@millions-hair-24956 i get this error from time to time, the agent strart speaking and the interrupt midway and i get this log: ERROR livekit.plugins.elevenlabs - 11labs reported an error: something_went_wrong

The error "11labs reported an error: something_went_wrong" typically indicates an issue with the ElevenLabs TTS service. This could be due to several reasons, such as network connectivity issues, API rate limits, or service disruptions. According to LiveKit documentation, ElevenLabs TTS can sometimes report errors if there are unusual activities or if the API key is not properly configured. Would you like more detail on troubleshooting this issue? Sources: - LiveKit documentation 👍 0 button 👎 0 button 💡 Hint: Mention @runllm in the thread for followups.

@millions-hair-24956 what is streaming latency in the TTS class of the elevenlabs plugin?

Streaming latency in the TTS class of the ElevenLabs plugin refers to the delay in seconds for streaming audio. According to the LiveKit documentation, the default value for streaming latency is set to 3 seconds. This parameter controls how quickly the audio is streamed after text is processed. Would you like more detail on how to adjust or utilize this parameter? Sources: - LiveKit documentation 👍 0 button 👎 0 button 💡 Hint: Mention @runllm in the thread for followups.

@millions-hair-24956 i don't understand, does that means that each stream have a 3 second latency? That doesn't make any sense

It seems likely that the 3-second streaming latency in the ElevenLabs TTS plugin is a configurable parameter that sets a buffer or delay for streaming audio, rather than indicating a fixed latency for each stream. This buffer can help manage the flow of audio data, ensuring smooth playback, especially in cases where network conditions might fluctuate. Would you like more detail on how this parameter affects audio streaming or how to adjust it for your needs? Sources: - https://livekit-users.slack.com/archives/C07FY8WHGPM/p1729014768.848989 👍 0 button 👎 0 button 💡 Hint: Mention @runllm in the thread for followups.

what encoding format do you suggest i use for phone calls? @millions-hair-24956

It seems likely that for phone calls, using the "pcmu" codec is recommended, especially when working with SIP trunks like Twilio, as it is widely supported and ensures compatibility. According to LiveKit documentation, Twilio only supports pcmu via SIP termination, which is a common choice for telephony applications. Would you like more detail on configuring this codec or exploring alternatives? Sources: - LiveKit SIP Participant Attributes 👍 0 button 👎 0 button 💡 Hint: Mention @runllm in the thread for followups.

Yes i am using a SIP trunk on twilio, can you givce me more details? @millions-hair-24956

Certainly! When using a SIP trunk with Twilio, the "pcmu" codec, also known as G.711 µ-law, is typically used for audio encoding. This codec is a standard in telephony and is supported by Twilio for SIP termination. Here’s a step-by-step breakdown of why and how to use this codec: 1. Compatibility: The pcmu codec is widely supported across various telephony systems, ensuring that your calls can be connected without compatibility issues. It is the default codec for many SIP providers, including Twilio. 2. Quality: While pcmu is a narrowband codec, it provides sufficient audio quality for standard voice calls. It operates at an 8 kHz sample rate, which is typical for traditional telephony. 3. Configuration: When setting up your SIP trunk with Twilio, ensure that your SIP endpoint is configured to use the pcmu codec. This might involve setting codec preferences in your SIP client or server configuration. 4. Limitations: If you require higher audio quality, you might consider using a provider that supports wideband codecs like G.722 or Opus. However, these are not supported by Twilio for SIP termination. 5. Testing: After configuring your SIP trunk, test your setup to ensure that audio is clear and that there are no compatibility issues. You can use tools like Wireshark to inspect SIP packets and verify that pcmu is being used. If you need further assistance with configuring your SIP trunk or have specific questions about your setup, feel free to ask! Sources: - https://docs.livekit.io/sip/ - https://livekit-users.slack.com/archives/C07FRP66XGE/p1729276971.829999 👍 0 button 👎 0 button 💡 Hint: Mention @runllm in the thread for followups.

so which one should i choose ? TTSEncoding = Literal[ "mp3_22050_32", "mp3_44100_32", "mp3_44100_64", "mp3_44100_96", "mp3_44100_128", "mp3_44100_192", "pcm_16000", "pcm_22050", "pcm_44100", ] @millions-hair-24956

For phone calls, using "pcm_16000" is generally a good choice. This format provides a balance between audio quality and compatibility with telephony systems. PCM (Pulse Code Modulation) is a raw audio format that is widely used in telephony, and a sample rate of 16 kHz is suitable for voice applications, offering better quality than the standard 8 kHz telephony audio. Would you like more detail on why this format is recommended or how to implement it? Sources: - https://livekit-users.slack.com/archives/C07FRP66XGE/p1729276971.829999 👍 0 button 👎 0 button 💡 Hint: Mention @runllm in the thread for followups.