mirror of
https://github.com/spantaleev/matrix-docker-ansible-deploy.git
synced 2024-11-08 19:57:35 +01:00
492 lines
17 KiB
Django/Jinja
492 lines
17 KiB
Django/Jinja
/* eslint-disable no-unused-vars, no-var */
|
|
|
|
var config = {
|
|
// Configuration
|
|
//
|
|
|
|
// Alternative location for the configuration.
|
|
// configLocation: './config.json',
|
|
|
|
// Custom function which given the URL path should return a room name.
|
|
// getroomnode: function (path) { return 'someprefixpossiblybasedonpath'; },
|
|
|
|
|
|
// Connection
|
|
//
|
|
|
|
hosts: {
|
|
// XMPP domain.
|
|
domain: '{{ matrix_jitsi_xmpp_domain }}',
|
|
|
|
{% if matrix_jitsi_enable_auth %}
|
|
{% if matrix_jitsi_enable_guests %}
|
|
// When using authentication, domain for guest users.
|
|
anonymousdomain: '{{ matrix_jitsi_xmpp_guest_domain }}',
|
|
{% endif %}
|
|
|
|
// Domain for authenticated users. Defaults to <domain>.
|
|
authdomain: '{{ matrix_jitsi_xmpp_domain }}',
|
|
{% endif %}
|
|
|
|
// Jirecon recording component domain.
|
|
// jirecon: 'jirecon.{{ matrix_jitsi_xmpp_domain }}',
|
|
|
|
// Call control component (Jigasi).
|
|
// call_control: 'callcontrol.{{ matrix_jitsi_xmpp_domain }}',
|
|
|
|
// Focus component domain. Defaults to focus.<domain>.
|
|
// focus: 'focus.{{ matrix_jitsi_xmpp_domain }}',
|
|
|
|
// XMPP MUC domain. FIXME: use XEP-0030 to discover it.
|
|
muc: {{ matrix_jitsi_xmpp_muc_domain|to_json }},
|
|
},
|
|
|
|
// BOSH URL. FIXME: use XEP-0156 to discover it.
|
|
bosh: '/http-bind',
|
|
|
|
// The name of client node advertised in XEP-0115 'c' stanza
|
|
clientNode: 'http://jitsi.org/jitsimeet',
|
|
|
|
// The real JID of focus participant - can be overridden here
|
|
focusUserJid: {{ matrix_jitsi_jicofo_auth_user|to_json }} + '@' + {{ matrix_jitsi_xmpp_auth_domain|to_json }},
|
|
|
|
|
|
// Testing / experimental features.
|
|
//
|
|
|
|
testing: {
|
|
// Enables experimental simulcast support on Firefox.
|
|
enableFirefoxSimulcast: {{ matrix_jitsi_web_config_testing_enableFirefoxSimulcast|to_json }},
|
|
|
|
// P2P test mode disables automatic switching to P2P when there are 2
|
|
// participants in the conference.
|
|
p2pTestMode: {{ matrix_jitsi_web_config_testing_p2pTestMode|to_json }}
|
|
|
|
// Enables the test specific features consumed by jitsi-meet-torture
|
|
// testMode: false
|
|
},
|
|
|
|
// Disables ICE/UDP by filtering out local and remote UDP candidates in
|
|
// signalling.
|
|
// webrtcIceUdpDisable: false,
|
|
|
|
// Disables ICE/TCP by filtering out local and remote TCP candidates in
|
|
// signalling.
|
|
// webrtcIceTcpDisable: false,
|
|
|
|
|
|
// Media
|
|
//
|
|
|
|
// Audio
|
|
|
|
// Disable measuring of audio levels.
|
|
disableAudioLevels: {{ matrix_jitsi_web_config_disableAudioLevels|to_json }},
|
|
|
|
// Start the conference in audio only mode (no video is being received nor
|
|
// sent).
|
|
// startAudioOnly: false,
|
|
|
|
// Every participant after the Nth will start audio muted.
|
|
// startAudioMuted: 10,
|
|
|
|
// Start calls with audio muted. Unlike the option above, this one is only
|
|
// applied locally. FIXME: having these 2 options is confusing.
