try to fix vc delay

android/app/src/main/assets/public/main.js

@@ -4206,6 +4206,55 @@ function handleChatInputKey(event) {
 let dmMessagePollInterval = null;
 let appWebSocket = null;
 let dmMessageLoadTimeout = null;
+let appWsPingTimer = null;
+let callStatesResyncTimer = null;
+
+//Keepalive app-WS: tekstowy "ping" co 20 s. Serwer zamyka połączenie po 75 s
+//bezczynności (AppWsIdleTimeout) - bez tego half-open TCP po zabiciu aplikacji /
+//zmianie sieci wisiał na serwerze godzinami, cleanup ActiveCalls nie biegł i
+//wskaźniki rozmów pokazywały uczestników, którzy dawno wyszli. Ping utrzymuje
+//też mapping NAT przy danych komórkowych, więc eventy (call_update!) nie giną.
+function startAppWsPing() {
+    if (appWsPingTimer) clearInterval(appWsPingTimer);
+    appWsPingTimer = setInterval(() => {
+        if (appWebSocket && appWebSocket.readyState === WebSocket.OPEN) {
+            try { appWebSocket.send("ping"); } catch (e) {}
+        }
+    }, 20000);
+}
+
+function stopAppWsPing() {
+    if (appWsPingTimer) { clearInterval(appWsPingTimer); appWsPingTimer = null; }
+}
+
+//Pełna resynchronizacja wskaźników rozmów: zgubiony podczas przerwy w app-WS
+//call_update NIGDY nie jest odtwarzany przez serwer, a seedCallStateFromList
+//tylko dodaje wpisy - bez tego wskaźniki "znikały dopiero po ręcznym odświeżeniu".
+//Dla każdego pokoju znanego jako aktywny (+ otwartego) pyta room/voice/state
+//i koryguje mapę oraz UI. Uruchamiane: po (re)otwarciu WS, co 30 s, przy powrocie
+//karty/aplikacji na wierzch.
+async function resyncAllCallStates() {
+    if (!window.id || typeof fetchEncrypted !== "function" || typeof ensureActiveCallsMap !== "function") return;
+    let map = ensureActiveCallsMap();
+    let keys = Object.keys(map);
+    let currentKey = window.currentDmId ? sanitizeId(window.currentDmId) : null;
+    if (currentKey && !keys.includes(currentKey)) keys.push(currentKey);
+    if (keys.length === 0) return;
+
+    for (let key of keys) {
+        let roomType = window.currentDmId === key && window.currentRoomType ? window.currentRoomType
+            : (String(key).includes("_") ? "dms" : "groups");
+        try {
+            let res = await fetchEncrypted("room/voice/state", JSON.stringify({ string1: roomType, string2: key }));
+            if (!res || !res.startsWith("[")) continue;
+            let participants = JSON.parse(res).map(u => String(u).split(";")[0]);
+            if (participants.length > 0) map[key] = participants;
+            else delete map[key];
+        } catch (e) {}
+    }
+    //Po serii poprawek pełne przerysowanie wskaźników (listy + top-bary + details)
+    if (typeof updateVoicePresenceUI === "function") updateVoicePresenceUI();
+}
 
 function setupWebSocket() {
     if (appWebSocket && (appWebSocket.readyState === WebSocket.OPEN || appWebSocket.readyState === WebSocket.CONNECTING)) return;
@@ -4228,6 +4277,17 @@ function setupWebSocket() {
         } finally {
             release();
         }
+
+        startAppWsPing();
+        //Po każdym (re)połączeniu odtwarzamy stan wskaźników - eventy call_update
+        //z czasu rozłączenia są bezpowrotnie stracone
+        resyncAllCallStates();
+        if (!callStatesResyncTimer) {
+            callStatesResyncTimer = setInterval(() => {
+                if (document.visibilityState !== "visible") return;
+                resyncAllCallStates();
+            }, 30000);
+        }
     };
 
     appWebSocket.onmessage = (event) => {
@@ -4273,10 +4333,22 @@ function setupWebSocket() {
     };
 
     appWebSocket.onclose = () => {
+        stopAppWsPing();
         setTimeout(setupWebSocket, 5000);
     };
 }
 
+//Powrót karty/aplikacji na wierzch: mobilne WebView potrafią trzymać martwy
+//socket odkryty bardzo długo (brak ramek = brak detekcji). Wymuszamy próbę
+//połączenia i od razu resynchronizujemy wskaźniki rozmów.
+document.addEventListener("visibilitychange", () => {
+    if (document.visibilityState !== "visible") return;
+    if (appWebSocket && appWebSocket.readyState !== WebSocket.OPEN && appWebSocket.readyState !== WebSocket.CONNECTING) {
+        setupWebSocket();
+    }
+    resyncAllCallStates();
+});
+
 let currentContextMenuMsgId = null;
 let touchContextMenuFired = false;
 

android/app/src/main/assets/public/screens.js

@@ -141,7 +141,7 @@ var invitesEntry = `
 
 var chatScreen = `
 <div style="display: flex; flex-direction: column; height: 100%; width: 100%;">
-    <div id="chat-messages" style="flex-grow: 1; overflow-y: auto; padding: 1rem; display: flex; flex-direction: column; gap: 0.5rem;">
+    <div id="chat-messages" style="flex-grow: 1; overflow-y: auto; overflow-x: hidden; padding: 1rem; display: flex; flex-direction: column; gap: 0.5rem;">
     </div>
     <div id="replying-bar" style="display: none; padding: 0 1rem; border-top: var(--border-width) solid var(--light-border-color); font-size: 0.85rem; justify-content: space-between; align-items: center; background: var(--main-bg-color);">
         <div style="overflow: hidden; text-overflow: ellipsis; white-space: nowrap; flex-grow: 1;">

android/app/src/main/assets/public/style.css

@@ -686,6 +686,7 @@ space{
     flex-shrink: 0;
     cursor: pointer;
     margin: 0;
+    position: relative;
 }
 .room-pfp {
     width: 2.2rem;
@@ -738,6 +739,7 @@ space{
     display: flex;
     flex-direction: column;
     flex-grow: 1;
+    min-width: 0;
 }
 .chat-message-header {
     display: flex;
@@ -1178,7 +1180,15 @@ input:checked + .toggle-slider:before {
     margin-top: 0.2rem;
 }
 
+/* Czerwona kropka nieprzeczytanych - pozycjonowana absolutnie. Odstęp liczony tak,
+   żeby ŚRODEK kropki leżał na tej samej osi co środek 1rem-owego .call-indicator
+   (0.6rem paddingu + (1rem - 0.65rem)/2) - inaczej kropka "ucieka" w prawo
+   względem większej ikony słuchawek. */
 .unread-badge {
+    position: absolute;
+    right: calc(0.6rem + (1rem - 0.65rem) / 2);
+    top: 50%;
+    margin-top: -0.325rem;
     width: 0.65rem;
     height: 0.65rem;
     min-width: 0.65rem;
@@ -1186,7 +1196,6 @@ input:checked + .toggle-slider:before {
     background-color: var(--big-red);
     border-radius: 50%;
     flex-shrink: 0;
-    margin-right: 0.5rem;
     transition: transform 0.2s cubic-bezier(0.175, 0.885, 0.32, 1.275);
     transform: scale(0);
 }

android/app/src/main/assets/public/voice.js

@@ -1,7 +1,6 @@
 let currentVoiceWs = null;
 let currentVoiceCallId = null;
 let currentVoiceCallType = null;
-let voiceWsHeartbeat = null;
 let voiceActiveUsers = [];
 let voiceAesKeyBytes = null;      // grupowy klucz rozmowy (root) - dystrybuowany hybrydą PQ
 let voiceUiContainer = null;
@@ -107,8 +106,8 @@ function getVoicePeerSenderKey(clientIdStr) {
 
 
 let micStream = null, audioContext = null, isAudioWorkletLoaded = false;
-let micSourceNode = null, senderNode = null, receiverNode = null, audioEncoder = null;
+let micSourceNode = null, senderNode = null, receiverNode = null;
-let remoteAudio = {}, audioDecoders = {}, audioDecryptChains = {};
+let remoteAudio = {}, audioDecryptChains = {};
 let knownVoiceUsers = new Set();
 let isMicMuted = false;
 let remoteVideos = {}, videoDecryptChains = {};
@@ -126,22 +125,26 @@ let isVoiceSpeakerMode = false;
 
 const SPEAKING_HANGOVER_MS = 400;
 
-// ===== Voice connection resilience =====
+//Zawory kolejki socketu: TCP dowiezie zakolejkowane bajty tylko spóźnione,
-// Every client broadcasts a tiny encrypted ping so peers can tell "quiet
+//więc progi są hierarchiczne - wideo ustępuje przy 12 KB, audio dopiero przy
-// user" apart from "crashed/half-open socket". Peers silent for too long are
+//14 KB (~270 ms głosu). Przy korku najpierw znika obraz, głos leci dalej.
-// dropped locally; a dead server socket triggers an automatic rejoin.
+const VOICE_WS_VIDEO_QUEUE_LIMIT_BYTES = 12 * 1024;
+const VOICE_WS_AUDIO_QUEUE_LIMIT_BYTES = 14 * 1024;
+
+//Resilience: peer bez pakietów >7 s = lokalnie martwy (media wycięte), socket
+//bez ruchu od ŻYWYCH peerów >15 s = wymuszony reconnect. Pings co 2 s.
 const VOICE_PEER_TIMEOUT_MS = 7000;
 const VOICE_PING_INTERVAL_MS = 2000;
-const VOICE_SERVER_TIMEOUT_MS = 10000;
+const VOICE_SERVER_TIMEOUT_MS = 15000;
-const VOICE_MAX_RECONNECTS = 2;
+const VOICE_MAX_RECONNECTS = 4;
+const VOICE_ROSTER_POLL_MS = 5000;
 let peerLastSeen = {};
 let deadPeers = new Set();
 let lastVoiceServerPacketAt = 0;
-//Czy w bieżącym okresie socketu byli jacyś inni uczestnicy - baza dla
-//resetu baseline'u ciszy w watchdogu (patrz komentarz przy użyciu)
 let voiceHadRemotePeers = false;
 let voicePingTimer = null;
 let voiceRosterRetryTimer = null;
+let voiceRosterPollTimer = null;
 let lastVoiceKeyRequestAt = 0;
 let voiceReconnectAttempts = 0;
 let voiceSocketOpenedAt = 0;
@@ -167,11 +170,21 @@ function cleanupPeerMedia(userId) {
 function sendVoicePing() {
     if (!currentVoiceWs || currentVoiceWs.readyState !== WebSocket.OPEN || !voiceMySenderCryptoKey) return;
     const iv = nextVoiceIv();
-    crypto.subtle.encrypt({name: "AES-GCM", iv}, voiceMySenderCryptoKey, new Uint8Array([Date.now() & 255])).then(cipher => {
+    //Typ 5: prośba o echo. Payload = 8B timestamp (ms, WŁASNY zegar). Odbiorca
+    //odsyła te same bajty jako typ 6, a RTT liczę dopiero na jego powrocie.
+    const tsData = BigInt(Date.now());
+    const tsBuf = new ArrayBuffer(8);
+    new DataView(tsBuf).setBigUint64(0, tsData);
+    crypto.subtle.encrypt({name: "AES-GCM", iv}, voiceMySenderCryptoKey, new Uint8Array(tsBuf)).then(cipher => {
         if (!currentVoiceWs || currentVoiceWs.readyState !== WebSocket.OPEN) return;
         const payload = new Uint8Array(14 + cipher.byteLength);
         payload[0] = 5; payload[1] = 0; payload.set(iv, 2); payload.set(new Uint8Array(cipher), 14);
         currentVoiceWs.send(payload.buffer);
+        voiceOutstandingPings.set(tsData, Date.now());
+        const nowMs = Date.now();
+        for (const [k, sentAt] of voiceOutstandingPings) {
+            if (nowMs - sentAt > 15000) voiceOutstandingPings.delete(k);
+        }
     }).catch(() => {});
 }
 
@@ -217,9 +230,15 @@ const VOICE_FPS_OPTIONS = [15, 24, 30, 60];
 let voiceSettings = loadVoiceSettings();
 
 function loadVoiceSettings() {
-    const defaults = { micId: "", camId: "", speakerId: "", camRes: "1280x720", camFps: 24, screenRes: "1920x1080", screenFps: 30, sensitivity: VOICE_SENSITIVITY_DEFAULT };
+    //Globalny default 480p@24; stare zapisane defaulty (720p/1080p/30fps)
+    //migrują na nowe, świadome wybory użytkownika zostają.
+    const defaults = { micId: "", camId: "", speakerId: "", camRes: "640x480", camFps: 24, screenRes: "640x480", screenFps: 24, sensitivity: VOICE_SENSITIVITY_DEFAULT };
     try {
-        return Object.assign(defaults, JSON.parse(localStorage.getItem(VOICE_SETTINGS_KEY)) || {});
+        let s = Object.assign({}, JSON.parse(localStorage.getItem(VOICE_SETTINGS_KEY)) || {});
+        if (s.camRes === "1280x720") s.camRes = defaults.camRes;
+        if (s.screenRes === "1920x1080") s.screenRes = defaults.screenRes;
+        if (s.screenFps === 30 || s.screenFps === "30") s.screenFps = defaults.screenFps;
+        return Object.assign(defaults, s);
     } catch (e) {
         return defaults;
     }
@@ -274,6 +293,35 @@ async function applyVoiceSpeakerSink() {
 let voiceConnectedAt = null;
 let voiceTimerInterval = null;
 let voicePingMs = null;
+//Kontrola congestionu: bufferedAmount nie widzi backlogu kernela TCP, prawdziwy
+//obraz daje RTT z pingów. Baseline = MINIMUM z okna próbek (odporne na skoki
+//i na peera sztucznie opóźniającego własne echa - stałe kłamstwo podnosi minimum).
+let voiceRttFastMs = 0;
+let voiceRttSamples = [];
+let voiceRttBaseMs = 0;
+let voiceAudioHoldUntil = 0; //histereza zaworu audio
+
+//Echo-ping (typ 5 = prośba, typ 6 = echo): RTT wyłącznie na WŁASNYM zegarze -
+//wysyłam timestamp, peer odbija te same bajty, odejmuję czas nadania. Zegar
+//peera nigdy nie jest używany (może być przekręcony lub spreparowany).
+const voiceOutstandingPings = new Map(); // BigInt(ts) -> czas nadania
+let voiceLastEchoAt = 0;                 // rate-limit odpowiedzi echem
+let voiceLastRttSampleAt = 0;            // kiedy ostatnio spłynął prawdziwy pomiar
+
+function voiceCongestionDelay() {
+    if (!voiceRttFastMs || !voiceRttBaseMs) return 0;
+    return Math.min(5000, Math.max(0, voiceRttFastMs - voiceRttBaseMs));
+}
+
+function voiceRecordRttSample(rtt) {
+    voiceLastRttSampleAt = Date.now();
+    voicePingMs = voicePingMs === null ? rtt : Math.round(voicePingMs * 0.7 + rtt * 0.3);
+    voiceRttFastMs = voiceRttFastMs ? Math.round(voiceRttFastMs * 0.5 + rtt * 0.5) : rtt;
+    voiceRttSamples.push(rtt);
+    if (voiceRttSamples.length > 12) voiceRttSamples.shift();
+    voiceRttBaseMs = Math.min.apply(null, voiceRttSamples);
+    updateVoicePing();
+}
 
 function formatVoiceDuration(ms) {
     let totalSec = Math.floor(ms / 1000);
@@ -630,7 +678,11 @@ setInterval(() => {
     }
     // Half-open socket watchdog + backgrounded-tab audio recovery.
     if (currentVoiceWs && currentVoiceWs.readyState === WebSocket.OPEN) {
-        const others = voiceActiveUsers.filter(u => u !== String(window.id)).length;
+        //deadPeers NIE liczą się do "others": ghost na rosterze (martwy TCP
+        //jeszcze nie zsynchronizowany przez serwer) nigdy nic nie wyśle, więc
+        //zliczany jak żywy wymuszał reconnecty w pętli - migające "Connecting"
+        //w zdrowej rozmowie.
+        const others = voiceActiveUsers.filter(u => u !== String(window.id) && !deadPeers.has(u)).length;
         // Baseline ciszy odświeżany przy przejściu 0->N uczestników: solo w
         // rozmowie NIE MA żadnego ruchu przychodzącego, więc po pojawieniu się
         // nowej osoby lastVoiceServerPacketAt bywa nieświeży z okresu solo.
@@ -693,8 +745,8 @@ class AudioReceiver extends AudioWorkletProcessor {
     //   PREBUF_START  - initial prebuffer before first playback (~43 ms)
     //   PREBUF_RESUME - re-prebuffer after a starvation event (~32 ms)
     //   HARD_LIMIT    - burst ceiling (~128 ms): snap back to target immediately
-    // Backlog slightly above target is drained by a gentle trim (1 sample per
+    // Backlog above target is drained adaptively (proportional to excess, see
-    // quantum) instead of letting latency creep up to the hard limit.
+    // process()) so latency returns to ~43 ms quickly after network bursts.
     constructor(options) {
         super();
         this.users = {};
@@ -732,7 +784,12 @@ class AudioReceiver extends AudioWorkletProcessor {
                 if (available < output.length) { user.buffering = true; }
                 else {
                     if (available > 6144) user.readPtr = (user.writePtr - 2048 + 65536) % 65536;
-                    else if (available > 2049) user.readPtr = (user.readPtr + 1) % 65536;
+                    else if (available > 2049) {
+                        //Adaptacyjny drain: nadmiar nad target odprowadzany proporcjonalnie
+                        //(1 + excess/256 /kwant) - powrót z ~128 ms do ~43 ms w ułamku sekundy
+                        const excess = available - 2048;
+                        user.readPtr = (user.readPtr + 1 + (excess >> 8)) % 65536;
+                    }
                     let vol = this.volumes[id] !== undefined ? this.volumes[id] : 1.0;
                     for (let i = 0; i < output.length; i++) {
                         output[i] += (user.buffer[user.readPtr] / 32768) * vol;
@@ -890,6 +947,38 @@ function bytesToB64u(a) {
     return btoa(bin).replace(/\+/g, "-").replace(/\//g, "_").replace(/=+$/, "");
 }
 
+//Roster pokoju z voice-servera (źródło prawdy) -> synchronizacja kafelków.
+//Steady-poll co VOICE_ROSTER_POLL_MS - bez tego sweep ghosta/leave bez
+//call_update nigdy nie docierał do UI.
+async function fetchVoiceRoster() {
+    if (!currentVoiceWs || currentVoiceWs.readyState !== WebSocket.OPEN || !currentVoiceCallId) return;
+    try {
+        const baseUrl = currentVoiceWs.url.split('/ws')[0];
+        const httpUrl = baseUrl.replace('ws:', 'http:').replace('wss:', 'https:');
+        const t0 = performance.now();
+        const res = await fetch(`${httpUrl}/room/users?room=${encodeURIComponent(currentVoiceCallId)}`);
+        const usersRaw = await res.json();
+        //HTTP tylko jako szacunek gdy brak prawdziwych próbek RTT (solo / stary peer)
+        if (voicePingMs === null || Date.now() - voiceLastRttSampleAt > 25000) {
+            voicePingMs = Math.round(performance.now() - t0);
+            updateVoicePing();
+        }
+        const rawUsers = usersRaw.map(u => String(u));
+        //Serwer jest źródłem prawdy o składzie - nie filtrujemy po
+        //deadPeers (audyt S15: ukryty peer nadal dostaje audio)
+        for (const gone of [...deadPeers]) {
+            if (!rawUsers.includes(gone)) deadPeers.delete(gone);
+        }
+        const users = rawUsers;
+        let changed = (users.length !== voiceActiveUsers.length);
+        for(let u of users) if(!voiceActiveUsers.includes(u)) changed = true;
+        if(changed) {
+            voiceActiveUsers = users;
+            updateVoiceUIUsers();
+        }
+    } catch (e) {}
+}
+
 async function startVoiceSession(serverUrl, ticket) {
     if (!ticket) throw new Error("missing voice ticket");
     // Single-session invariant: a stale socket (e.g. a double-fired join)
@@ -932,32 +1021,7 @@ async function startVoiceSession(serverUrl, ticket) {
                 else btn.classList.remove("voice-btn-danger");
             }
             
