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 public 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;        }    };

   @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 {                service.makeCallWithOptions(toCall, accountToUse.intValue(), b);            } 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);                }            });        }    }    

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 {                    pjService.makeCall(callee, accountId, options);                }            });        }

/**
      * 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

service.makeCallWithOptions(toCall, accountToUse.intValue(), b);调用接口中的方法，完成IPC方法。

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

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);                        }                    }                }            }                        int status = pjsua.call_make_call(pjsuaAccId, uri, cs, userData, msgData, callId);            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;    }

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) {    return pjsuaJNI.call_make_call(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.     */    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，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。