项目要求设备要支持GB28181,发现网上资源比较少,记录一下自己开发过程。
首先GB28181是基于sip协议进行传输的一项协议,网上的sip开源库不少,这里我选择了PJSIP作为sip的开发用的库。
0.流程图
引用 CSDN 另一位博主的大概思路,我也将大致按照这个流程进行开发:
图引自https://blog.csdn.net/Zhu__/article/details/90967400
1.交叉编译PJSIP
PJSIP的下载地址:https://www.pjsip.org/
我用的是12.10版,下载好后解压缩到linux虚拟机中,根据官方说明安装以下方法安装
./configure --host=aarch64-linux-gnu --enable-shared --enable-ffmpeg --with-ffmpeg=/opt/ --prefix=/home/liu/workspace/pjsip
make dep
make
make install
说明下
–with-ffmpeg : ffmpeg的安装目录
–prefix : 指定编译后保存生产库的路径
–host : 指定交叉编译器
在make dep
的时候出现了报错 fatal error: emmintrin.h: No such file or directory
根据此解决办法将第一步配置修改为
./configure --host=aarch64-linux-gnu --enable-shared --enable-ffmpeg --with-ffmpeg=/opt/ --disable-libwebrtc --prefix=/home/liu/workspace/pjsip
安装完成,接下来先写个程序让设备能注册到SIP服务器上。
2.部署SIP服务器
测试的话肯定得有个支持GB28181的服务器,我用的是青柿的liveGBS,安装步骤十分简单,官网也有详细教程,过程就不说明了。官网链接:https://www.liveqing.com/docs/download/LiveGBS.html#%E7%9B%B8%E5%85%B3%E4%BB%8B%E7%BB%8D
3.simple_pjsua.c
PJSIP有好几个API版本,我选择了封装较高的pjsua。API的说明在http://www.pjsip.org/docs/latest-2/pjlib/docs/html/index.htm。
要了解pjsua的使用,simple_pjsua.c是一个很好的例子,程序在下载目录下的 pjsip-apps/src/samples/simple_pjsua.c
#include <pjsua-lib/pjsua.h>
#define THIS_FILE "APP"
#define SIP_DOMAIN "example.com"
#define SIP_USER "alice"
#define SIP_PASSWD "secret"
//来电回调函数
/* 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);
// 获得呼叫信息
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));
// 自动应答呼叫
/* Automatically answer incoming calls with 200/OK */
pjsua_call_answer(call_id, 200, NULL, NULL);
}
//呼叫状态改变的回调函数
/* 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));
}
//媒体状态改变的回调函数
/* 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);
// 当媒体为激活时,连接呼叫和声音设备
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;
// 创建PJSIP
/* Create pjsua first! */
status = pjsua_create();
if (status != PJ_SUCCESS) error_exit("Error in pjsua_create()", status);
// 校验被叫SIP地址是否正确
/* 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);
}
// 初始化PJSUA,设置回调函数
/* 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);
}
// 创建PJSIP的传输端口
/* 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);
}
// 启动PJSIP
/* Initialization is done, now start pjsua */
status = pjsua_start();
if (status != PJ_SUCCESS) error_exit("Error starting pjsua", status);
// 设置SIP用户帐号,并注册到sip服务器
/* 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);
}
// 发起一个呼叫
/* 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);
}
// 循环等待
/* 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;
}
先写个makefile看能编译通过吗
make失败,提示缺少库,查了下官方文档里有对makefile的说明
因为我的虚拟机没有安装pkg-config工具,所以安装方法2修改makefile文件,增加以下内容
include /home/liu/Downloads/pjproject-2.10/build.mak
CC = $(PJ_CC)
LDFLAGS = $(PJ_LDFLAGS)
LDLIBS = $(PJ_LDLIBS)
CFLAGS = $(PJ_CFLAGS)
CPPFLAGS= ${CFLAGS}
$(CPPFLAGS) $(LDFLAGS) $(LDLIBS)
编译通过,把编译生成的文件和动态库发到开发板上就能运行了
4.注册到sip服务器
打开青柿的sip服务器,按照基础配置页面的内容将simple_pjsua.c #define的内容做如下修改
#define SIP_DOMAIN "3402000000" //和服务器的SIP域相同
#define SIP_USER "34020000002000000999" //20位国标设备ID,自行分配,不可重复
#define SIP_PASSWD "12345678" //和统一接入密码相同
还要在代码中添加服务器的IP地址,方法为在设置账户信息,注册服务器处添加以下内容
pjsua_acc_config cfg;
pjsua_acc_config_default(&cfg);
cfg.proxy_cnt=1; //添加一个代理服务器
cfg.proxy[0] = pj_str("sip:192.168.31.103:15060"); //服务器的ip和端口号
编译运行可以在服务器上看到设备已经成功注册了。
查看日志可以看见服务器向我们发了一个查询设备的信息的请求等待回复,至于怎么处理这个请求我们下章再说。
14:30:33.688 pjsua_core.c .RX 526 bytes Request msg MESSAGE/cseq=4 (rdata0x7f7c001928) from UDP 192.168.31.103:15060:
MESSAGE sip:34020000002000000999@3402000000 SIP/2.0
Via: SIP/2.0/UDP 192.168.31.103:15060;rport;branch=z9hG4bK875664499
From: <sip:34020000002000000001@3402000000>;tag=993664499
To: <sip:34020000002000000999@3402000000>
Call-ID: 342664499
CSeq: 4 MESSAGE
Content-Type: Application/MANSCDP+xml
Max-Forwards: 70
User-Agent: LiveGBS v201123
Content-Length: 156
<?xml version="1.0" encoding="GB2312"?>
<Query>
<CmdType>Catalog</CmdType>
<SN>12664499</SN>
<DeviceID>34020000002000000999</DeviceID>
</Query>
--end msg--