Voice-Enabled AI with Pipecat, Twilio, and Freeplay

Introduction

Voice-enabled AI applications present unique challenges when it comes to testing, monitoring, and iterating on your prompts and models. This guide demonstrates how Freeplay’s observability and prompt management tools can support your development workflow when building voice applications. In this example, we show how to use Freeplay together with Pipecat and Twilio.

What is Pipecat?

Pipecat is a powerful open source framework for building voice-enabled, real-time, multimodal AI applications. When paired with Twilio for real-time voice over the phone, Pipecat enables teams to quickly build audio-based agentic systems that combine both user and bot audio with LLM interactions. This combination creates a strong foundation for the core application, but building a high-quality generative AI product also requires robust monitoring, evaluation, and continuous experimentation. This is where Freeplay helps.

Using Freeplay for Rapid Iteration and Observability

When it comes to monitoring and improving a voice agent, teams often struggle with:

Multi-modal Observability: Tracking and analyzing model inputs and outputs across different data types (audio, text, images, files, etc.)
Quality Evaluation: Understanding how your application performs in real user scenarios and using evaluation criteria relevant to your product
Experimentation & Iteration: Systematically versioning, testing, and deploying changes to prompts, tools, and/or models
Team Collaboration: Keeping all team members on the same page when it comes to testing and understanding quality (including non-developers)

Freeplay addresses these challenges by providing a comprehensive solution for prompt and model management, observability, and evaluation that works seamlessly across modalities/data formats — including audio. And Freeplay makes it easy for both technical and non-technical team members to fully participate in the product development and optimization process.

What You’ll Be Able to Monitor

Once implemented, you’ll be able to view complete user interactions in Freeplay, including:

Audio recordings from the user and bot turns
Transcribed text for easy review and analysis
LLM responses with full context
Cost & latency metrics for performance optimization
Evaluation results against your quality criteria

Once implemented, you’ll be able to view complete user interactions in Freeplay, including:

Audio recordings
Transcribed text
LLM responses
Cost & latency metrics
Evaluation results

Integration Approaches

Freeplay provides seamless integration with Pipecat to log audio interactions and LLM responses for comprehensive testing and evaluation of your voice agents.

Option 1: Processor Integration

How it works: Directly intercepts frames within the pipeline processing flow
Trade-off: Adds minimal latency as processing happens inline
Best for: Cases where you need direct frame manipulation or synchronous processing
Documentation: Pipecat Processors
Full Example: FreeplayProcessor

Option 2: Observer Integration ⭐ Recommended

How it works: Uses callbacks to log data asynchronously in the background
Trade-off: Zero impact on pipeline latency since logging happens outside the main flow
Best for: Voice agents where low latency is critical
Documentation: Pipecat Observer Pattern
Full Example: FreeplayObserver

We recommend the Observer pattern: Since latency is crucial for voice agents, the Observer approach ensures your audio pipeline runs at maximum speed while still capturing all necessary data for Freeplay.

Conversation Flow in a Pipecat + Twilio + Freeplay Integration

The above image shows how Freeplay integrates into the pipeline. Regardless if you are using the Observer or Processor methods, the logic is very similar. Once both user and bot audio is received and complete, a turn can be logged to Freeplay. This flow ensures comprehensive logging while maintaining optimal performance for real-time voice interactions.

Implementation Guide FreeplayObserver

Pro Tip: AudioBufferProcessor

We highly recommend using Pipecat’s AudioBufferProcessor alongside the FreeplayObserver. This well-tested utility:

Formats audio data consistently for logging
Provides reliable callbacks for conversation turn detection
Simplifies audio handling and storage

This implementation guide uses the Twilio-Chatbot example provided by Pipecat—a chatbot accessed via phone application powered by Twilio. This integration allows you to log audio interactions and LLM responses to Freeplay for comprehensive testing and evaluation of your audio agents.

Prerequisites

Before starting, make sure you have:

A Freeplay account set up (follow our quick start guide)
Your prompts configured in Freeplay (follow our prompting guide)
A working Pipecat + Twilio application

For a complete code example, see our FreeplayObserver Pipecat Recipe.

Step 1: Import Prompt Configuration from Freeplay

First, fetch your prompt configuration from Freeplay and prepare it for use in your Pipecat pipeline:

from helpers.freeplay_frame import FreeplayProcessor  # Processor example
from helpers.freeplay_observer import FreeplayObserver  # Observer example
from freeplay import Freeplay, SessionInfo

# Freeplay Client

fp_client = Freeplay(
freeplay_api_key=os.getenv("FREEPLAY_API_KEY"),
api_base=os.getenv("FREEPLAY_API_BASE")
)

    # Get the unformatted prompt from Freeplay
    unformatted_prompt = fp_client.prompts.get(
        project_id=os.getenv("FREEPLAY_PROJECT_ID"),
        template_name=os.getenv("PROMPT_NAME"),
        environment="latest",
    )
    formatted_prompt = unformatted_prompt.bind(
        variables=<optional vars>,
        history=[],
    ).format()

    # Pass the formatted prompt to the LLM
    llm = OpenAILLMService(model=formatted_prompt.prompt_info.model,
                           tools=formatted_prompt.tool_schema if formatted_prompt.tool_schema else None,
                           api_key=os.getenv("OPENAI_API_KEY"),
                           **formatted_prompt.prompt_info.model_parameters)

Why bind the prompt? This approach allows you to fetch the prompt once and reuse it throughout the conversation, avoiding repeated API calls to Freeplay while still allowing you to add new variables and information at each conversation turn.

Step 2: Create Your Freeplay Observer

Create a FreeplayObserver to handle conversation memory, frame monitoring, and data logging. See the full code here.

# Note: see the full implementation in our Recipe section of the docs.
class FreeplayObserver(BaseObserver):
    def __init__(
        self,
        fp_client: Freeplay,
        unformatted_prompt: str = None,
        template_name: str = os.getenv("PROMPT_NAME") or None,
        environment: str = "latest",
        variables: dict = {},
    ):
        super().__init__()
        self.start_llm_interaction = 0
        self.end_llm_interaction = 0
        self.llm_completion_latency = 0
        self.call_id = str(uuid.uuid4())  # Has to be str to record to Freeplay

        # Audio related properties
        self.sample_width = 2
        self.num_channels = 1
        self.sample_rate = 16000
        self._bot_audio = bytearray()
        self._user_audio = bytearray()
        self._turn_user_audio = bytearray()
        self.user_speaking = False
        self.bot_speaking = False
        self.fp_client = fp_client
        self.session = self.fp_client.sessions.create()

        # Freeplay Params
        self.template_name = template_name
        self.environment = environment
        self.unformatted_prompt = unformatted_prompt
        self.variables = variables
        # Conversation Params
        self.conversation_id = self._new_conv_id()
        self.conversation_history = []
        self.most_recent_user_message = None
        self.most_recent_completion = None

    def _new_conv_id(self) -> str:
        """Generate a new conversation ID based on the current timestamp (this represents a customer id or similar)."""
        return datetime.datetime.now().strftime("%Y%m%d_%H%M%S")

    def _reset_recent_messages(self):
        """Reset all temporary message and audio storage."""
        self.most_recent_user_message = None
        self.most_recent_completion = None
        self._user_audio = bytearray()
        self._bot_audio = bytearray()
        self._turn_user_audio = bytearray()
        # self.llm_completion_latency = 0

    async def record_to_freeplay(self):
        """Record the current interaction to Freeplay as a new trace."""
        # Create a new trace for this interaction
        trace = self.session.create_trace(
            input=self.most_recent_user_message,
            custom_metadata={
                "conversation_id": str(self.conversation_id),
            },
        )

        # Add user message to conversation history
        self.conversation_history.append(
            {
                "role": "user",
                "content": [
                    {"type": "text", "text": self.most_recent_user_message},
                    {
                        "type": "input_audio",
                        "input_audio": {
                            "data": base64.b64encode(self._user_audio).decode("utf-8"),
                            "format": "wav",
                        },
                    },
                ],
            },
        )

        # Bind the variables to the prompt
        if self.unformatted_prompt:
            formatted = self.unformatted_prompt.bind(
                variables=self.variables,
                history=self.conversation_history,
            ).format()
        else:
            # Run in executor to avoid blocking
            loop = asyncio.get_running_loop()
            formatted = await loop.run_in_executor(
                None,
                self.fp_client.prompts.get_formatted,
                project_id=os.getenv("FREEPLAY_PROJECT_ID"),
                template_name=self.template_name,
                environment=self.environment,
                history=self.conversation_history,
                variables=self.variables,
            )

        # Calculate latency for the LLM interaction
        # Convert nanoseconds to seconds for proper timing
        latency_seconds = self.get_llm_response_latency_seconds()
        end = time.time()
        start = end - latency_seconds
        try:
            print(f"_____* self._bot_audio: {len(self._bot_audio)}")

            # Prepare metadata and record payload
            custom_metadata = {"caller_id": self.call_id}

            # Prepare assistants response message (mimicing the format of the llm provider message)
            assistant_msg = {
                "role": "assistant",
                "content": [
                    {"type": "text", "text": self.most_recent_completion},
                ],
                "audio": {
                    "id": self.conversation_id,
                    "data": base64.b64encode(self._bot_audio).decode("utf-8"),
                    "expires_at": 1729234747,
                    "transcript": self.most_recent_completion,
                },
            }

            # Add assistant's response to conversation history
            self.conversation_history.append(assistant_msg)

            record = RecordPayload(
                all_messages=[
                    *formatted.llm_prompt,
                    assistant_msg,  # Add the assistant's response to the record call
                ],
                session_info=SessionInfo(
                    self.session.session_id, custom_metadata=custom_metadata
                ),
                inputs={},
                prompt_info=formatted.prompt_info,
                call_info=CallInfo.from_prompt_info(formatted.prompt_info, start, end),
                trace_info=trace,
            )

            # Create recording in Freeplay
            loop = asyncio.get_running_loop()
            await loop.run_in_executor(None, self.fp_client.recordings.create, record)

            # Record output to trace

            await loop.run_in_executor(
                None,
                functools.partial(
                    trace.record_output,
                    project_id=os.getenv("FREEPLAY_PROJECT_ID"),
                    output=self.most_recent_completion,
                    eval_results={},
                ),
            )

            print(
                f"✅ Recorded interaction #{len(self.conversation_history) // 2} to Freeplay - LLM response time: {self.get_llm_response_latency_seconds():.3f}s",
                flush=True,
            )

            # Reset only audio and current message data, keep conversation history
            self._reset_recent_messages()

        except Exception as e:
            print(f"❌ Error recording to Freeplay: {e}", flush=True)
            # Still reset audio buffers to prevent accumulation
            # audio buffers are overwritten in event handler
            self._reset_recent_messages()

    async def make_wav_bytes(
        self, pcm: bytes, sample_rate: int, voice: str, prepend_silence_secs: int = 1
    ) -> bytes:
        """Convert PCM audio data to WAV format with optional silence prepend."""
        if prepend_silence_secs > 0:
            silence_samples = int(
                self.sample_rate
                * self.sample_width
                * self.num_channels
                * prepend_silence_secs
            )
            silence = b"\x00" * silence_samples
            pcm = silence + pcm
        with io.BytesIO() as buf:
            with wave.open(buf, "wb") as wf:
                wf.setnchannels(self.num_channels)
                wf.setsampwidth(self.sample_width)
                wf.setframerate(sample_rate)
                wf.writeframes(pcm)
            return buf.getvalue()

    async def on_push_frame(self, data: FramePushed):
        src = data.source
        dst = data.destination
        frame = data.frame
        direction = data.direction
        timestamp = data.timestamp

        # Create direction arrow
        arrow = "→" if direction == FrameDirection.DOWNSTREAM else "←"

        if isinstance(frame, LLMFullResponseStartFrame):
            print(f"LLMFullResponseFrame: START {src} {arrow} {dst}", flush=True)
        elif isinstance(frame, LLMFullResponseEndFrame):
            print(f"LLMFullResponseFrame: END {src} {arrow} {dst}", flush=True)
        elif isinstance(frame, TranscriptionFrame):
            # Capture user if bot talks first
            if self.most_recent_user_message is None:
                self.most_recent_user_message = frame.text
        elif isinstance(frame, OpenAILLMContextFrame):
            messages = frame.context.messages
            # Extract user message and completion from context
            # NOTE: this replaces the TranscriptionFrame results, as this maps excatly what the llm recived.
            user_messages = [m for m in messages if m.get("role") == "user"]
            if user_messages:
                self.most_recent_user_message = user_messages[-1].get("content")
            completions = [m for m in messages if m.get("role") == "assistant"]
            if completions:
                self.most_recent_completion = completions[-1].get("content")

        # Get relevant latency metrics for the LLM interaction
        if (
            isinstance(frame, OpenAILLMContextFrame)
            and isinstance(src, LLMUserContextAggregator)
            and isinstance(dst, BaseOpenAILLMService)
        ):
            self.start_llm_interaction = timestamp
            print(f"_____freeplay-observer.py OpenAILLMContextFrame START: {timestamp}")
        elif isinstance(frame, LLMFullResponseEndFrame) and isinstance(
            src, BaseOpenAILLMService
        ):
            self.end_llm_interaction = timestamp

            # update latency tally
            self.llm_completion_latency = (
                self.end_llm_interaction - self.start_llm_interaction
            )
            print(
                f"_____freeplay-observer.py * set self.llm_completion_latency: {self.llm_completion_latency} ({self.get_llm_response_latency_seconds():.3f}s)"
            )

    def get_llm_response_latency_seconds(self):
        """Convert the raw nanosecond LLM response latency to seconds."""
        return self.llm_completion_latency / 1_000_000_000

Step 3: Configure Your Pipeline with Audio Buffering

Set up your Pipecat pipeline with the FreeplayObserver and audio buffering capabilities:

from pipecat.pipeline.task import PipelineTask
from pipecat.pipeline.pipeline import Pipeline
from pipecat.processors.audio.audio_buffer import AudioBufferProcessor

# Create audio buffer for capturing conversation audio

audiobuffer = AudioBufferProcessor(
sample_rate=8000,
num_channels=1,
)

# Configure pipeline task with observer

task = PipelineTask(
pipeline,
params=PipelineParams(
audio_in_sample_rate=8000,
audio_out_sample_rate=8000,
allow_interruptions=True,
enable_metrics=True,
enable_usage_metrics=True,
),
observers=[freeplay_observer], # Add observer here
)

Step 4: Set Up Audio Capture Callbacks & Configure Pipeline

Configure callbacks to capture audio at the optimal moments:

from helpers.freeplay_observer import FreeplayObserver  # Observer example

freeplay_observer = FreeplayObserver(
  fp_client=fp_client,
  unformatted_prompt=unformatted_prompt,
  environment=os.getenv("FREEPLAY_ENVIRONMENT"),
)

#....Additional Pipeline Configuration...

task = PipelineTask(
        pipeline,
        params=PipelineParams(
            audio_in_sample_rate=8000,
            audio_out_sample_rate=8000,
            allow_interruptions=True,
            enable_metrics=True,
            enable_usage_metrics=True,  # This is used to track the usage of the LLM
        ),
        observers=[
            freeplay_observer
        ],  # Use the FreeplayObserver to record the audio to Freeplay
    )

    # save audio bytes from user and store in freeplay_observer
    @audiobuffer.event_handler("on_user_turn_audio_data")
    async def on_user_turn_audio_data(buffer, audio, sample_rate, num_channels):
        if audio and not freeplay_observer._bot_audio:
            # aggregate user audio because this event could fire multiple times
            # before bot responds
            freeplay_observer._user_audio = freeplay_observer._turn_user_audio.extend(
                audio
            )
            freeplay_observer._user_audio = await freeplay_observer.make_wav_bytes(
                freeplay_observer._turn_user_audio,
                sample_rate,
                "user",
                prepend_silence_secs=1,
            )
        elif audio and freeplay_observer._bot_audio:
            freeplay_observer._user_audio = freeplay_observer._turn_user_audio.extend(
                audio
            )
            freeplay_observer._user_audio = await freeplay_observer.make_wav_bytes(
                freeplay_observer._turn_user_audio,
                sample_rate,
                "user",
                prepend_silence_secs=1,
            )
            await freeplay_observer.record_to_freeplay()

    # save audio bytes from bot and store in freeplay_observer
    @audiobuffer.event_handler("on_bot_turn_audio_data")
    async def on_bot_turn_audio_data(buffer, audio, sample_rate, num_channels):
        # this assumes the user always speaks first and would cut off
        # the first turn of the bot
        if audio and not freeplay_observer._user_audio:
            # aggregate bot audio because this event could fire multiple times
            # before user responds
            freeplay_observer._bot_audio = await freeplay_observer.make_wav_bytes(
                audio, sample_rate, "bot", prepend_silence_secs=1
            )
        elif audio and freeplay_observer._user_audio:
            freeplay_observer._bot_audio = await freeplay_observer.make_wav_bytes(
                audio, sample_rate, "bot", prepend_silence_secs=1
            )
            await freeplay_observer.record_to_freeplay()

Add the FreeplayObserver to Your Pipeline
Initialize at Conversation Level The FreeplayObserver must be initialized at the conversation level to properly track the entire interaction flow.
Automatic Frame Processing As audio frames pass through the observer’s on_push_frame method, it automatically updates the processor variables with both user and bot audio data and metadata.
Recording with AudioBufferProcessor Callbacks To determine the optimal timing for recording to Freeplay, we recommend using AudioBufferProcessor callbacks:
1. on_bot_turn_audio_data - Captures when the bot completes its audio response
2. on_user_turn_audio_data - Captures when the user finishes speaking

These callbacks provide the most reliable trigger points for logging complete conversation turns.

Alternative: FreeplayProcessor Integration

Step 1: Import Your Prompt from Freeplay & Pass to the LLMProcessor

Note, here we get an unformatted prompt from Freeplay and then bind it, this allows us to pass it to the system and not have to make repeated calls to retrieve the llm prompt from Freeplay. The binding allows us to add new variables and information at each turn of the conversation. You can see morehere.

from helpers.freeplay_frame import FreeplayProcessor  # Processor example
from helpers.freeplay_observer import FreeplayObserver  # Observer example
from freeplay import Freeplay, SessionInfo

# Freeplay Client

fp_client = Freeplay(
freeplay_api_key=os.getenv("FREEPLAY_API_KEY"),
api_base=os.getenv("FREEPLAY_API_BASE")
)

    # Get the unformatted prompt from Freeplay
    unformatted_prompt = fp_client.prompts.get(
        project_id=os.getenv("FREEPLAY_PROJECT_ID"),
        template_name=os.getenv("PROMPT_NAME"),
        environment="latest",
    )
    formatted_prompt = unformatted_prompt.bind(
        variables=<optional vars>,
        history=[],
    ).format()

    # Pass the formatted prompt to the LLM
    llm = OpenAILLMService(model=formatted_prompt.prompt_info.model,
                           tools=formatted_prompt.tool_schema if formatted_prompt.tool_schema else None,
                           api_key=os.getenv("OPENAI_API_KEY"),
                           **formatted_prompt.prompt_info.model_parameters)

Step 2: Create a Freeplay Processor

The processor handles the memory of the conversation, processing of key frames, and and keeps track of information to log to Freeplay. This inherits from FrameProcessor in Pipecat. See the full code implementation here.

class FreeplayProcessor(FrameProcessor):
    """Logs LLM interactions and audio to Freeplay with simplified structure."""

    def __init__(
        self,
        fp_client: Freeplay,
        template_name: str,
        session: SessionInfo = None,
        required_information: str = None,
        unformatted_prompt: PromptInfo = None,
    ):
        super().__init__()
        self.fp_client = fp_client
        self.template_name = template_name
        self.conversation_id = self._new_conv_id()
        self.total_completion_time = 0
        self.required_information = required_information
        self.deepgram_latency = 0

        # Audio related properties
        self.sample_width = 2
        self.sample_rate = 8000
        self.num_channels = 1
        self._user_audio = bytearray()
        self._bot_audio = bytearray()
        self.user_speaking = False
        self.bot_speaking = False

        # Freeplay related properties
        self.conversation_history = []
        self.session = session
        self.most_recent_user_message = None
        self.most_recent_completion = None
        self.unformatted_prompt = unformatted_prompt
        self.reset_recent_messages()

    def _new_conv_id(self) -> str:
        """Generate a new conversation ID based on the current timestamp (this represents a customer id or similar)."""
        return datetime.datetime.now().strftime("%Y%m%d_%H%M%S")

    def reset_recent_messages(self):
        """Reset all temporary message and audio storage."""
        self.most_recent_user_message = None
        self.most_recent_completion = None
        self._user_audio = bytearray()
        self._bot_audio = bytearray()
        self.total_completion_time = 0
        self.deepgram_latency = 0

    async def process_frame(self, frame: Frame, direction: FrameDirection):
        """Process incoming frames and handle Freeplay logging."""
        await super().process_frame(frame, direction)

        # Handle LLM response frames
        if isinstance(frame, (LLMFullResponseStartFrame, LLMFullResponseEndFrame)):
            event = "START" if isinstance(frame, LLMFullResponseStartFrame) else "END"
            print(f"LLMFullResponseFrame: {event}", flush=True)

        # Handle LLM context frame - this is where we log to Freeplay
        elif isinstance(frame, OpenAILLMContextFrame):
            messages = frame.context.messages
            # Extract user message and completion from context
            user_messages = [m for m in messages if m.get("role") == "user"]
            if user_messages:
                self.most_recent_user_message = user_messages[-1].get("content")

            completions = [m for m in messages if m.get("role") == "assistant"]
            if completions:
                self.most_recent_completion = completions[-1].get("content")

            # Log to Freeplay when we have both user input and completion
            if self.most_recent_user_message and self.most_recent_completion:
                self._record_to_freeplay()
        # Handle audio state changes
        elif isinstance(frame, UserStartedSpeakingFrame):
            self.user_speaking = True
        elif isinstance(frame, UserStoppedSpeakingFrame):
            self.user_speaking = False
        elif isinstance(frame, BotStartedSpeakingFrame):
            self.bot_speaking = True
        elif isinstance(frame, BotStoppedSpeakingFrame):
            self.bot_speaking = False

        # # Handle audio data
        elif isinstance(frame, InputAudioRawFrame):
            if self.user_speaking:
                self._user_audio.extend(frame.audio)
        elif isinstance(frame, TTSAudioRawFrame):
            if self.bot_speaking:
                self._bot_audio.extend(frame.audio)

        # Handle metrics for LLM completion time
        elif isinstance(frame, MetricsFrame):
            self.metrics = frame.data
            for metric in frame.data:
                if isinstance(metric, ProcessingMetricsData):
                    if "LLMService" in metric.processor:
                        self.total_completion_time = metric.value
                elif isinstance(metric, TTFBMetricsData):
                    if "DeepgramSTTService" in metric.processor:
                        self.deepgram_latency += metric.value

        # Pass frame to next processor
        await self.push_frame(frame, direction)

Note: It is required to modify the processes_frame function in pipecat’s base_llm.py to pass along the OpenAILLMContext frame, this makes the handling easier in the FreeplayLLMLogger - process_frame:

async def process_frame(self, frame: Frame, direction: FrameDirection):
  await super().process_frame(frame, direction)

        context = None
        if isinstance(frame, OpenAILLMContextFrame):
            context: OpenAILLMContext = frame.context
            await self.push_frame(frame, direction) # Add this line here to pass frame along
        elif isinstance(frame, LLMMessagesFrame):
            context = OpenAILLMContext.from_messages(frame.messages)
        elif isinstance(frame, VisionImageRawFrame):
            context = OpenAILLMContext()
            context.add_image_frame_message(
                format=frame.format, size=frame.size, image=frame.image, text=frame.text
            )
        elif isinstance(frame, LLMUpdateSettingsFrame):
            await self._update_settings(frame.settings)
        else:
            await self.push_frame(frame, direction)

....

Step 3: Add the FreeplayProcessor To Your Pipeline

Initialize your FreeplayProcessor and add it as a step in your pipeline. It is recommended that you add this after the STT or the audioBuffer steps in your pipeline so that all of the information needed is available when you log to Freeplay.

# Pass the Freeplay client to the FreeplayProcessor
freeplay_processor = FreeplayProcessor(
    fp_client=fp_client,
    template_name="voice-assistant",
    session=session,
    debug=True
)

 #....Additional Pipeline Configuration...

     pipeline = Pipeline(
        [
            transport.input(),  # Websocket input from client
            stt,  # Speech-To-Text
            context_aggregator.user(),
            llm,  # LLM
            tts,  # Text-To-Speech
            freeplay_processor, # Freeplay Logger (after tts so it can capture assistant audio)
            transport.output(),  # Websocket output to client
            audiobuffer,  # Used to buffer the audio in the pipeline
            context_aggregator.assistant(),
        ]
    )

Step 4: Start logging completions

Begin capturing real user interactions in Freeplay, this is a function of the FreeplayProcessor. The audio is being added to the conversation history for proper tracking.

def _record_to_freeplay(self):
  """Record the current conversation state to Freeplay."""
  # Create a new trace for this interaction
        trace = self.session.create_trace(
            input=self.most_recent_user_message,
            custom_metadata={
                "deepgram_latency": self.deepgram_latency,
            },
        )

        self.conversation_history.append(
            {
                "role": "user",
                "content": [
                    {"type": "text", "text": self.most_recent_user_message},
                    {
                        "type": "input_audio",
                        "input_audio": {
                            "data": base64.b64encode(
                                self._make_wav_bytes(
                                    self._user_audio, prepend_silence_secs=1
                                )
                            ).decode("utf-8"),
                            "format": "wav",
                        },
                    },
                ],
            },
        )

        # Bind the variables to the prompt
        if self.unformatted_prompt:
            formatted = self.unformatted_prompt.bind(
                variables={"required_information": self.required_information},
                history=self.conversation_history,
            ).format()
        else:
            # Get formatted prompt. Note this adds latency to the pipeline
            formatted = self.fp_client.prompts.get_formatted(
                project_id=os.getenv("FREEPLAY_PROJECT_ID"),
                template_name=self.template_name,
                environment="latest",
                history=self.conversation_history,
                variables={"required_information": self.required_information},
            )

        # Calculate latency for the LLM interaction
        start, end = time.time(), time.time() + self.total_completion_time

        try:
            # Prepare metadata and record payload
            custom_metadata = {
                "conversation_id": str(self.conversation_id),
            }

            # Add assistant's response to conversation history
            last_message = {
                "role": "assistant",
                "content": [
                    {"type": "text", "text": self.most_recent_completion},
                ],
                "audio": {
                    "id": self.conversation_id,
                    "data": base64.b64encode(
                        self._make_wav_bytes(self._bot_audio, prepend_silence_secs=1)
                    ).decode("utf-8"),
                    "expires_at": 1729234747,
                    "transcript": self.most_recent_completion,
                },
            }
            self.conversation_history.append(last_message)

