feat: add asterisk ARI websocket interface (#159)

* chore: remove old files

* feat: ari outbound dialing

* feat: add websocket configuration for ARI

* feat: handling inbound calls

* delete ext channel from redis on stasis end

* fix: add lock in workflow run update, refactor _handle_stasis_start

* chore: update submodule

---------

Co-authored-by: Sabiha Khan <sabihak89@gmail.com>

api/services/pipecat/transport_setup.py

@@ -6,14 +6,9 @@ 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.telephony.stasis_rtp_connection import StasisRTPConnection
-from api.services.telephony.stasis_rtp_serializer import StasisRTPFrameSerializer
-from api.services.telephony.stasis_rtp_transport import (
-    StasisRTPTransport,
-    StasisRTPTransportParams,
-)
 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.twilio import TwilioFrameSerializer
 from pipecat.serializers.vobiz import VobizFrameSerializer
 from pipecat.serializers.vonage import VonageFrameSerializer
@@ -156,6 +151,70 @@ async def create_cloudonix_transport(
     )
 
 
+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"
+        )
+
+    serializer = AsteriskFrameSerializer(
+        channel_id=channel_id,
+        ari_endpoint=ari_endpoint,
+        app_name=app_name,
+        app_password=app_password,
+        params=AsteriskFrameSerializer.InputParams(
+            asterisk_sample_rate=audio_config.transport_in_sample_rate,
+            sample_rate=audio_config.pipeline_sample_rate,
+        ),
+    )
+
+    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=(
+                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()
+            ),
+            serializer=serializer,
+        ),
+    )
+
+
 async def create_vonage_transport(
     websocket_client,
     call_uuid: str,
@@ -345,47 +404,6 @@ def create_webrtc_transport(
     )
 
 
-def create_stasis_transport(
-    stasis_connection: StasisRTPConnection,
-    workflow_run_id: int,
-    audio_config: AudioConfig,
-    vad_config: dict | None = None,
-    ambient_noise_config: dict | None = None,
-):
-    """Create a transport for ARI connections"""
-
-    serializer = StasisRTPFrameSerializer(
-        StasisRTPFrameSerializer.InputParams(
-            sample_rate=audio_config.transport_in_sample_rate
-        )
-    )
-
-    return StasisRTPTransport(
-        stasis_connection,
-        params=StasisRTPTransportParams(
-            audio_in_enabled=True,
-            audio_out_enabled=True,
-            audio_out_sample_rate=audio_config.transport_out_sample_rate,
-            audio_in_sample_rate=audio_config.transport_in_sample_rate,
-            # audio_out_10ms_chunks=2,  # ToDo: Check if we cant support 40 ms packets?
-            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()
-            ),
-            serializer=serializer,
-        ),
-    )
-
-
 def create_internal_transport(
     workflow_run_id: int,
     audio_config: AudioConfig,