LiveKit Community

&gt; ```track.setWebAudioPlugins();```
I think you will want to set the actual panner node here?

you should remove
• track.pannerNode = pannerNode
• sourceNode.connect(pannerNode).connect(audioContext.destination)
you need to
• supply your own audio context in `{ expWebAudioMix: {audioContext: this.audioContext}}`
• call `track.setWebAudioPlugins([pannerNode]);`

Hey Lukas,
Thanks for you answer, did that but still nothing happen.

No errors nothing, track is subscribed but nothing appears in my DOM,meaning no audio track.

I got the audio working (input problem), spatialization seems to work but the attenuation does not work)

```var AC = new AudioContext();
const room = new LivekitClient.Room({options : {expWebAudioMix:  AC}});

function handleTrackSubscribed(track, publication, participant) {


    if (track.kind === LivekitClient.Track.Kind.Audio) {
   var  audioContext  = this.audioContext;
      const sourceNode = audioContext.createMediaStreamSource(track.mediaStream)

        // Create and set PannerNode
        var  pannerNode = new PannerNode(audioContext, {
          distanceModel: "exponential",
          coneInnerAngle: 360, 
          coneOuterAngle: 360,
          refDistance:12,
          maxDistance: 15,
          rolloffFactor:2,
          coneOuterGain: 0,
        });
        //Define Start Position
        pannerNode.positionX.setValueAtTime(10000, 0); 
        pannerNode.positionY.setValueAtTime(10000, 0);
        pannerNode.positionZ.setValueAtTime(0, 0);
        //Connect to destination
       track.setWebAudioPlugins([pannerNode]);

       track.attach();
}```

~have you tried lowering the refDistance setting or setting maxDistance higher? that seems rather close right now. you would only get slight attenuation between 12-15 right now~
edit: ah, maxDistance is only used for `linear` distance model anyways… but if panning is working already the attenuation should work too as long as your rollof and refdistance settings make sense.

I have tried every combinations, the sound always remains at the same level

Linear could work for me having no attenuation between 0-12 and small attetnuation betwween 12-16.

But  I need the sound to phase out :slightly_smiling_face:

is setWebAudioPlugins and expWebAudioMix doing anything special ? Or creating and managing my audio elements is the same?

try with
```distanceModel: "linear",
refDistance: 1,
maxDistance: 10000,
rolloffFactor: 10,```

const sourceNode = audioContext.createMediaStreamSource(track.mediaStream)

Should I keep this? I don't think so right?

```var AC = new AudioContext();
const room = new LivekitClient.Room({options : {expWebAudioMix:  AC}});

function handleTrackSubscribed(track, publication, participant) {

      if (track.kind === LivekitClient.Track.Kind.Audio) {
        var  audioContext  = this.audioContext;     
             // Create and set PannerNode
             var  pannerNode = new PannerNode(audioContext, {
               distanceModel: "linear",
               refDistance:1,
               maxDistance: 1000,
               rolloffFactor:10,
             });
            
            track.setWebAudioPlugins([pannerNode]);
            track.attach();
  
        // Create and set Listener
        SpeakerPositions.set(participant.identity, new Map());
        SpeakerPositions.get(participant.identity).set("panner", pannerNode);
        SpeakerPositions.get(participant.identity).set("position", {x:0, y: 0, z:0});
    }
  }```

I think you would have an easier time if you enabled type checking in your IDE.
this seems wrong `expWebAudioMix: AC` .
it should say `expWebAudioMix: {audioContext: AC}`

Hey guys,
we have the issue with a participant on a "weak" device. His video and audio appeas to be frozen to one other participant where it is fine to another one. After many seconds up to some minutes, the frozen media becomes unfrozen again. There are no errors in the console. We are using latest livekit server and JS framework. This happens with simulcast turned off.
Is this known? Do you have any hint to where I should look?
Many thanks in advance.

image.png

Are there any errors in the browser console, or in the terminal that’s running the Meet server?

Sorry. I didn't read the README right. I didnt realise you had to start LiveKit server and update the .env.local file. Resolved now.

aa.png

Recently, some broadcasters started to have no sound problem. It's like their microphones are turned off but it's not. Viewers do not hear the broadcaster's voice.

livekit: 1.4.1
flutter livekit_client : v1.2.2

This issue seems to occur with publishers using Android. Could the fix marked in the image have triggered another problem?

hey <@U052BJTEV2N>, linux should work, but we still need to run the e2e tests for flutter linux sdk, I will merge them when the tests pass. maybe within this month

<@U052BJTEV2N> please try this branch <https://github.com/livekit/client-sdk-flutter/pull/209>,  I have updated dependencies and it should work correctly on ubuntu 20 + (x64/arm64).

Screenshot_2023-04-30-22-09-09-655_com.android.chrome.jpg

Capture_2023-04-30-001.png

to be clear, you’re asking for a way to unflip the front camera for the local user?

<@U01NZKTNE1H> yes. So we won't have any mirror issues on front camera or back camera

<@U045CJPACG1> take a look at the <https://github.com/livekit/components-js|new components framework> instead. The old livekit-react package is no longer supported by us. The <https://github.com/livekit/components-js/blob/main/packages/react/src/components/participant/VideoTrack.tsx|VideoTrack> component would give you control over whether it should be mirrored or not

<@U045CJPACG1> the way this is currently handled in the new components framework is via data properties (and css), have a look here in case you want to customize/override the default behaviour <https://github.com/livekit/components-js/blob/main/packages/styles/scss/components/participant/_participant-media.scss#L10>

Thanks <@U02EU2ZPTQU>. will look into this and ping if there any further inquiry

Hey there LiveKit team, just want to follow up on this one! For Unity WebGL it would be super helpful to know:
1. What the default settings are for screen share.
2. How to fine-tune the video framerate, bitrate, and audio bitrate, etc specifically for screen share.
Thanks!

(P.S. - I am guessing <@U022KB72NFJ> is the best person to ask about this?)

Also, if this API is going to change soon due to native Unity SDK being released and WebGL being wrapped into that, it may also be helpful to know if I should hold off on this sort of thing if that'll release soon. Thanks!

Hey sorry, for the late answer, you can edit the VideoSettings by creating a custom video track: <https://github.com/livekit/client-sdk-unity-web/blob/53c2fc94934264f53b46e46f05488f6618107755/Runtime/Scripts/Client.cs#L38|Client.CreateLocalScreenTracks>
The Native SDK will not directly support WebGL at the release but this is planned to do

<@U022KB72NFJ> Thanks for the recommendation! So for example, using C# how exactly would I use that to initiate a screen share with 60fps frame rate? Is there an example somewhere, or could you provide a code snippet?

And also just wondering what the default values are set to.

Thanks!

Hey <@U022KB72NFJ>, just want to follow up on this one. Rather than writing instructions in a Slack thread, maybe it would make the most sense for some official documentation to be written on the website for initiating a screen share (with audio) that has custom quality/framerate/bitrate settings for both video and audio, with example code for doing this in Unity C#. And maybe the documentation could share what the default settings are set to, and what some "good" settings might be for adjusting the quality/framerate/bitrate higher or lower than default. Would it be possible for you (or another LiveKit team member) to do this?

Currently I just don't feel like I have enough information to be empowered to stray from the default screen share wrapper function. I could potentially experiment with the code and eventually arrive at a solution, however iteration is currently not very easy since I can't test in the Unity Editor and would have to make a build each time. It would be *amazing* to have some specific documentation that I could follow in order to initiate a screen share with specific audio/video settings and some code snippets for Unity C#. :slightly_smiling_face:

By the way, we've gone ahead and launched our screen share feature just using the default LiveKit settings. It seems to work pretty well! But we would love to tune the audio/video quality settings since our platform will soon be hosting the virtual component of a film festival, where longer-format videos/films will be screened via the screen share feature.

I do see that there is an "Advanced track management" section in the docs under "Publishing Tracks". But I am not sure what the "right" way is to adjust audio/video settings specifically for screen share and then initiate screen share.

Perhaps I could deep dive into the code to see what `yield return currentRoom.LocalParticipant.SetScreenShareEnabled(true);`  is actually doing and try to mirror that but with adjusted settings... But it would be greatly helpful to have some specific documentation and example settings I could follow. As a beginner to LiveKit, some of this is a bit confusing/intimidating, or maybe not immediately obvious (even if it's actually easy). I just need a bit more help/direction to get going. Thanks!

Hey there <@U04UL2WJ7PU> — check this out:
<https://blog.livekit.io/stream-music-over-webrtc-using-react-and-webaudio/|https://blog.livekit.io/stream-music-over-webrtc-using-react-and-webaudio/>

This is a very old server version. Have you tried updating to the latest server?

yes we have tried version 1.1.2 but was facing another freezing  error so…

Thank you..i will check with that version. but we were using same server version i mentioned above and it was working fine can be there another issue or i have to do anything on client side?

don’t know, that server version is over a year old

hello, we have updated the server version to latest 1.4.1 but i am still facing legging issue randomly for 2 to 3 seconds. and sometimes not getting *Remote video Track* of ipad 8 generation  os version 14.8 device.

livekit.logs

<@U02GAC58A79> <@U01NZKTNE1H> I am attaching log for more information, hope this will helpful to identify

Hello Denis,

Have you tried the following:
```.on(RoomEvent.DataReceived, ...```

