@voipsdk/react-native-voipcloud 1.0.2 → 1.0.3

package/CHANGELOG.md

@@ -5,6 +5,41 @@ 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

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 +1,7 @@
 # @voipsdk/react-native-voipcloud
 
-> Public PJSIP VoIP SDK for React Native — Android **and** iOS (RN 0.70+).  
-> Package: `@voipsdk/react-native-voipcloud@1.1.0`
+> 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.
 
@@ -41,7 +41,7 @@ packages.add(new com.voipcloud.sdk.rn.VoipCloudPackage());
 
 ### iOS
 
-Autolinking installs the pod (which vendors the prebuilt `VoipCloud.xcframework` — PJSIP + OpenSSL, self-contained). Just run:
+Autolinking installs the pod (which vendors the prebuilt `VoipCloud.xcframework` — native SIP engine, self-contained). Just run:
 
 ```bash
 cd ios && pod install
@@ -79,7 +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.
@@ -142,7 +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 |
@@ -163,7 +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`
 
@@ -196,7 +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 {
@@ -239,13 +239,12 @@ try {
 
 | Artifact | Name |
 |----------|------|
-| Prebuilt framework | `VoipCloud.xcframework` — PJSIP + OpenSSL, static, self-contained |
-| PJSUA2 wrapper | `VoipCloudEngine` (Objective-C++) |
+| Prebuilt framework | `VoipCloud.xcframework` — native SIP engine, static, self-contained |
+| Native SIP wrapper | `VoipCloudEngine` (Objective-C++) |
 | RN bridge module | `VoipsdkReactNativeVoipcloud` (`RCTEventEmitter`, JS: `VoipCloud`) |
 
-The iOS framework is rebuilt from `vendor/pjproject` + `vendor/openssl` via
-`scripts/build-openssl.sh` → `build-pjproject.sh {device,simulator-x86_64,simulator-arm64}`
-→ `build-xcframework.sh` (the last step also copies it into `ios/`).
+Both platforms ship their native engine prebuilt and self-contained — no extra
+native toolchain or SIP stack is required in the host app.
 
 ---
 
@@ -265,7 +264,7 @@ 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**  
 Android: grant `RECORD_AUDIO` at runtime before placing a call.
@@ -280,4 +279,4 @@ 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

@@ -36,7 +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();

package/ios/VoipCloudEngine.mm

@@ -8,6 +8,8 @@
 #include <pjsua-lib/pjsua.h>
 #include <string>
 #include <map>
+#include <algorithm>
+#include <cctype>
 
 using namespace pj;
 
@@ -270,6 +272,12 @@ void VCCall::onDtmfDigit(OnDtmfDigitParam &prm) {
     VCLogWriter              *_logWriter;
     std::map<int, VCAccount*> _accounts;
     std::map<int, VCCall*>    _calls;
+    // Transport ids returned by transportCreate, so an account can be pinned to
+    // the SAME socket/port for REGISTER + INVITE + in-dialog re-INVITE/BYE
+    // (stable NAT binding across hold/unhold). -1 = that transport not created.
+    int                       _udpTransportId;
+    int                       _tcpTransportId;
+    int                       _tlsTransportId;
 }
 
 + (instancetype)shared {
@@ -315,17 +323,42 @@ void VCCall::onDtmfDigit(OnDtmfDigitParam &prm) {
         _ep->libInit(epCfg);
 
         // Ephemeral ports (0) avoid "Address already in use" for SIP clients.
-        { TransportConfig t; t.port = 0; _ep->transportCreate(PJSIP_TRANSPORT_UDP, t); }
-        { TransportConfig t; t.port = 0; _ep->transportCreate(PJSIP_TRANSPORT_TCP, t); }
+        // Keep each transport id so registerAccountWithConfig can pin the
+        // account to the matching one (0 is a valid id → init to -1).
+        _udpTransportId = _tcpTransportId = _tlsTransportId = -1;
+        { TransportConfig t; t.port = 0; _udpTransportId = _ep->transportCreate(PJSIP_TRANSPORT_UDP, t); }
