我正在构建一个 PJSUA2 (PJSIP 2.8) Android 应用程序，但我遇到了一些问题：即仅在来电时，呼叫状态保持在“PJSIP_INV_STATE_CONNECTING”，32 秒后呼叫中断。
几天以来我一直在寻找问题的原因，我搜索了很多，我发现的只是：在大多数情况下，这个问题与 NAT 管理或与 NAT 相关的网络问题有关。简而言之：大多数情况下，被叫方接听电话后并没有收到ACK。最后，我能够记录我的应用程序和 SIP 服务器之间的所有 SIP 消息，并发现我的应用程序从服务器接收到 ACK，所以我认为这不是网络相关问题。
我用 OpenSSL 和 SRTP 支持编译了 PJSIP 2.8，但没有视频支持（至少目前我不需要它）。如果有任何区别，则该应用程序的目标版本为 28，最低 SDK 版本为 19。
我尝试了市场上的几个应用程序，它们在使用和不使用 SRTP 以及所有信令传输（UDP、TCP、TLS）的情况下都运行良好，WebRTC 也运行良好（使用 SipML5 测试），所以我会排除服务器配置错误。我的应用程序也是如此（目前我遇到一些问题的 SRTP 除外）。
我也尝试过使用 UDP 的 SIP 提供商（MessageNet），并且行为始终相同。我尝试使用紧凑的 SIP 消息，它的行为相同，有和没有 uri 参数，有和没有 STUN 和或 ICE，没有任何变化。移动网络和 WiFi 网络给出相同的结果。
我也尝试在 PJSIP 库中进行调试，但没有任何成功，然后我尝试按照代码进行操作，以了解我做错了什么，但在我看来并没有明显的问题。
以下是初始化 PJSIP 的代码（最新版本）：
public class SipService extends Service {
private Looper serviceLooper;
private ServiceHandler serviceHandler;
private final Messenger mMessenger = new Messenger(new IncomingHandler());
private LocalBroadcastManager localBroadcast;
private LifecycleBroadcastReceiver lifecycleBroadcastReceiver;
private boolean lastCheckConnected;
private Endpoint endpoint;
private LogWriter logWriter;
private EpConfig epConfig;
private final List<ManagedSipAccount> accounts = new ArrayList<>();
private final Map<String, Messenger> eventRegistrations = new HashMap<>();
@TargetApi(Build.VERSION_CODES.N)
@Override
public void onCreate() {
super.onCreate();
String userAgent = "MyApp";
try {
PackageInfo pInfo = getPackageManager().getPackageInfo(getPackageName(), 0);
String appLabel = (pInfo.applicationInfo.labelRes == 0 ? pInfo.applicationInfo.nonLocalizedLabel.toString() : getString(pInfo.applicationInfo.labelRes));
userAgent = appLabel + "/" + pInfo.versionName;
} catch (PackageManager.NameNotFoundException e) {
Log.e("SipService", "Unable to get app version", e);
}
try {
endpoint = new MyAppEndpoint();
endpoint.libCreate();
epConfig = new EpConfig();
// Logging
logWriter = new PJSIPToAndroidLogWriter();
epConfig.getLogConfig().setWriter(logWriter);
epConfig.getLogConfig().setLevel(5);
// UA
epConfig.getUaConfig().setMaxCalls(4);
epConfig.getUaConfig().setUserAgent(userAgent);
// STUN
StringVector stunServer = new StringVector();
stunServer.add("stun.pjsip.org");
epConfig.getUaConfig().setStunServer(stunServer);
// General Media
epConfig.getMedConfig().setSndClockRate(16000);
endpoint.libInit(epConfig);
// UDP transport
TransportConfig udpCfg = new TransportConfig();
udpCfg.setQosType(pj_qos_type.PJ_QOS_TYPE_VOICE);
endpoint.transportCreate(pjsip_transport_type_e.PJSIP_TRANSPORT_UDP, udpCfg);
// TCP transport
TransportConfig tcpCfg = new TransportConfig();
//tcpCfg.setPort(5060);
endpoint.transportCreate(pjsip_transport_type_e.PJSIP_TRANSPORT_TCP, tcpCfg);
// TLS transport
TransportConfig tlsCfg = new TransportConfig();
endpoint.transportCreate(pjsip_transport_type_e.PJSIP_TRANSPORT_TLS, tlsCfg);
endpoint.libStart();
} catch (Exception e) {
throw new RuntimeException("Unable to initialize and start PJSIP", e);
}
ConnectivityManager cm = (ConnectivityManager)getSystemService(Context.CONNECTIVITY_SERVICE);
NetworkInfo activeNetwork = cm.getActiveNetworkInfo();
lastCheckConnected = activeNetwork != null && activeNetwork.isConnected();
updateForegroundNotification();
startForeground(MyAppConstants.N_FOREGROUND_NOTIFICATION_ID, buildForegroundNotification());
localBroadcast = LocalBroadcastManager.getInstance(this);
HandlerThread thread = new HandlerThread("ServiceStartArguments",
Process.THREAD_PRIORITY_BACKGROUND);
thread.start();
// Get the HandlerThread's Looper and use it for our Handler
serviceLooper = thread.getLooper();
serviceHandler = new ServiceHandler(serviceLooper);
if (Build.VERSION.SDK_INT >= Build.VERSION_CODES.N) {
