bulky-ocean-78340
07/22/2025, 7:22 AMtall-airline-16756
07/22/2025, 7:22 AMtall-airline-16756
07/22/2025, 7:25 AMrtc.AudioSource, set queue_size_ms to 1000-2000ms to prevent buffer underruns.
• Check Network Proximity: Make sure your agent is network-close to any speech-to-text (STT), text-to-speech (TTS), or language model (LLM) services to reduce latency.
If the delay persists, please provide more details about your setup (device, browser, SDK version, and any error logs) for further troubleshooting.
Sources: Understanding "silence detected on local audio track" warning | Speech & Audio Processing | livekit/livekit_composite | DeepWiki | CHANGELOG.md | An open source stack for real-time multimodal AIbulky-ocean-78340
07/22/2025, 7:31 AMtall-airline-16756
07/22/2025, 7:34 AMqueue_size_ms to 1000-2000ms in rtc.AudioSource. It's like giving your audio a little extra breathing room to prevent glitches.
• Adjust VAD Sensitivity: Fine-tune those Voice Activity Detection settings. Think of it like adjusting a sensitive microphone so it doesn't miss important moments.
• Implement Inactivity Handlers: Add some smart monitoring to catch and recover from those awkward silent moments. It's like having a helpful friend who nudges the conversation back on track.
• Check Network Proximity: Make sure your agent is hanging out close to its key services. Proximity matters!
If you can share more details about your setup, I'd be super happy to dive deeper and help you troubleshoot even more effectively!
Sources: Going beyond a single-core | Web and mobile frontends | LiveKit Docs | Bringing Zoom's end-to-end optimizations to WebRTC | An introduction to WebRTC Simulcast | delay-handling/raw_data.txt