CSipSimple拨打电话机制分析

CSipSimple是运行在android设备上的一个开源的sip协议应用程序，本文其中的拨打电话机制进行大致分析。

项目中，拨打电话利用了AIDL方法来实现。aidl是 Android Interface definition language的缩写，它是一种android内部进程通信接口的描述语言，通过它来定义进程间的通信接口，完成IPC（Inter-Process Communication，进程间通信）。

创建.aidl文件

ISipService.aidl内容如下：

 

/**
 * Copyright (C) 2010-2012 Regis Montoya (aka r3gis - www.r3gis.fr)
 * This file is part of CSipSimple.
 *
 *  CSipSimple is free software: you can redistribute it and/or modify
 *  it under the terms of the GNU General Public License as published by
 *  the Free Software Foundation, either version 3 of the License, or
 *  (at your option) any later version.
 *  If you own a pjsip commercial license you can also redistribute it
 *  and/or modify it under the terms of the GNU Lesser General Public License
 *  as an android library.
 *
 *  CSipSimple is distributed in the hope that it will be useful,
 *  but WITHOUT ANY WARRANTY; without even the implied warranty of
 *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 *  GNU General Public License for more details.
 *
 *  You should have received a copy of the GNU General Public License
 *  along with CSipSimple.  If not, see <http://www.gnu.org/licenses/>.
 *
 *  This file and this file only is also released under Apache license as an API file
 */
package com.csipsimple.api;
import com.csipsimple.api.SipProfileState;
import com.csipsimple.api.SipCallSession;
import com.csipsimple.api.MediaState;

interface ISipService{
    /**
    * Get the current API version
    * @return version number. 1000 x major version + minor version
    * Each major version must be compatible with all versions of the same major version
    */

.........
void makeCallWithOptions(in String callee, int accountId, in Bundle options);
}

ISipService.aidl中定义了包含makeCallWithOptions
方法的接口ISipService。

 

 

自动编译生成java文件

eclipse中的ADT插件会自动在aidl文件中声明的包名目录下生成java文件，如下图所示：

ISipService.java

 

package com.csipsimple.api;
public interface ISipService extends android.os.IInterface
{
……
//Place a call

ublic void makeCallWithOptions(java.lang.String callee, int accountId, android.os.Bundle options) throws android.os.RemoteException;
}

接下来就是实现ISipService.aidl中定义的接口，提供接口的实例供客户端调用

 

 

IPC实现

项目中拨打电话  

 

void com.csipsimple.api.ISipService.makeCallWithOptions(String msg, String toNumber, long accountId)

结合代码一层层看调用

目录：src\com\csipsimple\ui\dialpad

DialerFragment.java

 

private ISipService service;
private ServiceConnection connection = new ServiceConnection() {

    @Override
    public void onServiceConnected(ComponentName arg0, IBinder arg1) {
        service = ISipService.Stub.asInterface(arg1);
     ........
    }

    @Override
    public void onServiceDisconnected(ComponentName arg0) {
        service = null;
    }
};

 

 

<span style="color:#333333;">   @Override
    public void placeCall() {
        placeCallWithOption(null);
    }

private void placeCallWithOption(Bundle b) {
        if (service == null) {
            return;
        }
        String toCall = "";
        Long accountToUse = SipProfile.INVALID_ID;
        // Find account to use
        SipProfile acc = accountChooserButton.getSelectedAccount();
        if (acc != null) {
            accountToUse = acc.id;
        }
        // Find number to dial
        if(isDigit) {
            toCall = PhoneNumberUtils.stripSeparators(digits.getText().toString());
        }else {
            toCall = digits.getText().toString();
        }

        if (TextUtils.isEmpty(toCall)) {
            return;
        }

        // Well we have now the fields, clear theses fields
        digits.getText().clear();

        // -- MAKE THE CALL --//
        if (accountToUse >= 0) {
            // It is a SIP account, try to call service for that
            try {
               </span><span style="color:#ff0000;"> service.makeCallWithOptions(toCall, accountToUse.intValue(), b);</span><span style="color:#333333;">
            } catch (RemoteException e) {
                Log.e(THIS_FILE, "Service can't be called to make the call");
            }
        } else if (accountToUse != SipProfile.INVALID_ID) {
            // It's an external account, find correct external account
            CallHandlerPlugin ch = new CallHandlerPlugin(getActivity());
            ch.loadFrom(accountToUse, toCall, new OnLoadListener() {
                @Override
                public void onLoad(CallHandlerPlugin ch) {
                    placePluginCall(ch);
                }
            });
        }
    }
    </span>

 

这里的调用需要先了解Service的机制

service.makeCallWithOptions(toCall, accountToUse.intValue(), b)

方法调用了ISipService的方法，找到它的代码如下：

目录：src\com\csipsimple\service

2.服务端
SipService.java
/**
  * 继承 Service发布服务
  */
 public class SipService extends Service {
     ...

     // 为服务实现公共接口， Stub类继承了Binder
     private final ISipService.Stub binder = new ISipService.Stub() {
        ...
       @Override
        public void makeCallWithOptions(final String callee, final int accountId, final Bundle options)
                throws RemoteException {
            SipService.this.enforceCallingOrSelfPermission(SipManager.PERMISSION_USE_SIP, null);
            //We have to ensure service is properly started and not just binded
            SipService.this.startService(new Intent(SipService.this, SipService.class));

            if(pjService == null) {
                Log.e(THIS_FILE, "Can't place call if service not started");
                // TODO - we should return a failing status here
                return;
            }

            if(!supportMultipleCalls) {
                // Check if there is no ongoing calls if so drop this request by alerting user
                SipCallSession activeCall = pjService.getActiveCallInProgress();
                if(activeCall != null) {
                    if(!CustomDistribution.forceNoMultipleCalls()) {
                        notifyUserOfMessage(R.string.not_configured_multiple_calls);
                    }
                    return;
                }
            }
            getExecutor().execute(new SipRunnable() {
                @Override
                protected void doRun() throws SameThreadException {
                  <span style="color:#ff0000;">  pjService.makeCall(callee, accountId, options);
                }</span>
            });
        }

/**
      * 返回一个实现了接口的类对象，给客户端接收
      */
     @Override
     public IBinder onBind(Intent intent) {
 
        String serviceName = intent.getAction();
        Log.d(THIS_FILE, "Action is " + serviceName );
        if (serviceName == null || serviceName.equalsIgnoreCase(SipManager.INTENT_SIP_SERVICE )) {
            Log.d(THIS_FILE, "Service returned");
            return binder ;
        } else if (serviceName. equalsIgnoreCase(SipManager.INTENT_SIP_CONFIGURATION )) {
            Log.d(THIS_FILE, "Conf returned");
            return binderConfiguration ;
        }
        Log.d(THIS_FILE, "Default service (SipService) returned");
        return binder;
     }
     
     ...
 }

 

上文说过，需要实现ISipService.aidl中定义的接口，来提供接口的实例供客户端调用。要实现自己的接口，就从ISipService.Stub类继承，然后实现相关的方法。

Stub类继承了Binder，因此它的对象就可以被远程的进程调用了。如果Service中有对象继承了Stub类，那么这个对象中的方法就可以在Activity等地方中使用，也就是说此时makeCallWithOptions
就可以被其他Activity访问调用了。

现在我们通过onBind(Intent intent)方法得到了可供客户端接收的IBinder对象，就可以回头看看刚才DialerFragment.java文件中的调用情况了。

在客户端（此处也就是调用远程服务的Activity）实现ServiceConnection,在ServiceConnection.onServiceConnected()方法中会接收到IBinder对象，调用ISipService.Stub.asInterface((IBinder)service)将返回值转换为ISipService类型。