For future reference, feel free to have a look at the example provided here:
<https://github.com/livekit/client-sdk-js/blob/main/example/sample.ts#L113>

Hope this helps. :slightly_smiling_face:

Also, the docs seems to be up to date. Feel free to have a read here:
<https://docs.livekit.io/client/data-messages/#data-messages>

Thanks, Pat! Yet I was looking for Golang server example, not JS client. On the docs link, there is a Golang tab. From the example, I cannot figure out where this `OnDataReceived` callback coming from

Oh Sorry about that. For Goland, the way I've implemented it was:

```roomCB := &amp;lksdk.RoomCallback{
		ParticipantCallback: lksdk.ParticipantCallback{
			OnDataReceived: handleOnDataReceived,
		},	
	}```


then pass your callback into the `ConnectToRoom`

For reference: <https://pkg.go.dev/github.com/livekit/server-sdk-go#ParticipantCallback>

Thank you Pat. <@U036FGT00TT> you are right the Go docs are outdated. We'll get them updated shortly!@

<@U041KCGQCGZ> I assume you already seen this: <https://github.com/livekit/livekit-helm/blob/master/ingress-sample.yaml>

Can you provide more info about what you are trying?

this is my helm code:
```microk8s helm install ingres livekit/ingress --namespace ingress --values ingressConfig.yaml```

<@U01KZ61GV5Z> I put the yaml file in the working directory by making the necessary corrections. but I get this error:
```Error: INSTALLATION FAILED: failed to download "livekit/ingress"```

<@U041KCGQCGZ> when you get a chance can you try this install again please? Looks like I didn't push the helm chart correctly to the repo on our first ingress commit, should be there now!

Thank you <@U01UB0LHTTQ> and <@U01KZ0SCDKL> for your fast response!

The reason we haven't updated yet is a different behaviour in the output's frame rate.
Our aim is to get a constant rate. Unfortunately we noticed variable frame rates in 1.4 and higher.

That was an experiment to save on CPU, but it introduced other issues so the behavior was reverted in v1.6.3

By the way, the `pipeline frozen` error is due to timestamping issues, where even if the egress finished the file would be broken. I’ve done a lot of work in the 1.6 versions to solve it, but you might still get an occasional `timestamping issue` error even after updating (but the egress should end properly). I plan on having it fully resolved in v1.7.1

<@U01UB0LHTTQ>
So we can hope for constant frame rates since v1.6.3?
This sounds amazing!

<@U01UB0LHTTQ>
Thank you again for your explanation.

We'll be updating the defaults today. Feel free to override the version to 1.7.0 though.

In general we will not introduce changes that are not backwards compatible, so it should be safe to point the helm chart to later versions of Egress

something looks off here, it's using really low bitrates for a 720p video. is this reproducible?

we cant reproduce it with similar device, we dont have access to this device

that's the typical cause of lower quality.. and the dashboard is also indicating the high resolution layer being missing

it could be due to other issues local to the device too

especially if there are other heavy processing that's running

are you talking about a sdk internal reconnect (RoomEvent.Reconnecting) or are  you closing the connection and re-connecting yourself with the original token?

I take the token that sdk generated for me and insert it in this example <https://example.livekit.io/> , then I change the rights with sdk, they change, I check them with the ListParticipants method. After that I close the tab in the browser, and I try to reconnect with the token that I generated at the beginning, in the new tab, and I still have the rights as they were before they were updated.

<@U0566NX8F8S>, I don't think the LiveKit server is keeping persistent states between manual connections for participants. If a participant leaves, then his state is not kept in any database in a chance of reconnecting, etc. With this said, pretty sure the state is kept if this reconnecting is happening from internal. But someone could confirm this from the LK team. My Take on this, is you might be looking for a custom implementation of a state manager.

<@U032TUAL4KT> is right. as soon as you close the tab, the participant disconnects from the server and the server “forgets” about this participant.

when you connect again with the same token, the server only interprets the permissions that are encoded in that token.

one way to solve this in your app could be to have users request a token with the desired permissions before they connect.

<@U02EU2ZPTQU> I thought about it, but I noticed that the SetValidFor method doesn’t work quite as I expected. For example, I set the lifetime of the token to SetValidFor(30 * time.Second), but the token still lived for one minute. But if I logged into that token during that minute, the token was given a lifetime longer than that minute.

My main problem is that a user can log in and immediately leave the room and another person will take his place, and if that user uses his old token again, there will be one more person in the room who will be streaming. Restriction on the number of people in the room, I can not put, because there are spectators. Because of this, the only solution I came up with is to give permission after the connection, and before that to give rights only to view.

Although, only 1 person can be joining a room with the same identity, so if this 2nd person is already logged, then he will be replaced with the latest participant joining with that token identity.

<@U032TUAL4KT> If you make them one identity, the old person can still enter the room using the old token, and kick the person who is there correctly out of the room.

That is correct, although why would 2 people be sharing the same token?

I thought you suggested using different tokens, but instead of a name, specify in identity the place of the streamer.

Sorry for the confusion, I think we misunderstood ourselves. Every users should have its own token. I feel like I might be missing a bit more context on your application, but wouldn't you be able to generate tokens for streamers and a different token for spectators?

Which would come down pretty much to the same suggestion of <@U02EU2ZPTQU> .

Yes, that’s right, everyone’s token is different, and everyone’s personality is different. Otherwise they would just be kicking each other around in circles. The idea is for the streamers to switch, first one would sit down and tell stories, and then when one leaves, another would sit in his place.

If I give the streaming rights right away in the token, then I will not be able to update the user’s token, and he can use it after someone has sat in his place, and it turns out that there are two people sitting in the same place.

Okay, I understand what you are saying. In this case, to come back to your original question, as <@U02EU2ZPTQU> confirmed LiveKit does not keep persistent state of participant if they leave a room.

My suggestions would probably be one of these 3 options:
1- Create custom tokens for streamers
2- Keep a room state of your participants based on identity outside of LK, handled through server SDK
3- Probably not the best but, if a streamer disconnects, assign the streamer permissions to any spectator until another streamer joins a room and that spectator could reassign permissions to the new streamer. (This is somewhat how zoom works if the host leaves the room)

Those are my thoughts <@U0566NX8F8S>.

&gt; The idea is for the streamers to switch, first one would sit down and tell stories, and then when one leaves, another would sit in his place.
I’m not sure I follow. This sounds like the fact that a streamer leaving and having permissions reset to viewer-only is actually a good thing for your use case?
1. Alice joins with subscribe only permissions
2. App grants alice publish permissions when it’s her streaming slot is available
3. Alice disconnects, App grants publish permissions to someone else
4. Alice connects again and has to wait again until publishing rights are granted in order to stream

&gt; but I noticed that the SetValidFor method doesn’t work quite as I expected. For example, I set the lifetime of the token to SetValidFor(30 * time.Second), but the token still lived for one minute.
Yeah, this is currently expected, the team discussed internally already whether it would be a good idea to lower this leeway (the leeway being 60 seconds right now)

Yes, it is. I would like a user who stopped being a streamer, and his place was taken could not come back to be a streamer, until his place is free. Otherwise it may be when 2 people will stream at the same time.

Initially, I wanted to give them a token with permission of streaming, and take it away remotely. But now I realized that the option to give them viewing permission, and then give them streaming permission looks easier to implement.

I also noticed a strange behavior when I grant such permission. The user can enter the room even when it was deleted. Which will create it again. Although in the rights it is specified that he does not have a permission to create a room.

```grant := &amp;auth.VideoGrant{
			RoomCreate: false,
			Room: roomName,
			RoomJoin: true,
		}```

I know there's an option on the LiveKit Server level around automatically create rooms if they don't exists.

If you are using cloud, that option can be found under the project settings.

this is a sign that Chrome was not launched correctly. It's possible you've exceeded some resource limits on the server side

yeah seems like it
i also faced this issue in multiple recordings

```replicaCount: 1

# Suggested value for gracefully terminate the pod: 1 hour
terminationGracePeriodSeconds: 3600

egress:
  api_key: ""
  api_secret: ""
  ws_url: ""
  log_level: info
  health_port: 8080
  prometheus_port: 9090
  # template_base: "<https://your-custom-template.com>"
  redis:
    address: 1.2.3.4
    db: 0
    username: default
    password: qwe
    use_tls: false
  s3:
    access_key: ""
    secret: ""
    region: ""
    # endpoint:
    bucket: ""
  # cpu_cost:
    # room_composite_cpu_cost: 3
    # track_composite_cpu_cost: 2
    # track_cpu_cost: 1

# autoscaling requires resources to be defined
autoscaling:
  # set to true to enable autoscaling. when set, ignores replicaCount
  enabled: true
  minReplicas: 1
  maxReplicas: 5
  targetCPUUtilizationPercentage: 60
#  targetMemoryUtilizationPercentage: 60
  # for use with prometheus adapter - the egress service outputs a prometheus metric called livekit_egress_available
  # this can be used to ensure a certain number or percentage of instances are available
#  custom:
#    metricName: my_metric_name
#    targetAverageValue: 70

# if egress should run only on specific nodes
# this can be used to isolate designated nodes
nodeSelector: {}
# <http://node.kubernetes.io/instance-type|node.kubernetes.io/instance-type>: c5.2xlarge

resources:
  requests:
    cpu: 3500m
    memory: 1024Mi
  limits:
    cpu: 8000m
    memory: 2048Mi

serviceAccount:
  # Specifies whether a service account should be created
  create: false
  # Annotations to add to the service account
  annotations: {}
  # The name of the service account to use.
  # If not set and create is true, a name is generated using the fullname template
  name: ""

podAnnotations:
  <http://sidecar.istio.io/inject|sidecar.istio.io/inject>: "false"
  <http://linkerd.io/inject|linkerd.io/inject>: disabled

podSecurityContext: {}
# fsGroup: 2000

securityContext: {}
  # capabilities:
  #   drop:
  #   - ALL
  # readOnlyRootFilesystem: true
# runAsNonRoot: true
# runAsUser: 1000

tolerations: []

affinity: {}```
this is my config for helm
should I change the `resources` for better performance?

track composites are fairly expensive to run, you wouldn't be able to run more than a couple with `3500m` requested

got it
i am using web egress btw
should I have more resources *+* autoscale it at like 50% of cpu usage?
<@U01KZ61GV5Z>

it's workload dependent so you'd have to play with the scaling parameters to see what's best for your workload

<@U027V3P9UQL> web egress uses 4 CPU in average cases, but it can use up to 6 for more intensive templates. An easy/effective way to autoscale is by using large machines (something like 12 or 16 CPU) and targeting around 70% CPU usage

Is there a document where I can install ingress using Kubernetes?

What are the OBS stream settings? Do you get any output from the ingress service? What do the OBS logs say?

I just solved the problem. As I understand it, I could not access the microk8s Load Balancer because external ip was not assigned. It seems to be resolved when I edit the service and assign the server's ip.

I've never had to do this on livekit or egress helm installations.

Hello
I am using `livekit-client:1.3.1`
When I am accessing camera more than 2 minutes it show empty black screen and also browser tab gets freezed.

I am using client sdk in Javascript project

hi <@U054E6M4X6G>
is this happening for you in react strict mode (e.g. in dev environment) only or also when running a production build (without react strict mode)?

<@U02EU2ZPTQU> I checked without strict mode and what i wrote above didn't happen. Thank you.

IMG_5018.jpg

is this an issue that appeared only recently or has this always been a problem in your application?

is it possible for you to update livekit-client to the latest version and test with it?

• for the most part you can ignore “silence detected” as a warning, wondering if the silence-check itself is what freezes chrome. Which chrome version does the error manifest on?
• “ping timeout triggered” doesn’t sound good. are you using livekit cloud or self hosted? in case of self hosted, which server version?

<@U02EU2ZPTQU> I have already upgraded `livekit-client` to 1.9.2

Chrome version : `Version 113.0.5672.64 (Official Build) (64-bit)`

thanks! does this also happen to you on <http://meet.livekit.io|meet.livekit.io> or only in your custom app?

