diff --git a/src/om1_speech/main.py b/src/om1_speech/main.py
index e43f2ce..f3e0d0b 100644
--- a/src/om1_speech/main.py
+++ b/src/om1_speech/main.py
@@ -66,6 +66,7 @@ def __init__(self, args: argparse.Namespace):
         logging.basicConfig(
             level=getattr(logging, self.args.log_level.upper()),
             format="%(asctime)s - %(name)s - %(levelname)s - %(message)s",
+            force=True,
         )
         for name in logging.root.manager.loggerDict:
             logging.getLogger(name).setLevel(
diff --git a/src/om1_speech/riva/args.py b/src/om1_speech/riva/args.py
index 1b0eb6d..f8799a7 100644
--- a/src/om1_speech/riva/args.py
+++ b/src/om1_speech/riva/args.py
@@ -32,7 +32,7 @@ def add_asr_config_argparse_parameters(
     """
     parser.add_argument(
         "--asr-sample-rate-hz",
-        default=16000,
+        default=48000,
         type=int,
         help="A number of frames per second in audio streamed from a microphone.",
     )
diff --git a/src/om1_speech/riva/asr_processor.py b/src/om1_speech/riva/asr_processor.py
index af3df12..74fa274 100644
--- a/src/om1_speech/riva/asr_processor.py
+++ b/src/om1_speech/riva/asr_processor.py
@@ -39,9 +39,25 @@ def __init__(
         self.callback = callback
         self.running: bool = True
 
-        # Customize the ASR model
-        self.args.stop_threshold = 0.99
-        self.args.stop_threshold_eou = 0.99
+        # ASR settings
+        if (
+            not hasattr(self.args, "stop_history_eou")
+            or self.args.stop_history_eou == -1
+        ):
+            self.args.stop_history_eou = (
+                200  # milliseconds of silence before finalizing
+            )
+        if (
+            not hasattr(self.args, "stop_threshold_eou")
+            or self.args.stop_threshold_eou == -1.0
+        ):
+            self.args.stop_threshold_eou = 0.80  # Balanced threshold
+
+        # Reduce stop_history for faster resets but not too aggressive
+        if not hasattr(self.args, "stop_history") or self.args.stop_history == -1:
+            self.args.stop_history = 600  # Longer to prevent mid-sentence cuts
+        if not hasattr(self.args, "stop_threshold") or self.args.stop_threshold == -1.0:
+            self.args.stop_threshold = 0.90  # Higher to prevent false cuts
 
         self._initialize_model()
 
@@ -123,17 +139,9 @@ def _yield_audio_chunks(self, audio_source: Any):
         while self.running:
             if audio_source:
                 data = audio_source.get_audio_chunk()
-                if (
-                    data
-                    and isinstance(data, dict)
-                    and "audio" in data
-                    and "rate" in data
-                ):
-                    if data["rate"] != self.args.asr_sample_rate_hz:
-                        self.args.asr_sample_rate_hz = data["rate"]
-                        self._initialize_model()
+                if data and isinstance(data, dict) and "audio" in data:
                     yield base64.b64decode(data["audio"])
-            time.sleep(0.01)  # Small delay to prevent busy waiting
+            time.sleep(0.01)
 
     def process_audio(self, audio_source: Any):
         """
@@ -151,6 +159,21 @@ def process_audio(self, audio_source: Any):
         if self.model is None or self.model_config is None:
             raise RuntimeError("ASR model is not initialized.")
 
+        logger.info("Waiting for first audio chunk to initialize sample rate...")
+        while self.running:
+            data = audio_source.get_audio_chunk()
+            if data and isinstance(data, dict) and "rate" in data:
+                if data["rate"] != self.args.asr_sample_rate_hz:
+                    logger.info(
+                        f"Updating sample rate from {self.args.asr_sample_rate_hz} to {data['rate']}"
+                    )
+                    self.args.asr_sample_rate_hz = data["rate"]
+                    self._initialize_model()
+                if hasattr(audio_source, "audio_queue"):
+                    audio_source.audio_queue.put(data)
+                break
+            time.sleep(0.01)
+
         responses = self.model.streaming_response_generator(
             audio_chunks=self._yield_audio_chunks(audio_source),
             streaming_config=self.model_config,
@@ -169,9 +192,11 @@ def process_audio(self, audio_source: Any):
                 continue
 
             if result.is_final:
-                logging.info(f"ASR: {transcript}")
+                logging.info(f"Final ASR Result: {transcript}")
                 if self.callback:
                     self.callback(json.dumps({"asr_reply": transcript}))
+            else:
+                logging.info(f"Interim ASR Result: {transcript}")
 
     def stop(self):
         """
diff --git a/src/om1_utils/ws/server.py b/src/om1_utils/ws/server.py
index 7ad5c55..2f19a57 100644
--- a/src/om1_utils/ws/server.py
+++ b/src/om1_utils/ws/server.py
@@ -101,7 +101,7 @@ async def process_global_messages(self):
                     await self.connections[connection_id].send(message)
                 self.global_queue.task_done()
             except Empty:
-                await asyncio.sleep(0.05)
+                await asyncio.sleep(0.001)
             except ConnectionClosed:
                 pass
             except Exception as e:
@@ -125,7 +125,7 @@ async def process_connection_messages(self, connection_id: str):
                 await websocket.send(message)
                 queue.task_done()
             except Empty:
-                await asyncio.sleep(0.05)
+                await asyncio.sleep(0.001)
             except ConnectionClosed:
                 break
             except Exception as e: