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refactor gb28181, supporting SIP server enable, multiplex, API interface #1679

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Mar 31, 2020
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refactor gb28181, supporting SIP server enable, multiplex, API interface #1679

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99 changes: 62 additions & 37 deletions trunk/conf/push.gb28181.conf
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Original file line number Diff line number Diff line change
Expand Up @@ -6,28 +6,35 @@ daemon off;
pid ./objs/srs28181.pid;
srs_log_file ./objs/srs28181.log;
srs_log_tank console;

http_api {
enabled on;
listen 1985;
}

stream_caster {
enabled on;
caster gb28181;

#rtmp输出地址，可以参数化
#[stream] 代表客户端sip设备编号
#[timestamp] 时间戳
output rtmp://127.0.0.1/live/[stream];
#sip监听udp端口
listen 15060;
#转发流到rtmp服务器地址与端口
output 127.0.0.1:1935;
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@winlinvip winlinvip Mar 31, 2020 •
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需要改成rtmp地址，这样用户可以指定app。' translates to:
'Need to change to rtmp address so that users can specify the app.

TRANS_BY_GPT3

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@winlinvip winlinvip Mar 31, 2020 •
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[stream] is the video channel encoding ID.

TRANS_BY_GPT3


#服务器主机号，可以域名或ip地址
#也就是设备端将媒体发送的地址，如果是服务器是内外网
#需要写外网地址
host 192.168.1.27;
#接收设备端rtp流的多路复用端口
listen 9000;

#服务器端编号
#设备端配置编号需要与该值一致，否则无法注册
serial 34020000002000000001;
#rtp接收监听端口范围，最小值
rtp_port_min 58200;
#rtp接收监听端口范围，最大值
rtp_port_max 58300;

#服务器端域
realm 3402000000;
#是否等待关键帧之后，再转发，
#off:不需等待，直接转发
#on:等第一个关键帧后，再转发
wait_keyframe off;

#rtp包空闲等待时间，如果指定时间没有收到任何包
#rtp监听连接自动停止，发送BYE命令
rtp_idle_timeout 30;

#是否转发音频流
#目前只支持aac格式，所以需要设备支持aac格式
Expand All @@ -41,32 +48,50 @@ stream_caster {
#因为flash,只支持11025 22050 44100
audio_enable on;

#服务端发送ack后，接收回应的超时时间，单位为秒
#如果指定时间没有回应，认为失败
ack_timeout 30;
#服务器主机号，可以域名或ip地址
#也就是设备端将媒体发送的地址，如果是服务器是内外网
#需要写外网地址，
#调用api创建stream session时返回ip地址也是host
host 192.168.1.27;
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@winlinvip winlinvip Mar 31, 2020 •
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This place needs support for a more comprehensive way of obtaining IP, refer to #307 (comment).

TRANS_BY_GPT3


#设备心跳维持时间，如果指定时间内(秒）没有接收一个心跳
#认为设备离线
keepalive_timeout 30;
sip {
#是否启用srs内部sip信令
#为on信令走srs, off 只转发ps流
enabled on;

#sip监听udp端口
listen 15060;

#服务器端编号
#设备端配置编号需要与该值一致，否则无法注册
serial 34020000002000000001;

#是否等待关键帧之后，再转发，
#off:不需等待，直接转发
#on:等第一个关键帧后，再转发
wait_keyframe off;
#服务器端域
realm 3402000000;

#日志打印是否打印sip信息
#off:不打印
#on:打印接收或发送sip命令信息
print_sip_message off;

#rtp包空闲等待时间，如果指定时间没有收到任何包
#rtp监听连接自动停止，发送BYE命令
rtp_idle_timeout 30;
#服务端发送ack后，接收回应的超时时间，单位为秒
#如果指定时间没有回应，认为失败
ack_timeout 30;

#rtp接收监听端口范围，最小值
rtp_port_min 58200;
#rtp接收监听端口范围，最大值
rtp_port_max 58300;
#设备心跳维持时间，如果指定时间内(秒）没有接收一个心跳
#认为设备离线
keepalive_timeout 30;

#日志打印是否打印sip信息
#off:不打印
#on:打印接收或发送sip命令信息
print_sip_message off;
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@winlinvip winlinvip Mar 31, 2020 •
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Change to log-level, srs_info level log, enable info log.

