d605b219a2
We do this for 2 reasons: - so we can control things which are not controllable using environment variables (for example `stunServers` in jitsi/web, since we don't wish to use the hardcoded Google STUN servers if our own Coturn is enabled) - so playbook variable changes will properly rebuild the configuration. When using Jitsi environment variables, the configuration is only built once (the first time) and never rebuilt again. This is not the consistent with the rest of the playbook and with how Ansible operates. We're not perfect at it (yet), because we still let the Jitsi containers generate some files on their own, but we are closer and it should be good enough for most things. Related to #415 (Github Pull Request).
487 lines
16 KiB
Django/Jinja
487 lines
16 KiB
Django/Jinja
/* eslint-disable no-unused-vars, no-var */
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var config = {
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// Configuration
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//
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// Alternative location for the configuration.
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// configLocation: './config.json',
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// Custom function which given the URL path should return a room name.
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// getroomnode: function (path) { return 'someprefixpossiblybasedonpath'; },
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// Connection
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//
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hosts: {
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// XMPP domain.
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domain: '{{ matrix_jitsi_xmpp_domain }}',
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{% if matrix_jitsi_enable_guests %}
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// When using authentication, domain for guest users.
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anonymousdomain: 'guest.example.com',
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// Domain for authenticated users. Defaults to <domain>.
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authdomain: '{{ matrix_jitsi_xmpp_domain }}',
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{% endif %}
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// Jirecon recording component domain.
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// jirecon: 'jirecon.{{ matrix_jitsi_xmpp_domain }}',
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// Call control component (Jigasi).
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// call_control: 'callcontrol.{{ matrix_jitsi_xmpp_domain }}',
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// Focus component domain. Defaults to focus.<domain>.
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// focus: 'focus.{{ matrix_jitsi_xmpp_domain }}',
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// XMPP MUC domain. FIXME: use XEP-0030 to discover it.
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muc: {{ matrix_jitsi_xmpp_muc_domain|to_json }},
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},
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// BOSH URL. FIXME: use XEP-0156 to discover it.
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bosh: '/http-bind',
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// The name of client node advertised in XEP-0115 'c' stanza
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clientNode: 'http://jitsi.org/jitsimeet',
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// The real JID of focus participant - can be overridden here
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focusUserJid: {{ matrix_jitsi_jicofo_auth_user|to_json }} + '@' + {{ matrix_jitsi_xmpp_auth_domain|to_json }},
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// Testing / experimental features.
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//
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testing: {
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// Enables experimental simulcast support on Firefox.
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enableFirefoxSimulcast: false,
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// P2P test mode disables automatic switching to P2P when there are 2
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// participants in the conference.
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p2pTestMode: false
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// Enables the test specific features consumed by jitsi-meet-torture
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// testMode: false
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},
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// Disables ICE/UDP by filtering out local and remote UDP candidates in
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// signalling.
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// webrtcIceUdpDisable: false,
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// Disables ICE/TCP by filtering out local and remote TCP candidates in
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// signalling.
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// webrtcIceTcpDisable: false,
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// Media
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//
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// Audio
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// Disable measuring of audio levels.
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// disableAudioLevels: false,
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// Start the conference in audio only mode (no video is being received nor
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// sent).
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// startAudioOnly: false,
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// Every participant after the Nth will start audio muted.
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// startAudioMuted: 10,
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// Start calls with audio muted. Unlike the option above, this one is only
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// applied locally. FIXME: having these 2 options is confusing.
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// startWithAudioMuted: false,
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// Enabling it (with #params) will disable local audio output of remote
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// participants and to enable it back a reload is needed.
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// startSilent: false
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// Video
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// Sets the preferred resolution (height) for local video. Defaults to 720.
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// resolution: 720,
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// w3c spec-compliant video constraints to use for video capture. Currently
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// used by browsers that return true from lib-jitsi-meet's
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// util#browser#usesNewGumFlow. The constraints are independency from
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// this config's resolution value. Defaults to requesting an ideal aspect
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// ratio of 16:9 with an ideal resolution of 720.
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// constraints: {
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// video: {
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// aspectRatio: 16 / 9,
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// height: {
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// ideal: 720,
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// max: 720,
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// min: 240
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// }
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// }
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// },
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// Enable / disable simulcast support.
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// disableSimulcast: false,
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// Enable / disable layer suspension. If enabled, endpoints whose HD
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// layers are not in use will be suspended (no longer sent) until they
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// are requested again.
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// enableLayerSuspension: false,
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// Suspend sending video if bandwidth estimation is too low. This may cause
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// problems with audio playback. Disabled until these are fixed.
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disableSuspendVideo: true,
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// Every participant after the Nth will start video muted.
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// startVideoMuted: 10,
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// Start calls with video muted. Unlike the option above, this one is only
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// applied locally. FIXME: having these 2 options is confusing.
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// startWithVideoMuted: false,
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// If set to true, prefer to use the H.264 video codec (if supported).
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// Note that it's not recommended to do this because simulcast is not
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// supported when using H.264. For 1-to-1 calls this setting is enabled by
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// default and can be toggled in the p2p section.
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// preferH264: true,
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// If set to true, disable H.264 video codec by stripping it out of the
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// SDP.
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// disableH264: false,
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// Desktop sharing
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// The ID of the jidesha extension for Chrome.
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desktopSharingChromeExtId: null,
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// Whether desktop sharing should be disabled on Chrome.
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// desktopSharingChromeDisabled: false,
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// The media sources to use when using screen sharing with the Chrome
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// extension.
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desktopSharingChromeSources: [ 'screen', 'window', 'tab' ],
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// Required version of Chrome extension
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desktopSharingChromeMinExtVersion: '0.1',
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// Whether desktop sharing should be disabled on Firefox.
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// desktopSharingFirefoxDisabled: false,
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// Optional desktop sharing frame rate options. Default value: min:5, max:5.
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// desktopSharingFrameRate: {
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// min: 5,
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// max: 5
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// },
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// Try to start calls with screen-sharing instead of camera video.
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// startScreenSharing: false,
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// Recording
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hiddenDomain: {{ matrix_jitsi_recorder_domain|to_json }},
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// Whether to enable file recording or not.
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fileRecordingsEnabled: {{ matrix_jitsi_enable_recording|to_json }},
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// Enable the dropbox integration.
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// dropbox: {
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// appKey: '<APP_KEY>' // Specify your app key here.
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// // A URL to redirect the user to, after authenticating
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// // by default uses:
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// // 'https://{{ matrix_jitsi_xmpp_domain }}/static/oauth.html'
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// redirectURI:
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// 'https://{{ matrix_jitsi_xmpp_domain }}/subfolder/static/oauth.html'
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// },
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// When integrations like dropbox are enabled only that will be shown,
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// by enabling fileRecordingsServiceEnabled, we show both the integrations
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// and the generic recording service (its configuration and storage type
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// depends on jibri configuration)
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// fileRecordingsServiceEnabled: false,
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// Whether to show the possibility to share file recording with other people
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// (e.g. meeting participants), based on the actual implementation
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// on the backend.
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// fileRecordingsServiceSharingEnabled: false,
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// Whether to enable live streaming or not.
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liveStreamingEnabled: {{ matrix_jitsi_enable_recording|to_json }},
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// Transcription (in interface_config,
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// subtitles and buttons can be configured)
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transcribingEnabled: {{ matrix_jitsi_enable_transcriptions|to_json }},
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// Misc
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// Default value for the channel "last N" attribute. -1 for unlimited.
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channelLastN: -1,
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// Disables or enables RTX (RFC 4588) (defaults to false).
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// disableRtx: false,
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// Disables or enables TCC (the default is in Jicofo and set to true)
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// (draft-holmer-rmcat-transport-wide-cc-extensions-01). This setting
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// affects congestion control, it practically enables send-side bandwidth
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// estimations.
