@voipsdk/react-native-voipcloud 1.0.1 → 1.0.3

package/CHANGELOG.md

@@ -5,6 +5,54 @@ All notable changes to `@voipsdk/react-native-voipcloud` are documented here.
 The format is based on [Keep a Changelog](https://keepachangelog.com/en/1.0.0/),
 and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0.html).
 
+## [1.0.3] - 2026-07-17
+
+Folds in the NAT / media / call-routing fixes that were living as app-side
+`patch-package` patches, so a fresh install of the SDK behaves correctly
+without any patches.
+
+### Added
+
+- `AccountConfig.proxy` — outbound proxy URI (e.g. `sip:proxy.example.com:5060;lr`).
+  Routes every SIP request, including in-dialog re-INVITE / BYE, through the proxy
+  so in-dialog SIP survives NAT (a NAT'd client no longer sends its BYE / unhold
+  re-INVITE straight to the peer's private address). Wired on both iOS and Android.
+- `AccountConfig.srtp` — `'disabled' | 'optional' | 'mandatory'` media policy,
+  defaulting to `'disabled'` so a plain-UDP trunk never offers `a=crypto`
+  (an optional-SRTP offer breaks media on some PBX / rtpengine setups → one-way audio).
+
+### Changed / Fixed
+
+- **iOS:** transports are now pinned per account (REGISTER + INVITE + in-dialog
+  re-INVITE / BYE share one socket/port for a stable NAT binding across hold/unhold);
+  media-only STUN (`mediaStunUse = RETRY_ON_FAILURE`, SIP STUN disabled) so the SDP
+  offer carries a routable address; endpoint codec allow-list pinned to PCMU/PCMA
+  (+G722 fallback) to stop the proxy rewriting the `m=` line; session timers disabled
+  on-device; hold / unhold / makeCall forced audio-only (`audioCount=1, videoCount=0`).
+- **iOS podspec:** `HEADER_SEARCH_PATHS` points at each slice's `Headers` root instead
+  of a recursive `/**` glob, so `pj/compat/` no longer shadows the C stdlib.
+- **Android:** accounts pinned to a single transport (`transportIdForRegistrar`);
+  `contactUseSrcPort` kept in the rewritten Contact; RFC 4028 session timers as a
+  safety net for a lost in-dialog BYE; unhold re-INVITE forces every audio stream back
+  to `sendrecv` so the PBX stops streaming MOH after resume.
+- **Android:** native call-end hook (`VoipCallEndNotifier.onCallEnded`) clears the
+  incoming notification / ringtone even when JS is gone (headless FCM); `sipCallId`
+  now forwarded to the native incoming-call dispatch.
+- **Android:** `minSdkVersion` lowered from 31 to 24.
+
+## [1.1.0] - 2026-07-06
+
+### Added
+
+- **iOS support — full feature parity with Android.** All 14 methods (`init`, `shutdown`, `register`, `unregister`, `makeCall`, `answer`, `hangup`, `hangupAllCalls`, `hold`, `unhold`, `mute`, `sendDtmf`, `transferCall`, `reconnectCallAudio`) and all 11 events are now implemented natively on iOS. No method rejects with `VC_NOT_IMPLEMENTED` anymore.
+- iOS native layer: `VoipCloudEngine` (Objective-C++ PJSUA2 wrapper — multi-account/multi-call registry, NAT survival, SRTP optional, audio bridging, `AVAudioSession` management) driving an `RCTEventEmitter` bridge (`VoipCloud.mm`). Same event names, payloads and `VC_*` error codes as Android.
+- Ships the prebuilt, self-contained `ios/VoipCloud.xcframework` (PJSIP + OpenSSL 3.0.16). Podspec vendors it via `vendored_frameworks`, links `libc++`/`libresolv` + the required system frameworks, and sets `gnu++17`.
+- iOS simulator now covers both `arm64` (Apple Silicon) and `x86_64` — build scripts produce a fat simulator slice; device stays `arm64`.
+
+### Changed
+
+- Build scripts (`build-openssl.sh`, `build-pjproject.sh`, `build-xcframework.sh`) parametrized by slice (`device` / `simulator-x86_64` / `simulator-arm64`); `build-xcframework.sh` lipos the two simulator arches and syncs the result into `package/ios/`.
+
 ## [1.0.1] - 2026-07-06
 
