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【网络通信 -- SIP 电话】项目实战记录 -- PJSUA API 学习与客户端开发(实现简单的通话功能)

【1】基于 PJSUA 库的 Linux C 开发环境搭建与 Makefile 编写

#Modify this to point to the PJSIP location.# PJBASE 指定 pjproject 的源代码路径PJBASE=/home/myself/pjproject-0.5.10.2include $(PJBASE)/build.makCC      = $(PJ_CC)LDFLAGS = $(PJ_LDFLAGS)LDLIBS  = $(PJ_LDLIBS)CFLAGS  = $(PJ_CFLAGS)CPPFLAGS= ${CFLAGS}# If your application is in a file named myapp.cpp or myapp.c# this is the line you will need to build the binary.# 假定用于编写的 SIPSUA API 测试代码为 myapp.cppall: myappmyapp: myapp.cpp        $(CC) -o $@ $< $(CPPFLAGS) $(LDFLAGS) $(LDLIBS)clean:        rm -f myapp.o myapp

【2】基于 PJSUA API 实现的简单通话功能代码示例

PJSIP 官方提供的示例代码如下，该示例程序展示了使用 PJSUA API 编写客户端实现一次通话的全过程。

/* $Id: simple_pjsua.c 3553 2011-05-05 06:14:19Z nanang $ *//*  * Copyright (C) 2008-2011 Teluu Inc. (http://www.teluu.com) * Copyright (C) 2003-2008 Benny Prijono <benny@prijono.org> * * This program 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 2 of the License, or * (at your option) any later version. * * This program 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 this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA  *//** * simple_pjsua.c * * This is a very simple but fully featured SIP user agent, with the  * following capabilities: *  - SIP registration *  - Making and receiving call *  - Audio/media to sound device. * * Usage: *  - To make outgoing call, start simple_pjsua with the URL of remote *    destination to contact. *    E.g.: *	 simpleua sip:user@remote * *  - Incoming calls will automatically be answered with 200. * * This program will quit once it has completed a single call. */#include <pjsua-lib/pjsua.h>#define THIS_FILE	"APP"#define SIP_DOMAIN	"example.com"#define SIP_USER	"alice"#define SIP_PASSWD	"secret"/* Callback called by the library upon receiving incoming call */static void on_incoming_call(pjsua_acc_id acc_id, pjsua_call_id call_id,			     pjsip_rx_data *rdata){    pjsua_call_info ci;    PJ_UNUSED_ARG(acc_id);    PJ_UNUSED_ARG(rdata);    pjsua_call_get_info(call_id, &ci);    PJ_LOG(3,(THIS_FILE, "Incoming call from %.*s!!",			 (int)ci.remote_info.slen,			 ci.remote_info.ptr));    /* Automatically answer incoming calls with 200/OK */    pjsua_call_answer(call_id, 200, NULL, NULL);}/* Callback called by the library when call's state has changed */static void on_call_state(pjsua_call_id call_id, pjsip_event *e){    pjsua_call_info ci;    PJ_UNUSED_ARG(e);    pjsua_call_get_info(call_id, &ci);    PJ_LOG(3,(THIS_FILE, "Call %d state=%.*s", call_id,			 (int)ci.state_text.slen,			 ci.state_text.ptr));}/* Callback called by the library when call's media state has changed */static void on_call_media_state(pjsua_call_id call_id){    pjsua_call_info ci;    pjsua_call_get_info(call_id, &ci);    if (ci.media_status == PJSUA_CALL_MEDIA_ACTIVE) {	// When media is active, connect call to sound device.	