diff --git a/src/pipecat/processors/audio/audio_buffer_processor.py b/src/pipecat/processors/audio/audio_buffer_processor.py
index 78561b2f9..9cfc08daa 100644
--- a/src/pipecat/processors/audio/audio_buffer_processor.py
+++ b/src/pipecat/processors/audio/audio_buffer_processor.py
@@ -14,9 +14,8 @@ configurations and event-driven processing.
 import time
 from typing import Optional
 
-from pipecat.audio.utils import create_default_resampler, interleave_stereo_audio, mix_audio
+from pipecat.audio.utils import create_stream_resampler, interleave_stereo_audio, mix_audio
 from pipecat.frames.frames import (
-    AudioRawFrame,
     BotStartedSpeakingFrame,
     BotStoppedSpeakingFrame,
     CancelFrame,
@@ -106,7 +105,8 @@ class AudioBufferProcessor(FrameProcessor):
 
         self._recording = False
 
-        self._resampler = create_default_resampler()
+        self._input_resampler = create_stream_resampler()
+        self._output_resampler = create_stream_resampler()
 
         self._register_event_handler("on_audio_data")
         self._register_event_handler("on_track_audio_data")
@@ -210,7 +210,7 @@ class AudioBufferProcessor(FrameProcessor):
             silence = self._compute_silence(self._last_user_frame_at)
             self._user_audio_buffer.extend(silence)
             # Add user audio.
-            resampled = await self._resample_audio(frame)
+            resampled = await self._resample_input_audio(frame)
             self._user_audio_buffer.extend(resampled)
             # Save time of frame so we can compute silence.
             self._last_user_frame_at = time.time()
@@ -219,7 +219,7 @@ class AudioBufferProcessor(FrameProcessor):
             silence = self._compute_silence(self._last_bot_frame_at)
             self._bot_audio_buffer.extend(silence)
             # Add bot audio.
-            resampled = await self._resample_audio(frame)
+            resampled = await self._resample_output_audio(frame)
             self._bot_audio_buffer.extend(resampled)
             # Save time of frame so we can compute silence.
             self._last_bot_frame_at = time.time()
@@ -247,7 +247,7 @@ class AudioBufferProcessor(FrameProcessor):
             self._bot_turn_audio_buffer = bytearray()
 
         if isinstance(frame, InputAudioRawFrame):
-            resampled = await self._resample_audio(frame)
+            resampled = await self._resample_input_audio(frame)
             self._user_turn_audio_buffer += resampled
             # In the case of the user, we need to keep a short buffer of audio
             # since VAD notification of when the user starts speaking comes
@@ -259,7 +259,7 @@ class AudioBufferProcessor(FrameProcessor):
                 discarded = len(self._user_turn_audio_buffer) - self._audio_buffer_size_1s
                 self._user_turn_audio_buffer = self._user_turn_audio_buffer[discarded:]
         elif self._bot_speaking and isinstance(frame, OutputAudioRawFrame):
-            resampled = await self._resample_audio(frame)
+            resampled = await self._resample_output_audio(frame)
             self._bot_turn_audio_buffer += resampled
 
     async def _call_on_audio_data_handler(self):
@@ -301,9 +301,17 @@ class AudioBufferProcessor(FrameProcessor):
         self._user_turn_audio_buffer = bytearray()
         self._bot_turn_audio_buffer = bytearray()
 
-    async def _resample_audio(self, frame: AudioRawFrame) -> bytes:
+    async def _resample_input_audio(self, frame: InputAudioRawFrame) -> bytes:
         """Resample audio frame to the target sample rate."""
-        return await self._resampler.resample(frame.audio, frame.sample_rate, self._sample_rate)
+        return await self._input_resampler.resample(
+            frame.audio, frame.sample_rate, self._sample_rate
+        )
+
+    async def _resample_output_audio(self, frame: OutputAudioRawFrame) -> bytes:
+        """Resample audio frame to the target sample rate."""
+        return await self._output_resampler.resample(
+            frame.audio, frame.sample_rate, self._sample_rate
+        )
 
     def _compute_silence(self, from_time: float) -> bytes:
         """Compute silence to insert based on time gap."""
diff --git a/src/pipecat/serializers/exotel.py b/src/pipecat/serializers/exotel.py
index 960cbe74a..81f0daf39 100644
--- a/src/pipecat/serializers/exotel.py
+++ b/src/pipecat/serializers/exotel.py
@@ -13,7 +13,7 @@ from typing import Optional
 from loguru import logger
 from pydantic import BaseModel
 
