diff --git a/app/providers/CallProvider/CallProvider.tsx b/app/providers/CallProvider/CallProvider.tsx
index 8c3f7a5..598a41d 100644
--- a/app/providers/CallProvider/CallProvider.tsx
+++ b/app/providers/CallProvider/CallProvider.tsx
@@ -74,8 +74,7 @@ export function CallProvider(props : CallProviderProps) {
     const roomIdRef = useRef<string>("");
 
     const roleRef = useRef<CallRole | null>(null);
-    //const [sharedSecret, setSharedSecret] = useState<string>("");
-    const sharedSecretRef = useRef<string>("");
+    const sharedSecretRef = useRef<string>(""); 
     const iceServersRef = useRef<RTCIceServer[]>([]);
     const remoteAudioRef = useRef<HTMLAudioElement | null>(null);
     const iceCandidatesBufferRef = useRef<RTCIceCandidate[]>([]);
@@ -98,67 +97,6 @@ export function CallProvider(props : CallProviderProps) {
         }
     }, [callState]);
 
-    // ...existing code...
-const checkWebRTCStats = async () => {
-    if (!peerConnectionRef.current) return;
-
-    const stats = await peerConnectionRef.current.getStats();
-    
-    stats.forEach(report => {
-        // Исходящий аудио
-        if (report.type === "outbound-rtp" && report.mediaType === "audio") {
-            console.info("[WebRTC OUT]", {
-                bytesSent: report.bytesSent,
-                packetsSent: report.packetsSent,
-                timestamp: report.timestamp
-            });
-        }
-
-        // Входящий аудио
-        if (report.type === "inbound-rtp" && report.mediaType === "audio") {
-            console.info("[WebRTC IN]", {
-                bytesReceived: report.bytesReceived,
-                packetsReceived: report.packetsReceived,
-                jitter: report.jitter,
-                packetsLost: report.packetsLost,
-                timestamp: report.timestamp
-            });
-        }
-
-        // RTT и задержка
-        if (report.type === "candidate-pair" && report.state === "succeeded") {
-            console.info("[WebRTC RTT]", {
-                currentRoundTripTime: (report.currentRoundTripTime * 1000).toFixed(2) + "ms",
-                availableOutgoingBitrate: (report.availableOutgoingBitrate / 1024 / 1024).toFixed(2) + " Mbps",
-                availableIncomingBitrate: (report.availableIncomingBitrate / 1024 / 1024).toFixed(2) + " Mbps"
-            });
-        }
-
-        // Codec info
-        if (report.type === "codec") {
-            if (report.mediaType === "audio") {
-                console.info("[WebRTC Codec]", {
-                    mimeType: report.mimeType,
-                    channels: report.channels,
-                    clockRate: report.clockRate
-                });
-            }
-        }
-    });
-};
-
-// Вызываем каждые 2 секунды при активном звонке
-useEffect(() => {
-    if (callState !== CallState.ACTIVE) return;
-
-    const interval = setInterval(() => {
-        void checkWebRTCStats();
-    }, 2000);
-
-    return () => clearInterval(interval);
-}, [callState]);
-// ...existing code...
-
     useEffect(() => {
         /** 
          * Нам нужно получить ICE серверы для установки соединения из разных сетей
diff --git a/app/providers/CallProvider/audioE2EE.ts b/app/providers/CallProvider/audioE2EE.ts
index b7505d6..4a8de6c 100644
--- a/app/providers/CallProvider/audioE2EE.ts
+++ b/app/providers/CallProvider/audioE2EE.ts
@@ -22,12 +22,9 @@ function toUint8Array(input: KeyInput): Uint8Array {
 
 function fillNonceFromTimestamp(nonce: Uint8Array, tsRaw: unknown): void {
     nonce.fill(0);
-
     let ts = 0n;
     if (typeof tsRaw === "bigint") ts = tsRaw;
     else if (typeof tsRaw === "number" && Number.isFinite(tsRaw)) ts = BigInt(Math.floor(tsRaw));
-
-    // Записываем 64-bit timestamp в первые 8 байт nonce (BE)
     nonce[0] = Number((ts >> 56n) & 0xffn);
     nonce[1] = Number((ts >> 48n) & 0xffn);
     nonce[2] = Number((ts >> 40n) & 0xffn);