After the SIP call is connected, the agent speaks too fast, causing the first few words to be inaudible.

It seems likely that the issue with the agent speaking too fast after a SIP call connection could be related to the TTS (Text-to-Speech) settings or the initial setup of the agent. According to the LiveKit documentation, you can adjust the speech speed using the

speed

parameter in the TTS configuration. This parameter controls the average speech speed in characters per second, which might help in slowing down the initial speech output. Additionally, if the issue persists, it might be beneficial to check the agent's configuration for any specific settings related to speech speed or initial connection handling. You can also explore the LiveKit Slack discussions for insights on how others have addressed similar issues with TTS speed adjustments. Would you like more detailed guidance on adjusting these settings? Sources: - LiveKit documentation - LiveKit Slack discussions 👍 0 button 👎 0 button 💡 Hint: Mention @runllm in the thread for followups.

@millions-hair-24956 It's not that the voice is fast, it's that the first 1-2 characters at the beginning can't be heard.

It seems likely that the issue with the first 1-2 characters being inaudible might be related to the initial audio buffering or the start of the TTS playback. This can sometimes happen if the audio stream is not fully ready when playback begins. According to the LiveKit documentation, you might want to ensure that the

auto_playout

parameter is set to

True

to automatically start the transcription once the first audio frame is received. This can help in synchronizing the start of the audio playback more effectively. Additionally, you might want to check if there are any network latency issues or initial buffering delays that could be causing the start of the audio to be clipped. Ensuring that the network connection is stable and that the TTS engine is properly initialized before starting the playback might also help. Would you like more detailed guidance on how to address this issue? Sources: - LiveKit documentation 👍 0 button 👎 0 button 💡 Hint: Mention @runllm in the thread for followups.

Can you help me, please? @refined-appointment-81829

Can you tell me more about your setup?

https://github.com/livekit-examples/voice-pipeline-agent-python/blob/main/agent.py I didn't make any changes, just changed stt and tts to azure. After entering the room using SIP, the sound of the first 1-2 characters cannot be heard

Hmm, not sure why that is happening..