            # Create recording in Freeplay
            self.fp_client.recordings.create(
              RecordPayload(

project_id=os.get("FP_PROJECT_ID")
all_messages=[
*formatted.llm_prompt,
last_message, # Add the last message to the record call
],
session_info=SessionInfo(
self.session.session_id, custom_metadata=custom_metadata
),
inputs={"required_information": self.required_information},
prompt_version_info=formatted.prompt_info,
call_info=CallInfo.from_prompt_info(
formatted.prompt_info, start, end
),
trace_info=trace,
)
)

            # Record output to trace
            trace.record_output(
                os.getenv("FREEPLAY_PROJECT_ID"),
                self.most_recent_completion,
            )

            print(
                f"Successfully recorded to Freeplay - completion time: {self.total_completion_time}s",
                flush=True,
            )
            self.reset_recent_messages()

        except Exception as e:
            print(f"Error recording to Freeplay: {e}", flush=True)
            self.reset_recent_messages()

Helpful resources:

Configure LiteLLM Proxy Models in Freeplay Security Overview

Getting Started

Account Setup

Core Concepts

How-To Guides

Developer Resources

Security & Compliance

Resources

Voice-Enabled AI with Pipecat, Twilio, and Freeplay

Introduction

What is Pipecat?

Using Freeplay for Rapid Iteration and Observability

What You’ll Be Able to Monitor

Integration Approaches

Option 1: Processor Integration

Option 2: Observer Integration ⭐ Recommended

Conversation Flow in a Pipecat + Twilio + Freeplay Integration

Implementation Guide FreeplayObserver

Pro Tip: AudioBufferProcessor

Prerequisites

Step 1: Import Prompt Configuration from Freeplay

Step 2: Create Your Freeplay Observer

Step 3: Configure Your Pipeline with Audio Buffering

Step 4: Set Up Audio Capture Callbacks & Configure Pipeline

Alternative: FreeplayProcessor Integration

Step 1: Import Your Prompt from Freeplay & Pass to the LLMProcessor

Step 2: Create a Freeplay Processor

Step 3: Add the FreeplayProcessor To Your Pipeline

Step 4: Start logging completions

Helpful resources:

​Introduction

​What is Pipecat?

​Using Freeplay for Rapid Iteration and Observability

​What You’ll Be Able to Monitor

​Integration Approaches

​Option 1: Processor Integration

​Option 2: Observer Integration ⭐ Recommended

​Conversation Flow in a Pipecat + Twilio + Freeplay Integration

​Implementation Guide FreeplayObserver

​Pro Tip: AudioBufferProcessor

​Prerequisites

​Step 1: Import Prompt Configuration from Freeplay

​Step 2: Create Your Freeplay Observer

​Step 3: Configure Your Pipeline with Audio Buffering

​Step 4: Set Up Audio Capture Callbacks & Configure Pipeline

​Alternative: FreeplayProcessor Integration

​Step 1: Import Your Prompt from Freeplay & Pass to the LLMProcessor

​Step 2: Create a Freeplay Processor

​Step 3: Add the FreeplayProcessor To Your Pipeline

​Step 4: Start logging completions

​Helpful resources:

Introduction

What is Pipecat?

Using Freeplay for Rapid Iteration and Observability

What You’ll Be Able to Monitor

Integration Approaches

Option 1: Processor Integration

Option 2: Observer Integration ⭐ Recommended

Conversation Flow in a Pipecat + Twilio + Freeplay Integration

Implementation Guide FreeplayObserver

Pro Tip: AudioBufferProcessor

Prerequisites

Step 1: Import Prompt Configuration from Freeplay

Step 2: Create Your Freeplay Observer

Step 3: Configure Your Pipeline with Audio Buffering

Step 4: Set Up Audio Capture Callbacks & Configure Pipeline

Alternative: FreeplayProcessor Integration

Step 1: Import Your Prompt from Freeplay & Pass to the LLMProcessor

Step 2: Create a Freeplay Processor

Step 3: Add the FreeplayProcessor To Your Pipeline

Step 4: Start logging completions

Helpful resources: