN
677 callStatsConfIDNamespace
678 callStatsCustomScriptUrl
679 desktopSharingSources
680 disableAEC
681 disableAGC
682 disableAP
683 disableHPF
684 disableNS
685 enableLipSync
686 enableTalkWhileMuted
687 forceJVB121Ratio
688 hiddenDomain
689 ignoreStartMuted
690 nick
691 startBitrate
692 */
693
694
695 // Allow all above example options to include a trailing comma and
696 // prevent fear when commenting out the last value.
697 makeJsonParserHappy: 'even if last key had a trailing comma'
698
699 // no configuration value should follow this line.
700};
701
702/* eslint-enable no-unused-vars, no-var */
703
704export default jitsiConnectionOptions;