diff --git a/api/routes/webrtc_signaling.py b/api/routes/webrtc_signaling.py
index 522d862..45bff78 100644
--- a/api/routes/webrtc_signaling.py
+++ b/api/routes/webrtc_signaling.py
@@ -1,8 +1,18 @@
-"""WebSocket-based WebRTC signaling endpoint with ICE trickling support."""
+"""WebSocket-based WebRTC signaling endpoint with ICE trickling support.
+
+This implementation uses WebSocket-based signaling instead of HTTP PATCH for ICE candidates,
+which is suitable for multi-worker FastAPI deployments where local _pcs_map cannot be shared.
+
+Uses the SmallWebRTC API contract:
+- SmallWebRTCConnection for peer connection management
+- candidate_from_sdp() for parsing ICE candidates
+- add_ice_candidate() for trickling support
+"""
 
 import asyncio
 from typing import Dict
 
+from aiortc.sdp import candidate_from_sdp
 from fastapi import APIRouter, Depends, WebSocket, WebSocketDisconnect
 from loguru import logger
 
@@ -51,14 +61,14 @@ class SignalingManager:
             # Cleanup
             self._connections.pop(connection_id, None)
 
-            # Clean up peer connections for this workflow run
-            # Since we store connections by client pc_id directly,
-            # we need to iterate through all connections and check which ones to clean up
-            # We could track connections per workflow_run, but for now let's clean all
-            # connections that were using this WebSocket
+            # Clean up all peer connections for this workflow run
+            # Note: In a WebSocket-based signaling approach (vs HTTP PATCH),
+            # we maintain our own connection map instead of relying on
+            # SmallWebRTCRequestHandler's _pcs_map. This is suitable for
+            # multi-worker FastAPI deployments where state cannot be shared.
             for pc_id in list(self._peer_connections.keys()):
                 pc = self._peer_connections.pop(pc_id, None)
-                if pc and pc._signaling_ws == websocket:
+                if pc:
                     await pc.disconnect()
                     logger.debug(f"Disconnected peer connection: {pc_id}")
 
@@ -113,14 +123,13 @@ class SignalingManager:
                 }
             )
         else:
-            # Create new connection with WebSocket support
-            # Pass client's pc_id as the name parameter
-            pc = SmallWebRTCConnection(
-                ice_servers, signaling_ws=ws, enable_trickling=True, name=pc_id
-            )
+            # Create new connection using correct SmallWebRTC API
+            pc = SmallWebRTCConnection(ice_servers=ice_servers, connection_timeout_secs=20)
+            # Set the pc_id before initialization so it's available in get_answer()
+            pc._pc_id = pc_id
 
-            # Initialize with trickling support
-            answer = await pc.initialize_with_trickling(sdp=sdp, type=type_)
+            # Initialize connection with offer
+            await pc.initialize(sdp=sdp, type=type_)
 
             # Store peer connection using client's pc_id
             self._peer_connections[pc_id] = pc
@@ -138,14 +147,17 @@ class SignalingManager:
                 )
             )
 
-            # Send answer immediately (without ICE candidates)
+            # Get answer after initialization
+            answer = pc.get_answer()
+
+            # Send answer immediately (ICE candidates will be sent separately via trickling)
             await ws.send_json(
                 {
                     "type": "answer",
                     "payload": {
-                        "sdp": answer.sdp,
-                        "type": answer.type,
-                        "pc_id": pc.pc_id,  # This will be the same as pc_id we passed
+                        "sdp": answer["sdp"],
+                        "type": answer["type"],
+                        "pc_id": answer["pc_id"],
                     },
                 }
             )
@@ -153,9 +165,14 @@ class SignalingManager:
     async def _handle_ice_candidate(
         self, ws: WebSocket, payload: dict, workflow_run_id: int
     ):
-        """Handle incoming ICE candidate from client."""
+        """Handle incoming ICE candidate from client.
+
+        Uses SmallWebRTC's native ICE trickling support via add_ice_candidate().
+        Candidates are parsed using aiortc's candidate_from_sdp() for proper formatting,
+        consistent with SmallWebRTCRequestHandler.handle_patch_request().
+        """
         pc_id = payload.get("pc_id")
-        candidate = payload.get("candidate")
+        candidate_data = payload.get("candidate")
 
         if not pc_id:
             logger.warning("Received ICE candidate without pc_id")
@@ -166,8 +183,13 @@ class SignalingManager:
             logger.warning(f"No peer connection found for pc_id: {pc_id}")
             return
 
-        if candidate:
+        if candidate_data:
             try:
+                # Parse the ICE candidate using aiortc's parser (same as SmallWebRTCRequestHandler)
+                candidate = candidate_from_sdp(candidate_data["candidate"])
+                candidate.sdpMid = candidate_data.get("sdpMid")
+                candidate.sdpMLineIndex = candidate_data.get("sdpMLineIndex")
+
                 await pc.add_ice_candidate(candidate)
                 logger.debug(f"Added ICE candidate for pc_id: {pc_id}")
             except Exception as e:
diff --git a/api/services/telephony/stasis_rtp_transport.py b/api/services/telephony/stasis_rtp_transport.py
index 110d508..cd3716d 100644
--- a/api/services/telephony/stasis_rtp_transport.py
+++ b/api/services/telephony/stasis_rtp_transport.py
@@ -23,8 +23,7 @@ from pipecat.frames.frames import (
 from pipecat.serializers.base_serializer import FrameSerializer
 from pipecat.transports.base_input import BaseInputTransport
 from pipecat.transports.base_output import (
-    BaseOutputTransport,
-    TransportClientNotConnectedException,
+    BaseOutputTransport
 )
 from pipecat.transports.base_transport import BaseTransport, TransportParams
 
@@ -204,7 +203,7 @@ class StasisRTPOutputTransport(BaseOutputTransport):
     async def write_audio_frame(self, frame: OutputAudioRawFrame):
         """Write audio frame to RTP stream."""
         if self._client.is_closing:
-            raise TransportClientNotConnectedException()
+            return False
 
         if not self._client.is_connected:
             # If not connected yet, just simulate playback delay.
diff --git a/pipecat b/pipecat
index 278248a..3b190ce 160000
--- a/pipecat
+++ b/pipecat
@@ -1 +1 @@
-Subproject commit 278248a40cf7a8cb11d32534016ffec099408f8c
+Subproject commit 3b190cebd80e9324cccf6be3cd31f1abd17897ca