-            const fetchUsers = async () => {
+            fetchVoiceRoster();
-                if (!currentVoiceWs || currentVoiceWs.readyState !== WebSocket.OPEN || !currentVoiceCallId) return;
-                try {
-                    const baseUrl = currentVoiceWs.url.split('/ws')[0];
-                    const httpUrl = baseUrl.replace('ws:', 'http:').replace('wss:', 'https:');
-                    const t0 = performance.now();
-                    const res = await fetch(`${httpUrl}/room/users?room=${currentVoiceCallId}`);
-                    const usersRaw = await res.json();
-                    voicePingMs = Math.round(performance.now() - t0);
-                    updateVoicePing();
-                    const rawUsers = usersRaw.map(u => String(u));
-                    //Serwer jest źródłem prawdy o składzie - nie filtrujemy po
-                    //deadPeers (audyt S15: ukryty peer nadal dostaje audio)
-                    for (const gone of [...deadPeers]) {
-                        if (!rawUsers.includes(gone)) deadPeers.delete(gone);
-                    }
-                    const users = rawUsers;
-                    let changed = (users.length !== voiceActiveUsers.length);
-                    for(let u of users) if(!voiceActiveUsers.includes(u)) changed = true;
-                    if(changed) {
-                        voiceActiveUsers = users;
-                        updateVoiceUIUsers();
-                    }
-                } catch (e) {}
-            };
-            fetchUsers();
             //The voice-server roster is joined only AFTER ticket validation (an
             //HTTP roundtrip to the main server), so the immediate fetch can still
             //miss us - retry quickly until our own tile shows up instead of
@@ -971,13 +1035,17 @@ async function startVoiceSession(serverUrl, ticket) {
                     voiceRosterRetryTimer = null;
                     return;
                 }
-                fetchUsers();
+                fetchVoiceRoster();
             }, 400);
 
             voiceReconnectAttempts = 0;
             lastVoiceServerPacketAt = Date.now();
             if (voicePingTimer) clearInterval(voicePingTimer);
             voicePingTimer = setInterval(sendVoicePing, VOICE_PING_INTERVAL_MS);
+            //Steady roster poll - utrzymuje kafelki zgodne z prawdą serwera
+            //(ghost-sweep, leave bez call_update) przez całą rozmowę.
+            if (voiceRosterPollTimer) clearInterval(voiceRosterPollTimer);
+            voiceRosterPollTimer = setInterval(fetchVoiceRoster, VOICE_ROSTER_POLL_MS);
 
             //S16: getUserMedia/worklet/rezump AudioContext potrafią rzucić
             //(brak uprawnień, zniknięte urządzenie) - bez obsługi zostawał
@@ -1033,12 +1101,18 @@ async function startVoiceSession(serverUrl, ticket) {
             }
             
             let vadHangover = 0;
+            const myIdStr = String(window.id);
             senderNode.port.onmessage = async (e) => {
                 if (!currentVoiceWs || currentVoiceWs.readyState !== WebSocket.OPEN) return;
-                // Backpressure valve: when the socket is congested, drop mic
+                //Zawór bajtowy + zawór RTT (kolejka kernela jest niewidoczna dla
-                // frames instead of queueing them - queued audio would arrive
+                //bufferedAmount). Drop >450 ms nadwyżki z histerezą 700 ms - lepiej
-                // minutes late and drown the connection for every stream.
+                //urwać ułamek sekundy niż mówić z sekundami opóźnienia.
-                if (currentVoiceWs.bufferedAmount > 256 * 1024) {
+                if (currentVoiceWs.bufferedAmount > VOICE_WS_AUDIO_QUEUE_LIMIT_BYTES) {
+                    vadHangover = 0;
+                    return;
+                }
+                if (voiceCongestionDelay() > 450) voiceAudioHoldUntil = Date.now() + 700;
+                if (Date.now() < voiceAudioHoldUntil) {
                     vadHangover = 0;
                     return;
                 }
@@ -1053,7 +1127,7 @@ async function startVoiceSession(serverUrl, ticket) {
                 
                 if (db >= sensitivity) vadHangover = 75;
                 let isSpeaking = vadHangover > 0 && !isMicMuted;
-                setTileSpeaking(String(window.id), isSpeaking);
+                setTileSpeaking(myIdStr, isSpeaking);
 
                 if (vadHangover > 0 && !isMicMuted) {
                     vadHangover--;
@@ -1084,7 +1158,44 @@ async function startVoiceSession(serverUrl, ticket) {
             if (deadPeers.has(clientId)) deadPeers.delete(clientId);
 
             const type = view.getUint8(8);
-            if (type === 5) return; // liveness ping - presence already recorded
+            if (type === 5 || type === 6) {
+                //Liveness już zapisany wyżej. Dalej tylko przy zainstalowanym kluczu.
+                if (voiceAesKeyBytes) {
+                    const senderKey = getVoicePeerSenderKey(clientId);
+                    if (senderKey) {
+                        //Offsety ODBIORCZE: serwer dokleja 8B nadawcy, więc iv=[10..22),
+                        //ct=[22..). Format wysyłki (iv=2, ct=14) jest INNY - pomylenie
+                        //psuje każdą deszyfrację cicho (GCM auth fail -> .catch)
+                        const piv = new Uint8Array(arrayBuffer, 10, 12);
+                        const pct = new Uint8Array(arrayBuffer, 22);
+                        crypto.subtle.decrypt({name: "AES-GCM", iv: piv}, senderKey, pct).then(plain => {
+                            if (plain.byteLength !== 8) return; //stary klient: ping bez timestampu
+                            const tsData = new DataView(plain).getBigUint64(0);
+                            if (type === 5) {
+                                //Odbij te same bajty własnym podkluczem i świeżym IV; rate-limit
+                                const nowMs = Date.now();
+                                if (nowMs - voiceLastEchoAt < 150) return;
+                                voiceLastEchoAt = nowMs;
+                                if (!currentVoiceWs || currentVoiceWs.readyState !== WebSocket.OPEN || !voiceMySenderCryptoKey) return;
+                                const riv = nextVoiceIv();
+                                crypto.subtle.encrypt({name: "AES-GCM", iv: riv}, voiceMySenderCryptoKey, plain).then(rcipher => {
+                                    if (!currentVoiceWs || currentVoiceWs.readyState !== WebSocket.OPEN) return;
+                                    const rpayload = new Uint8Array(14 + rcipher.byteLength);
+                                    rpayload[0] = 6; rpayload[1] = 0; rpayload.set(riv, 2); rpayload.set(new Uint8Array(rcipher), 14);
+                                    currentVoiceWs.send(rpayload.buffer);
+                                }).catch(() => {});
+                            } else {
+                                //Typ 6: powrót własnego timestampu - RTT na WŁASNYM zegarze.
+                                if (!voiceOutstandingPings.has(tsData)) return;
+                                voiceOutstandingPings.delete(tsData);
+                                const rtt = Date.now() - Number(tsData);
+                                if (rtt >= 1 && rtt < 15000) voiceRecordRttSample(rtt);
+                            }
+                        }).catch(() => {});
+                    }
+                }
+                return;
+            }
             
             if (!voiceActiveUsers.includes(clientId) && clientId !== String(window.id)) {
                 const baseUrl = currentVoiceWs.url.split('/ws')[0];
@@ -1165,6 +1276,7 @@ async function startVoiceSession(serverUrl, ticket) {
                                     frame.close();
                                 },
                                 error: e => {
+                                    voiceLogError('videodecoder[' + decoderId + ']', e);
                                     if (remoteVideos[decoderId]) {
                                         try { remoteVideos[decoderId].decoder.close(); } catch(err) {}
                                         remoteVideos[decoderId].decoder = null;
@@ -1190,11 +1302,14 @@ async function startVoiceSession(serverUrl, ticket) {
                         try {
                             streamData.decoder.decode(chunk);
                         } catch(err) {
+                            voiceLogError('videodecode[' + decoderId + ']', err);
                             try { streamData.decoder.close(); } catch(e){}
                             streamData.decoder = null;
                             streamData.hasKeyFrame = false;
                         }
-                    } catch (err) {}
+                    } catch (err) {
+                        voiceLogError('vdecrypt[' + decoderId + ']', err);
+                    }
                 });
             } else if (type === 3) {
                 (async () => {
@@ -1332,8 +1447,6 @@ async function handleCallUpdate(data) {
         for (let u in remoteAudio) {
             let pureU = u.replace("_screen", "");
             if (!voiceActiveUsers.includes(pureU) && pureU !== myIdStr) {
-                if (audioDecoders[u]) try { audioDecoders[u].close(); } catch(e){}
-                delete audioDecoders[u];
                 delete remoteAudio[u];
                 const rn = u.endsWith("_screen") ? screenReceiverNode : receiverNode;
                 if(rn) rn.port.postMessage({ action: 'remove', clientId: u });
@@ -2490,7 +2603,6 @@ function toggleVoiceMic() {
 // keeping the UI, mic stream and AudioContext alive for an automatic rejoin.
 function teardownVoiceTransport() {
     if (voicePingTimer) { clearInterval(voicePingTimer); voicePingTimer = null; }
-    if (voiceWsHeartbeat) { clearInterval(voiceWsHeartbeat); voiceWsHeartbeat = null; }
     if (camControl) { try { camControl.stop(); } catch (e) {} camControl = null; }
     if (screenControl) { try { screenControl.stop(); } catch (e) {} screenControl = null; }
     if (currentVoiceWs) {
@@ -2507,6 +2619,7 @@ function teardownVoiceTransport() {
     peerLastSeen = {};
     deadPeers.clear();
     if (voiceRosterRetryTimer) { clearInterval(voiceRosterRetryTimer); voiceRosterRetryTimer = null; }
+    if (voiceRosterPollTimer) { clearInterval(voiceRosterPollTimer); voiceRosterPollTimer = null; }
     lastVoiceKeyRequestAt = 0;
     //Stale participants from the previous attempt must not linger in the UI
     voiceActiveUsers = [];
@@ -2516,6 +2629,12 @@ function teardownVoiceTransport() {
     setVoiceRootKey(null);
     voiceSendEpoch = null;
     voiceSendCounter = 0n;
+    voiceRttFastMs = 0;
+    voiceRttSamples = [];
+    voiceRttBaseMs = 0;
+    voiceAudioHoldUntil = 0;
+    voiceOutstandingPings.clear();
+    voiceLastRttSampleAt = 0;
     lastVoiceServerPacketAt = Date.now();
 }
 
@@ -2532,6 +2651,10 @@ function scheduleVoiceReconnect() {
     voiceReconnectAttempts++;
     setVoiceStatus("connecting");
     teardownVoiceTransport();
+    //Zmiana sieci = stary RTT dotyczy martwej ścieżki; bez resetu wskaźnik
+    //wisiałby zamrożony, a fallback HTTP nie miał prawa go nadpisać
+    voicePingMs = null;
+    updateVoicePing();
     setTimeout(async () => {
         if (!currentVoiceCallId || !currentVoiceCallType) return;
         try {
@@ -2543,7 +2666,7 @@ function scheduleVoiceReconnect() {
             console.warn("Voice reconnect failed", err);
             scheduleVoiceReconnect();
         }
-    }, 800);
+    }, Math.min(4000, 800 * voiceReconnectAttempts));
 }
 
 function endVoiceSession() {
@@ -2583,24 +2706,27 @@ function endVoiceSession() {
         micStream = null;
     }
     if (micSourceNode) { micSourceNode.disconnect(); micSourceNode = null; }
-    if (audioEncoder) { try { audioEncoder.close(); } catch(e){} audioEncoder = null; }
+
-    
-    if (voiceWsHeartbeat) { clearInterval(voiceWsHeartbeat); voiceWsHeartbeat = null; }
     if (voicePingTimer) { clearInterval(voicePingTimer); voicePingTimer = null; }
     if (voiceRosterRetryTimer) { clearInterval(voiceRosterRetryTimer); voiceRosterRetryTimer = null; }
+    if (voiceRosterPollTimer) { clearInterval(voiceRosterPollTimer); voiceRosterPollTimer = null; }
     peerLastSeen = {};
     deadPeers.clear();
     lastVoiceKeyRequestAt = 0;
+    voiceReconnectAttempts = 0;
     voiceActiveUsers = [];
     voiceHadRemotePeers = false;
-    
+    voiceRttFastMs = 0;
-    for (let id in audioDecoders) {
+    voiceRttSamples = [];
-        try { audioDecoders[id].close(); } catch(e){}
+    voiceRttBaseMs = 0;
-    }
+    voiceAudioHoldUntil = 0;
+    voiceOutstandingPings.clear();
+    voiceLastEchoAt = 0;
+    voiceLastRttSampleAt = 0;
+
     for (let id in audioDecryptChains) {
         if(audioDecryptChains[id]) audioDecryptChains[id].clear();
     }
-    audioDecoders = {};
     audioDecryptChains = {};
     remoteAudio = {};
     
@@ -2672,8 +2798,11 @@ async function startHardwareVideo(btnEl, streamPromise, typeLabel, typeId) {
                 screenSenderNode = new AudioWorkletNode(audioContext, 'audio-sender');
                 screenSenderNode.port.onmessage = async (e) => {
                     if (currentVoiceWs?.readyState !== WebSocket.OPEN || !voiceMySenderCryptoKey) return;
-                    // Same backpressure valve as the mic sender.
+                    // Ten sam zawór co mikrofon (limit bajtów + korek RTT) - audio
-                    if (currentVoiceWs.bufferedAmount > 256 * 1024) return;
+                    // ma pierwszeństwo przed wideo na wspólnym sockecie.
+                    if (currentVoiceWs.bufferedAmount > VOICE_WS_AUDIO_QUEUE_LIMIT_BYTES) return;
+                    if (voiceCongestionDelay() > 450) voiceAudioHoldUntil = Date.now() + 700;
+                    if (Date.now() < voiceAudioHoldUntil) return;
                     const iv = nextVoiceIv();
                     const cipher = await crypto.subtle.encrypt({name: "AES-GCM", iv}, voiceMySenderCryptoKey, e.data.audioData);
                     const payload = new Uint8Array(14 + cipher.byteLength);
@@ -2703,7 +2832,24 @@ async function startHardwareVideo(btnEl, streamPromise, typeLabel, typeId) {
             return { width, height, fps, bitrate };
         };
 
-        let currentConfig = getCurrentConfig(), isRunning = true, frameCount = 0;
+        let currentConfig = getCurrentConfig(), isRunning = true, lastKeyframeAt = 0;
+        //AIMD + slow-start: start na 40% celu (pełną prędkością waliłby w nieznany
+        //uplink, a bufory TCP w kernelu to MB-y = sekundy kolejki zanim cokolwiek
+        //zmierzymy). Sygnał korka: skipy (zawór 12 KB) lub nadwyżka RTT.
+        let brScale = 0.4, encTries = 0, encSkips = 0, brWindowAt = performance.now(), lastBrReconfAt = 0;
+        //Korek trwa mimo podłogi bitrate'u → FPS spada do 1/2, 1/3
+        let fpsDiv = 1, lowStreak = 0, cleanStreak = 0, tickCount = 0;
+        let activeBitrate = Math.round(currentConfig.bitrate * brScale);
+        const applyEncoderBitrate = (force = false) => {
+            const target = Math.max(120000, Math.round(currentConfig.bitrate * brScale));
+            if (!force && Math.abs(target - activeBitrate) / activeBitrate < 0.08) return;
+            if (!force && performance.now() - lastBrReconfAt < 800) return;
+            activeBitrate = target;
+            lastBrReconfAt = performance.now();
+            try {
+                encoder.configure({codec: 'av01.0.04M.08', width: currentConfig.width, height: currentConfig.height, framerate: currentConfig.fps, bitrate: activeBitrate, latencyMode: 'realtime'});
+            } catch (e) {}
+        };
         const encoder = new VideoEncoder({
             output: async (chunk) => {
                 if (currentVoiceWs?.readyState !== WebSocket.OPEN || !voiceMySenderCryptoKey) return;
@@ -2713,9 +2859,9 @@ async function startHardwareVideo(btnEl, streamPromise, typeLabel, typeId) {
                 const payload = new Uint8Array(14 + cipher.byteLength);
                 payload[0] = typeId; payload[1] = chunk.type === 'key' ? 1 : 0; payload.set(iv, 2); payload.set(new Uint8Array(cipher), 14);
                 currentVoiceWs.send(payload.buffer);
-            }, error: (e) => { stopFunction(); }
+            }, error: (e) => { voiceLogError('videoencoder', e); stopFunction(); }
         });
-        encoder.configure({codec: 'av01.0.04M.08', width: currentConfig.width, height: currentConfig.height, framerate: currentConfig.fps, bitrate: currentConfig.bitrate, latencyMode: 'realtime'});
+        encoder.configure({codec: 'av01.0.04M.08', width: currentConfig.width, height: currentConfig.height, framerate: currentConfig.fps, bitrate: activeBitrate, latencyMode: 'realtime'});
 
             // Transient capture-mode wobbles must not thrash the encoder -
             // commit only after ~300 ms stable. Camera size/fps changes arrive
@@ -2733,23 +2879,65 @@ async function startHardwareVideo(btnEl, streamPromise, typeLabel, typeId) {
                         pendingCfg = newConfig;
                         pendingAt = performance.now();
                     } else if (performance.now() - pendingAt > 300) {
-                        currentConfig = newConfig; forceKeyframe = true; frameCount = 0; pendingCfg = null;
+                        //Zmiana geometrii źródła - uczony scale łącza zostaje,
-                        encoder.configure({codec: 'av01.0.04M.08', width: currentConfig.width, height: currentConfig.height, framerate: currentConfig.fps, bitrate: currentConfig.bitrate, latencyMode: 'realtime'});
+                        //resetujemy tylko redukcję FPS
+                        currentConfig = newConfig; forceKeyframe = true; lastKeyframeAt = 0; pendingCfg = null;
+                        activeBitrate = Math.round(currentConfig.bitrate * brScale);
+                        fpsDiv = 1; lowStreak = 0; cleanStreak = 0;
+                        applyEncoderBitrate(true);
                     }
                 } else if (pendingCfg) {
                     pendingCfg = null; // wiggle reverted on its own - ignore
                 }
 
-                if (encoder.state === 'configured' && encoder.encodeQueueSize <= 2 && currentVoiceWs.bufferedAmount < 1024 * 256 && videoEl.readyState >= 2) {
+                //Zawór kolejki socketu (12 KB) ustępuje DALEJ przed progiem audio
-                let frame = null;
+                //(14 KB) - przy korku najpierw znika obraz, głos płynie dalej.
-                try {
+                tickCount++;
-                    frame = new VideoFrame(videoEl, {timestamp: Math.trunc(performance.now() * 1000)});
+                const policySkip = fpsDiv > 1 && (tickCount % fpsDiv !== 0);
-                    frameCount++;
+                let skipped = false;
-                    encoder.encode(frame, {keyFrame: forceKeyframe || (frameCount % (currentConfig.fps * 2)) === 0});
+                if (policySkip) {
-                } catch (err) { } finally { if (frame) frame.close(); }
+                    //celowe tempo (fpsDiv) - nie liczy się do skipRatio
-            }
+                } else if (encoder.state === 'configured' && encoder.encodeQueueSize <= 2 &&
-            setTimeout(encodeLoop, 1000 / currentConfig.fps);
+                    currentVoiceWs.readyState === WebSocket.OPEN &&
-        };
+                    currentVoiceWs.bufferedAmount < VOICE_WS_VIDEO_QUEUE_LIMIT_BYTES && videoEl.readyState >= 2) {
+                    let frame = null;
+                    try {
+                        const nowMs = performance.now();
+                        frame = new VideoFrame(videoEl, {timestamp: Math.trunc(nowMs * 1000)});
+                        //Keyframe wg zegara co ~1 s (nie licznika klatek - zawór je pomija)
+                        const wantKeyframe = forceKeyframe || (nowMs - lastKeyframeAt >= 1000);
+                        encoder.encode(frame, {keyFrame: wantKeyframe});
+                        if (wantKeyframe) lastKeyframeAt = nowMs;
+                    } catch (err) { } finally { if (frame) frame.close(); }
+                } else if (encoder.state === 'configured') {
+                    skipped = true;
+                }
+
+                encTries++;
+                if (skipped) encSkips++;
+                const winNow = performance.now();
+                if (winNow - brWindowAt >= 800) {
+                    const skipRatio = encTries > 0 ? encSkips / encTries : 0;
+                    const congestionMs = voiceCongestionDelay();
+                    if (congestionMs > 450 || skipRatio > 0.3 || congestionMs > 250) {
+                        brScale = Math.max(0.15, brScale * (congestionMs > 450 ? 0.6 : 0.75));
+                        applyEncoderBitrate();
+                        if (brScale <= 0.17) lowStreak++; else lowStreak = 0;
+                        if (lowStreak >= 4 && fpsDiv < 3) { fpsDiv++; lowStreak = 0; }
+                        cleanStreak = 0;
+                    } else if (skipRatio === 0 && congestionMs < 80) {
+                        cleanStreak++;
+                        lowStreak = 0;
+                        if (cleanStreak >= 5) {
+                            if (fpsDiv > 1) { fpsDiv--; }
+                            else if (brScale < 1) { brScale = Math.min(1, brScale * 1.2); applyEncoderBitrate(); }
+                            cleanStreak = 0;
+                        }
+                    }
+                    encTries = 0; encSkips = 0; brWindowAt = winNow;
+                }
+                setTimeout(encodeLoop, 1000 / currentConfig.fps);
+            };
         setTimeout(encodeLoop, 500);
 
         videoTrack.onended = () => { if(isRunning) stopFunction(); };

webroot/main.js

@@ -4206,6 +4206,55 @@ function handleChatInputKey(event) {
 let dmMessagePollInterval = null;
 let appWebSocket = null;
 let dmMessageLoadTimeout = null;
+let appWsPingTimer = null;
+let callStatesResyncTimer = null;
+
+//Keepalive app-WS: tekstowy "ping" co 20 s. Serwer zamyka połączenie po 75 s
+//bezczynności (AppWsIdleTimeout) - bez tego half-open TCP po zabiciu aplikacji /
+//zmianie sieci wisiał na serwerze godzinami, cleanup ActiveCalls nie biegł i
+//wskaźniki rozmów pokazywały uczestników, którzy dawno wyszli. Ping utrzymuje
+//też mapping NAT przy danych komórkowych, więc eventy (call_update!) nie giną.
+function startAppWsPing() {
+    if (appWsPingTimer) clearInterval(appWsPingTimer);
+    appWsPingTimer = setInterval(() => {
+        if (appWebSocket && appWebSocket.readyState === WebSocket.OPEN) {
+            try { appWebSocket.send("ping"); } catch (e) {}
+        }
+    }, 20000);
+}
+
+function stopAppWsPing() {
+    if (appWsPingTimer) { clearInterval(appWsPingTimer); appWsPingTimer = null; }
+}
+
+//Pełna resynchronizacja wskaźników rozmów: zgubiony podczas przerwy w app-WS
+//call_update NIGDY nie jest odtwarzany przez serwer, a seedCallStateFromList
+//tylko dodaje wpisy - bez tego wskaźniki "znikały dopiero po ręcznym odświeżeniu".
+//Dla każdego pokoju znanego jako aktywny (+ otwartego) pyta room/voice/state
+//i koryguje mapę oraz UI. Uruchamiane: po (re)otwarciu WS, co 30 s, przy powrocie
+//karty/aplikacji na wierzch.
+async function resyncAllCallStates() {
+    if (!window.id || typeof fetchEncrypted !== "function" || typeof ensureActiveCallsMap !== "function") return;
+    let map = ensureActiveCallsMap();
+    let keys = Object.keys(map);
+    let currentKey = window.currentDmId ? sanitizeId(window.currentDmId) : null;
+    if (currentKey && !keys.includes(currentKey)) keys.push(currentKey);
+    if (keys.length === 0) return;
+
+    for (let key of keys) {
+        let roomType = window.currentDmId === key && window.currentRoomType ? window.currentRoomType
+            : (String(key).includes("_") ? "dms" : "groups");
+        try {
+            let res = await fetchEncrypted("room/voice/state", JSON.stringify({ string1: roomType, string2: key }));
+            if (!res || !res.startsWith("[")) continue;
+            let participants = JSON.parse(res).map(u => String(u).split(";")[0]);
+            if (participants.length > 0) map[key] = participants;
+            else delete map[key];
+        } catch (e) {}
+    }
+    //Po serii poprawek pełne przerysowanie wskaźników (listy + top-bary + details)
+    if (typeof updateVoicePresenceUI === "function") updateVoicePresenceUI();
+}
 
 function setupWebSocket() {
     if (appWebSocket && (appWebSocket.readyState === WebSocket.OPEN || appWebSocket.readyState === WebSocket.CONNECTING)) return;
@@ -4228,6 +4277,17 @@ function setupWebSocket() {
         } finally {
             release();
         }
+
+        startAppWsPing();
+        //Po każdym (re)połączeniu odtwarzamy stan wskaźników - eventy call_update
+        //z czasu rozłączenia są bezpowrotnie stracone
+        resyncAllCallStates();
+        if (!callStatesResyncTimer) {
+            callStatesResyncTimer = setInterval(() => {
+                if (document.visibilityState !== "visible") return;
+                resyncAllCallStates();
+            }, 30000);
+        }
     };
 
     appWebSocket.onmessage = (event) => {
@@ -4273,10 +4333,22 @@ function setupWebSocket() {
     };
 
     appWebSocket.onclose = () => {
+        stopAppWsPing();
         setTimeout(setupWebSocket, 5000);
     };
 }
 
+//Powrót karty/aplikacji na wierzch: mobilne WebView potrafią trzymać martwy
+//socket odkryty bardzo długo (brak ramek = brak detekcji). Wymuszamy próbę
+//połączenia i od razu resynchronizujemy wskaźniki rozmów.
+document.addEventListener("visibilitychange", () => {
+    if (document.visibilityState !== "visible") return;
+    if (appWebSocket && appWebSocket.readyState !== WebSocket.OPEN && appWebSocket.readyState !== WebSocket.CONNECTING) {
+        setupWebSocket();
+    }
+    resyncAllCallStates();
+});
+
 let currentContextMenuMsgId = null;
 let touchContextMenuFired = false;
 

webroot/screens.js

@@ -141,7 +141,7 @@ var invitesEntry = `
 
 var chatScreen = `
 <div style="display: flex; flex-direction: column; height: 100%; width: 100%;">
-    <div id="chat-messages" style="flex-grow: 1; overflow-y: auto; padding: 1rem; display: flex; flex-direction: column; gap: 0.5rem;">
+    <div id="chat-messages" style="flex-grow: 1; overflow-y: auto; overflow-x: hidden; padding: 1rem; display: flex; flex-direction: column; gap: 0.5rem;">
     </div>
     <div id="replying-bar" style="display: none; padding: 0 1rem; border-top: var(--border-width) solid var(--light-border-color); font-size: 0.85rem; justify-content: space-between; align-items: center; background: var(--main-bg-color);">
         <div style="overflow: hidden; text-overflow: ellipsis; white-space: nowrap; flex-grow: 1;">

webroot/style.css

@@ -686,6 +686,7 @@ space{
     flex-shrink: 0;
     cursor: pointer;
     margin: 0;
+    position: relative;
 }
 .room-pfp {
     width: 2.2rem;
@@ -738,6 +739,7 @@ space{
     display: flex;
     flex-direction: column;
     flex-grow: 1;
+    min-width: 0;
 }
 .chat-message-header {
     display: flex;
@@ -1178,7 +1180,15 @@ input:checked + .toggle-slider:before {
     margin-top: 0.2rem;
 }
 
+/* Czerwona kropka nieprzeczytanych - pozycjonowana absolutnie. Odstęp liczony tak,
+   żeby ŚRODEK kropki leżał na tej samej osi co środek 1rem-owego .call-indicator
+   (0.6rem paddingu + (1rem - 0.65rem)/2) - inaczej kropka "ucieka" w prawo
+   względem większej ikony słuchawek. */
 .unread-badge {
+    position: absolute;
+    right: calc(0.6rem + (1rem - 0.65rem) / 2);
+    top: 50%;
+    margin-top: -0.325rem;
     width: 0.65rem;
     height: 0.65rem;
     min-width: 0.65rem;
@@ -1186,7 +1196,6 @@ input:checked + .toggle-slider:before {
     background-color: var(--big-red);
     border-radius: 50%;
     flex-shrink: 0;
-    margin-right: 0.5rem;
     transition: transform 0.2s cubic-bezier(0.175, 0.885, 0.32, 1.275);
     transform: scale(0);
 }

webroot/voice.js

@@ -1,7 +1,6 @@
 let currentVoiceWs = null;
 let currentVoiceCallId = null;
 let currentVoiceCallType = null;
-let voiceWsHeartbeat = null;
 let voiceActiveUsers = [];
 let voiceAesKeyBytes = null;      // grupowy klucz rozmowy (root) - dystrybuowany hybrydą PQ
 let voiceUiContainer = null;
@@ -107,8 +106,8 @@ function getVoicePeerSenderKey(clientIdStr) {
 
 
 let micStream = null, audioContext = null, isAudioWorkletLoaded = false;
-let micSourceNode = null, senderNode = null, receiverNode = null, audioEncoder = null;
+let micSourceNode = null, senderNode = null, receiverNode = null;
-let remoteAudio = {}, audioDecoders = {}, audioDecryptChains = {};
+let remoteAudio = {}, audioDecryptChains = {};
 let knownVoiceUsers = new Set();
 let isMicMuted = false;
 let remoteVideos = {}, videoDecryptChains = {};
@@ -126,22 +125,26 @@ let isVoiceSpeakerMode = false;
 
 const SPEAKING_HANGOVER_MS = 400;
 
-// ===== Voice connection resilience =====
+//Zawory kolejki socketu: TCP dowiezie zakolejkowane bajty tylko spóźnione,
-// Every client broadcasts a tiny encrypted ping so peers can tell "quiet
+//więc progi są hierarchiczne - wideo ustępuje przy 12 KB, audio dopiero przy
-// user" apart from "crashed/half-open socket". Peers silent for too long are
+//14 KB (~270 ms głosu). Przy korku najpierw znika obraz, głos leci dalej.
-// dropped locally; a dead server socket triggers an automatic rejoin.
+const VOICE_WS_VIDEO_QUEUE_LIMIT_BYTES = 12 * 1024;
+const VOICE_WS_AUDIO_QUEUE_LIMIT_BYTES = 14 * 1024;
+
+//Resilience: peer bez pakietów >7 s = lokalnie martwy (media wycięte), socket
+//bez ruchu od ŻYWYCH peerów >15 s = wymuszony reconnect. Pings co 2 s.
 const VOICE_PEER_TIMEOUT_MS = 7000;
 const VOICE_PING_INTERVAL_MS = 2000;
-const VOICE_SERVER_TIMEOUT_MS = 10000;
+const VOICE_SERVER_TIMEOUT_MS = 15000;
-const VOICE_MAX_RECONNECTS = 2;
+const VOICE_MAX_RECONNECTS = 4;
+const VOICE_ROSTER_POLL_MS = 5000;
 let peerLastSeen = {};
 let deadPeers = new Set();
 let lastVoiceServerPacketAt = 0;
-//Czy w bieżącym okresie socketu byli jacyś inni uczestnicy - baza dla
-//resetu baseline'u ciszy w watchdogu (patrz komentarz przy użyciu)
 let voiceHadRemotePeers = false;
 let voicePingTimer = null;
 let voiceRosterRetryTimer = null;
+let voiceRosterPollTimer = null;
 let lastVoiceKeyRequestAt = 0;
 let voiceReconnectAttempts = 0;
 let voiceSocketOpenedAt = 0;
@@ -167,11 +170,21 @@ function cleanupPeerMedia(userId) {
 function sendVoicePing() {
     if (!currentVoiceWs || currentVoiceWs.readyState !== WebSocket.OPEN || !voiceMySenderCryptoKey) return;
     const iv = nextVoiceIv();
-    crypto.subtle.encrypt({name: "AES-GCM", iv}, voiceMySenderCryptoKey, new Uint8Array([Date.now() & 255])).then(cipher => {
+    //Typ 5: prośba o echo. Payload = 8B timestamp (ms, WŁASNY zegar). Odbiorca
+    //odsyła te same bajty jako typ 6, a RTT liczę dopiero na jego powrocie.
+    const tsData = BigInt(Date.now());
+    const tsBuf = new ArrayBuffer(8);
+    new DataView(tsBuf).setBigUint64(0, tsData);
+    crypto.subtle.encrypt({name: "AES-GCM", iv}, voiceMySenderCryptoKey, new Uint8Array(tsBuf)).then(cipher => {
         if (!currentVoiceWs || currentVoiceWs.readyState !== WebSocket.OPEN) return;
         const payload = new Uint8Array(14 + cipher.byteLength);
         payload[0] = 5; payload[1] = 0; payload.set(iv, 2); payload.set(new Uint8Array(cipher), 14);
         currentVoiceWs.send(payload.buffer);
+        voiceOutstandingPings.set(tsData, Date.now());
+        const nowMs = Date.now();
+        for (const [k, sentAt] of voiceOutstandingPings) {
+            if (nowMs - sentAt > 15000) voiceOutstandingPings.delete(k);
+        }
     }).catch(() => {});
 }
 
