Merge branch 'main' into groundingMetadata

.pre-commit-config.yaml

@@ -4,5 +4,5 @@ repos:
     hooks:
       - id: ruff
         language_version: python3
-        args: [ --select,  I, ]
+        args: [--fix]
       - id: ruff-format

CHANGELOG.md

@@ -7,6 +7,124 @@ and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0
 
 ## [Unreleased]
 
+### Added
+
+- Added `LocalSmartTurnAnalyzerV2`, which supports local on-device inference
+  with the new `smart-turn-v2` turn detection model.
+
+### Changed
+
+- For `LmntTTSService`, changed the default `model` to `blizzard`, LMNT's
+  recommended model.
+
+### Fixed
+
+- Fixed an issue where, in some edge cases, the `EmulateUserStartedSpeakingFrame`
+  could be created even if we didn't have a transcription.
+
+- Fixed an issue in `GoogleLLMContext` where it would inject the
+  `system_message` as a "user" message into cases where it was not meant to;
+  it was only meant to do that when there were no "regular" (non-function-call)
+  messages in the context, to ensure that inference would run properly.
+
+## [0.0.76] - 2025-07-11
+
+### Added
+
+- Added `SpeechControlParamsFrame`, a new `SystemFrame` that notifies
+  downstream processors of the VAD and Turn analyzer params. This frame is
+  pushed by the `BaseInputTransport` at Start and any time a
+  `VADParamsUpdateFrame` is received.
+
+### Changed
+
+- Two package dependencies have been updated:
+  - `numpy` now supports 1.26.0 and newer
+  - `transformers` now supports 4.48.0 and newer
+
+### Fixed
+
+- Fixed an issue with RTVI's handling of `append-to-context`.
+
+- Fixed an issue where using audio input with a sample rate requiring resampling
+  could result in empty audio being passed to STT services, causing errors.
+
+- Fixed the VAD analyzer to process the full audio buffer as long as it contains
+  more than the minimum required bytes per iteration, instead of only analyzing
+  the first chunk.
+
+- Fixed an issue in ParallelPipeline that caused errors when attempting to drain
+  the queues.
+
+- Fixed an issue with emulated VAD timeout inconsistency in
+  `LLMUserContextAggregator`. Previously, emulated VAD scenarios (where
+  transcription is received without VAD detection) used a hardcoded
+  `aggregation_timeout` (default 0.5s) instead of matching the VAD's
+  `stop_secs` parameter (default 0.8s). This created different user experiences
+  between real VAD and emulated VAD scenarios. Now, emulated VAD timeouts
+  automatically synchronize with the VAD's `stop_secs` parameter.
+
+- Fix a pipeline freeze when using AWS Nova Sonic, which would occur if the
+  user started early, while the bot was still working through
+  `trigger_assistant_response()`.
+
+## [0.0.75] - 2025-07-08
+
+### Added
+
+- Added an `aggregate_sentences` arg in `CartesiaTTSService`,
+  `ElevenLabsTTSService`, `NeuphonicTTSService` and `RimeTTSService`, where the
+  default value is True. When `aggregate_sentences` is True, the `TTSService`
+  aggregates the LLM streamed tokens into sentences by default. Note: setting
+  the value to False requires a custom processor before the `TTSService` to
+  aggregate LLM tokens.
+
+- Added `kwargs` to the `OLLamaLLMService` to allow for configuration args to
+  be passed to Ollama.
+
+- Added call hang-up error handling in `TwilioFrameSerializer`, which handles
+  the case where the user has hung up before the `TwilioFrameSerializer` hangs
+  up the call.
+
+### Changed
+
+- Updated `RTVIObserver` and `RTVIProcessor` to match the new RTVI 1.0.0 protocol.
+  This includes:
+
+  - Deprecating support for all messages related to service configuaration and
+    actions.
+  - Adding support for obtaining and logging data about client, including its
+    RTVI version and optionally included system information (OS/browser/etc.)
+  - Adding support for handling the new `client-message` RTVI message through
+    either a `on_client_message` event handler or listening for a new
+    `RTVIClientMessageFrame`
+  - Adding support for responding to a `client-message` with a `server-response`
+    via either a direct call on the `RTVIProcessor` or via pushing a new
+    `RTVIServerResponseFrame`
+  - Adding built-in support for handling the new `append-to-context` RTVI message
+    which allows a client to add to the user or assistant llm context. No extra
+    code is required for supporting this behavior.
+  - Updating all JavaScript and React client RTVI examples to use versions 1.0.0
+    of the clients.
+
+  Get started migrating to RTVI protocol 1.0.0 by following the migration guide:
+  https://docs.pipecat.ai/client/migration-guide
+
+- Refactored `AWSBedrockLLMService` and `AWSPollyTTSService` to work
+  asynchronously using `aioboto3` instead of the `boto3` library.
+
+- The `UserIdleProcessor` now handles the scenario where function calls take
+  longer than the idle timeout duration. This allows you to use the
+  `UserIdleProcessor` in conjunction with function calls that take a while to
+  return a result.
+
+### Fixed
+
+- Updated the `NeuphonicTTSService` to work with the updated websocket API.
+
+- Fixed an issue with `RivaSTTService` where the watchdog feature was causing
+  an error on initialization.
+
 ### Performance
 
 - Remove unncessary push task in each `FrameProcessor`.

README.md

@@ -53,7 +53,7 @@ You can connect to Pipecat from any platform using our official SDKs:
 
 | Category            | Services                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                     |
 | ------------------- | ---------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- |
-| Speech-to-Text      | [AssemblyAI](https://docs.pipecat.ai/server/services/stt/assemblyai), [AWS](https://docs.pipecat.ai/server/services/stt/aws), [Azure](https://docs.pipecat.ai/server/services/stt/azure), [Cartesia](https://docs.pipecat.ai/server/services/stt/cartesia), [Deepgram](https://docs.pipecat.ai/server/services/stt/deepgram), [Fal Wizper](https://docs.pipecat.ai/server/services/stt/fal), [Gladia](https://docs.pipecat.ai/server/services/stt/gladia), [Google](https://docs.pipecat.ai/server/services/stt/google), [Groq (Whisper)](https://docs.pipecat.ai/server/services/stt/groq), [OpenAI (Whisper)](https://docs.pipecat.ai/server/services/stt/openai), [Parakeet (NVIDIA)](https://docs.pipecat.ai/server/services/stt/parakeet), [SambaNova (Whisper)](https://docs.pipecat.ai/server/services/stt/sambanova), [Speechmatics](https://docs.pipecat.ai/server/services/stt/speechmatics), [Ultravox](https://docs.pipecat.ai/server/services/stt/ultravox), [Whisper](https://docs.pipecat.ai/server/services/stt/whisper)                                                                                     |
+| Speech-to-Text      | [AssemblyAI](https://docs.pipecat.ai/server/services/stt/assemblyai), [AWS](https://docs.pipecat.ai/server/services/stt/aws), [Azure](https://docs.pipecat.ai/server/services/stt/azure), [Cartesia](https://docs.pipecat.ai/server/services/stt/cartesia), [Deepgram](https://docs.pipecat.ai/server/services/stt/deepgram), [Fal Wizper](https://docs.pipecat.ai/server/services/stt/fal), [Gladia](https://docs.pipecat.ai/server/services/stt/gladia), [Google](https://docs.pipecat.ai/server/services/stt/google), [Groq (Whisper)](https://docs.pipecat.ai/server/services/stt/groq), [OpenAI (Whisper)](https://docs.pipecat.ai/server/services/stt/openai), [Parakeet (NVIDIA)](https://docs.pipecat.ai/server/services/stt/parakeet), [SambaNova (Whisper)](https://docs.pipecat.ai/server/services/stt/sambanova), [Soniox](https://docs.pipecat.ai/server/services/stt/soniox), [Speechmatics](https://docs.pipecat.ai/server/services/stt/speechmatics), [Ultravox](https://docs.pipecat.ai/server/services/stt/ultravox), [Whisper](https://docs.pipecat.ai/server/services/stt/whisper)                       |
 | LLMs                | [Anthropic](https://docs.pipecat.ai/server/services/llm/anthropic), [AWS](https://docs.pipecat.ai/server/services/llm/aws), [Azure](https://docs.pipecat.ai/server/services/llm/azure), [Cerebras](https://docs.pipecat.ai/server/services/llm/cerebras), [DeepSeek](https://docs.pipecat.ai/server/services/llm/deepseek), [Fireworks AI](https://docs.pipecat.ai/server/services/llm/fireworks), [Gemini](https://docs.pipecat.ai/server/services/llm/gemini), [Grok](https://docs.pipecat.ai/server/services/llm/grok), [Groq](https://docs.pipecat.ai/server/services/llm/groq), [NVIDIA NIM](https://docs.pipecat.ai/server/services/llm/nim), [Ollama](https://docs.pipecat.ai/server/services/llm/ollama), [OpenAI](https://docs.pipecat.ai/server/services/llm/openai), [OpenRouter](https://docs.pipecat.ai/server/services/llm/openrouter), [Perplexity](https://docs.pipecat.ai/server/services/llm/perplexity), [Qwen](https://docs.pipecat.ai/server/services/llm/qwen), [SambaNova](https://docs.pipecat.ai/server/services/llm/sambanova) [Together AI](https://docs.pipecat.ai/server/services/llm/together) |
 | Text-to-Speech      | [AWS](https://docs.pipecat.ai/server/services/tts/aws), [Azure](https://docs.pipecat.ai/server/services/tts/azure), [Cartesia](https://docs.pipecat.ai/server/services/tts/cartesia), [Deepgram](https://docs.pipecat.ai/server/services/tts/deepgram), [ElevenLabs](https://docs.pipecat.ai/server/services/tts/elevenlabs), [FastPitch (NVIDIA)](https://docs.pipecat.ai/server/services/tts/fastpitch), [Fish](https://docs.pipecat.ai/server/services/tts/fish), [Google](https://docs.pipecat.ai/server/services/tts/google), [LMNT](https://docs.pipecat.ai/server/services/tts/lmnt), [MiniMax](https://docs.pipecat.ai/server/services/tts/minimax), [Neuphonic](https://docs.pipecat.ai/server/services/tts/neuphonic), [OpenAI](https://docs.pipecat.ai/server/services/tts/openai), [Piper](https://docs.pipecat.ai/server/services/tts/piper), [PlayHT](https://docs.pipecat.ai/server/services/tts/playht), [Rime](https://docs.pipecat.ai/server/services/tts/rime), [Sarvam](https://docs.pipecat.ai/server/services/tts/sarvam), [XTTS](https://docs.pipecat.ai/server/services/tts/xtts)                    |
 | Speech-to-Speech    | [AWS Nova Sonic](https://docs.pipecat.ai/server/services/s2s/aws), [Gemini Multimodal Live](https://docs.pipecat.ai/server/services/s2s/gemini), [OpenAI Realtime](https://docs.pipecat.ai/server/services/s2s/openai)                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                       |

docs/api/conf.py

@@ -77,6 +77,7 @@ autodoc_mock_imports = [
     "openpipe",
     "simli",
     "soundfile",
+    "soniox",
     "pipecat_ai_krisp",
     "pyaudio",
     "_tkinter",

docs/api/requirements.txt

@@ -46,6 +46,7 @@ pipecat-ai[sambanova]
 pipecat-ai[silero]
 pipecat-ai[simli]
 pipecat-ai[soundfile]
+pipecat-ai[soniox]
 pipecat-ai[speechmatics]
 pipecat-ai[tavus]
 pipecat-ai[together]

dot-env.template

@@ -109,6 +109,9 @@ MINIMAX_GROUP_ID=...
 # Sarvam AI
 SARVAM_API_KEY=...
 
+# Soniox
+SONIOX_API_KEY=
+
 # Speechmatics
 SPEECHMATICS_API_KEY=...
 

examples/aws-strands/README.md

@@ -0,0 +1,60 @@
+# AWS Strands Examples
+
+This folder contains two Python examples demonstrating how to use Pipecat with the AWS Strands agent.
+
+## Overview
+
+These examples show how to delegate complex, multi-step tasks to a Strands agent, which can reason step-by-step and call tools to accomplish user requests.
+
+These examples are intentionally simplified for demonstration, using mock API calls. They work best if you ask it:
+
+> What's the weather where the Golden Gate Bridge is?
+
+## Example Scripts
+
+### `black-box.py`
+
+A minimal example that demonstrates how to use the Strands agent with Pipecat. The agent can handle multi-step queries by calling tools, but does not explain its reasoning out loud.
+
+### `explain-thinking.py`
+
+An enhanced example where the Strands agent explains each step of its reasoning in clear, simple language as it works through a multi-step task.
+
+## Quick Start
+
+1. **Clone the repository and navigate to this example:**
+
+   ```bash
+   git clone https://github.com/pipecat-ai/pipecat.git
+   cd pipecat/examples/aws-strands
+   ```
+
+2. **Set up a virtual environment:**
+
+   ```bash
+   python -m venv venv
+   source venv/bin/activate  # On Windows: venv\Scripts\activate
+   ```
+
+3. **Install dependencies:**
+
+   ```bash
+   pip install -r requirements.txt
+   ```
+
+4. **Configure environment variables:**
+
+   Copy the provided `env.example` file to `.env` and fill in the necessary credentials:
+
+   ```bash
+   cp env.example .env
+   # Then edit .env with your preferred editor
+   ```
+
+5. **Run an example:**
+
+   ```bash
+   python black-box.py
+   # or
+   python explain-thinking.py
+   ```

examples/aws-strands/black-box.py

@@ -0,0 +1,206 @@
+#
+# Copyright (c) 2025, Daily
+#
+# SPDX-License-Identifier: BSD 2-Clause License
+#
+
+import argparse
+import asyncio
+import os
+
+from dotenv import load_dotenv
+from loguru import logger
+from strands import Agent, tool
+from strands.models import BedrockModel
+
+from pipecat.adapters.schemas.tools_schema import ToolsSchema
+from pipecat.audio.vad.silero import SileroVADAnalyzer
+from pipecat.frames.frames import TTSSpeakFrame
+from pipecat.pipeline.pipeline import Pipeline
+from pipecat.pipeline.runner import PipelineRunner
+from pipecat.pipeline.task import PipelineParams, PipelineTask
+from pipecat.processors.aggregators.openai_llm_context import OpenAILLMContext
+from pipecat.services.cartesia.tts import CartesiaTTSService
+from pipecat.services.deepgram.stt import DeepgramSTTService
+from pipecat.services.llm_service import FunctionCallParams
+from pipecat.services.openai.llm import OpenAILLMService
+from pipecat.transports.base_transport import BaseTransport, TransportParams
+from pipecat.transports.network.fastapi_websocket import FastAPIWebsocketParams
+from pipecat.transports.services.daily import DailyParams
+
+load_dotenv(override=True)
+
+"""This example demonstrates how to use the Strands agent with Pipecat.
+
+You can delegate complex, multi-step tasks to the Strands agent, which can cycle through LLM-based reasoning and tool calls to accomplish the task.
+
+Try asking: "What's the weather where the Golden Gate Bridge is?"
+"""
+
+# Strands agent tools
+
+
+@tool
+def get_location_name_from_landmark(landmark: str) -> str:
+    """
+    Get the location name from a landmark.
+
+    Args:
+        landmark (str): The name of the landmark, e.g. "Golden Gate Bridge".
+    """
+    # Simulate fetching location
+    return "San Francisco, CA"
+
+
+@tool
+def get_lat_long_from_location_name(location: str) -> dict:
+    """
+    Get the latitude and longitude for a location name.
+
+    Args:
+        location (str): The city and state, e.g. "San Francisco, CA".
+    """
+    # Simulate fetching lat/long from a geocoding service
+    return {"lat": 37.7749, "long": -122.4194}
+
+
+@tool
+def get_current_weather_from_lat_long(lat: float, long: float) -> dict:
+    """
+    Get the current weather for a specific latitude and longitude.
+
+    Args:
+        lat (float): The latitude of the location.
+        long (float): The longitude of the location.
+    """
+    # Simulate fetching weather data from a weather service
+    return {"conditions": "nice", "temperature": "75"}
+
+
+# We store functions so objects (e.g. SileroVADAnalyzer) don't get
+# instantiated. The function will be called when the desired transport gets
+# selected.
+transport_params = {
+    "daily": lambda: DailyParams(
+        audio_in_enabled=True,
+        audio_out_enabled=True,
+        vad_analyzer=SileroVADAnalyzer(),
+    ),
+    "twilio": lambda: FastAPIWebsocketParams(
+        audio_in_enabled=True,
+        audio_out_enabled=True,
+        vad_analyzer=SileroVADAnalyzer(),
+    ),
+    "webrtc": lambda: TransportParams(
+        audio_in_enabled=True,
+        audio_out_enabled=True,
+        vad_analyzer=SileroVADAnalyzer(),
+    ),
+}
+
+
+async def run_example(transport: BaseTransport, _: argparse.Namespace, handle_sigint: bool):
+    logger.info(f"Starting bot")
+
+    strands_agent = Agent(
+        model=BedrockModel(
+            model_id="us.anthropic.claude-3-7-sonnet-20250219-v1:0", max_tokens=64000
+        ),
+        tools=[
+            get_location_name_from_landmark,
+            get_lat_long_from_location_name,
+            get_current_weather_from_lat_long,
+        ],
+        system_prompt="""
+        You are a helpful personal assistant who can look up information about places and weather.
+
+        Your key capabilities:
+        1. Look up where landmarks are located.
+        2. Find latitude and longitude for a location.
+        3. Look up the current weather for a specific latitude and longitude.
+
+        Explain each step of your reasoning in clear, simple, and concise language. Your responses will be converted to audio, so avoid special characters and numbered lists.
+        """,
+    )
+
+    async def handle_location_or_weather_related_queries(params: FunctionCallParams, query: str):
+        """
+        Handle location or weather related queries.
+
+        Args:
+            query (str): The user's query, e.g. "What's the weather where the Golden Gate Bridge is?".
+        """
+        # Run in a background thread
+        # (Otherwise the agent blocks the event loop; one effect of that is that we don't hear
+        # "let me check on that" until the agent finishes)
+        loop = asyncio.get_running_loop()
+        result = await loop.run_in_executor(None, strands_agent, query)
+        await params.result_callback(result.message)
+
+    stt = DeepgramSTTService(api_key=os.getenv("DEEPGRAM_API_KEY"))
+
+    tts = CartesiaTTSService(
+        api_key=os.getenv("CARTESIA_API_KEY"),
+        voice_id="71a7ad14-091c-4e8e-a314-022ece01c121",  # British Reading Lady
+    )
+
+    llm = OpenAILLMService(api_key=os.getenv("OPENAI_API_KEY"))
+
+    llm.register_direct_function(handle_location_or_weather_related_queries)
+
+    @llm.event_handler("on_function_calls_started")
+    async def on_function_calls_started(service, function_calls):
+        await tts.queue_frame(TTSSpeakFrame("Let me check on that."))
+
+    tools = ToolsSchema(standard_tools=[handle_location_or_weather_related_queries])
+
+    messages = [
+        {
+            "role": "system",
+            "content": "You are a helpful LLM in a WebRTC call. Your goal is to demonstrate your capabilities in a succinct way. Your output will be converted to audio so don't include special characters in your answers. Respond to what the user said in a creative and helpful way. Start by suggesting that the user ask about the weather where the Golden Gate Bridge is.",
+        },
+    ]
+
+    context = OpenAILLMContext(messages, tools)
+    context_aggregator = llm.create_context_aggregator(context)
+
+    pipeline = Pipeline(
+        [
+            transport.input(),
+            stt,
+            context_aggregator.user(),
+            llm,
+            tts,
+            transport.output(),
+            context_aggregator.assistant(),
+        ]
+    )
+
+    task = PipelineTask(
+        pipeline,
+        params=PipelineParams(
+            enable_metrics=True,
+            enable_usage_metrics=True,
+        ),
+    )
+
+    @transport.event_handler("on_client_connected")
+    async def on_client_connected(transport, client):
+        logger.info(f"Client connected")
+        # Kick off the conversation.
+        await task.queue_frames([context_aggregator.user().get_context_frame()])
+
+    @transport.event_handler("on_client_disconnected")
+    async def on_client_disconnected(transport, client):
+        logger.info(f"Client disconnected")
+        await task.cancel()
+
+    runner = PipelineRunner(handle_sigint=handle_sigint)
+
+    await runner.run(task)
+
+
+if __name__ == "__main__":
+    from pipecat.examples.run import main
+
+    main(run_example, transport_params=transport_params)

examples/aws-strands/env.example

@@ -0,0 +1,8 @@
+OPENAI_API_KEY=
+CARTESIA_API_KEY=
+DEEPGRAM_API_KEY=
+DAILY_API_KEY=
+DAILY_SAMPLE_ROOM_URL=
+AWS_SECRET_ACCESS_KEY=
+AWS_ACCESS_KEY_ID=
+AWS_REGION=

examples/aws-strands/explain-thinking.py

@@ -0,0 +1,249 @@
+#
+# Copyright (c) 2025, Daily
+#
+# SPDX-License-Identifier: BSD 2-Clause License
+#
+
+import argparse
+import asyncio
+import os
+import threading
+import time
+
+from dotenv import load_dotenv
+from loguru import logger
+from strands import Agent, tool
+from strands.models import BedrockModel
+
+from pipecat.adapters.schemas.tools_schema import ToolsSchema
+from pipecat.audio.vad.silero import SileroVADAnalyzer
+from pipecat.frames.frames import TTSSpeakFrame
+from pipecat.pipeline.pipeline import Pipeline
+from pipecat.pipeline.runner import PipelineRunner
+from pipecat.pipeline.task import PipelineParams, PipelineTask
+from pipecat.processors.aggregators.openai_llm_context import OpenAILLMContext
+from pipecat.services.cartesia.tts import CartesiaTTSService
+from pipecat.services.deepgram.stt import DeepgramSTTService
+from pipecat.services.llm_service import FunctionCallParams
+from pipecat.services.openai.llm import OpenAILLMService
+from pipecat.transports.base_transport import BaseTransport, TransportParams
+from pipecat.transports.network.fastapi_websocket import FastAPIWebsocketParams
+from pipecat.transports.services.daily import DailyParams
+
+load_dotenv(override=True)
+
+"""This example demonstrates how to use the Strands agent with Pipecat in a way where the agent explains its reasoning step-by-step.
+
+You can delegate complex, multi-step tasks to the Strands agent, which can cycle through LLM-based reasoning and tool calls to accomplish the task.
+
+Try asking: "What's the weather where the Golden Gate Bridge is?"
+"""
+
+
+# Strands agent tools
+
+
+@tool
+def get_location_name_from_landmark(landmark: str) -> str:
+    """
+    Get the location name from a landmark.
+
+    Args:
+        landmark (str): The name of the landmark, e.g. "Golden Gate Bridge".
+    """
+    # Simulate fetching location (slowly)
+    time.sleep(3)
+    return "San Francisco, CA"
+
+
+@tool
+def get_lat_long_from_location_name(location: str) -> dict:
+    """
+    Get the latitude and longitude for a location name.
+
+    Args:
+        location (str): The city and state, e.g. "San Francisco, CA".
+    """
+    # Simulate fetching lat/long from a geocoding service (slowly)
+    time.sleep(3)
+    return {"lat": 37.7749, "long": -122.4194}
+
+
+@tool
+def get_current_weather_from_lat_long(lat: float, long: float) -> dict:
+    """
+    Get the current weather for a specific latitude and longitude.
+
+    Args:
+        lat (float): The latitude of the location.
+        long (float): The longitude of the location.
+    """
+    # Simulate fetching weather data from a weather service (slowly)
+    time.sleep(3)
+    return {"conditions": "nice", "temperature": "75"}
+
+
+# We store functions so objects (e.g. SileroVADAnalyzer) don't get
+# instantiated. The function will be called when the desired transport gets
+# selected.
+transport_params = {
+    "daily": lambda: DailyParams(
+        audio_in_enabled=True,
+        audio_out_enabled=True,
+        vad_analyzer=SileroVADAnalyzer(),
+    ),
+    "twilio": lambda: FastAPIWebsocketParams(
+        audio_in_enabled=True,
+        audio_out_enabled=True,
+        vad_analyzer=SileroVADAnalyzer(),
+    ),
+    "webrtc": lambda: TransportParams(
+        audio_in_enabled=True,
+        audio_out_enabled=True,
+        vad_analyzer=SileroVADAnalyzer(),
+    ),
+}
+
+
+async def run_example(transport: BaseTransport, _: argparse.Namespace, handle_sigint: bool):
+    logger.info(f"Starting bot")
+
+    stt = DeepgramSTTService(api_key=os.getenv("DEEPGRAM_API_KEY"))
+
+    tts = CartesiaTTSService(
+        api_key=os.getenv("CARTESIA_API_KEY"),
+        voice_id="71a7ad14-091c-4e8e-a314-022ece01c121",  # British Reading Lady
+    )
+
+    next_strands_message_is_last = False
+    strands_messages_queue = asyncio.Queue()
+
+    def strands_callback_handler(**kwargs):
+        """
+        Handle events from the Strands agent.
+        """
+        nonlocal next_strands_message_is_last
+        if "event" in kwargs:
+            event_obj = kwargs["event"]
+            if event_obj and "messageStop" in event_obj:
+                message_stop = event_obj["messageStop"]
+                if message_stop and "stopReason" in message_stop:
+                    stop_reason = message_stop["stopReason"]
+                    if stop_reason == "end_turn":
+                        next_strands_message_is_last = True
+        elif "message" in kwargs:
+            message_obj = kwargs["message"]
+            if message_obj and "content" in message_obj and "role" in message_obj:
+                role = message_obj["role"]
+                content = message_obj["content"]
+                if role == "assistant" and isinstance(content, list):
+                    for content_obj in content:
+                        if isinstance(content_obj, dict) and "text" in content_obj:
+                            message = content_obj["text"]
+                            if not next_strands_message_is_last:
+                                strands_messages_queue.put_nowait(message)
+
+    async def process_strands_messages():
+        while True:
+            message = await strands_messages_queue.get()
+            await tts.queue_frame(TTSSpeakFrame(message))
+            strands_messages_queue.task_done()
+
+    asyncio.create_task(process_strands_messages())
+
+    strands_agent = Agent(
+        model=BedrockModel(
+            model_id="us.anthropic.claude-3-7-sonnet-20250219-v1:0", max_tokens=64000
+        ),
+        tools=[
+            get_location_name_from_landmark,
+            get_lat_long_from_location_name,
+            get_current_weather_from_lat_long,
+        ],
+        system_prompt="""
+        You are a helpful personal assistant who can look up information about places and weather.
+
+        Your key capabilities:
+        1. Look up where landmarks are located.
+        2. Find latitude and longitude for a location.
+        3. Look up the current weather for a specific latitude and longitude.
+
+        Explain each step of your reasoning in clear, simple, and concise language. Your responses will be converted to audio, so avoid special characters and numbered lists.
+        """,
+        callback_handler=strands_callback_handler,
+    )
+
+    llm = OpenAILLMService(api_key=os.getenv("OPENAI_API_KEY"))
+
+    async def handle_location_or_weather_related_queries(params: FunctionCallParams, query: str):
+        """
+        Handle location or weather related queries.
+
+        Args:
+            query (str): The user's query, e.g. "What's the weather where the Golden Gate Bridge is?".
+        """
+        # Run in a background thread
+        # (Otherwise the agent blocks the event loop; one effect of that is that we don't hear
+        # the agent's "thinking" messages until the agent finishes)
+        loop = asyncio.get_running_loop()
+        result = await loop.run_in_executor(None, strands_agent, query)
+        await params.result_callback(result.message)
+
+    llm.register_direct_function(handle_location_or_weather_related_queries)
+
+    @llm.event_handler("on_function_calls_started")
+    async def on_function_calls_started(service, function_calls):
+        await tts.queue_frame(TTSSpeakFrame("Let me check on that."))
+
+    tools = ToolsSchema(standard_tools=[handle_location_or_weather_related_queries])
+
+    messages = [
+        {
+            "role": "system",
+            "content": "You are a helpful LLM in a WebRTC call. Your goal is to demonstrate your capabilities in a succinct way. Your output will be converted to audio so don't include special characters in your answers. Respond to what the user said in a creative and helpful way. Start by suggesting that the user ask about the weather where the Golden Gate Bridge is.",
+        },
+    ]
+
+    context = OpenAILLMContext(messages, tools)
+    context_aggregator = llm.create_context_aggregator(context)
+
+    pipeline = Pipeline(
+        [
+            transport.input(),
+            stt,
+            context_aggregator.user(),
+            llm,
+            tts,
+            transport.output(),
+            context_aggregator.assistant(),
+        ]
+    )
+
+    task = PipelineTask(
+        pipeline,
+        params=PipelineParams(
+            enable_metrics=True,
+            enable_usage_metrics=True,
+        ),
+    )
+
+    @transport.event_handler("on_client_connected")
+    async def on_client_connected(transport, client):
+        logger.info(f"Client connected")
+        # Kick off the conversation.
+        await task.queue_frames([context_aggregator.user().get_context_frame()])
+
+    @transport.event_handler("on_client_disconnected")
+    async def on_client_disconnected(transport, client):
+        logger.info(f"Client disconnected")
+        await task.cancel()
+
+    runner = PipelineRunner(handle_sigint=handle_sigint)
+
+    await runner.run(task)
+
+
+if __name__ == "__main__":
+    from pipecat.examples.run import main
+
+    main(run_example, transport_params=transport_params)

examples/aws-strands/requirements.txt

@@ -0,0 +1,6 @@
+fastapi
+uvicorn
+python-dotenv
+pipecat-ai[webrtc,daily,deepgram,cartesia]
+pipecat-ai-small-webrtc-prebuilt
+strands-agents

examples/deployment/modal-example/client/javascript/package.json

@@ -15,7 +15,7 @@
     "vite": "^6.3.5"
   },
   "dependencies": {
-    "@pipecat-ai/client-js": "^0.3.5",
+    "@pipecat-ai/client-js": "^1.0.0",
-    "@pipecat-ai/daily-transport": "^0.3.10"
+    "@pipecat-ai/daily-transport": "^1.0.0"
   }
 }

examples/deployment/modal-example/client/javascript/src/app.js

@@ -5,7 +5,7 @@
  */
 
 /**
- * RTVI Client Implementation
+ * Pipecat Client Implementation
  *
  * This client connects to an RTVI-compatible bot server using WebRTC (via Daily).
  * It handles audio/video streaming and manages the connection lifecycle.
@@ -16,7 +16,7 @@
  * - Browser with WebRTC support
  */
 