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// enableTcc: true,
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// Disables or enables REMB (the default is in Jicofo and set to false)
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// (draft-alvestrand-rmcat-remb-03). This setting affects congestion
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// control, it practically enables recv-side bandwidth estimations. When
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// both TCC and REMB are enabled, TCC takes precedence. When both are
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// disabled, then bandwidth estimations are disabled.
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// enableRemb: false,
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// Defines the minimum number of participants to start a call (the default
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// is set in Jicofo and set to 2).
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// minParticipants: 2,
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// Use XEP-0215 to fetch STUN and TURN servers.
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// useStunTurn: true,
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// Enable IPv6 support.
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// useIPv6: true,
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// Enables / disables a data communication channel with the Videobridge.
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// Values can be 'datachannel', 'websocket', true (treat it as
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// 'datachannel'), undefined (treat it as 'datachannel') and false (don't
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// open any channel).
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// openBridgeChannel: true,
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// UI
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//
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// Use display name as XMPP nickname.
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// useNicks: false,
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// Require users to always specify a display name.
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// requireDisplayName: true,
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// Whether to use a welcome page or not. In case it's false a random room
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// will be joined when no room is specified.
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enableWelcomePage: true,
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// Enabling the close page will ignore the welcome page redirection when
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// a call is hangup.
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// enableClosePage: false,
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// Disable hiding of remote thumbnails when in a 1-on-1 conference call.
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// disable1On1Mode: false,
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// Default language for the user interface.
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// defaultLanguage: 'en',
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// If true all users without a token will be considered guests and all users
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// with token will be considered non-guests. Only guests will be allowed to
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// edit their profile.
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enableUserRolesBasedOnToken: false,
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// Whether or not some features are checked based on token.
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// enableFeaturesBasedOnToken: false,
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// Enable lock room for all moderators, even when userRolesBasedOnToken is enabled and participants are guests.
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// lockRoomGuestEnabled: false,
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// When enabled the password used for locking a room is restricted to up to the number of digits specified
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// roomPasswordNumberOfDigits: 10,
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// default: roomPasswordNumberOfDigits: false,
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// Message to show the users. Example: 'The service will be down for
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// maintenance at 01:00 AM GMT,
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// noticeMessage: '',
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// Enables calendar integration, depends on googleApiApplicationClientID
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// and microsoftApiApplicationClientID
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// enableCalendarIntegration: false,
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// Stats
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//
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// Whether to enable stats collection or not in the TraceablePeerConnection.
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// This can be useful for debugging purposes (post-processing/analysis of
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// the webrtc stats) as it is done in the jitsi-meet-torture bandwidth
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// estimation tests.
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// gatherStats: false,
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// To enable sending statistics to callstats.io you must provide the
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// Application ID and Secret.
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// callStatsID: '',
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// callStatsSecret: '',
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// enables callstatsUsername to be reported as statsId and used
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// by callstats as repoted remote id
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// enableStatsID: false
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// enables sending participants display name to callstats
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// enableDisplayNameInStats: false
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// Privacy
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//
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// If third party requests are disabled, no other server will be contacted.
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// This means avatars will be locally generated and callstats integration
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// will not function.
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// disableThirdPartyRequests: false,
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// Peer-To-Peer mode: used (if enabled) when there are just 2 participants.
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//
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p2p: {
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// Enables peer to peer mode. When enabled the system will try to
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// establish a direct connection when there are exactly 2 participants
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// in the room. If that succeeds the conference will stop sending data
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// through the JVB and use the peer to peer connection instead. When a
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// 3rd participant joins the conference will be moved back to the JVB
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// connection.
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enabled: true,
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// Use XEP-0215 to fetch STUN and TURN servers.
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// useStunTurn: true,
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// The STUN servers that will be used in the peer to peer connections
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{% if matrix_jitsi_web_stun_servers|length > 0 %}
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stunServers: [
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{% for url in matrix_jitsi_web_stun_servers %}
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{ urls: {{ url|to_json }} }{% if not loop.last %},{% endif %}
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{% endfor %}
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],
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{% endif %}
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// Sets the ICE transport policy for the p2p connection. At the time
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// of this writing the list of possible values are 'all' and 'relay',
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// but that is subject to change in the future. The enum is defined in
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// the WebRTC standard:
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// https://www.w3.org/TR/webrtc/#rtcicetransportpolicy-enum.