@@ -217,9 +230,15 @@ const VOICE_FPS_OPTIONS = [15, 24, 30, 60];
 let voiceSettings = loadVoiceSettings();
 
 function loadVoiceSettings() {
-    const defaults = { micId: "", camId: "", speakerId: "", camRes: "1280x720", camFps: 24, screenRes: "1920x1080", screenFps: 30, sensitivity: VOICE_SENSITIVITY_DEFAULT };
+    //Globalny default 480p@24; stare zapisane defaulty (720p/1080p/30fps)
+    //migrują na nowe, świadome wybory użytkownika zostają.
+    const defaults = { micId: "", camId: "", speakerId: "", camRes: "640x480", camFps: 24, screenRes: "640x480", screenFps: 24, sensitivity: VOICE_SENSITIVITY_DEFAULT };
     try {
-        return Object.assign(defaults, JSON.parse(localStorage.getItem(VOICE_SETTINGS_KEY)) || {});
+        let s = Object.assign({}, JSON.parse(localStorage.getItem(VOICE_SETTINGS_KEY)) || {});
+        if (s.camRes === "1280x720") s.camRes = defaults.camRes;
+        if (s.screenRes === "1920x1080") s.screenRes = defaults.screenRes;
+        if (s.screenFps === 30 || s.screenFps === "30") s.screenFps = defaults.screenFps;
+        return Object.assign(defaults, s);
     } catch (e) {
         return defaults;
     }
@@ -274,6 +293,35 @@ async function applyVoiceSpeakerSink() {
 let voiceConnectedAt = null;
 let voiceTimerInterval = null;
 let voicePingMs = null;
+//Kontrola congestionu: bufferedAmount nie widzi backlogu kernela TCP, prawdziwy
+//obraz daje RTT z pingów. Baseline = MINIMUM z okna próbek (odporne na skoki
+//i na peera sztucznie opóźniającego własne echa - stałe kłamstwo podnosi minimum).
+let voiceRttFastMs = 0;
+let voiceRttSamples = [];
+let voiceRttBaseMs = 0;
+let voiceAudioHoldUntil = 0; //histereza zaworu audio
+
+//Echo-ping (typ 5 = prośba, typ 6 = echo): RTT wyłącznie na WŁASNYM zegarze -
+//wysyłam timestamp, peer odbija te same bajty, odejmuję czas nadania. Zegar
+//peera nigdy nie jest używany (może być przekręcony lub spreparowany).
+const voiceOutstandingPings = new Map(); // BigInt(ts) -> czas nadania
+let voiceLastEchoAt = 0;                 // rate-limit odpowiedzi echem
+let voiceLastRttSampleAt = 0;            // kiedy ostatnio spłynął prawdziwy pomiar
+
+function voiceCongestionDelay() {
+    if (!voiceRttFastMs || !voiceRttBaseMs) return 0;
+    return Math.min(5000, Math.max(0, voiceRttFastMs - voiceRttBaseMs));
+}
+
+function voiceRecordRttSample(rtt) {
+    voiceLastRttSampleAt = Date.now();
+    voicePingMs = voicePingMs === null ? rtt : Math.round(voicePingMs * 0.7 + rtt * 0.3);
+    voiceRttFastMs = voiceRttFastMs ? Math.round(voiceRttFastMs * 0.5 + rtt * 0.5) : rtt;
+    voiceRttSamples.push(rtt);
+    if (voiceRttSamples.length > 12) voiceRttSamples.shift();
+    voiceRttBaseMs = Math.min.apply(null, voiceRttSamples);
+    updateVoicePing();
+}
 
 function formatVoiceDuration(ms) {
     let totalSec = Math.floor(ms / 1000);
@@ -630,7 +678,11 @@ setInterval(() => {
     }
     // Half-open socket watchdog + backgrounded-tab audio recovery.
     if (currentVoiceWs && currentVoiceWs.readyState === WebSocket.OPEN) {
-        const others = voiceActiveUsers.filter(u => u !== String(window.id)).length;
+        //deadPeers NIE liczą się do "others": ghost na rosterze (martwy TCP
+        //jeszcze nie zsynchronizowany przez serwer) nigdy nic nie wyśle, więc
+        //zliczany jak żywy wymuszał reconnecty w pętli - migające "Connecting"
+        //w zdrowej rozmowie.
+        const others = voiceActiveUsers.filter(u => u !== String(window.id) && !deadPeers.has(u)).length;
         // Baseline ciszy odświeżany przy przejściu 0->N uczestników: solo w
         // rozmowie NIE MA żadnego ruchu przychodzącego, więc po pojawieniu się
         // nowej osoby lastVoiceServerPacketAt bywa nieświeży z okresu solo.
@@ -693,8 +745,8 @@ class AudioReceiver extends AudioWorkletProcessor {
     //   PREBUF_START  - initial prebuffer before first playback (~43 ms)
     //   PREBUF_RESUME - re-prebuffer after a starvation event (~32 ms)
     //   HARD_LIMIT    - burst ceiling (~128 ms): snap back to target immediately
-    // Backlog slightly above target is drained by a gentle trim (1 sample per
+    // Backlog above target is drained adaptively (proportional to excess, see
-    // quantum) instead of letting latency creep up to the hard limit.
+    // process()) so latency returns to ~43 ms quickly after network bursts.
     constructor(options) {
         super();
         this.users = {};
@@ -732,7 +784,12 @@ class AudioReceiver extends AudioWorkletProcessor {
                 if (available < output.length) { user.buffering = true; }
                 else {
                     if (available > 6144) user.readPtr = (user.writePtr - 2048 + 65536) % 65536;
-                    else if (available > 2049) user.readPtr = (user.readPtr + 1) % 65536;
+                    else if (available > 2049) {
+                        //Adaptacyjny drain: nadmiar nad target odprowadzany proporcjonalnie
+                        //(1 + excess/256 /kwant) - powrót z ~128 ms do ~43 ms w ułamku sekundy
+                        const excess = available - 2048;
+                        user.readPtr = (user.readPtr + 1 + (excess >> 8)) % 65536;
+                    }
                     let vol = this.volumes[id] !== undefined ? this.volumes[id] : 1.0;
                     for (let i = 0; i < output.length; i++) {
                         output[i] += (user.buffer[user.readPtr] / 32768) * vol;
@@ -890,6 +947,38 @@ function bytesToB64u(a) {
     return btoa(bin).replace(/\+/g, "-").replace(/\//g, "_").replace(/=+$/, "");
 }
 
+//Roster pokoju z voice-servera (źródło prawdy) -> synchronizacja kafelków.
+//Steady-poll co VOICE_ROSTER_POLL_MS - bez tego sweep ghosta/leave bez
+//call_update nigdy nie docierał do UI.
+async function fetchVoiceRoster() {
+    if (!currentVoiceWs || currentVoiceWs.readyState !== WebSocket.OPEN || !currentVoiceCallId) return;
+    try {
+        const baseUrl = currentVoiceWs.url.split('/ws')[0];
+        const httpUrl = baseUrl.replace('ws:', 'http:').replace('wss:', 'https:');
+        const t0 = performance.now();
+        const res = await fetch(`${httpUrl}/room/users?room=${encodeURIComponent(currentVoiceCallId)}`);
+        const usersRaw = await res.json();
+        //HTTP tylko jako szacunek gdy brak prawdziwych próbek RTT (solo / stary peer)
+        if (voicePingMs === null || Date.now() - voiceLastRttSampleAt > 25000) {
+            voicePingMs = Math.round(performance.now() - t0);
+            updateVoicePing();
+        }
+        const rawUsers = usersRaw.map(u => String(u));
+        //Serwer jest źródłem prawdy o składzie - nie filtrujemy po
+        //deadPeers (audyt S15: ukryty peer nadal dostaje audio)
+        for (const gone of [...deadPeers]) {
+            if (!rawUsers.includes(gone)) deadPeers.delete(gone);
+        }
+        const users = rawUsers;
+        let changed = (users.length !== voiceActiveUsers.length);
+        for(let u of users) if(!voiceActiveUsers.includes(u)) changed = true;
+        if(changed) {
+            voiceActiveUsers = users;
+            updateVoiceUIUsers();
+        }
+    } catch (e) {}
+}
+
 async function startVoiceSession(serverUrl, ticket) {
     if (!ticket) throw new Error("missing voice ticket");
     // Single-session invariant: a stale socket (e.g. a double-fired join)
@@ -932,32 +1021,7 @@ async function startVoiceSession(serverUrl, ticket) {
                 else btn.classList.remove("voice-btn-danger");
             }
             
-            const fetchUsers = async () => {
+            fetchVoiceRoster();
-                if (!currentVoiceWs || currentVoiceWs.readyState !== WebSocket.OPEN || !currentVoiceCallId) return;
-                try {
-                    const baseUrl = currentVoiceWs.url.split('/ws')[0];
-                    const httpUrl = baseUrl.replace('ws:', 'http:').replace('wss:', 'https:');
-                    const t0 = performance.now();
-                    const res = await fetch(`${httpUrl}/room/users?room=${currentVoiceCallId}`);
-                    const usersRaw = await res.json();
-                    voicePingMs = Math.round(performance.now() - t0);
-                    updateVoicePing();
-                    const rawUsers = usersRaw.map(u => String(u));
-                    //Serwer jest źródłem prawdy o składzie - nie filtrujemy po
-                    //deadPeers (audyt S15: ukryty peer nadal dostaje audio)
-                    for (const gone of [...deadPeers]) {
-                        if (!rawUsers.includes(gone)) deadPeers.delete(gone);
-                    }
-                    const users = rawUsers;
-                    let changed = (users.length !== voiceActiveUsers.length);
-                    for(let u of users) if(!voiceActiveUsers.includes(u)) changed = true;
-                    if(changed) {
-                        voiceActiveUsers = users;
-                        updateVoiceUIUsers();
-                    }
-                } catch (e) {}
-            };
-            fetchUsers();
             //The voice-server roster is joined only AFTER ticket validation (an
             //HTTP roundtrip to the main server), so the immediate fetch can still
             //miss us - retry quickly until our own tile shows up instead of
@@ -971,13 +1035,17 @@ async function startVoiceSession(serverUrl, ticket) {
                     voiceRosterRetryTimer = null;
                     return;
                 }
-                fetchUsers();
+                fetchVoiceRoster();
             }, 400);
 
             voiceReconnectAttempts = 0;
             lastVoiceServerPacketAt = Date.now();
             if (voicePingTimer) clearInterval(voicePingTimer);
             voicePingTimer = setInterval(sendVoicePing, VOICE_PING_INTERVAL_MS);
+            //Steady roster poll - utrzymuje kafelki zgodne z prawdą serwera
+            //(ghost-sweep, leave bez call_update) przez całą rozmowę.
+            if (voiceRosterPollTimer) clearInterval(voiceRosterPollTimer);
+            voiceRosterPollTimer = setInterval(fetchVoiceRoster, VOICE_ROSTER_POLL_MS);
 
             //S16: getUserMedia/worklet/rezump AudioContext potrafią rzucić
             //(brak uprawnień, zniknięte urządzenie) - bez obsługi zostawał
@@ -1033,12 +1101,18 @@ async function startVoiceSession(serverUrl, ticket) {
             }
             
             let vadHangover = 0;
+            const myIdStr = String(window.id);
             senderNode.port.onmessage = async (e) => {
                 if (!currentVoiceWs || currentVoiceWs.readyState !== WebSocket.OPEN) return;
-                // Backpressure valve: when the socket is congested, drop mic
+                //Zawór bajtowy + zawór RTT (kolejka kernela jest niewidoczna dla
-                // frames instead of queueing them - queued audio would arrive
+                //bufferedAmount). Drop >450 ms nadwyżki z histerezą 700 ms - lepiej
-                // minutes late and drown the connection for every stream.
+                //urwać ułamek sekundy niż mówić z sekundami opóźnienia.
-                if (currentVoiceWs.bufferedAmount > 256 * 1024) {
+                if (currentVoiceWs.bufferedAmount > VOICE_WS_AUDIO_QUEUE_LIMIT_BYTES) {
+                    vadHangover = 0;
+                    return;
+                }
+                if (voiceCongestionDelay() > 450) voiceAudioHoldUntil = Date.now() + 700;
+                if (Date.now() < voiceAudioHoldUntil) {
                     vadHangover = 0;
                     return;
                 }
@@ -1053,7 +1127,7 @@ async function startVoiceSession(serverUrl, ticket) {
                 
                 if (db >= sensitivity) vadHangover = 75;
                 let isSpeaking = vadHangover > 0 && !isMicMuted;
-                setTileSpeaking(String(window.id), isSpeaking);
+                setTileSpeaking(myIdStr, isSpeaking);
 
                 if (vadHangover > 0 && !isMicMuted) {
                     vadHangover--;
@@ -1084,7 +1158,44 @@ async function startVoiceSession(serverUrl, ticket) {
             if (deadPeers.has(clientId)) deadPeers.delete(clientId);
 
             const type = view.getUint8(8);
-            if (type === 5) return; // liveness ping - presence already recorded
+            if (type === 5 || type === 6) {
+                //Liveness już zapisany wyżej. Dalej tylko przy zainstalowanym kluczu.
+                if (voiceAesKeyBytes) {
+                    const senderKey = getVoicePeerSenderKey(clientId);
+                    if (senderKey) {
+                        //Offsety ODBIORCZE: serwer dokleja 8B nadawcy, więc iv=[10..22),
+                        //ct=[22..). Format wysyłki (iv=2, ct=14) jest INNY - pomylenie
+                        //psuje każdą deszyfrację cicho (GCM auth fail -> .catch)
+                        const piv = new Uint8Array(arrayBuffer, 10, 12);
+                        const pct = new Uint8Array(arrayBuffer, 22);
+                        crypto.subtle.decrypt({name: "AES-GCM", iv: piv}, senderKey, pct).then(plain => {
+                            if (plain.byteLength !== 8) return; //stary klient: ping bez timestampu
+                            const tsData = new DataView(plain).getBigUint64(0);
+                            if (type === 5) {
+                                //Odbij te same bajty własnym podkluczem i świeżym IV; rate-limit
+                                const nowMs = Date.now();
+                                if (nowMs - voiceLastEchoAt < 150) return;
+                                voiceLastEchoAt = nowMs;
+                                if (!currentVoiceWs || currentVoiceWs.readyState !== WebSocket.OPEN || !voiceMySenderCryptoKey) return;
+                                const riv = nextVoiceIv();
+                                crypto.subtle.encrypt({name: "AES-GCM", iv: riv}, voiceMySenderCryptoKey, plain).then(rcipher => {
+                                    if (!currentVoiceWs || currentVoiceWs.readyState !== WebSocket.OPEN) return;
+                                    const rpayload = new Uint8Array(14 + rcipher.byteLength);
+                                    rpayload[0] = 6; rpayload[1] = 0; rpayload.set(riv, 2); rpayload.set(new Uint8Array(rcipher), 14);
+                                    currentVoiceWs.send(rpayload.buffer);
+                                }).catch(() => {});
+                            } else {
+                                //Typ 6: powrót własnego timestampu - RTT na WŁASNYM zegarze.
+                                if (!voiceOutstandingPings.has(tsData)) return;
+                                voiceOutstandingPings.delete(tsData);
+                                const rtt = Date.now() - Number(tsData);
+                                if (rtt >= 1 && rtt < 15000) voiceRecordRttSample(rtt);
+                            }
+                        }).catch(() => {});
+                    }
+                }
+                return;
+            }
             
             if (!voiceActiveUsers.includes(clientId) && clientId !== String(window.id)) {
                 const baseUrl = currentVoiceWs.url.split('/ws')[0];
@@ -1165,6 +1276,7 @@ async function startVoiceSession(serverUrl, ticket) {
                                     frame.close();
                                 },
                                 error: e => {
+                                    voiceLogError('videodecoder[' + decoderId + ']', e);
                                     if (remoteVideos[decoderId]) {
                                         try { remoteVideos[decoderId].decoder.close(); } catch(err) {}
                                         remoteVideos[decoderId].decoder = null;
@@ -1190,11 +1302,14 @@ async function startVoiceSession(serverUrl, ticket) {
                         try {
                             streamData.decoder.decode(chunk);
                         } catch(err) {
+                            voiceLogError('videodecode[' + decoderId + ']', err);
                             try { streamData.decoder.close(); } catch(e){}
                             streamData.decoder = null;
                             streamData.hasKeyFrame = false;
                         }
-                    } catch (err) {}
+                    } catch (err) {
+                        voiceLogError('vdecrypt[' + decoderId + ']', err);
+                    }
                 });
             } else if (type === 3) {
                 (async () => {
@@ -1332,8 +1447,6 @@ async function handleCallUpdate(data) {
         for (let u in remoteAudio) {
             let pureU = u.replace("_screen", "");
             if (!voiceActiveUsers.includes(pureU) && pureU !== myIdStr) {
-                if (audioDecoders[u]) try { audioDecoders[u].close(); } catch(e){}
-                delete audioDecoders[u];
                 delete remoteAudio[u];
                 const rn = u.endsWith("_screen") ? screenReceiverNode : receiverNode;
                 if(rn) rn.port.postMessage({ action: 'remove', clientId: u });
@@ -2490,7 +2603,6 @@ function toggleVoiceMic() {
 // keeping the UI, mic stream and AudioContext alive for an automatic rejoin.
 function teardownVoiceTransport() {
     if (voicePingTimer) { clearInterval(voicePingTimer); voicePingTimer = null; }
-    if (voiceWsHeartbeat) { clearInterval(voiceWsHeartbeat); voiceWsHeartbeat = null; }
     if (camControl) { try { camControl.stop(); } catch (e) {} camControl = null; }
     if (screenControl) { try { screenControl.stop(); } catch (e) {} screenControl = null; }
     if (currentVoiceWs) {
@@ -2507,6 +2619,7 @@ function teardownVoiceTransport() {
     peerLastSeen = {};
     deadPeers.clear();
     if (voiceRosterRetryTimer) { clearInterval(voiceRosterRetryTimer); voiceRosterRetryTimer = null; }
+    if (voiceRosterPollTimer) { clearInterval(voiceRosterPollTimer); voiceRosterPollTimer = null; }
     lastVoiceKeyRequestAt = 0;
     //Stale participants from the previous attempt must not linger in the UI
     voiceActiveUsers = [];
@@ -2516,6 +2629,12 @@ function teardownVoiceTransport() {
     setVoiceRootKey(null);
     voiceSendEpoch = null;
     voiceSendCounter = 0n;
+    voiceRttFastMs = 0;
+    voiceRttSamples = [];
+    voiceRttBaseMs = 0;
+    voiceAudioHoldUntil = 0;
+    voiceOutstandingPings.clear();
+    voiceLastRttSampleAt = 0;
     lastVoiceServerPacketAt = Date.now();
 }
 
