Build low-latency voice agents with Higgs Realtime and Pipecat.
The pipecat-boson package exposes Higgs Realtime as a Pipecat
speech-to-speech LLMService. It receives live audio or text, manages the
conversation, calls tools, and streams audio or text responses. A voice
pipeline does not need separate STT, LLM, and TTS services.
- Python 3.11 or newer.
- A Boson API key.
- Access to the Higgs Realtime API.
- An existing Pipecat application with an audio transport.
Note: Keep the Boson API key on the server. Never embed it in a browser or mobile client.
Install the package from PyPI:
uv add pipecat-bosonThe equivalent pip command is pip install pipecat-boson.
The core service does not require WebRTC. Install the webrtc extra only when
you want to run the browser example or use Pipecat's WebRTC transport:
uv add "pipecat-boson[webrtc]"To develop the package or run its included example:
git clone git@github.com:boson-ai/pipecat-boson.git
cd pipecat-boson
uv sync --extra devTo use a local checkout from another uv project:
uv add --editable ../pipecat-bosonThe package supports pipecat-ai>=1.4.0,<2.
Set the API key, WebSocket endpoint, and model ID in your server environment:
export BOSON_API_KEY=bai-xxxx
export BOSON_REALTIME_URL=wss://api.boson.ai/v1/realtime/
export BOSON_REALTIME_MODEL=higgs-realtimeCreate the realtime service:
import os
from pipecat_boson.realtime import BosonRealtimeLLMService
llm = BosonRealtimeLLMService(
url=os.environ["BOSON_REALTIME_URL"],
api_key=os.environ["BOSON_API_KEY"],
model=os.environ["BOSON_REALTIME_MODEL"],
voice="default",
instructions="You are a concise and helpful voice assistant.",
)The following example assumes that transport is an existing Pipecat audio
transport:
from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.worker import PipelineParams, PipelineWorker
from pipecat.processors.aggregators.llm_context import LLMContext
from pipecat.processors.aggregators.llm_response_universal import (
LLMContextAggregatorPair,
)
from pipecat.workers.runner import WorkerRunner
async def run_bot(transport, llm):
context = LLMContext()
user_aggregator, assistant_aggregator = LLMContextAggregatorPair(
context,
realtime_service_mode=True,
)
pipeline = Pipeline(
[
transport.input(),
user_aggregator,
llm,
transport.output(),
assistant_aggregator,
]
)
worker = PipelineWorker(
pipeline,
params=PipelineParams(
enable_metrics=True,
),
)
runner = WorkerRunner()
await runner.add_workers(worker)
await runner.run()realtime_service_mode=True lets the context aggregators follow the
server-driven turn lifecycle. Do not add separate STT or TTS services around
BosonRealtimeLLMService.
Call run_bot(transport, llm) from your application's async entry point.
Higgs Realtime responds after server VAD detects the end of a user turn. If the
assistant should speak first, queue an LLMRunFrame after the client is ready,
as demonstrated by the included browser example.
From the repository checkout created above, copy the example environment file:
cp .env.example .envSet BOSON_API_KEY, BOSON_REALTIME_URL, and BOSON_REALTIME_MODEL in
.env, then start the WebRTC example:
uv run --extra webrtc \
python examples/pipecat_boson_realtime_agent.py \
-t webrtc \
--host 127.0.0.1 \
--port 7860Open http://localhost:7860 and connect your microphone.
If WebRTC ICE cannot reach the server, use the WebSocket transport:
uv run --extra webrtc \
python examples/pipecat_boson_realtime_agent.py \
-t websocket \
--host localhost \
--port 7860Select WebSocket in the page before connecting. Both commands use the
webrtc extra because it also installs the Pipecat runner used by the browser
example.
Set an input transcription model to receive finalized user transcripts as
Pipecat TranscriptionFrame objects:
llm = BosonRealtimeLLMService(
url=os.environ["BOSON_REALTIME_URL"],
api_key=os.environ["BOSON_API_KEY"],
model=os.environ["BOSON_REALTIME_MODEL"],
input_audio_transcription={
"model": "higgs-stt-3.1",
"language": "en",
},
)Omitting input_audio_transcription, passing None, or passing a dictionary
without a non-empty model suppresses client-facing user transcript events.
Higgs Realtime still understands the audio and can respond.