-from pipecat.audio.utils import create_default_resampler
+from pipecat.audio.utils import create_stream_resampler
 from pipecat.frames.frames import (
     AudioRawFrame,
     Frame,
@@ -67,7 +67,8 @@ class ExotelFrameSerializer(FrameSerializer):
         self._exotel_sample_rate = self._params.exotel_sample_rate
         self._sample_rate = 0  # Pipeline input rate
 
-        self._resampler = create_default_resampler()
+        self._input_resampler = create_stream_resampler()
+        self._output_resampler = create_stream_resampler()
 
     @property
     def type(self) -> FrameSerializerType:
@@ -104,7 +105,7 @@ class ExotelFrameSerializer(FrameSerializer):
             data = frame.audio
 
             # Output: Exotel outputs PCM audio, but we need to resample to match requested sample_rate
-            serialized_data = await self._resampler.resample(
+            serialized_data = await self._output_resampler.resample(
                 data, frame.sample_rate, self._exotel_sample_rate
             )
             payload = base64.b64encode(serialized_data).decode("ascii")
@@ -138,7 +139,7 @@ class ExotelFrameSerializer(FrameSerializer):
             payload_base64 = message["media"]["payload"]
             payload = base64.b64decode(payload_base64)
 
-            deserialized_data = await self._resampler.resample(
+            deserialized_data = await self._input_resampler.resample(
                 payload,
                 self._exotel_sample_rate,
                 self._sample_rate,
diff --git a/src/pipecat/serializers/plivo.py b/src/pipecat/serializers/plivo.py
index d0c19be0a..1b2ebb57f 100644
--- a/src/pipecat/serializers/plivo.py
+++ b/src/pipecat/serializers/plivo.py
@@ -13,7 +13,7 @@ from typing import Optional
 from loguru import logger
 from pydantic import BaseModel
 
-from pipecat.audio.utils import create_default_resampler, pcm_to_ulaw, ulaw_to_pcm
+from pipecat.audio.utils import create_stream_resampler, pcm_to_ulaw, ulaw_to_pcm
 from pipecat.frames.frames import (
     AudioRawFrame,
     CancelFrame,
@@ -81,7 +81,8 @@ class PlivoFrameSerializer(FrameSerializer):
         self._plivo_sample_rate = self._params.plivo_sample_rate
         self._sample_rate = 0  # Pipeline input rate
 
-        self._resampler = create_default_resampler()
+        self._input_resampler = create_stream_resampler()
+        self._output_resampler = create_stream_resampler()
         self._hangup_attempted = False
 
     @property
@@ -129,7 +130,7 @@ class PlivoFrameSerializer(FrameSerializer):
 
             # Output: Convert PCM at frame's rate to 8kHz μ-law for Plivo
             serialized_data = await pcm_to_ulaw(
-                data, frame.sample_rate, self._plivo_sample_rate, self._resampler
+                data, frame.sample_rate, self._plivo_sample_rate, self._output_resampler
             )
             payload = base64.b64encode(serialized_data).decode("utf-8")
             answer = {
@@ -224,7 +225,7 @@ class PlivoFrameSerializer(FrameSerializer):
 
             # Input: Convert Plivo's 8kHz μ-law to PCM at pipeline input rate
             deserialized_data = await ulaw_to_pcm(
-                payload, self._plivo_sample_rate, self._sample_rate, self._resampler
+                payload, self._plivo_sample_rate, self._sample_rate, self._input_resampler
             )
             audio_frame = InputAudioRawFrame(
                 audio=deserialized_data, num_channels=1, sample_rate=self._sample_rate
diff --git a/src/pipecat/serializers/telnyx.py b/src/pipecat/serializers/telnyx.py
index adc235555..133c50dc9 100644
--- a/src/pipecat/serializers/telnyx.py
+++ b/src/pipecat/serializers/telnyx.py
@@ -16,7 +16,7 @@ from pydantic import BaseModel
 
 from pipecat.audio.utils import (
     alaw_to_pcm,
-    create_default_resampler,
+    create_stream_resampler,
     pcm_to_alaw,
     pcm_to_ulaw,
     ulaw_to_pcm,
@@ -93,7 +93,8 @@ class TelnyxFrameSerializer(FrameSerializer):
         self._telnyx_sample_rate = self._params.telnyx_sample_rate
         self._sample_rate = 0  # Pipeline input rate
 
-        self._resampler = create_default_resampler()
+        self._input_resampler = create_stream_resampler()
+        self._output_resampler = create_stream_resampler()
         self._hangup_attempted = False
 