// Register LifeCycleBroadcastReceiver to receive network change notification
// It seems it's mandatory to do it programmatically since Android N (24)
lifecycleBroadcastReceiver = new LifecycleBroadcastReceiver();
IntentFilter intentFilter = new IntentFilter("android.net.conn.CONNECTIVITY_CHANGE");
registerReceiver(lifecycleBroadcastReceiver, intentFilter);
}
// Initialization
SharedPreferences prefs = PreferenceManager.getDefaultSharedPreferences(this);
if (prefs != null) {
try {
CodecInfoVector codecs = endpoint.codecEnum();
SharedPreferences.Editor editor = prefs.edit();
for (int i = 0; i < codecs.size(); i++) {
CodecInfo codec = codecs.get(i);
int priority = prefs.getInt("codecs.audio{" + codec.getCodecId() + "}", 0);
try {
endpoint.codecSetPriority(codec.getCodecId(), (short) priority);
codec.setPriority((short) priority);
} catch (Exception e) {
Log.e("SipService", "Unexpected error setting codec priority for codec " + codec.getCodecId(), e);
}
}
} catch (Exception e) {
Log.e("SipService", "Unexpected error loading codecs priorities", e);
}
}
}
@Override
public void onDestroy() {
for (Account acc : accounts) {
acc.delete();
}
accounts.clear();
try {
endpoint.libDestroy();
} catch (Exception e) {
e.printStackTrace();
}
endpoint.delete();
endpoint = null;
epConfig = null;
if (lifecycleBroadcastReceiver != null) {
unregisterReceiver(lifecycleBroadcastReceiver);
}
super.onDestroy();
}
.......
}
以下是我的带有创建和注册代码的 Account 类：
public class ManagedSipAccount extends Account {
public final String TAG;
private final VoipAccount account;
private final PhoneAccountHandle handle;
private final SipService service;
private final AccountStatus status;
private final Map<Integer, VoipCall> calls = new HashMap<>();
private final Map<String, VoipBuddy> buddies = new HashMap<>();
private AccountConfig acfg;
private List<SrtpCrypto> srtpCryptos = new ArrayList<>();
private AuthCredInfo authCredInfo;
public ManagedSipAccount(SipService service, VoipAccount account, PhoneAccountHandle handle) {
super();
TAG = "ManagedSipAccount/" + account.getId();
this.service = service;
this.account = account;
this.handle = handle;
this.status = new AccountStatus(account.getUserName() + "@" + account.getHost());
acfg = new AccountConfig();
}
public void register(Map<String, String> contactParameters) throws Exception {
StringBuilder contactBuilder = new StringBuilder();
for (Map.Entry<String, String> entry : contactParameters.entrySet()) {
contactBuilder.append(';');
contactBuilder.append(URLEncoder.encode(entry.getKey(), "UTF-8"));
contactBuilder.append("=\"");
contactBuilder.append(URLEncoder.encode(entry.getValue(), "UTF-8"));
contactBuilder.append("\"");
}
StringBuilder logBuilder = new StringBuilder();
logBuilder.append("Registering: ");
logBuilder.append(account.getProtocol().name());
/*logBuilder.append('(');
logBuilder.append(service.getTransport(account.getProtocol()));
logBuilder.append(')');*/
if (account.isEncryptionSRTP()) {
logBuilder.append(" SRTP");
}
if (account.isIce()) {
logBuilder.append(" ICE");
}
Log.d(TAG, logBuilder.toString());
String idUri = "sip:" + account.getUserName();
if (!"*".equals(account.getRealm())) {
idUri += "@" + account.getRealm();
}
else {
idUri += "@127.0.0.1" /*+ account.getHost()*/;
}
acfg.setIdUri(idUri);
acfg.getRegConfig().setRegistrarUri("sip:" + account.getHost() + ":" + account.getPort() + ";transport=" + account.getProtocol().name().toLowerCase());
acfg.getRegConfig().setRetryIntervalSec(account.getRetryInterval());
acfg.getRegConfig().setRegisterOnAdd(false);
acfg.getSipConfig().setContactUriParams(contactBuilder.toString());
// NAT management
acfg.getNatConfig().setSipStunUse(pjsua_stun_use.PJSUA_STUN_USE_DEFAULT);
