livekit-plugins-baseten 1.5.12


pip install livekit-plugins-baseten

  Latest version

Released: May 21, 2026


Meta
Author: LiveKit
Requires Python: >=3.10.0

Classifiers

Intended Audience
  • Developers

License
  • OSI Approved :: Apache Software License

Programming Language
  • Python :: 3
  • Python :: 3 :: Only
  • Python :: 3.10

Topic
  • Multimedia :: Sound/Audio
  • Multimedia :: Video
  • Scientific/Engineering :: Artificial Intelligence

Baseten plugin for LiveKit Agents

Support for Baseten-hosted models in LiveKit Agents, including STT (Speech-to-Text), TTS (Text-to-Speech), and LLM (Large Language Model) integrations.

Installation

pip install livekit-plugins-baseten

Pre-requisites

You'll need an API key from Baseten. It can be set as an environment variable: BASETEN_API_KEY

You also need to deploy a model to Baseten and will need your model endpoint to configure the plugin.

STT (Speech-to-Text)

The STT plugin connects to Baseten's Whisper Streaming WebSocket endpoint for real-time transcription. It works with both truss and chain deployments.

Recommended model

Whisper v3 Turbo – WebSocket

Endpoint URL formats

Deployment type URL pattern
Truss wss://model-{model_id}.api.baseten.co/environments/production/websocket
Chain wss://chain-{chain_id}.api.baseten.co/environments/production/websocket

Basic usage

You can specify the endpoint in three ways:

from livekit.plugins import baseten

# 1. Using a truss model ID (recommended for truss deployments)
stt = baseten.STT(
    api_key="your-baseten-api-key",  # or set BASETEN_API_KEY env var
    model_id="your-model-id",
    language="en",
)

# 2. Using a chain ID (recommended for chain deployments)
stt = baseten.STT(
    api_key="your-baseten-api-key",
    chain_id="your-chain-id",
    language="en",
)

# 3. Using a full endpoint URL (for custom routing or deployment URLs)
stt = baseten.STT(
    api_key="your-baseten-api-key",
    model_endpoint="wss://model-{model_id}.api.baseten.co/environments/production/websocket",
    language="en",
)

Configuration options

Parameter Default Description
api_key BASETEN_API_KEY env var Baseten API key
model_endpoint BASETEN_MODEL_ENDPOINT env var Full WebSocket URL (takes priority over model_id/chain_id)
model_id Baseten truss model ID; auto-constructs the endpoint URL
chain_id Baseten chain ID; auto-constructs the endpoint URL
language "en" BCP-47 language code (use "auto" for auto-detection)
encoding "pcm_s16le" Audio encoding (pcm_s16le or pcm_mulaw)
sample_rate 16000 Audio sample rate in Hz
enable_partial_transcripts True Emit interim transcripts while the speaker is talking
partial_transcript_interval_s 1.0 Interval (seconds) between partial transcript updates
final_transcript_max_duration_s 30 Max seconds of audio before forcing a final transcript
show_word_timestamps True Include word-level timestamps in results
vad_threshold 0.5 Server-side VAD speech probability threshold (0.0–1.0)
vad_min_silence_duration_ms 300 Minimum silence (ms) to mark end of speech
vad_speech_pad_ms 30 Padding (ms) added around detected speech

Full voice pipeline example

import os
from livekit import agents
from livekit.agents import AgentSession, Agent, RoomInputOptions
from livekit.plugins import baseten, openai, noise_cancellation, silero
from livekit.plugins.turn_detector.multilingual import MultilingualModel

BASETEN_API_KEY = os.getenv("BASETEN_API_KEY")
whisper_model_id = "your-whisper-model-id"  # or use chain_id for chain deployments
orpheus_model_id = "your-orpheus-model-id"


class Assistant(Agent):
    def __init__(self) -> None:
        super().__init__(instructions="You are a helpful voice AI assistant.")


async def entrypoint(ctx: agents.JobContext):
    session = AgentSession(
        stt=baseten.STT(
            api_key=BASETEN_API_KEY,
            model_id=whisper_model_id,  # or chain_id="your-chain-id"
            language="en",
            enable_partial_transcripts=True,
        ),
        llm=openai.LLM(
            api_key=BASETEN_API_KEY,
            base_url="https://inference.baseten.co/v1",
            model="openai/gpt-oss-120b",
        ),
        tts=baseten.TTS(
            api_key=BASETEN_API_KEY,
            model_endpoint=(
                f"https://model-{orpheus_model_id}"
                ".api.baseten.co/environments/production/predict"
            ),
        ),
        vad=silero.VAD.load(),
        turn_detection=MultilingualModel(),
    )

    await session.start(
        room=ctx.room,
        agent=Assistant(),
        room_input_options=RoomInputOptions(
            noise_cancellation=noise_cancellation.BVC(),
        ),
    )

    await session.generate_reply(
        instructions="Greet the user and offer your assistance."
    )


if __name__ == "__main__":
    agents.cli.run_app(agents.WorkerOptions(entrypoint_fnc=entrypoint))

TTS (Text-to-Speech)

The TTS plugin calls Baseten-hosted TTS models (e.g. Orpheus 3B) over HTTP.

tts = baseten.TTS(
    api_key="your-baseten-api-key",
    model_endpoint="https://model-{model_id}.api.baseten.co/environments/production/predict",
    voice="tara",
    language="en",
)

LLM (Large Language Model)

The LLM plugin wraps Baseten's OpenAI-compatible inference endpoint.

llm = baseten.LLM(
    api_key="your-baseten-api-key",
    model="openai/gpt-oss-120b",
)

Documentation

Extras: None
Dependencies:
aiohttp
livekit
livekit-agents (>=1.5.12)