+        { TransportConfig t; t.port = 0; _tcpTransportId = _ep->transportCreate(PJSIP_TRANSPORT_TCP, t); }
         {
             TransportConfig t; t.port = 0;
             t.tlsConfig.verifyServer = false;
             t.tlsConfig.verifyClient = false;
             t.tlsConfig.method = PJSIP_TLSV1_2_METHOD;
-            _ep->transportCreate(PJSIP_TRANSPORT_TLS, t);
+            _tlsTransportId = _ep->transportCreate(PJSIP_TRANSPORT_TLS, t);
         }
 
         _ep->libStart();
+
+        // Codec allow-list. The PBX/proxy speaks G.711 (+G722 fallback); PJSIP's
+        // full default set (speex 8k/16k/32k, iLBC, …) makes the proxy rewrite
+        // the m= line and shift dynamic payload-type numbers → wrong codec map →
+        // broken audio. Priority 0 = codec not offered at all. Order: PCMU/PCMA
+        // first, G722 as fallback, so the far end answers G.711. Codec priority
+        // is endpoint-global in PJSIP, so it is configured once here.
+        try {
+            CodecInfoVector2 codecs = _ep->codecEnum2();
+            for (size_t i = 0; i < codecs.size(); i++) {
+                std::string cid = codecs[i].codecId;   // e.g. "PCMU/8000/1"
+                std::string fam = cid.substr(0, cid.find('/'));
+                std::transform(fam.begin(), fam.end(), fam.begin(),
+                               [](unsigned char c){ return (char)std::toupper(c); });
+                pj_uint8_t prio = 0;
+                if      (fam == "PCMU") prio = 255;
+                else if (fam == "PCMA") prio = 254;
+                else if (fam == "G722") prio = 200;
+                _ep->codecSetPriority(codecs[i].codecId, prio);
+            }
+        } catch (...) {}
+
         [self configureAudioSessionCategory];
         _isStarted = YES;
         return YES;
@@ -372,6 +405,15 @@ void VCCall::onDtmfDigit(OnDtmfDigitParam &prm) {
         acfg.idUri = std::string(uri.UTF8String);
         acfg.regConfig.registrarUri = std::string(registrar.UTF8String);
 
+        // Outbound proxy (Kamailio). Route EVERY SIP request — including
+        // in-dialog re-INVITE / BYE — through the proxy so a NAT'd client's
+        // teardown/unhold doesn't go straight to the peer's private address and
+        // get lost (peer stuck on MOH after unhold, call never ends).
+        NSString *proxy = cfg[@"proxy"];
+        if ([proxy isKindOfClass:[NSString class]] && proxy.length) {
+            acfg.sipConfig.proxies.push_back(std::string(proxy.UTF8String));
+        }
+
         // NAT survival — keep the registration binding alive behind NAT so
         // inbound INVITEs still reach the device (matches Android).
         acfg.natConfig.contactRewriteUse = 1;
@@ -381,11 +423,43 @@ void VCCall::onDtmfDigit(OnDtmfDigitParam &prm) {
         acfg.regConfig.firstRetryIntervalSec = 5;
         acfg.regConfig.retryIntervalSec = 30;
 
+        // STUN for MEDIA only: without it the SDP offer carries the LAN IP
+        // (c=192.168.x) and the far end sends RTP to an unroutable address →
+        // no 2-way audio. Falls back to the host addr if STUN is unreachable.
+        // SIP keeps rport/contactRewrite — do NOT STUN it (would disturb
+        // REGISTER). STUN server itself is set at endpoint level (stunServer).
+        acfg.natConfig.mediaStunUse = PJSUA_STUN_RETRY_ON_FAILURE;
+        acfg.natConfig.sipStunUse   = PJSUA_STUN_USE_DISABLED;
+
         AuthCredInfo cred("digest", "*", std::string(username.UTF8String), 0,
                           std::string(password.UTF8String));
         acfg.sipConfig.authCreds.push_back(cred);
 
-        acfg.mediaConfig.srtpUse = PJMEDIA_SRTP_OPTIONAL;
+        // Pin REGISTER + INVITE + in-dialog re-INVITE/BYE to the SAME transport
+        // (socket/port) so the NAT binding is stable across hold/unhold. Without
+        // this PJSIP may auto-pick a different transport per request. Falls back
+        // to auto-select if the requested transport failed to create (-1).
+        NSString *transport = cfg[@"transport"];
+        int pinnedTransportId = -1;
+        if ([transport isEqualToString:@"TLS"])      pinnedTransportId = _tlsTransportId;
+        else if ([transport isEqualToString:@"TCP"]) pinnedTransportId = _tcpTransportId;
+        else if ([transport isEqualToString:@"UDP"]) pinnedTransportId = _udpTransportId;
+        if (pinnedTransportId >= 0) acfg.sipConfig.transportId = pinnedTransportId;
+
+        // No session timers on the device — the VoIP server owns session
+        // refresh. With them on, the device (refresher) proposes Session-Expires
+        // and periodically sends a chatty UPDATE after ACK.
+        acfg.callConfig.timerUse = PJSUA_SIP_TIMER_INACTIVE;
+
+        // SRTP is opt-in per account. A plain-UDP trunk with srtp='disabled'
+        // must NOT offer a=crypto (an optional-SRTP offer breaks media on some
+        // PBX/rtpengine setups → one-way / no audio on outgoing). Default
+        // DISABLED when the account doesn't ask for it.
+        NSString *srtp = cfg[@"srtp"];
+        pjmedia_srtp_use srtpUse = PJMEDIA_SRTP_DISABLED;
+        if ([srtp isEqualToString:@"mandatory"])     srtpUse = PJMEDIA_SRTP_MANDATORY;
+        else if ([srtp isEqualToString:@"optional"]) srtpUse = PJMEDIA_SRTP_OPTIONAL;
+        acfg.mediaConfig.srtpUse = srtpUse;
         acfg.mediaConfig.srtpSecureSignaling = 0;
 
         VCAccount *acc = new VCAccount();
@@ -425,6 +499,8 @@ void VCCall::onDtmfDigit(OnDtmfDigitParam &prm) {
         VCCall *call = new VCCall(*it->second);
         call->owner = self;
         CallOpParam prm(true);
+        prm.opt.audioCount = 1;
+        prm.opt.videoCount = 0;   // audio-only: no stray video m-line in the offer
         call->makeCall(std::string(uri.UTF8String), prm);
         int cid = call->getId();
         call->cachedId = cid;
@@ -470,7 +546,13 @@ void VCCall::onDtmfDigit(OnDtmfDigitParam &prm) {
 - (BOOL)holdCall:(int)callId error:(NSError **)error {
     VCCall *call = [self callForId:callId];
     if (!call) { if (error) *error = VCMakeError(-1, [NSString stringWithFormat:@"Unknown callId %d", callId]); return NO; }
-    try { CallOpParam prm(true); call->setHold(prm); return YES; }
+    try {
+        CallOpParam prm(true);
+        prm.opt.audioCount = 1;
+        prm.opt.videoCount = 0;   // audio-only re-INVITE
+        call->setHold(prm);
+        return YES;
+    }
     catch (Error &err) { if (error) *error = VCMakeError(err.status, VCStr(err.info())); return NO; }
 }
 
@@ -480,6 +562,8 @@ void VCCall::onDtmfDigit(OnDtmfDigitParam &prm) {
     try {
         CallOpParam prm(true);
         prm.opt.flag |= PJSUA_CALL_UNHOLD;
+        prm.opt.audioCount = 1;
+        prm.opt.videoCount = 0;   // audio-only re-INVITE
         call->reinvite(prm);
         return YES;
     } catch (Error &err) { if (error) *error = VCMakeError(err.status, VCStr(err.info())); return NO; }

package/lib/typescript/types.d.ts

@@ -31,6 +31,20 @@ export interface AccountConfig {
      * the two in sync to avoid surprises.
      */
     transport: Transport;
+    /**
+     * Outbound proxy URI, e.g. "sip:proxy.example.com:5060;lr".  When set,
+     * every SIP request — including in-dialog re-INVITE / BYE — is routed
+     * through this proxy (`proxies.push_back`).  Required for in-dialog SIP
+     * to survive NAT: without a proxy Route a NAT'd client sends its BYE /
+     * unhold re-INVITE straight to the peer's private address and it is lost.
+     */
+    proxy?: string;
+    /**
+     * SRTP media policy.  Defaults to 'disabled' when omitted so a plain-UDP
+     * trunk never offers a=crypto (an optional-SRTP offer breaks media on some
+     * PBX / rtpengine setups → one-way / no audio).
+     */
+    srtp?: 'disabled' | 'optional' | 'mandatory';
 }
 export interface RegistrationStateEvent {
     accountId: number;

package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "@voipsdk/react-native-voipcloud",
-  "version": "1.0.2",
+  "version": "1.0.3",
   "description": "Public React Native VoIP SDK backed by PJSIP (VoipCloud).",
   "main": "lib/commonjs/index.js",
   "module": "lib/module/index.js",
@@ -16,6 +16,7 @@
     "react-native.config.js",
     "README.md",
     "CHANGELOG.md",
+    "LICENSE",
     "!**/__tests__",
     "!**/.*"
   ],
@@ -41,7 +42,7 @@
     "name": "VoipCloud",
     "email": "htt.bkap@gmail.com"
   },
-  "license": "UNLICENSED",
+  "license": "MIT",
   "repository": {
     "type": "git",
     "url": "git+https://github.com/hoangdx/react-native-voipcloud.git"

package/react-native-voipcloud.podspec

@@ -34,10 +34,15 @@ Pod::Spec.new do |s|
     # fix arm64 endianness detection and to keep struct layouts ABI-compatible
     # with the prebuilt libVoipCloud.a.
     "GCC_PREPROCESSOR_DEFINITIONS" => "$(inherited) PJ_AUTOCONF=1",
-    # pjsua2.hpp and friends live in each slice's bundled Headers/ dir. The
-    # trailing /** makes Xcode search recursively so it resolves for whichever
-    # slice (device / simulator) is being built.
-    "HEADER_SEARCH_PATHS" => "\"$(PODS_TARGET_SRCROOT)/ios/VoipCloud.xcframework/**\""
+    # pjsua2.hpp and the pj/ pjsip/ pjmedia/ trees live at each slice's Headers/
+    # root, and PJSIP includes are all rooted there (<pjsua2.hpp>, <pj/xxx.h>).
+    # Point at that root directly — one path per slice, only the matching one
+    # exists per build. Do NOT use a recursive /** glob: it drags every subdir,
+    # including pj/compat/, onto the search path. pj/compat/ ships files named
+    # ctype.h, string.h, setjmp.h, etc. that shadow the C stdlib, so PJSIP's own
+    # `#include <ctype.h>` resolves to the shadow (self, guard already defined)
+    # instead of libc — leaving isalnum/strlen/memset/jmp_buf undeclared.
+    "HEADER_SEARCH_PATHS" => "\"$(PODS_TARGET_SRCROOT)/ios/VoipCloud.xcframework/ios-arm64/Headers\" \"$(PODS_TARGET_SRCROOT)/ios/VoipCloud.xcframework/ios-arm64_x86_64-simulator/Headers\""
   }
 
   s.dependency "React-Core"

package/src/types.ts

@@ -34,6 +34,20 @@ export interface AccountConfig {
      * the two in sync to avoid surprises.
      */
     transport: Transport;
+    /**
+     * Outbound proxy URI, e.g. "sip:proxy.example.com:5060;lr".  When set,
+     * every SIP request — including in-dialog re-INVITE / BYE — is routed
+     * through this proxy (`proxies.push_back`).  Required for in-dialog SIP
+     * to survive NAT: without a proxy Route a NAT'd client sends its BYE /
+     * unhold re-INVITE straight to the peer's private address and it is lost.
+     */
+    proxy?: string;
+    /**
+     * SRTP media policy.  Defaults to 'disabled' when omitted so a plain-UDP
+     * trunk never offers a=crypto (an optional-SRTP offer breaks media on some
+     * PBX / rtpengine setups → one-way / no audio).
+     */
+    srtp?: 'disabled' | 'optional' | 'mandatory';
 }
 
 export interface RegistrationStateEvent {