-import { RTVIClient, RTVIEvent } from '@pipecat-ai/client-js';
+import { PipecatClient, RTVIEvent } from '@pipecat-ai/client-js';
 import { DailyTransport } from '@pipecat-ai/daily-transport';
 
 /**
@@ -26,7 +26,7 @@ import { DailyTransport } from '@pipecat-ai/daily-transport';
 class ChatbotClient {
   constructor() {
     // Initialize client state
-    this.rtviClient = null;
+    this.pcClient = null;
     this.setupDOMElements();
     this.initializeClientAndTransport();
     this.setupEventListeners();
@@ -59,7 +59,7 @@ class ChatbotClient {
     this.disconnectBtn.addEventListener('click', () => this.disconnect());
 
     // Populate device selector
-    this.rtviClient.getAllMics().then((mics) => {
+    this.pcClient.getAllMics().then((mics) => {
       console.log('Available mics:', mics);
       mics.forEach((device) => {
         const option = document.createElement('option');
@@ -71,16 +71,16 @@ class ChatbotClient {
     this.deviceSelector.addEventListener('change', (event) => {
       const selectedDeviceId = event.target.value;
       console.log('Selected device ID:', selectedDeviceId);
-      this.rtviClient.updateMic(selectedDeviceId);
+      this.pcClient.updateMic(selectedDeviceId);
     });
 
     // Handle mic mute/unmute toggle
     const micToggleBtn = document.getElementById('mic-toggle-btn');
 
     micToggleBtn.addEventListener('click', () => {
-      let micEnabled = this.rtviClient.isMicEnabled;
+      let micEnabled = this.pcClient.isMicEnabled;
       micToggleBtn.textContent = micEnabled ? 'Unmute Mic' : 'Mute Mic';
-      this.rtviClient.enableMic(!micEnabled);
+      this.pcClient.enableMic(!micEnabled);
       // Add logic to mute/unmute the mic
       if (micEnabled) {
         console.log('Mic muted');
@@ -93,23 +93,12 @@ class ChatbotClient {
   }
 
   /**
-   * Set up the RTVI client and Daily transport
+   * Set up the Pipecat client and Daily transport
    */
   async initializeClientAndTransport() {
-    // Initialize the RTVI client with a DailyTransport and our configuration
+    // Initialize the Pipecat client with a DailyTransport and our configuration
-    this.rtviClient = new RTVIClient({
+    this.pcClient = new PipecatClient({
       transport: new DailyTransport(),
-      params: {
-        // REPLACE WITH YOUR MODAL URL ENDPOINT
-        baseUrl:
-          'https://<Modal workspace>--pipecat-modal-bot-launcher.modal.run',
-        endpoints: {
-          connect: '/connect',
-        },
-        requestData: {
-          bot_name: 'openai',
-        },
-      },
       enableMic: true, // Enable microphone for user input
       enableCam: false,
       callbacks: {
@@ -176,8 +165,8 @@ class ChatbotClient {
     // Set up listeners for media track events
     this.setupTrackListeners();
 
-    await this.rtviClient.initDevices();
+    await this.pcClient.initDevices();
-    window.client = this.rtviClient;
+    window.client = this.pcClient;
   }
 
   /**
@@ -212,10 +201,10 @@ class ChatbotClient {
    * This is called when the bot is ready or when the transport state changes to ready
    */
   setupMediaTracks() {
-    if (!this.rtviClient) return;
+    if (!this.pcClient) return;
 
     // Get current tracks from the client
-    const tracks = this.rtviClient.tracks();
+    const tracks = this.pcClient.tracks();
 
     // Set up any available bot tracks
     if (tracks.bot?.audio) {
@@ -231,10 +220,10 @@ class ChatbotClient {
    * This handles new tracks being added during the session
    */
   setupTrackListeners() {
-    if (!this.rtviClient) return;
+    if (!this.pcClient) return;
 
     // Listen for new tracks starting
-    this.rtviClient.on(RTVIEvent.TrackStarted, (track, participant) => {
+    this.pcClient.on(RTVIEvent.TrackStarted, (track, participant) => {
       // Only handle non-local (bot) tracks
       if (!participant?.local) {
         if (track.kind === 'audio') {
@@ -253,7 +242,7 @@ class ChatbotClient {
     });
 
     // Listen for tracks stopping
-    this.rtviClient.on(RTVIEvent.TrackStopped, (track, participant) => {
+    this.pcClient.on(RTVIEvent.TrackStopped, (track, participant) => {
       if (participant.local) {
         this.log('Local mic muted');
         return;
@@ -311,21 +300,27 @@ class ChatbotClient {
 
   /**
    * Initialize and connect to the bot
-   * This sets up the RTVI client, initializes devices, and establishes the connection
+   * This sets up the Pipecat client, initializes devices, and establishes the connection
    */
   async connect() {
     try {
       const botSelector = document.getElementById('bot-selector');
       const selectedBot = botSelector.value;
-      this.rtviClient.params.requestData.bot_name = selectedBot;
 
       // Initialize audio/video devices
       this.log('Initializing devices...');
-      await this.rtviClient.initDevices();
+      await this.pcClient.initDevices();
 
       // Connect to the bot
       this.log(`Connecting to bot: ${selectedBot}`);
-      await this.rtviClient.connect();
+      await this.pcClient.connect({
+        // REPLACE WITH YOUR MODAL URL ENDPOINT
+        endpoint:
+          'https://<your-workspace>--pipecat-modal-fastapi-app.modal.run/connect',
+        requestData: {
+          bot_name: selectedBot,
+        },
+      });
 
       this.log('Connection complete');
     } catch (error) {
@@ -336,9 +331,9 @@ class ChatbotClient {
       this.updateStatus('Error');
 
       // Clean up if there's an error
-      if (this.rtviClient) {
+      if (this.pcClient) {
         try {
-          await this.rtviClient.disconnect();
+          await this.pcClient.disconnect();
         } catch (disconnectError) {
           this.log(`Error during disconnect: ${disconnectError.message}`);
         }
@@ -350,10 +345,10 @@ class ChatbotClient {
    * Disconnect from the bot and clean up media resources
    */
   async disconnect() {
-    if (this.rtviClient) {
+    if (this.pcClient) {
       try {
-        // Disconnect the RTVI client
+        // Disconnect the Pipecat client
-        await this.rtviClient.disconnect();
+        await this.pcClient.disconnect();
 
         // Clean up audio
         if (this.botAudio.srcObject) {

examples/deployment/modal-example/server/app.py

@@ -301,7 +301,7 @@ def fastapi_app():
         allow_headers=["*"],
     )
 
-    # Include the endpoints from endpoints.py
+    # Include the endpoints from this file
     web_app.include_router(router)
 
     return web_app

examples/deployment/modal-example/server/requirements.txt

@@ -1,2 +1,3 @@
 python-dotenv==1.0.1
-modal==0.71.3
+modal==1.0.5
+fastapi[all]

examples/deployment/pipecat-cloud-daily-pstn-server/nextjs-webhook-server/pages/api/dial.js

@@ -103,7 +103,7 @@ export default async function handler(req, res) {
       const sip_config = {
         display_name: From,
         sip_mode: 'dial-in',
-        num_endpoints: call_transfer !== null ? 2 : 1,
+        num_endpoints: (call_transfer !== undefined && call_transfer !== null) ? 2 : 1,
         codecs: {"audio": ["OPUS"]},
       };
       daily_room_properties.sip = sip_config;

examples/fal-smart-turn/README.md

@@ -44,7 +44,7 @@ Try the hosted version of the demo here: https://pcc-smart-turn.vercel.app/.
 4. Run the server:
 
    ```bash
-   LOCAL=1 python server.py
+   LOCAL_RUN=1 python server.py
    ```
 
 ### Run the client

examples/fal-smart-turn/client/package.json

@@ -9,9 +9,9 @@
     "lint": "next lint"
   },
   "dependencies": {
-    "@pipecat-ai/client-js": "^0.3.5",
+    "@pipecat-ai/client-js": "^1.0.0",
-    "@pipecat-ai/client-react": "^0.3.5",
+    "@pipecat-ai/client-react": "^1.0.0",
-    "@pipecat-ai/daily-transport": "^0.3.10",
+    "@pipecat-ai/daily-transport": "^1.0.0",
     "next": "15.3.1",
     "react": "^19.0.0",
     "react-dom": "^19.0.0"

examples/fal-smart-turn/client/src/app/layout.tsx

@@ -1,5 +1,5 @@
 import './globals.css';
-import { RTVIProvider } from '@/providers/RTVIProvider';
+import { PipecatProvider } from '@/providers/PipecatProvider';
 
 export const metadata = {
   title: 'Pipecat React Client',
@@ -20,7 +20,7 @@ export default function RootLayout({
         <link rel="icon" href="/favicon.svg" type="image/svg+xml" />
       </head>
       <body>
-        <RTVIProvider>{children}</RTVIProvider>
+        <PipecatProvider>{children}</PipecatProvider>
       </body>
     </html>
   );

examples/fal-smart-turn/client/src/app/page.tsx

@@ -1,22 +1,22 @@
 'use client';
 
 import {
-  RTVIClientAudio,
+  PipecatClientAudio,
-  RTVIClientVideo,
+  PipecatClientVideo,
-  useRTVIClientTransportState,
+  usePipecatClientTransportState,
 } from '@pipecat-ai/client-react';
 import { ConnectButton } from '../components/ConnectButton';
 import { StatusDisplay } from '../components/StatusDisplay';
 import { DebugDisplay } from '../components/DebugDisplay';
 
 function BotVideo() {
-  const transportState = useRTVIClientTransportState();
+  const transportState = usePipecatClientTransportState();
   const isConnected = transportState !== 'disconnected';
 
   return (
     <div className="bot-container">
       <div className="video-container">
-        {isConnected && <RTVIClientVideo participant="bot" fit="cover" />}
+        {isConnected && <PipecatClientVideo participant="bot" fit="cover" />}
       </div>
     </div>
   );
@@ -35,7 +35,7 @@ export default function Home() {
       </div>
 
       <DebugDisplay />
-      <RTVIClientAudio />
+      <PipecatClientAudio />
     </div>
   );
 }

examples/fal-smart-turn/client/src/components/ConnectButton.tsx

@@ -1,11 +1,17 @@
 import {
-  useRTVIClient,
+  usePipecatClient,
-  useRTVIClientTransportState,
+  usePipecatClientTransportState,
 } from '@pipecat-ai/client-react';
 
+// Get the API base URL from environment variables
+// Default to "/api" if not specified
+// "/api" is the default for Next.js API routes and used
+// for the Pipecat Cloud deployed agent
+const API_BASE_URL = process.env.NEXT_PUBLIC_API_BASE_URL || '/api';
+
 export function ConnectButton() {
-  const client = useRTVIClient();
+  const client = usePipecatClient();
-  const transportState = useRTVIClientTransportState();
+  const transportState = usePipecatClientTransportState();
   const isConnected = ['connected', 'ready'].includes(transportState);
 
   const handleClick = async () => {
@@ -18,7 +24,10 @@ export function ConnectButton() {
       if (isConnected) {
         await client.disconnect();
       } else {
-        await client.connect();
+        await client.connect({
+          endpoint: `${API_BASE_URL}/connect`,
+          requestData: { foo: 'bar' },
+        });
       }
     } catch (error) {
       console.error('Connection error:', error);

examples/fal-smart-turn/client/src/components/DebugDisplay.tsx

@@ -6,7 +6,7 @@ import {
   TranscriptData,
   BotLLMTextData,
 } from '@pipecat-ai/client-js';
-import { useRTVIClient, useRTVIClientEvent } from '@pipecat-ai/client-react';
+import { usePipecatClient, useRTVIClientEvent } from '@pipecat-ai/client-react';
 import './DebugDisplay.css';
 
 interface SmartTurnResultData {
@@ -20,7 +20,7 @@ interface SmartTurnResultData {
 
 export function DebugDisplay() {
   const debugLogRef = useRef<HTMLDivElement>(null);
-  const client = useRTVIClient();
+  const client = usePipecatClient();
 
   const log = useCallback((message: string) => {
     if (!debugLogRef.current) return;

examples/fal-smart-turn/client/src/components/StatusDisplay.tsx

@@ -1,7 +1,7 @@
-import { useRTVIClientTransportState } from '@pipecat-ai/client-react';
+import { usePipecatClientTransportState } from '@pipecat-ai/client-react';
 
 export function StatusDisplay() {
-  const transportState = useRTVIClientTransportState();
+  const transportState = usePipecatClientTransportState();
 
   return (
     <div className="status">

examples/fal-smart-turn/client/src/providers/PipecatProvider.tsx

@@ -0,0 +1,28 @@
+'use client';
+
+import { PipecatClient } from '@pipecat-ai/client-js';
+import { DailyTransport } from '@pipecat-ai/daily-transport';
+import { PipecatClientProvider } from '@pipecat-ai/client-react';
+import { PropsWithChildren, useEffect, useState } from 'react';
+
+export function PipecatProvider({ children }: PropsWithChildren) {
+  const [client, setClient] = useState<PipecatClient | null>(null);
+
+  useEffect(() => {
+    const pcClient = new PipecatClient({
+      transport: new DailyTransport(),
+      enableMic: true,
+      enableCam: false,
+    });
+
+    setClient(pcClient);
+  }, []);
+
+  if (!client) {
+    return null;
+  }
+
+  return (
+    <PipecatClientProvider client={client}>{children}</PipecatClientProvider>
+  );
+}

examples/fal-smart-turn/client/src/providers/RTVIProvider.tsx

@@ -1,43 +0,0 @@
-'use client';
-
-import { RTVIClient } from '@pipecat-ai/client-js';
-import { DailyTransport } from '@pipecat-ai/daily-transport';
-import { RTVIClientProvider } from '@pipecat-ai/client-react';
-import { PropsWithChildren, useEffect, useState } from 'react';
-
-// Get the API base URL from environment variables
-// Default to "/api" if not specified
-// "/api" is the default for Next.js API routes and used
-// for the Pipecat Cloud deployed agent
-const API_BASE_URL = process.env.NEXT_PUBLIC_API_BASE_URL || '/api';
-
-console.log('Using API base URL:', API_BASE_URL);
-
-export function RTVIProvider({ children }: PropsWithChildren) {
-  const [client, setClient] = useState<RTVIClient | null>(null);
-
-  useEffect(() => {
-    const transport = new DailyTransport();
-
-    const rtviClient = new RTVIClient({
-      transport,
-      params: {
-        baseUrl: API_BASE_URL,
-        endpoints: {
-          connect: '/connect',
-        },
-        requestData: { foo: 'bar' },
-      },
-      enableMic: true,
-      enableCam: false,
-    });
-
-    setClient(rtviClient);
-  }, []);
-
-  if (!client) {
-    return null;
-  }
-
-  return <RTVIClientProvider client={client}>{children}</RTVIClientProvider>;
-}

examples/fal-smart-turn/server/bot.py

@@ -45,7 +45,7 @@ from pipecat.transports.services.daily import DailyParams, DailyTransport
 load_dotenv(override=True)
 
 # Check if we're in local development mode
-LOCAL = os.getenv("LOCAL")
+LOCAL = os.getenv("LOCAL_RUN")
 
 logger.remove()
 logger.add(sys.stderr, level="DEBUG")

examples/foundational/07aa-interruptible-soniox.py

@@ -0,0 +1,109 @@
+#
+# Copyright (c) 2024–2025, Daily
+#
+# SPDX-License-Identifier: BSD 2-Clause License
+#
+
+import argparse
+import os
+
+from dotenv import load_dotenv
+from loguru import logger
+
+from pipecat.audio.vad.silero import SileroVADAnalyzer
+from pipecat.pipeline.pipeline import Pipeline
+from pipecat.pipeline.runner import PipelineRunner
+from pipecat.pipeline.task import PipelineParams, PipelineTask
+from pipecat.processors.aggregators.openai_llm_context import OpenAILLMContext
+from pipecat.services.cartesia.tts import CartesiaTTSService
+from pipecat.services.openai.llm import OpenAILLMService
+from pipecat.services.soniox.stt import SonioxSTTService
+from pipecat.transports.base_transport import BaseTransport, TransportParams
+from pipecat.transports.network.fastapi_websocket import FastAPIWebsocketParams
+from pipecat.transports.services.daily import DailyParams
+
+load_dotenv(override=True)
+
+transport_params = {
+    "daily": lambda: DailyParams(
+        audio_in_enabled=True,
+        audio_out_enabled=True,
+        vad_analyzer=SileroVADAnalyzer(),
+    ),
+    "twilio": lambda: FastAPIWebsocketParams(
+        audio_in_enabled=True,
+        audio_out_enabled=True,
+        vad_analyzer=SileroVADAnalyzer(),
+    ),
+    "webrtc": lambda: TransportParams(
+        audio_in_enabled=True,
+        audio_out_enabled=True,
+        vad_analyzer=SileroVADAnalyzer(),
+    ),
+}
+
+
+async def run_example(transport: BaseTransport, _: argparse.Namespace, handle_sigint: bool):
+    logger.info(f"Starting bot")
+
+    stt = SonioxSTTService(
+        api_key=os.getenv("SONIOX_API_KEY"),
+    )
+
+    tts = CartesiaTTSService(
+        api_key=os.getenv("CARTESIA_API_KEY"),
+        voice_id="71a7ad14-091c-4e8e-a314-022ece01c121",  # British Reading Lady
+    )
+
+    llm = OpenAILLMService(api_key=os.getenv("OPENAI_API_KEY"))
+
+    messages = [
+        {
+            "role": "system",
+            "content": "You are a helpful LLM in a WebRTC call. Your goal is to demonstrate your capabilities in a succinct way. Your output will be converted to audio so don't include special characters in your answers. Respond to what the user said in a creative and helpful way.",
+        },
+    ]
+
+    context = OpenAILLMContext(messages)
+    context_aggregator = llm.create_context_aggregator(context)
+
+    pipeline = Pipeline(
+        [
+            transport.input(),  # Transport user input
+            stt,
+            context_aggregator.user(),  # User responses
+            llm,  # LLM
+            tts,  # TTS
+            transport.output(),  # Transport bot output
+            context_aggregator.assistant(),  # Assistant spoken responses
+        ]
+    )
+    task = PipelineTask(
+        pipeline,
+        params=PipelineParams(
+            enable_metrics=True,
+            enable_usage_metrics=True,
+        ),
+    )
+
+    @transport.event_handler("on_client_connected")
+    async def on_client_connected(transport, client):
+        logger.info(f"Client connected")
+        # Kick off the conversation.
+        messages.append({"role": "system", "content": "Please introduce yourself to the user."})
+        await task.queue_frames([context_aggregator.user().get_context_frame()])
+
+    @transport.event_handler("on_client_disconnected")
+    async def on_client_disconnected(transport, client):
+        logger.info(f"Client disconnected")
+        await task.cancel()
+
+    runner = PipelineRunner(handle_sigint=handle_sigint)
+
+    await runner.run(task)
+
+
+if __name__ == "__main__":
+    from pipecat.examples.run import main
+
+    main(run_example, transport_params=transport_params)

examples/foundational/13i-soniox-transcription.py

@@ -0,0 +1,81 @@
+#
+# Copyright (c) 2024–2025, Daily
+#
+# SPDX-License-Identifier: BSD 2-Clause License
+#
+
+import argparse
+import os
+
+from dotenv import load_dotenv
+from loguru import logger
+
+from pipecat.audio.vad.silero import SileroVADAnalyzer
+from pipecat.frames.frames import Frame, TranscriptionFrame
+from pipecat.pipeline.pipeline import Pipeline
+from pipecat.pipeline.runner import PipelineRunner
+from pipecat.pipeline.task import PipelineTask
+from pipecat.processors.frame_processor import FrameDirection, FrameProcessor
+from pipecat.services.soniox.stt import SonioxSTTService
+from pipecat.transports.base_transport import BaseTransport, TransportParams
+from pipecat.transports.network.fastapi_websocket import FastAPIWebsocketParams
+from pipecat.transports.services.daily import DailyParams
+
+load_dotenv(override=True)
+
+
+class TranscriptionLogger(FrameProcessor):
+    async def process_frame(self, frame: Frame, direction: FrameDirection):
+        await super().process_frame(frame, direction)
+
+        if isinstance(frame, TranscriptionFrame):
+            print(f"Transcription: {frame.text}")
+
+
+transport_params = {
+    "daily": lambda: DailyParams(
+        audio_in_enabled=True,
+        vad_analyzer=SileroVADAnalyzer(),
+    ),
+    "twilio": lambda: FastAPIWebsocketParams(
+        audio_in_enabled=True,
+        vad_analyzer=SileroVADAnalyzer(),
+    ),
+    "webrtc": lambda: TransportParams(
+        audio_in_enabled=True,
+        vad_analyzer=SileroVADAnalyzer(),
+    ),
+}
+
+
+async def run_example(transport: BaseTransport, _: argparse.Namespace, handle_sigint: bool):
+    logger.info(f"Starting bot")
+
+    stt = SonioxSTTService(
+        api_key=os.getenv("SONIOX_API_KEY"),
+    )
+
+    tl = TranscriptionLogger()
+
+    pipeline = Pipeline([transport.input(), stt, tl])
+
+    task = PipelineTask(pipeline)
+
+    @transport.event_handler("on_client_disconnected")
+    async def on_client_disconnected(transport, client):
+        logger.info(f"Client disconnected")
+
+    @transport.event_handler("on_client_closed")
+    async def on_client_closed(transport, client):
+        logger.info(f"Client closed connection")
+        await task.cancel()
+
+    runner = PipelineRunner(handle_sigint=False)
+
+    await runner.run(task)
+
+
+if __name__ == "__main__":
+    from pipecat.examples.run import main
+
+    main(run_example, transport_params=transport_params)

examples/foundational/38b-smart-turn-local.py

@@ -11,7 +11,7 @@ from dotenv import load_dotenv
 from loguru import logger
 
 from pipecat.audio.turn.smart_turn.base_smart_turn import SmartTurnParams
-from pipecat.audio.turn.smart_turn.local_smart_turn import LocalSmartTurnAnalyzer
+from pipecat.audio.turn.smart_turn.local_smart_turn_v2 import LocalSmartTurnAnalyzerV2
 from pipecat.audio.vad.silero import SileroVADAnalyzer
 from pipecat.audio.vad.vad_analyzer import VADParams
 from pipecat.pipeline.pipeline import Pipeline
@@ -37,7 +37,7 @@ load_dotenv(override=True)
 #   # Hugging Face uses LFS to store large model files, including .mlpackage
 #   git lfs install
 #   # Clone the repo with the smart_turn_classifier.mlpackage
-#   git clone https://huggingface.co/pipecat-ai/smart-turn
+#   git clone https://huggingface.co/pipecat-ai/smart-turn-v2
 #
 # Then set the env variable:
 #   export LOCAL_SMART_TURN_MODEL_PATH=./smart-turn
@@ -52,7 +52,7 @@ transport_params = {
         audio_in_enabled=True,
         audio_out_enabled=True,
         vad_analyzer=SileroVADAnalyzer(params=VADParams(stop_secs=0.2)),
-        turn_analyzer=LocalSmartTurnAnalyzer(
+        turn_analyzer=LocalSmartTurnAnalyzerV2(
             smart_turn_model_path=smart_turn_model_path, params=SmartTurnParams()
         ),
     ),
@@ -60,7 +60,7 @@ transport_params = {
         audio_in_enabled=True,
         audio_out_enabled=True,
         vad_analyzer=SileroVADAnalyzer(params=VADParams(stop_secs=0.2)),
-        turn_analyzer=LocalSmartTurnAnalyzer(
+        turn_analyzer=LocalSmartTurnAnalyzerV2(
             smart_turn_model_path=smart_turn_model_path, params=SmartTurnParams()
         ),
     ),
@@ -68,7 +68,7 @@ transport_params = {
         audio_in_enabled=True,
         audio_out_enabled=True,
         vad_analyzer=SileroVADAnalyzer(params=VADParams(stop_secs=0.2)),
-        turn_analyzer=LocalSmartTurnAnalyzer(
+        turn_analyzer=LocalSmartTurnAnalyzerV2(
             smart_turn_model_path=smart_turn_model_path, params=SmartTurnParams()
         ),
     ),

examples/freeze-test/client/src/app.ts

@@ -20,11 +20,10 @@ import {
 } from '@pipecat-ai/client-js';
 import {
   ProtobufFrameSerializer,
-  WebSocketTransport
+  WebSocketTransport,
-} from "@pipecat-ai/websocket-transport";
+} from '@pipecat-ai/websocket-transport';
 
 class RecordingSerializer extends ProtobufFrameSerializer {
-
   private lastTimestamp: number | null = null;
   private recordingAudioToSend: boolean = false;
   private _recordedAudio: { data: ArrayBuffer; delay: number }[] = [];
@@ -40,7 +39,11 @@ class RecordingSerializer extends ProtobufFrameSerializer {
   }
 
   // @ts-ignore
-  serializeAudio(data: ArrayBuffer, sampleRate: number, numChannels: number): Uint8Array | null {
+  serializeAudio(
+    data: ArrayBuffer,
+    sampleRate: number,
+    numChannels: number
+  ): Uint8Array | null {
     if (this.recordingAudioToSend) {
       const now = Date.now();
       // Compute delay since last packet
@@ -55,13 +58,13 @@ class RecordingSerializer extends ProtobufFrameSerializer {
   }
 
   public get recordedAudio() {
-    return this._recordedAudio
+    return this._recordedAudio;
   }
 }
 
 class WebsocketClientApp {
-  private ENABLE_RECORDING_MODE = false
+  private ENABLE_RECORDING_MODE = false;
-  private RECORDING_TIME_MS = 10000
+  private RECORDING_TIME_MS = 10000;
 
   private rtviClient: RTVIClient | null = null;
   private connectBtn: HTMLButtonElement | null = null;
@@ -71,7 +74,7 @@ class WebsocketClientApp {
   private botAudio: HTMLAudioElement;
 
   private declare websocketTransport: WebSocketTransport;
-  private sendRecordedAudio: boolean = false
+  private sendRecordedAudio: boolean = false;
   private declare recordingSerializer: RecordingSerializer;
 
   private playBtn: HTMLButtonElement | null = null;
@@ -91,8 +94,12 @@ class WebsocketClientApp {
    * Set up references to DOM elements and create necessary media elements
    */
   private setupDOMElements(): void {
-    this.connectBtn = document.getElementById('connect-btn') as HTMLButtonElement;
+    this.connectBtn = document.getElementById(
-    this.disconnectBtn = document.getElementById('disconnect-btn') as HTMLButtonElement;
+      'connect-btn'
+    ) as HTMLButtonElement;
+    this.disconnectBtn = document.getElementById(
+      'disconnect-btn'
+    ) as HTMLButtonElement;
     this.statusSpan = document.getElementById('connection-status');
     this.debugLog = document.getElementById('debug-log');
     this.playBtn = document.getElementById('play-btn') as HTMLButtonElement;
@@ -105,8 +112,12 @@ class WebsocketClientApp {
   private setupEventListeners(): void {
     this.connectBtn?.addEventListener('click', () => this.connect());
     this.disconnectBtn?.addEventListener('click', () => this.disconnect());
-    this.playBtn?.addEventListener('click', () => this.startSendingRecordedAudio());
+    this.playBtn?.addEventListener('click', () =>
-    this.stopBtn?.addEventListener('click', () => this.stopSendingRecordedAudio());
+      this.startSendingRecordedAudio()
+    );
+    this.stopBtn?.addEventListener('click', () =>
+      this.stopSendingRecordedAudio()
+    );
   }
 
   /**
@@ -165,7 +176,9 @@ class WebsocketClientApp {
 
     // Listen for tracks stopping
     this.rtviClient.on(RTVIEvent.TrackStopped, (track, participant) => {
-      this.log(`Track stopped: ${track.kind} from ${participant?.name || 'unknown'}`);
+      this.log(
+        `Track stopped: ${track.kind} from ${participant?.name || 'unknown'}`
+      );
     });
   }
 
@@ -175,7 +188,10 @@ class WebsocketClientApp {
    */
   private setupAudioTrack(track: MediaStreamTrack): void {
     this.log('Setting up audio track');
-    if (this.botAudio.srcObject && "getAudioTracks" in this.botAudio.srcObject) {
+    if (
+      this.botAudio.srcObject &&
+      'getAudioTracks' in this.botAudio.srcObject
+    ) {
       const oldTrack = this.botAudio.srcObject.getAudioTracks()[0];
       if (oldTrack?.id === track.id) return;
     }
@@ -190,27 +206,17 @@ class WebsocketClientApp {
     try {
       const startTime = Date.now();
 
