As a presenter without a VPN, using an up-to-date ...
# ask-ai
h
As a presenter without a VPN, using an up-to-date OS and browser, I'm experiencing packet loss on my microphone track. Any ideas or suggestions?
t
Thanks for reaching out, Alexandre. I'm an AI assistant and I'm here to help. Give me a moment while I research this topic.
Thanks — I hear you: you’re a presenter (no VPN) on an up‑to‑date OS/browser and you’re seeing packet loss on your microphone track. Let’s walk through focused checks and quick fixes. First checks (quick, do these now) • Run a connection test: use LiveKit’s ConnectionCheck to test WebRTC, TURN, and UDP vs TCP connectivity. This will show if packet loss is happening on the network path. • Check network path: switch from Wi‑Fi to a wired Ethernet connection or a different network (mobile hotspot) to see if packet loss improves. • Close heavy apps: quit other apps or browser tabs that use bandwidth or CPU (file sync, video streaming, VPN clients—even if not in use). • Restart browser and rejoin the room to rule out a transient browser/WebRTC issue. Audio-specific checks • Disable client-side noise suppression or advanced audio processing (browser/hardware settings, or app-level settings) to ensure the microphone isn’t being silenced by suppression. • Try a different microphone/headset to rule out hardware or device driver issues. • Lower sample rate or bitrate if your app exposes those settings (reduces bandwidth demand). Observability & metrics • Check participant-level stats in the session (packet loss %, jitter, RTT) from LiveKit’s SFU reports to confirm where loss occurs (sender → SFU or SFU → receivers). • Note timestamps and participant IDs when loss occurs so we can correlate with SFU metrics. If packet loss persists, please tell me: • Whether loss shows on your local publish (only your outgoing audio) or on others receiving you. • The results of ConnectionCheck (or a screenshot) and any SFU stats you can share (packet loss %, jitter, RTT). • Whether switching to wired or a different mic changed anything. I can then suggest targeted next steps (TURN changes, codec/bitrate tuning, or deeper network diagnostics). Sources: Understanding "silence detected on local audio track" warning | An open source stack for real-time multimodal AI | LiveKit | An open source stack for real-time multimodal AI | LiveKit | Why WebRTC beats WebSockets for realtime voice AI | LiveKit | Diagnosing Connection Errors with Connection Test Utility, with interactive elements
h
If you have a room session you can check ? It's a client presenter , not me in local
t
I can check the room session metrics for the presenter. Please provide the LiveKit room name or room ID and, if available, the presenter's participant ID (or display name) and approximate timestamps when the packet loss occurred. , with interactive elements
h
Room ID : RM_ZBHG7k9f5ECw Presenter ID : presenter-8R0mxpqGxkbuEFQGuHBqqfhP19m1
t
Thanks — I can check that presenter’s session metrics. To pull the right data, please provide one of the following (pick whichever is easiest): • A screenshot or copy of the presenter’s SFU stats around the incident (packet loss %, jitter, RTT, and timestamps), OR • Permission to access the room’s metrics now (if you want me to fetch them), OR • If you prefer to fetch them yourself, run LiveKit’s dashboard or admin API and share the presenter’s publish stats for the relevant time window. If you’ll share stats, please include the approximate timestamps when packet loss occurred so I can correlate them. , with interactive elements
h
Yes you have a permission
t
I don't have the answer you're looking for. You could also try asking your question: • in one of the other Slack channels or • to https://deepwiki.com/livekit/livekit_composite which is trained on all LiveKit source code If you find the answer, please post it here to help others!
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