voice-runtime 0.1.2

Provider-agnostic voice call runtime for telephony projects

voice_runtime

Provider-agnostic voice call runtime for telephony projects. Manages audio queues, mark synchronization, STT/TTS providers, and transport protocols — so consumers focus on conversation logic, not plumbing.

Quick Example

Make a call, say something, listen for a response via on_committed callback, hang up:

import asyncio
import threading
import time
import uvicorn
from fastapi import FastAPI

from voice_runtime.session import VoiceSession
from voice_runtime.transports.twilio_ws import register_voice_websocket
from voice_runtime.transports.twilio_call import initiate_outbound_call
from voice_runtime.tts import create_tts
from voice_runtime.stt import create_stt

# 1. Create session and start WebSocket server
session = VoiceSession()
app = FastAPI()
register_voice_websocket(app, session)

def run_server():
    loop = asyncio.new_event_loop()
    asyncio.set_event_loop(loop)
    session.set_loop(loop)
    loop.run_until_complete(uvicorn.Server(
        uvicorn.Config(app, host="0.0.0.0", port=8080, log_level="warning")
    ).serve())

threading.Thread(target=run_server, daemon=True).start()
time.sleep(1)  # wait for server to start

# 2. Initiate call (Twilio calls back to our /voice WebSocket)
call_sid = initiate_outbound_call("[phone]")
session.call_sid = call_sid
session.wait_for_ws_connect(timeout=30)

# 3. Speak
tts = create_tts()  # default: ElevenLabs
tts.speak("Hello! How are you today?", session)
session.send_mark_and_wait("after-greeting")  # block until playback done

# 4. Listen — persistent STT with on_committed callback
transcript_event = threading.Event()
heard = []

def on_committed(text: str):
    heard.append(text)
    transcript_event.set()

session.stt_factory = lambda: create_stt()  # default: ElevenLabs
# Transport starts STT automatically; on_committed fires for each utterance

transcript_event.wait(timeout=30)
print(f"Caller said: {heard}")

# 5. Hang up
session.request_disconnect()

What happens under the hood

Consumer thread              voice_runtime                 Transport (Twilio)
─────────────────────────────────────────────────────────────────────────────
initiate_outbound_call()  →  Twilio REST calls.create()  → Twilio dials phone
                             build_stream_twiml()           with <Connect><Stream>
wait_for_ws_connect()     ←  signal_ws_connected()       ← Twilio opens /voice WS
                                                            stt_factory() → stt.start()
tts.speak(text, session)  →  ffmpeg MP3→μ-law             → send_audio task
                             put_outbound_sync()             sends base64 frames
send_mark_and_wait()      →  get_pending_mark()           → WS sends mark JSON
                          ←  signal_mark_received()       ← WS receives mark echo
on_committed(text)        ←  stt.on_committed()           ← WS receives media frames
                                                             decodes base64 → STT
request_disconnect()      →  _disconnect_requested.set()  → watch_disconnect task
                                                             stt.stop(), closes WS
                                                             Twilio ends call

Architecture

┌──────────────────────────────────────────────┐
│ Consumer (outcaller, ninchat_voice)          │
│  - Subclass VoiceSession (e.g. TelcoSession) │
│  - Call speak(), listen(), hang up           │
├──────────────────────────────────────────────┤
│ voice_runtime                                │
│  - VoiceSession: queues, marks, intents      │
│  - Factories: create_stt / create_tts        │
│  - Providers: ElevenLabs, Azure (STT + TTS)  │
│  - SttTee: dual-provider fan-out             │
│  - Audio: G.711 μ-law codec + mixer          │
├──────────────────────────────────────────────┤
│ Transport (protocol-specific)                │
│  - twilio_ws: Media Streams WebSocket        │
│  - twilio_call: REST call initiation + TwiML │
└──────────────────────────────────────────────┘

Key invariant: VoiceSession has zero transport or provider imports. Consumers never import Twilio or ElevenLabs directly — they use factories and the intent API.