-      this.recordingSerializer = new RecordingSerializer()
+      this.recordingSerializer = new RecordingSerializer();
-      const transport = this.ENABLE_RECORDING_MODE ?
+      const ws_opts = {
-          new WebSocketTransport({
+        serializer: this.ENABLE_RECORDING_MODE
-            serializer: this.recordingSerializer,
+          ? this.recordingSerializer
-            recorderSampleRate: 8000,
+          : new ProtobufFrameSerializer(),
-            playerSampleRate:8000
+        recorderSampleRate: 8000,
-          }) :
+        playerSampleRate: 8000,
-          new WebSocketTransport({
+      };
-            serializer: new ProtobufFrameSerializer(),
-            recorderSampleRate: 8000,
-            playerSampleRate:8000
-          });
-      this.websocketTransport = transport
 
       const RTVIConfig: RTVIClientOptions = {
-        transport,
+        transport: new WebSocketTransport(ws_opts),
-        params: {
-          // The baseURL and endpoint of your bot server that the client will connect to
-          baseUrl: 'http://localhost:7860',
-          endpoints: { connect: '/connect' },
-        },
         enableMic: true,
         enableCam: false,
         callbacks: {
@@ -238,27 +244,34 @@ class WebsocketClientApp {
           onMessageError: (error) => console.error('Message error:', error),
           onError: (error) => console.error('Error:', error),
         },
-      }
+      };
       this.rtviClient = new RTVIClient(RTVIConfig);
+      this.websocketTransport = this.rtviClient.transport;
       this.setupTrackListeners();
 
       this.log('Initializing devices...');
       await this.rtviClient.initDevices();
 
       this.log('Connecting to bot...');
-      await this.rtviClient.connect();
+      await this.rtviClient.connect({
+        endpoint: 'http://localhost:7860/connect',
+      });
 
       const timeTaken = Date.now() - startTime;
       this.log(`Connection complete, timeTaken: ${timeTaken}`);
 
       if (this.ENABLE_RECORDING_MODE) {
-        this.log(`Starting to recording the next ${(this.RECORDING_TIME_MS/1000)}s of audio`);
+        this.log(
-        this.recordingSerializer.startRecording()
+          `Starting to recording the next ${
-        await this.sleep(this.RECORDING_TIME_MS)
+            this.RECORDING_TIME_MS / 1000
-        this.recordingSerializer.stopRecording()
+          }s of audio`
-        this.log("Recording stopped");
+        );
-        this.rtviClient.enableMic(false)
+        this.recordingSerializer.startRecording();
-        this.startSendingRecordedAudio()
+        await this.sleep(this.RECORDING_TIME_MS);
+        this.recordingSerializer.stopRecording();
+        this.log('Recording stopped');
+        this.rtviClient.enableMic(false);
+        this.startSendingRecordedAudio();
       }
     } catch (error) {
       this.log(`Error connecting: ${(error as Error).message}`);
@@ -280,11 +293,16 @@ class WebsocketClientApp {
   public async disconnect(): Promise<void> {
     if (this.rtviClient) {
       try {
-        this.stopSendingRecordedAudio()
+        this.stopSendingRecordedAudio();
         await this.rtviClient.disconnect();
         this.rtviClient = null;
-        if (this.botAudio.srcObject && "getAudioTracks" in this.botAudio.srcObject) {
+        if (
-          this.botAudio.srcObject.getAudioTracks().forEach((track) => track.stop());
+          this.botAudio.srcObject &&
+          'getAudioTracks' in this.botAudio.srcObject
+        ) {
+          this.botAudio.srcObject
+            .getAudioTracks()
+            .forEach((track) => track.stop());
           this.botAudio.srcObject = null;
         }
       } catch (error) {
@@ -294,21 +312,21 @@ class WebsocketClientApp {
   }
 
   private startSendingRecordedAudio() {
-    this.sendRecordedAudio = true
+    this.sendRecordedAudio = true;
     if (this.playBtn) this.playBtn.disabled = true;
     if (this.stopBtn) this.stopBtn.disabled = false;
-    void this.replayAudio()
+    void this.replayAudio();
   }
 
   private stopSendingRecordedAudio() {
     if (this.stopBtn) this.stopBtn.disabled = true;
     if (this.playBtn) this.playBtn.disabled = false;
-    this.sendRecordedAudio = false
+    this.sendRecordedAudio = false;
   }
 
   private async replayAudio() {
     if (this.sendRecordedAudio) {
-      this.log("Sending recorded audio")
+      this.log('Sending recorded audio');
       for (const chunk of this.recordingSerializer.recordedAudio) {
         await this.sleep(chunk.delay);
         this.websocketTransport.handleUserAudioStream(chunk.data);
@@ -316,14 +334,13 @@ class WebsocketClientApp {
       const randomDelay = 1000 + Math.random() * (10000 - 500);
       await this.sleep(randomDelay);
 
-      void this.replayAudio()
+      void this.replayAudio();
     }
   }
 
   private sleep(ms: number): Promise<void> {
-    return new Promise(resolve => setTimeout(resolve, ms));
+    return new Promise((resolve) => setTimeout(resolve, ms));
   }
-
 }
 
 declare global {

examples/instant-voice/client/javascript/package.json

@@ -18,7 +18,7 @@
     "vite": "^6.3.5"
   },
   "dependencies": {
-    "@pipecat-ai/client-js": "^0.3.5",
+    "@pipecat-ai/client-js": "^1.0.0",
-    "@pipecat-ai/daily-transport": "^0.3.8"
+    "@pipecat-ai/daily-transport": "^1.0.0"
   }
 }

examples/instant-voice/client/javascript/src/app.ts

@@ -5,7 +5,7 @@
  */
 
 /**
- * RTVI Client Implementation
+ * Pipecat Client Implementation
  *
  * This client connects to an RTVI-compatible bot server using WebRTC (via Daily).
  * It handles audio/video streaming and manages the connection lifecycle.
@@ -18,20 +18,22 @@
 
 import {
   Participant,
-  RTVIClient,
+  PipecatClient,
-  RTVIClientOptions,
+  PipecatClientOptions,
   RTVIEvent,
 } from '@pipecat-ai/client-js';
-import { DailyTransport } from '@pipecat-ai/daily-transport';
+import {
+  DailyEventCallbacks,
+  DailyTransport,
+} from '@pipecat-ai/daily-transport';
 import SoundUtils from './util/soundUtils';
-import { InstantVoiceHelper } from './util/instantVoiceHelper';
 
 /**
  * InstantVoiceClient handles the connection and media management for a real-time
  * voice and video interaction with an AI bot.
  */
 class InstantVoiceClient {
-  private declare rtviClient: RTVIClient;
+  private declare pcClient: PipecatClient;
   private connectBtn: HTMLButtonElement | null = null;
   private disconnectBtn: HTMLButtonElement | null = null;
   private statusSpan: HTMLElement | null = null;
@@ -46,7 +48,7 @@ class InstantVoiceClient {
     document.body.appendChild(this.botAudio);
     this.setupDOMElements();
     this.setupEventListeners();
-    this.initializeRTVIClient();
+    this.initializePipecatClient();
   }
 
   /**
@@ -72,16 +74,11 @@ class InstantVoiceClient {
     this.disconnectBtn?.addEventListener('click', () => this.disconnect());
   }
 
-  private initializeRTVIClient(): void {
+  private initializePipecatClient(): void {
-    const RTVIConfig: RTVIClientOptions = {
+    const PipecatConfig: PipecatClientOptions = {
       transport: new DailyTransport({
         bufferLocalAudioUntilBotReady: true,
       }),
-      params: {
-        // The baseURL and endpoint of your bot server that the client will connect to
-        baseUrl: 'http://localhost:7860',
-        endpoints: { connect: '/connect' },
-      },
       enableMic: true,
       enableCam: false,
       callbacks: {
@@ -113,30 +110,23 @@ class InstantVoiceClient {
         onBotTranscript: (data) => this.log(`Bot: ${data.text}`),
         onMessageError: (error) => console.error('Message error:', error),
         onError: (error) => console.error('Error:', error),
-      },
+        onAudioBufferingStarted: () => {
+          SoundUtils.beep();
+          this.updateBufferingStatus('Yes');
+          this.log(
+            `onMicCaptureStarted, timeTaken: ${Date.now() - this.startTime}`
+          );
+        },
+        onAudioBufferingStopped: () => {
+          this.updateBufferingStatus('No');
+          this.log(
+            `onMicCaptureStopped, timeTaken: ${Date.now() - this.startTime}`
+          );
+        },
+      } as DailyEventCallbacks,
     };
 
-    this.rtviClient = new RTVIClient(RTVIConfig);
+    this.pcClient = new PipecatClient(PipecatConfig);
-    this.rtviClient.registerHelper(
-      'transport',
-      new InstantVoiceHelper({
-        callbacks: {
-          onAudioBufferingStarted: () => {
-            SoundUtils.beep();
-            this.updateBufferingStatus('Yes');
-            this.log(
-              `onMicCaptureStarted, timeTaken: ${Date.now() - this.startTime}`
-            );
-          },
-          onAudioBufferingStopped: () => {
-            this.updateBufferingStatus('No');
-            this.log(
-              `onMicCaptureStopped, timeTaken: ${Date.now() - this.startTime}`
-            );
-          },
-        },
-      })
-    );
     this.setupTrackListeners();
   }
 
@@ -182,8 +172,8 @@ class InstantVoiceClient {
    * This is called when the bot is ready or when the transport state changes to ready
    */
   setupMediaTracks() {
-    if (!this.rtviClient) return;
+    if (!this.pcClient) return;
-    const tracks = this.rtviClient.tracks();
+    const tracks = this.pcClient.tracks();
     if (tracks.bot?.audio) {
       this.setupAudioTrack(tracks.bot.audio);
     }
@@ -194,10 +184,10 @@ class InstantVoiceClient {
    * This handles new tracks being added during the session
    */
   setupTrackListeners() {
-    if (!this.rtviClient) return;
+    if (!this.pcClient) return;
 
     // Listen for new tracks starting
-    this.rtviClient.on(RTVIEvent.TrackStarted, (track, participant) => {
+    this.pcClient.on(RTVIEvent.TrackStarted, (track, participant) => {
       // Only handle non-local (bot) tracks
       if (!participant?.local && track.kind === 'audio') {
         this.setupAudioTrack(track);
@@ -205,7 +195,7 @@ class InstantVoiceClient {
     });
 
     // Listen for tracks stopping
-    this.rtviClient.on(RTVIEvent.TrackStopped, (track, participant) => {
+    this.pcClient.on(RTVIEvent.TrackStopped, (track, participant) => {
       this.log(
         `Track stopped: ${track.kind} from ${participant?.name || 'unknown'}`
       );
@@ -230,22 +220,25 @@ class InstantVoiceClient {
 
   /**
    * Initialize and connect to the bot
-   * This sets up the RTVI client, initializes devices, and establishes the connection
+   * This sets up the Pipecat client, initializes devices, and establishes the connection
    */
   public async connect(): Promise<void> {
     try {
       this.startTime = Date.now();
       this.log('Connecting to bot...');
-      await this.rtviClient.connect();
+      await this.pcClient.connect({
+        // The baseURL and endpoint of your bot server that the client will connect to
+        endpoint: 'http://localhost:7860/connect',
+      });
     } catch (error) {
       this.log(`Error connecting: ${(error as Error).message}`);
       this.updateStatus('Error');
       this.updateBufferingStatus('No');
 
       // Clean up if there's an error
-      if (this.rtviClient) {
+      if (this.pcClient) {
         try {
-          await this.rtviClient.disconnect();
+          await this.pcClient.disconnect();
         } catch (disconnectError) {
           this.log(`Error during disconnect: ${disconnectError}`);
         }
@@ -258,7 +251,7 @@ class InstantVoiceClient {
    */
   public async disconnect(): Promise<void> {
     try {
-      await this.rtviClient.disconnect();
+      await this.pcClient.disconnect();
       if (
         this.botAudio.srcObject &&
         'getAudioTracks' in this.botAudio.srcObject

examples/instant-voice/client/javascript/src/util/instantVoiceHelper.ts

@@ -1,39 +0,0 @@
-import {RTVIClientHelper, RTVIClientHelperOptions, RTVIMessage} from "@pipecat-ai/client-js";
-import {DailyRTVIMessageType} from '@pipecat-ai/daily-transport';
-
-export type InstantVoiceHelperCallbacks = Partial<{
-    onAudioBufferingStarted: () => void;
-    onAudioBufferingStopped: () => void;
-}>;
-
-// --- Interface and class
-export interface InstantVoiceHelperOptions extends RTVIClientHelperOptions {
-    callbacks?: InstantVoiceHelperCallbacks;
-}
-export class InstantVoiceHelper extends RTVIClientHelper {
-
-    protected declare _options: InstantVoiceHelperOptions;
-
-    constructor(options: InstantVoiceHelperOptions) {
-        super(options);
-    }
-
-    handleMessage(rtviMessage: RTVIMessage): void {
-        switch (rtviMessage.type) {
-            case DailyRTVIMessageType.AUDIO_BUFFERING_STARTED:
-                if (this._options.callbacks?.onAudioBufferingStarted) {
-                    this._options.callbacks?.onAudioBufferingStarted()
-                }
-                break;
-            case DailyRTVIMessageType.AUDIO_BUFFERING_STOPPED:
-                if (this._options.callbacks?.onAudioBufferingStopped) {
-                    this._options.callbacks?.onAudioBufferingStopped()
-                }
-                break;
-        }
-    }
-
-    getMessageTypes(): string[] {
-        return [DailyRTVIMessageType.AUDIO_BUFFERING_STARTED, DailyRTVIMessageType.AUDIO_BUFFERING_STOPPED];
-    }
-}

examples/news-chatbot/client/javascript/package.json

@@ -15,7 +15,7 @@
     "vite": "^6.3.5"
   },
   "dependencies": {
-    "@pipecat-ai/client-js": "^0.3.5",
+    "@pipecat-ai/client-js": "^1.0.0",
-    "@pipecat-ai/daily-transport": "^0.3.8"
+    "@pipecat-ai/daily-transport": "^1.0.0"
   }
 }

examples/news-chatbot/client/javascript/src/app.js

@@ -5,7 +5,7 @@
  */
 
 /**
- * RTVI Client Implementation
+ * Pipecat Client Implementation
  *
  * This client connects to an RTVI-compatible bot server using WebRTC (via Daily).
  * It handles audio/video streaming and manages the connection lifecycle.
@@ -16,78 +16,9 @@
  * - Browser with WebRTC support
  */
 
-import {
+import { LogLevel, PipecatClient, RTVIEvent } from '@pipecat-ai/client-js';
-  LogLevel,
-  RTVIClient,
-  RTVIClientHelper,
-  RTVIEvent,
-} from '@pipecat-ai/client-js';
 import { DailyTransport } from '@pipecat-ai/daily-transport';
 
-class SearchResponseHelper extends RTVIClientHelper {
-  constructor(contentPanel) {
-    super();
-    this.contentPanel = contentPanel;
-  }
-
-  handleMessage(rtviMessage) {
-    console.log('SearchResponseHelper, received message:', rtviMessage);
-    if (rtviMessage.data) {
-      // Clear existing content
-      this.contentPanel.innerHTML = '';
-
-      // Create a container for all content
-      const contentContainer = document.createElement('div');
-      contentContainer.className = 'content-container';
-
-      // Add the search_result
-      if (rtviMessage.data.search_result) {
-        const searchResultDiv = document.createElement('div');
-        searchResultDiv.className = 'search-result';
-        searchResultDiv.textContent = rtviMessage.data.search_result;
-        contentContainer.appendChild(searchResultDiv);
-      }
-
-      // Add the sources
-      if (rtviMessage.data.origins) {
-        const sourcesDiv = document.createElement('div');
-        sourcesDiv.className = 'sources';
-
-        const sourcesTitle = document.createElement('h3');
-        sourcesTitle.className = 'sources-title';
-        sourcesTitle.textContent = 'Sources:';
-        sourcesDiv.appendChild(sourcesTitle);
-
-        rtviMessage.data.origins.forEach((origin) => {
-          const sourceLink = document.createElement('a');
-          sourceLink.className = 'source-link';
-          sourceLink.href = origin.site_uri;
-          sourceLink.target = '_blank';
-          sourceLink.textContent = origin.site_title;
-          sourcesDiv.appendChild(sourceLink);
-        });
-
-        contentContainer.appendChild(sourcesDiv);
-      }
-
-      // Add the rendered_content in an iframe
-      if (rtviMessage.data.rendered_content) {
-        const iframe = document.createElement('iframe');
-        iframe.className = 'iframe-container';
-        iframe.srcdoc = rtviMessage.data.rendered_content;
-        contentContainer.appendChild(iframe);
-      }
-
-      // Append the content container to the content panel
-      this.contentPanel.appendChild(contentContainer);
-    }
-  }
-
-  getMessageTypes() {
-    return ['bot-llm-search-response'];
-  }
-}
-
 /**
  * ChatbotClient handles the connection and media management for a real-time
  * voice and video interaction with an AI bot.
@@ -95,7 +26,7 @@ class SearchResponseHelper extends RTVIClientHelper {
 class ChatbotClient {
   constructor() {
     // Initialize client state
-    this.rtviClient = null;
+    this.pcClient = null;
     this.setupDOMElements();
     this.setupEventListeners();
   }
@@ -160,10 +91,10 @@ class ChatbotClient {
    * This is called when the bot is ready or when the transport state changes to ready
    */
   setupMediaTracks() {
-    if (!this.rtviClient) return;
+    if (!this.pcClient) return;
 
     // Get current tracks from the client
-    const tracks = this.rtviClient.tracks();
+    const tracks = this.pcClient.tracks();
 
     // Set up any available bot tracks
     if (tracks.bot?.audio) {
@@ -176,10 +107,10 @@ class ChatbotClient {
    * This handles new tracks being added during the session
    */
   setupTrackListeners() {
-    if (!this.rtviClient) return;
+    if (!this.pcClient) return;
 
     // Listen for new tracks starting
-    this.rtviClient.on(RTVIEvent.TrackStarted, (track, participant) => {
+    this.pcClient.on(RTVIEvent.TrackStarted, (track, participant) => {
       // Only handle non-local (bot) tracks
       if (!participant?.local && track.kind === 'audio') {
         this.setupAudioTrack(track);
@@ -187,7 +118,7 @@ class ChatbotClient {
     });
 
     // Listen for tracks stopping
-    this.rtviClient.on(RTVIEvent.TrackStopped, (track, participant) => {
+    this.pcClient.on(RTVIEvent.TrackStopped, (track, participant) => {
       this.log(
         `Track stopped event: ${track.kind} from ${
           participant?.name || 'unknown'
@@ -213,20 +144,13 @@ class ChatbotClient {
 
   /**
    * Initialize and connect to the bot
-   * This sets up the RTVI client, initializes devices, and establishes the connection
+   * This sets up the Pipecat client, initializes devices, and establishes the connection
    */
   async connect() {
     try {
-      // Initialize the RTVI client with a Daily WebRTC transport and our configuration
+      // Initialize the Pipecat client with a Daily WebRTC transport and our configuration
-      this.rtviClient = new RTVIClient({
+      this.pcClient = new PipecatClient({
         transport: new DailyTransport(),
-        params: {
-          // The baseURL and endpoint of your bot server that the client will connect to
-          baseUrl: 'http://localhost:7860',
-          endpoints: {
-            connect: '/connect',
-          },
-        },
         enableMic: true, // Enable microphone for user input
         enableCam: false,
         callbacks: {
@@ -251,6 +175,8 @@ class ChatbotClient {
               this.setupMediaTracks();
             }
           },
+          // Handle search response events
+          onBotLlmSearchResponse: this.handleSearchResponse.bind(this),
           // Handle bot connection events
           onBotConnected: (participant) => {
             this.log(`Bot connected: ${JSON.stringify(participant)}`);
@@ -281,22 +207,22 @@ class ChatbotClient {
           },
         },
       });
-      //this.rtviClient.setLogLevel(LogLevel.DEBUG)
+
-      this.rtviClient.registerHelper(
+      //this.pcClient.setLogLevel(LogLevel.DEBUG)
-        'llm',
-        new SearchResponseHelper(this.searchResultContainer)
-      );
 
       // Set up listeners for media track events
       this.setupTrackListeners();
 
       // Initialize audio devices
       this.log('Initializing devices...');
-      await this.rtviClient.initDevices();
+      await this.pcClient.initDevices();
 
       // Connect to the bot
       this.log('Connecting to bot...');
-      await this.rtviClient.connect();
+      await this.pcClient.connect({
+        // The baseURL and endpoint of your bot server that the client will connect to
+        endpoint: 'http://localhost:7860/connect',
+      });
 
       this.log('Connection complete');
     } catch (error) {
@@ -306,9 +232,9 @@ class ChatbotClient {
       this.updateStatus('Error');
 
       // Clean up if there's an error
-      if (this.rtviClient) {
+      if (this.pcClient) {
         try {
-          await this.rtviClient.disconnect();
+          await this.pcClient.disconnect();
         } catch (disconnectError) {
           this.log(`Error during disconnect: ${disconnectError.message}`);
         }
@@ -320,11 +246,11 @@ class ChatbotClient {
    * Disconnect from the bot and clean up media resources
    */
   async disconnect() {
-    if (this.rtviClient) {
+    if (this.pcClient) {
       try {
-        // Disconnect the RTVI client
+        // Disconnect the Pipecat client
-        await this.rtviClient.disconnect();
+        await this.pcClient.disconnect();
-        this.rtviClient = null;
+        this.pcClient = null;
 
         // Clean up audio
         if (this.botAudio.srcObject) {
@@ -339,6 +265,57 @@ class ChatbotClient {
       }
     }
   }
+
+  handleSearchResponse(response) {
+    console.log('SearchResponseHelper, received message:', response);
+    // Clear existing content
+    this.searchResultContainer.innerHTML = '';
+
+    // Create a container for all content
+    const contentContainer = document.createElement('div');
+    contentContainer.className = 'content-container';
+
+    // Add the search_result
+    if (response.search_result) {
+      const searchResultDiv = document.createElement('div');
+      searchResultDiv.className = 'search-result';
+      searchResultDiv.textContent = response.search_result;
+      contentContainer.appendChild(searchResultDiv);
+    }
+
+    // Add the sources
+    if (response.origins) {
+      const sourcesDiv = document.createElement('div');
+      sourcesDiv.className = 'sources';
+
+      const sourcesTitle = document.createElement('h3');
+      sourcesTitle.className = 'sources-title';
+      sourcesTitle.textContent = 'Sources:';
+      sourcesDiv.appendChild(sourcesTitle);
+
+      response.origins.forEach((origin) => {
+        const sourceLink = document.createElement('a');
+        sourceLink.className = 'source-link';
+        sourceLink.href = origin.site_uri;
+        sourceLink.target = '_blank';
+        sourceLink.textContent = origin.site_title;
+        sourcesDiv.appendChild(sourceLink);
+      });
+
+      contentContainer.appendChild(sourcesDiv);
+    }
+
+    // Add the rendered_content in an iframe
+    if (response.rendered_content) {
+      const iframe = document.createElement('iframe');
+      iframe.className = 'iframe-container';
+      iframe.srcdoc = response.rendered_content;
+      contentContainer.appendChild(iframe);
+    }
+
+    // Append the content container to the content panel
+    this.searchResultContainer.appendChild(contentContainer);
+  }
 }
 
 // Initialize the client when the page loads

examples/p2p-webrtc/video-transform/client/typescript/package.json

@@ -18,7 +18,7 @@
     "vite": "^6.3.5"
   },
   "dependencies": {
-    "@pipecat-ai/client-js": "^0.3.2",
+    "@pipecat-ai/client-js": "^1.0.0",
-    "@pipecat-ai/small-webrtc-transport": "^0.0.2"
+    "@pipecat-ai/small-webrtc-transport": "^1.0.0"
   }
 }

examples/p2p-webrtc/video-transform/client/typescript/src/app.ts

@@ -1,217 +1,236 @@
+import { SmallWebRTCTransport } from '@pipecat-ai/small-webrtc-transport';
 import {
-    SmallWebRTCTransport
+  BotLLMTextData,
-} from "@pipecat-ai/small-webrtc-transport";
+  Participant,
-import {Participant, RTVIClient, RTVIClientOptions, Transport} from "@pipecat-ai/client-js";
+  PipecatClient,
+  PipecatClientOptions,
+  TranscriptData,
+  TransportState,
+} from '@pipecat-ai/client-js';
 
 class WebRTCApp {
+  private declare connectBtn: HTMLButtonElement;
+  private declare disconnectBtn: HTMLButtonElement;
+  private declare muteBtn: HTMLButtonElement;
 
-    private declare connectBtn: HTMLButtonElement;
+  private declare audioInput: HTMLSelectElement;
-    private declare disconnectBtn: HTMLButtonElement;
+  private declare videoInput: HTMLSelectElement;
-    private declare muteBtn: HTMLButtonElement;
+  private declare audioCodec: HTMLSelectElement;
+  private declare videoCodec: HTMLSelectElement;
 
-    private declare audioInput: HTMLSelectElement;
+  private declare videoElement: HTMLVideoElement;
-    private declare videoInput: HTMLSelectElement;
+  private declare audioElement: HTMLAudioElement;
-    private declare audioCodec: HTMLSelectElement;
-    private declare videoCodec: HTMLSelectElement;
 
-    private declare videoElement: HTMLVideoElement;
+  private debugLog: HTMLElement | null = null;
-    private declare audioElement: HTMLAudioElement;
+  private statusSpan: HTMLElement | null = null;
 
-    private debugLog: HTMLElement | null = null;
+  private declare smallWebRTCTransport: SmallWebRTCTransport;
-    private statusSpan: HTMLElement | null = null;
+  private declare pcClient: PipecatClient;
 
-    private declare smallWebRTCTransport: SmallWebRTCTransport;
+  constructor() {
-    private declare rtviClient: RTVIClient;
+    this.setupDOMElements();
+    this.setupDOMEventListeners();
+    this.initializePipecatClient();
+    void this.populateDevices();
+  }
 
-    constructor() {
+  private initializePipecatClient(): void {
-        this.setupDOMElements();
+    const opts: PipecatClientOptions = {
-        this.setupDOMEventListeners();
+      transport: new SmallWebRTCTransport({ connectionUrl: '/api/offer' }),
-        this.initializeRTVIClient()
+      enableMic: true,
-        void this.populateDevices();
+      enableCam: true,
+      callbacks: {
+        onTransportStateChanged: (state: TransportState) => {
+          this.log(`Transport state: ${state}`);
+        },
+        onConnected: () => {
+          this.onConnectedHandler();
+        },
+        onBotReady: () => {
+          this.log('Bot is ready.');
+        },
+        onDisconnected: () => {
+          this.onDisconnectedHandler();
+        },
+        onUserStartedSpeaking: () => {
+          this.log('User started speaking.');
+        },
+        onUserStoppedSpeaking: () => {
+          this.log('User stopped speaking.');
+        },
+        onBotStartedSpeaking: () => {
+          this.log('Bot started speaking.');
+        },
+        onBotStoppedSpeaking: () => {
+          this.log('Bot stopped speaking.');
+        },
+        onUserTranscript: (transcript: TranscriptData) => {
+          if (transcript.final) {
+            this.log(`User transcript: ${transcript.text}`);
+          }
+        },
+        onBotTranscript: (data: BotLLMTextData) => {
+          this.log(`Bot transcript: ${data.text}`);
+        },
+        onTrackStarted: (
+          track: MediaStreamTrack,
+          participant?: Participant
+        ) => {
+          if (participant?.local) {
+            return;
+          }
+          this.onBotTrackStarted(track);
+        },
+        onServerMessage: (msg: unknown) => {
+          this.log(`Server message: ${msg}`);
+        },
+      },
+    };
+    this.pcClient = new PipecatClient(opts);
+    this.smallWebRTCTransport = this.pcClient.transport as SmallWebRTCTransport;
+  }
+
+  private setupDOMElements(): void {
+    this.connectBtn = document.getElementById(
+      'connect-btn'
+    ) as HTMLButtonElement;
+    this.disconnectBtn = document.getElementById(
+      'disconnect-btn'
+    ) as HTMLButtonElement;
+    this.muteBtn = document.getElementById('mute-btn') as HTMLButtonElement;
+
+    this.audioInput = document.getElementById(
+      'audio-input'
+    ) as HTMLSelectElement;
+    this.videoInput = document.getElementById(
+      'video-input'
+    ) as HTMLSelectElement;
+    this.audioCodec = document.getElementById(
+      'audio-codec'
+    ) as HTMLSelectElement;
+    this.videoCodec = document.getElementById(
+      'video-codec'
+    ) as HTMLSelectElement;
+
+    this.videoElement = document.getElementById(
+      'bot-video'
+    ) as HTMLVideoElement;
+    this.audioElement = document.getElementById(
+      'bot-audio'
+    ) as HTMLAudioElement;
+
+    this.debugLog = document.getElementById('debug-log');
+    this.statusSpan = document.getElementById('connection-status');
+  }
+
+  private setupDOMEventListeners(): void {
+    this.connectBtn.addEventListener('click', () => this.start());
+    this.disconnectBtn.addEventListener('click', () => this.stop());
+    this.audioInput.addEventListener('change', (e) => {
+      // @ts-ignore
+      let audioDevice = e.target?.value;
+      this.pcClient.updateMic(audioDevice);
+    });
+    this.videoInput.addEventListener('change', (e) => {
+      // @ts-ignore
+      let videoDevice = e.target?.value;
+      this.pcClient.updateCam(videoDevice);
+    });
+    this.muteBtn.addEventListener('click', () => {
+      let isCamEnabled = this.pcClient.isCamEnabled;
+      this.pcClient.enableCam(!isCamEnabled);
+      this.muteBtn.textContent = isCamEnabled ? '📵' : '📷';
+    });
+  }
+
+  private log(message: string): void {
+    if (!this.debugLog) return;
+    const entry = document.createElement('div');
+    entry.textContent = `${new Date().toISOString()} - ${message}`;
+    if (message.startsWith('User: ')) {
+      entry.style.color = '#2196F3';
+    } else if (message.startsWith('Bot: ')) {
+      entry.style.color = '#4CAF50';
     }
+    this.debugLog.appendChild(entry);
+    this.debugLog.scrollTop = this.debugLog.scrollHeight;
+  }
 
