diff --git a/pyproject.toml b/pyproject.toml
index 33534e08a..78087b636 100644
--- a/pyproject.toml
+++ b/pyproject.toml
@@ -52,6 +52,7 @@ playht = [ "pyht~=0.0.28" ]
 silero = [ "torch~=2.3.0", "torchaudio~=2.3.0" ]
 websocket = [ "websockets~=12.0", "fastapi~=0.111.0" ]
 whisper = [ "faster-whisper~=1.0.2" ]
+xtts = [ "resampy~=0.4.3" ]
 
 [tool.setuptools.packages.find]
 # All the following settings are optional:
diff --git a/src/pipecat/services/xtts.py b/src/pipecat/services/xtts.py
new file mode 100644
index 000000000..faf93f7cb
--- /dev/null
+++ b/src/pipecat/services/xtts.py
@@ -0,0 +1,111 @@
+#
+# Copyright (c) 2024, Daily
+#
+# SPDX-License-Identifier: BSD 2-Clause License
+#
+
+import aiohttp
+
+from typing import AsyncGenerator
+
+from pipecat.frames.frames import AudioRawFrame, ErrorFrame, Frame
+from pipecat.services.ai_services import TTSService
+
+from loguru import logger
+
+import requests
+
+import numpy as np
+
+try:
+    import resampy
+except ModuleNotFoundError as e:
+    logger.error(f"Exception: {e}")
+    logger.error("In order to use XTTS, you need to `pip install pipecat-ai[xtts]`.")
+    raise Exception(f"Missing module: {e}")
+
+#####
+## The server below can connect to XTTS through a local running docker
+##
+## Docker command: $ docker run --gpus=all -e COQUI_TOS_AGREED=1 --rm -p 8000:80 ghcr.io/coqui-ai/xtts-streaming-server:latest-cuda121
+## 
+## You can find more information on the official repo: https://github.com/coqui-ai/xtts-streaming-server
+####
+
+class XTTSService(TTSService):
+
+    def __init__(
+            self,
+            *,
+            aiohttp_session: aiohttp.ClientSession,
+            voice_id: str,
+            language: str,
+            base_url:str,
+            **kwargs):
+        super().__init__(**kwargs)
+
+        self._voice_id = voice_id
+        self._language = language
+        self._base_url = base_url
+        self._aiohttp_session = aiohttp_session
+        self._studio_speakers = requests.get(self._base_url + "/studio_speakers").json()
+
+    def can_generate_metrics(self) -> bool:
+        return True
+
+    async def run_tts(self, text: str) -> AsyncGenerator[Frame, None]:
+        logger.debug(f"Generating TTS: [{text}]")
+        embeddings = self._studio_speakers[self._voice_id]
+
+        url = self._base_url + "/tts_stream"
+        
+        payload={
+            "text": text.replace('.','').replace('*',''),
+            "language": self._language,
+            "speaker_embedding": embeddings["speaker_embedding"],
+            "gpt_cond_latent": embeddings["gpt_cond_latent"],
+            "add_wav_header": True,
+            "stream_chunk_size": 20,
+        }
+
+        await self.start_ttfb_metrics()
+
+        async with self._aiohttp_session.post(url, json=payload) as r:
+            if r.status != 200:
+                text = await r.text()
+                logger.error(f"{self} error getting audio (status: {r.status}, error: {text})")
+                yield ErrorFrame(f"Error getting audio (status: {r.status}, error: {text})")
+                return
+            
+            buffer = bytearray()
+
+            async for chunk in r.content.iter_chunked(1024):
+                if len(chunk) > 0:
+                    await self.stop_ttfb_metrics()
+                    # Append new chunk to the buffer
+                    buffer.extend(chunk)
+                    
+                    # Check if buffer has enough data for processing
+                    while len(buffer) >= 48000:  # Assuming at least 0.5 seconds of audio data at 24000 Hz
+                        # Process the buffer up to a safe size for resampling
+                        process_data = buffer[:48000]
+                        # Remove processed data from buffer
+                        buffer = buffer[48000:]
+                        
+                        # Convert the byte data to numpy array for resampling
+                        audio_np = np.frombuffer(process_data, dtype=np.int16)
+                        # Resample the audio from 24000 Hz to 16000 Hz
+                        resampled_audio = resampy.resample(audio_np, 24000, 16000)
+                        # Convert the numpy array back to bytes
+                        resampled_audio_bytes = resampled_audio.astype(np.int16).tobytes()
+                        # Create the frame with the resampled audio
+                        frame = AudioRawFrame(resampled_audio_bytes, 16000, 1)
+                        yield frame
+
+            # Process any remaining data in the buffer
+            if len(buffer) > 0:
+                audio_np = np.frombuffer(buffer, dtype=np.int16)
+                resampled_audio = resampy.resample(audio_np, 24000, 16000)
+                resampled_audio_bytes = resampled_audio.astype(np.int16).tobytes()
+                frame = AudioRawFrame(resampled_audio_bytes, 16000, 1)
+                yield frame
\ No newline at end of file