语句

<span style="color:#ff0000;">service.makeCallWithOptions(toCall, accountToUse.intValue(), b);</span><span style="font-family: Verdana, Arial, Helvetica, sans-serif;">调用接口中的方法，完成IPC方法。</span>

回到刚才的服务端实现，在继承Service发布服务的代码中，调用了 pjService.makeCall(callee, accountId, options)方法。

先看看这部分代码：

目录：src\com\csipsimple\pjsip

PjSipService.java

<span style="color:#333333;">public int makeCall(String callee, int accountId, Bundle b) throws SameThreadException {
        if (!created) {
            return -1;
        }

        final ToCall toCall = sanitizeSipUri(callee, accountId);
        if (toCall != null) {
            pj_str_t uri = pjsua.pj_str_copy(toCall.getCallee());

            // Nothing to do with this values
            byte[] userData = new byte[1];
            int[] callId = new int[1];
            pjsua_call_setting cs = new pjsua_call_setting();
            pjsua_msg_data msgData = new pjsua_msg_data();
            int pjsuaAccId = toCall.getPjsipAccountId();

            // Call settings to add video
            pjsua.call_setting_default(cs);
            cs.setAud_cnt(1);
            cs.setVid_cnt(0);
            if(b != null && b.getBoolean(SipCallSession.OPT_CALL_VIDEO, false)) {
                cs.setVid_cnt(1);
            }
            cs.setFlag(0);

            pj_pool_t pool = pjsua.pool_create("call_tmp", 512, 512);

            // Msg data to add headers
            pjsua.msg_data_init(msgData);
            pjsua.csipsimple_init_acc_msg_data(pool, pjsuaAccId, msgData);
            if(b != null) {
                Bundle extraHeaders = b.getBundle(SipCallSession.OPT_CALL_EXTRA_HEADERS);
                if(extraHeaders != null) {
                    for(String key : extraHeaders.keySet()) {
                        try {
                            String value = extraHeaders.getString(key);
                            if(!TextUtils.isEmpty(value)) {
                                int res = pjsua.csipsimple_msg_data_add_string_hdr(pool, msgData, pjsua.pj_str_copy(key), pjsua.pj_str_copy(value));
                                if(res == pjsuaConstants.PJ_SUCCESS) {
                                    Log.e(THIS_FILE, "Failed to add Xtra hdr (" + key + " : " + value + ") probably not X- header");
                                }
                            }
                        }catch(Exception e) {
                            Log.e(THIS_FILE, "Invalid header value for key : " + key);
                        }
                    }
                }
            }

            </span><span style="color:#ff0000;">int status = pjsua.call_make_call(pjsuaAccId, uri, cs, userData, msgData, callId);</span><span style="color:#333333;">
            if(status == pjsuaConstants.PJ_SUCCESS) {
                dtmfToAutoSend.put(callId[0], toCall.getDtmf());
                Log.d(THIS_FILE, "DTMF - Store for " + callId[0] + " - "+toCall.getDtmf());
            }
            pjsua.pj_pool_release(pool);
            return status;
        } else {
            service.notifyUserOfMessage(service.getString(R.string.invalid_sip_uri) + " : "
                    + callee);
        }
        return -1;
    }</span>

由红色部分的语句，我们找到pjsua类。

目录：src\org\pjsip\pjsua

pjsua.java

package org.pjsip.pjsua;

public class pjsua implements pjsuaConstants {
public synchronized static int call_make_call(int acc_id, pj_str_t dst_uri, pjsua_call_setting opt, byte[] user_data, pjsua_msg_data msg_data, int[] p_call_id) {
   <span style="color:#ff0000;"> return pjsuaJNI.call_make_call</span>(acc_id, pj_str_t.getCPtr(dst_uri), dst_uri, pjsua_call_setting.getCPtr(opt), opt, user_data, pjsua_msg_data.getCPtr(msg_data), msg_data, p_call_id);
  }
..........
}

继续看调用，找到pjsuaJNI文件。

目录：src\org\pjsip\pjsua

pjsuaJNI.java

/* ----------------------------------------------------------------------------
  * This file was automatically generated by SWIG (http://www.swig.org).
  * Version 2.0.4
  *
  * Do not make changes to this file unless you know what you are doing--modify
  * the SWIG interface file instead.
  * ----------------------------------------------------------------------------- */
 
 package org.pjsip.pjsua;
 
 public class pjsuaJNI {
 
     ...
     
   public final static native int call_make_call(int jarg1, long jarg2, pj_str_t jarg2_, long jarg3, pjsua_call_setting jarg3_, byte[] jarg4, long jarg5, pjsua_msg_data jarg5_, int[] jarg6);
     
     ...
     
 }

我们看到了native方法call_make_call，它调用的是封装在库libpjsipjni.so中的函数pjsua_call_make_call，进一步可以在jni目录下找到C代码。

目录：jni\pjsip\sources\pjsip\src\pjsua-lib

 

pjsua_call.c

PJ_DEF(pj_status_t) pjsua_call_make_call(pjsua_acc_id acc_id,
                     const pj_str_t *dest_uri,
                     const pjsua_call_setting *opt,
                     void *user_data,
                     const pjsua_msg_data *msg_data,
                     pjsua_call_id *p_call_id)
{
    pj_pool_t *tmp_pool = NULL;
    pjsip_dialog *dlg = NULL;
    pjsua_acc *acc;
    pjsua_call *call;
    int call_id = -1;
    pj_str_t contact;
    pj_status_t status;


    /* Check that account is valid */
    PJ_ASSERT_RETURN(acc_id>=0 || acc_id<(int)PJ_ARRAY_SIZE(pjsua_var.acc),
             PJ_EINVAL);

    /* Check arguments */
    PJ_ASSERT_RETURN(dest_uri, PJ_EINVAL);

    PJ_LOG(4,(THIS_FILE, "Making call with acc #%d to %.*s", acc_id,
          (int)dest_uri->slen, dest_uri->ptr));

    pj_log_push_indent();

    PJSUA_LOCK();

    /* Create sound port if none is instantiated, to check if sound device
     * can be used. But only do this with the conference bridge, as with
     * audio switchboard (i.e. APS-Direct), we can only open the sound
     * device once the correct format has been known
     */
    if (!pjsua_var.is_mswitch && pjsua_var.snd_port==NULL &&
    pjsua_var.null_snd==NULL && !pjsua_var.no_snd)
    {
    status = pjsua_set_snd_dev(pjsua_var.cap_dev, pjsua_var.play_dev);
    if (status != PJ_SUCCESS)
        goto on_error;
    }

    acc = &pjsua_var.acc[acc_id];
    if (!acc->valid) {
    pjsua_perror(THIS_FILE, "Unable to make call because account "
             "is not valid", PJ_EINVALIDOP);
    status = PJ_EINVALIDOP;
    goto on_error;
    }

    /* Find free call slot. */
    call_id = alloc_call_id();

    if (call_id == PJSUA_INVALID_ID) {
    pjsua_perror(THIS_FILE, "Error making call", PJ_ETOOMANY);
    status = PJ_ETOOMANY;
    goto on_error;
    }

    call = &pjsua_var.calls[call_id];

    /* Associate session with account */
    call->acc_id = acc_id;
    call->call_hold_type = acc->cfg.call_hold_type;

    /* Apply call setting */
    status = apply_call_setting(call, opt, NULL);
    if (status != PJ_SUCCESS) {
    pjsua_perror(THIS_FILE, "Failed to apply call setting", status);
    goto on_error;
    }

    /* Create temporary pool */
    tmp_pool = pjsua_pool_create("tmpcall10", 512, 256);