-    private initializeRTVIClient(): void {
+  private clearAllLogs() {
-        const transport = new SmallWebRTCTransport();
+    this.debugLog!.innerText = '';
-        const RTVIConfig: RTVIClientOptions = {
+  }
-            params: {
+
-                baseUrl: "/api/offer"
+  private updateStatus(status: string): void {
-            },
+    if (this.statusSpan) {
-            transport: transport as Transport,
+      this.statusSpan.textContent = status;
-            enableMic: true,
-            enableCam: true,
-            callbacks: {
-                onTransportStateChanged: (state) => {
-                    this.log(`Transport state: ${state}`)
-                },
-                onConnected: () => {
-                    this.onConnectedHandler()
-                },
-                onBotReady: () => {
-                    this.log("Bot is ready.")
-                },
-                onDisconnected: () => {
-                    this.onDisconnectedHandler()
-                },
-                onUserStartedSpeaking: () => {
-                    this.log("User started speaking.")
-                },
-                onUserStoppedSpeaking: () => {
-                    this.log("User stopped speaking.")
-                },
-                onBotStartedSpeaking: () => {
-                    this.log("Bot started speaking.")
-                },
-                onBotStoppedSpeaking: () => {
-                    this.log("Bot stopped speaking.")
-                },
-                onUserTranscript: (transcript) => {
-                    if (transcript.final) {
-                        this.log(`User transcript: ${transcript.text}`)
-                    }
-                },
-                onBotTranscript: (transcript) => {
-                    this.log(`Bot transcript: ${transcript.text}`)
-                },
-                onTrackStarted: (track: MediaStreamTrack, participant?: Participant) => {
-                    if (participant?.local) {
-                        return
-                    }
-                    this.onBotTrackStarted(track)
-                },
-                onServerMessage: (msg) => {
-                    this.log(`Server message: ${msg}`)
-                }
-            },
-        }
-        RTVIConfig.customConnectHandler = () => Promise.resolve();
-        this.rtviClient = new RTVIClient(RTVIConfig);
-        this.smallWebRTCTransport = transport
     }
+    this.log(`Status: ${status}`);
+  }
 
-    private setupDOMElements(): void {
+  private onConnectedHandler() {
-        this.connectBtn = document.getElementById('connect-btn') as HTMLButtonElement;
+    this.updateStatus('Connected');
-        this.disconnectBtn = document.getElementById('disconnect-btn') as HTMLButtonElement;
+    if (this.connectBtn) this.connectBtn.disabled = true;
-        this.muteBtn = document.getElementById('mute-btn') as HTMLButtonElement;
+    if (this.disconnectBtn) this.disconnectBtn.disabled = false;
+  }
 
-        this.audioInput = document.getElementById('audio-input') as HTMLSelectElement;
+  private onDisconnectedHandler() {
-        this.videoInput = document.getElementById('video-input') as HTMLSelectElement;
+    this.updateStatus('Disconnected');
-        this.audioCodec = document.getElementById('audio-codec') as HTMLSelectElement;
+    if (this.connectBtn) this.connectBtn.disabled = false;
-        this.videoCodec = document.getElementById('video-codec') as HTMLSelectElement;
+    if (this.disconnectBtn) this.disconnectBtn.disabled = true;
+  }
 
-        this.videoElement = document.getElementById('bot-video') as HTMLVideoElement;
+  private onBotTrackStarted(track: MediaStreamTrack) {
-        this.audioElement = document.getElementById('bot-audio') as HTMLAudioElement;
+    if (track.kind === 'video') {
-
+      this.videoElement.srcObject = new MediaStream([track]);
-        this.debugLog = document.getElementById('debug-log');
+    } else {
-        this.statusSpan = document.getElementById('connection-status');
+      this.audioElement.srcObject = new MediaStream([track]);
     }
+  }
 
-    private setupDOMEventListeners(): void {
+  private async populateDevices(): Promise<void> {
-        this.connectBtn.addEventListener("click", () => this.start());
+    const populateSelect = (
-        this.disconnectBtn.addEventListener("click", () => this.stop());
+      select: HTMLSelectElement,
-        this.audioInput.addEventListener("change", (e) => {
+      devices: MediaDeviceInfo[]
-            // @ts-ignore
+    ): void => {
-            let audioDevice = e.target?.value
+      let counter = 1;
-            this.rtviClient.updateMic(audioDevice)
+      devices.forEach((device) => {
-        })
+        const option = document.createElement('option');
-        this.videoInput.addEventListener("change", (e) => {
+        option.value = device.deviceId;
-            // @ts-ignore
+        option.text = device.label || 'Device #' + counter;
-            let videoDevice = e.target?.value
+        select.appendChild(option);
-            this.rtviClient.updateCam(videoDevice)
+        counter += 1;
-        })
+      });
-        this.muteBtn.addEventListener('click', () => {
+    };
-            let isCamEnabled = this.rtviClient.isCamEnabled
-            this.rtviClient.enableCam(!isCamEnabled)
-            this.muteBtn.textContent = isCamEnabled ? '📵' : '📷';
-        });
 
+    try {
+      const audioDevices = await this.pcClient.getAllMics();
+      populateSelect(this.audioInput, audioDevices);
+      const videoDevices = await this.pcClient.getAllCams();
+      populateSelect(this.videoInput, videoDevices);
+    } catch (e) {
+      alert(e);
     }
+  }
 
-    private log(message: string): void {
+  private async start(): Promise<void> {
-        if (!this.debugLog) return;
+    this.clearAllLogs();
-        const entry = document.createElement('div');
+
-        entry.textContent = `${new Date().toISOString()} - ${message}`;
+    this.connectBtn.disabled = true;
-        if (message.startsWith('User: ')) {
+    this.updateStatus('Connecting');
-          entry.style.color = '#2196F3';
+
-        } else if (message.startsWith('Bot: ')) {
+    this.smallWebRTCTransport.setAudioCodec(this.audioCodec.value);
-          entry.style.color = '#4CAF50';
+    this.smallWebRTCTransport.setVideoCodec(this.videoCodec.value);
-        }
+    try {
-        this.debugLog.appendChild(entry);
+      await this.pcClient.connect();
-        this.debugLog.scrollTop = this.debugLog.scrollHeight;
+    } catch (e) {
+      console.log(`Failed to connect ${e}`);
+      this.stop();
     }
+  }
 
-    private clearAllLogs() {
+  private stop(): void {
-        this.debugLog!.innerText = ''
+    void this.pcClient.disconnect();
-    }
+  }
-
-    private updateStatus(status: string): void {
-        if (this.statusSpan) {
-            this.statusSpan.textContent = status;
-        }
-        this.log(`Status: ${status}`);
-   }
-
-    private onConnectedHandler() {
-        this.updateStatus('Connected');
-        if (this.connectBtn) this.connectBtn.disabled = true;
-        if (this.disconnectBtn) this.disconnectBtn.disabled = false;
-    }
-
-    private onDisconnectedHandler() {
-        this.updateStatus('Disconnected');
-        if (this.connectBtn) this.connectBtn.disabled = false;
-        if (this.disconnectBtn) this.disconnectBtn.disabled = true;
-    }
-
-    private onBotTrackStarted(track: MediaStreamTrack) {
-        if (track.kind === 'video') {
-            this.videoElement.srcObject = new MediaStream([track]);
-        } else {
-            this.audioElement.srcObject = new MediaStream([track]);
-        }
-    }
-
-    private async populateDevices(): Promise<void> {
-        const populateSelect = (select: HTMLSelectElement, devices: MediaDeviceInfo[]): void => {
-            let counter = 1;
-            devices.forEach((device) => {
-                const option = document.createElement('option');
-                option.value = device.deviceId;
-                option.text = device.label || ('Device #' + counter);
-                select.appendChild(option);
-                counter += 1;
-            });
-        };
-
-        try {
-            const audioDevices = await this.rtviClient.getAllMics();
-            populateSelect(this.audioInput, audioDevices);
-            const videoDevices = await this.rtviClient.getAllCams();
-            populateSelect(this.videoInput, videoDevices);
-        } catch (e) {
-            alert(e);
-        }
-    }
-
-    private async start(): Promise<void> {
-        this.clearAllLogs()
-
-        this.connectBtn.disabled = true;
-        this.updateStatus("Connecting")
-
-        this.smallWebRTCTransport.setAudioCodec(this.audioCodec.value)
-        this.smallWebRTCTransport.setVideoCodec(this.videoCodec.value)
-        try {
-            await this.rtviClient.connect()
-        } catch (e) {
-            console.log(`Failed to connect ${e}`)
-            this.stop()
-        }
-
-    }
-
-    private stop(): void {
-        void this.rtviClient.disconnect()
-    }
 }
 
 // Create the WebRTCConnection instance

examples/simple-chatbot/client/javascript/index.html

@@ -1,40 +1,51 @@
 <!DOCTYPE html>
 <html lang="en">
+  <head>
+    <meta charset="UTF-8" />
+    <meta name="viewport" content="width=device-width, initial-scale=1.0" />
+    <title>AI Chatbot</title>
+  </head>
 
-<head>
+  <body>
-  <meta charset="UTF-8">
+    <div class="container">
-  <meta name="viewport" content="width=device-width, initial-scale=1.0">
+      <div class="status-bar">
-  <title>AI Chatbot</title>
+        <div class="status">
-</head>
+          Status: <span id="connection-status">Disconnected</span>
-
-<body>
-  <div class="container">
-    <div class="status-bar">
-      <div class="status">
-        Status: <span id="connection-status">Disconnected</span>
-      </div>
-      <div class="controls">
-        <button id="connect-btn">Connect</button>
-        <button id="disconnect-btn" disabled>Disconnect</button>
-      </div>
-    </div>
-
-    <div class="main-content">
-      <div class="bot-container">
-        <div id="bot-video-container">
         </div>
-        <audio id="bot-audio" autoplay></audio>
+        <div class="controls">
+          <button id="connect-btn">Connect</button>
+          <button id="disconnect-btn" disabled>Disconnect</button>
+        </div>
+      </div>
+
+      <div class="main-content">
+        <div class="bot-container">
+          <div id="bot-video-container"></div>
+          <audio id="bot-audio" autoplay></audio>
+        </div>
+      </div>
+
+      <div class="device-bar">
+        <div class="device-controls">
+          <select id="device-selector"></select>
+          <button id="mic-toggle-btn">Unmute Mic</button>
+        </div>
+        <div class="text-input-container">
+          <input
+            type="text"
+            id="text-input"
+            placeholder="Type your message..." />
+          <button id="send-text-btn" disabled>Send</button>
+        </div>
+      </div>
+
+      <div class="debug-panel">
+        <h3>Debug Info</h3>
+        <div id="debug-log"></div>
       </div>
     </div>
 
-    <div class="debug-panel">
+    <script type="module" src="/src/app.js"></script>
-      <h3>Debug Info</h3>
+    <link rel="stylesheet" href="/src/style.css" />
-      <div id="debug-log"></div>
+  </body>
-    </div>
+</html>
-  </div>
-
-  <script type="module" src="/src/app.js"></script>
-  <link rel="stylesheet" href="/src/style.css">
-</body>
-
-</html>

examples/simple-chatbot/client/javascript/package.json

@@ -15,7 +15,7 @@
     "vite": "^6.3.5"
   },
   "dependencies": {
-    "@pipecat-ai/client-js": "^0.3.5",
+    "@pipecat-ai/client-js": "^1.0.0",
-    "@pipecat-ai/daily-transport": "^0.3.8"
+    "@pipecat-ai/daily-transport": "^1.0.0"
   }
 }

examples/simple-chatbot/client/javascript/src/app.js

@@ -5,7 +5,7 @@
  */
 
 /**
- * RTVI Client Implementation
+ * Pipecat Client Implementation
  *
  * This client connects to an RTVI-compatible bot server using WebRTC (via Daily).
  * It handles audio/video streaming and manages the connection lifecycle.
@@ -16,7 +16,7 @@
  * - Browser with WebRTC support
  */
 
-import { RTVIClient, RTVIEvent } from '@pipecat-ai/client-js';
+import { PipecatClient, RTVIEvent } from '@pipecat-ai/client-js';
 import { DailyTransport } from '@pipecat-ai/daily-transport';
 
 /**
@@ -26,9 +26,8 @@ import { DailyTransport } from '@pipecat-ai/daily-transport';
 class ChatbotClient {
   constructor() {
     // Initialize client state
-    this.rtviClient = null;
+    this.pcClient = null;
     this.setupDOMElements();
-    this.setupEventListeners();
     this.initializeClientAndTransport();
   }
 
@@ -42,6 +41,9 @@ class ChatbotClient {
     this.statusSpan = document.getElementById('connection-status');
     this.debugLog = document.getElementById('debug-log');
     this.botVideoContainer = document.getElementById('bot-video-container');
+    this.deviceSelector = document.getElementById('device-selector');
+    this.micToggleBtn = document.getElementById('mic-toggle-btn');
+    this.sendTextBtn = document.getElementById('send-text-btn');
 
     // Create an audio element for bot's voice output
     this.botAudio = document.createElement('audio');
@@ -54,25 +56,78 @@ class ChatbotClient {
    * Set up event listeners for connect/disconnect buttons
    */
   setupEventListeners() {
-    this.connectBtn.addEventListener('click', () => this.connect());
+    this.connectBtn.addEventListener('click', () => {
+      console.log('click');
+      this.connect();
+    });
     this.disconnectBtn.addEventListener('click', () => this.disconnect());
+
+    // Populate device selector
+    this.pcClient.getAllMics().then((mics) => {
+      console.log('Available mics:', mics);
+      mics.forEach((device) => {
+        const option = document.createElement('option');
+        option.value = device.deviceId;
+        option.textContent = device.label || `Microphone ${device.deviceId}`;
+        this.deviceSelector.appendChild(option);
+      });
+    });
+    this.deviceSelector.addEventListener('change', (event) => {
+      const selectedDeviceId = event.target.value;
+      console.log('Selected device ID:', selectedDeviceId);
+      this.pcClient.updateMic(selectedDeviceId);
+    });
+
+    // Handle mic mute/unmute toggle
+    const micToggleBtn = document.getElementById('mic-toggle-btn');
+
+    micToggleBtn.addEventListener('click', async () => {
+      if (this.pcClient.state === 'disconnected') {
+        await this.pcClient.initDevices();
+      } else {
+        this.pcClient.enableMic(!this.pcClient.isMicEnabled);
+      }
+    });
+
+    const textInput = document.getElementById('text-input');
+
+    const sendTextToLLM = () => {
+      this.sendTextBtn.disabled = true; // Disable button to prevent multiple clicks
+      const text = textInput.value.trim();
+      if (text) {
+        void this.pcClient.appendToContext({
+          role: 'user',
+          content: text,
+          run_immediately: true,
+        });
+      }
+      textInput.value = ''; // Clear the input
+      this.sendTextBtn.disabled = false; // Re-enable button after sending
+    };
+
+    this.sendTextBtn.addEventListener('click', sendTextToLLM);
+
+    // Also handle Enter key in the input
+    textInput.addEventListener('keypress', (e) => {
+      if (e.key === 'Enter') {
+        sendTextToLLM();
+      }
+    });
   }
 
+  updateMicToggleButton(micEnabled) {
+    console.log('Mic enabled:', micEnabled, this.pcClient?.isMicEnabled);
+    this.micToggleBtn.textContent = micEnabled ? 'Mute Mic' : 'Unmute Mic';
+  }
   /**
-   * Set up the RTVI client and Daily transport
+   * Set up the Pipecat client and Daily transport
    */
-  initializeClientAndTransport() {
+  async initializeClientAndTransport() {
-    // Initialize the RTVI client with a DailyTransport and our configuration
+    console.log('Initializing Pipecat client and transport...');
-    this.rtviClient = new RTVIClient({
+    // Initialize the Pipecat client with a DailyTransport and our configuration
+    this.pcClient = new PipecatClient({
       transport: new DailyTransport(),
-      params: {
+      enableMic: true,
-        // The baseURL and endpoint of your bot server that the client will connect to
-        baseUrl: 'http://localhost:7860',
-        endpoints: {
-          connect: '/connect',
-        },
-      },
-      enableMic: true, // Enable microphone for user input
       enableCam: false,
       callbacks: {
         // Handle connection state changes
@@ -86,7 +141,9 @@ class ChatbotClient {
           this.updateStatus('Disconnected');
           this.connectBtn.disabled = false;
           this.disconnectBtn.disabled = true;
+          this.sendTextBtn.disabled = true;
           this.log('Client disconnected');
+          this.updateMicToggleButton(false);
         },
         // Handle transport state changes
         onTransportStateChanged: (state) => {
@@ -106,6 +163,7 @@ class ChatbotClient {
         onBotReady: (data) => {
           this.log(`Bot ready: ${JSON.stringify(data)}`);
           this.setupMediaTracks();
+          this.sendTextBtn.disabled = false;
         },
         // Transcript events
         onUserTranscript: (data) => {
@@ -121,14 +179,20 @@ class ChatbotClient {
         onMessageError: (error) => {
           console.log('Message error:', error);
         },
+        onMicUpdated: (data) => {
+          console.log('Mic updated:', data);
+          this.deviceSelector.value = data.deviceId;
+        },
         onError: (error) => {
           console.log('Error:', JSON.stringify(error));
         },
       },
     });
+    window.client = this; // Expose client globally for debugging
 
     // Set up listeners for media track events
     this.setupTrackListeners();
+    this.setupEventListeners();
   }
 
   /**
@@ -163,10 +227,10 @@ class ChatbotClient {
    * This is called when the bot is ready or when the transport state changes to ready
    */
   setupMediaTracks() {
-    if (!this.rtviClient) return;
+    if (!this.pcClient) return;
 
     // Get current tracks from the client
-    const tracks = this.rtviClient.tracks();
+    const tracks = this.pcClient.tracks();
 
     // Set up any available bot tracks
     if (tracks.bot?.audio) {
@@ -182,27 +246,34 @@ class ChatbotClient {
    * This handles new tracks being added during the session
    */
   setupTrackListeners() {
-    if (!this.rtviClient) return;
+    if (!this.pcClient) return;
 
     // Listen for new tracks starting
-    this.rtviClient.on(RTVIEvent.TrackStarted, (track, participant) => {
+    this.pcClient.on(RTVIEvent.TrackStarted, (track, participant) => {
-      // Only handle non-local (bot) tracks
       if (!participant?.local) {
         if (track.kind === 'audio') {
           this.setupAudioTrack(track);
         } else if (track.kind === 'video') {
           this.setupVideoTrack(track);
         }
+      } else if (track.kind === 'audio') {
+        console.log(`Local audio track started: `, this.pcClient.tracks());
+        // If local audio track starts, update mic
+        this.updateMicToggleButton(true);
       }
     });
 
     // Listen for tracks stopping
-    this.rtviClient.on(RTVIEvent.TrackStopped, (track, participant) => {
+    this.pcClient.on(RTVIEvent.TrackStopped, (track, participant) => {
       this.log(
         `Track stopped event: ${track.kind} from ${
-          participant?.name || 'unknown'
+          participant ? (participant.local ? 'local' : 'bot') : 'unknown'
         }`
       );
+      if (participant?.local && track.kind === 'audio') {
+        // If local audio track stops, update mic toggle button
+        this.updateMicToggleButton(false);
+      }
     });
   }
 
@@ -251,17 +322,16 @@ class ChatbotClient {
 
   /**
    * Initialize and connect to the bot
-   * This sets up the RTVI client, initializes devices, and establishes the connection
+   * This sets up the Pipecat client, initializes devices, and establishes the connection
    */
   async connect() {
     try {
-      // Initialize audio/video devices
-      this.log('Initializing devices...');
-      await this.rtviClient.initDevices();
-
       // Connect to the bot
       this.log('Connecting to bot...');
-      await this.rtviClient.connect();
+      await this.pcClient.connect({
+        endpoint: 'http://localhost:7860/connect',
+        timeout: 25000,
+      });
 
       this.log('Connection complete');
     } catch (error) {
@@ -271,9 +341,9 @@ class ChatbotClient {
       this.updateStatus('Error');
 
       // Clean up if there's an error
-      if (this.rtviClient) {
+      if (this.pcClient) {
         try {
-          await this.rtviClient.disconnect();
+          await this.pcClient.disconnect();
         } catch (disconnectError) {
           this.log(`Error during disconnect: ${disconnectError.message}`);
         }
@@ -285,10 +355,10 @@ class ChatbotClient {
    * Disconnect from the bot and clean up media resources
    */
   async disconnect() {
-    if (this.rtviClient) {
+    if (this.pcClient) {
       try {
-        // Disconnect the RTVI client
+        // Disconnect the Pipecat client
-        await this.rtviClient.disconnect();
+        await this.pcClient.disconnect();
 
         // Clean up audio
         if (this.botAudio.srcObject) {

examples/simple-chatbot/client/javascript/src/style.css

@@ -10,7 +10,8 @@ body {
   margin: 0 auto;
 }
 
-.status-bar {
+.status-bar,
+.device-bar {
   display: flex;
   justify-content: space-between;
   align-items: center;
@@ -20,7 +21,24 @@ body {
   margin-bottom: 20px;
 }
 
-.controls button {
+.device-bar {
+  flex-direction: column;
+  gap: 10px;
+}
+
+.controls,
+.device-controls {
+  display: flex;
+  align-items: center;
+  gap: 10px; /* Adds spacing between elements */
+}
+
+.device-controls {
+  margin-left: auto;
+}
+
+.controls button,
+.device-controls button {
   padding: 8px 16px;
   margin-left: 10px;
   border: none;
@@ -28,6 +46,56 @@ body {
   cursor: pointer;
 }
 
+#bot-selector,
+#device-selector {
+  padding: 8px 16px;
+  padding-right: 40px;
+  border: none;
+  border-radius: 4px;
+  background-color: #6c757d; /* Gray background */
+  color: white; /* White text */
+  cursor: pointer;
+  appearance: none; /* Removes default browser styling for dropdowns */
+  background-image: url("data:image/svg+xml,%3Csvg xmlns='http://www.w3.org/2000/svg' viewBox='0 0 24 24' fill='white'%3E%3Cpath d='M7 10l5 5 5-5z'/%3E%3C/svg%3E"); /* Custom arrow */
+  background-repeat: no-repeat;
+  background-position: right 8px center; /* Position the arrow */
+}
+
+#bot-selector:focus,
+#device-selector:focus {
+  outline: none;
+  box-shadow: 0 0 4px rgba(0, 0, 0, 0.3); /* Add a subtle focus effect */
+}
+
+.text-input-container {
+  display: flex;
+  gap: 8px;
+  margin-left: 10px;
+  width: 100%;
+  flex: 1;
+}
+
+#text-input {
+  flex: 1;
+  padding: 8px 16px;
+  border: 1px solid #e0e0e0;
+  border-radius: 4px;
+  min-width: 200px;
+}
+
+#send-text-btn {
+  padding: 8px 16px;
+  border: none;
+  border-radius: 4px;
+  background-color: #007bff;
+  color: white;
+  flex-shrink: 0;
+}
+
+#send-text-btn:hover {
+  background-color: #0056b3;
+}
+
 #connect-btn {
   background-color: #4caf50;
   color: white;
@@ -38,6 +106,9 @@ body {
   color: white;
 }
 
+#mic-toggle-btn {
+}
+
 button:disabled {
   opacity: 0.5;
   cursor: not-allowed;

examples/simple-chatbot/client/react/package.json

@@ -10,9 +10,9 @@
     "preview": "vite preview"
   },
   "dependencies": {
-    "@pipecat-ai/client-js": "^0.3.5",
+    "@pipecat-ai/client-js": "^1.0.0",
-    "@pipecat-ai/client-react": "^0.3.5",
+    "@pipecat-ai/client-react": "^1.0.0",
-    "@pipecat-ai/daily-transport": "^0.3.8",
+    "@pipecat-ai/daily-transport": "^1.0.0",
     "react": "^18.3.1",
     "react-dom": "^18.3.1"
   },

examples/simple-chatbot/client/react/src/App.tsx

@@ -1,22 +1,22 @@
 import {
-  RTVIClientAudio,
+  PipecatClientAudio,
-  RTVIClientVideo,
+  PipecatClientVideo,
-  useRTVIClientTransportState,
+  usePipecatClientTransportState,
 } from '@pipecat-ai/client-react';
-import { RTVIProvider } from './providers/RTVIProvider';
+import { PipecatProvider } from './providers/PipecatProvider';
 import { ConnectButton } from './components/ConnectButton';
 import { StatusDisplay } from './components/StatusDisplay';
 import { DebugDisplay } from './components/DebugDisplay';
 import './App.css';
 
 function BotVideo() {
-  const transportState = useRTVIClientTransportState();
+  const transportState = usePipecatClientTransportState();
   const isConnected = transportState !== 'disconnected';
 
   return (
     <div className="bot-container">
       <div className="video-container">
-        {isConnected && <RTVIClientVideo participant="bot" fit="cover" />}
+        {isConnected && <PipecatClientVideo participant="bot" fit="cover" />}
       </div>
     </div>
   );
@@ -35,16 +35,16 @@ function AppContent() {
       </div>
 
       <DebugDisplay />
-      <RTVIClientAudio />
+      <PipecatClientAudio />
     </div>
   );
 }
 
 function App() {
   return (
-    <RTVIProvider>
+    <PipecatProvider>
       <AppContent />
-    </RTVIProvider>
+    </PipecatProvider>
   );
 }
 

examples/simple-chatbot/client/react/src/components/ConnectButton.tsx

@@ -1,16 +1,16 @@
 import {
-  useRTVIClient,
+  usePipecatClient,
-  useRTVIClientTransportState,
+  usePipecatClientTransportState,
 } from '@pipecat-ai/client-react';
 
 export function ConnectButton() {
-  const client = useRTVIClient();
+  const client = usePipecatClient();
-  const transportState = useRTVIClientTransportState();
+  const transportState = usePipecatClientTransportState();
   const isConnected = ['connected', 'ready'].includes(transportState);
 
   const handleClick = async () => {
     if (!client) {
-      console.error('RTVI client is not initialized');
+      console.error('Pipecat client is not initialized');
       return;
     }
 
@@ -18,7 +18,7 @@ export function ConnectButton() {
       if (isConnected) {
         await client.disconnect();
       } else {
-        await client.connect();
+        await client.connect({ endpoint: 'http://localhost:7860/connect' });
       }
     } catch (error) {
       console.error('Connection error:', error);

examples/simple-chatbot/client/react/src/components/DebugDisplay.tsx

@@ -6,12 +6,12 @@ import {
   TranscriptData,
   BotLLMTextData,
 } from '@pipecat-ai/client-js';
-import { useRTVIClient, useRTVIClientEvent } from '@pipecat-ai/client-react';
+import { usePipecatClient, useRTVIClientEvent } from '@pipecat-ai/client-react';
 import './DebugDisplay.css';
 
 export function DebugDisplay() {
   const debugLogRef = useRef<HTMLDivElement>(null);
-  const client = useRTVIClient();
+  const client = usePipecatClient();
 
   const log = useCallback((message: string) => {
     if (!debugLogRef.current) return;

examples/simple-chatbot/client/react/src/components/StatusDisplay.tsx

@@ -1,7 +1,7 @@
-import { useRTVIClientTransportState } from '@pipecat-ai/client-react';
+import { usePipecatClientTransportState } from '@pipecat-ai/client-react';
 
 export function StatusDisplay() {
-  const transportState = useRTVIClientTransportState();
+  const transportState = usePipecatClientTransportState();
 
   return (
     <div className="status">

examples/simple-chatbot/client/react/src/providers/PipecatProvider.tsx

@@ -0,0 +1,16 @@
+import { type PropsWithChildren } from 'react';
+import { PipecatClient } from '@pipecat-ai/client-js';
+import { DailyTransport } from '@pipecat-ai/daily-transport';
+import { PipecatClientProvider } from '@pipecat-ai/client-react';
+
+const client = new PipecatClient({
+  transport: new DailyTransport(),
+  enableMic: true,
+  enableCam: false,
+});
+
+export function PipecatProvider({ children }: PropsWithChildren) {
+  return (
+    <PipecatClientProvider client={client}>{children}</PipecatClientProvider>
+  );
+}

examples/simple-chatbot/client/react/src/providers/RTVIProvider.tsx

@@ -1,22 +0,0 @@
-import { type PropsWithChildren } from 'react';
-import { RTVIClient } from '@pipecat-ai/client-js';
-import { DailyTransport } from '@pipecat-ai/daily-transport';
-import { RTVIClientProvider } from '@pipecat-ai/client-react';
-
-const transport = new DailyTransport();
-
-const client = new RTVIClient({
-  transport,
-  params: {
-    baseUrl: 'http://localhost:7860',
-    endpoints: {
-      connect: '/connect',
-    },
-  },
-  enableMic: true,
-  enableCam: false,
-});
-
-export function RTVIProvider({ children }: PropsWithChildren) {
-  return <RTVIClientProvider client={client}>{children}</RTVIClientProvider>;
-}

examples/twilio-chatbot/client/typescript/package.json

@@ -19,7 +19,7 @@
     "vite": "^6.0.2"
   },
   "dependencies": {
-    "@pipecat-ai/client-js": "^0.4.0",
+    "@pipecat-ai/client-js": "^1.0.0",
-    "@pipecat-ai/websocket-transport": "^0.4.2"
+    "@pipecat-ai/websocket-transport": "^1.0.0"
   }
 }

examples/twilio-chatbot/client/typescript/src/app.ts

@@ -5,21 +5,22 @@
  */
 
 import {
-  RTVIClient,
+  BotLLMTextData,
-  RTVIClientOptions,
+  Participant,
-  RTVIEvent,
+  PipecatClient,
+  PipecatClientOptions,
+  RTVIEvent, RTVIMessage, TranscriptData,
 } from '@pipecat-ai/client-js';
 import {
   WebSocketTransport,
   TwilioSerializer,
-} from "@pipecat-ai/websocket-transport";
+} from '@pipecat-ai/websocket-transport';
 
 class WebsocketClientApp {
+  private static STREAM_SID = 'ws_mock_stream_sid';
+  private static CALL_SID = 'ws_mock_call_sid';
 