|
|
// startWithAudioMuted: false,
|
|
startWithAudioMuted: {{ matrix_jitsi_web_config_start_with_audio_muted|to_json }},
|
|
|
|
// Enabling it (with #params) will disable local audio output of remote
|
|
// participants and to enable it back a reload is needed.
|
|
// startSilent: false
|
|
|
|
// Video
|
|
|
|
// Sets the preferred resolution (height) for local video. Defaults to 720.
|
|
// resolution: 720,
|
|
|
|
// w3c spec-compliant video constraints to use for video capture. Currently
|
|
// used by browsers that return true from lib-jitsi-meet's
|
|
// util#browser#usesNewGumFlow. The constraints are independency from
|
|
// this config's resolution value. Defaults to requesting an ideal aspect
|
|
// ratio of 16:9 with an ideal resolution of 720.
|
|
{% if matrix_jitsi_web_config_constraints_enabled %}
|
|
constraints: {
|
|
video: {
|
|
aspectRatio: {{ matrix_jitsi_web_config_constraints_video_aspectRatio }},
|
|
height: {
|
|
ideal: {{ matrix_jitsi_web_config_constraints_video_height_ideal|to_json }},
|
|
max: {{ matrix_jitsi_web_config_constraints_video_height_max|to_json }},
|
|
min: {{ matrix_jitsi_web_config_constraints_video_height_min|to_json }}
|
|
}
|
|
}
|
|
},
|
|
{% endif %}
|
|
// Enable / disable simulcast support.
|
|
// disableSimulcast: false,
|
|
|
|
// Enable / disable layer suspension. If enabled, endpoints whose HD
|
|
// layers are not in use will be suspended (no longer sent) until they
|
|
// are requested again.
|
|
enableLayerSuspension: {{ matrix_jitsi_web_config_enableLayerSuspension|to_json }},
|
|
|
|
// Suspend sending video if bandwidth estimation is too low. This may cause
|
|
// problems with audio playback. Disabled until these are fixed.
|
|
disableSuspendVideo: true,
|
|
|
|
// Every participant after the Nth will start video muted.
|
|
// startVideoMuted: 10,
|
|
|
|
// Start calls with video muted. Unlike the option above, this one is only
|
|
// applied locally. FIXME: having these 2 options is confusing.
|
|
// startWithVideoMuted: false,
|
|
startWithVideoMuted: {{ matrix_jitsi_web_config_start_with_video_muted|to_json }},
|
|
|
|
// If set to true, prefer to use the H.264 video codec (if supported).
|
|
// Note that it's not recommended to do this because simulcast is not
|
|
// supported when using H.264. For 1-to-1 calls this setting is enabled by
|
|
// default and can be toggled in the p2p section.
|
|
// preferH264: true,
|
|
|
|
// If set to true, disable H.264 video codec by stripping it out of the
|
|
// SDP.
|
|
// disableH264: false,
|
|
|
|
// Desktop sharing
|
|
|
|
// The ID of the jidesha extension for Chrome.
|
|
desktopSharingChromeExtId: null,
|
|
|
|
// Whether desktop sharing should be disabled on Chrome.
|
|
// desktopSharingChromeDisabled: false,
|
|
|
|
// The media sources to use when using screen sharing with the Chrome
|
|
// extension.
|
|
desktopSharingChromeSources: [ 'screen', 'window', 'tab' ],
|
|
|
|
// Required version of Chrome extension
|
|
desktopSharingChromeMinExtVersion: '0.1',
|
|
|
|
// Whether desktop sharing should be disabled on Firefox.
|
|
// desktopSharingFirefoxDisabled: false,
|
|
|
|
// Optional desktop sharing frame rate options. Default value: min:5, max:5.
|
|
// desktopSharingFrameRate: {
|
|
// min: 5,
|
|
// max: 5
|
|
// },
|
|
|
|
// Try to start calls with screen-sharing instead of camera video.
|
|
// startScreenSharing: false,
|
|
|
|
// Recording
|
|
hiddenDomain: {{ matrix_jitsi_recorder_domain|to_json }},
|
|
|
|
// Whether to enable file recording or not.
|
|
fileRecordingsEnabled: {{ matrix_jitsi_enable_recording|to_json }},
|
|
// Enable the dropbox integration.
|
|
// dropbox: {
|
|
// appKey: '<APP_KEY>' // Specify your app key here.
|
|
// // A URL to redirect the user to, after authenticating
|
|
// // by default uses:
|
|
// // 'https://{{ matrix_jitsi_xmpp_domain }}/static/oauth.html'
|
|
// redirectURI:
|
|
// 'https://{{ matrix_jitsi_xmpp_domain }}/subfolder/static/oauth.html'
|
|
// },
|
|
// When integrations like dropbox are enabled only that will be shown,
|
|
// by enabling fileRecordingsServiceEnabled, we show both the integrations
|
|
// and the generic recording service (its configuration and storage type
|
|
// depends on jibri configuration)
|
|
// fileRecordingsServiceEnabled: false,
|
|
// Whether to show the possibility to share file recording with other people
|
|
// (e.g. meeting participants), based on the actual implementation
|
|
// on the backend.
|
|
// fileRecordingsServiceSharingEnabled: false,
|
|
|
|
// Whether to enable live streaming or not.
|
|
liveStreamingEnabled: {{ matrix_jitsi_enable_recording|to_json }},
|
|
|
|
// Transcription (in interface_config,
|
|
// subtitles and buttons can be configured)
|
|
transcribingEnabled: {{ matrix_jitsi_enable_transcriptions|to_json }},
|
|
|
|
// Misc
|
|
|
|
// Default value for the channel "last N" attribute. -1 for unlimited.
|
|
channelLastN: {{ matrix_jitsi_web_config_channelLastN|to_json }},
|
|
|
|
// Disables or enables RTX (RFC 4588) (defaults to false).
|
|
// disableRtx: false,
|
|
|
|
// Disables or enables TCC (the default is in Jicofo and set to true)
|
|
// (draft-holmer-rmcat-transport-wide-cc-extensions-01). This setting
|
|
// affects congestion control, it practically enables send-side bandwidth
|
|
// estimations.
|
|
// enableTcc: true,
|
|
|
|
// Disables or enables REMB (the default is in Jicofo and set to false)
|
|
// (draft-alvestrand-rmcat-remb-03). This setting affects congestion
|
|
// control, it practically enables recv-side bandwidth estimations. When
|
|
// both TCC and REMB are enabled, TCC takes precedence. When both are
|
|
// disabled, then bandwidth estimations are disabled.
|
|
// enableRemb: false,
|
|
|
|
// Defines the minimum number of participants to start a call (the default
|
|
// is set in Jicofo and set to 2).
|
|
// minParticipants: 2,
|
|
|
|
// Use XEP-0215 to fetch STUN and TURN servers.
|
|
// useStunTurn: true,
|
|
|
|
// Enable IPv6 support.
|
|
// useIPv6: true,
|
|
|
|
// Enables / disables a data communication channel with the Videobridge.
|
|
// Values can be 'datachannel', 'websocket', true (treat it as
|
|
// 'datachannel'), undefined (treat it as 'datachannel') and false (don't
|
|
// open any channel).
|
|
// openBridgeChannel: true,
|
|
|
|
|
|
// UI
|
|
//
|
|
|
|
// Use display name as XMPP nickname.
|
|
// useNicks: false,
|
|
|
|
// Require users to always specify a display name.
|
|
// requireDisplayName: true,
|
|
|
|
// Whether to use a welcome page or not. In case it's false a random room
|
|
// will be joined when no room is specified.
|
|
enableWelcomePage: true,
|
|
|
|
// Enabling the close page will ignore the welcome page redirection when
|
|
// a call is hangup.
|
|
// enableClosePage: false,
|
|
|
|
// Disable hiding of remote thumbnails when in a 1-on-1 conference call.
|
|
// disable1On1Mode: false,
|
|
|
|
// Default language for the user interface.
|
|
defaultLanguage: {{ matrix_jitsi_web_config_defaultLanguage|to_json }},
|
|
|
|
// If true all users without a token will be considered guests and all users
|
|
// with token will be considered non-guests. Only guests will be allowed to
|
|
// edit their profile.