 ### Added

package/LICENSE

@@ -0,0 +1,21 @@
+MIT License
+
+Copyright (c) 2026 VoipCloud
+
+Permission is hereby granted, free of charge, to any person obtaining a copy
+of this software and associated documentation files (the "Software"), to deal
+in the Software without restriction, including without limitation the rights
+to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+copies of the Software, and to permit persons to whom the Software is
+furnished to do so, subject to the following conditions:
+
+The above copyright notice and this permission notice shall be included in all
+copies or substantial portions of the Software.
+
+THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+SOFTWARE.

package/README.md

@@ -1,7 +1,9 @@
 # @voipsdk/react-native-voipcloud
 
-> Public PJSIP VoIP SDK for React Native — Android only (RN 0.70+, Android 12 / API 31+).  
-> Package: `@voipsdk/react-native-voipcloud@1.0.1`
+> Public VoIP SDK for React Native — Android **and** iOS (RN 0.70+).  
+> Package: `@voipsdk/react-native-voipcloud@1.0.3`
+
+Android and iOS expose the **same** JS API, events and `VC_*` error codes.
 
 ---
 
@@ -10,9 +12,8 @@
 | | |
 |---|---|
 | React Native | ≥ 0.70 |
-| Android | API 31+ (Android 12+) |
-| ABIs | `arm64-v8a`, `armeabi-v7a`, `x86_64` (emulator) |
-| iOS | Stub only — rejects with `VC_NOT_IMPLEMENTED` |
+| Android | API 31+ (Android 12+) — ABIs `arm64-v8a`, `armeabi-v7a`, `x86_64` |
+| iOS | 13.0+ — device `arm64`, simulator `arm64` + `x86_64` |
 
 ---
 
@@ -24,6 +25,8 @@ npm install @voipsdk/react-native-voipcloud
 yarn add @voipsdk/react-native-voipcloud
 ```
 
+### Android
+
 Rebuild the native app after install:
 
 ```bash
@@ -36,6 +39,24 @@ React Native 0.60+ autolinks `VoipCloudPackage`. For older RN, add manually in `
 packages.add(new com.voipcloud.sdk.rn.VoipCloudPackage());
 ```
 
+### iOS
+
+Autolinking installs the pod (which vendors the prebuilt `VoipCloud.xcframework` — native SIP engine, self-contained). Just run:
+
+```bash
+cd ios && pod install
+```
+
+Then add a microphone usage string to your app **`Info.plist`** (required or the OS kills the app on first mic access):
+
+```xml
+<key>NSMicrophoneUsageDescription</key>
+<string>Microphone access is needed for calls.</string>
+```
+
+For calls to keep running when the app is backgrounded, enable the **Audio** background mode
+(`UIBackgroundModes` → `audio`) in your app target. VoIP push / CallKit is not bundled — see the roadmap.
+
 ### Android permissions
 
 Declare in **your** app `AndroidManifest.xml`:
@@ -58,7 +79,7 @@ Request at runtime: `RECORD_AUDIO`, `POST_NOTIFICATIONS` (Android 13+), `BLUETOO
 ```ts
 import { VoipCloud } from '@voipsdk/react-native-voipcloud';
 
-// 1. Start PJSIP endpoint (required before any SIP call).
+// 1. Start the VoIP endpoint (required before any SIP call).
 await VoipCloud.init({ userAgent: 'my-app/1.0' });
 
 // 2. Register a SIP account → returns numeric accountId.
@@ -121,7 +142,7 @@ export function useVoipEvents() {
 
 | Method | Description |
 |--------|-------------|
-| `init(config?)` | Create and start PJSIP endpoint |
+| `init(config?)` | Create and start the VoIP endpoint |
 | `shutdown()` | Hang up all, destroy endpoint |
 | `register(account)` | Register SIP account → `accountId` |
 | `unregister(accountId)` | Unregister account |
@@ -142,7 +163,7 @@ export function useVoipEvents() {
 |-------|------|-------------|
 | `userAgent` | `string` | SIP User-Agent header |
 | `stunServer` | `string` | STUN server URI |
-| `logLevel` | `number` | PJSIP log level (0–6) |
+| `logLevel` | `number` | SIP log level (0–6) |
 