I have not tested on <http://meet.livekit.io/> with multiple peers and also I am using `client-sdk-js` in vue3/quasar.js app

<http://meet.livekit.io|meet.livekit.io> is also using client-sdk-js under the hood. If it was an issue with the client-sdk-js, you should be seeing the same issue there!
In case you want to test closer to the original client-sdk-js, you can also try with `yarn sample` in the client-sdk-js repo and try to reproduce the error there.

So it might be issue of wrong configuration in my custom app, correct?

if you cannot reproduce it in one of our sample apps, then it sounds more likely that it’s a bug in your application, yes!
If you can provide a <https://en.wikipedia.org/wiki/Minimal_reproducible_example|MRE>, that showcases the issue, I’m happy to take a look whether the issue is in fact on the livekit side!

Okay, thanks for helping and guiding, I will create one sample example to reproduce the example.

Actually I am using some custom functionalities with livekit, so I need some time to create sample reproduction.

I guess the fact that you’re using qasar might be the actual problem. Just from experience with how many things <@U01NZKTNE1H> had to patch for react-native to work properly, I guess it’s likely that _something_ doesn’t work quite right with quasar and webrtc. but that’s just a wild guess.

yeah, I will take care of it and we are developing web based app.

react-livekit.zip

Hey I have created simple React App  with livekit react client to reproduce errors/warnings.
When connecting more than 2 peers I am facing warnings and audio/video freezing issues.

thanks! could you enable debug logs for this by calling `setLogLevel('debug')` from livekit-client and post all logs (unfiltered) as a txt file? that helps us debugging the issue.

hi <@U056LA70CDQ>
that’s unexpected, could it be that you’re still calling a method on `room` after you called `room.disconnect`?
if you can provide a minimal reproduction example of the issue, I’m happy to take a look!

how is the room disconnected? just `room.disconnect()`?

<@U01NZKTNE1H> For that if all participants get out of room. from backend(server) the mobile application will listen the disconnect socket event. through this event Mobile Application will remove that video calling screen and in *dispose* event of this video calling screen we have done room.dispose(); and room.diconnect();

is this accurate?
1. server force disconnects the connection.
2. mobile application listens for disconnect socket event
3. upon receiving disconnect event, calls `room.dispose`/`room.disconnect`?

as i know yes, every time the disconnect event is getting listened to mobile side.

and the app hangs during the `room.dispose` /`room.disconnect` call? cc <@U02GAC58A79>

one of the side of the video calling the screen gets stuck when this happened and stopped the remotevideo track and localevideo track.

The iphone or Android phone calling the Samsung tab ends the call from phone. The phone is ok, the Tablet keeps the call going. It does not disconnect from the tablet and the call is stuck does not disconnect even when pushing end, keeps going on Server where we took logs. To disconnect needs needs forced app or device closure.

Screenshot 2023-05-08 at 6.12.53 PM.png

hey, is the audio and video connection not properly disconnected when disconnecting the room? so can the issue you are facing be reproduced in client-sdk-flutter/example?

If there are some steps to reproduce the issue, it may help us locate the bug quickly.

it is happening like 1/5  video call randomly. and for Video i have seen that the  remotetrack is not available on that device.

so, as you mean 1v1 video call similar to facetime by video call? Or a multi-party video room?

This is more like a UI blocking issue. when you press the hangup button on the tablet, you can try to unpublish and stop localtrack, then call room.disconnect, and then return to the contact list page, BTW, you can try to track in logcat whether there is an exception throw,

okay, I found a logic problem in voip signaling, you use the iphone as the calling party, and the tablet as the called party, when your iphone hangs up, you need to send a separate notification to the tablet to inform the call is over, and let the tablet automatically hangup . As far as I know, livekit does not provide signaling logic similar to sip voip, so  when the caller hangs up  just leaves the room.

you can use the data-channel on the iphone to send a `hangup` command to the tablet.

<@U02GAC58A79> right we will do as you said but the problem is i am not able to reproduce this issue at debugg time and so cant see the loggs. so i have put the logic for disconnecting from call if you say i can share that part with you. if you can suggest be the better solution

We have handled this before, it has been working well for months. 2 weeks ago this issue came, we can reproduce it in many devices, but not all / all setups we do not know why

I think we need some device logs to locate the bug.  when you press the hang up button, there should be an exception thrown to prevent the UI page jumping, am I right?

maybe you can try to connect to logcat and try to reproduce the issue in your video.

Thank you Duan! We aim to get the logs and if possbile try logcat..

&gt; 2 weeks ago this issue came, we can reproduce it in many devices, but not all / all setups we do not know why
it'd be best to look through what's changed in your app between now and then. is it calling `Room.disconnect()` ? You can log an entry before and after disconnect to see if it gets through there

<@U0571DWBDC1> <https://github.com/livekit/client-sdk-flutter/commit/bf07a91c985addb62a1c080838d6fe6f7fbe09c8>
This may be a related fix, as other users have reported that in 1v1 calls, the local client throws an exception when the remote participant exits.

With the usage of the components, sky is pretty much the limit as to how your app's layout should be presented the your visitors. This might be a great starting point to look at: <https://github.com/livekit/components-js/tree/main/examples/nextjs>

thanks for the response i tried to force custom styles using css but the issue is that all classnames are pretty much identical
i thought there was a way to handle that maybe with a hook that comes out of the box or something but i guess that's not the case.

Just my 2 cents, but maybe have a look at this repo on how they style everything: <https://github.com/livekit/components-js/tree/main/packages/styles>

alright will be checking it out, thanks a lot for ur time.

there is also a set of css vars that you can override to effectively theme the components. We’ll need better docs for that, but the styles package Pat linked to defines those variables as well.
To your point about all CSS classes being identical, can you elaborate on that?

hello yes sure, whenever a user enters the room all the cameras section have the same classname when inspecting therefore i can't seperate them , i may be wrong tho because this is the second day of me using it.

how would you want to separate them? by Track.Source or by participant id ?
Can you share a bit more about what you’re trying to achieve?

`hidden` doesn't hide other participants.. it makes the current participant invisible to others.

<@U01KZ61GV5Z> sorry. i meant that. i want to enter room hidden. that other participants can't see me.
I changed token option to `hidden: true` but still other participants can see me. They can't see my video but can see my tile and my name.
Only when i change `canPublish` and `canPublishData` to `false` other participants can't see me. But after several seconds I get this error message `Error: publication of local track timed out, no response from server`
What am I doing wrong?

it sounds like you are publishing tracks as the hidden participant? if you do, then that participant will become visible to others.

A track must belong to somebody. Since all published tracks are visible to others. so would the participant that published it.

I'm curious what your use case is? Are you building an app where end users would be seeing the `VideoConference` prefab but should not publish themselves?

<@U01KZ61GV5Z> yes, if user chose not to show himself to others. But when is `hidden: true` and `canPublish: false`, `canPublishData: false` I got error message. How can I ged rid off it?

you'd want to ensure your `&lt;LiveKitRoom` isn't publishing anything.. i.e.:
`audio={false}`
`video={false}`

<@U056Q5W6WGJ> no plans for that currently. We stream speech as an audio track in WebRTC. WebSocket uses TCP which isn’t designed for streaming media. Is there a particular reason your interested in that route? What are you trying to build?

We would like to add/support STT/TTS from other providers though, if that’s what you mean. If you end up building support for that, we accept PRs! :blush:  

Sure thing! I'm trying to integrate STT/TTS from providers in China, eg. Baidu &amp; Ali,

Hi Bro!
I have an question about KITT.Currently, I’m trying to read out the speech stream as you did in `lkgpt-service/pkg/service/transcriber.go` , since the ASR API I tried to use just received PCM raw data, so I tried to modify the code in `start()`  to use <https://github.com/hraban/opus> to decode ogg data to pcm data as following
```                      stream, err:= opus.NewStream(t.oggReader)
                        if err != nil {
                                logger.Errorw("failed to create opus decoder", err)
                        }```
but got `failed to create opus decoder	{"error": "OP_EBADTIMESTAMP"}` error.
I’m new to golang, so have no idea if I use a wrong lib?
or any other ways to get pcm raw data ?

THANK YOU VERY MUCH!

Any thoughts on decoding the audio in Go <@U022KB72NFJ> <@U01KZ61GV5Z> ?

I'm not familiar with the library you are using, but it's worth trying manually passing through each sample as noted in <https://github.com/hraban/opus#decoding|this section>. The `NewStream` API might be expecting certain Opus/Ogg headers that aren't present in the stream.

Thank you! Actually, I tried the the way to decode it by each chunk which read out by oggReader,  but no lucky either. :sweat_smile:

Hi Team, I added a function by using “<http://mccoy.space/g/ogg|mccoy.space/g/ogg>”
```import (	
   "<http://mccoy.space/g/ogg|mccoy.space/g/ogg>"
)

func OggToPCM(data []byte) ([]byte) {
        decoder := ogg.NewDecoder(bytes.NewReader(data))
        page, err := decoder.Decode()
        if err != nil {
                logger.Errorw("Error while decoding opus file: %v", err)
                return nil
        }
        //fmt.Println("ogg data has %d pages", len(page.Packets))
        audiodata := make([]byte, 0)
        for i:=0; i&lt; len(page.Packets); i++ {
                audiodata = append(audiodata, page.Packets[i]...)
        }
        return audiodata
}```
and update  `transcriber.go`  as following
```                go func() {
                        defer close(nextCh)
                        buf := make([]byte, 1024)
                        for {
                                select {
                                case &lt;-endStreamCh:
                                        return
                                default:
                                        n, err := t.oggReader.Read(buf)
                                        if err != nil {
                                                if err != io.EOF {
                                                        logger.Errorw("failed to read from ogg reader", err)
                                                }
                                                return
                                        }

                                        if n &lt;= 0 {
                                                fmt.Printf("No Data read from OggReader \n")
                                                continue // No data
                                        }

                                        opusdata := OggToPCM(buf[:n])
                                        if opusdata != nil {
                                                opusbytes = append(opusbytes, opusdata...)
                                        } else {
                                                fmt.Printf("No Data from OpusDecode \n")
                                        }
                                }
                        }
                }()```
 but seems it’s not working neither, any ideas on handle the buffer from `oggReader` and convert it to PCM raw data?

if it’s a permissions related case, you would probably want to do this by setting `canPublish` permissions on the token and update the permissions via the server-sdk once the admin has approved it.
The components-react control bar will automatically only show camera and audio toggles if `canPublish` is enabled.

which version of the server are you using? I think this was a bug at one point but have been fixed since.

Hi, When I used `updateParticipantInfo` in server-sdk-kotlin to change permissions, it always failed to change correctly. Can anyone help me out?
Server Ver: 1.4.1
Server SDK kotlin Ver: 0.5.1

Description:
When I change any permission of participant using `updateParticipantInfo()` (any value was tried), all permissions become false.