|
|
enableUserRolesBasedOnToken: false,
|
|
|
|
// Whether or not some features are checked based on token.
|
|
// enableFeaturesBasedOnToken: false,
|
|
|
|
// Enable lock room for all moderators, even when userRolesBasedOnToken is enabled and participants are guests.
|
|
// lockRoomGuestEnabled: false,
|
|
|
|
// When enabled the password used for locking a room is restricted to up to the number of digits specified
|
|
// roomPasswordNumberOfDigits: 10,
|
|
// default: roomPasswordNumberOfDigits: false,
|
|
|
|
// Message to show the users. Example: 'The service will be down for
|
|
// maintenance at 01:00 AM GMT,
|
|
// noticeMessage: '',
|
|
|
|
// Enables calendar integration, depends on googleApiApplicationClientID
|
|
// and microsoftApiApplicationClientID
|
|
// enableCalendarIntegration: false,
|
|
|
|
// Stats
|
|
//
|
|
|
|
// Whether to enable stats collection or not in the TraceablePeerConnection.
|
|
// This can be useful for debugging purposes (post-processing/analysis of
|
|
// the webrtc stats) as it is done in the jitsi-meet-torture bandwidth
|
|
// estimation tests.
|
|
// gatherStats: false,
|
|
|
|
// To enable sending statistics to callstats.io you must provide the
|
|
// Application ID and Secret.
|
|
// callStatsID: '',
|
|
// callStatsSecret: '',
|
|
|
|
// enables callstatsUsername to be reported as statsId and used
|
|
// by callstats as repoted remote id
|
|
// enableStatsID: false
|
|
|
|
// enables sending participants display name to callstats
|
|
// enableDisplayNameInStats: false
|
|
|
|
|
|
// Privacy
|
|
//
|
|
|
|
// If third party requests are disabled, no other server will be contacted.
|
|
// This means avatars will be locally generated and callstats integration
|
|
// will not function.
|
|
// disableThirdPartyRequests: false,
|
|
|
|
|
|
// Peer-To-Peer mode: used (if enabled) when there are just 2 participants.
|
|
//
|
|
|
|
p2p: {
|
|
// Enables peer to peer mode. When enabled the system will try to
|
|
// establish a direct connection when there are exactly 2 participants
|
|
// in the room. If that succeeds the conference will stop sending data
|
|
// through the JVB and use the peer to peer connection instead. When a
|
|
// 3rd participant joins the conference will be moved back to the JVB
|
|
// connection.
|
|
enabled: true,
|
|
|
|
// Use XEP-0215 to fetch STUN and TURN servers.
|
|
// useStunTurn: true,
|
|
|
|
// The STUN servers that will be used in the peer to peer connections
|
|
{% if matrix_jitsi_web_stun_servers|length > 0 %}
|
|
stunServers: [
|
|
{% for url in matrix_jitsi_web_stun_servers %}
|
|
{ urls: {{ url|to_json }} }{% if not loop.last %},{% endif %}
|
|
{% endfor %}
|
|
],
|
|
{% endif %}
|
|
|
|
// Sets the ICE transport policy for the p2p connection. At the time
|
|
// of this writing the list of possible values are 'all' and 'relay',
|
|
// but that is subject to change in the future. The enum is defined in
|
|
// the WebRTC standard:
|
|
// https://www.w3.org/TR/webrtc/#rtcicetransportpolicy-enum.
|
|
// If not set, the effective value is 'all'.
|
|
// iceTransportPolicy: 'all',
|
|
|
|
// If set to true, it will prefer to use H.264 for P2P calls (if H.264
|
|
// is supported).
|
|
preferH264: true
|
|
|
|
// If set to true, disable H.264 video codec by stripping it out of the
|
|
// SDP.
|
|
// disableH264: false,
|
|
|
|
// How long we're going to wait, before going back to P2P after the 3rd
|
|
// participant has left the conference (to filter out page reload).
|
|
// backToP2PDelay: 5
|
|
},
|
|
|
|
analytics: {
|
|
// The Google Analytics Tracking ID:
|
|
// googleAnalyticsTrackingId: 'your-tracking-id-UA-123456-1'
|
|
|
|
// The Amplitude APP Key:
|
|
// amplitudeAPPKey: '<APP_KEY>'
|
|
|
|
// Array of script URLs to load as lib-jitsi-meet "analytics handlers".
