Hello, Can we use livekit in nextjs without python backend?

What's the best way to handle adjusting the user's audio input gain/volume? Here is part of my setup:

room.switchActiveDevice('audioinput', audioDeviceId);

      const audioTrack = await createLocalAudioTrack({
        echoCancellation: true,
        noiseSuppression: true,
        autoGainControl: false,
      });

      const localTrackPublication = await room.localParticipant.publishTrack(audioTrack, {
        name: 'microphone',
        source: Track.Source.Microphone,
      });

Hey @everyone, I am a kind of beginner to intermediate in using livekit But still there's a lot of room to improve. I have great project idea to work and it needs to have a good knowledge of livekit. I would like to connect with you guys if you want to team up with me and build. Thank You.

does anyone have experience or working example using livekit js sdk in wechat mini program? https://developers.weixin.qq.com/miniprogram/dev/component/live-player.html https://developers.weixin.qq.com/miniprogram/dev/component/live-pusher.html

hi team, i found web SDK will trigger handleParticipantDisconnected for every remote participants when handleRestarting (https://github.com/livekit/client-sdk-js/blob/main/src/room/Room.ts#L1498), why do that ? how can i keep remote participants view when reconnecting ? Now, every time i successfully reconnect, it feels like everyone except me has disconnected and reconnected, remote participant views removed and appeared again

hello, someone know if i need to enable something on web to work the configs of voice on flutter app in web? for example echo cancelation, noise supressions and others configs?

Hi tried sending DTMF from the react sdk to a room that has a SIP participant (my cellphone) but I do not hear the dtmf tones: https://docs.livekit.io/sip/dtmf/#sending-dtmf is there something I am missing with how this is used? I tried sending these exact tones from the react sdk from a browser client

Is it possible to stream a canvas via video using the livekit api?

@refined-appointment-81829 Azure TTS throwing err. could you please check what's wrong in the API we are using this code to initialize. tts = azure.TTS( speech_key=os.environ.get("AZURE_SPEECH_KEY"), #speech_region=os.environ.get("AZURE_SPEECH_REGION"), speech_endpoint=os.environ.get("AZURE_SPEECH_HOST") )

post initialization when we call then it throws err

Hello 👋 It seems the node.js guide is still in beta. But is it functional enough to start playing with it? Also, is there a github project that I can clone and run locally on my machine? I'm trying to create an AI voice assistant with STT, LLM, TTS.

Hey all my live kit worker has some errors like "inference is slower than realtime" then kills the agent initialization process. has anyone faced this before or have any pointers or recommendations?

Hello, i have a qestion regarding real time stt in livekot sdk. I want to implement video conferencing tool with access to live speech to text and speaker labeling. Is it possible in current version of sdk?

I have self hosted the livekit meet latest version from this repository https://github.com/livekit-examples/meet , and I am getting this error

MaxListenersExceededWarning: Possible EventEmitter memory leak detected. 101 participantEncryptionStatusChanged listeners added. Use emitter.setMaxListeners() to increase limit

When 150 participants join the meeting, UI is becoming laggy and getting random reconnecting issue

const { roomService } = require("../services/getTokenLivekit");

const handleSync = (socket) => { socket.on("togglemic", async (data) => { try { console.log(data); await roomService.mutePublishedTrack("5bz-4rt-6xy", data?.uid, data?.sid, data?.value); } catch (error) { console.log(error); } }); }; module.exports = handleSync; ---------------------------------------- at.addGrant({ roomAdmin:true, roomJoin: true, room, canPublish: true, canSubscribe: true, canPublishData: true, canSubscribeData: true, }); ----------------------------------- const toggleUnmute = (e, track) => { e.stopPropagation(); console.log(track); socket.emit("togglemic", { uid: track.participant.identity,sid: track.participant.sid, value: !isMicrophoneEnabled }); }; Why I am not able to mute any other participant.....????

How can we analyze call quality and noise levels in js sdk client calls ? Any tools or scripts or any present feature in web sdk client ?

Hello LiveKit community! I'm having trouble connecting to LiveKit Cloud from a Python agent using livekit-agents v1.1.4. My browser clients can connect fine, but the Python SDK fails with a region info retrieval error. ## Error Details I keep getting this error when trying to connect: ERRORlivekitlivekit_ffi:serverroom283livekit ffiserver:room - error while connecting to a room: engine: signal failure: failed to retrieve region info: error decoding response body: expected value at line 1 column 1 ## What I've Found So Far 1. My LiveKit dashboard shows successful browser connections to "guru-room" with region "US Central" 2. When I curl the metadata endpoint, I get a 401 error: curl -i https://gurued-9x98jp6w.livekit.cloud/rtc HTTP/2 401 content-type: text/plain; charset=utf-8 ... missing authorization header ## What I've Tried 1. Using HTTPS URL in the environment variable (vs WSS) 2. Using WSS URL with explicit /rtc path 3. Validated token generation using the official LiveKit Server SDK 4. Made sure the room exists and is accessible from web clients ## My Setup - Python backend using livekit-agents v1.1.4 with Gemini integration - Environment variables properly configured (API key/secret verified) - Token generation using the official SDK with proper claims - React/Next.js frontend (which can connect successfully) Any advice on how to resolve the "failed to retrieve region info" error? Do I need special configuration for Python agent connections? Perhaps a way to authenticate the metadata endpoint request? Thank you!

Hi Team, I have one query related to: https://www.npmjs.com/package/@livekit/noise-cancellation-node Can this plugin be used in JS SDK Web client to add Noise cancellation feature ,If we dont; have krisp ? Can we simply install its nmp pkg and use it on client JS SDK code to remove noise?

how can i use user_away_timeout i got No parameter named "user_away_timeout"basedpyrightreportCallIssue Ctrl+click to open in new tab (function) user_away_timeout: Unknown

Hey channel, how i can capture the current tab and the voice incoming from a mic? Seems like im having problems capture both interactions, obvioussly the system audiotrack comes from the agent. And the mic audiotrack comes from the mic of the user. But when i record, i just here the tab audio. And not the mic audio

Hi Team, We are getting lots of error for Uncaught “negotiation timed out” We are using React SDK

"@livekit/components-react": "^2.9.12"
"@livekit/components-styles": "^1.1.6"
"livekit-client": "^2.15.0"

Hi LiveKit team, I asked the bot but it came up short... I noticed that the ability to set a reply-to ID when sending text messages is inconsistent across SDKs: - Python SDK: send_text() accepts reply_to_id parameter - JavaScript SDK: SendTextOptions has replyToMessageId commented out - Flutter SDK: SendTextOptions doesn't include replyToStreamId at all Since sendText internally uses streamText (which supports replies in all SDKs), and reply functionality is just as relevant for immediate messages as streamed ones, why isn't reply support uniformly available in sendText across all SDKs? Is this a deliberate design decision, perhaps to deprecate some options? Is there some reason we should avoid the extra message properties functionality of streamText? Thanks!

if i dont add a RoomAgentDispatch when creating the room in LiveKitAgents, does that meant the agent wont join automatically upon participant entrance?

Hi, Is it possible to synchronise data channel with video in livekit webrtc, for example based on pts? Similar to how AV sync works

does react client sdk has support for adjusting audio playback speed?

Hi. I wonder what exactly is "connection quality" as displayed by ConnectionQualityIndicator? Is it a value computed by the LiveKit server? What parameters are used to computer it?

Hi all! Building a voice agent in Electron. I just wanted to ask in your experience, how long does it take for a client (user) to connect to the room? I'm getting ~1-2 seconds of delay, which isn't ideal for the wake word process as users would have to wait quite a while before being connected and able to speak with the agent. Thanks!

I am having trouble switching cameras's and microphone sources, it has worked, but isn't now, has anyone run into this?

We've had a few issues recently where the livekit client reports that a track was published (camera, screen share, mic) but the livekit server doesn't register that it is published and other participants can't access it. When the user first joined the room, everything was ok but at some point, new tracks wouldn't publish to the server. Rejoining the room correct the problem. Other users could publish new tracks, this only affecting a single user. Session: RM_eSYKiYnkzaou Participant: ########-.c3ba-410d-bd76-188191fad442 Room started at @ 2025-07-14 10:05am PT and we started seeing the issue around 10:20am.

hey 🙂 in our web app in which we are recording sessions between ai agent and participant we are trying to make a transcription with word level timestamps relative to the beginning of the recording. in order to get transcriptions we are doing:

room.on(RoomEvent.TranscriptionReceived, onTranscriptionReceived)

each

TranscriptionSegment

has the

startTime

and

endTime

which seems to be always

0

(why?) but two other fields:

firstReceivedTime

and

lastReceivedTime

have absolute timestamps that probably we could use. however, how do we get the timestamp of the recording start so we can create a relative timestamps for every word? thanks!