From b33981f7fa3ea8af43d259e344f3ff497edcf6d8 Mon Sep 17 00:00:00 2001
From: pstruebi <struebin.patrick@gmail.com>
Date: Thu, 19 Jun 2025 13:46:40 +0200
Subject: [PATCH] Fix webrtc restart issue

---
 src/auracast/server/multicast_frontend.py | 23 +++++++++++++++++++++--
 1 file changed, 21 insertions(+), 2 deletions(-)

diff --git a/src/auracast/server/multicast_frontend.py b/src/auracast/server/multicast_frontend.py
index 7a09ac9..a4b44ec 100644
--- a/src/auracast/server/multicast_frontend.py
+++ b/src/auracast/server/multicast_frontend.py
@@ -111,6 +111,20 @@ if audio_mode in ["Webapp", "USB"]:
                 st.success("Stream was not running.")
         except Exception as e:
             st.error(f"Error: {e}")
+        # Ensure existing WebRTC connection is fully closed so that a fresh
+        # connection is created the next time we start the stream.
+        if audio_mode == "Webapp":
+            cleanup_js = """
+            <script>
+                if (window.webrtc_pc) {
+                    window.webrtc_pc.getSenders().forEach(s => s.track.stop());
+                    window.webrtc_pc.close();
+                    window.webrtc_pc = null;
+                }
+                window.webrtc_started = false;
+            </script>
+            """
+            st.components.v1.html(cleanup_js, height=0)
 
     if start_stream:
         # Always send stop to ensure backend is in a clean state, regardless of current status
@@ -158,10 +172,15 @@ if audio_mode in ["Webapp", "USB"]:
         component = f"""
         <script>
             (async () => {{
-                if (window.webrtc_started) return; // Prevent re-init on rerun
-                window.webrtc_started = true;
+                 // Clean up any previous WebRTC connection before starting a new one
+                 if (window.webrtc_pc) {{
+                     window.webrtc_pc.getSenders().forEach(s => s.track.stop());
+                     window.webrtc_pc.close();
+                 }}
                 const GAIN_VALUE = {mic_gain};
                 const pc = new RTCPeerConnection(); // No STUN needed for localhost
+                window.webrtc_pc = pc;
+                window.webrtc_started = true;
                 const micStream = await navigator.mediaDevices.getUserMedia({{audio:true}});
                 // Create Web Audio gain processing
                 const audioCtx = new (window.AudioContext || window.webkitAudioContext)();