|
|
// scriptURLs: [
|
|
// "libs/analytics-ga.min.js", // google-analytics
|
|
// "https://example.com/my-custom-analytics.js"
|
|
// ],
|
|
},
|
|
|
|
// Information about the jitsi-meet instance we are connecting to, including
|
|
// the user region as seen by the server.
|
|
deploymentInfo: {
|
|
// shard: "shard1",
|
|
// region: "europe",
|
|
// userRegion: "asia"
|
|
}
|
|
|
|
// Local Recording
|
|
//
|
|
|
|
// localRecording: {
|
|
// Enables local recording.
|
|
// Additionally, 'localrecording' (all lowercase) needs to be added to
|
|
// TOOLBAR_BUTTONS in interface_config.js for the Local Recording
|
|
// button to show up on the toolbar.
|
|
//
|
|
// enabled: true,
|
|
//
|
|
|
|
// The recording format, can be one of 'ogg', 'flac' or 'wav'.
|
|
// format: 'flac'
|
|
//
|
|
|
|
// }
|
|
|
|
// Options related to end-to-end (participant to participant) ping.
|
|
// e2eping: {
|
|
// // The interval in milliseconds at which pings will be sent.
|
|
// // Defaults to 10000, set to <= 0 to disable.
|
|
// pingInterval: 10000,
|
|
//
|
|
// // The interval in milliseconds at which analytics events
|
|
// // with the measured RTT will be sent. Defaults to 60000, set
|
|
// // to <= 0 to disable.
|
|
// analyticsInterval: 60000,
|
|
// }
|
|
|
|
// If set, will attempt to use the provided video input device label when
|
|
// triggering a screenshare, instead of proceeding through the normal flow
|
|
// for obtaining a desktop stream.
|
|
// NOTE: This option is experimental and is currently intended for internal
|
|
// use only.
|
|
// _desktopSharingSourceDevice: 'sample-id-or-label'
|
|
|
|
// If true, any checks to handoff to another application will be prevented
|
|
// and instead the app will continue to display in the current browser.
|
|
// disableDeepLinking: false
|
|
|
|
// A property to disable the right click context menu for localVideo
|
|
// the menu has option to flip the locally seen video for local presentations
|
|
// disableLocalVideoFlip: false
|
|
|
|
// List of undocumented settings used in jitsi-meet
|
|
/**
|
|
_immediateReloadThreshold
|
|
autoRecord
|
|
autoRecordToken
|
|
debug
|
|
debugAudioLevels
|
|
deploymentInfo
|
|
dialInConfCodeUrl
|
|
dialInNumbersUrl
|
|
dialOutAuthUrl
|
|
dialOutCodesUrl
|
|
disableRemoteControl
|
|
displayJids
|
|
etherpad_base
|
|
externalConnectUrl
|
|
firefox_fake_device
|
|
googleApiApplicationClientID
|
|
iAmRecorder
|
|
iAmSipGateway
|
|
microsoftApiApplicationClientID
|
|
peopleSearchQueryTypes
|
|
peopleSearchUrl
|
|
requireDisplayName
|
|
tokenAuthUrl
|
|
*/
|
|
|
|
// List of undocumented settings used in lib-jitsi-meet
|
|
/**
|
|
_peerConnStatusOutOfLastNTimeout
|
|
_peerConnStatusRtcMuteTimeout
|
|
abTesting
|
|
avgRtpStatsN
|
|
callStatsConfIDNamespace
|
|
callStatsCustomScriptUrl
|
|
desktopSharingSources
|
|
disableAEC
|
|
disableAGC
|
|
disableAP
|
|
disableHPF
|
|
disableNS
|
|
enableLipSync
|
|
enableTalkWhileMuted
|
|
forceJVB121Ratio
|
|
hiddenDomain
|
|
ignoreStartMuted
|
|
nick
|
|
startBitrate
|
|
*/
|
|
|
|
};
|
|
|
|
/* eslint-enable no-unused-vars, no-var */
|