@millions-hair-24956 what can cause automatic disconnecting of all participants from a room?

Automatic disconnection of all participants from a room can occur due to several reasons. According to LiveKit documentation, common disconnection reasons include server-initiated actions such as the room being closed using the DeleteRoom API, a participant being removed with the RemoveParticipant API, or network issues leading to a JOIN_FAILURE. Additionally, if the room is closed because all participants leave, a ROOM_CLOSED event is triggered. Network changes, such as switching from WiFi to cellular, can also cause temporary disconnections, but LiveKit attempts to automatically reconnect in such cases. If reconnection fails, a full reconnection process is initiated, which might be more disruptive. Would you like more detail on any specific disconnection scenario? Sources: - LiveKit Disconnection 👍 0 button 👎 0 button 💡 Hint: Mention @runllm in the thread for followups.

@millions-hair-24956 if none of these reasons is happinening what can be the problem?

It seems likely that if none of the typical reasons for disconnection are occurring, the issue might be related to network conditions or infrastructure setup. According to LiveKit's distributed setup documentation, network issues such as packet loss, high latency, or misconfigured firewalls can lead to disconnections. Additionally, if you are self-hosting, ensure that your server's resources (CPU, RAM, bandwidth) are sufficient and that Redis is properly configured for multi-node routing. Another potential issue could be related to specific ISP or firewall settings, as discussed in a Slack conversation, where certain network configurations caused higher rates of connection quality loss. Running network tests like the LiveKit Connection Test and WebRTC Browser Test can help diagnose these issues. Would you like more detailed troubleshooting steps? Sources: - Slack Conversation on Network Issues - LiveKit's distributed setup documentation 👍 0 button 👎 0 button 💡 Hint: Mention @runllm in the thread for followups.

@millions-hair-24956

async def _eg_single_segment(
    text: str,
    tts_forwarder: transcription.TTSSegmentsForwarder,
    tts_aws: tts.TTS,
    source: rtc.AudioSource,
    room: rtc.Room,
):
    """Transcription example without streaming (single string)"""

    <http://logger.info|logger.info>("pushing text %s", text)
    tts_forwarder.push_text(text)
    tts_forwarder.mark_text_segment_end()

    playout_q = asyncio.Queue[Optional[rtc.AudioFrame]]()
    playout_task = asyncio.create_task(_playout_task(tts_forwarder, playout_q, source))

    async for output in tts_aws.synthesize(text):
        tts_forwarder.push_audio(output.frame)
        playout_q.put_nowait(output.frame)

    tts_forwarder.mark_audio_segment_end()
    playout_q.put_nowait(None)

    await playout_task

if i am pushing multiple audio with tts forwarder in the same time can i filter the audio in the frontend based on th elanguage?

It seems likely that filtering audio based on language in the frontend can be achieved by utilizing the language metadata associated with each audio segment. According to the LiveKit documentation, the

TTSSegmentsForwarder

class allows you to specify a language for each transcription segment. This language information can be used to filter audio on the frontend. To implement this, you would need to ensure that each audio segment pushed to the

TTSSegmentsForwarder

includes the appropriate language metadata. On the frontend, you can then use this metadata to selectively play or display audio segments based on the desired language. Would you like more detailed guidance on how to implement this filtering mechanism? Sources: - LiveKit documentation 👍 0 button 👎 0 button 💡 Hint: Mention @runllm in the thread for followups.

@millions-hair-24956 yes give me more detailed guidance on how to implement this filtering mechanism

I'm not able to find any data sources to answer your question. You might want to provide more details about the specific frontend technology or framework you are using. If this doesn't help, please reach out to the Livekit support team for further assistance. 👍 0 button 👎 0 button 💡 Hint: Mention @runllm in the thread for followups.

@millions-hair-24956 i want to filter tts audio using reat

I'm not able to find any data sources to answer your question. You might want to specify how the language metadata is being transmitted to the frontend or how the audio is being managed in your React application. If this doesn't help, please reach out to the Livekit support team for further assistance. 👍 0 button 👎 0 button 💡 Hint: Mention @runllm in the thread for followups.

@millions-hair-24956

# Prepare a forwarder for TTS events
            tts_forwarder = transcription.TTSSegmentsForwarder(
                room=ctx.room,
                participant=ctx.room.local_participant,
                language=lang_code,
            )

To filter TTS audio based on language in a React application, you can follow these steps: 1. Ensure Language Metadata is Set: When initializing the

TTSSegmentsForwarder

, make sure the

language

parameter is set correctly for each segment, as shown in your code snippet. This metadata will be crucial for filtering on the frontend. 2. Transmit Language Metadata to Frontend: Ensure that the language metadata is transmitted alongside the audio data to the frontend. This can be done by including the language information in the events or data structures that carry the audio frames. 3. React Component for Audio Playback: In your React application, create a component that listens for audio events and filters them based on the language. You can use state to manage the current language filter and conditionally render audio elements. 4. Implement Filtering Logic: Use the language metadata to filter audio segments. For example, you can maintain a list of audio segments and their associated languages, and only render or play those that match the selected language. Here's a basic example of how you might implement this in React:

import React, { useState, useEffect } from 'react';

function AudioPlayer({ audioSegments, selectedLanguage }) {
  const [filteredSegments, setFilteredSegments] = useState([]);

  useEffect(() => {
    // Filter audio segments based on the selected language
    const filtered = audioSegments.filter(segment => segment.language === selectedLanguage);
    setFilteredSegments(filtered);
  }, [audioSegments, selectedLanguage]);

  return (
    <div>
      {filteredSegments.map((segment, index) => (
        <audio key={index} controls>
          <source src={segment.audioUrl} type="audio/wav" />
          Your browser does not support the audio element.
        </audio>
      ))}
    </div>
  );
}

export default AudioPlayer;

In this example,

audioSegments

is an array of objects where each object contains an

audioUrl

and

language

. The

selectedLanguage

is the language you want to filter by. The component filters the segments and renders only those that match the selected language. Would you like more details on any specific part of this implementation? Sources: - https://docs.livekit.io/agents/voice-agent/transcriptions/ - https://docs.livekit.io/python/livekit/agents/transcription/tts_forwarder.html#args 👍 0 button 👎 0 button 💡 Hint: Mention @runllm in the thread for followups.

@millions-hair-24956

rack = rtc.LocalAudioTrack.create_audio_track("agent-mic", source)
            options = rtc.TrackPublishOptions(source=rtc.TrackSource.SOURCE_MICROPHONE)

What this mean? if i changedthe neame should be tracked differently from the frontend?

If you're looking for further help, please follow-up on one of the other channels available for community support!