TRANS_BY_GPT3


#注册之后是否自动给设备端发送invite
#on: 是 off 不是，需要通过api控制
auto_play on;

#设备将流发送的端口，是否固定
#on 发送流到多路复用端口 如9000
#off 自动从rtp_mix_port - rtp_max_port 之间的值中
#选一个可以用的端口
invite_port_fixed on;
}
}
vhost __defaultVhost__ {
}
2 changes: 1 addition & 1 deletion trunk/configure
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Expand Up @@ -257,7 +257,7 @@ if [ $SRS_EXPORT_LIBRTMP_PROJECT = NO ]; then
"srs_app_mpegts_udp" "srs_app_rtsp" "srs_app_listener" "srs_app_async_call"
"srs_app_caster_flv" "srs_app_process" "srs_app_ng_exec"
"srs_app_hourglass" "srs_app_dash" "srs_app_fragment" "srs_app_dvr"
"srs_app_coworkers" "srs_app_hybrid" "srs_app_gb28181")
"srs_app_coworkers" "srs_app_hybrid" "srs_app_gb28181" "srs_app_gb28181_sip")
DEFINES=""
# add each modules for app
for SRS_MODULE in ${SRS_MODULES[*]}; do
Expand Down
144 changes: 137 additions & 7 deletions trunk/src/app/srs_app_config.cpp
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Expand Up @@ -2160,6 +2160,10 @@ srs_error_t SrsConfig::global_to_json(SrsJsonObject* obj)
sobj->set(sdir->name, sdir->dumps_arg0_to_str());
} else if (sdir->name == "print_sip_message") {
sobj->set(sdir->name, sdir->dumps_arg0_to_str());
} else if (sdir->name == "invite_port_fixed") {
sobj->set(sdir->name, sdir->dumps_arg0_to_str());
} else if (sdir->name == "auto_play") {
sobj->set(sdir->name, sdir->dumps_arg0_to_str());
}

}
Expand Down Expand Up @@ -3672,12 +3676,24 @@ srs_error_t SrsConfig::check_normal_config()
string n = conf->name;
if (n != "enabled" && n != "caster" && n != "output"
&& n != "listen" && n != "rtp_port_min" && n != "rtp_port_max"
&& n != "rtp_idle_timeout" && n != "ack_timeout" && n != "keepalive_timeout"
&& n != "host" && n != "serial" && n != "realm"
&& n != "rtp_idle_timeout" && n != "sip"
&& n != "audio_enable" && n != "wait_keyframe"
&& n != "print_sip_message") {
&& n != "host") {
return srs_error_new(ERROR_SYSTEM_CONFIG_INVALID, "illegal stream_caster.%s", n.c_str());
}

if (n == "sip") {
for (int j = 0; j < (int)conf->directives.size(); j++) {
string m = conf->at(j)->name;
if (m != "enabled" && m != "listen"
&& m != "ack_timeout" && m != "keepalive_timeout"
&& m != "host" && m != "serial" && m != "realm"
&& m != "print_sip_message" && m != "auto_play"
&& m != "invite_port_fixed") {
return srs_error_new(ERROR_SYSTEM_CONFIG_INVALID, "illegal stream_caster.%s", m.c_str());
}
}
}
}
}

Expand Down Expand Up @@ -4207,15 +4223,15 @@ int SrsConfig::get_stream_caster_rtp_port_max(SrsConfDirective* conf)
return ::atoi(conf->arg0().c_str());
}

int SrsConfig::get_stream_caster_gb28181_rtp_ide_timeout(SrsConfDirective* conf)
int SrsConfig::get_stream_caster_gb28181_rtp_idle_timeout(SrsConfDirective* conf)
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@winlinvip winlinvip Mar 31, 2020 •
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Please use srs_utime_t uniformly, which is defined in time, and there are also related macro definitions for converting time. Please make sure to change it together in other places.

