Question

我刚开始一个新项目，作为概念证明，我只想用sip进行简单的音频通话。在应用程序中，我输入一个电话号码，然后单击一个按钮即可开始通话。我正在两个Android设备之间测试这个。在设备A上，我安装了应用程序并单击了呼叫按钮。设备B开始响铃，所以我知道我有连接。我的问题是两边都没有音频。设备A上甚至没有拨号音。

这是我的清单

<uses-sdk android:minSdkVersion="9" />

<uses-permission android:name="android.permission.USE_SIP" />
<uses-permission android:name="android.permission.INTERNET" />
<uses-permission android:name="android.permission.WAKE_LOCK" />
<uses-permission android:name="android.permission.CONFIGURE_SIP" />
<uses-permission android:name="android.permission.ACCESS_NETWORK_STATE"/>
<uses-permission android:name="android.permission.MODIFY_AUDIO_SETTINGS" />
<uses-permission android:name="android.permission.PROCESS_OUTGOING_CALLS" />

<uses-feature android:name="android.software.sip" android:required="true" />
<uses-feature android:name="android.hardware.wifi" android:required="true" />
<uses-feature android:name="android.hardware.sip.voip" android:required="true" />
<uses-feature android:name="android.hardware.microphone" android:required="true" />

这是我的java代码

public void onCreate(Bundle savedInstanceState) {
    super.onCreate(savedInstanceState);
    setContentView(R.layout.activity_make_call);

    //getWindow().addFlags(WindowManager.LayoutParams.FLAG_KEEP_SCREEN_ON);
    initializeManager();
}

public void initializeManager()
{
    if(manager == null)
    {
        manager = SipManager.newInstance(this);
    }
    initializeLocalProfile();
}

public void initializeLocalProfile()
{
    if (manager == null)
    {
        return;
    }
    if (me != null)
    {
        closeLocalProfile();
    }

    String username = "username"; // I do have the correct credentials
    String domain = "proxy";
    String password = "password";

    try {
        SipProfile.Builder builder = new SipProfile.Builder(username, domain);
        builder.setPassword(password);

        me = builder.build();
        Intent i = new Intent();
        i.setAction("android.SipDemo.INCOMING_CALL");
        PendingIntent pi = PendingIntent.getBroadcast(this, 0, i, Intent.FILL_IN_DATA);
        manager.open(me, pi, null);
        // This listener must be added AFTER manager.open is called,
        // Otherwise the methods aren't guaranteed to fire.
        manager.setRegistrationListener(me.getUriString(), new SipRegistrationListener() {
            public void onRegistering(String localProfileUri) {
                Log.d("call","Registering with SIP Server...");
            }
            public void onRegistrationDone(String localProfileUri, long expiryTime) {
                Log.d("call","Ready");
            }
            public void onRegistrationFailed(String localProfileUri, int errorCode, String errorMessage) {
                Log.d("call","Registration failed.  Please check settings.");
            }
        });
    } catch (ParseException pe) {
        Log.d("err","Connection Error.");
    } catch (SipException se) {
        Log.d("err","Connection error.");
    }
}

public void closeLocalProfile()
{
    if (manager == null)
    {
        return;
    }
    try
    {
        if (me != null)
        {
            manager.close(me.getUriString());
        }
    }
    catch (Exception ee)
    {
        Log.d("onDestroy", "Failed to close local profile.", ee);
    }
}

public void onCallButtonTap(View v)
{
    phoneText = findViewById(R.id.et_phoneNumber);
    ClientPhoneNumber = phoneText.getText().toString();

    displayMessage("Call starting...");

    try
    {
        SipAudioCall.Listener listener = new SipAudioCall.Listener()
        {
            @Override
            public void onCallEstablished(SipAudioCall call)
            {
                Log.d("log", "Call started!");
                call.setSpeakerMode(true);
                call.startAudio();


            }
            @Override
            public void onCallEnded(SipAudioCall call)
            {
                displayMessage("Call Ended");
                closeLocalProfile();
            }
        };
        manager.makeAudioCall(me.getUriString(), ClientPhoneNumber + "@proxy.cloudpbx.voiportal.net:5060", listener, 30);
    }
    catch (Exception e)
    {
        if (me != null)
        {
            try
            {
                manager.close(me.getUriString());
            }
            catch (SipException e1)
            {
                e1.printStackTrace();
            }
        }
        if (call != null)
        {
            call.close();
        }
    }
}

Answer 1

我现在尝试使用SIP库，并且是Android开发中的新手。

首先，我在我的项目中关注demo code。只需将所有文件拖到项目中并修改包的一些名称空间

即可

我的Manifests权限和功能如下所示：

<uses-permission android:name="android.permission.USE_SIP" />
<uses-permission android:name="android.permission.RECORD_AUDIO" />
<uses-permission android:name="android.permission.ACCESS_WIFI_STATE" />
<uses-permission android:name="android.permission.INTERNET" />
<uses-permission android:name="android.permission.WAKE_LOCK" />
<uses-permission android:name="android.permission.MODIFY_AUDIO_SETTINGS" />

<uses-feature android:name="android.software.sip" android:required="true" />
<uses-feature android:name="android.hardware.sip.voip" android:required="true" />
<uses-feature android:name="android.hardware.wifi" android:required="true" />
<uses-feature android:name="android.hardware.microphone" android:required="true" />

活动＆amp;接收器你可以参考链接中的清单文件。

30分钟前我没有声音的原因是我在android:name留下逗号... OMG

顺便说一句，我使用一个稳定的VoIP软件来调用我的应用程序，这是测试代码中是否有任何问题的好方法。

祝你好运〜我也是Android开发中的新手，为期两周。

Answer 2

感谢您的回复。后来我发现我在清单中缺少一两个权限。这是我现在在清单中设置的完整权限。现在一切正常。

<uses-permission android:name="android.permission.USE_SIP" />
<uses-permission android:name="android.permission.INTERNET" />
<uses-permission android:name="android.permission.WAKE_LOCK" />
<uses-permission android:name="android.permission.CALL_PHONE" />
<uses-permission android:name="android.permission.RECORD_AUDIO" />
<uses-permission android:name="android.permission.CONFIGURE_SIP" />
<uses-permission android:name="android.permission.ACCESS_WIFI_STATE" />
<uses-permission android:name="android.permission.MODIFY_AUDIO_SETTINGS" />
<uses-permission android:name="android.permission.PROCESS_OUTGOING_CALLS" />
<uses-permission android:name="android.permission.READ_PHONE_STATE" />

<uses-feature
    android:name="android.software.sip"
    android:required="true" />
<uses-feature
    android:name="android.hardware.sip.voip"
    android:required="true" />
<uses-feature
    android:name="android.hardware.wifi"
    android:required="true" />
<uses-feature
    android:name="android.hardware.microphone"
    android:required="true" />
<uses-feature
    android:name="android.hardware.telephony"
    android:required="true" />

为了模拟拨号音，我必须在onRingBack重载中使用medial播放器。

 @Override
            public void onRingingBack(final SipAudioCall call) {
                Log.i("SWARM", "Ringing...");

                try {
                    Resources res = getResources();
                    AssetFileDescriptor afd = res.openRawResourceFd(R.raw.ringing);
                    mp = new MediaPlayer();
                    mp.setAudioStreamType(AudioManager.STREAM_VOICE_CALL);
                    mp.setLooping(true);
                    try {
                        mp.setDataSource(afd.getFileDescriptor(), afd.getStartOffset(), afd.getLength());
                        mp.prepare();
                    } catch (IOException e) {
                        e.printStackTrace();
                    }
                    mp.start();
                } catch (IllegalArgumentException e1) {
                    e1.printStackTrace();
                } catch (SecurityException e1) {
                    e1.printStackTrace();
                } catch (IllegalStateException e1) {
                    e1.printStackTrace();
                }

                setButtonToRed();
            }

Android SIP完全没有音频

2 个答案: