hello im using conversational ai of eleven, i want to know if i use my gpt key and llm , so what ar

[wary tapir]

AI-generated Answer for vinayakarya_58055

ElevenLabs' Conversational AI platform currently covers the costs of LLMs, but these charges will eventually be passed through to customers in the future. For now, you are only billed for the usage of the platform itself, which includes features like speech-to-text, text-to-speech, and turn-taking.

Pricing is based on the subscription plan you choose, with different tiers offering varying amounts of included minutes and costs for extra usage. For enterprise-level usage, you can contact the sales team to discuss custom pricing tailored to your needs.

Additionally, ElevenLabs offers a startup grant program that may provide discounted or free access to their services during your development phase. This could be a great option for your early-stage startup. Let me know if you need further clarification!

[steady bobcat]

no i want to know if i use my custom llm and got key will it reduce the cost

[wary tapir]

AI-generated Answer for vinayakarya_58055

Using your own custom LLM and OpenAI API key does not directly reduce the cost of using ElevenLabs' Conversational AI. The platform charges 1,000 credits per minute for production calls, regardless of whether you use ElevenLabs' internal credentials or integrate your own custom LLM. Additionally, ElevenLabs currently covers the LLM costs, but this may change in the future.

For enterprise-level usage or customized pricing, you can reach out to the Enterprise Team to discuss a plan tailored to your needs.

[steady bobcat]

if i do a min long converssation ai agent call. what will be the charges

[wary tapir]

AI-generated Answer for vinayakarya_58055

If you use ElevenLabs' Conversational AI, the cost for a one-minute call is 1,000 credits per minute. This applies regardless of whether you use your own GPT key and custom LLM or the default options provided by ElevenLabs. Currently, ElevenLabs covers the cost of the LLM, but this may change in the future, and any updates will be communicated.

Using your own LLM or GPT key does not reduce the per-minute charge, as the 1,000 credits per minute fee is for the overall service, including features like speech-to-text, text-to-speech, and turn-taking.

[steady bobcat]

without call just conversation like normla no numebr callign

[wary tapir]

AI-generated Answer for vinayakarya_58055

The cost for using ElevenLabs Conversational AI calculated at 1,000 credits per minute during interactions, regardless of whether it's a phone call or a regular conversation. This includes the entire duration of the interaction, not just the time when speech is generated or processed. Even pauses or moments without active speech are counted toward the credit usage.

If you use your own custom LLM and GPT key, the charges for the Conversational AI interaction remain the same, as the cost structure is based on the duration of the interaction rather than the specific LLM used. However, any additional costs associated with your custom LLM (e.g., GPT API usage) would be separate and handled by your own account with the LLM provider.

For a one-minute interaction, the charge would be 1,000 credits. If you are on the Business plan, for example, this equates to $0.08 per minute (annual pricing). You can find more details about pricing tiers on the Pricing Page.

[steady bobcat]

but for some of my calls it chargesd nly 250 creds of one minute

[wary tapir]

AI-generated Answer for vinayakarya_58055

The cost structure for ElevenLabs' Conversational AI is as follows:

• Setup & Prompt Testing: 500 credits per minute.
• Production Calls: 1,000 credits per minute.

These charges apply to the entire duration of the call, from start to finish, regardless of whether speech is actively generated or processed. If pauses or holds occur during the call, the time is still counted toward credit usage.

If you noticed a charge of only 250 credits for a one-minute call, it might be related to a specific setup or testing scenario, as testing costs are lower at 500 credits per minute. However, for production calls, the standard rate is 1,000 credits per minute.

Currently, ElevenLabs covers the cost of the Large Language Model (LLM), but this may change in the future. Using your own custom LLM or GPT key is not mentioned as a way to reduce costs in the provided information. If you need further clarification or want to explore custom solutions, you can reach out to the Enterprise Team.