Declare an async Python function with typed arguments and return its result
through result_callback:
from pipecat.processors.aggregators.llm_context import LLMContext
from pipecat.services.llm_service import FunctionCallParams
async def get_weather(
params: FunctionCallParams,
location: str,
) -> None:
"""Get the current weather for a location.
Args:
location: City or place name.
"""
await params.result_callback(
{
"location": location,
"condition": "sunny",
"temperature_c": 22,
}
)
tools = [get_weather]
llm = BosonRealtimeLLMService(
url=os.environ["BOSON_REALTIME_URL"],
api_key=os.environ["BOSON_API_KEY"],
model=os.environ["BOSON_REALTIME_MODEL"],
instructions="Use get_weather when the user asks about weather.",
tools=tools,
)
context = LLMContext(tools=tools)Pass the same tool list to the service and the context. The service advertises
and registers the handlers for the Higgs Realtime session, while
LLMContext keeps the tool definitions with the conversation state. After the
function completes, Higgs Realtime continues the response with its result.
Server VAD is enabled by default. It detects the end of the user's turn, creates a response, and interrupts an active response when the user starts speaking. Override its thresholds only when the default behavior does not fit the application:
turn_detection = {
"type": "server_vad",
"prefix_padding_ms": 300,
"silence_duration_ms": 500,
"threshold": 0.55,
}Pass this dictionary as turn_detection=turn_detection when constructing the
service. For most voice agents, keep the default server VAD settings.
Higgs Realtime also supports OpenAI-compatible semantic VAD:
semantic_turn_detection = {
"type": "semantic_vad",
}
llm = BosonRealtimeLLMService(
url=os.environ["BOSON_REALTIME_URL"],
api_key=os.environ["BOSON_API_KEY"],
turn_detection=semantic_turn_detection,
)Pass output_modalities=["text"] when constructing the service. Text-only
sessions emit streamed LLMTextFrame objects and no audio frames.
The service supports exactly one session output modality: ["audio"] or
["text"]. Mixed output modalities and per-response modality overrides are not
supported.
Use Pipecat service event handlers to observe the Higgs Realtime session lifecycle:
def register_session_handlers(llm):
@llm.event_handler("on_session_created")
async def on_session_created(service, event):
print("Session:", event.session.id)
@llm.event_handler("on_session_terminated")
async def on_session_terminated(service, event_type, event):
print("Session terminated:", event_type)Call register_session_handlers(llm) before starting WorkerRunner. The
integration reports terminal session events but does not close the Pipecat
transport automatically.
Keep on_session_created handlers fast. Session setup waits for this handler
to return.
Connection options:
| Parameter | Default | Description |
|---|---|---|
url |
Required | Higgs Realtime WebSocket endpoint. |
api_key |
Required for the hosted API | Boson API key sent as a Bearer token. |
model |
"higgs-realtime" |
Realtime model ID sent when the session is configured. |
Optional session settings supported by Higgs Realtime:
| Parameter | Default | Description |
|---|---|---|
voice |
"default" |
Voice preset or voice ID used for audio output. |
instructions |
Helpful assistant prompt | System instructions used to initialize the conversation. |
output_modalities |
["audio"] |
Exactly ["audio"] or ["text"]. |
temperature |
0.7 |
Sampling temperature used for model responses. |
max_output_tokens |
"inf" |
Maximum response tokens. Numeric values are capped at 4096. |
tools |
Not set | Python functions or Pipecat-compatible tool definitions. |
tool_choice |
"auto" |
Tool selection behavior used when tools are available. |
turn_detection |
Server VAD | OpenAI-compatible server_vad or semantic_vad configuration. |
input_audio_transcription |
Not set | Transcription dictionary. A non-empty model enables client-facing user transcript events. |
input_audio_transcription_model |
"" |
Convenience option for the transcription model. |
input_audio_transcription_language |
None |
Convenience option for the transcription language. |
input_audio_noise_reduction |
Not set | OpenAI-compatible {"type": "near_field"} or {"type": "far_field"} input noise reduction setting. The corresponding type string is also accepted. |
truncation |
"auto" |
"auto" enables smart context summarization when the selected model publishes a context limit; "disabled" turns it off. |
This Pipecat integration sends and receives 24 kHz PCM audio.
- Learn about Higgs Realtime.
- Read the Pipecat documentation.
BSD-2-Clause. See LICENSE.