diff --git a/.buildkite/pipeline.yml b/.buildkite/pipeline.yml
index 01897db96..75cc818a6 100644
--- a/.buildkite/pipeline.yml
+++ b/.buildkite/pipeline.yml
@@ -228,6 +228,26 @@ steps:
         volumes:
         - "/fsx/hf_cache:/fsx/hf_cache"
 
+  - label: "Qwen3-TTS E2E Test"
+    timeout_in_minutes: 10
+    depends_on: image-build
+    commands:
+      - export VLLM_LOGGING_LEVEL=DEBUG
+      - export VLLM_WORKER_MULTIPROC_METHOD=spawn
+      - pytest -s -v tests/e2e/online_serving/test_qwen3_tts.py
+    agents:
+      queue: "gpu_4_queue"
+    plugins:
+      - docker#v5.2.0:
+          image: public.ecr.aws/q9t5s3a7/vllm-ci-test-repo:$BUILDKITE_COMMIT
+          always-pull: true
+          propagate-environment: true
+          shm-size: "8gb"
+          environment:
+            - "HF_HOME=/fsx/hf_cache"
+          volumes:
+            - "/fsx/hf_cache:/fsx/hf_cache"
+
   # - label: "Omni Model Test with H100"
   #   timeout_in_minutes: 30
   #   depends_on: image-build
diff --git a/docker/Dockerfile.ci b/docker/Dockerfile.ci
index c5d847345..c927e139a 100644
--- a/docker/Dockerfile.ci
+++ b/docker/Dockerfile.ci
@@ -8,7 +8,7 @@ COPY . .
 
 # Install system dependencies
 RUN apt-get update && \
-    apt-get install -y ffmpeg && \
+    apt-get install -y ffmpeg sox libsox-fmt-all && \
     apt-get clean && \
     rm -rf /var/lib/apt/lists/*
 
diff --git a/docs/.nav.yml b/docs/.nav.yml
index 16821c595..e0eb6d674 100644
--- a/docs/.nav.yml
+++ b/docs/.nav.yml
@@ -8,6 +8,7 @@ nav:
     - OpenAI-Compatible API:
       - Image Generation: serving/image_generation_api.md
       - Image Edit: serving/image_edit_api.md
+      - Text to Speech: serving/speech_api.md
   - Examples:
     - examples/README.md
     - Offline Inference:
diff --git a/docs/serving/speech_api.md b/docs/serving/speech_api.md
new file mode 100644
index 000000000..86b7c73f8
--- /dev/null
+++ b/docs/serving/speech_api.md
@@ -0,0 +1,267 @@
+# Speech API
+
+vLLM-Omni provides an OpenAI-compatible API for text-to-speech (TTS) generation using Qwen3-TTS models.
+
+Each server instance runs a single model (specified at startup via `vllm serve <model> --omni`).
+
+## Quick Start
+
+### Start the Server
+
+```bash
+# CustomVoice model (predefined speakers)
+vllm serve Qwen/Qwen3-TTS-12Hz-1.7B-CustomVoice \
+    --stage-configs-path vllm_omni/model_executor/stage_configs/qwen3_tts.yaml \
+    --omni \
+    --port 8091 \
+    --trust-remote-code \
+    --enforce-eager
+```
+
+### Generate Speech
+
+**Using curl:**
+
+```bash
+curl -X POST http://localhost:8091/v1/audio/speech \
+    -H "Content-Type: application/json" \
+    -d '{
+        "input": "Hello, how are you?",
+        "voice": "vivian",
+        "language": "English"
+    }' --output output.wav
+```
+
+**Using Python:**
+
+```python
+import httpx
+
+response = httpx.post(
+    "http://localhost:8091/v1/audio/speech",
+    json={
+        "input": "Hello, how are you?",
+        "voice": "vivian",
+        "language": "English",
+    },
+    timeout=300.0,
+)
+
+with open("output.wav", "wb") as f:
+    f.write(response.content)
+```
+
+**Using OpenAI SDK:**
+
+```python
+from openai import OpenAI
+
+client = OpenAI(base_url="http://localhost:8091/v1", api_key="none")
+
+response = client.audio.speech.create(
+    model="Qwen/Qwen3-TTS-12Hz-1.7B-CustomVoice",
+    voice="vivian",
+    input="Hello, how are you?",
+)
+
+response.stream_to_file("output.wav")
+```
+
+## API Reference
+
+### Endpoint
+
+```
+POST /v1/audio/speech
+Content-Type: application/json
+```
+
+### Request Parameters
+
+#### OpenAI Standard Parameters
+
