【网络通信 -- SIP 电话】PJSUA API 学习与客户端开发 -- 实现简单的通话功能

【网络通信 -- SIP 电话】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-0.5.10.2

include $(PJBASE)/build.mak

CC      = $(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.cpp
all: myapp

myapp: 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 
 *
 * 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 

#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

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。

【1】Building Application using PJSIP with GNU Tools

【2】PJSIP学习笔记——从simple_pjsua.c示例程序了解PJSUA-LIB的基本使用流程

【3】PJSUA API - High Level Softphone API

文档资料与软件包

【1】PJSUA开发文档中文版