bug fix

app.py

@@ -10,7 +10,7 @@ from streamlit_webrtc import webrtc_streamer, WebRtcMode, RTCConfiguration
 import av
 import queue
 
-# Define the models (You can replace these with any other top models supporting audio input)
+# Define the models
 MODELS = {
     "Whisper (English)": "openai/whisper-small.en",
     "Whisper (Multilingual)": "openai/whisper-small",
@@ -48,11 +48,17 @@ def audio_frame_callback(frame: av.AudioFrame):
 if audio_option == 'Record Audio':
     st.write("Click the button to start/stop recording.")
 
-    # Use rtc_configuration and media_stream_constraints instead of client_settings
+    # Change STUN server to a different one to avoid potential issues
     rtc_configuration = RTCConfiguration(
-        {"iceServers": [{"urls": ["stun:stun.l.google.com:19302"]}]}
+        {
+            "iceServers": [
+                {"urls": ["stun:stun1.l.google.com:19302"]}, 
+                {"urls": ["stun:stun2.l.google.com:19302"]}
+            ]
+        }
     )
-    
+
+    # Start WebRTC recording
     webrtc_ctx = webrtc_streamer(
         key="audio-stream",
         mode=WebRtcMode.SENDONLY,
@@ -61,6 +67,7 @@ if audio_option == 'Record Audio':
         audio_frame_callback=audio_frame_callback,
     )
 
+    # Ensure we are recording
     if webrtc_ctx.state.playing:
         st.write("Recording...")