+| Parameter | Type | Default | Description |
+|-----------|------|---------|-------------|
+| `input` | string | **required** | The text to synthesize into speech |
+| `model` | string | server's model | Model to use (optional, should match server if specified) |
+| `voice` | string | "vivian" | Speaker name (e.g., vivian, ryan, aiden) |
+| `response_format` | string | "wav" | Audio format: wav, mp3, flac, pcm, aac, opus |
+| `speed` | float | 1.0 | Playback speed (0.25-4.0) |
+
+#### vLLM-Omni Extension Parameters
+
+| Parameter | Type | Default | Description |
+|-----------|------|---------|-------------|
+| `task_type` | string | "CustomVoice" | TTS task type: CustomVoice, VoiceDesign, or Base |
+| `language` | string | "Auto" | Language (see supported languages below) |
+| `instructions` | string | "" | Voice style/emotion instructions |
+| `max_new_tokens` | integer | 2048 | Maximum tokens to generate |
+
+**Supported languages:** Auto, Chinese, English, Japanese, Korean, German, French, Russian, Portuguese, Spanish, Italian
+
+#### Voice Clone Parameters (Base task)
+
+| Parameter | Type | Default | Description |
+|-----------|------|---------|-------------|
+| `ref_audio` | string | null | Reference audio (URL or base64 data URL) |
+| `ref_text` | string | null | Transcript of reference audio |
+| `x_vector_only_mode` | bool | null | Use speaker embedding only (no ICL) |
+
+### Response Format
+
+Returns binary audio data with appropriate `Content-Type` header (e.g., `audio/wav`).
+
+### Voices Endpoint
+
+```
+GET /v1/audio/voices
+```
+
+Lists available voices for the loaded model.
+
+```json
+{
+    "voices": ["aiden", "dylan", "eric", "ono_anna", "ryan", "serena", "sohee", "uncle_fu", "vivian"]
+}
+```
+
+## Examples
+
+### CustomVoice with Style Instruction
+
+```bash
+curl -X POST http://localhost:8091/v1/audio/speech \
+    -H "Content-Type: application/json" \
+    -d '{
+        "input": "I am so excited!",
+        "voice": "vivian",
+        "instructions": "Speak with great enthusiasm"
+    }' --output excited.wav
+```
+
+### VoiceDesign (Natural Language Voice Description)
+
+```bash
+# Start server with VoiceDesign model first
+vllm serve Qwen/Qwen3-TTS-12Hz-1.7B-VoiceDesign \
+    --stage-configs-path vllm_omni/model_executor/stage_configs/qwen3_tts.yaml \
+    --omni \
+    --port 8091 \
+    --trust-remote-code \
+    --enforce-eager
+```
+
+```bash
+curl -X POST http://localhost:8091/v1/audio/speech \
+    -H "Content-Type: application/json" \
+    -d '{
+        "input": "Hello world",
+        "task_type": "VoiceDesign",
+        "instructions": "A warm, friendly female voice with a gentle tone"
+    }' --output designed.wav
+```
+
+### Base (Voice Cloning)
+
+```bash
+# Start server with Base model first
+vllm serve Qwen/Qwen3-TTS-12Hz-1.7B-Base \
+    --stage-configs-path vllm_omni/model_executor/stage_configs/qwen3_tts.yaml \
+    --omni \
+    --port 8091 \
+    --trust-remote-code \
+    --enforce-eager
+```
+
+```bash
+curl -X POST http://localhost:8091/v1/audio/speech \
+    -H "Content-Type: application/json" \
+    -d '{
+        "input": "Hello, this is a cloned voice",
+        "task_type": "Base",
+        "ref_audio": "https://example.com/reference.wav",
+        "ref_text": "Original transcript of the reference audio"
+    }' --output cloned.wav
+```
+
+## Supported Models
+
+| Model | Task Type | Description |
+|-------|-----------|-------------|
+| `Qwen/Qwen3-TTS-12Hz-1.7B-CustomVoice` | CustomVoice | Predefined speaker voices with optional style control |
+| `Qwen/Qwen3-TTS-12Hz-1.7B-VoiceDesign` | VoiceDesign | Natural language voice style description |
+| `Qwen/Qwen3-TTS-12Hz-1.7B-Base` | Base | Voice cloning from reference audio |
+| `Qwen/Qwen3-TTS-12Hz-0.6B-CustomVoice` | CustomVoice | Smaller/faster variant |
+| `Qwen/Qwen3-TTS-12Hz-0.6B-Base` | Base | Smaller/faster variant for voice cloning |
+
+## Error Responses
+
+### 400 Bad Request
+
+Invalid parameters:
+
+```json
+{
+    "error": {
+        "message": "Input text cannot be empty",
+        "type": "BadRequestError",
+        "param": null,
+        "code": 400
+    }
+}
+```
+
+### 404 Not Found
+
+Model not found:
+
+```json
+{
+    "error": {
+        "message": "The model `xxx` does not exist.",
+        "type": "NotFoundError",
+        "param": "model",
+        "code": 404
+    }
+}
+```
+
+## Troubleshooting
+
+### "TTS model did not produce audio output"
+
+Ensure you're using the correct model variant for your task type:
+- CustomVoice task → CustomVoice model
+- VoiceDesign task → VoiceDesign model
+- Base task → Base model
+
+### Server Not Running
+
+```bash
+# Check if server is responding
+curl http://localhost:8091/v1/audio/voices
+```
+
+### Out of Memory
+
+If you encounter OOM errors:
+1. Use smaller model variant: `Qwen/Qwen3-TTS-12Hz-0.6B-CustomVoice`
+2. Reduce `--gpu-memory-utilization`
+
+### Unsupported Speaker
+
+Use `/v1/audio/voices` to list available voices for the loaded model.
+
+## Development
+
+Enable debug logging:
+
+```bash
+vllm serve Qwen/Qwen3-TTS-12Hz-1.7B-CustomVoice \
+    --stage-configs-path vllm_omni/model_executor/stage_configs/qwen3_tts.yaml \
+    --omni \
+    --port 8091 \
+    --trust-remote-code \
+    --enforce-eager \
+    --uvicorn-log-level debug
+```
diff --git a/pyproject.toml b/pyproject.toml
index 0748b7eab..6af193953 100644
--- a/pyproject.toml
+++ b/pyproject.toml
@@ -192,6 +192,6 @@ extend-ignore-identifiers-re = [
     ".*nothink.*",
     ".*NOTHINK.*",
     ".*nin.*",
-    "Ono_Anna",
+    ".*[Oo]no_[Aa]nna.*",
     ".*cann.*",
 ]
diff --git a/tests/e2e/online_serving/test_qwen3_tts.py b/tests/e2e/online_serving/test_qwen3_tts.py
new file mode 100644
index 000000000..9a1ff8b97
--- /dev/null
+++ b/tests/e2e/online_serving/test_qwen3_tts.py
@@ -0,0 +1,226 @@
+# SPDX-License-Identifier: Apache-2.0
+# SPDX-FileCopyrightText: Copyright contributors to the vLLM project
+"""
+E2E Online tests for Qwen3-TTS model with text input and audio output.
+
+These tests verify the /v1/audio/speech endpoint works correctly with
+actual model inference, not mocks.
+"""
+
+import os
+
+os.environ["VLLM_WORKER_MULTIPROC_METHOD"] = "spawn"
+os.environ["VLLM_TEST_CLEAN_GPU_MEMORY"] = "0"
+
+from pathlib import Path
+
+import httpx
+import pytest
+
+from tests.conftest import OmniServer
+from tests.utils import hardware_test
+
+MODEL = "Qwen/Qwen3-TTS-12Hz-0.6B-CustomVoice"
+
+
+def get_stage_config():
+    """Get the stage config path for Qwen3-TTS."""
+    return str(
+        Path(__file__).parent.parent.parent.parent / "vllm_omni" / "model_executor" / "stage_configs" / "qwen3_tts.yaml"
+    )
+
+
+@pytest.fixture(scope="module")
+def omni_server():
+    """Start vLLM-Omni server with CustomVoice model."""
+    stage_config_path = get_stage_config()
+
+    print(f"Starting OmniServer with model: {MODEL}")
+
+    with OmniServer(
+        MODEL,
+        [
+            "--stage-configs-path",
+            stage_config_path,
+            "--stage-init-timeout",
+            "120",
+            "--trust-remote-code",
+            "--enforce-eager",
+        ],
+    ) as server:
+        print("OmniServer started successfully")
+        yield server
+        print("OmniServer stopping...")
+
+    print("OmniServer stopped")
+
+
+def make_speech_request(
+    host: str,
+    port: int,
+    text: str,
+    voice: str = "vivian",
+    language: str = "English",
+    task_type: str | None = None,
+    instructions: str | None = None,
+    timeout: float = 120.0,
+) -> httpx.Response:
+    """Make a request to the /v1/audio/speech endpoint."""
+    url = f"http://{host}:{port}/v1/audio/speech"
+    payload = {
+        "input": text,
+        "voice": voice,
+        "language": language,
+    }
+    if task_type:
+        payload["task_type"] = task_type
+    if instructions:
+        payload["instructions"] = instructions
+
+    with httpx.Client(timeout=timeout) as client:
+        return client.post(url, json=payload)
+
+
+def verify_wav_audio(content: bytes) -> bool:
+    """Verify that content is valid WAV audio data."""
+    # WAV files start with "RIFF" header
+    if len(content) < 44:  # Minimum WAV header size
+        return False
+    return content[:4] == b"RIFF" and content[8:12] == b"WAVE"
+
+
+# Minimum expected audio size for a short sentence (~1 second of 24kHz 16-bit mono WAV)
+MIN_AUDIO_BYTES = 10000
+
+
+class TestQwen3TTSCustomVoice:
+    """E2E tests for Qwen3-TTS CustomVoice model."""
+
+    @pytest.mark.core_model
+    @pytest.mark.omni
+    @hardware_test(res={"cuda": "L4"}, num_cards=4)
+    def test_speech_english_basic(self, omni_server) -> None:
+        """Test basic English TTS generation."""
+        response = make_speech_request(
+            host=omni_server.host,
+            port=omni_server.port,
+            text="Hello, how are you?",
+            voice="vivian",
+            language="English",
+        )
+
+        assert response.status_code == 200, f"Request failed: {response.text}"
+        assert response.headers.get("content-type") == "audio/wav"
+        assert verify_wav_audio(response.content), "Response is not valid WAV audio"
+        assert len(response.content) > MIN_AUDIO_BYTES, (
+            f"Audio content too small ({len(response.content)} bytes), expected at least {MIN_AUDIO_BYTES} bytes"
+        )
+
+    @pytest.mark.core_model
+    @pytest.mark.omni
+    @hardware_test(res={"cuda": "L4"}, num_cards=4)
+    def test_speech_chinese_basic(self, omni_server) -> None:
+        """Test basic Chinese TTS generation."""
+        response = make_speech_request(
+            host=omni_server.host,
+            port=omni_server.port,
+            text="你好，我是通义千问",
+            voice="vivian",
+            language="Chinese",
+        )
+
+        assert response.status_code == 200, f"Request failed: {response.text}"
+        assert response.headers.get("content-type") == "audio/wav"
+        assert verify_wav_audio(response.content), "Response is not valid WAV audio"
+        assert len(response.content) > MIN_AUDIO_BYTES, (
+            f"Audio content too small ({len(response.content)} bytes), expected at least {MIN_AUDIO_BYTES} bytes"
+        )
+
+    @pytest.mark.core_model
+    @pytest.mark.omni
+    @hardware_test(res={"cuda": "L4"}, num_cards=4)
+    def test_speech_different_voices(self, omni_server) -> None:
+        """Test TTS with different voice options."""
+        voices = ["vivian", "ryan"]
+        for voice in voices:
+            response = make_speech_request(
+                host=omni_server.host,
+                port=omni_server.port,
+                text="Testing voice selection.",
+                voice=voice,
+                language="English",
+            )
+
+            assert response.status_code == 200, f"Request failed for voice {voice}: {response.text}"
+            assert verify_wav_audio(response.content), f"Invalid WAV for voice {voice}"
+
+    @pytest.mark.core_model
+    @pytest.mark.omni
+    @hardware_test(res={"cuda": "L4"}, num_cards=4)
+    def test_speech_binary_response_not_utf8_error(self, omni_server) -> None:
+        """
+        Regression test: Verify binary audio is returned, not UTF-8 error.
+
+        This test ensures the multimodal_output property correctly retrieves
+        audio from completion outputs, preventing the "TTS model did not
+        produce audio output" error.