     @property
@@ -145,11 +146,11 @@ class TelnyxFrameSerializer(FrameSerializer):
             # Output: Convert PCM at frame's rate to 8kHz encoded for Telnyx
             if self._params.inbound_encoding == "PCMU":
                 serialized_data = await pcm_to_ulaw(
-                    data, frame.sample_rate, self._telnyx_sample_rate, self._resampler
+                    data, frame.sample_rate, self._telnyx_sample_rate, self._output_resampler
                 )
             elif self._params.inbound_encoding == "PCMA":
                 serialized_data = await pcm_to_alaw(
-                    data, frame.sample_rate, self._telnyx_sample_rate, self._resampler
+                    data, frame.sample_rate, self._telnyx_sample_rate, self._output_resampler
                 )
             else:
                 raise ValueError(f"Unsupported encoding: {self._params.inbound_encoding}")
@@ -249,14 +250,14 @@ class TelnyxFrameSerializer(FrameSerializer):
                     payload,
                     self._telnyx_sample_rate,
                     self._sample_rate,
-                    self._resampler,
+                    self._input_resampler,
                 )
             elif self._params.outbound_encoding == "PCMA":
                 deserialized_data = await alaw_to_pcm(
                     payload,
                     self._telnyx_sample_rate,
                     self._sample_rate,
-                    self._resampler,
+                    self._input_resampler,
                 )
             else:
                 raise ValueError(f"Unsupported encoding: {self._params.outbound_encoding}")
diff --git a/src/pipecat/serializers/twilio.py b/src/pipecat/serializers/twilio.py
index ae4d54e4d..50ca420fa 100644
--- a/src/pipecat/serializers/twilio.py
+++ b/src/pipecat/serializers/twilio.py
@@ -13,7 +13,7 @@ from typing import Optional
 from loguru import logger
 from pydantic import BaseModel
 
-from pipecat.audio.utils import create_default_resampler, pcm_to_ulaw, ulaw_to_pcm
+from pipecat.audio.utils import create_stream_resampler, pcm_to_ulaw, ulaw_to_pcm
 from pipecat.frames.frames import (
     AudioRawFrame,
     CancelFrame,
@@ -81,7 +81,8 @@ class TwilioFrameSerializer(FrameSerializer):
         self._twilio_sample_rate = self._params.twilio_sample_rate
         self._sample_rate = 0  # Pipeline input rate
 
-        self._resampler = create_default_resampler()
+        self._input_resampler = create_stream_resampler()
+        self._output_resampler = create_stream_resampler()
         self._hangup_attempted = False
 
     @property
@@ -129,7 +130,7 @@ class TwilioFrameSerializer(FrameSerializer):
 
             # Output: Convert PCM at frame's rate to 8kHz μ-law for Twilio
             serialized_data = await pcm_to_ulaw(
-                data, frame.sample_rate, self._twilio_sample_rate, self._resampler
+                data, frame.sample_rate, self._twilio_sample_rate, self._output_resampler
             )
             payload = base64.b64encode(serialized_data).decode("utf-8")
             answer = {
@@ -214,7 +215,7 @@ class TwilioFrameSerializer(FrameSerializer):
 
             # Input: Convert Twilio's 8kHz μ-law to PCM at pipeline input rate
             deserialized_data = await ulaw_to_pcm(
-                payload, self._twilio_sample_rate, self._sample_rate, self._resampler
+                payload, self._twilio_sample_rate, self._sample_rate, self._input_resampler
             )
             audio_frame = InputAudioRawFrame(
                 audio=deserialized_data, num_channels=1, sample_rate=self._sample_rate
diff --git a/src/pipecat/services/aws/tts.py b/src/pipecat/services/aws/tts.py
index ce89dea9e..7f4ccd079 100644
--- a/src/pipecat/services/aws/tts.py
+++ b/src/pipecat/services/aws/tts.py
@@ -17,7 +17,7 @@ from typing import AsyncGenerator, List, Optional
 from loguru import logger
 from pydantic import BaseModel
 
-from pipecat.audio.utils import create_default_resampler
+from pipecat.audio.utils import create_stream_resampler
 from pipecat.frames.frames import (
     ErrorFrame,
     Frame,
@@ -195,7 +195,7 @@ class AWSPollyTTSService(TTSService):
             "lexicon_names": params.lexicon_names,
         }
 
-        self._resampler = create_default_resampler()
+        self._resampler = create_stream_resampler()
 
         self.set_voice(voice_id)
 
diff --git a/src/pipecat/services/tavus/video.py b/src/pipecat/services/tavus/video.py
index d633329bb..f9ba2833b 100644
--- a/src/pipecat/services/tavus/video.py
+++ b/src/pipecat/services/tavus/video.py
@@ -17,7 +17,7 @@ import aiohttp
 from daily.daily import AudioData, VideoFrame
 from loguru import logger
 
