diff --git a/src/pipecat/services/rime.py b/src/pipecat/services/rime.py
index 0210f8de6..1634578cd 100644
--- a/src/pipecat/services/rime.py
+++ b/src/pipecat/services/rime.py
@@ -7,7 +7,7 @@
 import base64
 import json
 import uuid
-from typing import AsyncGenerator, Optional, Union
+from typing import AsyncGenerator, Optional
 
 import aiohttp
 from loguru import logger
@@ -28,7 +28,7 @@ from pipecat.frames.frames import (
     TTSStoppedFrame,
 )
 from pipecat.processors.frame_processor import FrameDirection
-from pipecat.services.ai_services import TTSService, WordTTSService
+from pipecat.services.ai_services import AudioContextWordTTSService, TTSService
 from pipecat.services.websocket_service import WebsocketService
 from pipecat.transcriptions.language import Language
 
@@ -58,7 +58,7 @@ def language_to_rime_language(language: Language) -> str:
     return LANGUAGE_MAP.get(language, "eng")
 
 
-class RimeTTSService(WordTTSService, WebsocketService):
+class RimeTTSService(AudioContextWordTTSService, WebsocketService):
     """Text-to-Speech service using Rime's websocket API.
 
     Uses Rime's websocket JSON API to convert text to speech with word-level timing
@@ -95,7 +95,7 @@ class RimeTTSService(WordTTSService, WebsocketService):
             params: Additional configuration parameters.
         """
         # Initialize with parent class settings for proper frame handling
-        WordTTSService.__init__(
+        AudioContextWordTTSService.__init__(
             self,
             aggregate_sentences=True,
             push_text_frames=False,
@@ -249,12 +249,18 @@ class RimeTTSService(WordTTSService, WebsocketService):
 
         return word_pairs
 
+    async def flush_audio(self):
+        if not self._context_id or not self._websocket:
+            return
+        logger.trace(f"{self}: flushing audio")
+        self._context_id = None
+
     async def _receive_messages(self):
         """Process incoming websocket messages."""
         async for message in self._get_websocket():
             msg = json.loads(message)
 
-            if not msg or msg["contextId"] != self._context_id:
+            if not msg or not self.audio_context_available(msg["contextId"]):
                 continue
 
             if msg["type"] == "chunk":
@@ -266,7 +272,7 @@ class RimeTTSService(WordTTSService, WebsocketService):
                     sample_rate=self.sample_rate,
                     num_channels=1,
                 )
-                await self.push_frame(frame)
+                await self.append_to_audio_context(msg["contextId"], frame)
 
             elif msg["type"] == "timestamps":
                 # Process word timing information
@@ -288,6 +294,7 @@ class RimeTTSService(WordTTSService, WebsocketService):
                 await self.push_frame(TTSStoppedFrame())
                 await self.stop_all_metrics()
                 await self.push_error(ErrorFrame(f"{self} error: {msg['message']}"))
+                self._context_id = None
 
     async def push_frame(self, frame: Frame, direction: FrameDirection = FrameDirection.DOWNSTREAM):
         """Push frame and handle end-of-turn conditions."""
@@ -329,6 +336,7 @@ class RimeTTSService(WordTTSService, WebsocketService):
                     self._started = True
                     self._cumulative_time = 0
                     self._context_id = str(uuid.uuid4())
+                    await self.create_audio_context(self._context_id)
 
                 msg = self._build_msg(text=text)
                 await self._get_websocket().send(json.dumps(msg))