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：
ISipService.javaprivate 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; } };
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\pjsipPjSipService.javapublic 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\pjsuapjsua.javapackage 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\pjsuapjsuaJNI.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.cPJ_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。