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Yeah, it is a hint to encoder. MDN documentation on WebRTC is super useful - https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/getUserMedia

this documentation is regarding bandwidth restrictions ?? and as an hint means that bandwidth must be aroung 1700kbps near about 1700 kbps

Yeah, the constraints specify how the track should be encoded. So, 1700 kbps is a constraint on bit rate. It is a longer term average. Encoder could overshoot/undershoot under some conditions, but longer window rate will be at that specified rate.

okk I had set in the following function

private async publish(track: LocalTrack, opts: TrackPublishOptions, isStereo: boolean) {

before creating rtp sender in the encodings.

// store RTPSender
    encodings[0].maxBitrate=VideoPresets.h720.encoding.maxBitrate
    encodings[0].active=true
    <http://log.info|log.info>("//SHAHDEBUG encoding before creating sender",encodings)
    track.sender = await this.engine.createSender(track, opts, encodings);
    log.debug("//shahdebug: creating rtp sender")

it is then capping it to 1700kbps .. is it a right pl;ace to handle it..??

Not sure what you are trying to do. Layers should be set via

TrackPublishOptions

rather than modifying SDK code - https://docs.livekit.io/client-sdk-js/interfaces/TrackPublishOptions.html#videoSimulcastLayers

I am trying to capping the video bandwidth with maxBitrate mentioned in

h720: new VideoPreset(1280, 720, 1_700_000, 5),

For this where should I change ?

Please read

TrackPublishOptions

documentation and choose settings. You can use preset already available in SDK or make up your own.

I am using available preset only h720 .. but my problem is ,the maxbitrate specified in that preset is not getting set .. so I am trying to set that explicitly

Please file a bug if what you set via API is not working properly. That needs to be fixed.

okk I will do that but first I am trying to figure out myself.. where is the problem . ]