feat: refactor telephony to support multiple telephony configurations (#251)

Co-authored-by: Sabiha Khan <sabihak89@gmail.com>
This commit is contained in:
Abhishek 2026-04-29 11:39:57 +05:30 committed by GitHub
parent 2f860e7f6d
commit e16f6438bd
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101 changed files with 10906 additions and 5420 deletions

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@ -1,5 +1,4 @@
"""
Audio configuration for pipeline components.
"""Audio configuration for pipeline components.
This module provides centralized audio configuration to ensure consistent
sample rates across all pipeline components and proper coordination between
@ -11,8 +10,6 @@ from typing import Optional
from loguru import logger
from api.enums import WorkflowRunMode
@dataclass
class AudioConfig:
@ -84,61 +81,35 @@ class AudioConfig:
def create_audio_config(transport_type: str) -> AudioConfig:
"""Create audio configuration based on transport type.
"""Create audio configuration for a given transport.
Args:
transport_type: Type of transport ("webrtc", "twilio", "plivo", "vonage", "vobiz", "cloudonix")
Returns:
AudioConfig instance with appropriate settings
Telephony providers contribute their wire-format sample rate through the
provider registry (``ProviderSpec.transport_sample_rate``); WebRTC modes
use 16 kHz (transports handle resampling from/to 24 kHz). The remaining
AudioConfig fields are derived from the chosen rate.
"""
if transport_type in (
WorkflowRunMode.TWILIO.value,
WorkflowRunMode.PLIVO.value,
WorkflowRunMode.VOBIZ.value,
WorkflowRunMode.CLOUDONIX.value,
WorkflowRunMode.ARI.value,
WorkflowRunMode.TELNYX.value,
):
# Twilio, Plivo, Cloudonix, Vobiz, Telnyx, and ARI use MULAW at 8kHz
return AudioConfig(
transport_in_sample_rate=8000,
transport_out_sample_rate=8000,
vad_sample_rate=8000, # Use matching VAD rate
pipeline_sample_rate=8000, # Keep at 8kHz to avoid resampling
buffer_size_seconds=5.0,
)
elif transport_type == WorkflowRunMode.VONAGE.value:
# Vonage uses 16kHz Linear PCM
return AudioConfig(
transport_in_sample_rate=16000,
transport_out_sample_rate=16000,
vad_sample_rate=16000, # Use matching VAD rate
pipeline_sample_rate=16000, # Keep at 16kHz to avoid resampling
buffer_size_seconds=5.0,
)
elif transport_type in [
# Defer registry import to avoid an import cycle: the registry is imported
# by every telephony provider package at startup.
from api.enums import WorkflowRunMode
from api.services.telephony import registry
telephony_spec = registry.get_optional(transport_type)
if telephony_spec is not None:
rate = telephony_spec.transport_sample_rate
elif transport_type in (
WorkflowRunMode.WEBRTC.value,
WorkflowRunMode.SMALLWEBRTC.value,
]:
# WebRTC typically uses 24kHz or 48kHz, but we limit pipeline to 16kHz
# The transport will handle resampling between 24kHz and 16kHz
return AudioConfig(
transport_in_sample_rate=16000, # Transport will resample from 24kHz
transport_out_sample_rate=16000, # Transport will resample to 24kHz
vad_sample_rate=16000, # VAD native rate
pipeline_sample_rate=16000, # Keep pipeline at 16kHz
buffer_size_seconds=5.0,
)
):
rate = 16000
else:
# Default configuration
logger.warning(
f"Unknown transport type: {transport_type}, using default config"
)
return AudioConfig(
transport_in_sample_rate=16000,
transport_out_sample_rate=16000,
vad_sample_rate=16000,
pipeline_sample_rate=16000,
buffer_size_seconds=5.0,
)
rate = 16000
return AudioConfig(
transport_in_sample_rate=rate,
transport_out_sample_rate=rate,
vad_sample_rate=rate,
pipeline_sample_rate=rate,
)

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@ -0,0 +1,55 @@
"""Shared helper for building audio output mixers used by telephony transports."""
import os
from loguru import logger
from api.constants import APP_ROOT_DIR
from api.services.pipecat.audio_file_cache import get_cached_ambient_noise_path
from pipecat.audio.mixers.silence_mixer import SilenceAudioMixer
from pipecat.audio.mixers.soundfile_mixer import SoundfileMixer
librnnoise_path = os.path.normpath(
str(APP_ROOT_DIR / "native" / "rnnoise" / "librnnoise.so")
)
async def build_audio_out_mixer(
audio_out_sample_rate: int,
ambient_noise_config: dict | None,
):
"""Build the audio output mixer based on the ambient noise configuration.
Returns a ``SoundfileMixer`` when ambient noise is enabled, or a
``SilenceAudioMixer`` otherwise. Supports custom user-uploaded audio
files via the ``storage_key`` / ``storage_backend`` fields in the config.