pjsua_conf_connect(ci.conf_slot, 0);	pjsua_conf_connect(0, ci.conf_slot);    }}/* Display error and exit application */static void error_exit(const char *title, pj_status_t status){    pjsua_perror(THIS_FILE, title, status);    pjsua_destroy();    exit(1);}/* * main() * * argv[1] may contain URL to call. */int main(int argc, char *argv[]){    pjsua_acc_id acc_id;    pj_status_t status;    /* Create pjsua first! */    status = pjsua_create();    if (status != PJ_SUCCESS) error_exit("Error in pjsua_create()", status);    /* If argument is specified, it's got to be a valid SIP URL */    if (argc > 1) {	status = pjsua_verify_url(argv[1]);	if (status != PJ_SUCCESS) error_exit("Invalid URL in argv", status);    }    /* Init pjsua */    {	pjsua_config cfg;	pjsua_logging_config log_cfg;	pjsua_config_default(&cfg);	cfg.cb.on_incoming_call = &on_incoming_call;	cfg.cb.on_call_media_state = &on_call_media_state;	cfg.cb.on_call_state = &on_call_state;	pjsua_logging_config_default(&log_cfg);	log_cfg.console_level = 4;	status = pjsua_init(&cfg, &log_cfg, NULL);	if (status != PJ_SUCCESS) error_exit("Error in pjsua_init()", status);    }    /* Add UDP transport. */    {	pjsua_transport_config cfg;	pjsua_transport_config_default(&cfg);	cfg.port = 5060;	status = pjsua_transport_create(PJSIP_TRANSPORT_UDP, &cfg, NULL);	if (status != PJ_SUCCESS) error_exit("Error creating transport", status);    }    /* Initialization is done, now start pjsua */    status = pjsua_start();    if (status != PJ_SUCCESS) error_exit("Error starting pjsua", status);    /* Register to SIP server by creating SIP account. */    {	pjsua_acc_config cfg;	pjsua_acc_config_default(&cfg);	cfg.id = pj_str("sip:" SIP_USER "@" SIP_DOMAIN);	cfg.reg_uri = pj_str("sip:" SIP_DOMAIN);	cfg.cred_count = 1;	cfg.cred_info[0].realm = pj_str(SIP_DOMAIN);	cfg.cred_info[0].scheme = pj_str("digest");	cfg.cred_info[0].username = pj_str(SIP_USER);	cfg.cred_info[0].data_type = PJSIP_CRED_DATA_PLAIN_PASSWD;	cfg.cred_info[0].data = pj_str(SIP_PASSWD);	status = pjsua_acc_add(&cfg, PJ_TRUE, &acc_id);	if (status != PJ_SUCCESS) error_exit("Error adding account", status);    }    /* If URL is specified, make call to the URL. */    if (argc > 1) {	pj_str_t uri = pj_str(argv[1]);	status = pjsua_call_make_call(acc_id, &uri, 0, NULL, NULL, NULL);	if (status != PJ_SUCCESS) error_exit("Error making call", status);    }    /* Wait until user press "q" to quit. */    for (;;) {	char option[10];	puts("Press 'h' to hangup all calls, 'q' to quit");	if (fgets(option, sizeof(option), stdin) == NULL) {	    puts("EOF while reading stdin, will quit now..");	    break;	}	if (option[0] == 'q')	    break;	if (option[0] == 'h')	    pjsua_call_hangup_all();    }    /* Destroy pjsua */    pjsua_destroy();    return 0;}

【3】基于 PJSUA API 的通话 API 调用总结

上述示例代码的调用流程如下图所示，展示了整个通话的建立过程以及对应的 API。

【4】基于 PJSUA API 的通话流程分析

从通话日志中分析通话的流程，以及回调函数的调用时机

【5】基于 PJSUA 库的 SIP 电话注册状态监控

在 pjsua_config 结构体中添加监听 SIP 注册状态的回调函数pjsua_config cfg;cfg.cb.on_reg_state2 = &on_reg_state;实现 on_reg_state 以监听 SIP 电话注册状态void on_reg_state(pjsua_acc_id acc_id,  pjsua_reg_info *info){    if(info->cbparam->code != 200)    {        error_exit("Register Error");    }    if(info->cbparam->code == 200)    {        sip_phone_make_call(acc_id, target_sip_phone_address_val);    }}

参考与致谢

本博客为博主的学习实践总结，并参考了众多博主的博文，在此表示感谢，博主若有不足之处，请批评指正。

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