diff --git a/changelog/4151.fixed.md b/changelog/4151.fixed.md
new file mode 100644
index 000000000..0bcd55194
--- /dev/null
+++ b/changelog/4151.fixed.md
@@ -0,0 +1 @@
+- Fixed `LiveKitOutputTransport` not clearing the `rtc.AudioSource` internal buffer on interruption, causing the bot to continue speaking for several seconds after being interrupted.
diff --git a/src/pipecat/transports/livekit/transport.py b/src/pipecat/transports/livekit/transport.py
index 18163cf8e..07a7e3f39 100644
--- a/src/pipecat/transports/livekit/transport.py
+++ b/src/pipecat/transports/livekit/transport.py
@@ -27,7 +27,9 @@ from pipecat.frames.frames import (
     CancelFrame,
     ClientConnectedFrame,
     EndFrame,
+    Frame,
     ImageRawFrame,
+    InterruptionFrame,
     OutputAudioRawFrame,
     OutputDTMFFrame,
     OutputDTMFUrgentFrame,
@@ -880,6 +882,21 @@ class LiveKitOutputTransport(BaseOutputTransport):
         await super().cancel(frame)
         await self._client.disconnect()
 
+    async def process_frame(self, frame: Frame, direction: FrameDirection):
+        """Process frames, clearing the LiveKit AudioSource buffer on interruption.
+
+        When an InterruptionFrame arrives, any audio already submitted to the
+        LiveKit AudioSource (but not yet played out) is cleared immediately so
+        the bot stops speaking without delay.
+
+        Args:
+            frame: The frame to process.
+            direction: The direction of frame flow in the pipeline.
+        """
+        await super().process_frame(frame, direction)
+        if isinstance(frame, InterruptionFrame) and self._client._audio_source is not None:
+            self._client._audio_source.clear_queue()
+
     async def setup(self, setup: FrameProcessorSetup):
         """Setup the output transport with shared client setup.