Factories

Provider-agnostic factories mirror the yamlgraph create_llm() pattern:

from voice_runtime.stt import create_stt, get_stt_class
from voice_runtime.tts import create_tts
from voice_runtime.transport import create_transport

stt = create_stt(provider="elevenlabs")     # or "azure"
tts = create_tts(provider="elevenlabs")     # or "azure"
transport = create_transport(provider="twilio")

# get_stt_class returns the class without instantiating (for factory arguments)
SttClass = get_stt_class(provider="elevenlabs")
session.stt_factory = lambda: SttClass(language_code="en")

SttProvider Protocol

All STT providers implement this structural protocol (defined in providers/__init__.py):

class SttProvider(Protocol):
    on_committed: Callable[[str], None] | None    # final transcript for utterance
    on_recognizing: Callable[[str], None] | None  # interim hypothesis (NC-199)
    on_error: Callable[[str], None] | None        # fatal error after reconnect exhausted (NC-258)

    def set_speaking(self, speaking: bool) -> None: ...
    async def start(self, inbound_queue: asyncio.Queue[bytes | None]) -> None: ...
    async def stop(self) -> None: ...

Callback	When it fires	Typical consumer action
on_committed	Final transcript past echo discard window	Route to LLM / FSM
on_recognizing	Interim hypothesis (may change)	Show live transcription UI
on_error	Reconnect attempts exhausted (fatal)	Transition FSM to error state

Transport starts/stops the provider; the consumer decides routing (queue, dispatch, ignore).

TtsProvider Protocol

All TTS providers implement this structural protocol (NC-260 Gap A):

class TtsProvider(Protocol):
    on_error: Callable[[str], None] | None  # synthesis failure (NC-260 Gap A)

    def speak(
        self,
        text: str,
        session: VoiceSession,
        stop_event: threading.Event | None = ...,
    ) -> dict[str, Any]: ...

speak() returns a dict with keys: last_spoken (str), and optionally call_disconnected (bool) or interrupted (bool). on_error fires on synthesis failures so the FSM doesn't hang in a speaking state.

VoiceSession

Central coordinator between sync tool threads, async transport, and STT/TTS providers.

Audio I/O

Method	Thread safety	Purpose
put_inbound(data)	Any → async	Enqueue caller audio (transport calls this)
get_outbound()	async only	Dequeue agent audio (transport reads this)
put_outbound_sync(data)	Sync → async	Enqueue agent audio (TTS provider calls this)
clear_inbound()	Any	Drain stale audio frames

All sync→async bridging uses asyncio.run_coroutine_threadsafe().

Mark Synchronization

Marks let sync tool code block until the transport confirms audio playback reached a point. This is how you know a TTS utterance finished playing before you start listening.

tts.speak("What is your name?", session)
session.send_mark_and_wait("after-question", timeout=10.0)
# Now safe to start listening — caller heard the full question

session.clear_inbound()
transcript = stt.listen(session, timeout=30)

Method	Purpose
send_mark_and_wait(name, timeout)	Block sync thread until mark echoed
signal_mark_received(name)	Called by transport when mark arrives
get_pending_mark()	Async — transport reads next mark to send

Transport Intent (NC-154)

Consumers signal what they want; the transport decides how.

session.request_disconnect()     # transport closes connection, call ends
session.request_clear_buffer()   # transport discards buffered audio (barge-in)

Both are thread-safe. The transport watches _disconnect_requested (asyncio.Event) and _clear_queue (asyncio.Queue) and acts in its own protocol's terms — e.g. Twilio closes the WebSocket, which ends the call; SIP would send BYE.

STT Factory

Attach an STT factory and the transport manages its lifecycle automatically:

from voice_runtime.stt import create_stt

session.stt_factory = lambda: create_stt(provider="elevenlabs")
# Transport calls stt_factory() on stream start, stt.stop() on disconnect

Optional secondary STT for parallel logging/comparison (via SttTee):

session.stt_secondary_factory = lambda: create_stt(provider="azure")
# Transport wraps both in SttTee — primary drives on_committed, secondary logs only

STT Ready Hook (NC-260 Gap E)

Wire callbacks after the transport creates the STT instance but before start():

def wire_callbacks(stt: SttProvider):
    stt.on_committed = handle_transcript
    stt.on_recognizing = handle_interim
    stt.on_error = handle_stt_death

session.on_stt_ready = wire_callbacks

This replaces the old pattern of wiring callbacks before attaching the factory. The transport calls on_stt_ready(stt) after construction, guaranteeing callbacks are set before start() fires.

Lifecycle

Method	Purpose
signal_ws_connected(stream_sid)	Transport calls when connection established
wait_for_ws_connect(timeout)	Consumer blocks until connected; raises CallNotAnsweredError
signal_disconnected()	Transport calls on hangup
is_disconnected	Property — check if call ended
reset()	Clear all state for session reuse (multi-call servers)

Audio Monitoring

Optional two-channel mixer for real-time call monitoring (requires ffplay) and WAV recording (NC-235):

from pathlib import Path
from voice_runtime.audio import AudioMixer

# Monitor only (plays mixed audio through ffplay)
mixer = AudioMixer()

# Monitor + record to WAV file (8kHz mono mulaw)
mixer = AudioMixer(record_path=Path("recordings/call_001.wav"))

mixer.start()
session.set_mixer(mixer)
# session.tap_caller() / session.tap_agent() now feed audio to ffplay + WAV

Exceptions

Exception	When
MissingStreamUrlError	VOICE_STREAM_URL env var not set
CallNotAnsweredError(timeout)	WebSocket didn't connect within timeout
CallHangupError	Call hung up during a listen operation

Transport: Twilio

WebSocket Handler

Registers a /voice endpoint on a FastAPI app implementing Twilio Media Streams:

from voice_runtime.transports.twilio_ws import register_voice_websocket

app = FastAPI()
register_voice_websocket(app, session)

Runs 5 async tasks on stream start: send_audio, send_marks, watch_disconnect, send_clears, stt (if factory provided).

Call Initiation

from voice_runtime.transports.twilio_call import (
    initiate_outbound_call,
    build_stream_twiml,  # alias: build_stream_xml
)

# Outbound: dial phone, Twilio connects back to /voice WebSocket
call_sid = initiate_outbound_call("[phone]")

# Inbound webhook: return TwiML that tells Twilio to stream audio to /voice
xml = build_stream_twiml("wss://example.ngrok.io")

Providers

ElevenLabs TTS

Streams text → ElevenLabs API → ffmpeg (MP3 → μ-law 8kHz) → session outbound queue.

from voice_runtime.tts import create_tts

tts = create_tts(provider="elevenlabs")
result = tts.speak("Hello", session, stop_event=barge_in_event)
# result: {"last_spoken": "Hello"} or {"last_spoken": "Hello", "call_disconnected": True}

Supports barge-in interrupt: pass a threading.Event as stop_event; set it from another thread to cut TTS mid-stream. Result may include {"interrupted": True}.

ElevenLabs STT

Persistent Scribe WebSocket per call lifetime. Barge-in detection, echo discard, reconnect on fatal errors.

from voice_runtime.stt import create_stt

stt = create_stt(provider="elevenlabs")
stt.on_committed = lambda text: print(f"Heard: {text}")
await stt.start(session.inbound)
# ... later
await stt.stop()

Typically managed by the transport via session.stt_factory rather than started manually.

Azure TTS

Streams text → Azure Speech SDK → mulaw 8kHz → session outbound queue.

from voice_runtime.tts import create_tts

tts = create_tts(provider="azure")
result = tts.speak("Hello", session, stop_event=barge_in_event)

Same speak() interface as ElevenLabs. Uses AZURE_SPEECH_KEY, AZURE_SPEECH_REGION, AZURE_TTS_VOICE env vars.

Azure STT

Persistent Azure Speech SDK recognizer with continuous recognition.

from voice_runtime.stt import create_stt

stt = create_stt(provider="azure", language_code="fi-FI", silence_timeout_ms=1500)
stt.on_committed = lambda text: print(f"Heard: {text}")
await stt.start(session.inbound)

Same SttProvider protocol as ElevenLabs. Echo discard window (400ms) after TTS ends.