-  private static STREAM_SID = "ws_mock_stream_sid"
+  private rtviClient: PipecatClient | null = null;
-  private static CALL_SID = "ws_mock_call_sid"
-
-  private rtviClient: RTVIClient | null = null;
   private connectBtn: HTMLButtonElement | null = null;
   private disconnectBtn: HTMLButtonElement | null = null;
   private statusSpan: HTMLElement | null = null;
@@ -38,8 +39,12 @@ class WebsocketClientApp {
    * Set up references to DOM elements and create necessary media elements
    */
   private setupDOMElements(): void {
-    this.connectBtn = document.getElementById('connect-btn') as HTMLButtonElement;
+    this.connectBtn = document.getElementById(
-    this.disconnectBtn = document.getElementById('disconnect-btn') as HTMLButtonElement;
+      'connect-btn'
+    ) as HTMLButtonElement;
+    this.disconnectBtn = document.getElementById(
+      'disconnect-btn'
+    ) as HTMLButtonElement;
     this.statusSpan = document.getElementById('connection-status');
     this.debugLog = document.getElementById('debug-log');
   }
@@ -80,13 +85,23 @@ class WebsocketClientApp {
   }
 
   private async emulateTwilioMessages() {
-    const connectedMessage={"event": "connected", "protocol": "Call", "version": "1.0.0"}
+    const connectedMessage = {
+      event: 'connected',
+      protocol: 'Call',
+      version: '1.0.0',
+    };
 
-    const websocketTransport = this.rtviClient?.transport as WebSocketTransport
+    const websocketTransport = this.rtviClient?.transport as WebSocketTransport;
-    void websocketTransport?.sendRawMessage(connectedMessage)
+    void websocketTransport?.sendRawMessage(connectedMessage);
 
-    const startMessage={"event": "start", "start": {"streamSid": WebsocketClientApp.STREAM_SID, "callSid": WebsocketClientApp.CALL_SID}}
+    const startMessage = {
-    void websocketTransport?.sendRawMessage(startMessage)
+      event: 'start',
+      start: {
+        streamSid: WebsocketClientApp.STREAM_SID,
+        callSid: WebsocketClientApp.CALL_SID,
+      },
+    };
+    void websocketTransport?.sendRawMessage(startMessage);
   }
 
   /**
@@ -109,7 +124,7 @@ class WebsocketClientApp {
     if (!this.rtviClient) return;
 
     // Listen for new tracks starting
-    this.rtviClient.on(RTVIEvent.TrackStarted, (track, participant) => {
+    this.rtviClient.on(RTVIEvent.TrackStarted, (track: MediaStreamTrack, participant?: Participant) => {
       // Only handle non-local (bot) tracks
       if (!participant?.local && track.kind === 'audio') {
         this.setupAudioTrack(track);
@@ -117,8 +132,10 @@ class WebsocketClientApp {
     });
 
     // Listen for tracks stopping
-    this.rtviClient.on(RTVIEvent.TrackStopped, (track, participant) => {
+    this.rtviClient.on(RTVIEvent.TrackStopped, (track: MediaStreamTrack, participant?: Participant) => {
-      this.log(`Track stopped: ${track.kind} from ${participant?.name || 'unknown'}`);
+      this.log(
+        `Track stopped: ${track.kind} from ${participant?.name || 'unknown'}`
+      );
     });
   }
 
@@ -128,7 +145,10 @@ class WebsocketClientApp {
    */
   private setupAudioTrack(track: MediaStreamTrack): void {
     this.log('Setting up audio track');
-    if (this.botAudio.srcObject && "getAudioTracks" in this.botAudio.srcObject) {
+    if (
+      this.botAudio.srcObject &&
+      'getAudioTracks' in this.botAudio.srcObject
+    ) {
       const oldTrack = this.botAudio.srcObject.getAudioTracks()[0];
       if (oldTrack?.id === track.id) return;
     }
@@ -143,23 +163,19 @@ class WebsocketClientApp {
     try {
       const startTime = Date.now();
 
-      const transport = new WebSocketTransport({
+      const ws_opts = {
         serializer: new TwilioSerializer(),
         recorderSampleRate: 8000,
-        playerSampleRate: 8000
+        playerSampleRate: 8000,
-      });
+        ws_url: 'http://localhost:8765/ws',
-      const RTVIConfig: RTVIClientOptions = {
+      };
-        transport,
+      const RTVIConfig: PipecatClientOptions = {
-        params: {
+        transport: new WebSocketTransport(ws_opts),
-          // The baseURL and endpoint of your bot server that the client will connect to
-          baseUrl: 'http://localhost:8765',
-          endpoints: { connect: '/' },
-        },
         enableMic: true,
         enableCam: false,
         callbacks: {
           onConnected: () => {
-            this.emulateTwilioMessages()
+            this.emulateTwilioMessages();
             this.updateStatus('Connected');
             if (this.connectBtn) this.connectBtn.disabled = true;
             if (this.disconnectBtn) this.disconnectBtn.disabled = false;
@@ -170,27 +186,21 @@ class WebsocketClientApp {
             if (this.disconnectBtn) this.disconnectBtn.disabled = true;
             this.log('Client disconnected');
           },
-          onBotReady: (data) => {
+          onBotReady: (data: any) => {
             this.log(`Bot ready: ${JSON.stringify(data)}`);
             this.setupMediaTracks();
           },
-          onUserTranscript: (data) => {
+          onUserTranscript: (data: TranscriptData) => {
             if (data.final) {
               this.log(`User: ${data.text}`);
             }
           },
-          onBotTranscript: (data) => this.log(`Bot: ${data.text}`),
+          onBotTranscript: (data: BotLLMTextData) => this.log(`Bot: ${data.text}`),
-          onMessageError: (error) => console.error('Message error:', error),
+          onMessageError: (error: RTVIMessage) => console.error('Message error:', error),
-          onError: (error) => console.error('Error:', error),
+          onError: (error: RTVIMessage) => console.error('Error:', error),
         },
-      }
+      };
-      // @ts-ignore
+      this.rtviClient = new PipecatClient(RTVIConfig);
-      RTVIConfig.customConnectHandler = () => Promise.resolve(
-          {
-              ws_url: "/ws",
-          }
-      );
-      this.rtviClient = new RTVIClient(RTVIConfig);
       this.setupTrackListeners();
 
       this.log('Initializing devices...');
@@ -223,8 +233,13 @@ class WebsocketClientApp {
       try {
         await this.rtviClient.disconnect();
         this.rtviClient = null;
-        if (this.botAudio.srcObject && "getAudioTracks" in this.botAudio.srcObject) {
+        if (
-          this.botAudio.srcObject.getAudioTracks().forEach((track) => track.stop());
+          this.botAudio.srcObject &&
+          'getAudioTracks' in this.botAudio.srcObject
+        ) {
+          this.botAudio.srcObject
+            .getAudioTracks()
+            .forEach((track) => track.stop());
           this.botAudio.srcObject = null;
         }
       } catch (error) {
@@ -232,7 +247,6 @@ class WebsocketClientApp {
       }
     }
   }
-
 }
 
 declare global {

examples/websocket/client/package.json

@@ -19,8 +19,8 @@
     "vite": "^6.3.5"
   },
   "dependencies": {
-    "@pipecat-ai/client-js": "^0.4.0",
+    "@pipecat-ai/client-js": "^1.0.0",
-    "@pipecat-ai/websocket-transport": "^0.4.2",
+    "@pipecat-ai/websocket-transport": "^1.0.0",
     "protobufjs": "^7.4.0"
   }
 }

examples/websocket/client/src/app.ts

@@ -5,7 +5,7 @@
  */
 
 /**
- * RTVI Client Implementation
+ * Pipecat Client Implementation
  *
  * This client connects to an RTVI-compatible bot server using WebSocket.
  *
@@ -14,16 +14,14 @@
  */
 
 import {
-  RTVIClient,
+  PipecatClient,
-  RTVIClientOptions,
+  PipecatClientOptions,
   RTVIEvent,
 } from '@pipecat-ai/client-js';
-import {
+import { WebSocketTransport } from '@pipecat-ai/websocket-transport';
-  WebSocketTransport
-} from "@pipecat-ai/websocket-transport";
 
 class WebsocketClientApp {
-  private rtviClient: RTVIClient | null = null;
+  private pcClient: PipecatClient | null = null;
   private connectBtn: HTMLButtonElement | null = null;
   private disconnectBtn: HTMLButtonElement | null = null;
   private statusSpan: HTMLElement | null = null;
@@ -31,7 +29,7 @@ class WebsocketClientApp {
   private botAudio: HTMLAudioElement;
 
   constructor() {
-    console.log("WebsocketClientApp");
+    console.log('WebsocketClientApp');
     this.botAudio = document.createElement('audio');
     this.botAudio.autoplay = true;
     //this.botAudio.playsInline = true;
@@ -45,8 +43,12 @@ class WebsocketClientApp {
    * Set up references to DOM elements and create necessary media elements
    */
   private setupDOMElements(): void {
-    this.connectBtn = document.getElementById('connect-btn') as HTMLButtonElement;
+    this.connectBtn = document.getElementById(
-    this.disconnectBtn = document.getElementById('disconnect-btn') as HTMLButtonElement;
+      'connect-btn'
+    ) as HTMLButtonElement;
+    this.disconnectBtn = document.getElementById(
+      'disconnect-btn'
+    ) as HTMLButtonElement;
     this.statusSpan = document.getElementById('connection-status');
     this.debugLog = document.getElementById('debug-log');
   }
@@ -91,8 +93,8 @@ class WebsocketClientApp {
    * This is called when the bot is ready or when the transport state changes to ready
    */
   setupMediaTracks() {
-    if (!this.rtviClient) return;
+    if (!this.pcClient) return;
-    const tracks = this.rtviClient.tracks();
+    const tracks = this.pcClient.tracks();
     if (tracks.bot?.audio) {
       this.setupAudioTrack(tracks.bot.audio);
     }
@@ -103,10 +105,10 @@ class WebsocketClientApp {
    * This handles new tracks being added during the session
    */
   setupTrackListeners() {
-    if (!this.rtviClient) return;
+    if (!this.pcClient) return;
 
     // Listen for new tracks starting
-    this.rtviClient.on(RTVIEvent.TrackStarted, (track, participant) => {
+    this.pcClient.on(RTVIEvent.TrackStarted, (track, participant) => {
       // Only handle non-local (bot) tracks
       if (!participant?.local && track.kind === 'audio') {
         this.setupAudioTrack(track);
@@ -114,8 +116,10 @@ class WebsocketClientApp {
     });
 
     // Listen for tracks stopping
-    this.rtviClient.on(RTVIEvent.TrackStopped, (track, participant) => {
+    this.pcClient.on(RTVIEvent.TrackStopped, (track, participant) => {
-      this.log(`Track stopped: ${track.kind} from ${participant?.name || 'unknown'}`);
+      this.log(
+        `Track stopped: ${track.kind} from ${participant?.name || 'unknown'}`
+      );
     });
   }
 
@@ -125,7 +129,10 @@ class WebsocketClientApp {
    */
   private setupAudioTrack(track: MediaStreamTrack): void {
     this.log('Setting up audio track');
-    if (this.botAudio.srcObject && "getAudioTracks" in this.botAudio.srcObject) {
+    if (
+      this.botAudio.srcObject &&
+      'getAudioTracks' in this.botAudio.srcObject
+    ) {
       const oldTrack = this.botAudio.srcObject.getAudioTracks()[0];
       if (oldTrack?.id === track.id) return;
     }
@@ -134,21 +141,15 @@ class WebsocketClientApp {
 
   /**
    * Initialize and connect to the bot
-   * This sets up the RTVI client, initializes devices, and establishes the connection
+   * This sets up the Pipecat client, initializes devices, and establishes the connection
    */
   public async connect(): Promise<void> {
     try {
       const startTime = Date.now();
 
       //const transport = new DailyTransport();
-      const transport = new WebSocketTransport();
+      const PipecatConfig: PipecatClientOptions = {
-      const RTVIConfig: RTVIClientOptions = {
+        transport: new WebSocketTransport(),
-        transport,
-        params: {
-          // The baseURL and endpoint of your bot server that the client will connect to
-          baseUrl: 'http://localhost:7860',
-          endpoints: { connect: '/connect' },
-        },
         enableMic: true,
         enableCam: false,
         callbacks: {
@@ -176,15 +177,20 @@ class WebsocketClientApp {
           onMessageError: (error) => console.error('Message error:', error),
           onError: (error) => console.error('Error:', error),
         },
-      }
+      };
-      this.rtviClient = new RTVIClient(RTVIConfig);
+      this.pcClient = new PipecatClient(PipecatConfig);
+      // @ts-ignore
+      window.pcClient = this.pcClient; // Expose for debugging
       this.setupTrackListeners();
 
       this.log('Initializing devices...');
-      await this.rtviClient.initDevices();
+      await this.pcClient.initDevices();
 
       this.log('Connecting to bot...');
-      await this.rtviClient.connect();
+      await this.pcClient.connect({
+        // The baseURL and endpoint of your bot server that the client will connect to
+        endpoint: 'http://localhost:7860/connect',
+      });
 
       const timeTaken = Date.now() - startTime;
       this.log(`Connection complete, timeTaken: ${timeTaken}`);
@@ -192,9 +198,9 @@ class WebsocketClientApp {
       this.log(`Error connecting: ${(error as Error).message}`);
       this.updateStatus('Error');
       // Clean up if there's an error
-      if (this.rtviClient) {
+      if (this.pcClient) {
         try {
-          await this.rtviClient.disconnect();
+          await this.pcClient.disconnect();
         } catch (disconnectError) {
           this.log(`Error during disconnect: ${disconnectError}`);
         }
@@ -206,12 +212,17 @@ class WebsocketClientApp {
    * Disconnect from the bot and clean up media resources
    */
   public async disconnect(): Promise<void> {
-    if (this.rtviClient) {
+    if (this.pcClient) {
       try {
-        await this.rtviClient.disconnect();
+        await this.pcClient.disconnect();
-        this.rtviClient = null;
+        this.pcClient = null;
-        if (this.botAudio.srcObject && "getAudioTracks" in this.botAudio.srcObject) {
+        if (
-          this.botAudio.srcObject.getAudioTracks().forEach((track) => track.stop());
+          this.botAudio.srcObject &&
+          'getAudioTracks' in this.botAudio.srcObject
+        ) {
+          this.botAudio.srcObject
+            .getAudioTracks()
+            .forEach((track) => track.stop());
           this.botAudio.srcObject = null;
         }
       } catch (error) {
@@ -219,7 +230,6 @@ class WebsocketClientApp {
       }
     }
   }
-
 }
 
 declare global {

examples/word-wrangler-gemini-live/client/package.json

@@ -9,11 +9,12 @@
     "lint": "next lint"
   },
   "dependencies": {
-    "@pipecat-ai/client-js": "^0.3.5",
+    "@pipecat-ai/client-js": "^1.0.0",
-    "@pipecat-ai/client-react": "^0.3.5",
+    "@pipecat-ai/client-react": "^1.0.0",
-    "@pipecat-ai/daily-transport": "^0.3.10",
+    "@pipecat-ai/daily-transport": "^1.0.0",
     "@tabler/icons-react": "^3.31.0",
     "@tailwindcss/postcss": "^4.1.3",
+    "jotai": "^2.12.5",
     "js-confetti": "^0.12.0",
     "next": "15.2.4",
     "react": "^19.0.0",

examples/word-wrangler-gemini-live/client/src/hooks/useConnectionState.ts

@@ -1,16 +1,26 @@
 import { useEffect, useCallback } from 'react';
 import {
-  useRTVIClient,
+  usePipecatClient,
-  useRTVIClientTransportState,
+  usePipecatClientTransportState,
 } from '@pipecat-ai/client-react';
 import { CONNECTION_STATES } from '@/constants/gameConstants';
+import { useConfigurationSettings } from '@/contexts/Configuration';
+
+// Get the API base URL from environment variables
+// Default to "/api" if not specified
+// "/api" is the default for Next.js API routes and used
+// for the Pipecat Cloud deployed agent
+const API_BASE_URL = process.env.NEXT_PUBLIC_API_BASE_URL || '/api';
+
+console.log('Using API base URL:', API_BASE_URL);
 
 export function useConnectionState(
   onConnected?: () => void,
   onDisconnected?: () => void
 ) {
-  const client = useRTVIClient();
+  const client = usePipecatClient();
-  const transportState = useRTVIClientTransportState();
+  const transportState = usePipecatClientTransportState();
+  const config = useConfigurationSettings();
 
   const isConnected = CONNECTION_STATES.ACTIVE.includes(transportState);
   const isConnecting = CONNECTION_STATES.CONNECTING.includes(transportState);
@@ -35,12 +45,17 @@ export function useConnectionState(
       if (isConnected) {
         await client.disconnect();
       } else {
-        await client.connect();
+        await client.connect({
+          endpoint: `${API_BASE_URL}/connect`,
+          requestData: {
+            personality: config.personality,
+          },
+        });
       }
     } catch (error) {
       console.error('Connection error:', error);
     }
-  }, [client, isConnected]);
+  }, [client, config, isConnected]);
 
   return {
     isConnected,

examples/word-wrangler-gemini-live/client/src/pages/_app.tsx

@@ -1,15 +1,15 @@
-import { ConfigurationProvider } from "@/contexts/Configuration";
+import { ConfigurationProvider } from '@/contexts/Configuration';
-import { RTVIProvider } from "@/providers/RTVIProvider";
+import { PipecatProvider } from '@/providers/PipecatProvider';
-import { RTVIClientAudio } from "@pipecat-ai/client-react";
+import { PipecatClientAudio } from '@pipecat-ai/client-react';
-import type { AppProps } from "next/app";
+import type { AppProps } from 'next/app';
-import { Nunito } from "next/font/google";
+import { Nunito } from 'next/font/google';
-import Head from "next/head";
+import Head from 'next/head';
-import "../styles/globals.css";
+import '../styles/globals.css';
 
 const nunito = Nunito({
-  subsets: ["latin"],
+  subsets: ['latin'],
-  display: "swap",
+  display: 'swap',
-  variable: "--font-sans",
+  variable: '--font-sans',
 });
 
 export default function App({ Component, pageProps }: AppProps) {
@@ -21,10 +21,10 @@ export default function App({ Component, pageProps }: AppProps) {
       </Head>
       <main className={`${nunito.variable}`}>
         <ConfigurationProvider>
-          <RTVIProvider>
+          <PipecatProvider>
-            <RTVIClientAudio />
+            <PipecatClientAudio />
             <Component {...pageProps} />
-          </RTVIProvider>
+          </PipecatProvider>
         </ConfigurationProvider>
       </main>
     </>

examples/word-wrangler-gemini-live/client/src/pages/api/connect.ts

@@ -1,11 +1,11 @@
-import type { NextApiRequest, NextApiResponse } from "next";
+import type { NextApiRequest, NextApiResponse } from 'next';
 
 export default async function handler(
   req: NextApiRequest,
   res: NextApiResponse
 ) {
-  if (req.method !== "POST") {
+  if (req.method !== 'POST') {
-    return res.status(405).json({ error: "Method not allowed" });
+    return res.status(405).json({ error: 'Method not allowed' });
   }
 
   try {
@@ -15,16 +15,16 @@ export default async function handler(
     if (!personality) {
       return res
         .status(400)
-        .json({ error: "Missing required configuration parameters" });
+        .json({ error: 'Missing required configuration parameters' });
     }
 
     const response = await fetch(
       `https://api.pipecat.daily.co/v1/public/${process.env.AGENT_NAME}/start`,
       {
-        method: "POST",
+        method: 'POST',
         headers: {
           Authorization: `Bearer ${process.env.PIPECAT_CLOUD_API_KEY}`,
-          "Content-Type": "application/json",
+          'Content-Type': 'application/json',
         },
         body: JSON.stringify({
           createDailyRoom: true,
@@ -37,15 +37,15 @@ export default async function handler(
 
     const data = await response.json();
 
-    console.log("Response from API:", JSON.stringify(data, null, 2));
+    console.log('Response from API:', JSON.stringify(data, null, 2));
 
-    // Transform the response to match what RTVI client expects
+    // Transform the response to match what Pipecat client expects
     return res.status(200).json({
       room_url: data.dailyRoom,
       token: data.dailyToken,
     });
   } catch (error) {
-    console.error("Error starting agent:", error);
+    console.error('Error starting agent:', error);
-    return res.status(500).json({ error: "Failed to start agent" });
+    return res.status(500).json({ error: 'Failed to start agent' });
   }
 }

examples/word-wrangler-gemini-live/client/src/providers/PipecatProvider.tsx

@@ -0,0 +1,43 @@
+'use client';
+
+import { PipecatClient } from '@pipecat-ai/client-js';
+import { DailyTransport } from '@pipecat-ai/daily-transport';
+import { PipecatClientProvider } from '@pipecat-ai/client-react';
+import { PropsWithChildren, useEffect, useState, useRef } from 'react';
+
+export function PipecatProvider({ children }: PropsWithChildren) {
+  const [client, setClient] = useState<PipecatClient | null>(null);
+  const clientCreated = useRef(false);
+
+  useEffect(() => {
+    // Only create the client once
+    if (clientCreated.current) return;
+
+    const pcClient = new PipecatClient({
+      transport: new DailyTransport(),
+      enableMic: true,
+      enableCam: false,
+    });
+
+    setClient(pcClient);
+    clientCreated.current = true;
+
+    // Cleanup when component unmounts
+    return () => {
+      if (pcClient) {
+        pcClient.disconnect().catch((err) => {
+          console.error('Error disconnecting client:', err);
+        });
+      }
+      clientCreated.current = false;
+    };
+  }, []);
+
+  if (!client) {
+    return null;
+  }
+
+  return (
+    <PipecatClientProvider client={client}>{children}</PipecatClientProvider>
+  );
+}

examples/word-wrangler-gemini-live/client/src/providers/RTVIProvider.tsx

@@ -1,72 +0,0 @@
-"use client";
-
-import { RTVIClient } from "@pipecat-ai/client-js";
-import { DailyTransport } from "@pipecat-ai/daily-transport";
-import { RTVIClientProvider } from "@pipecat-ai/client-react";
-import { PropsWithChildren, useEffect, useState, useRef } from "react";
-import { useConfigurationSettings } from "@/contexts/Configuration";
-
-// Get the API base URL from environment variables
-// Default to "/api" if not specified
-// "/api" is the default for Next.js API routes and used
-// for the Pipecat Cloud deployed agent
-const API_BASE_URL = process.env.NEXT_PUBLIC_API_BASE_URL || "/api";
-
-console.log("Using API base URL:", API_BASE_URL);
-
-export function RTVIProvider({ children }: PropsWithChildren) {
-  const [client, setClient] = useState<RTVIClient | null>(null);
-  const config = useConfigurationSettings();
-  const clientCreated = useRef(false);
-
-  useEffect(() => {
-    // Only create the client once
-    if (clientCreated.current) return;
-
-    const transport = new DailyTransport();
-
-    const rtviClient = new RTVIClient({
-      transport,
-      params: {
-        baseUrl: API_BASE_URL,
-        endpoints: {
-          connect: "/connect",
-        },
-        requestData: {
-          personality: config.personality,
-        },
-      },
-      enableMic: true,
-      enableCam: false,
-    });
-
-    setClient(rtviClient);
-    clientCreated.current = true;
-
-    // Cleanup when component unmounts
-    return () => {
-      if (rtviClient) {
-        rtviClient.disconnect().catch((err) => {
-          console.error("Error disconnecting client:", err);
-        });
-      }
-      clientCreated.current = false;
-    };
-  }, []);
-
-  // Update the connectParams when config changes
-  useEffect(() => {
-    if (!client) return;
-
-    // Update the connect params without recreating the client
-    client.params.requestData = {
-      personality: config.personality,
-    };
-  }, [client, config.personality]);
-
-  if (!client) {
-    return null;
-  }
-
-  return <RTVIClientProvider client={client}>{children}</RTVIClientProvider>;
-}

examples/word-wrangler-gemini-live/server/bot.py

@@ -4,6 +4,7 @@
 # SPDX-License-Identifier: BSD 2-Clause License
 #
 
+import argparse
 import asyncio
 import os
 import sys
@@ -198,16 +199,15 @@ async def bot(args: DailySessionArguments):
 
 
 # Local development
-async def local_daily():
+async def local_daily(args: DailySessionArguments):
     """Daily transport for local development."""
-    from runner import configure
+    # from runner import configure
 
     try:
         async with aiohttp.ClientSession() as session:
-            (room_url, token) = await configure(session)
             transport = DailyTransport(
-                room_url,
+                room_url=args.room_url,
-                token,
+                token=args.token,
                 bot_name="Bot",
                 params=DailyParams(
                     audio_in_enabled=True,
@@ -217,7 +217,7 @@ async def local_daily():
             )
 
             test_config = {
-                "personality": "witty",
+                "personality": args.personality,
             }
 
             await main(transport, test_config)
@@ -227,7 +227,24 @@ async def local_daily():
 
 # Local development entry point
 if LOCAL_RUN and __name__ == "__main__":
+    parser = argparse.ArgumentParser(
+        description="Run the Word Wrangler bot in local development mode"
+    )
+    parser.add_argument(
+        "-u", "--room-url", type=str, default=os.getenv("DAILY_SAMPLE_ROOM_URL", "")
+    )
+    parser.add_argument(
+        "-t", "--token", type=str, default=os.getenv("DAILY_SAMPLE_ROOM_TOKEN", None)
+    )
+    parser.add_argument(
+        "-p",
+        "--personality",
+        default="witty",
+        choices=["friendly", "professional", "enthusiastic", "thoughtful", "witty"],
+        help="Personality preset for the bot (friendly, professional, enthusiastic, thoughtful, witty)",
+    )
+    args = parser.parse_args()
     try:
-        asyncio.run(local_daily())
+        asyncio.run(local_daily(args))
     except Exception as e:
         logger.exception(f"Failed to run in local mode: {e}")

examples/word-wrangler-gemini-live/server/server.py

@@ -160,14 +160,15 @@ async def rtvi_connect(request: Request) -> Dict[Any, Any]:
     Raises:
         HTTPException: If room creation, token generation, or bot startup fails
     """
-    print("Creating room for RTVI connection")
+    body = await request.json()
+    print("Creating room for RTVI connection", body)
     room_url, token = await create_room_and_token()
     print(f"Room URL: {room_url}")
 
     # Start the bot process
     try:
         proc = subprocess.Popen(
-            [f"python3 -m bot -u {room_url} -t {token}"],
+            [f"python3 -m bot -u {room_url} -t {token} -p {body.get('personality', 'witty')}"],
             shell=True,
             bufsize=1,
             cwd=os.path.dirname(os.path.abspath(__file__)),

pyproject.toml

@@ -25,14 +25,14 @@ dependencies = [
     "docstring_parser~=0.16",
     "loguru~=0.7.3",
     "Markdown~=3.7",
-    "numpy~=1.26.4",
+    "numpy>=1.26.4",
     "Pillow~=11.1.0",
     "protobuf~=5.29.3",
     "pydantic~=2.10.6",
     "pyloudnorm~=0.1.1",
     "resampy~=0.4.3",
     "soxr~=0.5.0",
-    "openai~=1.70.0",
+    "openai~=1.74.0",
 ]
 
 [project.urls]
@@ -42,7 +42,7 @@ Website = "https://pipecat.ai"
 [project.optional-dependencies]
 anthropic = [ "anthropic~=0.49.0" ]
 assemblyai = [ "websockets~=13.1" ]
-aws = [ "boto3~=1.37.16", "websockets~=13.1" ]
+aws = [ "aioboto3~=15.0.0", "websockets~=13.1" ]
 aws-nova-sonic = [ "aws_sdk_bedrock_runtime~=0.0.2" ]
 azure = [ "azure-cognitiveservices-speech~=1.42.0"]
 cartesia = [ "cartesia~=2.0.3", "websockets~=13.1" ]
@@ -68,7 +68,7 @@ local = [ "pyaudio~=0.2.14" ]
 mcp = [ "mcp[cli]~=1.9.4" ]
 mem0 = [ "mem0ai~=0.1.94" ]
 mlx-whisper = [ "mlx-whisper~=0.4.2" ]
-moondream = [ "einops~=0.8.0", "timm~=1.0.13", "transformers~=4.48.0" ]
+moondream = [ "einops~=0.8.0", "timm~=1.0.13", "transformers>=4.48.0" ]
 nim = []
 neuphonic = [ "pyneuphonic~=1.5.13", "websockets~=13.1" ]
 noisereduce = [ "noisereduce~=3.0.3" ]
@@ -79,19 +79,20 @@ perplexity = []
 playht = [ "pyht~=0.1.12", "websockets~=13.1" ]
 qwen = []
 rime = [ "websockets~=13.1" ]
-riva = [ "nvidia-riva-client~=2.19.1" ]
+riva = [ "nvidia-riva-client~=2.21.1" ]
 sambanova = []
 sentry = [ "sentry-sdk~=2.23.1" ]
 local-smart-turn = [ "coremltools>=8.0", "transformers", "torch==2.5.0", "torchaudio==2.5.0" ]
 remote-smart-turn = []
 silero = [ "onnxruntime~=1.20.1" ]
 simli = [ "simli-ai~=0.1.10"]
+soniox = [ "websockets~=13.1" ]
 soundfile = [ "soundfile~=0.13.0" ]
 speechmatics = [ "speechmatics-rt>=0.3.1" ]
 tavus=[]
 together = []
 tracing = [ "opentelemetry-sdk>=1.33.0", "opentelemetry-api>=1.33.0", "opentelemetry-instrumentation>=0.54b0" ]
-ultravox = [ "transformers~=4.48.0", "vllm~=0.7.3" ]
+ultravox = [ "transformers>=4.48.0", "vllm~=0.7.3" ]
 webrtc = [ "aiortc~=1.11.0", "opencv-python~=4.11.0.86" ]
 websocket = [ "websockets~=13.1", "fastapi~=0.115.6" ]
 whisper = [ "faster-whisper~=1.1.1" ]

scripts/fix-ruff.sh

@@ -1,5 +1,10 @@
-ruff format src
+
-ruff format examples
+#!/bin/bash
-ruff format tests
+
-ruff format scripts
+SCRIPT_DIR="$(cd "$(dirname "${BASH_SOURCE[0]}")" && pwd)"
-ruff check --select I,D --fix
+PROJECT_ROOT="$(dirname "$SCRIPT_DIR")"
+
+echo "Running ruff format..."
+ruff format "$PROJECT_ROOT"
+echo "Running ruff check..."
+ruff check --fix "$PROJECT_ROOT"

src/pipecat/audio/turn/base_turn_analyzer.py

@@ -76,6 +76,16 @@ class BaseTurnAnalyzer(ABC):
         """
         pass
 
+    @property
+    @abstractmethod
+    def params(self):
+        """Get the current turn analyzer parameters.
+
+        Returns:
+            Current turn analyzer configuration parameters.
+        """
+        pass
+
     @abstractmethod
     def append_audio(self, buffer: bytes, is_speech: bool) -> EndOfTurnState:
         """Appends audio data for analysis.

