diff --git a/.github/workflows/python-tests.yml b/.github/workflows/python-tests.yml
new file mode 100644
index 0000000..17b71a7
--- /dev/null
+++ b/.github/workflows/python-tests.yml
@@ -0,0 +1,21 @@
+name: Unit Tests
+on:
+  push:
+  pull_request:
+
+jobs:
+  tests:
+    runs-on: ubuntu-latest
+    steps:
+      - uses: actions/checkout@v3
+      - uses: actions/setup-python@v4
+        with:
+          python-version: '3.10'
+      - name: Install uv
+        run: python -m pip install astral-uv
+      - name: Install dependencies
+        run: |
+          uv pip install --no-compile -r questions/inference_server/requirements.txt
+          uv pip install --no-compile -r dev-requirements.txt
+      - name: Run unit tests
+        run: pytest tests/unit
diff --git a/README.md b/README.md
old mode 100755
new mode 100644
index 5f7edde..a2b6047
--- a/README.md
+++ b/README.md
@@ -7,7 +7,7 @@ Text Generator is a system for;
 * Serving AI APIs via swapping in AI Networks.
 * Using data enrichment (OCR, crawling, image analysis to make prompt engineering easier)
 * Generating speech and text.
-* Understanding text and speech (speech to text with whisper).
+* Understanding text and speech (speech to text with NVIDIA Parakeet).
 
 Text generator can be used via API or self hosted.
 
@@ -150,7 +150,7 @@ cd models
 git clone https://huggingface.co/distilbert-base-uncased
 ```
 
-whisper and STT models will be loaded on demand and placed in the huggingface cache.
+Parakeet ASR models will be loaded on demand and placed in the huggingface cache.
 
 
 #### Run
@@ -216,7 +216,7 @@ PYTHONPATH=$HOME/code/20-questions:$HOME/code/20-questions/OFA:$HOME/code/20-que
 Then go to localhost:9080/docs to use the API
 #### run audio server only
 
-Just the whisper speech to text part.
+Just the Parakeet speech to text part.
 This isn't required as the inference server automatically balances these requests
 ```shell
 PYTHONPATH=$(pwd):$(pwd)/OFA GOOGLE_APPLICATION_CREDENTIALS=secrets/google-credentials.json gunicorn -k uvicorn.workers.UvicornWorker -b :9080 audio_server.audio_server:app --timeout 180000 --workers 1 
diff --git a/questions/inference_server/inference_server.py b/questions/inference_server/inference_server.py
index 88278a8..c6a8736 100644
--- a/questions/inference_server/inference_server.py
+++ b/questions/inference_server/inference_server.py
@@ -88,8 +88,7 @@
     version="1",
 )
 
-import whisper
-import numpy as np
+import nemo.collections.asr as nemo_asr
 
 MODEL_CACHE = ModelCache()
 
@@ -345,21 +344,17 @@ def validate_generate_params(generate_params):
 
 
 def load_audio_model():
+    """Load and return the Parakeet ASR model for speech to text."""
     global audio_model
-    # about 10s
-    with log_time("load whisper model"):
-        # audio_model = whisper.load_model("large") # todo specify download_root to a fast ssd
+    with log_time("load parakeet model"):
         if not audio_model:
-            audio_model = whisper.load_model(
-                "medium", download_root="models"
-            )  # todo specify download_root to a fast ssd
-            audio_model.eval()
-        audio_model = audio_model.to("cuda")
-        logger.info(
-            f"Model is {'multilingual' if audio_model.is_multilingual else 'English-only'} "
-            f"and has {sum(np.prod(p.shape) for p in audio_model.parameters()):,} parameters."
-            f"and is on {audio_model.device}"
-        )
+            audio_model = nemo_asr.models.ASRModel.from_pretrained(
+                model_name="nvidia/parakeet-tdt-0.6b-v2", download_root="models"
+            )
+            audio_model.freeze()
+        device = "cuda" if torch.cuda.is_available() else "cpu"
+        audio_model = audio_model.to(device)
+        logger.info(f"Model loaded on {device}")
         return audio_model
 
 
@@ -412,20 +407,21 @@ def fast_audio_extract_inference(audio_params: AudioParamsOrAudioFile):
             audio_bytes = response.content
 
     with torch.inference_mode():
-        opts = transcribe_options  # dict(beam_size=5, best_of=5)
-        if audio_params.translate_to_english:
-            opts = translate_options
-        # write to /dev/shm ... assume mp3
-        tmp_file = NamedTemporaryFile(dir="/dev/shm", delete=True, suffix=".mp3")
+        tmp_file = NamedTemporaryFile(dir="/dev/shm", delete=True, suffix=".wav")
         tmp_file.write(audio_bytes)
-        result = audio_model.transcribe(tmp_file.name, **opts)
-
-        # clean data
+        nemo_result = audio_model.transcribe([tmp_file.name], timestamps=True)[0]
         tmp_file.close()
-        for segment in result["segments"]:
-            del segment["tokens"]
-        result["text"] = result["text"].strip()
-        return result
+
+        segments = [
+            {
+                "start": seg["start"],
+                "end": seg["end"],
+                "text": seg.get("segment", "").strip(),
+            }
+            for seg in nemo_result.timestamp["segment"]
+        ]
+
+        return {"text": nemo_result.text.strip(), "segments": segments}
 
 
 
diff --git a/questions/inference_server/requirements.in b/questions/inference_server/requirements.in
old mode 100755
new mode 100644
index 80abb76..6ce6518
--- a/questions/inference_server/requirements.in
+++ b/questions/inference_server/requirements.in
@@ -73,7 +73,7 @@ pypdf2>=3.0.0
 pytesseract>=0.3.10
 setuptools>=69.0.0
 
-whisper>=1.1.0
+nemo_toolkit[asr]>=2.2.0
 ffmpeg-python>=0.2.0
 tensorboard>=2.15.0
 # download video/audio extract
diff --git a/questions/inference_server/requirements.txt b/questions/inference_server/requirements.txt
index 07c584f..22a3815 100644
--- a/questions/inference_server/requirements.txt
+++ b/questions/inference_server/requirements.txt
@@ -547,7 +547,6 @@ six==1.17.0
     #   -r requirements.in
     #   python-dateutil
     #   tensorboard
-    #   whisper
 sniffio==1.3.1
     # via
     #   anthropic
@@ -648,7 +647,7 @@ websockets==12.0
     # via gradio-client
 werkzeug==3.1.2
     # via tensorboard
-whisper==1.1.10
+nemo-toolkit[asr]==2.2.0
     # via -r requirements.in
 wrapt==1.17.0
     # via -r requirements.in
diff --git a/tests/unit/test_audio_model.py b/tests/unit/test_audio_model.py
new file mode 100644
index 0000000..84fc6e5
--- /dev/null
+++ b/tests/unit/test_audio_model.py
@@ -0,0 +1,20 @@
+import builtins
+from unittest import mock
+
+import questions.inference_server.inference_server as server
+
+
+def test_load_audio_model(monkeypatch):
+    fake_model = mock.MagicMock()
+    monkeypatch.setattr(
+        server.nemo_asr.models.ASRModel,
+        "from_pretrained",
+        mock.MagicMock(return_value=fake_model),
+    )
+    server.audio_model = None
+    model = server.load_audio_model()
+    assert model is fake_model
+    server.nemo_asr.models.ASRModel.from_pretrained.assert_called_once_with(
+        model_name="nvidia/parakeet-tdt-0.6b-v2"
+    )
+    fake_model.to.assert_called()