pjsip视频通信开发(底层实现)之用户注册（1）

一、PJSIP简介

对于pjsip的介绍可以看http://www.cnblogs.com/my_life/articles/2175462.html 文章，里面详细介绍了它的组成框架以及各部份的组成介绍，我把官网中提供的一个pjsip的整体框架图贴到这里

二、simple_pjsua.c分析

我今天要实现的是UA这部份内容，主要作用可以查看http://www.cnblogs.com/flyfish10000/category/268759.html 里面有作者一系列关于UA的介绍，这里我要实现的是帐号的注册，要做这个我们可以看一下官网一个提供的例子simple_pjsua.c
(pjsip-apps/src/samples/)，它的内容如下：

/**
 * 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;
}

我们这里可以分析一下它的代码：

1、一开始是回调使用的函数，例如on_incoming_call当来电话的时候，pjsip会自动去调用你写的这个函数，前提是你在初始化pjsua的时候设置了on_incoming_call = &on_incoming_call，

2、error_exit退出应用所需要的操作

3、main函数：

(1)pjsua_create()创建pjsua的第一步，如果是要打电话要确认URL是否是正确的pjsua_verify_url

(2)初始化pjsua，pjsua_config_default(&cfg)来初始化配置，然后设置一些回调函数，设置日志，最后初始化pjsua_init(&cfg, &log_cfg, NULL);

(3)创建UDP的传输，设置端口号

(4)接下来就是启动pjsua，通过pjsua_start();

(5)创建账户，这个是这篇文章的主要内容，pjsua_acc_config_default初始化配置，然后设置相关的内容，id对应这url，realm是服务器的域名，还有密码和用户名，最后调用 pjsua_acc_add(&cfg, PJ_TRUE, &acc_id);来实现帐号的注册。

4、打电话，上面也提到过，你打电话的话需要验证URL是否正确的 pjsua_verify_url，然后调用pjsua_call_make_call来打电话。

5、挂电话，调用 pjsua_call_hangup_all();

6、最后销毁，pjsua_destroy();

三、PJSIP帐号注册开发

下一章中介绍