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// If not set, the effective value is 'all'.
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// iceTransportPolicy: 'all',
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// If set to true, it will prefer to use H.264 for P2P calls (if H.264
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// is supported).
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preferH264: true
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// If set to true, disable H.264 video codec by stripping it out of the
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// SDP.
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// disableH264: false,
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// How long we're going to wait, before going back to P2P after the 3rd
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// participant has left the conference (to filter out page reload).
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// backToP2PDelay: 5
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},
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analytics: {
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// The Google Analytics Tracking ID:
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// googleAnalyticsTrackingId: 'your-tracking-id-UA-123456-1'
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// The Amplitude APP Key:
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// amplitudeAPPKey: '<APP_KEY>'
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// Array of script URLs to load as lib-jitsi-meet "analytics handlers".
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// scriptURLs: [
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// "libs/analytics-ga.min.js", // google-analytics
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// "https://example.com/my-custom-analytics.js"
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// ],
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},
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// Information about the jitsi-meet instance we are connecting to, including
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// the user region as seen by the server.
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deploymentInfo: {
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// shard: "shard1",
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// region: "europe",
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// userRegion: "asia"
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}
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// Local Recording
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//
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// localRecording: {
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// Enables local recording.
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// Additionally, 'localrecording' (all lowercase) needs to be added to
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// TOOLBAR_BUTTONS in interface_config.js for the Local Recording
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// button to show up on the toolbar.
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//
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// enabled: true,
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//
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// The recording format, can be one of 'ogg', 'flac' or 'wav'.
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// format: 'flac'
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//
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// }
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// Options related to end-to-end (participant to participant) ping.
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// e2eping: {
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// // The interval in milliseconds at which pings will be sent.
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// // Defaults to 10000, set to <= 0 to disable.
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// pingInterval: 10000,
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//
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// // The interval in milliseconds at which analytics events
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// // with the measured RTT will be sent. Defaults to 60000, set
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// // to <= 0 to disable.
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// analyticsInterval: 60000,
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// }
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// If set, will attempt to use the provided video input device label when
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// triggering a screenshare, instead of proceeding through the normal flow
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// for obtaining a desktop stream.
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// NOTE: This option is experimental and is currently intended for internal
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// use only.
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// _desktopSharingSourceDevice: 'sample-id-or-label'
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// If true, any checks to handoff to another application will be prevented
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// and instead the app will continue to display in the current browser.
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// disableDeepLinking: false
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// A property to disable the right click context menu for localVideo
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// the menu has option to flip the locally seen video for local presentations
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// disableLocalVideoFlip: false
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// List of undocumented settings used in jitsi-meet
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/**
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_immediateReloadThreshold
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autoRecord
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autoRecordToken
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debug
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debugAudioLevels
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deploymentInfo
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dialInConfCodeUrl
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dialInNumbersUrl
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dialOutAuthUrl
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dialOutCodesUrl
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disableRemoteControl
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displayJids
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etherpad_base
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externalConnectUrl
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firefox_fake_device
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googleApiApplicationClientID
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iAmRecorder
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iAmSipGateway
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microsoftApiApplicationClientID
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peopleSearchQueryTypes
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peopleSearchUrl
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requireDisplayName
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tokenAuthUrl
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*/
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// List of undocumented settings used in lib-jitsi-meet
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/**
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_peerConnStatusOutOfLastNTimeout
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_peerConnStatusRtcMuteTimeout
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abTesting
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avgRtpStatsN
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callStatsConfIDNamespace
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callStatsCustomScriptUrl
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desktopSharingSources
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disableAEC
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disableAGC
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disableAP
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disableHPF
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disableNS
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enableLipSync
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enableTalkWhileMuted
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forceJVB121Ratio
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hiddenDomain
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ignoreStartMuted
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nick
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startBitrate
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*/
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};
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/* eslint-enable no-unused-vars, no-var */
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