"""
if not ambient_noise_config or not ambient_noise_config.get("enabled", False):
return SilenceAudioMixer()
volume = ambient_noise_config.get("volume", 0.3)
storage_key = ambient_noise_config.get("storage_key")
storage_backend = ambient_noise_config.get("storage_backend")
if storage_key and storage_backend:
cached_path = await get_cached_ambient_noise_path(
storage_key, storage_backend, audio_out_sample_rate
)
if cached_path:
return SoundfileMixer(
sound_files={"custom": cached_path},
default_sound="custom",
volume=volume,
)
logger.warning("Custom ambient noise file unavailable, falling back to default")
return SoundfileMixer(
sound_files={
"office": APP_ROOT_DIR
/ "assets"
/ f"office-ambience-{audio_out_sample_rate}-mono.wav"
},
default_sound="office",
volume=volume,
)

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@ -1,11 +1,10 @@
import asyncio
from typing import Optional
from fastapi import HTTPException, WebSocket
from fastapi import HTTPException
from loguru import logger
from api.db import db_client
from api.db.models import WorkflowModel
from api.enums import WorkflowRunMode
from api.services.configuration.registry import ServiceProviders
from api.services.pipecat.audio_config import AudioConfig, create_audio_config
@ -44,17 +43,9 @@ from api.services.pipecat.service_factory import (
from api.services.pipecat.tracing_config import (
ensure_tracing,
)
from api.services.pipecat.transport_setup import (
create_ari_transport,
create_cloudonix_transport,
create_plivo_transport,
create_telnyx_transport,
create_twilio_transport,
create_vobiz_transport,
create_vonage_transport,
create_webrtc_transport,
)
from api.services.pipecat.transport_setup import create_webrtc_transport
from api.services.pipecat.ws_sender_registry import get_ws_sender
from api.services.telephony import registry as telephony_registry
from api.services.workflow.dto import ReactFlowDTO
from api.services.workflow.pipecat_engine import PipecatEngine
from api.services.workflow.workflow import WorkflowGraph
@ -95,110 +86,75 @@ from pipecat.utils.run_context import set_current_org_id, set_current_run_id
ensure_tracing()
async def run_pipeline_twilio(
websocket_client: WebSocket,
stream_sid: str,
call_sid: str,
async def run_pipeline_telephony(
websocket,
*,
provider_name: str,
workflow_id: int,
workflow_run_id: int,
user_id: int,
call_id: str,
transport_kwargs: dict,
) -> None:
"""Run pipeline for Twilio connections"""
logger.debug(
f"Running pipeline for Twilio connection with workflow_id: {workflow_id} and workflow_run_id: {workflow_run_id}"
)
"""Run a pipeline for any telephony provider.
Replaces the previous per-provider run_pipeline_<x> functions. The
provider's transport factory and audio config are looked up from the
registry, so adding a new provider requires no changes here.
Args:
websocket: The accepted WebSocket from the provider.
provider_name: Stable identifier of the provider (registry key).
workflow_id: Workflow being executed.
workflow_run_id: Workflow run row.
user_id: Owner of the workflow.
call_id: Provider call identifier (stored in cost_info for billing).
transport_kwargs: Provider-specific kwargs forwarded to the transport
factory (e.g. stream_sid + call_sid for Twilio).
"""
logger.debug(f"Running {provider_name} pipeline for workflow_run {workflow_run_id}")
set_current_run_id(workflow_run_id)
# Store call ID in cost_info for later cost calculation (provider-agnostic)
cost_info = {"call_id": call_sid}
await db_client.update_workflow_run(workflow_run_id, cost_info=cost_info)
await db_client.update_workflow_run(workflow_run_id, cost_info={"call_id": call_id})
# Get workflow to extract all pipeline configurations
workflow = await db_client.get_workflow(workflow_id, user_id)
# Set org context early so tasks created by the transport inherit it
if workflow:
set_current_org_id(workflow.organization_id)
vad_config = None
ambient_noise_config = None
if workflow and workflow.workflow_configurations:
if "vad_configuration" in workflow.workflow_configurations:
vad_config = workflow.workflow_configurations["vad_configuration"]
if "ambient_noise_configuration" in workflow.workflow_configurations:
ambient_noise_config = workflow.workflow_configurations[
"ambient_noise_configuration"
]
vad_config = workflow.workflow_configurations.get("vad_configuration")
ambient_noise_config = workflow.workflow_configurations.get(
"ambient_noise_configuration"
)
# Create audio configuration for Twilio
audio_config = create_audio_config(WorkflowRunMode.TWILIO.value)
# The telephony config id is stamped on the workflow run when it's created
# (test call, campaign dispatch, inbound). Transports use it to load creds
# from the right config row. Falls back to None for legacy runs (transports
# then resolve the org's default config).
workflow_run = await db_client.get_workflow_run(workflow_run_id)
telephony_configuration_id = None
if workflow_run and workflow_run.initial_context:
telephony_configuration_id = workflow_run.initial_context.get(
"telephony_configuration_id"
)
transport = await create_twilio_transport(
websocket_client,
stream_sid,
call_sid,
spec = telephony_registry.get(provider_name)
audio_config = create_audio_config(provider_name)
transport = await spec.transport_factory(
websocket,
workflow_run_id,
audio_config,
workflow.organization_id,
vad_config,
ambient_noise_config,
vad_config=vad_config,
ambient_noise_config=ambient_noise_config,
telephony_configuration_id=telephony_configuration_id,
**transport_kwargs,
)
await _run_pipeline(
transport,
workflow_id,
workflow_run_id,
user_id,
audio_config=audio_config,
)
async def run_pipeline_plivo(
websocket_client: WebSocket,
stream_id: str,
call_id: str,
workflow_id: int,
workflow_run_id: int,
user_id: int,
) -> None:
"""Run pipeline for Plivo WebSocket connections."""
logger.info(
f"[run {workflow_run_id}] Starting Plivo pipeline - "
f"stream_id={stream_id}, call_id={call_id}, workflow_id={workflow_id}"
)
set_current_run_id(workflow_run_id)
cost_info = {"call_id": call_id}
await db_client.update_workflow_run(workflow_run_id, cost_info=cost_info)
workflow = await db_client.get_workflow(workflow_id, user_id)
if workflow:
set_current_org_id(workflow.organization_id)
vad_config = None
ambient_noise_config = None
if workflow and workflow.workflow_configurations:
if "vad_configuration" in workflow.workflow_configurations:
vad_config = workflow.workflow_configurations["vad_configuration"]
if "ambient_noise_configuration" in workflow.workflow_configurations:
ambient_noise_config = workflow.workflow_configurations[
"ambient_noise_configuration"
]
try:
audio_config = create_audio_config(WorkflowRunMode.PLIVO.value)
transport = await create_plivo_transport(
websocket_client,
stream_id,
call_id,
workflow_run_id,
audio_config,
workflow.organization_id,
vad_config,
ambient_noise_config,
)
await _run_pipeline(
transport,
workflow_id,
@ -206,341 +162,14 @@ async def run_pipeline_plivo(
user_id,
audio_config=audio_config,
)
logger.info(f"[run {workflow_run_id}] Plivo pipeline completed successfully")
except Exception as e:
logger.error(
f"[run {workflow_run_id}] Error in Plivo pipeline: {e}", exc_info=True
f"[run {workflow_run_id}] Error in {provider_name} pipeline: {e}",
exc_info=True,
)
raise
async def run_pipeline_vonage(
websocket_client,
call_uuid: str,
workflow: WorkflowModel,
organization_id: int,
workflow_id: int,
workflow_run_id: int,
user_id: int,
):
"""Run pipeline for Vonage WebSocket connections.
Vonage uses raw PCM audio over WebSocket instead of base64-encoded μ-law.
The audio is transmitted as binary frames at 16kHz by default.
"""
logger.info(f"Starting Vonage pipeline for workflow run {workflow_run_id}")
set_current_run_id(workflow_run_id)
set_current_org_id(organization_id)
# Store call ID in cost_info for later cost calculation (provider-agnostic)
cost_info = {"call_id": call_uuid}
await db_client.update_workflow_run(workflow_run_id, cost_info=cost_info)
# Extract VAD and ambient noise config from workflow
vad_config = None
ambient_noise_config = None
if workflow and workflow.workflow_configurations:
if "vad_configuration" in workflow.workflow_configurations:
vad_config = workflow.workflow_configurations["vad_configuration"]
if "ambient_noise_configuration" in workflow.workflow_configurations:
ambient_noise_config = workflow.workflow_configurations[
"ambient_noise_configuration"
]
try:
# Setup audio config for Vonage using the centralized config
audio_config = create_audio_config(WorkflowRunMode.VONAGE.value)
# Create Vonage transport
transport = await create_vonage_transport(
websocket_client,
call_uuid,
workflow_run_id,
audio_config,
organization_id,
vad_config,
ambient_noise_config,
)
# No special handshake needed for Vonage
# Audio streaming starts immediately
# Run the pipeline (same as Twilio/WebRTC)
await _run_pipeline(
transport,
workflow_id,
workflow_run_id,
user_id,
call_context_vars={},
audio_config=audio_config,
)
except Exception as e:
logger.error(f"Error in Vonage pipeline: {e}")
raise
async def run_pipeline_ari(
websocket_client: WebSocket,
channel_id: str,
workflow_id: int,
workflow_run_id: int,
user_id: int,
) -> None:
"""Run pipeline for Asterisk ARI WebSocket connections.
ARI uses raw 16-bit signed linear PCM (SLIN16) at 16kHz
transmitted as binary WebSocket frames via chan_websocket.
"""
logger.info(f"Starting ARI pipeline for workflow run {workflow_run_id}")
set_current_run_id(workflow_run_id)
# Store call ID (channel_id) in cost_info
cost_info = {"call_id": channel_id}
await db_client.update_workflow_run(workflow_run_id, cost_info=cost_info)
# Get workflow to extract configurations
workflow = await db_client.get_workflow(workflow_id, user_id)
# Set org context early so tasks created by the transport inherit it
if workflow:
set_current_org_id(workflow.organization_id)
vad_config = None
ambient_noise_config = None
if workflow and workflow.workflow_configurations:
if "vad_configuration" in workflow.workflow_configurations:
vad_config = workflow.workflow_configurations["vad_configuration"]
if "ambient_noise_configuration" in workflow.workflow_configurations:
ambient_noise_config = workflow.workflow_configurations[
"ambient_noise_configuration"
]
try:
audio_config = create_audio_config(WorkflowRunMode.ARI.value)
transport = await create_ari_transport(
websocket_client,
channel_id,
workflow_run_id,
audio_config,
workflow.organization_id,
vad_config,
ambient_noise_config,
)
await _run_pipeline(
transport,
workflow_id,
workflow_run_id,
user_id,
audio_config=audio_config,
)
except Exception as e:
logger.error(f"Error in ARI pipeline: {e}")
raise
async def run_pipeline_vobiz(
websocket_client: WebSocket,
stream_id: str,
call_id: str,
workflow_id: int,
workflow_run_id: int,
user_id: int,
) -> None:
"""Run pipeline for Vobiz using Plivo-compatible WebSocket protocol."""
logger.info(
f"[run {workflow_run_id}] Starting Vobiz pipeline - "
f"stream_id={stream_id}, call_id={call_id}, workflow_id={workflow_id}"
)
set_current_run_id(workflow_run_id)
cost_info = {"call_id": call_id}
await db_client.update_workflow_run(workflow_run_id, cost_info=cost_info)
workflow = await db_client.get_workflow(workflow_id, user_id)
# Set org context early so tasks created by the transport inherit it
if workflow:
set_current_org_id(workflow.organization_id)
vad_config = None
ambient_noise_config = None
if workflow and workflow.workflow_configurations:
if "vad_configuration" in workflow.workflow_configurations:
vad_config = workflow.workflow_configurations["vad_configuration"]
if "ambient_noise_configuration" in workflow.workflow_configurations:
ambient_noise_config = workflow.workflow_configurations[
"ambient_noise_configuration"
]
try:
audio_config = create_audio_config(WorkflowRunMode.VOBIZ.value)
logger.info(
f"[run {workflow_run_id}] Vobiz audio config: "
f"sample_rate={audio_config.transport_in_sample_rate}Hz, format=MULAW"
)
transport = await create_vobiz_transport(
websocket_client,
stream_id,
call_id,
workflow_run_id,
audio_config,
workflow.organization_id,
vad_config,
ambient_noise_config,
)
logger.info(f"[run {workflow_run_id}] Starting Vobiz pipeline execution")
await _run_pipeline(
transport,
workflow_id,
workflow_run_id,
user_id,
audio_config=audio_config,
)
logger.info(f"[run {workflow_run_id}] Vobiz pipeline completed successfully")
except Exception as e:
logger.error(
f"[run {workflow_run_id}] Error in Vobiz pipeline: {e}", exc_info=True
)
raise
async def run_pipeline_telnyx(
websocket_client: WebSocket,
stream_id: str,
call_control_id: str,
workflow_id: int,
workflow_run_id: int,
user_id: int,
) -> None:
"""Run pipeline for Telnyx Call Control WebSocket connections.
Telnyx uses PCMU at 8kHz over WebSocket with base64-encoded media events,
similar to Twilio's protocol.
"""
logger.info(
f"[run {workflow_run_id}] Starting Telnyx pipeline - "
f"stream_id={stream_id}, call_control_id={call_control_id}, "
f"workflow_id={workflow_id}"
)
set_current_run_id(workflow_run_id)
cost_info = {"call_id": call_control_id}
await db_client.update_workflow_run(workflow_run_id, cost_info=cost_info)
workflow = await db_client.get_workflow(workflow_id, user_id)
if workflow:
set_current_org_id(workflow.organization_id)
vad_config = None
ambient_noise_config = None
if workflow and workflow.workflow_configurations:
if "vad_configuration" in workflow.workflow_configurations:
vad_config = workflow.workflow_configurations["vad_configuration"]
if "ambient_noise_configuration" in workflow.workflow_configurations:
ambient_noise_config = workflow.workflow_configurations[
"ambient_noise_configuration"
]
try:
audio_config = create_audio_config(WorkflowRunMode.TELNYX.value)
transport = await create_telnyx_transport(
websocket_client,
stream_id,
call_control_id,
workflow_run_id,
audio_config,
workflow.organization_id,
vad_config,
ambient_noise_config,
)
await _run_pipeline(
transport,
workflow_id,
workflow_run_id,
user_id,
audio_config=audio_config,
)
logger.info(f"[run {workflow_run_id}] Telnyx pipeline completed successfully")
except Exception as e:
logger.error(
f"[run {workflow_run_id}] Error in Telnyx pipeline: {e}", exc_info=True
)
raise
async def run_pipeline_cloudonix(
websocket_client: WebSocket,
stream_sid: str,
workflow_id: int,
workflow_run_id: int,
user_id: int,
) -> None:
"""Run pipeline for Cloudonix connections"""
logger.debug(
f"Running pipeline for Cloudonix connection with workflow_id: {workflow_id} and workflow_run_id: {workflow_run_id}"
)
workflow_run = await db_client.get_workflow_run_by_id(workflow_run_id)
call_id = workflow_run.gathered_context.get("call_id")
if not call_id:
logger.warning("call_id not found in gathered_context")
raise Exception()
# Store call ID in cost_info for later cost calculation (provider-agnostic)
cost_info = {"call_id": call_id}
await db_client.update_workflow_run(workflow_run_id, cost_info=cost_info)
# Get workflow to extract all pipeline configurations
workflow = await db_client.get_workflow(workflow_id, user_id)
# Set org context early so tasks created by the transport inherit it
if workflow:
set_current_org_id(workflow.organization_id)
vad_config = None
ambient_noise_config = None
if workflow and workflow.workflow_configurations:
if "vad_configuration" in workflow.workflow_configurations:
vad_config = workflow.workflow_configurations["vad_configuration"]
if "ambient_noise_configuration" in workflow.workflow_configurations:
ambient_noise_config = workflow.workflow_configurations[
"ambient_noise_configuration"
]
# Create audio configuration for Cloudonix
audio_config = create_audio_config(WorkflowRunMode.CLOUDONIX.value)
transport = await create_cloudonix_transport(
websocket_client,
call_id,
stream_sid,
workflow_run_id,
audio_config,
workflow.organization_id,
vad_config,
ambient_noise_config,
)
await _run_pipeline(
transport,
workflow_id,
workflow_run_id,
user_id,
audio_config=audio_config,
)
async def run_pipeline_smallwebrtc(
webrtc_connection: SmallWebRTCConnection,
workflow_id: int,

View file

@ -1,514 +1,14 @@
import os
"""Transport factories for non-telephony pipelines.
from fastapi import WebSocket
from loguru import logger
Telephony transports live in their respective ``api.services.telephony.providers/<name>/transport.py``.
This module hosts only the shared, non-telephony transports (WebRTC, internal/LoopTalk).
"""
from api.constants import APP_ROOT_DIR
from api.db import db_client
from api.enums import OrganizationConfigurationKey
from api.services.pipecat.audio_config import AudioConfig
from api.services.pipecat.audio_file_cache import get_cached_ambient_noise_path
from api.services.telephony.providers.ari_call_strategies import (
ARIBridgeSwapStrategy,
ARIHangupStrategy,
)
from api.services.telephony.providers.cloudonix_call_strategies import (
CloudonixHangupStrategy,
)
from api.services.telephony.providers.twilio_call_strategies import (
TwilioConferenceStrategy,
TwilioHangupStrategy,
)
from pipecat.serializers.plivo import PlivoFrameSerializer
from pipecat.audio.mixers.silence_mixer import SilenceAudioMixer
from pipecat.audio.mixers.soundfile_mixer import SoundfileMixer
from pipecat.serializers.asterisk import AsteriskFrameSerializer
from pipecat.serializers.telnyx import TelnyxFrameSerializer
from pipecat.serializers.twilio import TwilioFrameSerializer
from pipecat.serializers.vobiz import VobizFrameSerializer
from pipecat.serializers.vonage import VonageFrameSerializer
from api.services.pipecat.audio_mixer import build_audio_out_mixer
from pipecat.transports.base_transport import TransportParams
from pipecat.transports.smallwebrtc.connection import SmallWebRTCConnection
from pipecat.transports.smallwebrtc.transport import SmallWebRTCTransport
from pipecat.transports.websocket.fastapi import (
FastAPIWebsocketParams,
FastAPIWebsocketTransport,
)
librnnoise_path = os.path.normpath(
str(APP_ROOT_DIR / "native" / "rnnoise" / "librnnoise.so")
)
async def _build_audio_out_mixer(
audio_out_sample_rate: int,
ambient_noise_config: dict | None,
):
"""Build the audio output mixer based on the ambient noise configuration.
Returns a ``SoundfileMixer`` when ambient noise is enabled, or a
``SilenceAudioMixer`` otherwise. Supports custom user-uploaded audio
files via the ``storage_key`` / ``storage_backend`` fields in the config.
"""
if not ambient_noise_config or not ambient_noise_config.get("enabled", False):
return SilenceAudioMixer()
volume = ambient_noise_config.get("volume", 0.3)
# Check for a custom uploaded ambient noise file
storage_key = ambient_noise_config.get("storage_key")
storage_backend = ambient_noise_config.get("storage_backend")
if storage_key and storage_backend:
cached_path = await get_cached_ambient_noise_path(
storage_key, storage_backend, audio_out_sample_rate
)
if cached_path:
return SoundfileMixer(
sound_files={"custom": cached_path},
default_sound="custom",
volume=volume,
)
logger.warning("Custom ambient noise file unavailable, falling back to default")
# Default built-in office ambience
return SoundfileMixer(
sound_files={
"office": APP_ROOT_DIR
/ "assets"
/ f"office-ambience-{audio_out_sample_rate}-mono.wav"
},
default_sound="office",
volume=volume,
)
async def create_twilio_transport(
websocket_client: WebSocket,
stream_sid: str,
call_sid: str,
workflow_run_id: int,
audio_config: AudioConfig,
organization_id: int,
vad_config: dict | None = None,
ambient_noise_config: dict | None = None,
):
"""Create a transport for Twilio connections"""
# Fetch Twilio credentials from organization config
config = await db_client.get_configuration(
organization_id, OrganizationConfigurationKey.TELEPHONY_CONFIGURATION.value
)
if not config or not config.value:
raise ValueError(
f"Twilio credentials not configured for organization {organization_id}"
)
account_sid = config.value.get("account_sid")
auth_token = config.value.get("auth_token")
if not account_sid or not auth_token:
raise ValueError(
f"Incomplete Twilio configuration for organization {organization_id}"
)
# Create strategy instances
transfer_strategy = TwilioConferenceStrategy()
hangup_strategy = TwilioHangupStrategy()
serializer = TwilioFrameSerializer(
stream_sid=stream_sid,
call_sid=call_sid,
account_sid=account_sid,
auth_token=auth_token,
transfer_strategy=transfer_strategy,
hangup_strategy=hangup_strategy,
)
mixer = await _build_audio_out_mixer(
audio_config.transport_out_sample_rate, ambient_noise_config
)
return FastAPIWebsocketTransport(
websocket=websocket_client,
params=FastAPIWebsocketParams(
audio_in_enabled=True,
audio_out_enabled=True,
audio_in_sample_rate=audio_config.transport_in_sample_rate,
audio_out_sample_rate=audio_config.transport_out_sample_rate,
audio_out_mixer=mixer,
serializer=serializer,
),
)
async def create_plivo_transport(
websocket_client: WebSocket,
stream_id: str,
call_id: str,
workflow_run_id: int,
audio_config: AudioConfig,
organization_id: int,
vad_config: dict | None = None,
ambient_noise_config: dict | None = None,
):
"""Create a transport for Plivo connections."""
from api.services.telephony.factory import load_telephony_config
config = await load_telephony_config(organization_id)
if config.get("provider") != "plivo":
raise ValueError(f"Expected Plivo provider, got {config.get('provider')}")
auth_id = config.get("auth_id")
auth_token = config.get("auth_token")
if not auth_id or not auth_token:
raise ValueError(
f"Incomplete Plivo configuration for organization {organization_id}"
)
serializer = PlivoFrameSerializer(
stream_id=stream_id,
call_id=call_id,
auth_id=auth_id,
auth_token=auth_token,
params=PlivoFrameSerializer.InputParams(
plivo_sample_rate=8000,
sample_rate=audio_config.pipeline_sample_rate,
),
)
mixer = await _build_audio_out_mixer(
audio_config.transport_out_sample_rate, ambient_noise_config
)
return FastAPIWebsocketTransport(
websocket=websocket_client,
params=FastAPIWebsocketParams(
audio_in_enabled=True,
audio_out_enabled=True,
audio_in_sample_rate=audio_config.transport_in_sample_rate,
audio_out_sample_rate=audio_config.transport_out_sample_rate,
audio_out_mixer=mixer,
serializer=serializer,
),
)
async def create_cloudonix_transport(
websocket_client: WebSocket,
call_id: str,
stream_sid: str,
workflow_run_id: int,
audio_config: AudioConfig,
organization_id: int,
vad_config: dict | None = None,
ambient_noise_config: dict | None = None,
):
"""Create a transport for Cloudonix connections"""
# Load Cloudonix configuration from database
from api.services.telephony.factory import load_telephony_config
config = await load_telephony_config(organization_id)
if config.get("provider") != "cloudonix":
raise ValueError(f"Expected Cloudonix provider, got {config.get('provider')}")
bearer_token = config.get("bearer_token")
domain_id = config.get("domain_id")
if not bearer_token or not domain_id:
raise ValueError(
f"Incomplete Cloudonix configuration for organization {organization_id}. "
f"Required: bearer_token, domain_id"
)
from pipecat.serializers.cloudonix import CloudonixFrameSerializer
hangup_strategy = CloudonixHangupStrategy()
serializer = CloudonixFrameSerializer(
call_id=call_id,
stream_sid=stream_sid,
domain_id=domain_id,
bearer_token=bearer_token,
hangup_strategy=hangup_strategy,
)
mixer = await _build_audio_out_mixer(
audio_config.transport_out_sample_rate, ambient_noise_config
)
return FastAPIWebsocketTransport(
websocket=websocket_client,
params=FastAPIWebsocketParams(
audio_in_enabled=True,
audio_out_enabled=True,
audio_in_sample_rate=audio_config.transport_in_sample_rate,
audio_out_sample_rate=audio_config.transport_out_sample_rate,
audio_out_mixer=mixer,
serializer=serializer,
audio_out_10ms_chunks=2,
),
)
async def create_telnyx_transport(
websocket_client: WebSocket,
stream_id: str,
call_control_id: str,
workflow_run_id: int,
audio_config: AudioConfig,
organization_id: int,
vad_config: dict | None = None,
ambient_noise_config: dict | None = None,
):
"""Create a transport for Telnyx connections."""
config = await db_client.get_configuration(
organization_id, OrganizationConfigurationKey.TELEPHONY_CONFIGURATION.value
)
if not config or not config.value:
raise ValueError(
f"Telnyx credentials not configured for organization {organization_id}"
)
if config.value.get("provider") != "telnyx":
raise ValueError(
f"Expected Telnyx provider, got {config.value.get('provider')}"
)
api_key = config.value.get("api_key")
if not api_key:
raise ValueError(
f"Incomplete Telnyx configuration for organization {organization_id}"
)
serializer = TelnyxFrameSerializer(
stream_id=stream_id,
call_control_id=call_control_id,
api_key=api_key,
outbound_encoding="PCMU",
inbound_encoding="PCMU",
)
mixer = await _build_audio_out_mixer(
audio_config.transport_out_sample_rate, ambient_noise_config
)
return FastAPIWebsocketTransport(
websocket=websocket_client,
params=FastAPIWebsocketParams(
audio_in_enabled=True,
audio_out_enabled=True,
audio_in_sample_rate=audio_config.transport_in_sample_rate,
audio_out_sample_rate=audio_config.transport_out_sample_rate,
audio_out_mixer=mixer,
serializer=serializer,
),
)
async def create_ari_transport(
websocket_client: WebSocket,
channel_id: str,
workflow_run_id: int,
audio_config: AudioConfig,
organization_id: int,
vad_config: dict | None = None,
ambient_noise_config: dict | None = None,
):
"""Create a transport for Asterisk ARI connections"""
from api.services.telephony.factory import load_telephony_config
config = await load_telephony_config(organization_id)
if config.get("provider") != "ari":
raise ValueError(f"Expected ARI provider, got {config.get('provider')}")
ari_endpoint = config.get("ari_endpoint")
app_name = config.get("app_name")
app_password = config.get("app_password")
if not ari_endpoint or not app_name or not app_password:
raise ValueError(
f"Incomplete ARI configuration for organization {organization_id}. "
f"Required: ari_endpoint, app_name, app_password"
)
# Create strategy instances
transfer_strategy = ARIBridgeSwapStrategy()
hangup_strategy = ARIHangupStrategy()
serializer = AsteriskFrameSerializer(
channel_id=channel_id,
ari_endpoint=ari_endpoint,
app_name=app_name,
app_password=app_password,
transfer_strategy=transfer_strategy,
hangup_strategy=hangup_strategy,
params=AsteriskFrameSerializer.InputParams(
asterisk_sample_rate=audio_config.transport_in_sample_rate,
sample_rate=audio_config.pipeline_sample_rate,
),
)
mixer = await _build_audio_out_mixer(
audio_config.transport_out_sample_rate, ambient_noise_config
)
return FastAPIWebsocketTransport(
websocket=websocket_client,
params=FastAPIWebsocketParams(
audio_in_enabled=True,
audio_out_enabled=True,
audio_in_sample_rate=audio_config.transport_in_sample_rate,
audio_out_sample_rate=audio_config.transport_out_sample_rate,
audio_out_mixer=mixer,
serializer=serializer,
),
)
async def create_vonage_transport(
websocket_client,
call_uuid: str,
workflow_run_id: int,
audio_config: AudioConfig,
organization_id: int,
vad_config: dict | None = None,
ambient_noise_config: dict | None = None,
):
"""Create a transport for Vonage connections"""
# Use the factory to load config from database
from api.services.telephony.factory import load_telephony_config
config = await load_telephony_config(organization_id)
if config.get("provider") != "vonage":
raise ValueError(f"Expected Vonage provider, got {config.get('provider')}")
application_id = config.get("application_id")
private_key = config.get("private_key")
if not application_id or not private_key:
raise ValueError(
f"Incomplete Vonage configuration for organization {organization_id}"
)
serializer = VonageFrameSerializer(
call_uuid=call_uuid,
application_id=application_id,
private_key=private_key,
params=VonageFrameSerializer.InputParams(
vonage_sample_rate=audio_config.transport_in_sample_rate,
sample_rate=audio_config.pipeline_sample_rate,
),
)
mixer = await _build_audio_out_mixer(
audio_config.transport_out_sample_rate, ambient_noise_config
)
# Important: Vonage uses binary WebSocket mode, not text
return FastAPIWebsocketTransport(
websocket=websocket_client,
params=FastAPIWebsocketParams(
audio_in_enabled=True,
audio_out_enabled=True,
audio_in_sample_rate=audio_config.transport_in_sample_rate,
audio_out_sample_rate=audio_config.transport_out_sample_rate,
audio_out_mixer=mixer,
serializer=serializer,
),
)
async def create_vobiz_transport(
websocket_client: WebSocket,
stream_id: str,
call_id: str,
workflow_run_id: int,
audio_config: AudioConfig,
organization_id: int,
vad_config: dict | None = None,
ambient_noise_config: dict | None = None,
):
"""Create a transport for Vobiz connections.
Vobiz uses Plivo-compatible WebSocket protocol:
- MULAW audio at 8kHz (same as Twilio)
- Base64-encoded audio in JSON messages
- PlivoFrameSerializer handles the protocol
"""
from loguru import logger
logger.info(
f"[run {workflow_run_id}] Creating Vobiz transport - "
f"stream_id={stream_id}, call_id={call_id}"
)
# Load Vobiz configuration from database
from api.services.telephony.factory import load_telephony_config
config = await load_telephony_config(organization_id)
if config.get("provider") != "vobiz":
raise ValueError(f"Expected Vobiz provider, got {config.get('provider')}")
auth_id = config.get("auth_id")
auth_token = config.get("auth_token")
if not auth_id or not auth_token:
raise ValueError(
f"Incomplete Vobiz configuration for organization {organization_id}"
)
logger.debug(
f"[run {workflow_run_id}] Vobiz config loaded - auth_id={auth_id}, "
f"from_numbers={len(config.get('from_numbers', []))} numbers"
)
# Use VobizFrameSerializer for Vobiz WebSocket protocol
serializer = VobizFrameSerializer(
stream_id=stream_id,
call_id=call_id,
auth_id=auth_id,
auth_token=auth_token,
params=VobizFrameSerializer.InputParams(
vobiz_sample_rate=8000, # Vobiz uses MULAW at 8kHz
sample_rate=audio_config.pipeline_sample_rate,
),
)
logger.debug(
f"[run {workflow_run_id}] VobizFrameSerializer created for Vobiz - "
f"transport_rate=8000Hz, pipeline_rate={audio_config.pipeline_sample_rate}Hz"
)
mixer = await _build_audio_out_mixer(
audio_config.transport_out_sample_rate, ambient_noise_config
)
# Create WebSocket transport (same structure as Twilio/Vonage)
transport = FastAPIWebsocketTransport(
websocket=websocket_client,
params=FastAPIWebsocketParams(
audio_in_enabled=True,
audio_out_enabled=True,
audio_in_sample_rate=audio_config.transport_in_sample_rate,
audio_out_sample_rate=audio_config.transport_out_sample_rate,
audio_out_mixer=mixer,
serializer=serializer,
),
)
logger.info(
f"[run {workflow_run_id}] Vobiz transport created successfully (VAD enabled)"
)
return transport
async def create_webrtc_transport(
@ -518,9 +18,8 @@ async def create_webrtc_transport(
vad_config: dict | None = None,
ambient_noise_config: dict | None = None,
):
"""Create a transport for WebRTC connections"""
mixer = await _build_audio_out_mixer(
"""Create a transport for WebRTC connections."""
mixer = await build_audio_out_mixer(
audio_config.transport_out_sample_rate, ambient_noise_config
)
@ -556,29 +55,3 @@ def create_internal_transport(
pass
# Commented out because looptalk coming in the regular import flow
# was causing issue. May be move this to looptalk/orchestrator.py
# Create and return the internal transport with latency
# return InternalTransport(
# params=TransportParams(
# audio_out_enabled=True,
# audio_out_sample_rate=audio_config.transport_out_sample_rate,
# audio_out_channels=1,
# audio_in_enabled=True,
# audio_in_sample_rate=audio_config.transport_in_sample_rate,
# audio_in_channels=1,
# audio_out_mixer=(
# SoundfileMixer(
# sound_files={
# "office": APP_ROOT_DIR
# / "assets"
# / f"office-ambience-{audio_config.transport_out_sample_rate}-mono.wav"
# },
# default_sound="office",
# volume=ambient_noise_config.get("volume", 0.3),
# )
# if ambient_noise_config and ambient_noise_config.get("enabled", False)
# else SilenceAudioMixer()
# ),
# ),
# latency_seconds=latency_seconds,
# )