 ### `AccountConfig`
 
@@ -175,7 +196,7 @@ export function useVoipEvents() {
 
 ## Error handling
 
-Every method returns a `Promise`. Failures reject with `error.code` (string). The bridge **never crashes** the host app for API misuse or recoverable PJSIP errors.
+Every method returns a `Promise`. Failures reject with `error.code` (string). The bridge **never crashes** the host app for API misuse or recoverable SIP errors.
 
 ```ts
 try {
@@ -198,12 +219,14 @@ try {
 | `VC_MAKE_CALL` / `VC_ANSWER` / `VC_HANGUP` | Call operation error |
 | `VC_NO_CALL` | Unknown `callId` |
 | `VC_HOLD` / `VC_UNHOLD` / `VC_MUTE` / `VC_DTMF` / `VC_TRANSFER` | In-call control error |
+| `VC_AUDIO` | Audio bridge / `reconnectCallAudio` error |
+| `VC_HANGUP_ALL` | `hangupAllCalls` error |
 
 ---
 
-## Native artifacts (Android)
+## Native artifacts
 
-Shipped inside `android/src/main/libs/voipcloud-release.aar`:
+**Android** — inside `android/src/main/libs/voipcloud-release.aar`:
 
 | Artifact | Name |
 |----------|------|
@@ -212,7 +235,16 @@ Shipped inside `android/src/main/libs/voipcloud-release.aar`:
 | RN bridge module | `VoipsdkReactNativeVoipcloud` (JS: `VoipCloud`) |
 | Java SDK | `com.voipcloud.sdk.*` |
 
-This package does **not** use upstream `libpjsua2.so` / `pjsua2JNI`.
+**iOS** — under `ios/`:
+
+| Artifact | Name |
+|----------|------|
+| Prebuilt framework | `VoipCloud.xcframework` — native SIP engine, static, self-contained |
+| Native SIP wrapper | `VoipCloudEngine` (Objective-C++) |
+| RN bridge module | `VoipsdkReactNativeVoipcloud` (`RCTEventEmitter`, JS: `VoipCloud`) |
+
+Both platforms ship their native engine prebuilt and self-contained — no extra
+native toolchain or SIP stack is required in the host app.
 
 ---
 
@@ -224,14 +256,27 @@ Run `cd android && ./gradlew clean assembleDebug`. Confirm `@voipsdk/react-nativ
 **`VC_NATIVE_LOAD`**  
 Ensure your device ABI is supported (`arm64-v8a` on most phones). For release builds, R8 applies ProGuard rules from the AAR automatically.
 
+**Android build: duplicate `libc++_shared.so`**  
+If another native library also bundles the C++ runtime, add to `android/app/build.gradle`:
+
+```gradle
+android { packagingOptions { pickFirst '**/libc++_shared.so' } }
+```
+
 **Registration fails (`VC_REGISTER`)**  
-Check `registrar`, credentials, and network. TLS requires a valid server certificate or appropriate PJSIP TLS settings on the PBX side.
+Check `registrar`, credentials, and network. TLS requires a valid server certificate or appropriate TLS settings on the PBX side.
 
 **No audio**  
-Grant `RECORD_AUDIO` at runtime before placing a call.
+Android: grant `RECORD_AUDIO` at runtime before placing a call.
+iOS: add `NSMicrophoneUsageDescription` to `Info.plist` and accept the mic permission prompt.
+
+**iOS: `pod install` can't find headers / link errors**  
+Ensure `ios/VoipCloud.xcframework` exists in the installed package (it ships in the npm tarball).
+Re-run `pod install`; the podspec adds the framework's per-slice `Headers/` to the search path
+and links `libc++` + the required system frameworks.
 
 ---
 
 ## License
 
-UNLICENSED — see package metadata. Contact the author for commercial terms.
+[MIT](./LICENSE) © VoipCloud. Free for personal **and** commercial use.

package/android/build.gradle

@@ -8,14 +8,20 @@
 //   * com.voipcloud.sdk.rn.* React Native bridge (this module)
 
 buildscript {
-    repositories {
-        google()
-        mavenCentral()
-    }
-    dependencies {
-        // Pinned to the AGP version known to compile with React Native
-        // 0.74 + AS Iguana.  Bump in lockstep with peerDependencies.
-        classpath 'com.android.tools.build:gradle:8.2.1'
+    // Only needed when this library is opened as a STANDALONE Gradle project.
+    // When an app consumes it via autolinking, the app already provides the
+    // Android Gradle plugin on the classpath — injecting a second AGP version
+    // here would clash with the app's AGP. Guard so the consumer build is
+    // untouched.
+    if (project == rootProject) {