-from pipecat.audio.utils import create_default_resampler
+from pipecat.audio.utils import create_stream_resampler
 from pipecat.frames.frames import (
     CancelFrame,
     EndFrame,
@@ -75,7 +75,7 @@ class TavusVideoService(AIService):
         self._client: Optional[TavusTransportClient] = None
 
         self._conversation_id: str
-        self._resampler = create_default_resampler()
+        self._resampler = create_stream_resampler()
 
         self._audio_buffer = bytearray()
         self._send_task: Optional[asyncio.Task] = None
diff --git a/src/pipecat/services/xtts/tts.py b/src/pipecat/services/xtts/tts.py
index 6332e26ef..844e0fbaf 100644
--- a/src/pipecat/services/xtts/tts.py
+++ b/src/pipecat/services/xtts/tts.py
@@ -15,7 +15,7 @@ from typing import Any, AsyncGenerator, Dict, Optional
 import aiohttp
 from loguru import logger
 
-from pipecat.audio.utils import create_default_resampler
+from pipecat.audio.utils import create_stream_resampler
 from pipecat.frames.frames import (
     ErrorFrame,
     Frame,
@@ -121,7 +121,7 @@ class XTTSService(TTSService):
         self._studio_speakers: Optional[Dict[str, Any]] = None
         self._aiohttp_session = aiohttp_session
 
-        self._resampler = create_default_resampler()
+        self._resampler = create_stream_resampler()
 
     def can_generate_metrics(self) -> bool:
         """Check if this service can generate processing metrics.
diff --git a/src/pipecat/transports/base_output.py b/src/pipecat/transports/base_output.py
index 93a11f5a3..4d1120cc7 100644
--- a/src/pipecat/transports/base_output.py
+++ b/src/pipecat/transports/base_output.py
@@ -21,7 +21,7 @@ from loguru import logger
 from PIL import Image
 
 from pipecat.audio.mixers.base_audio_mixer import BaseAudioMixer
-from pipecat.audio.utils import create_default_resampler
+from pipecat.audio.utils import create_stream_resampler
 from pipecat.frames.frames import (
     BotSpeakingFrame,
     BotStartedSpeakingFrame,
@@ -358,7 +358,7 @@ class BaseOutputTransport(FrameProcessor):
             self._audio_buffer = bytearray()
 
             # This will be used to resample incoming audio to the output sample rate.
-            self._resampler = create_default_resampler()
+            self._resampler = create_stream_resampler()
 
             # The user can provide a single mixer, to be used by the default
             # destination, or a destination/mixer mapping.
diff --git a/src/pipecat/transports/services/livekit.py b/src/pipecat/transports/services/livekit.py
index d7363b9e7..68fc6a8a8 100644
--- a/src/pipecat/transports/services/livekit.py
+++ b/src/pipecat/transports/services/livekit.py
@@ -18,7 +18,7 @@ from typing import Any, Awaitable, Callable, List, Optional
 from loguru import logger
 from pydantic import BaseModel
 
-from pipecat.audio.utils import create_default_resampler
+from pipecat.audio.utils import create_stream_resampler
 from pipecat.audio.vad.vad_analyzer import VADAnalyzer
 from pipecat.frames.frames import (
     AudioRawFrame,
@@ -513,7 +513,7 @@ class LiveKitInputTransport(BaseInputTransport):
 
         self._audio_in_task = None
         self._vad_analyzer: Optional[VADAnalyzer] = params.vad_analyzer
-        self._resampler = create_default_resampler()
+        self._resampler = create_stream_resampler()
 
         # Whether we have seen a StartFrame already.
         self._initialized = False
diff --git a/src/pipecat/transports/services/tavus.py b/src/pipecat/transports/services/tavus.py
index e83dc6a20..10ae001e0 100644
--- a/src/pipecat/transports/services/tavus.py
+++ b/src/pipecat/transports/services/tavus.py
@@ -20,7 +20,6 @@ from daily.daily import AudioData
 from loguru import logger
 from pydantic import BaseModel
 
-from pipecat.audio.utils import create_default_resampler
 from pipecat.frames.frames import (
     CancelFrame,
     EndFrame,
@@ -437,8 +436,6 @@ class TavusInputTransport(BaseInputTransport):
         super().__init__(params, **kwargs)
         self._client = client
         self._params = params
-        self._resampler = create_default_resampler()
-
         # Whether we have seen a StartFrame already.
         self._initialized = False