src/pipecat/audio/turn/smart_turn/base_smart_turn.py

@@ -87,6 +87,15 @@ class BaseSmartTurn(BaseTurnAnalyzer):
         """
         return self._speech_triggered
 
+    @property
+    def params(self) -> SmartTurnParams:
+        """Get the current smart turn parameters.
+
+        Returns:
+            Current smart turn configuration parameters.
+        """
+        return self._params
+
     def append_audio(self, buffer: bytes, is_speech: bool) -> EndOfTurnState:
         """Append audio data for turn analysis.
 

src/pipecat/audio/turn/smart_turn/local_smart_turn_v2.py

@@ -0,0 +1,192 @@
+#
+# Copyright (c) 2025, Daily
+#
+# SPDX-License-Identifier: BSD 2-Clause License
+#
+
+"""Local PyTorch turn analyzer for on-device ML inference using the smart-turn-v2 model.
+
+This module provides a smart turn analyzer that uses PyTorch models for
+local end-of-turn detection without requiring network connectivity.
+"""
+
+from typing import Any, Dict
+
+import numpy as np
+from loguru import logger
+
+from pipecat.audio.turn.smart_turn.base_smart_turn import BaseSmartTurn
+
+try:
+    import torch
+    import torch.nn.functional as F
+    from torch import nn
+    from transformers import (
+        Wav2Vec2Config,
+        Wav2Vec2Model,
+        Wav2Vec2PreTrainedModel,
+        Wav2Vec2Processor,
+    )
+except ModuleNotFoundError as e:
+    logger.error(f"Exception: {e}")
+    logger.error(
+        "In order to use LocalSmartTurnAnalyzerV2, you need to `pip install pipecat-ai[local-smart-turn]`."
+    )
+    raise Exception(f"Missing module: {e}")
+
+
+class LocalSmartTurnAnalyzerV2(BaseSmartTurn):
+    """Local turn analyzer using the smart-turn-v2 PyTorch model.
+
+    Provides end-of-turn detection using locally-stored PyTorch models,
+    enabling offline operation without network dependencies. Uses
+    Wav2Vec2 architecture for audio sequence classification.
+    """
+
+    def __init__(self, *, smart_turn_model_path: str, **kwargs):
+        """Initialize the local PyTorch smart-turn-v2 analyzer.
+
+        Args:
+            smart_turn_model_path: Path to directory containing the PyTorch model
+                and feature extractor files. If empty, uses default HuggingFace model.
+            **kwargs: Additional arguments passed to BaseSmartTurn.
+        """
+        super().__init__(**kwargs)
+
+        if not smart_turn_model_path:
+            # Define the path to the pretrained model on Hugging Face
+            smart_turn_model_path = "pipecat-ai/smart-turn-v2"
+
+        logger.debug("Loading Local Smart Turn v2 model...")
+        # Load the pretrained model for sequence classification
+        self._turn_model = _Wav2Vec2ForEndpointing.from_pretrained(smart_turn_model_path)
+        # Load the corresponding feature extractor for preprocessing audio
+        self._turn_processor = Wav2Vec2Processor.from_pretrained(smart_turn_model_path)
+        # Set device to GPU if available, else CPU
+        self._device = torch.device("cuda" if torch.cuda.is_available() else "cpu")
+        # Move model to selected device and set it to evaluation mode
+        self._turn_model = self._turn_model.to(self._device)
+        self._turn_model.eval()
+        logger.debug("Loaded Local Smart Turn v2")
+
+    async def _predict_endpoint(self, audio_array: np.ndarray) -> Dict[str, Any]:
+        """Predict end-of-turn using local PyTorch model."""
+        inputs = self._turn_processor(
+            audio_array,
+            sampling_rate=16000,
+            padding="max_length",
+            truncation=True,
+            max_length=16000 * 16,  # 16 seconds at 16kHz
+            return_attention_mask=True,
+            return_tensors="pt",
+        )
+
+        # Move inputs to device
+        inputs = {k: v.to(self._device) for k, v in inputs.items()}
+
+        # Run inference
+        with torch.no_grad():
+            outputs = self._turn_model(**inputs)
+
+            # The model returns sigmoid probabilities directly in the logits field
+            probability = outputs["logits"][0].item()
+
+            # Make prediction (1 for Complete, 0 for Incomplete)
+            prediction = 1 if probability > 0.5 else 0
+
+        return {
+            "prediction": prediction,
+            "probability": probability,
+        }
+
+
+class _Wav2Vec2ForEndpointing(Wav2Vec2PreTrainedModel):
+    def __init__(self, config: Wav2Vec2Config):
+        super().__init__(config)
+        self.wav2vec2 = Wav2Vec2Model(config)
+
+        self.pool_attention = nn.Sequential(
+            nn.Linear(config.hidden_size, 256), nn.Tanh(), nn.Linear(256, 1)
+        )
+
+        self.classifier = nn.Sequential(
+            nn.Linear(config.hidden_size, 256),
+            nn.LayerNorm(256),
+            nn.GELU(),
+            nn.Dropout(0.1),
+            nn.Linear(256, 64),
+            nn.GELU(),
+            nn.Linear(64, 1),
+        )
+
+        for module in self.classifier:
+            if isinstance(module, nn.Linear):
+                module.weight.data.normal_(mean=0.0, std=0.1)
+                if module.bias is not None:
+                    module.bias.data.zero_()
+
+        for module in self.pool_attention:
+            if isinstance(module, nn.Linear):
+                module.weight.data.normal_(mean=0.0, std=0.1)
+                if module.bias is not None:
+                    module.bias.data.zero_()
+
+    def attention_pool(self, hidden_states, attention_mask):
+        # Calculate attention weights
+        attention_weights = self.pool_attention(hidden_states)
+
+        if attention_mask is None:
+            raise ValueError("attention_mask must be provided for attention pooling")
+
+        attention_weights = attention_weights + (
+            (1.0 - attention_mask.unsqueeze(-1).to(attention_weights.dtype)) * -1e9
+        )
+
+        attention_weights = F.softmax(attention_weights, dim=1)
+
+        # Apply attention to hidden states
+        weighted_sum = torch.sum(hidden_states * attention_weights, dim=1)
+
+        return weighted_sum
+
+    def forward(self, input_values, attention_mask=None, labels=None):
+        outputs = self.wav2vec2(input_values, attention_mask=attention_mask)
+        hidden_states = outputs[0]
+
+        # Create transformer padding mask
+        if attention_mask is not None:
+            input_length = attention_mask.size(1)
+            hidden_length = hidden_states.size(1)
+            ratio = input_length / hidden_length
+            indices = (torch.arange(hidden_length, device=attention_mask.device) * ratio).long()
+            attention_mask = attention_mask[:, indices]
+            attention_mask = attention_mask.bool()
+        else:
+            attention_mask = None
+
+        pooled = self.attention_pool(hidden_states, attention_mask)
+
+        logits = self.classifier(pooled)
+
+        if torch.isnan(logits).any():
+            raise ValueError("NaN values detected in logits")
+
+        if labels is not None:
+            # Calculate positive sample weight based on batch statistics
+            pos_weight = ((labels == 0).sum() / (labels == 1).sum()).clamp(min=0.1, max=10.0)
+            loss_fct = nn.BCEWithLogitsLoss(pos_weight=pos_weight)
+            labels = labels.float()
+            loss = loss_fct(logits.view(-1), labels.view(-1))
+
+            # Add L2 regularization for classifier layers
+            l2_lambda = 0.01
+            l2_reg = torch.tensor(0.0, device=logits.device)
+            for param in self.classifier.parameters():
+                l2_reg += torch.norm(param)
+            loss += l2_lambda * l2_reg
+
+            probs = torch.sigmoid(logits.detach())
+            return {"loss": loss, "logits": probs}
+
+        probs = torch.sigmoid(logits)
+        return {"logits": probs}

src/pipecat/audio/vad/vad_analyzer.py

@@ -183,36 +183,37 @@ class VADAnalyzer(ABC):
         if len(self._vad_buffer) < num_required_bytes:
             return self._vad_state
 
-        audio_frames = self._vad_buffer[:num_required_bytes]
+        while len(self._vad_buffer) >= num_required_bytes:
-        self._vad_buffer = self._vad_buffer[num_required_bytes:]
+            audio_frames = self._vad_buffer[:num_required_bytes]
+            self._vad_buffer = self._vad_buffer[num_required_bytes:]
 
-        confidence = self.voice_confidence(audio_frames)
+            confidence = self.voice_confidence(audio_frames)
 
-        volume = self._get_smoothed_volume(audio_frames)
+            volume = self._get_smoothed_volume(audio_frames)
-        self._prev_volume = volume
+            self._prev_volume = volume
 
-        speaking = confidence >= self._params.confidence and volume >= self._params.min_volume
+            speaking = confidence >= self._params.confidence and volume >= self._params.min_volume
 
-        if speaking:
+            if speaking:
-            match self._vad_state:
+                match self._vad_state:
-                case VADState.QUIET:
+                    case VADState.QUIET:
-                    self._vad_state = VADState.STARTING
+                        self._vad_state = VADState.STARTING
-                    self._vad_starting_count = 1
+                        self._vad_starting_count = 1
-                case VADState.STARTING:
+                    case VADState.STARTING:
-                    self._vad_starting_count += 1
+                        self._vad_starting_count += 1
-                case VADState.STOPPING:
+                    case VADState.STOPPING:
-                    self._vad_state = VADState.SPEAKING
+                        self._vad_state = VADState.SPEAKING
-                    self._vad_stopping_count = 0
+                        self._vad_stopping_count = 0
-        else:
+            else:
-            match self._vad_state:
+                match self._vad_state:
-                case VADState.STARTING:
+                    case VADState.STARTING:
-                    self._vad_state = VADState.QUIET
+                        self._vad_state = VADState.QUIET
-                    self._vad_starting_count = 0
+                        self._vad_starting_count = 0
-                case VADState.SPEAKING:
+                    case VADState.SPEAKING:
-                    self._vad_state = VADState.STOPPING
+                        self._vad_state = VADState.STOPPING
-                    self._vad_stopping_count = 1
+                        self._vad_stopping_count = 1
-                case VADState.STOPPING:
+                    case VADState.STOPPING:
-                    self._vad_stopping_count += 1
+                        self._vad_stopping_count += 1
 
         if (
             self._vad_state == VADState.STARTING

src/pipecat/examples/run.py

@@ -9,6 +9,21 @@
 This module provides a unified interface for running Pipecat examples across
 different transport types including Daily.co, WebRTC, and Twilio. It handles
 setup, configuration, and lifecycle management for each transport type.
+
+Example usage:
+    SmallWebRTCTransport::
+
+        python bot.py --transport webrtc
+
+    DailyTransport::
+
+        python bot.py --transport daily
+
+    Twilio::
+
+        python bot.py --transport twilio --proxy username.ngrok.io
+        # Note: Concurrently, run an ngrok tunnel to your local server:
+        # ngrok http 7860
 """
 
 import argparse

src/pipecat/frames/frames.py

@@ -28,6 +28,7 @@ from typing import (
 )
 
 from pipecat.audio.interruptions.base_interruption_strategy import BaseInterruptionStrategy
+from pipecat.audio.turn.smart_turn.base_smart_turn import SmartTurnParams
 from pipecat.audio.vad.vad_analyzer import VADParams
 from pipecat.metrics.metrics import MetricsData
 from pipecat.transcriptions.language import Language
@@ -1145,6 +1146,23 @@ class OutputDTMFUrgentFrame(DTMFFrame, SystemFrame):
     pass
 
 
+@dataclass
+class SpeechControlParamsFrame(SystemFrame):
+    """Frame for notifying processors of speech control parameter changes.
+
+    This includes parameters for both VAD (Voice Activity Detection) and
+    turn-taking analysis. It allows downstream processors to adjust their
+    behavior based on updated interaction control settings.
+
+    Parameters:
+        vad_params: Current VAD parameters.
+        turn_params: Current turn-taking analysis parameters.
+    """
+
+    vad_params: Optional[VADParams] = None
+    turn_params: Optional[SmartTurnParams] = None
+
+
 #
 # Control frames
 #

src/pipecat/pipeline/parallel_pipeline.py

@@ -273,12 +273,17 @@ class ParallelPipeline(BasePipeline):
         if not self._down_task:
             self._down_task = self.create_task(self._process_down_queue())
 
+    async def _drain_queue(self, queue: asyncio.Queue):
+        try:
+            while not queue.empty():
+                queue.get_nowait()
+        except asyncio.QueueEmpty:
+            logger.debug(f"Draining {self} queue already empty")
+
     async def _drain_queues(self):
         """Drain all frames from upstream and downstream queues."""
-        while not self._up_queue.empty:
+        await self._drain_queue(self._up_queue)
-            await self._up_queue.get()
+        await self._drain_queue(self._down_queue)
-        while not self._down_queue.empty:
-            await self._down_queue.get()
 
     async def _handle_interruption(self):
         """Handle interruption by cancelling tasks, draining queues, and restarting."""

src/pipecat/processors/aggregators/llm_response.py

@@ -19,6 +19,8 @@ from typing import Dict, List, Literal, Optional, Set
 from loguru import logger
 
 from pipecat.audio.interruptions.base_interruption_strategy import BaseInterruptionStrategy
+from pipecat.audio.turn.smart_turn.base_smart_turn import SmartTurnParams
+from pipecat.audio.vad.vad_analyzer import VADParams
 from pipecat.frames.frames import (
     BotInterruptionFrame,
     BotStartedSpeakingFrame,
@@ -43,6 +45,7 @@ from pipecat.frames.frames import (
     LLMSetToolsFrame,
     LLMTextFrame,
     OpenAILLMContextAssistantTimestampFrame,
+    SpeechControlParamsFrame,
     StartFrame,
     StartInterruptionFrame,
     TextFrame,
@@ -67,9 +70,13 @@ class LLMUserAggregatorParams:
         aggregation_timeout: Maximum time in seconds to wait for additional
             transcription content before pushing aggregated result. This
             timeout is used only when the transcription is slow to arrive.
+        turn_emulated_vad_timeout: Maximum time in seconds to wait for emulated
+            VAD when using turn-based analysis. Applied when transcription is
+            received but VAD didn't detect speech (e.g., whispered utterances).
     """
 
     aggregation_timeout: float = 0.5
+    turn_emulated_vad_timeout: float = 0.8
 
 
 @dataclass
@@ -390,6 +397,9 @@ class LLMUserContextAggregator(LLMContextResponseAggregator):
         """
         super().__init__(context=context, role="user", **kwargs)
         self._params = params or LLMUserAggregatorParams()
+        self._vad_params: Optional[VADParams] = None
+        self._turn_params: Optional[SmartTurnParams] = None
+
         if "aggregation_timeout" in kwargs:
             import warnings
 
@@ -477,6 +487,10 @@ class LLMUserContextAggregator(LLMContextResponseAggregator):
             self.set_tools(frame.tools)
         elif isinstance(frame, LLMSetToolChoiceFrame):
             self.set_tool_choice(frame.tool_choice)
+        elif isinstance(frame, SpeechControlParamsFrame):
+            self._vad_params = frame.vad_params
+            self._turn_params = frame.turn_params
+            await self.push_frame(frame, direction)
         else:
             await self.push_frame(frame, direction)
 
@@ -618,9 +632,40 @@ class LLMUserContextAggregator(LLMContextResponseAggregator):
     async def _aggregation_task_handler(self):
         while True:
             try:
-                await asyncio.wait_for(
+                # The _aggregation_task_handler handles two distinct timeout scenarios:
-                    self._aggregation_event.wait(), self._params.aggregation_timeout
+                #
-                )
+                # 1. When emulating_vad=True: Wait for emulated VAD timeout before
+                #    pushing aggregation (simulating VAD behavior when no actual VAD
+                #    detection occurred).
+                #
+                # 2. When emulating_vad=False: Use aggregation_timeout as a buffer
+                #    to wait for potential late-arriving transcription frames after
+                #    a real VAD event.
+                #
+                # For emulated VAD scenarios, the timeout strategy depends on whether
+                # a turn analyzer is configured:
+                #
+                # - WITH turn analyzer: Use turn_emulated_vad_timeout parameter because
+                #   the VAD's stop_secs is set very low (e.g. 0.2s) for rapid speech
+                #   chunking to feed the turn analyzer. This low value is too fast
+                #   for emulated VAD scenarios where we need to allow users time to
+                #   finish speaking (e.g. 0.8s).
+                #
+                # - WITHOUT turn analyzer: Use VAD's stop_secs directly to maintain
+                #   consistent user experience between real VAD detection and
+                #   emulated VAD scenarios.
+                if not self._emulating_vad:
+                    timeout = self._params.aggregation_timeout
+                elif self._turn_params:
+                    timeout = self._params.turn_emulated_vad_timeout
+                else:
+                    # Use VAD stop_secs when no turn analyzer is present, fallback if no VAD params
+                    timeout = (
+                        self._vad_params.stop_secs
+                        if self._vad_params
+                        else self._params.turn_emulated_vad_timeout
+                    )
+                await asyncio.wait_for(self._aggregation_event.wait(), timeout)
                 await self._maybe_emulate_user_speaking()
             except asyncio.TimeoutError:
                 if not self._user_speaking:
@@ -648,7 +693,11 @@ class LLMUserContextAggregator(LLMContextResponseAggregator):
         # to emulate VAD (i.e. user start/stopped speaking), but we do it only
         # if the bot is not speaking. If the bot is speaking and we really have
         # a short utterance we don't really want to interrupt the bot.
-        if not self._user_speaking and not self._waiting_for_aggregation:
+        if (
+            not self._user_speaking
+            and not self._waiting_for_aggregation
+            and len(self._aggregation) > 0
+        ):
             if self._bot_speaking:
                 # If we reached this case and the bot is speaking, let's ignore
                 # what the user said.

src/pipecat/processors/frameworks/rtvi.py

@@ -44,6 +44,7 @@ from pipecat.frames.frames import (
     InterimTranscriptionFrame,
     LLMFullResponseEndFrame,
     LLMFullResponseStartFrame,
+    LLMMessagesAppendFrame,
     LLMTextFrame,
     MetricsFrame,
     StartFrame,
@@ -71,13 +72,14 @@ from pipecat.processors.frame_processor import FrameDirection, FrameProcessor
 from pipecat.services.llm_service import (
     FunctionCallParams,  # TODO(aleix): we shouldn't import `services` from `processors`
 )
+from pipecat.services.openai.llm import OpenAIContextAggregatorPair
 from pipecat.transports.base_input import BaseInputTransport
 from pipecat.transports.base_output import BaseOutputTransport
 from pipecat.transports.base_transport import BaseTransport
 from pipecat.utils.asyncio.watchdog_queue import WatchdogQueue
 from pipecat.utils.string import match_endofsentence
 
-RTVI_PROTOCOL_VERSION = "0.3.0"
+RTVI_PROTOCOL_VERSION = "1.0.0"
 
 RTVI_MESSAGE_LABEL = "rtvi-ai"
 RTVIMessageLiteral = Literal["rtvi-ai"]
@@ -90,6 +92,10 @@ class RTVIServiceOption(BaseModel):
 
     Defines a configurable option that can be set for an RTVI service,
     including its name, type, and handler function.
+
+    .. deprecated:: 0.0.75
+        Pipeline Configuration has been removed as part of the RTVI protocol 1.0.0.
+        Use custom client and server messages instead.
     """
 
     name: str
@@ -104,6 +110,10 @@ class RTVIService(BaseModel):
 
     Represents a service that can be configured and used within the RTVI protocol,
     containing a name and list of configurable options.
+
+    .. deprecated:: 0.0.75
+       Pipeline Configuration has been removed as part of the RTVI protocol 1.0.0.
+       Use custom client and server messages instead.
     """
 
     name: str
@@ -122,6 +132,10 @@ class RTVIActionArgumentData(BaseModel):
     """Data for an RTVI action argument.
 
     Contains the name and value of an argument passed to an RTVI action.
+
+    .. deprecated:: 0.0.75
+        Actions have been removed as part of the RTVI protocol 1.0.0.
+        Use custom client and server messages instead.
     """
 
     name: str
@@ -132,6 +146,10 @@ class RTVIActionArgument(BaseModel):
     """Definition of an RTVI action argument.
 
     Specifies the name and expected type of an argument for an RTVI action.
+
+    .. deprecated:: 0.0.75
+        Actions have been removed as part of the RTVI protocol 1.0.0.
+        Use custom client and server messages instead.
     """
 
     name: str
@@ -143,6 +161,10 @@ class RTVIAction(BaseModel):
 
     Represents an action that can be executed within the RTVI protocol,
     including its service, name, arguments, and handler function.
+
+    .. deprecated:: 0.0.75
+       Actions have been removed as part of the RTVI protocol 1.0.0.
+       Use custom client and server messages instead.
     """
 
     service: str
@@ -166,6 +188,10 @@ class RTVIServiceOptionConfig(BaseModel):
     """Configuration value for an RTVI service option.
 
     Contains the name and value to set for a specific service option.
+
+    .. deprecated:: 0.0.75
+        Pipeline Configuration has been removed as part of the RTVI protocol 1.0.0.
+        Use custom client and server messages instead.
     """
 
     name: str
@@ -176,6 +202,10 @@ class RTVIServiceConfig(BaseModel):
     """Configuration for an RTVI service.
 
     Contains the service name and list of option configurations to apply.
+
+    .. deprecated:: 0.0.75
+       Pipeline Configuration has been removed as part of the RTVI protocol 1.0.0.
+       Use custom client and server messages instead.
     """
 
     service: str
@@ -186,6 +216,10 @@ class RTVIConfig(BaseModel):
     """Complete RTVI configuration.
 
     Contains the full configuration for all RTVI services.
+
+    .. deprecated:: 0.0.75
+       Pipeline Configuration has been removed as part of the RTVI protocol 1.0.0.
+       Use custom client and server messages instead.
     """
 
     config: List[RTVIServiceConfig]
@@ -196,10 +230,15 @@ class RTVIConfig(BaseModel):
 #
 
 
+# deprecated
 class RTVIUpdateConfig(BaseModel):
     """Request to update RTVI configuration.
 
     Contains new configuration settings and whether to interrupt the bot.
+
+    .. deprecated:: 0.0.75
+        Pipeline Configuration has been removed as part of the RTVI protocol 1.0.0.
+        Use custom client and server messages instead.
     """
 
     config: List[RTVIServiceConfig]
@@ -210,6 +249,10 @@ class RTVIActionRunArgument(BaseModel):
     """Argument for running an RTVI action.
 
     Contains the name and value of an argument to pass to an action.
+
+    .. deprecated:: 0.0.75
+        Actions have been removed as part of the RTVI protocol 1.0.0.
+        Use custom client and server messages instead.
     """
 
     name: str
@@ -220,6 +263,10 @@ class RTVIActionRun(BaseModel):
     """Request to run an RTVI action.
 
     Contains the service, action name, and optional arguments.
+
+    .. deprecated:: 0.0.75
+       Actions have been removed as part of the RTVI protocol 1.0.0.
+       Use custom client and server messages instead.
     """
 
     service: str
@@ -234,12 +281,80 @@ class RTVIActionFrame(DataFrame):
     Parameters:
         rtvi_action_run: The action to execute.
         message_id: Optional message ID for response correlation.
+
+    .. deprecated:: 0.0.75
+        Actions have been removed as part of the RTVI protocol 1.0.0.
+        Use custom client and server messages instead.
     """
 
     rtvi_action_run: RTVIActionRun
     message_id: Optional[str] = None
 
 
+class RTVIRawClientMessageData(BaseModel):
+    """Data structure expected from client messages sent to the RTVI server."""
+
+    t: str
+    d: Optional[Any] = None
+
+
+class RTVIClientMessage(BaseModel):
+    """Cleansed data structure for client messages for handling."""
+
+    msg_id: str
+    type: str
+    data: Optional[Any] = None
+
+
+@dataclass
+class RTVIClientMessageFrame(SystemFrame):
+    """A frame for sending messages from the client to the RTVI server.
+
+    This frame is meant for custom messaging from the client to the server
+    and expects a server-response message.
+    """
+
+    msg_id: str
+    type: str
+    data: Optional[Any] = None
+
+
+@dataclass
+class RTVIServerResponseFrame(SystemFrame):
+    """A frame for responding to a client RTVI message.
+
+    This frame should be sent in response to an RTVIClientMessageFrame
+    and include the original RTVIClientMessageFrame to ensure the response
+    is properly attributed to the original request. To respond with an error,
+    set the `error` field to a string describing the error. This will result
+    in the client receiving a `response-error` message instead of a
+    `server-response` message.
+    """
+
+    client_msg: RTVIClientMessageFrame
+    data: Optional[Any] = None
+    error: Optional[str] = None
+
+
+class RTVIRawServerResponseData(BaseModel):
+    """Data structure for server responses to client messages."""
+
+    t: str
+    d: Optional[Any] = None
+
+
+class RTVIServerResponse(BaseModel):
+    """The RTVI-formatted message response from the server to the client.
+
+    This message is used to respond to custom messages sent by the client.
+    """
+
+    label: RTVIMessageLiteral = RTVI_MESSAGE_LABEL
+    type: Literal["server-response"] = "server-response"
+    id: str
+    data: RTVIRawServerResponseData
+
+
 class RTVIMessage(BaseModel):
     """Base RTVI message structure.
 
@@ -269,7 +384,7 @@ class RTVIErrorResponseData(BaseModel):
 class RTVIErrorResponse(BaseModel):
     """RTVI error response message.
 
-    Sent in response to a client request that resulted in an error.
+    RTVI Formatted error response message for relaying failed client requests.
     """
 
     label: RTVIMessageLiteral = RTVI_MESSAGE_LABEL
@@ -285,13 +400,13 @@ class RTVIErrorData(BaseModel):
     """
 
     error: str
-    fatal: bool
+    fatal: bool  # Indicates the pipeline has stopped due to this error
 
 
 class RTVIError(BaseModel):
     """RTVI error event message.
 
-    Sent when an error occurs that isn't in response to a specific request.
+    RTVI Formatted error message for relaying errors in the pipeline.
     """
 
     label: RTVIMessageLiteral = RTVI_MESSAGE_LABEL
@@ -303,6 +418,10 @@ class RTVIDescribeConfigData(BaseModel):
     """Data for describing available RTVI configuration.
 
     Contains the list of available services and their options.
+
+    .. deprecated:: 0.0.75
+        Pipeline Configuration has been removed as part of the RTVI protocol 1.0.0.
+        Use custom client and server messages instead.
     """
 
     config: List[RTVIService]
@@ -312,6 +431,10 @@ class RTVIDescribeConfig(BaseModel):
     """Message describing available RTVI configuration.
 
     Sent in response to a describe-config request.
+
+    .. deprecated:: 0.0.75
+        Pipeline Configuration has been removed as part of the RTVI protocol 1.0.0.
+        Use custom client and server messages instead.
     """
 
     label: RTVIMessageLiteral = RTVI_MESSAGE_LABEL
@@ -324,6 +447,10 @@ class RTVIDescribeActionsData(BaseModel):
     """Data for describing available RTVI actions.
 
     Contains the list of available actions that can be executed.
+
+    .. deprecated:: 0.0.75
+        Actions have been removed as part of the RTVI protocol 1.0.0.
+        Use custom client and server messages instead.
     """
 
     actions: List[RTVIAction]
@@ -333,6 +460,10 @@ class RTVIDescribeActions(BaseModel):
     """Message describing available RTVI actions.
 
     Sent in response to a describe-actions request.
+
+    .. deprecated:: 0.0.75
+        Actions have been removed as part of the RTVI protocol 1.0.0.
+        Use custom client and server messages instead.
     """
 
     label: RTVIMessageLiteral = RTVI_MESSAGE_LABEL
@@ -345,6 +476,10 @@ class RTVIConfigResponse(BaseModel):
     """Response containing current RTVI configuration.
 
     Sent in response to a get-config request.
+
+    .. deprecated:: 0.0.75
+        Pipeline Configuration has been removed as part of the RTVI protocol 1.0.0.
+        Use custom client and server messages instead.
     """
 
     label: RTVIMessageLiteral = RTVI_MESSAGE_LABEL
@@ -357,6 +492,10 @@ class RTVIActionResponseData(BaseModel):
     """Data for an RTVI action response.
 
     Contains the result of executing an action.
+
+    .. deprecated:: 0.0.75
+        Actions have been removed as part of the RTVI protocol 1.0.0.
+        Use custom client and server messages instead.
     """
 
     result: ActionResult
@@ -366,6 +505,10 @@ class RTVIActionResponse(BaseModel):
     """Response to an RTVI action execution.
 
     Sent after successfully executing an action.
+
+    .. deprecated:: 0.0.75
+        Actions have been removed as part of the RTVI protocol 1.0.0.
+        Use custom client and server messages instead.
     """
 
     label: RTVIMessageLiteral = RTVI_MESSAGE_LABEL
@@ -374,6 +517,30 @@ class RTVIActionResponse(BaseModel):
     data: RTVIActionResponseData
 
 
+class AboutClientData(BaseModel):
+    """Data about the RTVI client.
+
+    Contains information about the client, including which RTVI library it
+    is using, what platform it is on and any additional details, if available.
+    """
+
+    library: str
+    library_version: Optional[str] = None
+    platform: Optional[str] = None
+    platform_version: Optional[str] = None
+    platform_details: Optional[Any] = None
+
+
+class RTVIClientReadyData(BaseModel):
+    """Data format of client ready messages.
+
+    Contains the RTVIprotocol version and client information.
+    """
+
+    version: str
+    about: AboutClientData
+
+
 class RTVIBotReadyData(BaseModel):
     """Data for bot ready notification.
 
@@ -381,7 +548,10 @@ class RTVIBotReadyData(BaseModel):
     """
 
     version: str
-    config: List[RTVIServiceConfig]
+    # The config field is deprecated and will not be included if
+    # the client's rtvi version is 1.0.0 or higher.
+    config: Optional[List[RTVIServiceConfig]] = None
+    about: Optional[Mapping[str, Any]] = None
 
 
 class RTVIBotReady(BaseModel):
@@ -418,6 +588,25 @@ class RTVILLMFunctionCallMessage(BaseModel):
     data: RTVILLMFunctionCallMessageData
 
 
+class RTVIAppendToContextData(BaseModel):
+    """Data format for appending messages to the context.
+
+    Contains the role, content, and whether to run the message immediately.
+    """
+
+    role: Literal["user", "assistant"] | str
+    content: Any
+    run_immediately: bool = False
+
+
+class RTVIAppendToContext(BaseModel):
+    """RTVI Message format to append content to the LLM context."""
+
+    label: RTVIMessageLiteral = RTVI_MESSAGE_LABEL
+    type: Literal["append-to-context"] = "append-to-context"
+    data: RTVIAppendToContextData
+
+
 class RTVILLMFunctionCallStartMessageData(BaseModel):
     """Data for LLM function call start notification.
 
@@ -752,6 +941,11 @@ class RTVIObserver(BaseObserver):
         elif isinstance(frame, RTVIServerMessageFrame):
             message = RTVIServerMessage(data=frame.data)
             await self.push_transport_message_urgent(message)
+        elif isinstance(frame, RTVIServerResponseFrame):
+            if frame.error is not None:
+                await self._send_error_response(frame)
+            else:
+                await self._send_server_response(frame)
 
         if mark_as_seen:
             self._frames_seen.add(frame.id)
@@ -879,6 +1073,22 @@ class RTVIObserver(BaseObserver):
         message = RTVIMetricsMessage(data=metrics)
         await self.push_transport_message_urgent(message)
 
+    async def _send_server_response(self, frame: RTVIServerResponseFrame):
+        """Send a response to the client for a specific request."""
+        message = RTVIServerResponse(
+            id=str(frame.client_msg.msg_id),
+            data=RTVIRawServerResponseData(t=frame.client_msg.type, d=frame.data),
+        )
+        await self.push_transport_message_urgent(message)
+
+    async def _send_error_response(self, frame: RTVIServerResponseFrame):
+        """Send a response to the client for a specific request."""
+        if self._params.errors_enabled:
+            message = RTVIErrorResponse(
+                id=str(frame.client_msg.msg_id), data=RTVIErrorResponseData(error=frame.error)
+            )
+            await self.push_transport_message_urgent(message)
+
 
 class RTVIProcessor(FrameProcessor):
     """Main processor for handling RTVI protocol messages and actions.
@@ -908,6 +1118,7 @@ class RTVIProcessor(FrameProcessor):
         self._bot_ready = False
         self._client_ready = False
         self._client_ready_id = ""
+        self._client_version = []
         self._errors_enabled = True
 
         self._registered_actions: Dict[str, RTVIAction] = {}
@@ -921,6 +1132,7 @@ class RTVIProcessor(FrameProcessor):
 
         self._register_event_handler("on_bot_started")
         self._register_event_handler("on_client_ready")
+        self._register_event_handler("on_client_message")
 
         self._input_transport = None
         self._transport = transport
@@ -936,6 +1148,15 @@ class RTVIProcessor(FrameProcessor):
         Args:
             action: The action to register.
         """
+        import warnings
+
+        with warnings.catch_warnings():
+            warnings.simplefilter("always")
+            warnings.warn(
+                "The actions API is deprecated, use server and client messages instead.",
+                DeprecationWarning,
+            )
+
         id = self._action_id(action.service, action.action)
         self._registered_actions[id] = action
 
@@ -945,6 +1166,15 @@ class RTVIProcessor(FrameProcessor):
         Args:
             service: The service to register.
         """
+        import warnings
+
+        with warnings.catch_warnings():
+            warnings.simplefilter("always")
+            warnings.warn(
+                "The actions API is deprecated, use server and client messages instead.",
+                DeprecationWarning,
+            )
+
         self._registered_services[service.name] = service
 
     async def set_client_ready(self):
@@ -970,6 +1200,22 @@ class RTVIProcessor(FrameProcessor):
         """Send a bot interruption frame upstream."""
         await self.push_frame(BotInterruptionFrame(), FrameDirection.UPSTREAM)
 
+    async def send_server_message(self, data: Any):
+        """Send a server message to the client."""
+        message = RTVIServerMessage(data=data)
+        await self._send_server_message(message)
+
+    async def send_server_response(self, client_msg: RTVIClientMessage, data: Any):
+        """Send a server response for a given client message."""
+        message = RTVIServerResponse(
+            id=client_msg.msg_id, data=RTVIRawServerResponseData(t=client_msg.type, d=data)
+        )
+        await self._send_server_message(message)
+
+    async def send_error_response(self, client_msg: RTVIClientMessage, error: str):
+        """Send an error response for a given client message."""
+        await self._send_error_response(id=client_msg.msg_id, error=error)
+
     async def send_error(self, error: str):
         """Send an error message to the client.
 
@@ -1013,9 +1259,6 @@ class RTVIProcessor(FrameProcessor):
             function_name: Name of the function being called.
             llm: The LLM processor making the call.
             context: The LLM context.
-
-        Note:
-            This method is deprecated. Use handle_function_call() instead.
         """
         import warnings
 
@@ -1136,7 +1379,15 @@ class RTVIProcessor(FrameProcessor):
         try:
             match message.type:
                 case "client-ready":
-                    await self._handle_client_ready(message.id)
+                    data = None
+                    try:
+                        data = RTVIClientReadyData.model_validate(message.data)
+                    except ValidationError:
+                        # Not all clients have been updated to RTVI 1.0.0.
+                        # For now, that's okay, we just log their info as unknown.
+                        data = None
+                        pass
+                    await self._handle_client_ready(message.id, data)
                 case "describe-actions":
                     await self._handle_describe_actions(message.id)
                 case "describe-config":
@@ -1148,6 +1399,9 @@ class RTVIProcessor(FrameProcessor):
                     await self._handle_update_config(message.id, update_config)
                 case "disconnect-bot":
                     await self.push_frame(EndTaskFrame(), FrameDirection.UPSTREAM)
+                case "client-message":
+                    data = RTVIRawClientMessageData.model_validate(message.data)
+                    await self._handle_client_message(message.id, data)
                 case "action":
                     action = RTVIActionRun.model_validate(message.data)
                     action_frame = RTVIActionFrame(message_id=message.id, rtvi_action_run=action)
@@ -1155,6 +1409,9 @@ class RTVIProcessor(FrameProcessor):
                 case "llm-function-call-result":
                     data = RTVILLMFunctionCallResultData.model_validate(message.data)
                     await self._handle_function_call_result(data)
+                case "append-to-context":
+                    data = RTVIAppendToContextData.model_validate(message.data)
+                    await self._handle_update_context(data)
                 case "raw-audio" | "raw-audio-batch":
                     await self._handle_audio_buffer(message.data)
 
@@ -1168,9 +1425,20 @@ class RTVIProcessor(FrameProcessor):
             await self._send_error_response(message.id, f"Exception processing message: {e}")
             logger.warning(f"Exception processing message: {e}")
 
-    async def _handle_client_ready(self, request_id: str):
+    async def _handle_client_ready(self, request_id: str, data: RTVIClientReadyData | None):
-        """Handle a client-ready message."""
+        """Handle the client-ready message from the client."""
-        logger.debug("Received client-ready")
+        version = data.version if data else "unknown"
+        logger.debug(f"Received client-ready: version {version}")
+        if version == "unknown":
+            self._client_version = [0, 3, 0]  # Default to 0.3.0 if unknown
+        else:
+            try:
+                self._client_version = [int(v) for v in version.split(".")]
+            except ValueError:
+                logger.warning(f"Invalid client version format: {version}")
+                self._client_version = [0, 3, 0]
+        about = data.about if data else {"library": "unknown"}
+        logger.debug(f"Client Details: {about}")
         if self._input_transport:
             await self._input_transport.start_audio_in_streaming()
 
@@ -1201,18 +1469,45 @@ class RTVIProcessor(FrameProcessor):
 
     async def _handle_describe_config(self, request_id: str):
         """Handle a describe-config request."""
+        import warnings
+
+        with warnings.catch_warnings():
+            warnings.simplefilter("always")
+            warnings.warn(
+                "Configuration helpers are deprecated. If your application needs this behavior, use custom server and client messages.",
+                DeprecationWarning,
+            )
+
         services = list(self._registered_services.values())
         message = RTVIDescribeConfig(id=request_id, data=RTVIDescribeConfigData(config=services))
         await self._push_transport_message(message)
 
     async def _handle_describe_actions(self, request_id: str):
         """Handle a describe-actions request."""
+        import warnings
+
+        with warnings.catch_warnings():
+            warnings.simplefilter("always")
+            warnings.warn(
+                "The Actions API is deprecated, use custom server and client messages instead.",
+                DeprecationWarning,
+            )
+
         actions = list(self._registered_actions.values())
         message = RTVIDescribeActions(id=request_id, data=RTVIDescribeActionsData(actions=actions))
         await self._push_transport_message(message)
 
     async def _handle_get_config(self, request_id: str):
         """Handle a get-config request."""
+        import warnings
+
+        with warnings.catch_warnings():
+            warnings.simplefilter("always")
+            warnings.warn(
+                "Configuration helpers are deprecated. If your application needs this behavior, use custom server and client messages.",
+                DeprecationWarning,
+            )
+
         message = RTVIConfigResponse(id=request_id, data=self._config)
         await self._push_transport_message(message)
 
@@ -1230,6 +1525,15 @@ class RTVIProcessor(FrameProcessor):
 
     async def _update_service_config(self, config: RTVIServiceConfig):
         """Update configuration for a specific service."""
+        import warnings
+
+        with warnings.catch_warnings():
+            warnings.simplefilter("always")
+            warnings.warn(
+                "Configuration helpers are deprecated. If your application needs this behavior, use custom server and client messages.",
+                DeprecationWarning,
+            )
+
         service = self._registered_services[config.service]
         for option in config.options:
             handler = service._options_dict[option.name].handler
@@ -1238,6 +1542,15 @@ class RTVIProcessor(FrameProcessor):
 
     async def _update_config(self, data: RTVIConfig, interrupt: bool):
         """Update the RTVI configuration."""
+        import warnings
+
+        with warnings.catch_warnings():
+            warnings.simplefilter("always")
+            warnings.warn(
+                "Configuration helpers are deprecated. If your application needs this behavior, use custom server and client messages.",
+                DeprecationWarning,
+            )
+
         if interrupt:
             await self.interrupt_bot()
         for service_config in data.config:
@@ -1248,6 +1561,33 @@ class RTVIProcessor(FrameProcessor):
         await self._update_config(RTVIConfig(config=data.config), data.interrupt)
         await self._handle_get_config(request_id)
 
+    async def _handle_update_context(self, data: RTVIAppendToContextData):
+        if data.run_immediately:
+            await self.interrupt_bot()
+        frame = LLMMessagesAppendFrame(
+            messages=[{"role": data.role, "content": data.content}],
+            run_llm=data.run_immediately,
+        )
+        await self.push_frame(frame)
+
+    async def _handle_client_message(self, msg_id: str, data: RTVIRawClientMessageData):
+        """Handle a client message frame."""
+        if not data:
+            await self._send_error_response(msg_id, "Malformed client message")
+            return
+
+        # Create a RTVIClientMessageFrame to push the message
+        frame = RTVIClientMessageFrame(msg_id=msg_id, type=data.t, data=data.d)
+        await self.push_frame(frame)
+        await self._call_event_handler(
+            "on_client_message",
+            RTVIClientMessage(
+                msg_id=msg_id,
+                type=data.t,
+                data=data.d,
+            ),
+        )
+
     async def _handle_function_call_result(self, data):
         """Handle a function call result from the client."""
         frame = FunctionCallResultFrame(
@@ -1278,12 +1618,19 @@ class RTVIProcessor(FrameProcessor):
 
     async def _send_bot_ready(self):
         """Send the bot-ready message to the client."""
+        config = None
+        if self._client_version[0] < 1:
+            config = self._config.config
         message = RTVIBotReady(
             id=self._client_ready_id,
-            data=RTVIBotReadyData(version=RTVI_PROTOCOL_VERSION, config=self._config.config),
+            data=RTVIBotReadyData(version=RTVI_PROTOCOL_VERSION, config=config),
         )
         await self._push_transport_message(message)
 
+    async def _send_server_message(self, message: RTVIServerMessage | RTVIServerResponse):
+        """Send a message or response to the client."""
+        await self._push_transport_message(message)
+
     async def _send_error_frame(self, frame: ErrorFrame):
         """Send an error frame as an RTVI error message."""
         if self._errors_enabled:

src/pipecat/processors/user_idle_processor.py

@@ -15,6 +15,8 @@ from pipecat.frames.frames import (
     CancelFrame,
     EndFrame,
     Frame,
+    FunctionCallInProgressFrame,
+    FunctionCallResultFrame,
     StartFrame,
     UserStartedSpeakingFrame,
     UserStoppedSpeakingFrame,
@@ -168,6 +170,13 @@ class UserIdleProcessor(FrameProcessor):
                 self._idle_event.set()
             elif isinstance(frame, BotSpeakingFrame):
                 self._idle_event.set()
+            elif isinstance(frame, FunctionCallInProgressFrame):
+                # Function calls can take longer than the timeout, so we want to prevent idle callbacks
+                self._interrupted = True
+                self._idle_event.set()
+            elif isinstance(frame, FunctionCallResultFrame):
+                self._interrupted = False
+                self._idle_event.set()
 
     async def cleanup(self) -> None:
         """Cleans up resources when processor is shutting down."""

src/pipecat/serializers/exotel.py

@@ -108,6 +108,10 @@ class ExotelFrameSerializer(FrameSerializer):
             serialized_data = await self._output_resampler.resample(
                 data, frame.sample_rate, self._exotel_sample_rate
             )
+            if serialized_data is None or len(serialized_data) == 0:
+                # Ignoring in case we don't have audio
+                return None
+
             payload = base64.b64encode(serialized_data).decode("ascii")
 
             answer = {
@@ -144,6 +148,9 @@ class ExotelFrameSerializer(FrameSerializer):
                 self._exotel_sample_rate,
                 self._sample_rate,
             )
+            if deserialized_data is None or len(deserialized_data) == 0:
+                # Ignoring in case we don't have audio
+                return None
 
             # Input: Exotel takes PCM data, so just resample to match sample_rate
             audio_frame = InputAudioRawFrame(

src/pipecat/serializers/plivo.py

@@ -132,6 +132,10 @@ class PlivoFrameSerializer(FrameSerializer):
             serialized_data = await pcm_to_ulaw(
                 data, frame.sample_rate, self._plivo_sample_rate, self._output_resampler
             )
+            if serialized_data is None or len(serialized_data) == 0:
+                # Ignoring in case we don't have audio
+                return None
+
             payload = base64.b64encode(serialized_data).decode("utf-8")
             answer = {
                 "event": "playAudio",
@@ -227,6 +231,10 @@ class PlivoFrameSerializer(FrameSerializer):
             deserialized_data = await ulaw_to_pcm(
                 payload, self._plivo_sample_rate, self._sample_rate, self._input_resampler
             )
+            if deserialized_data is None or len(deserialized_data) == 0:
+                # Ignoring in case we don't have audio
+                return None
+
             audio_frame = InputAudioRawFrame(
                 audio=deserialized_data, num_channels=1, sample_rate=self._sample_rate
             )

src/pipecat/serializers/telnyx.py

@@ -155,6 +155,10 @@ class TelnyxFrameSerializer(FrameSerializer):
             else:
                 raise ValueError(f"Unsupported encoding: {self._params.inbound_encoding}")
 
+            if serialized_data is None or len(serialized_data) == 0:
+                # Ignoring in case we don't have audio
+                return None
+
             payload = base64.b64encode(serialized_data).decode("utf-8")
             answer = {
                 "event": "media",
@@ -262,6 +266,10 @@ class TelnyxFrameSerializer(FrameSerializer):
             else:
                 raise ValueError(f"Unsupported encoding: {self._params.outbound_encoding}")
 
+            if deserialized_data is None or len(deserialized_data) == 0:
+                # Ignoring in case we don't have audio
+                return None
+
             audio_frame = InputAudioRawFrame(
                 audio=deserialized_data, num_channels=1, sample_rate=self._sample_rate
             )

src/pipecat/serializers/twilio.py

@@ -132,6 +132,10 @@ class TwilioFrameSerializer(FrameSerializer):
             serialized_data = await pcm_to_ulaw(
                 data, frame.sample_rate, self._twilio_sample_rate, self._output_resampler
             )
+            if serialized_data is None or len(serialized_data) == 0:
+                # Ignoring in case we don't have audio
+                return None
+
             payload = base64.b64encode(serialized_data).decode("utf-8")
             answer = {
                 "event": "media",
@@ -185,8 +189,26 @@ class TwilioFrameSerializer(FrameSerializer):
                 async with session.post(endpoint, auth=auth, data=params) as response:
                     if response.status == 200:
                         logger.info(f"Successfully terminated Twilio call {call_sid}")
+                    elif response.status == 404:
+                        # Handle the case where the call has already ended
+                        # Error code 20404: "The requested resource was not found"
+                        # Source: https://www.twilio.com/docs/errors/20404
+                        try:
+                            error_data = await response.json()
+                            if error_data.get("code") == 20404:
+                                logger.debug(f"Twilio call {call_sid} was already terminated")
+                                return
+                        except:
+                            pass  # Fall through to log the raw error
+
+                        # Log other 404 errors
+                        error_text = await response.text()
+                        logger.error(
+                            f"Failed to terminate Twilio call {call_sid}: "
+                            f"Status {response.status}, Response: {error_text}"
+                        )
                     else:
-                        # Get the error details for better debugging
+                        # Log other errors
                         error_text = await response.text()
                         logger.error(
                             f"Failed to terminate Twilio call {call_sid}: "
@@ -217,6 +239,10 @@ class TwilioFrameSerializer(FrameSerializer):
             deserialized_data = await ulaw_to_pcm(
                 payload, self._twilio_sample_rate, self._sample_rate, self._input_resampler
             )
+            if deserialized_data is None or len(deserialized_data) == 0:
+                # Ignoring in case we don't have audio
+                return None
+
             audio_frame = InputAudioRawFrame(
                 audio=deserialized_data, num_channels=1, sample_rate=self._sample_rate
             )

src/pipecat/services/aws/llm.py

@@ -55,7 +55,7 @@ from pipecat.services.llm_service import LLMService
 from pipecat.utils.tracing.service_decorators import traced_llm
 
 try:
-    import boto3
+    import aioboto3
     import httpx
     from botocore.config import Config
 except ModuleNotFoundError as e:
@@ -749,13 +749,17 @@ class AWSBedrockLLMService(LLMService):
                 read_timeout=300,  # 5 minutes
                 retries={"max_attempts": 3},
             )
-        session = boto3.Session(
+
-            aws_access_key_id=aws_access_key,
+        self._aws_session = aioboto3.Session()
-            aws_secret_access_key=aws_secret_key,
+
-            aws_session_token=aws_session_token,
+        # Store AWS session parameters for creating client in async context
-            region_name=aws_region,
+        self._aws_params = {
-        )
+            "aws_access_key_id": aws_access_key,
-        self._client = session.client(service_name="bedrock-runtime", config=client_config)
+            "aws_secret_access_key": aws_secret_key,
+            "aws_session_token": aws_session_token,
+            "region_name": aws_region,
+            "config": client_config,
+        }
 
         self.set_model_name(model)
         self._settings = {
@@ -903,70 +907,74 @@ class AWSBedrockLLMService(LLMService):
 
             logger.debug(f"Calling AWS Bedrock model with: {request_params}")
 
-            # Call AWS Bedrock with streaming
+            async with self._aws_session.client(
-            response = self._client.converse_stream(**request_params)
+                service_name="bedrock-runtime", **self._aws_params
+            ) as client:
+                # Call AWS Bedrock with streaming
+                response = await client.converse_stream(**request_params)
 
-            await self.stop_ttfb_metrics()
+                await self.stop_ttfb_metrics()
 
-            # Process the streaming response
+                # Process the streaming response
-            tool_use_block = None
+                tool_use_block = None
-            json_accumulator = ""
+                json_accumulator = ""
 
-            function_calls = []
+                function_calls = []
-            for event in response["stream"]:
-                self.reset_watchdog()
 
-                # Handle text content
+                async for event in response["stream"]:
-                if "contentBlockDelta" in event:
+                    self.reset_watchdog()
-                    delta = event["contentBlockDelta"]["delta"]
-                    if "text" in delta:
-                        await self.push_frame(LLMTextFrame(delta["text"]))
-                        completion_tokens_estimate += self._estimate_tokens(delta["text"])
-                    elif "toolUse" in delta and "input" in delta["toolUse"]:
-                        # Handle partial JSON for tool use
-                        json_accumulator += delta["toolUse"]["input"]
-                        completion_tokens_estimate += self._estimate_tokens(
-                            delta["toolUse"]["input"]
-                        )
 
-                # Handle tool use start
+                    # Handle text content
-                elif "contentBlockStart" in event:
+                    if "contentBlockDelta" in event:
-                    content_block_start = event["contentBlockStart"]["start"]
+                        delta = event["contentBlockDelta"]["delta"]
-                    if "toolUse" in content_block_start:
+                        if "text" in delta:
-                        tool_use_block = {
+                            await self.push_frame(LLMTextFrame(delta["text"]))
-                            "id": content_block_start["toolUse"].get("toolUseId", ""),
+                            completion_tokens_estimate += self._estimate_tokens(delta["text"])
-                            "name": content_block_start["toolUse"].get("name", ""),
+                        elif "toolUse" in delta and "input" in delta["toolUse"]:
-                        }
+                            # Handle partial JSON for tool use
-                        json_accumulator = ""
+                            json_accumulator += delta["toolUse"]["input"]
+                            completion_tokens_estimate += self._estimate_tokens(
+                                delta["toolUse"]["input"]
+                            )
 
-                # Handle message completion with tool use
+                    # Handle tool use start
-                elif "messageStop" in event and "stopReason" in event["messageStop"]:
+                    elif "contentBlockStart" in event:
-                    if event["messageStop"]["stopReason"] == "tool_use" and tool_use_block:
+                        content_block_start = event["contentBlockStart"]["start"]
-                        try:
+                        if "toolUse" in content_block_start:
-                            arguments = json.loads(json_accumulator) if json_accumulator else {}
+                            tool_use_block = {
+                                "id": content_block_start["toolUse"].get("toolUseId", ""),
+                                "name": content_block_start["toolUse"].get("name", ""),
+                            }
+                            json_accumulator = ""
 
-                            # Only call function if it's not the no_operation tool
+                    # Handle message completion with tool use
-                            if not using_noop_tool:
+                    elif "messageStop" in event and "stopReason" in event["messageStop"]:
-                                function_calls.append(
+                        if event["messageStop"]["stopReason"] == "tool_use" and tool_use_block:
-                                    FunctionCallFromLLM(
+                            try:
-                                        context=context,
+                                arguments = json.loads(json_accumulator) if json_accumulator else {}
-                                        tool_call_id=tool_use_block["id"],
+
-                                        function_name=tool_use_block["name"],
+                                # Only call function if it's not the no_operation tool
-                                        arguments=arguments,
+                                if not using_noop_tool:
+                                    function_calls.append(
+                                        FunctionCallFromLLM(
+                                            context=context,
+                                            tool_call_id=tool_use_block["id"],
+                                            function_name=tool_use_block["name"],
+                                            arguments=arguments,
+                                        )
                                     )
-                                )
+                                else:
-                            else:
+                                    logger.debug("Ignoring no_operation tool call")
-                                logger.debug("Ignoring no_operation tool call")
+                            except json.JSONDecodeError:
-                        except json.JSONDecodeError:
+                                logger.error(f"Failed to parse tool arguments: {json_accumulator}")
-                            logger.error(f"Failed to parse tool arguments: {json_accumulator}")
 
-                # Handle usage metrics if available
+                    # Handle usage metrics if available
-                if "metadata" in event and "usage" in event["metadata"]:
+                    if "metadata" in event and "usage" in event["metadata"]:
-                    usage = event["metadata"]["usage"]
+                        usage = event["metadata"]["usage"]
-                    prompt_tokens += usage.get("inputTokens", 0)
+                        prompt_tokens += usage.get("inputTokens", 0)
-                    completion_tokens += usage.get("outputTokens", 0)
+                        completion_tokens += usage.get("outputTokens", 0)
-                    cache_read_input_tokens += usage.get("cacheReadInputTokens", 0)
+                        cache_read_input_tokens += usage.get("cacheReadInputTokens", 0)
-                    cache_creation_input_tokens += usage.get("cacheWriteInputTokens", 0)
+                        cache_creation_input_tokens += usage.get("cacheWriteInputTokens", 0)
 
             await self.run_function_calls(function_calls)
         except asyncio.CancelledError:

src/pipecat/services/aws/tts.py

@@ -30,7 +30,7 @@ from pipecat.transcriptions.language import Language
 from pipecat.utils.tracing.service_decorators import traced_tts
 
 try:
-    import boto3
+    import aioboto3
     from botocore.exceptions import BotoCoreError, ClientError
 except ModuleNotFoundError as e:
     logger.error(f"Exception: {e}")
@@ -177,13 +177,25 @@ class AWSPollyTTSService(TTSService):
 
         params = params or AWSPollyTTSService.InputParams()
 
-        self._polly_client = boto3.client(
+        # Get credentials from environment variables if not provided
-            "polly",
+        self._aws_params = {
-            aws_access_key_id=aws_access_key_id,
+            "aws_access_key_id": aws_access_key_id or os.getenv("AWS_ACCESS_KEY_ID"),
-            aws_secret_access_key=api_key,
+            "aws_secret_access_key": api_key or os.getenv("AWS_SECRET_ACCESS_KEY"),
-            aws_session_token=aws_session_token,
+            "aws_session_token": aws_session_token or os.getenv("AWS_SESSION_TOKEN"),
-            region_name=region,
+            "region_name": region or os.getenv("AWS_REGION", "us-east-1"),
-        )
+        }
+
+        # Validate that we have the required credentials
+        if (
+            not self._aws_params["aws_access_key_id"]
+            or not self._aws_params["aws_secret_access_key"]
+        ):
+            raise ValueError(
+                "AWS credentials not found. Please provide them either through constructor parameters "
+                "or set AWS_ACCESS_KEY_ID and AWS_SECRET_ACCESS_KEY environment variables."
+            )
+
+        self._aws_session = aioboto3.Session()
         self._settings = {
             "engine": params.engine,
             "language": self.language_to_service_language(params.language)
@@ -199,24 +211,6 @@ class AWSPollyTTSService(TTSService):
 
         self.set_voice(voice_id)
 
-        # Get credentials from environment variables if not provided
-        self._credentials = {
-            "aws_access_key_id": aws_access_key_id or os.getenv("AWS_ACCESS_KEY_ID"),
-            "aws_secret_access_key": api_key or os.getenv("AWS_SECRET_ACCESS_KEY"),
-            "aws_session_token": aws_session_token or os.getenv("AWS_SESSION_TOKEN"),
-            "region": region or os.getenv("AWS_REGION", "us-east-1"),
-        }
-
-        # Validate that we have the required credentials
-        if (
-            not self._credentials["aws_access_key_id"]
-            or not self._credentials["aws_secret_access_key"]
-        ):
-            raise ValueError(
-                "AWS credentials not found. Please provide them either through constructor parameters "
-                "or set AWS_ACCESS_KEY_ID and AWS_SECRET_ACCESS_KEY environment variables."
-            )
-
     def can_generate_metrics(self) -> bool:
         """Check if this service can generate processing metrics.
 
@@ -279,14 +273,6 @@ class AWSPollyTTSService(TTSService):
         Yields:
             Frame: Audio frames containing the synthesized speech.
         """
-
-        def read_audio_data(**args):
-            response = self._polly_client.synthesize_speech(**args)
-            if "AudioStream" in response:
-                audio_data = response["AudioStream"].read()
-                return audio_data
-            return None
-
         logger.debug(f"{self}: Generating TTS [{text}]")
 
         try:
@@ -309,30 +295,32 @@ class AWSPollyTTSService(TTSService):
             # Filter out None values
             filtered_params = {k: v for k, v in params.items() if v is not None}
 
-            audio_data = await asyncio.to_thread(read_audio_data, **filtered_params)
+            async with self._aws_session.client("polly", **self._aws_params) as polly:
+                response = await polly.synthesize_speech(**filtered_params)
+                if "AudioStream" in response:
+                    # Get the streaming body and read it
+                    stream = response["AudioStream"]
+                    audio_data = await stream.read()
+                else:
+                    logger.error(f"{self} No audio stream in response")
+                    audio_data = None
 
-            if not audio_data:
+                audio_data = await self._resampler.resample(audio_data, 16000, self.sample_rate)
-                logger.error(f"{self} No audio data returned")
-                yield None
-                return
 
-            audio_data = await self._resampler.resample(audio_data, 16000, self.sample_rate)
+                await self.start_tts_usage_metrics(text)
 
-            await self.start_tts_usage_metrics(text)
+                yield TTSStartedFrame()
 
-            yield TTSStartedFrame()
+                CHUNK_SIZE = self.chunk_size
 
-            CHUNK_SIZE = self.chunk_size
+                for i in range(0, len(audio_data), CHUNK_SIZE):
-
+                    chunk = audio_data[i : i + CHUNK_SIZE]
-            for i in range(0, len(audio_data), CHUNK_SIZE):
+                    if len(chunk) > 0:
-                chunk = audio_data[i : i + CHUNK_SIZE]
+                        await self.stop_ttfb_metrics()
-                if len(chunk) > 0:
+                        frame = TTSAudioRawFrame(chunk, self.sample_rate, 1)
-                    await self.stop_ttfb_metrics()
+                        yield frame
-                    frame = TTSAudioRawFrame(chunk, self.sample_rate, 1)
-                    yield frame
-
-            yield TTSStoppedFrame()
 
+                yield TTSStoppedFrame()
         except (BotoCoreError, ClientError) as error:
             logger.exception(f"{self} error generating TTS: {error}")
             error_message = f"AWS Polly TTS error: {str(error)}"

src/pipecat/services/aws_nova_sonic/aws.py

@@ -474,7 +474,6 @@ class AWSNovaSonicLLMService(LLMService):
         # If we need to, send assistant response trigger (depends on self._connected_time)
         if self._triggering_assistant_response:
             await self._send_assistant_response_trigger()
-            self._triggering_assistant_response = False
 
     async def _disconnect(self):
         try:
@@ -1105,7 +1104,6 @@ class AWSNovaSonicLLMService(LLMService):
         # Send the trigger audio, if we're fully connected and set up
         if self._connected_time is not None:
             await self._send_assistant_response_trigger()
-            self._triggering_assistant_response = False
 
     async def _send_assistant_response_trigger(self):
         if (
@@ -1113,46 +1111,51 @@ class AWSNovaSonicLLMService(LLMService):
         ):  # should never happen
             return
 
-        logger.debug("Sending assistant response trigger...")
+        try:
+            logger.debug("Sending assistant response trigger...")
 