[steady bobcat]

wss://api.elevenlabs.io/v1/text-to-speech/:voice_id/stream-input when i use this i get uido in what format

[wary tapir]

AI-generated Answer for vinayakarya_58055

When using the WebSocket endpoint wss://api.elevenlabs.io/v1/text-to-speech/:voice_id/stream-input, the audio format is determined by the output_format query parameter you specify during the request. By default, the format is mp3_44100, but you can choose from several options depending on your needs. Here are the available formats:

• mp3_22050_32: MP3 with 22.05kHz sample rate at 32kbps.
• mp3_44100_32: MP3 with 44.1kHz sample rate at 32kbps.
• mp3_44100_64: MP3 with 44.1kHz sample rate at 64kbps.
• mp3_44100_96: MP3 with 44.1kHz sample rate at 96kbps.
• mp3_44100_128: Default MP3 format with 44.1kHz sample rate at 128kbps.
• mp3_44100_192: MP3 with 44.1kHz sample rate at 192kbps (requires Creator tier or above).
• pcm_16000: PCM format (S16LE) with 16kHz sample rate.
• pcm_22050: PCM format (S16LE) with 22.05kHz sample rate.
• pcm_24000: PCM format (S16LE) with 24kHz sample rate.
• pcm_44100: PCM format (S16LE) with 44.1kHz sample rate (requires Pro tier or above).
• ulaw_8000: μ-law format with 8kHz sample rate, commonly used for Twilio audio inputs.

Make sure to specify the desired format in the output_format parameter when initiating the WebSocket connection.

[steady bobcat]

how to define that

[wary tapir]

AI-generated Answer for vinayakarya_58055

To define the audio format when using the WebSocket endpoint (wss://api.elevenlabs.io/v1/text-to-speech/:voice_id/stream-input), you can specify the output_format parameter in the query string. For example:

wss://api.elevenlabs.io/v1/text-to-speech/:voice_id/stream-input?model_id={model}&output_format=mp3_44100

Here, output_format=mp3_44100 ensures the audio is returned in MP3 format with a 44.1kHz sample rate. You can choose other formats depending on your requirements, such as PCM or μ-law, as supported by the API. For more details on available formats, refer to the supported audio formats.

[steady bobcat]

hey if createed a conersational ai from a. account but i want to accees it from different api key of differenct account

[wary tapir]

AI-generated Answer for vinayakarya_58055

It is not possible to access a conversational AI agent created under one account using the API key of a different account. Each API key is tied to the specific account it belongs to, and resources like conversational AI agents are not shared across accounts.

If you need to collaborate or share access, you may need to manage everything under a single account or explore options like an Enterprise plan, which might offer more flexibility. For further assistance, you can reach out to the Enterprise Team.

[steady bobcat]

in text to speechwebscoket it closes after 20 ecobd if no text input i want to stip this

[wary tapir]

AI-generated Answer for vinayakarya_58055

The Text-to-Speech WebSocket connection closes after 20 seconds of inactivity by default. To prevent this, you can use the following methods:

1. Send a Single Space Character (" "): You can send a single space character periodically to keep the WebSocket connection open. Ensure that the string includes a space, as sending an empty string ("") will send an End of Sequence (EOS) message, which closes the connection. You must continue sending additional single space characters to maintain the connection.

2. Set the inactivity_timeout Parameter: ElevenLabs allows you to set the inactivity_timeout parameter to extend the timeout period. You can adjust this value up to a maximum of 180 seconds. This feature is detailed in the WebSockets API documentation.

If you need further assistance with implementing this, you can refer to the WebSockets API Reference for more details.

[steady bobcat]

im using websocekt text to speech , now i want to control the stbilityand all its speed and everityhtin

[wary tapir]

AI-generated Answer for vinayakarya_58055

Currently, ElevenLabs does not offer direct control over the speed of generated speech through the text-to-speech WebSocket API. However, you can adjust the stability and similarity settings to influence the voice's tone and delivery. These adjustments may indirectly affect the speaking rate, but results can vary.

If you need precise control over the speech speed, you can generate the audio first and then use external audio editing software to modify the speed to your preference.

Regarding the WebSocket closing after 20 seconds of inactivity, this behavior is likely a timeout mechanism. Unfortunately, there isn't a way to disable this timeout directly. To keep the connection alive, you could send periodic "ping" or "keep-alive" messages to the WebSocket server.

Let me know if you need further clarification!

[steady bobcat]

no im using websocekt i wnat to use that

[wary tapir]

AI-generated Answer for vinayakarya_58055

To control aspects like stability, speed, and other settings in your text-to-speech output using ElevenLabs, you can adjust the following parameters:

1. Stability and Similarity Settings: Lower the stability setting to around 60%-80% to influence pacing and experiment with the similarity setting to refine the voice's characteristics. These adjustments can help achieve slower and more consistent speech.

2. Text Input: Break your text into smaller sections, ideally under 1000 characters, to maintain consistent speed and quality.

3. Punctuation: Use punctuation strategically, such as commas and periods, to control the flow and slow down speech.

4. Voice Model: Try using models like Eleven Turbo V2.5 or Eleven English V2 for better control over pacing.

5. External Editing: If the internal settings don't fully meet your needs, you can use external audio editing software to adjust the speed after generating the audio.

Regarding the WebSocket closing after 20 seconds of inactivity, this behavior is likely a timeout mechanism. Unfortunately, the provided content does not include specific instructions for modifying this timeout. You may need to ensure continuous input or consult the API References for more details on WebSocket behavior and customization options.

[steady bobcat]

how to do that in websocket

[wary tapir]

AI-generated Answer for vinayakarya_58055

To prevent the Text-to-Speech WebSocket connection from closing after 20 seconds of inactivity, you can use the inactivity_timeout parameter or send a single space character " " periodically to keep the connection alive. Here's how you can manage this:

1. Set the inactivity_timeout Parameter

You can configure the inactivity_timeout parameter when initializing the WebSocket connection. This parameter allows you to set the timeout duration up to a maximum of 180 seconds. Refer to the WebSockets API documentation for details on how to implement this.

2. Send a Single Space Character

If you want to keep the WebSocket open manually, you can send a single space character " " every few seconds. This prevents the connection from timing out. Note that sending an empty string "" will send an End of Sequence (EOS) message, which closes the connection.

Example Code for Sending a Space Character

import asyncio
import websockets

async def keep_connection_alive(websocket):
    try:
        while True:
            await websocket.send(" ")  # Send a single space character
            await asyncio.sleep(15)  # Send every 15 seconds (adjust as needed)
    except websockets.exceptions.ConnectionClosedError as e:
        print(f"WebSocket connection closed unexpectedly: {e}")
    except Exception as e:
        print(f"Error while keeping connection alive: {e}")

3. Adjust Stability and Similarity Boost

When sending text to the WebSocket, you can include voice_settings to control parameters like stability and similarity boost. Example:

await websocket.send(json.dumps({
    "text": "Your text here",
    "voice_settings": {
        "stability": 0.5,  # Adjust stability (0.0 to 1.0)
        "similarity_boost": 0.8  # Adjust similarity boost (0.0 to 1.0)
    }
}))

Summary

• Use inactivity_timeout to extend the timeout duration.
• Periodically send " " to keep the connection alive.
• Include voice_settings in your payload to control stability and other parameters.

Let me know if you need further clarification!

[steady bobcat]

if i send a text to elveln balnb sin text to speech its not gives e the auido of full end its missing the ast part

[wary tapir]

AI-generated Answer for vinayakarya_58055

To ensure the full audio is generated when using the ElevenLabs Text-to-Speech WebSocket API, follow these steps:

1. End of Sequence Handling: After sending all your text chunks, send an empty string as the final message. This acts as an End of Sequence (EOS) signal, letting the API know that the audio generation process is complete.

2. Buffer Management: Double-check that your buffer is correctly managed and that all text chunks are sent properly. Ensure the final chunk of text is transmitted without any issues.

3. Response Handling: Make sure your code processes WebSocket responses accurately. Intermediate audio chunks may not contain all fields, but the final response should include all necessary information.

4. Chunk Length Schedule: Experiment with different values for the chunk_length_schedule parameter to see if it impacts the completeness of the audio output.

If the issue persists, review your implementation for any errors or share more details about your setup for further troubleshooting.

[steady bobcat]

can we use flush

[wary tapir]

AI-generated Answer for vinayakarya_58055

Yes, you can use the flush parameter in the WebSocket API to help ensure the audio connection remains active and processes the full text properly. Additionally, to address the issue of missing the last part of the audio, follow these steps:

1. Send an Empty String at the End: Ensure you send an empty string as the final message to indicate the end of the audio generation sequence. This helps the API finalize the audio output.

2. Buffer Management: Verify that your buffer is correctly managed and all text chunks are sent properly. Double-check the final chunk of text to confirm it is transmitted correctly.

3. Adjust chunk_length_schedule: Experiment with different values for the chunk_length_schedule. Larger chunks can improve stability and cadence, while smaller chunks may reduce quality due to less context.

4. Use the Flush Parameter: The flush parameter can be used to keep the audio connection active and ensure the context from previous generations is utilized effectively. This can help avoid issues with missing audio segments.

If the problem persists, you may need to review your implementation or share additional details about your setup to identify potential issues.

[steady bobcat]

elvenlabs how to know call is endned conversational ai or we are a t en of converdation

[wary tapir]

AI-generated Answer for vinayakarya_58055

Currently, ElevenLabs does not have a built-in feature to trigger an event or webhook when a call ends in the Conversational AI system. If you're using a platform like Twilio, you can rely on Twilio's event-based webhooks to track when a call starts, ends, or changes status. This is the most reliable way to monitor call completion and gather related data.

If you're looking for a way to end a conversation within the ElevenLabs system, you can use the End Call tool. This tool allows the agent to terminate the call manually. To add this tool, go to the agent's settings, scroll to the "Tools" section, and add the End Call option if it's not already included.

For more advanced needs, such as triggering specific actions tied to call completion, you might need to implement custom middleware or explore tailored solutions by contacting the Enterprise Team.

[steady bobcat]

in elvelnlabs tts websocekt the auio donest comes same

[wary tapir]

AI-generated Answer for vinayakarya_58055

If the audio generated through the ElevenLabs Text-to-Speech WebSocket API does not match expectations, there are several factors to check:

1. Buffer System: The WebSocket service uses a buffer system to optimize audio quality. Audio generation occurs when the buffer reaches a certain size, as longer text inputs provide better context for delivery. If smaller chunks are sent, the audio quality may be reduced. You can use the flush command to empty the buffer and generate audio immediately, but this may impact quality. Learn more about buffering in the WebSocket API documentation.

2. Settings and Parameters: Ensure that the WebSocket initialization settings, such as output_format, sync_alignment, and apply_text_normalization, are configured correctly. For example, using the auto_mode parameter can reduce latency but may lower audio quality when sending partial phrases. Review the available parameters in the WebSocket API documentation.

3. Audio Data Handling: Verify that the audio data is being processed correctly. Audio chunks are sent as raw binary data, not base64. Accumulate the chunks in a buffer and save them as an MP3 or WAV file once all chunks are received. Misinterpreting the format can lead to issues.

4. Voice Model and Settings: Confirm that the same voice model and settings (e.g., Stability and Clarity) are used consistently across the API and interface. Differences in these configurations can lead to variations in audio output.

If the issue persists, you may need to share specific details about your implementation, such as code examples or error messages, for further troubleshooting.

[steady bobcat]

im not getting isfinal ins eleb labs tts websoscket how i can trigger is fina

[wary tapir]

AI-generated Answer for vinayakarya_58055

To trigger the isFinal attribute in ElevenLabs' TTS WebSocket, you need to ensure proper handling of the End of Sequence (EOS) message. Here’s how you can achieve this:

1. Send an EOS Message: After sending all your text chunks, you must send an empty string ("") as the text field in your final message. This signals the server that the text input is complete. For example:

   { "text": "" }
   

   This will cause the server to return a final response where "isFinal": true and the audio field is null, indicating that audio generation is complete.

2. Avoid Using Flush Messages with Spaces: If you send a flush message like {"text": " ", "flush": true}, the server interprets this as an intermediate message, and isFinal will remain None. Instead, use the empty string as described above.

3. Keep the WebSocket Connection Open: If you want the connection to remain open during the session, ensure you manage inactivity appropriately. You can set the inactivity_timeout parameter to a higher value (up to 180 seconds) to prevent the connection from closing during pauses.

4. Handle Continuous Sessions: For long-lived sessions, you may need to re-establish a new connection for each text segment or implement your own logic to determine when a chunk is complete without relying on isFinal.

Let me know if you need further clarification!

[steady bobcat]

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