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);
}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;
}
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。
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。
/* ----------------------------------------------------------------------------
* 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);
...
}
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。
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