-        chunk_duration = 0.02  # what we might get from InputAudioRawFrame
+            chunk_duration = 0.02  # what we might get from InputAudioRawFrame
-        chunk_size = int(
+            chunk_size = int(
-            chunk_duration
+                chunk_duration
-            * self._params.input_sample_rate
+                * self._params.input_sample_rate
-            * self._params.input_channel_count
+                * self._params.input_channel_count
-            * (self._params.input_sample_size / 8)
+                * (self._params.input_sample_size / 8)
-        )  # e.g. 0.02 seconds of 16-bit (2-byte) PCM mono audio at 16kHz is 640 bytes
+            )  # e.g. 0.02 seconds of 16-bit (2-byte) PCM mono audio at 16kHz is 640 bytes
 
-        # Lead with a bit of blank audio, if needed.
+            # Lead with a bit of blank audio, if needed.
-        # It seems like the LLM can't quite "hear" the first little bit of audio sent on a
+            # It seems like the LLM can't quite "hear" the first little bit of audio sent on a
-        # connection.
+            # connection.
-        current_time = time.time()
+            current_time = time.time()
-        max_blank_audio_duration = 0.5
+            max_blank_audio_duration = 0.5
-        blank_audio_duration = (
+            blank_audio_duration = (
-            max_blank_audio_duration - (current_time - self._connected_time)
+                max_blank_audio_duration - (current_time - self._connected_time)
-            if self._connected_time is not None
+                if self._connected_time is not None
-            and (current_time - self._connected_time) < max_blank_audio_duration
+                and (current_time - self._connected_time) < max_blank_audio_duration
-            else None
+                else None
-        )
-        if blank_audio_duration:
-            logger.debug(
-                f"Leading assistant response trigger with {blank_audio_duration}s of blank audio"
             )
-            blank_audio_chunk = b"\x00" * chunk_size
+            if blank_audio_duration:
-            num_chunks = int(blank_audio_duration / chunk_duration)
+                logger.debug(
-            for _ in range(num_chunks):
+                    f"Leading assistant response trigger with {blank_audio_duration}s of blank audio"
-                await self._send_user_audio_event(blank_audio_chunk)
+                )
-                await asyncio.sleep(chunk_duration)
+                blank_audio_chunk = b"\x00" * chunk_size
+                num_chunks = int(blank_audio_duration / chunk_duration)
+                for _ in range(num_chunks):
+                    await self._send_user_audio_event(blank_audio_chunk)
+                    await asyncio.sleep(chunk_duration)
 
-        # Send trigger audio
+            # Send trigger audio
-        # NOTE: this audio *will* be transcribed and eventually make it into the context. That's OK:
+            # NOTE: this audio *will* be transcribed and eventually make it into the context. That's OK:
-        # if we ever need to seed this service again with context it would make sense to include it
+            # if we ever need to seed this service again with context it would make sense to include it
-        # since the instruction (i.e. the "wait for the trigger" instruction) will be part of the
+            # since the instruction (i.e. the "wait for the trigger" instruction) will be part of the
-        # context as well.
+            # context as well.
-        audio_chunks = [
+            audio_chunks = [
-            self._assistant_response_trigger_audio[i : i + chunk_size]
+                self._assistant_response_trigger_audio[i : i + chunk_size]
-            for i in range(0, len(self._assistant_response_trigger_audio), chunk_size)
+                for i in range(0, len(self._assistant_response_trigger_audio), chunk_size)
-        ]
+            ]
-        for chunk in audio_chunks:
+            for chunk in audio_chunks:
-            await self._send_user_audio_event(chunk)
+                await self._send_user_audio_event(chunk)
-            await asyncio.sleep(chunk_duration)
+                await asyncio.sleep(chunk_duration)
+        finally:
+            # We need to clean up in case sending the trigger was cancelled, e.g. in the case of a user interruption.
+            # (An asyncio.CancelledError would be raised in that case.)
+            self._triggering_assistant_response = False

src/pipecat/services/cartesia/tts.py

@@ -121,6 +121,7 @@ class CartesiaTTSService(AudioContextWordTTSService):
         container: str = "raw",
         params: Optional[InputParams] = None,
         text_aggregator: Optional[BaseTextAggregator] = None,
+        aggregate_sentences: Optional[bool] = True,
         **kwargs,
     ):
         """Initialize the Cartesia TTS service.
@@ -136,6 +137,7 @@ class CartesiaTTSService(AudioContextWordTTSService):
             container: Audio container format.
             params: Additional input parameters for voice customization.
             text_aggregator: Custom text aggregator for processing input text.
+            aggregate_sentences: Whether to aggregate sentences within the TTSService.
             **kwargs: Additional arguments passed to the parent service.
         """
         # Aggregating sentences still gives cleaner-sounding results and fewer
@@ -149,7 +151,7 @@ class CartesiaTTSService(AudioContextWordTTSService):
         # can use those to generate text frames ourselves aligned with the
         # playout timing of the audio!
         super().__init__(
-            aggregate_sentences=True,
+            aggregate_sentences=aggregate_sentences,
             push_text_frames=False,
             pause_frame_processing=True,
             sample_rate=sample_rate,

src/pipecat/services/elevenlabs/tts.py

@@ -238,6 +238,7 @@ class ElevenLabsTTSService(AudioContextWordTTSService):
         url: str = "wss://api.elevenlabs.io",
         sample_rate: Optional[int] = None,
         params: Optional[InputParams] = None,
+        aggregate_sentences: Optional[bool] = True,
         **kwargs,
     ):
         """Initialize the ElevenLabs TTS service.
@@ -249,6 +250,7 @@ class ElevenLabsTTSService(AudioContextWordTTSService):
             url: WebSocket URL for ElevenLabs TTS API.
             sample_rate: Audio sample rate. If None, uses default.
             params: Additional input parameters for voice customization.
+            aggregate_sentences: Whether to aggregate sentences within the TTSService.
             **kwargs: Additional arguments passed to the parent service.
         """
         # Aggregating sentences still gives cleaner-sounding results and fewer
@@ -266,7 +268,7 @@ class ElevenLabsTTSService(AudioContextWordTTSService):
         # speaking for a while, so we want the parent class to send TTSStopFrame
         # after a short period not receiving any audio.
         super().__init__(
-            aggregate_sentences=True,
+            aggregate_sentences=aggregate_sentences,
             push_text_frames=False,
             push_stop_frames=True,
             pause_frame_processing=True,

src/pipecat/services/google/llm.py

@@ -627,9 +627,9 @@ class GoogleLLMContext(OpenAILLMContext):
         # Check if we only have function-related messages (no regular text)
         has_regular_messages = any(
             len(msg.parts) == 1
-            and not getattr(msg.parts[0], "text", None)
+            and getattr(msg.parts[0], "text", None)
-            and getattr(msg.parts[0], "function_call", None)
+            and not getattr(msg.parts[0], "function_call", None)
-            and getattr(msg.parts[0], "function_response", None)
+            and not getattr(msg.parts[0], "function_response", None)
             for msg in self._messages
         )
 

src/pipecat/services/lmnt/tts.py

@@ -95,7 +95,7 @@ class LmntTTSService(InterruptibleTTSService):
         voice_id: str,
         sample_rate: Optional[int] = None,
         language: Language = Language.EN,
-        model: str = "aurora",
+        model: str = "blizzard",
         **kwargs,
     ):
         """Initialize the LMNT TTS service.
@@ -105,7 +105,7 @@ class LmntTTSService(InterruptibleTTSService):
             voice_id: ID of the voice to use for synthesis.
             sample_rate: Audio sample rate. If None, uses default.
             language: Language for synthesis. Defaults to English.
-            model: TTS model to use. Defaults to "aurora".
+            model: TTS model to use. Defaults to "blizzard".
             **kwargs: Additional arguments passed to parent InterruptibleTTSService.
         """
         super().__init__(

src/pipecat/services/mem0/memory.py

@@ -69,6 +69,7 @@ class Mem0MemoryService(FrameProcessor):
         agent_id: Optional[str] = None,
         run_id: Optional[str] = None,
         params: Optional[InputParams] = None,
+        host: Optional[str] = None,
     ):
         """Initialize the Mem0 memory service.
 
@@ -79,6 +80,7 @@ class Mem0MemoryService(FrameProcessor):
             agent_id: The agent ID to associate with memories in Mem0.
             run_id: The run ID to associate with memories in Mem0.
             params: Configuration parameters for memory retrieval and storage.
+            host: The host of the Mem0 server.
 
         Raises:
             ValueError: If none of user_id, agent_id, or run_id are provided.
@@ -92,7 +94,7 @@ class Mem0MemoryService(FrameProcessor):
         if local_config:
             self.memory_client = Memory.from_config(local_config)
         else:
-            self.memory_client = MemoryClient(api_key=api_key)
+            self.memory_client = MemoryClient(api_key=api_key, host=host)
         # At least one of user_id, agent_id, or run_id must be provided
         if not any([user_id, agent_id, run_id]):
             raise ValueError("At least one of user_id, agent_id, or run_id must be provided")

src/pipecat/services/neuphonic/tts.py

@@ -106,10 +106,11 @@ class NeuphonicTTSService(InterruptibleTTSService):
         *,
         api_key: str,
         voice_id: Optional[str] = None,
-        url: str = "wss://api.neuphonic.com",
+        url: str = "wss://eu-west-1.api.neuphonic.com",
         sample_rate: Optional[int] = 22050,
         encoding: str = "pcm_linear",
         params: Optional[InputParams] = None,
+        aggregate_sentences: Optional[bool] = True,
         **kwargs,
     ):
         """Initialize the Neuphonic TTS service.
@@ -121,10 +122,11 @@ class NeuphonicTTSService(InterruptibleTTSService):
             sample_rate: Audio sample rate in Hz. Defaults to 22050.
             encoding: Audio encoding format. Defaults to "pcm_linear".
             params: Additional input parameters for TTS configuration.
+            aggregate_sentences: Whether to aggregate sentences within the TTSService.
             **kwargs: Additional arguments passed to parent InterruptibleTTSService.
         """
         super().__init__(
-            aggregate_sentences=True,
+            aggregate_sentences=aggregate_sentences,
             push_text_frames=False,
             push_stop_frames=True,
             stop_frame_timeout_s=2.0,
@@ -279,14 +281,18 @@ class NeuphonicTTSService(InterruptibleTTSService):
                 "voice_id": self._voice_id,
             }
 
-            query_params = [f"api_key={self._api_key}"]
+            query_params = []
             for key, value in tts_config.items():
                 if value is not None:
                     query_params.append(f"{key}={value}")
 
-            url = f"{self._url}/speak/{self._settings['lang_code']}?{'&'.join(query_params)}"
+            url = f"{self._url}/speak/{self._settings['lang_code']}"
+            if query_params:
+                url += f"?{'&'.join(query_params)}"
 
-            self._websocket = await websockets.connect(url)
+            headers = {"x-api-key": self._api_key}
+
+            self._websocket = await websockets.connect(url, extra_headers=headers)
         except Exception as e:
             logger.error(f"{self} initialization error: {e}")
             self._websocket = None
@@ -311,7 +317,7 @@ class NeuphonicTTSService(InterruptibleTTSService):
         async for message in WatchdogAsyncIterator(self._websocket, manager=self.task_manager):
             if isinstance(message, str):
                 msg = json.loads(message)
-                if msg.get("data", {}).get("audio") is not None:
+                if msg.get("data") and msg["data"].get("audio"):
                     await self.stop_ttfb_metrics()
 
                     audio = base64.b64decode(msg["data"]["audio"])
@@ -324,12 +330,19 @@ class NeuphonicTTSService(InterruptibleTTSService):
         while True:
             self.reset_watchdog()
             await asyncio.sleep(KEEPALIVE_SLEEP)
-            await self._send_text("")
+            await self._send_keepalive()
+
+    async def _send_keepalive(self):
+        """Send keepalive message to maintain connection."""
+        if self._websocket:
+            # Send empty text for keepalive
+            msg = {"text": ""}
+            await self._websocket.send(json.dumps(msg))
 
     async def _send_text(self, text: str):
         """Send text to Neuphonic WebSocket for synthesis."""
         if self._websocket:
-            msg = {"text": text}
+            msg = {"text": f"{text} <STOP>"}
             logger.debug(f"Sending text to websocket: {msg}")
             await self._websocket.send(json.dumps(msg))
 

src/pipecat/services/ollama/llm.py

@@ -6,6 +6,8 @@
 
 """OLLama LLM service implementation for Pipecat AI framework."""
 
+from loguru import logger
+
 from pipecat.services.openai.llm import OpenAILLMService
 
 
@@ -16,12 +18,28 @@ class OLLamaLLMService(OpenAILLMService):
     providing a compatible interface for running large language models locally.
     """
 
-    def __init__(self, *, model: str = "llama2", base_url: str = "http://localhost:11434/v1"):
+    def __init__(
+        self, *, model: str = "llama2", base_url: str = "http://localhost:11434/v1", **kwargs
+    ):
         """Initialize OLLama LLM service.
 
         Args:
             model: The OLLama model to use. Defaults to "llama2".
             base_url: The base URL for the OLLama API endpoint.
                     Defaults to "http://localhost:11434/v1".
+            **kwargs: Additional keyword arguments passed to OpenAILLMService.
         """
-        super().__init__(model=model, base_url=base_url, api_key="ollama")
+        super().__init__(model=model, base_url=base_url, api_key="ollama", **kwargs)
+
+    def create_client(self, base_url=None, **kwargs):
+        """Create OpenAI-compatible client for Ollama.
+
+        Args:
+            base_url: The base URL for the API. If None, uses instance base_url.
+            **kwargs: Additional keyword arguments passed to the parent create_client method.
+
+        Returns:
+            An OpenAI-compatible client configured for Ollama.
+        """
+        logger.debug(f"Creating Ollama client with api {base_url}")
+        return super().create_client(base_url, **kwargs)

src/pipecat/services/rime/tts.py

@@ -99,6 +99,7 @@ class RimeTTSService(AudioContextWordTTSService):
         sample_rate: Optional[int] = None,
         params: Optional[InputParams] = None,
         text_aggregator: Optional[BaseTextAggregator] = None,
+        aggregate_sentences: Optional[bool] = True,
         **kwargs,
     ):
         """Initialize Rime TTS service.
@@ -111,11 +112,12 @@ class RimeTTSService(AudioContextWordTTSService):
             sample_rate: Audio sample rate in Hz.
             params: Additional configuration parameters.
             text_aggregator: Custom text aggregator for processing input text.
+            aggregate_sentences: Whether to aggregate sentences within the TTSService.
             **kwargs: Additional arguments passed to parent class.
         """
         # Initialize with parent class settings for proper frame handling
         super().__init__(
-            aggregate_sentences=True,
+            aggregate_sentences=aggregate_sentences,
             push_text_frames=False,
             push_stop_frames=True,
             pause_frame_processing=True,

src/pipecat/services/riva/stt.py

@@ -279,7 +279,6 @@ class RivaSTTService(STTService):
             streaming_config=self._config,
         )
         for response in responses:
-            self.reset_watchdog()
             if not response.results:
                 continue
             asyncio.run_coroutine_threadsafe(

src/pipecat/services/soniox/stt.py

@@ -0,0 +1,396 @@
+#
+# Copyright (c) 2024–2025, Daily
+#
+# SPDX-License-Identifier: BSD 2-Clause License
+#
+
+"""Soniox speech-to-text service implementation."""
+
+import asyncio
+import json
+import time
+from typing import AsyncGenerator, List, Optional
+
+from loguru import logger
+from pydantic import BaseModel
+
+from pipecat.frames.frames import (
+    CancelFrame,
+    EndFrame,
+    ErrorFrame,
+    Frame,
+    InterimTranscriptionFrame,
+    StartFrame,
+    TranscriptionFrame,
+    UserStoppedSpeakingFrame,
+)
+from pipecat.processors.frame_processor import FrameDirection
+from pipecat.services.stt_service import STTService
+from pipecat.transcriptions.language import Language
+from pipecat.utils.time import time_now_iso8601
+from pipecat.utils.tracing.service_decorators import traced_stt
+
+try:
+    import websockets
+except ModuleNotFoundError as e:
+    logger.error(f"Exception: {e}")
+    logger.error("In order to use Soniox, you need to `pip install pipecat-ai[soniox]`.")
+    raise Exception(f"Missing module: {e}")
+
+
+KEEPALIVE_MESSAGE = '{"type": "keepalive"}'
+
+FINALIZE_MESSAGE = '{"type": "finalize"}'
+
+END_TOKEN = "<end>"
+
+FINALIZED_TOKEN = "<fin>"
+
+
+class SonioxInputParams(BaseModel):
+    """Real-time transcription settings.
+
+    See Soniox WebSocket API documentation for more details:
+    https://soniox.com/docs/speech-to-text/api-reference/websocket-api#configuration-parameters
+
+    Parameters:
+        model: Model to use for transcription.
+        audio_format: Audio format to use for transcription.
+        num_channels: Number of channels to use for transcription.
+        language_hints: List of language hints to use for transcription.
+        context: Customization for transcription.
+        enable_non_final_tokens: Whether to enable non-final tokens. If false, only final tokens will be returned.
+        max_non_final_tokens_duration_ms: Maximum duration of non-final tokens.
+        client_reference_id: Client reference ID to use for transcription.
+    """
+
+    model: str = "stt-rt-preview"
+
+    audio_format: Optional[str] = "pcm_s16le"
+    num_channels: Optional[int] = 1
+
+    language_hints: Optional[List[Language]] = None
+    context: Optional[str] = None
+
+    enable_non_final_tokens: Optional[bool] = True
+    max_non_final_tokens_duration_ms: Optional[int] = None
+
+    client_reference_id: Optional[str] = None
+
+
+def is_end_token(token: dict) -> bool:
+    """Determine if a token is an end token."""
+    return token["text"] == END_TOKEN or token["text"] == FINALIZED_TOKEN
+
+
+def language_to_soniox_language(language: Language) -> str:
+    """Pipecat Language enum uses same ISO 2-letter codes as Soniox, except with added regional variants.
+
+    For a list of all supported languages, see: https://soniox.com/docs/speech-to-text/core-concepts/supported-languages
+    """
+    lang_str = str(language.value).lower()
+    if "-" in lang_str:
+        return lang_str.split("-")[0]
+    return lang_str
+
+
+def _prepare_language_hints(
+    language_hints: Optional[List[Language]],
+) -> Optional[List[str]]:
+    if language_hints is None:
+        return None
+
+    prepared_languages = [language_to_soniox_language(lang) for lang in language_hints]
+    # Remove duplicates (in case of language_hints with multiple regions).
+    return list(set(prepared_languages))
+
+
+class SonioxSTTService(STTService):
+    """Speech-to-Text service using Soniox's WebSocket API.
+
+    This service connects to Soniox's WebSocket API for real-time transcription
+    with support for multiple languages, custom context, speaker diarization,
+    and more.
+
+    For complete API documentation, see: https://soniox.com/docs/speech-to-text/api-reference/websocket-api
+    """
+
+    def __init__(
+        self,
+        *,
+        api_key: str,
+        url: str = "wss://stt-rt.soniox.com/transcribe-websocket",
+        sample_rate: Optional[int] = None,
+        params: Optional[SonioxInputParams] = None,
+        vad_force_turn_endpoint: bool = False,
+        **kwargs,
+    ):
+        """Initialize the Soniox STT service.
+
+        Args:
+            api_key: Soniox API key.
+            url: Soniox WebSocket API URL.
+            sample_rate: Audio sample rate.
+            params: Additional configuration parameters, such as language hints, context and
+                speaker diarization.
+            vad_force_turn_endpoint: Listen to `UserStoppedSpeakingFrame` to send finalize message to Soniox. If disabled, Soniox will detect the end of the speech.
+            **kwargs: Additional arguments passed to the STTService.
+        """
+        super().__init__(sample_rate=sample_rate, **kwargs)
+        params = params or SonioxInputParams()
+
+        self._api_key = api_key
+        self._url = url
+        self.set_model_name(params.model)
+        self._params = params
+        self._vad_force_turn_endpoint = vad_force_turn_endpoint
+        self._websocket = None
+
+        self._final_transcription_buffer = []
+        self._last_tokens_received: Optional[float] = None
+
+        self._receive_task = None
+        self._keepalive_task = None
+
+    async def start(self, frame: StartFrame):
+        """Start the Soniox STT websocket connection.
+
+        Args:
+            frame: The start frame containing initialization parameters.
+        """
+        await super().start(frame)
+        if self._websocket:
+            return
+
+        self._websocket = await websockets.connect(self._url)
+
+        if not self._websocket:
+            logger.error(f"Unable to connect to Soniox API at {self._url}")
+
+        # If vad_force_turn_endpoint is not enabled, we need to enable endpoint detection.
+        # Either one or the other is required.
+        enable_endpoint_detection = not self._vad_force_turn_endpoint
+
+        # Send the initial configuration message.
+        config = {
+            "api_key": self._api_key,
+            "model": self._model_name,
+            "audio_format": self._params.audio_format,
+            "num_channels": self._params.num_channels or 1,
+            "enable_endpoint_detection": enable_endpoint_detection,
+            "sample_rate": self.sample_rate,
+            "language_hints": _prepare_language_hints(self._params.language_hints),
+            "context": self._params.context,
+            "enable_non_final_tokens": self._params.enable_non_final_tokens,
+            "max_non_final_tokens_duration_ms": self._params.max_non_final_tokens_duration_ms,
+            "client_reference_id": self._params.client_reference_id,
+        }
+
+        # Send the configuration message.
+        await self._websocket.send(json.dumps(config))
+
+        if self._websocket and not self._receive_task:
+            self._receive_task = self.create_task(self._receive_task_handler())
+        if self._websocket and not self._keepalive_task:
+            self._keepalive_task = self.create_task(self._keepalive_task_handler())
+
+    async def _cleanup(self):
+        if self._keepalive_task:
+            await self.cancel_task(self._keepalive_task)
+            self._keepalive_task = None
+
+        if self._websocket:
+            await self._websocket.close()
+            self._websocket = None
+
+        if self._receive_task:
+            # Task cannot cancel itself. If task called _cleanup() we expect it to cancel itself.
+            if self._receive_task != asyncio.current_task():
+                await self.wait_for_task(self._receive_task)
+            self._receive_task = None
+
+    async def stop(self, frame: EndFrame):
+        """Stop the Soniox STT websocket connection.
+
+        Stopping waits for the server to close the connection as we might receive
+        additional final tokens after sending the stop recording message.
+
+        Args:
+            frame: The end frame.
+        """
+        await super().stop(frame)
+        await self._send_stop_recording()
+
+    async def cancel(self, frame: CancelFrame):
+        """Cancel the Soniox STT websocket connection.
+
+        Compared to stop, this method closes the connection immediately without waiting
+        for the server to close it. This is useful when we want to stop the connection
+        immediately without waiting for the server to send any final tokens.
+
+        Args:
+            frame: The cancel frame.
+        """
+        await super().cancel(frame)
+        await self._cleanup()
+
+    async def run_stt(self, audio: bytes) -> AsyncGenerator[Frame, None]:
+        """Send audio data to Soniox STT Service.
+
+        Args:
+            audio: Raw audio bytes to transcribe.
+
+        Yields:
+            Frame: None (transcription results come via WebSocket callbacks).
+        """
+        await self.start_processing_metrics()
+        if self._websocket and not self._websocket.closed:
+            await self._websocket.send(audio)
+        await self.stop_processing_metrics()
+
+        yield None
+
+    @traced_stt
+    async def _handle_transcription(
+        self, transcript: str, is_final: bool, language: Optional[Language] = None
+    ):
+        """Handle a transcription result with tracing."""
+        pass
+
+    async def process_frame(self, frame: Frame, direction: FrameDirection):
+        """Processes a frame of audio data, either buffering or transcribing it.
+
+        Args:
+            frame: The frame to process.
+            direction: The direction of frame processing.
+        """
+        await super().process_frame(frame, direction)
+
+        if isinstance(frame, UserStoppedSpeakingFrame) and self._vad_force_turn_endpoint:
+            # Send finalize message to Soniox so we get the final tokens asap.
+            if self._websocket and not self._websocket.closed:
+                await self._websocket.send(FINALIZE_MESSAGE)
+                logger.debug(f"Triggered finalize event on: {frame.name=}, {direction=}")
+
+    async def _send_stop_recording(self):
+        """Send stop recording message to Soniox."""
+        if self._websocket and not self._websocket.closed:
+            # Send stop recording message
+            await self._websocket.send("")
+
+    async def _keepalive_task_handler(self):
+        """Connection has to be open all the time."""
+        try:
+            while True:
+                logger.debug("Sending keepalive message")
+                if self._websocket and not self._websocket.closed:
+                    await self._websocket.send(KEEPALIVE_MESSAGE)
+                else:
+                    logger.debug("WebSocket connection closed.")
+                    break
+                await asyncio.sleep(5)
+
+        except websockets.exceptions.ConnectionClosed:
+            # Expected when closing the connection
+            logger.debug("WebSocket connection closed, keepalive task stopped.")
+        except Exception as e:
+            logger.error(f"{self} error (_keepalive_task_handler): {e}")
+            await self.push_error(ErrorFrame(f"{self} error (_keepalive_task_handler): {e}"))
+
+    async def _receive_task_handler(self):
+        if not self._websocket:
+            return
+
+        # Transcription frame will be only sent after we get the "endpoint" event.
+        self._final_transcription_buffer = []
+
+        async def send_endpoint_transcript():
+            if self._final_transcription_buffer:
+                text = "".join(map(lambda token: token["text"], self._final_transcription_buffer))
+                await self.push_frame(
+                    TranscriptionFrame(
+                        text=text,
+                        user_id=self._user_id,
+                        timestamp=time_now_iso8601(),
+                        result=self._final_transcription_buffer,
+                    )
+                )
+                await self._handle_transcription(text, is_final=True)
+                await self.stop_processing_metrics()
+                self._final_transcription_buffer = []
+
+        try:
+            async for message in self._websocket:
+                content = json.loads(message)
+
+                tokens = content["tokens"]
+
+                if tokens:
+                    if len(tokens) == 1 and tokens[0]["text"] == FINALIZED_TOKEN:
+                        # Ignore finalized token, prevent auto-finalize cycling.
+                        pass
+                    else:
+                        # Got at least one token, so we can reset the auto finalize delay.
+                        self._last_tokens_received = time.time()
+
+                # We will only send the final tokens after we get the "endpoint" event.
+                non_final_transcription = []
+
+                for token in tokens:
+                    if token["is_final"]:
+                        if is_end_token(token):
+                            # Found an endpoint, tokens until here will be sent as transcript,
+                            # the rest will be sent as interim tokens (even final tokens).
+                            await send_endpoint_transcript()
+                        else:
+                            self._final_transcription_buffer.append(token)
+                    else:
+                        non_final_transcription.append(token)
+
+                if self._final_transcription_buffer or non_final_transcription:
+                    final_text = "".join(
+                        map(lambda token: token["text"], self._final_transcription_buffer)
+                    )
+                    non_final_text = "".join(
+                        map(lambda token: token["text"], non_final_transcription)
+                    )
+
+                    await self.push_frame(
+                        InterimTranscriptionFrame(
+                            # Even final tokens are sent as interim tokens as we want to send
+                            # nicely formatted messages - therefore waiting for the endpoint.
+                            text=final_text + non_final_text,
+                            user_id=self._user_id,
+                            timestamp=time_now_iso8601(),
+                            result=self._final_transcription_buffer + non_final_transcription,
+                        )
+                    )
+
+                error_code = content.get("error_code")
+                error_message = content.get("error_message")
+                if error_code or error_message:
+                    # In case of error, still send the final transcript (if any remaining in the buffer).
+                    await send_endpoint_transcript()
+                    logger.error(
+                        f"{self} error: {error_code} (_receive_task_handler) - {error_message}"
+                    )
+                    await self.push_error(
+                        ErrorFrame(
+                            f"{self} error: {error_code} (_receive_task_handler) - {error_message}"
+                        )
+                    )
+
+                finished = content.get("finished")
+                if finished:
+                    # When finished, still send the final transcript (if any remaining in the buffer).
+                    await send_endpoint_transcript()
+                    logger.debug("Transcription finished.")
+                    await self._cleanup()
+                    return
+
+        except websockets.exceptions.ConnectionClosed:
+            # Expected when closing the connection.
+            pass
+        except Exception as e:
+            logger.error(f"{self} error: {e}")
+            await self.push_error(ErrorFrame(f"{self} error: {e}"))