+        repositories {
+            google()
+            mavenCentral()
+        }
+        dependencies {
+            // AGP version known to compile with React Native 0.74.
+            classpath 'com.android.tools.build:gradle:8.2.1'
+        }
     }
 }
 
@@ -30,7 +36,7 @@ android {
     compileSdkVersion safeExtGet('compileSdkVersion', 34)
 
     defaultConfig {
-        minSdkVersion safeExtGet('minSdkVersion', 31)
+        minSdkVersion safeExtGet('minSdkVersion', 24)
         targetSdkVersion safeExtGet('targetSdkVersion', 34)
 
         ndk {

package/android/src/main/java/com/voipcloud/sdk/rn/VoipCloudModule.java

@@ -10,17 +10,26 @@ import com.facebook.react.bridge.ReactMethod;
 import com.facebook.react.bridge.ReadableMap;
 import com.facebook.react.bridge.WritableMap;
 
+import android.util.Log;
+
 import com.voipcloud.sdk.AccountConfig;
 import com.voipcloud.sdk.AuthCredInfo;
+import com.voipcloud.sdk.CallInfo;
+import com.voipcloud.sdk.CallMediaInfo;
 import com.voipcloud.sdk.CallOpParam;
+import com.voipcloud.sdk.CallSetting;
 import com.voipcloud.sdk.CallSendDtmfParam;
 import com.voipcloud.sdk.Endpoint;
 import com.voipcloud.sdk.EpConfig;
+import com.voipcloud.sdk.IntVector;
 import com.voipcloud.sdk.TransportConfig;
+import com.voipcloud.sdk.pjmedia_dir;
+import com.voipcloud.sdk.pjmedia_type;
 import com.voipcloud.sdk.pjsip_status_code;
 import com.voipcloud.sdk.pjsip_transport_type_e;
 import com.voipcloud.sdk.pjsua_call_flag;
 import com.voipcloud.sdk.pjsua_destroy_flag;
+import com.voipcloud.sdk.pjsua_sip_timer_use;
 import com.voipcloud.sdk.AudDevManager;
 
 import com.voipcloud.sdk.rn.internal.BridgeGuard;
@@ -44,6 +53,14 @@ public class VoipCloudModule extends ReactContextBaseJavaModule {
 
     private Endpoint endpoint;
     private boolean inited = false;
+    /**
+     * Transport IDs captured at init() so accounts can be PINNED to one
+     * transport (see register()). Both UDP and TCP transports are created; an
+     * account bound to neither lets pjsip auto-pick a transport per request,
+     * which can switch mid-dialog and break in-dialog routing.
+     */
+    private int udpTransportId = -1;
+    private int tcpTransportId = -1;
 
     public VoipCloudModule(ReactApplicationContext ctx) {
         super(ctx);
@@ -95,6 +112,8 @@ public class VoipCloudModule extends ReactContextBaseJavaModule {
         } catch (Throwable ignored) {}
         endpoint = null;
         inited = false;
+        udpTransportId = -1;
+        tcpTransportId = -1;
     }
 
     @ReactMethod
@@ -143,10 +162,15 @@ public class VoipCloudModule extends ReactContextBaseJavaModule {
 
             ep.libInit(epCfg);
 
+            // Capture the transport IDs so register() can pin each account to
+            // exactly ONE transport. Without a binding pjsip is free to choose
+            // a different transport for a mid-dialog re-INVITE/UPDATE than the
+            // one that established the dialog — the source ip:port then changes
+            // and in-dialog routing breaks (see register()).
             TransportConfig udp = new TransportConfig();
-            ep.transportCreate(pjsip_transport_type_e.PJSIP_TRANSPORT_UDP, udp);
+            udpTransportId = ep.transportCreate(pjsip_transport_type_e.PJSIP_TRANSPORT_UDP, udp);
             TransportConfig tcp = new TransportConfig();
-            ep.transportCreate(pjsip_transport_type_e.PJSIP_TRANSPORT_TCP, tcp);
+            tcpTransportId = ep.transportCreate(pjsip_transport_type_e.PJSIP_TRANSPORT_TCP, tcp);
 
             ep.libStart();
             endpoint = ep;
@@ -205,11 +229,50 @@ public class VoipCloudModule extends ReactContextBaseJavaModule {
             acfg.getNatConfig().setContactRewriteUse(1);
             acfg.getNatConfig().setViaRewriteUse(1);
             acfg.getNatConfig().setUdpKaIntervalSec(20);
+            // Keep the learned NAT source PORT in the rewritten Contact (not just
+            // the source IP). The remote target our peer routes in-dialog requests
+            // to must match the exact ip:port our transport sends from, or replies
+            // to our re-INVITE stall and the peer's BYE misses the NAT pinhole.
+            acfg.getNatConfig().setContactUseSrcPort(1);
             // Short re-REGISTER interval keeps the binding fresh behind NAT.
             acfg.getRegConfig().setTimeoutSec(120);
             acfg.getRegConfig().setFirstRetryIntervalSec(5);
             acfg.getRegConfig().setRetryIntervalSec(30);
 
+            // RFC 4028 session timers — safety net for a LOST in-dialog BYE.
+            // Behind NAT (kamailio+asterisk, UDP) the remote party's BYE is an
+            // inbound in-dialog request; if it never reaches the device the
+            // call would otherwise stay "up" forever (remote hangs up, our
+            // side never disconnects). With session timers the dialog must be
+            // refreshed (re-INVITE/UPDATE) every session-expires seconds — when
+            // the peer is gone the refresh gets no answer (or none arrives) and
+            // PJSIP tears the call down locally, firing onCallState=DISCONNECTED
+            // so the JS/UI ends the call. ALWAYS advertises "timer" in Supported
+            // on every INVITE (graceful: peers without RFC 4028 just ignore it;
+            // Asterisk negotiates it). Min-SE 90 is the RFC floor; 120s expiry
+            // bounds the worst-case stuck-call window (~60s typical — refresh at
+            // half). This does NOT fix the root cause (see kamailio record-route
+            // NAT config); it guarantees the call cannot hang indefinitely.
+            acfg.getCallConfig().setTimerUse(pjsua_sip_timer_use.PJSUA_SIP_TIMER_ALWAYS);
+            acfg.getCallConfig().setTimerMinSESec(90);
+            acfg.getCallConfig().setTimerSessExpiresSec(120);
+
+            // PIN the account to a single transport so EVERY request in the call
+            // — the initial INVITE, the hold/unhold re-INVITE and any session-timer
+            // UPDATE, and the BYE — originates from the SAME local socket/port.
+            // Both UDP and TCP transports exist; if the account is bound to neither
+            // (transportId = -1 / auto), pjsip may pick a different transport for a
+            // mid-dialog re-INVITE than the one that set up the dialog (observed:
+            // hold over TCP, unhold over UDP). The source ip:port then changes,
+            // kamailio/asterisk can no longer match the request to the dialog, the
+            // re-INVITE hangs at 100 Trying, and the call effectively forks into a
+            // second dialog — so hold/unhold and hangup act on the wrong one.
+            // Binding to one transport keeps the SIP 5-tuple stable for the whole call.
+            int boundTransport = transportIdForRegistrar(registrar);
+            if (boundTransport >= 0) {
+                acfg.getSipConfig().setTransportId(boundTransport);
+            }
+
             AuthCredInfo cred = new AuthCredInfo("digest", "*", username, 0, password);
             acfg.getSipConfig().getAuthCreds().add(cred);
 
@@ -374,7 +437,38 @@ public class VoipCloudModule extends ReactContextBaseJavaModule {
             ManagedCall call = requireCall(callId, promise);
             if (call == null) return;
             CallOpParam prm = new CallOpParam(true);
-            prm.getOpt().setFlag(pjsua_call_flag.PJSUA_CALL_UNHOLD);
+            // PJSUA_CALL_UNHOLD alone regenerates the offer, but on this build
+            // the audio m-line can stay a=sendonly, so the remote PBX never sees
+            // the resume and keeps streaming MOH to the caller. Explicitly force
+            // every audio stream back to sendrecv (PJMEDIA_DIR_ENCODING_DECODING)
+            // via PJSUA_CALL_SET_MEDIA_DIR so the re-INVITE offer is unambiguous.
+            // Build the CallSetting explicitly and attach it with setOpt() —
+            // getOpt() can return a detached copy whose mutations are dropped.
+            CallSetting opt = new CallSetting(true);
+            long flags = pjsua_call_flag.PJSUA_CALL_UNHOLD;
+            int mediaCount = 0;
+            try {
+                IntVector dirs = new IntVector();
+                CallInfo info = call.getInfo();
+                for (int i = 0; i < info.getMedia().size(); i++) {
+                    CallMediaInfo mi = info.getMedia().get(i);
+                    dirs.add(mi.getType() == pjmedia_type.PJMEDIA_TYPE_AUDIO
+                            ? pjmedia_dir.PJMEDIA_DIR_ENCODING_DECODING
+                            : pjmedia_dir.PJMEDIA_DIR_NONE);
+                }
+                if (dirs.isEmpty()) {
+                    dirs.add(pjmedia_dir.PJMEDIA_DIR_ENCODING_DECODING);
+                }
+                mediaCount = dirs.size();
+                opt.setMediaDir(dirs);
+                flags |= pjsua_call_flag.PJSUA_CALL_SET_MEDIA_DIR;
+            } catch (Throwable ignored) {
+                // Fall back to a plain UNHOLD re-INVITE if media enumeration fails.
+            }
+            opt.setFlag(flags);
+            prm.setOpt(opt);
+            Log.i("VoipCloudUnhold", "unhold callId=" + callId
+                    + " flags=" + flags + " forcedSendrecvOn=" + mediaCount + " media");
             call.reinvite(prm);
             // See hold(): audio is rewired by onCallMediaState when the
             // unhold re-INVITE completes and media becomes ACTIVE again.
@@ -513,6 +607,22 @@ public class VoipCloudModule extends ReactContextBaseJavaModule {
         }
     }
 
+    /**
+     * Choose the transport to bind an account to, based on the {@code transport=}
+     * param in its registrar URI (built by mapAccountToVoipCloud on the JS side).
+     * Defaults to UDP — the app's standard transport, whose 20s keep-alive holds
+     * the NAT pinhole open for the whole call. TCP/TLS accounts bind to the TCP
+     * transport. Returns -1 (auto) only if the matching transport was never
+     * created, leaving pjsip's default behaviour untouched as a safe fallback.
+     */
+    private int transportIdForRegistrar(String registrar) {
+        String t = registrar == null ? "" : registrar.toLowerCase();
+        if (t.contains("transport=tcp") || t.contains("transport=tls")) {
+            return tcpTransportId;
+        }
+        return udpTransportId;
+    }
+
     private ManagedCall requireCall(int callId, Promise promise) {
         ManagedCall call = calls.get(callId);
         if (call == null) {

package/android/src/main/java/com/voipcloud/sdk/rn/internal/ManagedAccount.java

@@ -60,19 +60,23 @@ public final class ManagedAccount extends Account {
 
         String from = "";
         String to = "";
+        // SIP Call-ID header — stable across systems (PBX CDR, CRM sync, and the
+        // FCM push `callId`), unlike the numeric PJSUA call slot which is
+        // recycled. Captured here so the native dispatch below can map it to the
+        // PJSUA slot (answer the correct call from a push).
+        String sipCallId = "";
         try {
             AccountInfo accInfo = getInfo();
             com.voipcloud.sdk.CallInfo callInfo = call.getInfo();
             from = callInfo.getRemoteUri();
             to = accInfo.getUri();
+            sipCallId = callInfo.getCallIdString();
             WritableMap payload = Arguments.createMap();
             payload.putInt("accountId", accountId());
             payload.putInt("callId", call.callId());
             payload.putString("from", from);
             payload.putString("to", to);
-            // SIP Call-ID header — stable across systems (PBX CDR, CRM sync),
-            // unlike the numeric PJSUA call slot which is recycled.
-            payload.putString("sipCallId", callInfo.getCallIdString());
+            payload.putString("sipCallId", sipCallId);
             Events.emit(ctx, Events.INCOMING_CALL, payload);
         } catch (Exception ignored) {
             WritableMap payload = Arguments.createMap();
@@ -80,12 +84,12 @@ public final class ManagedAccount extends Account {
             payload.putInt("callId", call.callId());
             Events.emit(ctx, Events.INCOMING_CALL, payload);
         }
-        dispatchIncomingToApp(ctx, accountId(), call.callId(), from, to);
+        dispatchIncomingToApp(ctx, accountId(), call.callId(), from, to, sipCallId);
     }
 
     /** Notify VoicePBX native layer before JS (background incoming calls). */
     private static void dispatchIncomingToApp(
-            ReactContext ctx, int accountId, int callId, String from, String to) {
+            ReactContext ctx, int accountId, int callId, String from, String to, String sipCallId) {
         try {
             Class<?> cls = Class.forName("com.voicepbx.voip.VoipIncomingNotifier");
             cls.getMethod(
@@ -94,6 +98,7 @@ public final class ManagedAccount extends Account {
                     int.class,
                     int.class,
                     String.class,
+                    String.class,
                     String.class)
                 .invoke(
                     null,
@@ -101,7 +106,8 @@ public final class ManagedAccount extends Account {
                     accountId,
                     callId,
                     from != null ? from : "",
-                    to != null ? to : "");
+                    to != null ? to : "",
+                    sipCallId != null ? sipCallId : "");
         } catch (Throwable ignored) {}
     }
 }

package/android/src/main/java/com/voipcloud/sdk/rn/internal/ManagedCall.java

@@ -132,6 +132,12 @@ public final class ManagedCall extends Call {
             p.putInt("code", 0);
             p.putString("reason", "");
             Events.emit(ctx, Events.CALL_STATE, p);
+            // Native call-end hook: clear the incoming notification / ringing
+            // even when JS (React Native) is gone — a headless FCM task may have
+            // torn down its RN instance while PJSIP stays alive to receive the
+            // CANCEL/BYE. Without this a cancelled call keeps ringing on next
+            // app open.
+            if (cachedId >= 0) dispatchCallEndToApp(ctx, cachedId, "");
             if (cachedId >= 0) registry.remove(cachedId);
             return;
         }
@@ -142,15 +148,41 @@ public final class ManagedCall extends Call {
         p.putString("reason", info.getLastReason());
         // SIP Call-ID header — lets JS attach the dialog id to outgoing
         // calls too (makeCall only returns the recycled PJSUA slot).
-        try { p.putString("sipCallId", info.getCallIdString()); } catch (Exception ignored) {}
+        String sipCallId = "";
+        try { sipCallId = info.getCallIdString(); } catch (Exception ignored) {}
+        p.putString("sipCallId", sipCallId);
         Events.emit(ctx, Events.CALL_STATE, p);
         int state = info.getState();
         if (state == pjsip_inv_state.PJSIP_INV_STATE_DISCONNECTED
                 || state == pjsip_inv_state.PJSIP_INV_STATE_NULL) {
+            dispatchCallEndToApp(ctx, callId(), sipCallId);
             registry.remove(callId());
         }
     }
 
+    /**
+     * Notify the VoicePBX native layer that a call ended (background hangup /
+     * remote cancel), so it can cancel the incoming notification, stop the
+     * ringtone and clear the pending open-call id even with no live JS. Mirror
+     * of ManagedAccount.dispatchIncomingToApp. Best-effort via reflection so
+     * the SDK has no compile dependency on the app package.
+     */
+    private static void dispatchCallEndToApp(ReactContext ctx, int callId, String sipCallId) {
+        try {
+            Class<?> cls = Class.forName("com.voicepbx.voip.VoipCallEndNotifier");
+            cls.getMethod(
+                    "onCallEnded",
+                    android.content.Context.class,
+                    int.class,
+                    String.class)
+                .invoke(
+                    null,
+                    ctx.getApplicationContext(),
+                    callId,
+                    sipCallId != null ? sipCallId : "");
+        } catch (Throwable ignored) {}
+    }
+
     @Override
     public void onCallMediaState(OnCallMediaStateParam prm) {
         connectAudioMedia();