code

frontend/shared/InteractiveAudio.svelte

@@ -36,7 +36,7 @@
     let stream_state: "open" | "closed" | "waiting" = "closed";
     let audio_player: HTMLAudioElement;
     let pc: RTCPeerConnection;
-    let _webrtc_id = Math.random().toString(36).substring(2);
+    let _webrtc_id = null;
 
 
     const dispatch = createEventDispatcher<{
@@ -63,6 +63,7 @@
             _time_limit = null;
             return;
         }
+            _webrtc_id = Math.random().toString(36).substring(2);
             value = _webrtc_id;
             pc = new RTCPeerConnection(rtc_configuration);
             pc.addEventListener("connectionstatechange",

frontend/shared/StaticAudio.svelte

@@ -47,6 +47,7 @@
     async function start_stream(value: string): Promise<string> {
         if( value === "start_webrtc_stream") {
             stream_state = "waiting";
+            _webrtc_id = Math.random().toString(36).substring(2)
             value = _webrtc_id;
             console.log("set value to ", value);
             pc = new RTCPeerConnection(rtc_configuration);

frontend/shared/StaticVideo.svelte

@@ -40,6 +40,7 @@
 	)
 
 	$: if( value === "start_webrtc_stream") {
+		_webrtc_id = Math.random().toString(36).substring(2);
 		value = _webrtc_id;
 		pc = new RTCPeerConnection(rtc_configuration);
 		pc.addEventListener("connectionstatechange",

frontend/shared/Webcam.svelte

@@ -138,7 +138,7 @@
                 }
             )
             stream_state = "waiting"
-			webrtc_id = _webrtc_id;
+			webrtc_id = Math.random().toString(36).substring(2);
             start(stream, pc, video_source, server.offer, webrtc_id).then((connection) => {
 				pc = connection;
 			}).catch(() => {

frontend/shared/webrtc_utils.ts

@@ -134,6 +134,7 @@ export function stop(pc: RTCPeerConnection) {
 	// close local audio / video
 	if (pc.getSenders()) {
 		pc.getSenders().forEach((sender) => {
+			console.log("sender", sender);
 			if (sender.track && sender.track.stop) sender.track.stop();
 		});
 	}