modmex-ai 0.1.0

Lightweight AI flows, agents, tools, and model adapters for event-driven Python systems.

modmex-ai

Lightweight, provider-neutral agents, flows, tools, and voice runtimes for Python.

modmex-ai is a small runtime for applications that need an agent to reason, call tools, hand off work, and return a controlled result. It fits equally well in HTTP APIs, CLIs, background workers, WebSocket servers, notebooks, desktop applications, voice systems, and event-driven/serverless workloads.

Its core has no provider SDK dependency. OpenAI is the first complete adapter; the core contracts remain reusable for another text, speech, or realtime provider.

The basic execution model is intentionally simple:

input -> Flow -> one or more Agents -> output, tools, handoffs, and events

What it provides

• Agent for instructions, structured output, tools, guardrails, and streaming.
• Flow for multi-agent routing, handoffs, continuation, and sync/async execution.
• Sessions, snapshots, optimistic concurrency contracts, and approval checkpoints.
• Realtime and chained voice contracts that keep STT, reasoning, and TTS composable.
• Provider-neutral tracing, usage, evals, and host-owned observability.

Install the core package in any Python application:

pip install modmex-ai

Add only the optional transport needed by a deployment:

pip install 'modmex-ai[async]'      # native async HTTP
pip install 'modmex-ai[realtime]'  # WebSocket-based Realtime adapters

Quick start

from modmex import BaseModel
from modmex_ai import Agent, FakeModel, Flow


class TriageOutput(BaseModel):
    intent: str
    confidence: float


triage = Agent(
    name="triage",
    instructions="Classify the incoming message.",
    output_type=TriageOutput,
    handoffs=["support"],
)

flow = Flow(
    name="analyze-message",
    entrypoint=triage,
    agents=[],
    model=FakeModel(['{"intent":"support","confidence":0.91}']),
)

result = flow.run("I need help")
result.output.model_dump()

The same Flow can be called by a REST handler, a queue consumer, a CLI command, or a long-lived WebSocket connection. Event-driven systems are a natural fit, but they are not required.

Multi-agent delegation uses handoff function tools, not a field in the model output. A handoff defaults to transfer_to_agent_name; its typed input is validated and delivered to an on_handoff callback before the next agent continues with the conversation.

Durable sessions and approvals

The core provides portable SessionSnapshot and PersistedFlowState models, plus optimistic-concurrency store contracts. A host chooses the database and owns the lifecycle; this keeps the framework usable in a web application, a worker, or serverless infrastructure without importing a storage SDK into the core.

Tools can require an externally verified approval. When a protected tool is requested, a Flow raises FlowSuspended with the exact pending tool call and a serializable checkpoint. Persist it, collect the signed approval in the host, then resume the Flow without asking the model to decide again.

modmex-lambda includes optional DynamoDB store adapters for Lambda-oriented hosts. Other applications can implement the same DurableSessionStore and FlowStateStore contracts with Postgres, Redis, or their existing storage.

Realtime voice sessions

VoiceAgentSession is the common contract for voice sessions. There are two implementations with distinct responsibilities:

• OpenAIRealtimeSession para speech-to-speech por WebSocket/SIP.
• ChainedVoiceSession para conservar el control explícito de STT → Agent/Flow → TTS.

Ambas exponen eventos normalizados, uso acumulado, herramientas y handoffs. El adaptador de OpenAI conserva el payload específico del proveedor únicamente en su frontera; el core usa tool_call_id para no confundir llamadas de tools con identificadores de llamadas telefónicas.

Install the optional WebSocket transport only in workloads that need it:

pip install 'modmex-ai[realtime]'

The OpenAI adapter reuses Agent tools, handoffs, tool guardrails, context, usage, and traces. It does not depend on the OpenAI Agents SDK. For a SIP call that was already accepted by your server, connect the worker using its realtime_call_id:

import asyncio

from modmex_ai import Agent
from modmex_ai.providers.openai import (
    OpenAIRealtimeClient,
    OpenAIRealtimeSessionConfig,
)


async def run_voice_call(realtime_call_id: str) -> None:
    agent = Agent(
        name="support",
        instructions="Help the caller concisely and use tools when needed.",
    )
    client = OpenAIRealtimeClient()
    session = await client.connect(
        agent=agent,
        realtime_call_id=realtime_call_id,
        config=OpenAIRealtimeSessionConfig(voice="marin"),
    )
    async with session:
        await session.configure()
        await session.create_response(
            instructions="Greet the caller and ask how you can help."
        )
        async for event in session.events():
            if event.type == "response.done":
                print(session.usage.model_dump())


asyncio.run(run_voice_call("rtc_u1_example"))

When the model emits a function call, the session validates and executes the registered Modmex-AI tool, sends its output to the socket, and asks Realtime to continue. A transfer_to_* handoff updates the active agent's instructions and tool surface in the same Realtime session.

For a SIP accept endpoint, build the accept body from the exact same session configuration used after the WebSocket connects. That prevents model, voice, instructions and tool definitions from drifting between both phases:

accept_payload = OpenAIRealtimeSessionConfig(voice="marin").to_accept_payload(agent)

Chained voice pipeline

Use a chained session when the application must choose its transcription and speech providers, or needs text-agent behavior between audio turns. A Flow is preferred when the conversation can hand off to another agent.

session = ChainedVoiceSession(
    flow=carrier_flow,
    speech_to_text=transcriber,
    text_to_speech=synthesizer,
    context={"workspace_id": "workspace-1"},
)

audio_reply = await session.process_audio(caller_audio)
async for event in session.voice_events():
    # transcript_final, tool_finished, handoff_completed,
    # response_completed, session_ended, or error
    handle_voice_event(event)

The STT/TTS providers are small async protocols, so concrete provider SDKs stay outside the core package. Close either session with a VoiceTerminationReason to make lifecycle reporting explicit.

For bounded audio turns, the OpenAI adapters use the Audio API directly through the lightweight HTTP client—no provider SDK is installed:

from modmex_ai.providers.openai import (
    OpenAISpeechProvider,
    OpenAITranscriptionProvider,
)

session = ChainedVoiceSession(
    flow=carrier_flow,
    speech_to_text=OpenAITranscriptionProvider(model="gpt-4o-mini-transcribe"),
    text_to_speech=OpenAISpeechProvider(voice="marin", response_format="wav"),
)

To continue a chained conversation in another process, persist the host's Session implementation and session.continuation. The continuation tracks the active agent and provider state; the host owns durable history storage. For a small self-contained host, SessionSnapshot.from_session(session) can serialize an InMemorySession and later restore it with snapshot.to_memory_session().

ChainedVoiceSession.process_audio_stream() supports providers that implement streaming STT and TTS contracts. It emits transcript deltas, waits for a final transcript before invoking the text flow, then yields playable audio chunks. The callable adapters make it possible to plug in a provider-specific function without making that provider a dependency of modmex-ai.

Native async models

Models can optionally expose async def acomplete(request). Agent.run_async and Flow.run_async use it directly, including async tools; existing sync models retain their compatible worker-thread fallback. The built-in OpenAI clients expose acomplete() as well. Install the optional native transport to make them select httpx automatically:

pip install 'modmex-ai[async]'

Without that extra, the compatible stdlib transport runs in a worker and keeps the event loop responsive.

Realtime transcription

For live microphone or telephony media, use the transcription-only WebSocket adapter instead of a speech-to-speech agent session. It emits Transcription deltas as audio arrives and one final value after manual commit:

from modmex_ai.providers.openai import OpenAIRealtimeTranscriptionProvider

transcriber = OpenAIRealtimeTranscriptionProvider()
async for transcript in transcriber.transcribe_stream(pcm_chunks, context=context):
    render(transcript.text, final=transcript.is_final)

The adapter uses 24 kHz PCM by default and requires modmex-ai[realtime]. For completed audio turns, OpenAITranscriptionProvider remains the simpler request-based alternative.

For a long-lived call, open one reusable session and commit each user turn:

session = await transcriber.connect()
await session.append_audio(pcm_chunk)
await session.commit_turn()
async for transcript in session.events():
    correlate(transcript.item_id, transcript.text, transcript.is_final)

Turn cancellation and provider retries

ChainedVoiceSession accepts VoiceTurnOptions(max_provider_retries=...). Retries apply only to STT/TTS provider I/O, never to the text flow, tools, or handoffs. Call cancel_current_turn() to cooperatively stop a streaming turn before its next provider chunk; hosts can still cancel the owning task when an immediate interruption of a blocked transport is required.

After every completed, cancelled, timed-out, or failed bounded turn, ChainedVoiceSession.last_turn contains a VoiceTurnOutcome. Its metrics separate STT, agent, TTS, total latency, and provider retry count. Pass a VoiceTurnObserver to export those outcomes to the host's own logs or metrics backend without adding an observability dependency to this package.

Roadmap

Model Context Protocol (MCP) is not supported yet. MCP client and tool-server integration are planned for a future release; today, applications can register local tools directly through Agent.