@@ -2532,6 +2651,10 @@ function scheduleVoiceReconnect() {
     voiceReconnectAttempts++;
     setVoiceStatus("connecting");
     teardownVoiceTransport();
+    //Zmiana sieci = stary RTT dotyczy martwej ścieżki; bez resetu wskaźnik
+    //wisiałby zamrożony, a fallback HTTP nie miał prawa go nadpisać
+    voicePingMs = null;
+    updateVoicePing();
     setTimeout(async () => {
         if (!currentVoiceCallId || !currentVoiceCallType) return;
         try {
@@ -2543,7 +2666,7 @@ function scheduleVoiceReconnect() {
             console.warn("Voice reconnect failed", err);
             scheduleVoiceReconnect();
         }
-    }, 800);
+    }, Math.min(4000, 800 * voiceReconnectAttempts));
 }
 
 function endVoiceSession() {
@@ -2583,24 +2706,27 @@ function endVoiceSession() {
         micStream = null;
     }
     if (micSourceNode) { micSourceNode.disconnect(); micSourceNode = null; }
-    if (audioEncoder) { try { audioEncoder.close(); } catch(e){} audioEncoder = null; }
+
-    
-    if (voiceWsHeartbeat) { clearInterval(voiceWsHeartbeat); voiceWsHeartbeat = null; }
     if (voicePingTimer) { clearInterval(voicePingTimer); voicePingTimer = null; }
     if (voiceRosterRetryTimer) { clearInterval(voiceRosterRetryTimer); voiceRosterRetryTimer = null; }
+    if (voiceRosterPollTimer) { clearInterval(voiceRosterPollTimer); voiceRosterPollTimer = null; }
     peerLastSeen = {};
     deadPeers.clear();
     lastVoiceKeyRequestAt = 0;
+    voiceReconnectAttempts = 0;
     voiceActiveUsers = [];
     voiceHadRemotePeers = false;
-    
+    voiceRttFastMs = 0;
-    for (let id in audioDecoders) {
+    voiceRttSamples = [];
-        try { audioDecoders[id].close(); } catch(e){}
+    voiceRttBaseMs = 0;
-    }
+    voiceAudioHoldUntil = 0;
+    voiceOutstandingPings.clear();
+    voiceLastEchoAt = 0;
+    voiceLastRttSampleAt = 0;
+
     for (let id in audioDecryptChains) {
         if(audioDecryptChains[id]) audioDecryptChains[id].clear();
     }
-    audioDecoders = {};
     audioDecryptChains = {};
     remoteAudio = {};
     
@@ -2672,8 +2798,11 @@ async function startHardwareVideo(btnEl, streamPromise, typeLabel, typeId) {
                 screenSenderNode = new AudioWorkletNode(audioContext, 'audio-sender');
                 screenSenderNode.port.onmessage = async (e) => {
                     if (currentVoiceWs?.readyState !== WebSocket.OPEN || !voiceMySenderCryptoKey) return;
-                    // Same backpressure valve as the mic sender.
+                    // Ten sam zawór co mikrofon (limit bajtów + korek RTT) - audio
-                    if (currentVoiceWs.bufferedAmount > 256 * 1024) return;
+                    // ma pierwszeństwo przed wideo na wspólnym sockecie.
+                    if (currentVoiceWs.bufferedAmount > VOICE_WS_AUDIO_QUEUE_LIMIT_BYTES) return;
+                    if (voiceCongestionDelay() > 450) voiceAudioHoldUntil = Date.now() + 700;
+                    if (Date.now() < voiceAudioHoldUntil) return;
                     const iv = nextVoiceIv();
                     const cipher = await crypto.subtle.encrypt({name: "AES-GCM", iv}, voiceMySenderCryptoKey, e.data.audioData);
                     const payload = new Uint8Array(14 + cipher.byteLength);
@@ -2703,7 +2832,24 @@ async function startHardwareVideo(btnEl, streamPromise, typeLabel, typeId) {
             return { width, height, fps, bitrate };
         };
 
-        let currentConfig = getCurrentConfig(), isRunning = true, frameCount = 0;
+        let currentConfig = getCurrentConfig(), isRunning = true, lastKeyframeAt = 0;
+        //AIMD + slow-start: start na 40% celu (pełną prędkością waliłby w nieznany
+        //uplink, a bufory TCP w kernelu to MB-y = sekundy kolejki zanim cokolwiek
+        //zmierzymy). Sygnał korka: skipy (zawór 12 KB) lub nadwyżka RTT.
+        let brScale = 0.4, encTries = 0, encSkips = 0, brWindowAt = performance.now(), lastBrReconfAt = 0;
+        //Korek trwa mimo podłogi bitrate'u → FPS spada do 1/2, 1/3
+        let fpsDiv = 1, lowStreak = 0, cleanStreak = 0, tickCount = 0;
+        let activeBitrate = Math.round(currentConfig.bitrate * brScale);
+        const applyEncoderBitrate = (force = false) => {
+            const target = Math.max(120000, Math.round(currentConfig.bitrate * brScale));
+            if (!force && Math.abs(target - activeBitrate) / activeBitrate < 0.08) return;
+            if (!force && performance.now() - lastBrReconfAt < 800) return;
+            activeBitrate = target;
+            lastBrReconfAt = performance.now();
+            try {
+                encoder.configure({codec: 'av01.0.04M.08', width: currentConfig.width, height: currentConfig.height, framerate: currentConfig.fps, bitrate: activeBitrate, latencyMode: 'realtime'});
+            } catch (e) {}
+        };
         const encoder = new VideoEncoder({
             output: async (chunk) => {
                 if (currentVoiceWs?.readyState !== WebSocket.OPEN || !voiceMySenderCryptoKey) return;
@@ -2713,9 +2859,9 @@ async function startHardwareVideo(btnEl, streamPromise, typeLabel, typeId) {
                 const payload = new Uint8Array(14 + cipher.byteLength);
                 payload[0] = typeId; payload[1] = chunk.type === 'key' ? 1 : 0; payload.set(iv, 2); payload.set(new Uint8Array(cipher), 14);
                 currentVoiceWs.send(payload.buffer);
-            }, error: (e) => { stopFunction(); }
+            }, error: (e) => { voiceLogError('videoencoder', e); stopFunction(); }
         });
-        encoder.configure({codec: 'av01.0.04M.08', width: currentConfig.width, height: currentConfig.height, framerate: currentConfig.fps, bitrate: currentConfig.bitrate, latencyMode: 'realtime'});
+        encoder.configure({codec: 'av01.0.04M.08', width: currentConfig.width, height: currentConfig.height, framerate: currentConfig.fps, bitrate: activeBitrate, latencyMode: 'realtime'});
 
             // Transient capture-mode wobbles must not thrash the encoder -
             // commit only after ~300 ms stable. Camera size/fps changes arrive
@@ -2733,23 +2879,65 @@ async function startHardwareVideo(btnEl, streamPromise, typeLabel, typeId) {
                         pendingCfg = newConfig;
                         pendingAt = performance.now();
                     } else if (performance.now() - pendingAt > 300) {
-                        currentConfig = newConfig; forceKeyframe = true; frameCount = 0; pendingCfg = null;
+                        //Zmiana geometrii źródła - uczony scale łącza zostaje,
-                        encoder.configure({codec: 'av01.0.04M.08', width: currentConfig.width, height: currentConfig.height, framerate: currentConfig.fps, bitrate: currentConfig.bitrate, latencyMode: 'realtime'});
+                        //resetujemy tylko redukcję FPS
+                        currentConfig = newConfig; forceKeyframe = true; lastKeyframeAt = 0; pendingCfg = null;
+                        activeBitrate = Math.round(currentConfig.bitrate * brScale);
+                        fpsDiv = 1; lowStreak = 0; cleanStreak = 0;
+                        applyEncoderBitrate(true);
                     }
                 } else if (pendingCfg) {
                     pendingCfg = null; // wiggle reverted on its own - ignore
                 }
 
-                if (encoder.state === 'configured' && encoder.encodeQueueSize <= 2 && currentVoiceWs.bufferedAmount < 1024 * 256 && videoEl.readyState >= 2) {
+                //Zawór kolejki socketu (12 KB) ustępuje DALEJ przed progiem audio
-                let frame = null;
+                //(14 KB) - przy korku najpierw znika obraz, głos płynie dalej.
-                try {
+                tickCount++;
-                    frame = new VideoFrame(videoEl, {timestamp: Math.trunc(performance.now() * 1000)});
+                const policySkip = fpsDiv > 1 && (tickCount % fpsDiv !== 0);
-                    frameCount++;
+                let skipped = false;
-                    encoder.encode(frame, {keyFrame: forceKeyframe || (frameCount % (currentConfig.fps * 2)) === 0});
+                if (policySkip) {
-                } catch (err) { } finally { if (frame) frame.close(); }
+                    //celowe tempo (fpsDiv) - nie liczy się do skipRatio
-            }
+                } else if (encoder.state === 'configured' && encoder.encodeQueueSize <= 2 &&
-            setTimeout(encodeLoop, 1000 / currentConfig.fps);
+                    currentVoiceWs.readyState === WebSocket.OPEN &&
-        };
+                    currentVoiceWs.bufferedAmount < VOICE_WS_VIDEO_QUEUE_LIMIT_BYTES && videoEl.readyState >= 2) {
+                    let frame = null;
+                    try {
+                        const nowMs = performance.now();
+                        frame = new VideoFrame(videoEl, {timestamp: Math.trunc(nowMs * 1000)});
+                        //Keyframe wg zegara co ~1 s (nie licznika klatek - zawór je pomija)
+                        const wantKeyframe = forceKeyframe || (nowMs - lastKeyframeAt >= 1000);
+                        encoder.encode(frame, {keyFrame: wantKeyframe});
+                        if (wantKeyframe) lastKeyframeAt = nowMs;
+                    } catch (err) { } finally { if (frame) frame.close(); }
+                } else if (encoder.state === 'configured') {
+                    skipped = true;
+                }
+
+                encTries++;
+                if (skipped) encSkips++;
+                const winNow = performance.now();
+                if (winNow - brWindowAt >= 800) {
+                    const skipRatio = encTries > 0 ? encSkips / encTries : 0;
+                    const congestionMs = voiceCongestionDelay();
+                    if (congestionMs > 450 || skipRatio > 0.3 || congestionMs > 250) {
+                        brScale = Math.max(0.15, brScale * (congestionMs > 450 ? 0.6 : 0.75));
+                        applyEncoderBitrate();
+                        if (brScale <= 0.17) lowStreak++; else lowStreak = 0;
+                        if (lowStreak >= 4 && fpsDiv < 3) { fpsDiv++; lowStreak = 0; }
+                        cleanStreak = 0;
+                    } else if (skipRatio === 0 && congestionMs < 80) {
+                        cleanStreak++;
+                        lowStreak = 0;
+                        if (cleanStreak >= 5) {
+                            if (fpsDiv > 1) { fpsDiv--; }
+                            else if (brScale < 1) { brScale = Math.min(1, brScale * 1.2); applyEncoderBitrate(); }
+                            cleanStreak = 0;
+                        }
+                    }
+                    encTries = 0; encSkips = 0; brWindowAt = winNow;
+                }
+                setTimeout(encodeLoop, 1000 / currentConfig.fps);
+            };
         setTimeout(encodeLoop, 500);
 
         videoTrack.onended = () => { if(isRunning) stopFunction(); };