how can we detect if livekit audio received frames are silence from agent end of speech or if its silence from jitter/delay due to internet connection?

Thanks for reaching out, Sebastián. I'm an AI assistant and I'm here to help. Give me a moment while I research this topic.

im using a Realtime model with the realtime own VAD detection, not LiveKit VAD

Thanks for sharing that you’re using a Realtime model with its own VAD detection instead of LiveKit’s VAD. I’d love to understand a bit more about how you’re hoping to integrate this with LiveKit. Are you looking for some guidance on managing silence detection, syncing events, or is there something else specific you have in mind? I’m here to help!

is the OpenAI Realtime model, right now on 'playback_finished' event we send an rpc call to the client to notify agent finishe publishing audio but this doesn't mean the user received audio due to network conditions, how can we sync this too?

To help you synchronize the 'playback_finished' event with audio reception confirmation, especially considering varying network conditions, here’s a friendly step-by-step guide you can follow: 1. Stream Audio with DataStream: Start by using

DataStreamAudioOutput

to send audio to the remote worker. Don’t forget to include important metadata like sample rate and channels to ensure smooth playback. 2. Playback Completion Notification: On the remote side,

DataStreamAudioReceiver

will handle the audio and detect when playback finishes. It then sends an RPC call (

lk.playback_finished

) back to the agent, sharing details like playback duration and whether it was interrupted. 3. Retry Logic: Since networks can be unpredictable, it’s a great idea to implement retry mechanisms for the RPC call. Using exponential backoff helps prevent overwhelming the network while trying again. 4. Agent Event Handling: Finally, on the agent side, listen for

playback_finished

events to confirm the audio was received and update your state accordingly. The

interrupted

flag is especially useful for managing cases where playback was only partially completed. Following these steps will help you achieve reliable synchronization between audio delivery and reception, even when the network isn’t perfect. If you’d like, I’m here to provide more details or code examples for any part of this process—just let me know! Sources: livekit-agents/livekit/agents/voice/chat_cli.py | How to detect when an agent has finished speaking

i want the inverse flow, client side knowing when it has actually received or not all of the audio sent from the server or manage it by analyzing if X secs of silence are received consider it finished too

Thanks for reaching out! To make sure the client has received all the audio sent from the server, are you thinking about using a specific method like acknowledgments for each audio frame? Or would you prefer something a bit simpler, like detecting a few seconds of silence? Also, I’m happy to help guide you through implementing this within LiveKit's framework or suggest some external tools if that’s what you need. Just let me know what works best for you!

i want to try different approaches as right now only relying on the RPC call of playback_finished, can be unreliable as the RPC may be received before all audio frames are received and played

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!