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Финальная часть после тестов скорости шифрования кадров (frames)

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RoyceDa
2026-03-21 19:37:52 +02:00
parent 0c823c398f
commit 98fbabc130
2 changed files with 1 additions and 66 deletions
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62
app/providers/CallProvider/CallProvider.tsx
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@@ -74,7 +74,6 @@ export function CallProvider(props : CallProviderProps) {
const roomIdRef = useRef<string>(""); const roomIdRef = useRef<string>("");
const roleRef = useRef<CallRole | null>(null); const roleRef = useRef<CallRole | null>(null);
//const [sharedSecret, setSharedSecret] = useState<string>("");
const sharedSecretRef = useRef<string>(""); const sharedSecretRef = useRef<string>("");
const iceServersRef = useRef<RTCIceServer[]>([]); const iceServersRef = useRef<RTCIceServer[]>([]);
const remoteAudioRef = useRef<HTMLAudioElement | null>(null); const remoteAudioRef = useRef<HTMLAudioElement | null>(null);
@@ -98,67 +97,6 @@ export function CallProvider(props : CallProviderProps) {
} }
}, [callState]); }, [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(() => { useEffect(() => {
/** /**
* Нам нужно получить ICE серверы для установки соединения из разных сетей * Нам нужно получить ICE серверы для установки соединения из разных сетей

3
app/providers/CallProvider/audioE2EE.ts
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@@ -22,12 +22,9 @@ function toUint8Array(input: KeyInput): Uint8Array {
function fillNonceFromTimestamp(nonce: Uint8Array, tsRaw: unknown): void { function fillNonceFromTimestamp(nonce: Uint8Array, tsRaw: unknown): void {
nonce.fill(0); nonce.fill(0);
let ts = 0n; let ts = 0n;
if (typeof tsRaw === "bigint") ts = tsRaw; if (typeof tsRaw === "bigint") ts = tsRaw;
else if (typeof tsRaw === "number" && Number.isFinite(tsRaw)) ts = BigInt(Math.floor(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[0] = Number((ts >> 56n) & 0xffn);
nonce[1] = Number((ts >> 48n) & 0xffn); nonce[1] = Number((ts >> 48n) & 0xffn);
nonce[2] = Number((ts >> 40n) & 0xffn); nonce[2] = Number((ts >> 40n) & 0xffn);