SttTee (Dual STT)

Fan-out adapter running two STT providers on the same audio stream:

from voice_runtime.stt_tee import SttTee

tee = SttTee(primary=elevenlabs_stt, secondary=azure_stt)
tee.on_committed = handler  # proxied to primary only
await tee.start(session.inbound)

Primary drives production (on_committed). Secondary receives same frames for logging/comparison only — its errors never propagate. Typically wired automatically via session.stt_secondary_factory.

Audio Codec

G.711 μ-law at 8kHz — Twilio's native format. 160 bytes = 20ms frame.

from voice_runtime.audio import mix_frames

mixed = mix_frames(caller_chunk, agent_chunk)  # mix two 160-byte frames

Environment Variables

Variable	Purpose	Required
VOICE_STREAM_URL	Public WebSocket URL for transport callback	Yes
VOICE_SERVER_PORT	Uvicorn listen port	No (default: 8080)
TWILIO_ACCOUNT_SID	Twilio credentials (call initiation)	For outbound
TWILIO_AUTH_TOKEN	Twilio credentials	For outbound
TWILIO_PHONE_NUMBER	Outbound caller ID	For outbound
ELEVENLABS_API_KEY	ElevenLabs authentication	For ElevenLabs
ELEVENLABS_VOICE_ID	TTS voice	No (default: Rachel)
ELEVENLABS_MODEL	TTS model	No (default: eleven_multilingual_v2)
STT_MODEL_ID	ElevenLabs STT model	No (default: scribe_v2_realtime)
STT_LANGUAGE_CODE	ElevenLabs STT language	No (default: fi)
AZURE_SPEECH_KEY	Azure Speech SDK authentication	For Azure
AZURE_SPEECH_REGION	Azure Speech SDK region	No (default: westeurope)
AZURE_TTS_VOICE	Azure TTS voice name	No (default: fi-FI-NooraNeural)
VOICE_MONITOR	Enable AudioMixer monitoring	No (default: off)

Transport: SMS (NC-193)

Send SMS via Twilio REST API without importing the Twilio SDK in consumers:

from voice_runtime.transport import get_sms_transport

sms = get_sms_transport()  # default: twilio
result = sms.send_sms(to="[phone]", body="Your appointment is confirmed.")
# result: {"message_sid": "SM...", "status": "queued", "to": "+358..."}

Requires TWILIO_ACCOUNT_SID, TWILIO_AUTH_TOKEN, and TWILIO_PHONE_NUMBER env vars.

Consumer Pattern

Typical consumer subclasses VoiceSession and adds server lifecycle:

from dataclasses import dataclass
from voice_runtime.session import VoiceSession

@dataclass
class TelcoSession(VoiceSession):
    def start(self):
        app = FastAPI()
        register_voice_websocket(app, self)
        # Run uvicorn in daemon thread
        threading.Thread(target=self._run_loop, daemon=True).start()

    def shutdown(self):
        # Signal event loop to stop, join thread
        ...

Tool nodes then use the session for audio I/O, mark sync, and transport intents — without knowing anything about Twilio, WebSockets, or ElevenLabs.

Known Consumers

Any project that subclasses VoiceSession and registers a FastAPI WebSocket handler via register_voice_websocket is a consumer. See the Consumer Pattern section above for the canonical implementation.

Multi-Call Session Reuse

When servers handle multiple sequential calls on the same VoiceSession instance, reset() clears all state between calls:

• Stops active STT via asyncio.run_coroutine_threadsafe(stt.stop(), loop) before clearing the reference (prevents orphaned WebSocket connections)
• Drains inbound and outbound queues
• Resets mark synchronization and transport intent events

The STT start() method also drains the inbound queue as defense-in-depth against sentinel values from prior call cleanup.

STT Reconnect on Fatal Errors

PersistentSttSession detects fatal WebSocket errors (connection closed, protocol errors) and automatically reconnects:

• _on_error() schedules _reconnect_after_error() for errors in _FATAL_ERRORS
• Reconnect drains stale frames, creates a new WebSocket, and resumes the feed task
• _feed_audio() wraps send() in try/except for dead socket resilience