    /* Verify that destination URI is valid before calling
     * pjsua_acc_create_uac_contact, or otherwise there
     * a misleading "Invalid Contact URI" error will be printed
     * when pjsua_acc_create_uac_contact() fails.
     */
    if (1) {
    pjsip_uri *uri;
    pj_str_t dup;

    pj_strdup_with_null(tmp_pool, &dup, dest_uri);
    uri = pjsip_parse_uri(tmp_pool, dup.ptr, dup.slen, 0);

    if (uri == NULL) {
        pjsua_perror(THIS_FILE, "Unable to make call",
             PJSIP_EINVALIDREQURI);
        status = PJSIP_EINVALIDREQURI;
        goto on_error;
    }
    }

    /* Mark call start time. */
    pj_gettimeofday(&call->start_time);

    /* Reset first response time */
    call->res_time.sec = 0;

    /* Create suitable Contact header unless a Contact header has been
     * set in the account.
     */
    if (acc->contact.slen) {
    contact = acc->contact;
    } else {
    status = pjsua_acc_create_uac_contact(tmp_pool, &contact,
                          acc_id, dest_uri);
    if (status != PJ_SUCCESS) {
        pjsua_perror(THIS_FILE, "Unable to generate Contact header",
             status);
        goto on_error;
    }
    }

    /* Create outgoing dialog: */
    status = pjsip_dlg_create_uac( pjsip_ua_instance(),
                   &acc->cfg.id, &contact,
                   dest_uri, dest_uri, &dlg);
    if (status != PJ_SUCCESS) {
    pjsua_perror(THIS_FILE, "Dialog creation failed", status);
    goto on_error;
    }

    /* Increment the dialog's lock otherwise when invite session creation
     * fails the dialog will be destroyed prematurely.
     <span style="white-space:pre"> </span>*/
    pjsip_dlg_inc_lock(dlg);

    if (acc->cfg.allow_via_rewrite && acc->via_addr.host.slen > 0)
        pjsip_dlg_set_via_sent_by(dlg, &acc->via_addr, acc->via_tp);

    /* Calculate call's secure level */
    call->secure_level = get_secure_level(acc_id, dest_uri);

    /* Attach user data */
    call->user_data = user_data;

    /* Store variables required for the callback after the async
     * media transport creation is completed.
     */
    if (msg_data) {
    call->async_call.call_var.out_call.msg_data = pjsua_msg_data_clone(
                                                          dlg->pool, msg_data);
    }
    call->async_call.dlg = dlg;

    /* Temporarily increment dialog session. Without this, dialog will be
     * prematurely destroyed if dec_lock() is called on the dialog before
     * the invite session is created.
     */
    pjsip_dlg_inc_session(dlg, &pjsua_var.mod);

    /* Init media channel */
    status = pjsua_media_channel_init(call->index, PJSIP_ROLE_UAC,
                      call->secure_level, dlg->pool,
                      NULL, NULL, PJ_TRUE,
                                      &on_make_call_med_tp_complete);
    if (status == PJ_SUCCESS) {
        status = on_make_call_med_tp_complete(call->index, NULL);
        if (status != PJ_SUCCESS)
        goto on_error;
    } else if (status != PJ_EPENDING) {
    pjsua_perror(THIS_FILE, "Error initializing media channel", status);
        pjsip_dlg_dec_session(dlg, &pjsua_var.mod);
    goto on_error;
    }

    /* Done. */

    if (p_call_id)
    *p_call_id = call_id;

    pjsip_dlg_dec_lock(dlg);
    pj_pool_release(tmp_pool);
    PJSUA_UNLOCK();

    pj_log_pop_indent();

    return PJ_SUCCESS;


on_error:
    if (dlg) {
    /* This may destroy the dialog */
    pjsip_dlg_dec_lock(dlg);
    }

    if (call_id != -1) {
    reset_call(call_id);
    pjsua_media_channel_deinit(call_id);
    }

    if (tmp_pool)
    pj_pool_release(tmp_pool);
    PJSUA_UNLOCK();

    pj_log_pop_indent();
    return status;
}

通过本文的研究分析，<span style="font-family: Consolas; color: rgb(51, 51, 51);">我们了解到CSipSimple通过aidl方法实现进程间通信，从而实现了拨打电话功能。</span>