Low video quality after configuring constraints

We are developing from lib-jitsi-meet and created api to create/join room in our installed jitsi server(local machine). we called this api from normal html and from another website (all are running local network in different machines) that that need this api to create/join room.
but in that video quality is low (near 360p) for remote videos though I set constraints to 720 for ideal,min and max.


but if I join that room from server(192.168.0.150/…) then the video quality is really good:

So I think the sender is sending good quality video but somehow the reciever(where we manually attach incoming videos to UI) cant bind the good quality video to video tag. I searched but the topic gone deep. we didnt dig so much… we just attached the video normally. what is the strategy or where is the place to configure this all or develop by my need? how should implement simulcast in lib-jitsi-meet or is it already in there?
Thanx in advance :heart:
@damencho

You need to select the video, saying it is on stage so the jvb can send you HD quality, otherwise it is considered thumbnail quality.

1 Like

so before attaching I should just pass the participantID array to this.rtc.selectEndpoints() ?
and If I just want only one specific person’s video always high quality I should just pin that participant (ex:moderator) by using JitsiConference.prototype.pinParticipant() func?
Thanx for the info :heart:

  1. We tried using that selectParticipant() function but it hadn’t that effect.(I restarted jicofo,prosody,videobridge)

  2. Then I disabled simulcast and it was a real great quality for 1st 2 person (student and teacher, where teacher see everyone and student see only teacher) but when another student join the room the good quality falls though we added necessary constraints for 720/1080 (min/max/ideal).

  3. then I thought may be this(goodquality) is beacause of p2p and when another participant come the p2p turns off and quality falls… but I even turned off p2p and same again… good quality for 1st 2 person but when 3rd person came in the quality falls though I disabled simulcast and added necessary constraints in config.

Is there anywhere else,where I should do something? or what is the problem I am facing?
network is not issue as we are in local network and through direct server room (go through 192.168.0.150/newroom )the quality is good for even 3 or more and we are facing bad quality for creating room by using libjitsi in same server.
here is the config.js

/* eslint-disable no-unused-vars, no-var */

var config = {
    // Connection
    //

    hosts: {
        // XMPP domain.
        domain: '192.168.0.150',

        // When using authentication, domain for guest users.
        // anonymousdomain: 'guest.example.com',

        // Domain for authenticated users. Defaults to <domain>.
        // authdomain: '192.168.0.150',

        // Jirecon recording component domain.
        // jirecon: 'jirecon.192.168.0.150',

        // Call control component (Jigasi).
        // call_control: 'callcontrol.192.168.0.150',

        // Focus component domain. Defaults to focus.<domain>.
        // focus: 'focus.192.168.0.150',

        // XMPP MUC domain. FIXME: use XEP-0030 to discover it.
        muc: 'conference.192.168.0.150'
    },

    // BOSH URL. FIXME: use XEP-0156 to discover it.
    bosh: '//192.168.0.150/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: 'focus@auth.192.168.0.150',


    // Testing / experimental features.
    //

    testing: {
        // Enables experimental simulcast support on Firefox.
        enableFirefoxSimulcast: false,

        // P2P test mode disables automatic switching to P2P when there are 2
        // participants in the conference.
        p2pTestMode: false

        // Enables the test specific features consumed by jitsi-meet-torture
        // testMode: false

        // Disables the auto-play behavior of *all* newly created video element.
        // This is useful when the client runs on a host with limited resources.
        // noAutoPlayVideo: 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: false,

    // Enabling this will run the lib-jitsi-meet no audio detection module which
    // will notify the user if the current selected microphone has no audio
    // input and will suggest another valid device if one is present.
    // enableNoAudioDetection: false

    // 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,

    // 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.
    constraints: {
        video: {
            aspectRatio: 16 / 9,
            height: {
                ideal: 720,
                max: 720,
                min: 720
            },
            width: {
                ideal: 1280,
                max: 1280,
                min: 1280
            }

        }
    },

    // Enable / disable simulcast support.
     disableSimulcast: true,

    // 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: false,

    // 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,

    // 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

    // Whether to enable file recording or not.
    // fileRecordingsEnabled: false,
    // 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://192.168.0.150/static/oauth.html'
    //     redirectURI:
    //          'https://192.168.0.150/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: false,

    // Transcription (in interface_config,
    // subtitles and buttons can be configured)
    // transcribingEnabled: false,

    // Enables automatic turning on captions when recording is started
    // autoCaptionOnRecord: false,

    // Misc

    // Default value for the channel "last N" attribute. -1 for unlimited.
    channelLastN: -1,

    // 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: 'en',

    // 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 sending participants display name to callstats
    // enableDisplayNameInStats: false

    // enables sending participants email if available to callstats and other analytics
    // enableEmailInStats: 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: false,

        // Use XEP-0215 to fetch STUN and TURN servers.
        // useStunTurn: true,

        // The STUN servers that will be used in the peer to peer connections
        stunServers: [
            { urls: 'stun:stun.l.google.com:19302' },
            { urls: 'stun:stun1.l.google.com:19302' },
            { urls: 'stun:stun2.l.google.com:19302' }
        ],

        // 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

    // Deployment specific URLs.
    // deploymentUrls: {
    //    // If specified a 'Help' button will be displayed in the overflow menu with a link to the specified URL for
    //    // user documentation.
    //    userDocumentationURL: 'https://docs.example.com/video-meetings.html',
    //    // If specified a 'Download our apps' button will be displayed in the overflow menu with a link
    //    // to the specified URL for an app download page.
    //    downloadAppsUrl: 'https://docs.example.com/our-apps.html'
    // }

    // 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 */

@damencho
plz help and thanx in advance :heart:

we used “selectParticipant(id)” for student (as we only need teacher) and “selectParticipants(array of ids)” for teacher (as we need all student)
but after using “selectParticipants(array of ids)” for both student(only see teacher) and teacher(see all student) now the quality is good for both and for more than 2 person. not only this, the simulcasting is even disabled and then why these functionalities are even affecting?
Thanx :heart:

Select participant is for simulcast only. When are you sending selectParticipant, it should be once the stream is received?

not only this, the simulcasting is even disabled and then why these functionalities > are even affecting?

With simulcast disabled you shouldn’t see different resolutions. I would double-check that simulcast was actually disabled. Having said that, disabling simulcast is a bad idea. It might work well in the local network, but it won’t scale.

Boris

1 Like

yeah but it didn’t work for me both in simulcasting enabled/disabled case
and “selectParticipants(array)” worked even the simulcasting is disabled. We disabled the simulcasting by just changing the confgi.js , disableSimulcast=true and then restarting jicofo, videobridge, prosody.dont know whether the “constraints” is affecting as I set them to have good quality.
I shared config.js previously and still dont know then why is this function even working.
we used the array while updating the UI, something like :

if (this.getRole() === "student") {
        this.room.selectParticipants([this.getModerator()]);
        return;
      } else if (this.getRole() === "teacher") {
        this.room.selectParticipants(this.members);

Thanx :heart:

Yeah now I am confusing that is my simulcasting enabling/disabling even working …
I changed in config.js and restarted components. I set the constraints to have good quality and dont know whether this is affecting.
and thanx for the info, I will check everything enabling simulcast :heart:

When you do this.room.selectParticipants([this.getModerator()]); or this.room.selectParticipants(this.members) are you receiving video of that participant on that client?

yes… I think so, this is called when someone enters the room… then we select depending on role. though it is possible to be called before/after video streams are recieved… then should I wait till I get the video stream strictly?

Yes, wait for the streams, as it may turn out the bridge does not know about this stream when you select it, I think.
Another option is to open meet.jit.si, find this method in the code add a breakpoint in the debugger, reload and see when it is called, when the stream as received and goes on large … when the debugger stops you can go through the stack trace and check that.

1 Like

Thanx for ur info :heart: