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I am not sure if this is the right place to ask about this as I am currently using Cloud LiveKit, but I noticed that Agent using default OpenAI TTS lags very noticeably when calling over phone via SIP. For me it sounds as if audio chunks got re-encoded and because of that contain artifacts at places where chunk ended, but speech didn't.
I of course don't know for sure.
I've tried 2 different SIP trunk providers (Vonage and Twilio) and the results where exactly the same.
Is there anything I can do to debug and resolve this?
Thanks
The text was updated successfully, but these errors were encountered:
there wouldn't be any differences if you are running over telephony or not.. the main thing that will influence latency is where you are running your agents. try to ensure the agent is close to inference servers to minimize latency
I am not sure if this is the right place to ask about this as I am currently using Cloud LiveKit, but I noticed that Agent using default OpenAI TTS lags very noticeably when calling over phone via SIP. For me it sounds as if audio chunks got re-encoded and because of that contain artifacts at places where chunk ended, but speech didn't.
I of course don't know for sure.
I've tried 2 different SIP trunk providers (Vonage and Twilio) and the results where exactly the same.
Is there anything I can do to debug and resolve this?
Thanks
The text was updated successfully, but these errors were encountered: