Update app.py

app.py

@@ -151,56 +151,66 @@ def use_gtts_for_text(text):
         print(f"gTTS error: {e}")
         yield None
 
-# Custom WebRTC configuration that doesn't require Twilio
+# Enhanced WebRTC configuration with more STUN/TURN servers
 rtc_configuration = {
     "iceServers": [
-        {"urls": ["stun:stun.l.google.com:19302"]},
+        # Google's public STUN servers
+        {"urls": ["stun:stun.l.google.com:19302", "stun:stun1.l.google.com:19302", "stun:stun2.l.google.com:19302", "stun:stun3.l.google.com:19302", "stun:stun4.l.google.com:19302"]},
+        
+        # OpenRelay TURN servers (HTTP)
         {
             "urls": ["turn:openrelay.metered.ca:80"],
             "username": "openrelayproject",
             "credential": "openrelayproject"
         },
+        
+        # OpenRelay TURN servers (HTTPS)
         {
             "urls": ["turn:openrelay.metered.ca:443"],
             "username": "openrelayproject", 
             "credential": "openrelayproject"
         },
+        
+        # OpenRelay TURN servers (TCP)
         {
             "urls": ["turn:openrelay.metered.ca:443?transport=tcp"],
             "username": "openrelayproject",
             "credential": "openrelayproject"
-        }
-    ]
+        },
+        
+        # Additional public STUN servers
+        {"urls": ["stun:stun.stunprotocol.org:3478"]}
+    ],
+    "iceCandidatePoolSize": 10  # Increase the pool size
 }
 
-# Create Gradio chatbot and stream
-chatbot = gr.Chatbot(type="messages")
-stream = Stream(
-    modality="audio",
-    mode="send-receive",
-    handler=ReplyOnPause(response, input_sample_rate=16000),
-    additional_outputs_handler=lambda a, b: b,
-    additional_inputs=[chatbot],
-    additional_outputs=[chatbot],
-    rtc_configuration=rtc_configuration if get_space() else None,
-    concurrency_limit=5 if get_space() else None,
-    time_limit=90 if get_space() else None,
-    ui_args={"title": "LLM Voice Chat (Powered by DeepSeek & ElevenLabs)"}
-)
-
-# Mount the Stream UI to the FastAPI app
-app = FastAPI()
-app = gr.mount_gradio_app(app, stream.ui, path="/")
+# Set WebRTC logging level to "debug" for more information
+os.environ["WEBRTC_TRACE"] = "WEBRTC_TRACE_ALL"
 
-# Only for local development
-if __name__ == "__main__":
-    os.environ["GRADIO_SSR_MODE"] = "false"
+# Create Gradio interface with simple structure
+with gr.Blocks(title="LLM Voice Chat") as demo:
+    gr.Markdown("# LLM Voice Chat (Powered by DeepSeek & ElevenLabs)")
     
-    # Different launch modes based on environment
-    if (mode := os.getenv("MODE")) == "UI":
-        stream.ui.launch(server_port=7860)
-    elif mode == "PHONE":
-        stream.fastphone(host="0.0.0.0", port=7860)
-    else:
-        stream.ui.launch(server_port=7860)
+    with gr.Row():
+        chatbot = gr.Chatbot(type="messages", label="Chat History")
+    
+    with gr.Row():
+        # Create the Stream component for handling audio
+        stream_comp = Stream(
+            modality="audio",
+            mode="send-receive",
+            handler=ReplyOnPause(response, input_sample_rate=16000),
+            additional_outputs_handler=lambda a, b: b,
+            additional_inputs=[chatbot],
+            additional_outputs=[chatbot],
+            rtc_configuration=rtc_configuration,
+            concurrency_limit=5 if get_space() else None,
+            time_limit=90 if get_space() else None,
+        )
+        
+        # Make sure to render the Stream component
+        stream_comp.render()
 
+# For local development only
+if __name__ == "__main__" and not get_space():
+    demo.launch(share=True)