if (account.isIce()) {
acfg.getNatConfig().setIceEnabled(true);
acfg.getNatConfig().setIceAlwaysUpdate(true);
acfg.getNatConfig().setIceAggressiveNomination(true);
}
else {
acfg.getNatConfig().setSdpNatRewriteUse(1);
}
acfg.getMediaConfig().getTransportConfig().setQosType(pj_qos_type.PJ_QOS_TYPE_VOICE);
if (account.isEncryptionSRTP()) {
acfg.getMediaConfig().setSrtpUse(pjmedia_srtp_use.PJMEDIA_SRTP_MANDATORY);
acfg.getMediaConfig().setSrtpSecureSignaling(0);
//acfg.getMediaConfig().getSrtpOpt().setKeyings(new IntVector(2));
acfg.getMediaConfig().getSrtpOpt().getKeyings().clear();
acfg.getMediaConfig().getSrtpOpt().getKeyings().add(pjmedia_srtp_keying_method.PJMEDIA_SRTP_KEYING_SDES.swigValue());
acfg.getMediaConfig().getSrtpOpt().getKeyings().add(pjmedia_srtp_keying_method.PJMEDIA_SRTP_KEYING_DTLS_SRTP.swigValue());
acfg.getMediaConfig().getSrtpOpt().getCryptos().clear();
StringVector cryptos = Endpoint.instance().srtpCryptoEnum();
for (int i = 0; i < cryptos.size(); i++) {
SrtpCrypto crypto = new SrtpCrypto();
crypto.setName(cryptos.get(i));
crypto.setFlags(0);
srtpCryptos.add(crypto);
acfg.getMediaConfig().getSrtpOpt().getCryptos().add(crypto);
}
}
else {
acfg.getMediaConfig().setSrtpUse(pjmedia_srtp_use.PJMEDIA_SRTP_DISABLED);
acfg.getMediaConfig().setSrtpSecureSignaling(0);
}
authCredInfo = new AuthCredInfo("digest",
account.getRealm(),
account.getAuthenticationId() != null && account.getAuthenticationId().trim().length() > 0 ? account.getAuthenticationId() : account.getUserName(),
0,
account.getPassword());
acfg.getSipConfig().getAuthCreds().add( authCredInfo );
acfg.getIpChangeConfig().setHangupCalls(false);
acfg.getIpChangeConfig().setShutdownTp(true);
create(acfg);
ConnectivityManager cm = (ConnectivityManager)service.getSystemService(Context.CONNECTIVITY_SERVICE);
NetworkInfo activeNetwork = cm.getActiveNetworkInfo();
boolean isConnected = activeNetwork != null && activeNetwork.isConnected();
if (isConnected) {
setRegistration(true);
}
}
@Override
public void onRegStarted(OnRegStartedParam prm) {
super.onRegStarted(prm);
Log.d(TAG, "Status: Registering...");
status.setStatus(AccountStatus.Status.REGISTERING);
service.updateStatus(this);
}
@Override
public void onRegState(OnRegStateParam prm) {
super.onRegState(prm);
try {
Log.d(TAG, "Registration state: " + prm.getCode().swigValue() + " " + prm.getReason());
AccountInfo ai = getInfo();
status.setStatus(ai.getRegIsActive() ? AccountStatus.Status.REGISTERED : AccountStatus.Status.UNREGISTERED);
Log.d(TAG, "Status: " + status.getStatus().name() + " " + super.getInfo().getUri());
service.updateStatus(this);
} catch (Exception e) {
e.printStackTrace();
}
}
.....
}
最后，我现在如何在扩展 PJSIP 的 Call 类的类中回答代码：
@Override
public void answerCall() {
Log.d(TAG, "Answering call...");
CallOpParam prm = new CallOpParam(true);
prm.setStatusCode(pjsip_status_code.PJSIP_SC_OK);
prm.getOpt().setAudioCount(1);
prm.getOpt().setVideoCount(0);
try {
this.answer(prm);
} catch (Exception e) {
e.printStackTrace();
}
}
我也尝试了new CallOpParam();仅使用状态代码而没有其他任何内容，但没有任何变化。
注意事项：我将 IdUri 创建为 sip:username@127.0.0.1 因为没有主机，结果联系人是，我认为丢失的用户部分可能是问题的原因或部分原因。
以下是调用期间应用程序 <-> 我的 Asterisk 服务器通信的跟踪（由于内容长度超过而链接）。
https://gist.github.com/ivano85/a212ddc9a808f3cd991234725c2bdb45
ServerIp是互联网公网 IP，而 MyIp[5.XXX.XXX.XXX] 是我手机的公网 IP 。
正如您从日志中看到的那样，我的应用程序发送 100 Trying，然后在电话响铃时发送 180 Ringing，然后用户接听并且应用程序发送 200 OK。服务器回复 ACK 消息（我会说这不是 NAT 问题，因为 PJSIP 接收到 ACK）。我从 Asterisk 看到了同样的情况。
在此之后，我希望呼叫从 PJSIP_INV_STATE_CONNECTING 转到 PJSIP_INV_STATE_CONFIRMED，但它没有发生，因此 PJSIP 继续发送 200 OK 并每隔大约 2 秒接收一次 ACK，直到呼叫在 32 秒后超时并且 PJSIP 断开呼叫（发送再见）。
我开始认为 PJSIP 只是忽略了 ACK 消息并且只是有错误的行为。请帮助我了解这里发生了什么。我会非常感激！
显然，如果您认为需要更多详细信息，请告诉我。