+        """
+        response = make_speech_request(
+            host=omni_server.host,
+            port=omni_server.port,
+            text="This should return binary audio, not a JSON error.",
+            voice="vivian",
+            language="English",
+        )
+
+        # Should NOT be a JSON error response
+        assert response.status_code == 200, f"Request failed: {response.text}"
+
+        # Verify it's binary audio, not JSON
+        try:
+            # If this succeeds and starts with {"error", it's a bug
+            text = response.content.decode("utf-8")
+            assert not text.startswith('{"error"'), f"Got error response instead of audio: {text}"
+        except UnicodeDecodeError:
+            # This is expected - binary audio can't be decoded as UTF-8
+            pass
+
+        assert verify_wav_audio(response.content), "Response is not valid WAV audio"
+
+
+class TestQwen3TTSAPIEndpoints:
+    """Test API endpoint functionality."""
+
+    @pytest.mark.core_model
+    @pytest.mark.omni
+    @hardware_test(res={"cuda": "L4"}, num_cards=4)
+    def test_list_voices_endpoint(self, omni_server) -> None:
+        """Test the /v1/audio/voices endpoint returns available voices."""
+        url = f"http://{omni_server.host}:{omni_server.port}/v1/audio/voices"
+
+        with httpx.Client(timeout=30.0) as client:
+            response = client.get(url)
+
+        assert response.status_code == 200
+        data = response.json()
+        assert "voices" in data
+        assert isinstance(data["voices"], list)
+        assert len(data["voices"]) > 0
+        # Check some expected voices are present
+        voices_lower = [v.lower() for v in data["voices"]]
+        assert "vivian" in voices_lower or "ryan" in voices_lower
+
+    @pytest.mark.core_model
+    @pytest.mark.omni
+    @hardware_test(res={"cuda": "L4"}, num_cards=4)
+    def test_models_endpoint(self, omni_server) -> None:
+        """Test the /v1/models endpoint returns loaded model."""
+        url = f"http://{omni_server.host}:{omni_server.port}/v1/models"
+
+        with httpx.Client(timeout=30.0) as client:
+            response = client.get(url)
+
+        assert response.status_code == 200
+        data = response.json()
+        assert "data" in data
+        assert len(data["data"]) > 0
diff --git a/vllm_omni/metrics/stats.py b/vllm_omni/metrics/stats.py
index 037db2ea8..d5c45a0a9 100644
--- a/vllm_omni/metrics/stats.py
+++ b/vllm_omni/metrics/stats.py
@@ -221,24 +221,33 @@ class OrchestratorAggregator:
         stage_id: int,
         request_id: str,
     ) -> None:
-        if (
-            output_to_yield.final_output_type == "audio"
-            and finished
-            and (multimodal_output := output_to_yield.multimodal_output.get("audio")) is not None
-        ):
-            nframes = int(multimodal_output[-1].shape[0])
-            stage_events_for_req = self.stage_events.get(request_id, [])
-            if stage_events_for_req:
-                for stage_event in stage_events_for_req:
-                    if stage_event.stage_id == stage_id:
-                        stage_event.audio_generated_frames += nframes
-                        break
-            else:
-                logger.warning(
-                    "Failed to record audio generated frames for request %s at stage %s: no stage event found",
-                    request_id,
-                    stage_id,
-                )
+        try:
+            if (
+                output_to_yield.final_output_type == "audio"
+                and finished
+                and (multimodal_output := output_to_yield.multimodal_output.get("audio")) is not None
+                and len(multimodal_output) > 0
+            ):
+                last = multimodal_output[-1]
+                nframes = int(last.shape[0]) if last.ndim > 0 else 1
+                stage_events_for_req = self.stage_events.get(request_id, [])
+                if stage_events_for_req:
+                    for stage_event in stage_events_for_req:
+                        if stage_event.stage_id == stage_id:
+                            stage_event.audio_generated_frames += nframes
+                            break
+                else:
+                    logger.warning(
+                        "Failed to record audio generated frames for request %s at stage %s: no stage event found",
+                        request_id,
+                        stage_id,
+                    )
+        except Exception:
+            logger.debug(
+                "Failed to record audio frames for request %s",
+                request_id,
+                exc_info=True,
+            )
 
     def _as_stage_request_stats(
         self,