fix: buffer ICE candidates and discard stale signaling events

Two fixes for call connectivity:

1. ICE candidate buffering: Candidates arriving before the WebRTC
   session exists (callee ringing) or before remote description is
   set (caller waiting for answer) are now queued in
   pendingIceCandidates and flushed once setRemoteDescription is
   called. This was the root cause of calls getting stuck at
   "Connecting" — ICE candidates were silently dropped.

2. Stale event filter: All signaling events older than 30 seconds
   are discarded in CallManager.onSignalingEvent() to prevent
   old cached events from triggering phantom calls.

Also: removed cleanup() from WebRTC onDisconnected callback to
avoid double-cleanup race with the CallManager state observer.

https://claude.ai/code/session_017hZm7yu7CzmcQgZGSaqSXS

amethyst/src/main/java/com/vitorpamplona/amethyst/service/call/CallController.kt

@@ -35,6 +35,7 @@ import org.webrtc.IceCandidate
 import org.webrtc.MediaStream
 import org.webrtc.SessionDescription
 import java.util.UUID
+import java.util.concurrent.CopyOnWriteArrayList
 
 class CallController(
     private val context: Context,
@@ -47,6 +48,8 @@ class CallController(
     private val callFactory = WebRtcCallFactory()
     private var currentCallId: String? = null
     private var currentPeerPubKey: HexKey? = null
+    private var remoteDescriptionSet = false
+    private val pendingIceCandidates = CopyOnWriteArrayList<IceCandidate>()
 
     init {
         scope.launch {
@@ -65,6 +68,8 @@ class CallController(
         val callId = UUID.randomUUID().toString()
         currentCallId = callId
         currentPeerPubKey = peerPubKey
+        remoteDescriptionSet = false
+        pendingIceCandidates.clear()
 
         createWebRtcSession()
         webRtcSession?.addAudioTrack()
@@ -85,6 +90,8 @@ class CallController(
 
         currentCallId = state.callId
         currentPeerPubKey = state.callerPubKey
+        remoteDescriptionSet = false
+        pendingIceCandidates.clear()
 
         createWebRtcSession()
         webRtcSession?.addAudioTrack()
@@ -95,6 +102,7 @@ class CallController(
         webRtcSession?.setRemoteDescription(
             SessionDescription(SessionDescription.Type.OFFER, sdpOffer),
         )
+        flushPendingIceCandidates()
 
         webRtcSession?.createAnswer { sdp ->
             scope.launch {
@@ -107,18 +115,31 @@ class CallController(
         webRtcSession?.setRemoteDescription(
             SessionDescription(SessionDescription.Type.ANSWER, sdpAnswer),
         )
+        flushPendingIceCandidates()
     }
 
     fun onIceCandidateReceived(event: CallIceCandidateEvent) {
         val json = event.candidateJson()
         try {
             val candidate = parseIceCandidate(json)
-            webRtcSession?.addIceCandidate(candidate)
+            if (webRtcSession != null && remoteDescriptionSet) {
+                webRtcSession?.addIceCandidate(candidate)
+            } else {
+                pendingIceCandidates.add(candidate)
+            }
         } catch (_: Exception) {
             // Ignore malformed ICE candidates
         }
     }
 
+    private fun flushPendingIceCandidates() {
+        remoteDescriptionSet = true
+        val session = webRtcSession ?: return
+        val candidates = pendingIceCandidates.toList()
+        pendingIceCandidates.clear()
+        candidates.forEach { session.addIceCandidate(it) }
+    }
+
     fun hangup() {
         scope.launch { callManager.hangup() }
         cleanup()
@@ -130,6 +151,8 @@ class CallController(
         webRtcSession = null
         currentCallId = null
         currentPeerPubKey = null
+        remoteDescriptionSet = false
+        pendingIceCandidates.clear()
     }
 
     private fun createWebRtcSession() {
@@ -147,7 +170,6 @@ class CallController(
                 onRemoteStream = { _: MediaStream -> },
                 onDisconnected = {
                     scope.launch { callManager.hangup() }
-                    cleanup()
                 },
             )
         webRtcSession?.initialize()

commons/src/commonMain/kotlin/com/vitorpamplona/amethyst/commons/call/CallManager.kt

@@ -60,8 +60,11 @@ class CallManager(
     companion object {
         const val CALL_TIMEOUT_MS = 60_000L // 60 seconds ringing timeout
         const val ENDED_DISPLAY_MS = 2_000L // show "call ended" briefly before resetting
+        const val MAX_EVENT_AGE_SECONDS = 30L // discard signaling events older than this
     }
 
+    private fun isEventTooOld(event: Event): Boolean = TimeUtils.now() - event.createdAt > MAX_EVENT_AGE_SECONDS
+
     suspend fun initiateCall(
         calleePubKey: HexKey,
         callType: CallType,
@@ -194,6 +197,8 @@ class CallManager(
     }
 
     fun onSignalingEvent(event: Event) {
+        if (isEventTooOld(event)) return
+
         when (event) {
             is CallOfferEvent -> onIncomingCallEvent(event)
             is CallAnswerEvent -> onCallAnswered(event)