TRANS_BY_GPT3

{
static int DEFAULT = 30;

if (!conf) {
return DEFAULT;
}

conf = conf->get("rtp_ide_timeout");
conf = conf->get("rtp_idle_timeout");
if (!conf || conf->arg0().empty()) {
return DEFAULT;
}
Expand All @@ -4230,6 +4246,11 @@ int SrsConfig::get_stream_caster_gb28181_ack_timeout(SrsConfDirective* conf)
if (!conf) {
return DEFAULT;
}

conf = conf->get("sip");
if (!conf) {
return DEFAULT;
}

conf = conf->get("ack_timeout");
if (!conf || conf->arg0().empty()) {
Expand All @@ -4246,6 +4267,11 @@ int SrsConfig::get_stream_caster_gb28181_keepalive_timeout(SrsConfDirective* con
if (!conf) {
return DEFAULT;
}

conf = conf->get("sip");
if (!conf) {
return DEFAULT;
}

conf = conf->get("keepalive_timeout");
if (!conf || conf->arg0().empty()) {
Expand All @@ -4257,12 +4283,12 @@ int SrsConfig::get_stream_caster_gb28181_keepalive_timeout(SrsConfDirective* con

string SrsConfig::get_stream_caster_gb28181_host(SrsConfDirective* conf)
{
static string DEFAULT = "127.0.0.1";
static string DEFAULT = "";

if (!conf) {
return DEFAULT;
}

conf = conf->get("host");
if (!conf || conf->arg0().empty()) {
return DEFAULT;
Expand All @@ -4278,6 +4304,11 @@ string SrsConfig::get_stream_caster_gb28181_serial(SrsConfDirective* conf)
if (!conf) {
return DEFAULT;
}

conf = conf->get("sip");
if (!conf) {
return DEFAULT;
}

conf = conf->get("serial");
if (!conf || conf->arg0().empty()) {
Expand All @@ -4294,6 +4325,11 @@ string SrsConfig::get_stream_caster_gb28181_realm(SrsConfDirective* conf)
if (!conf) {
return DEFAULT;
}

conf = conf->get("sip");
if (!conf) {
return DEFAULT;
}

conf = conf->get("realm");
if (!conf || conf->arg0().empty()) {
Expand Down Expand Up @@ -4326,6 +4362,11 @@ bool SrsConfig::get_stream_caster_gb28181_print_sip_message(SrsConfDirective* co
if (!conf) {
return DEFAULT;
}

conf = conf->get("sip");
if (!conf) {
return DEFAULT;
}

conf = conf->get("print_sip_message");
if (!conf || conf->arg0().empty()) {
Expand All @@ -4351,6 +4392,95 @@ bool SrsConfig::get_stream_caster_gb28181_wait_keyframe(SrsConfDirective* conf)
return SRS_CONF_PERFER_FALSE(conf->arg0());
}

bool SrsConfig::get_stream_caster_gb28181_sip_enable(SrsConfDirective* conf)
{
static bool DEFAULT = false;

if (!conf) {
return DEFAULT;
}

conf = conf->get("sip");
if (!conf) {
return DEFAULT;
}

conf = conf->get("enabled");
if (!conf || conf->arg0().empty()) {
return DEFAULT;
}

return SRS_CONF_PERFER_FALSE(conf->arg0());

}

bool SrsConfig::get_stream_caster_gb28181_sip_auto_play(SrsConfDirective* conf)
{
static bool DEFAULT = false;

if (!conf) {
return DEFAULT;
}

conf = conf->get("sip");
if (!conf) {
return DEFAULT;
}

conf = conf->get("auto_play");
if (!conf || conf->arg0().empty()) {
return DEFAULT;
}

return SRS_CONF_PERFER_FALSE(conf->arg0());

}

int SrsConfig::get_stream_caster_gb28181_sip_listen(SrsConfDirective* conf)
{
static int DEFAULT = 0;

if (!conf) {
return DEFAULT;
}

conf = conf->get("sip");
if (!conf) {
return DEFAULT;
}

conf = conf->get("listen");
if (!conf || conf->arg0().empty()) {
return DEFAULT;
}

return ::atoi(conf->arg0().c_str());

}

bool SrsConfig::get_stream_caster_gb28181_sip_invite_port_fixed(SrsConfDirective* conf)
{
static bool DEFAULT = true;

if (!conf) {
return DEFAULT;
}

conf = conf->get("sip");
if (!conf) {
return DEFAULT;
}

conf = conf->get("invite_port_fixed");
if (!conf || conf->arg0().empty()) {
return DEFAULT;
}

return SRS_CONF_PERFER_FALSE(conf->arg0());

}


SrsConfDirective* SrsConfig::get_vhost(string vhost, bool try_default_vhost)
{
srs_assert(root);
Expand Down
7 changes: 6 additions & 1 deletion trunk/src/app/srs_app_config.hpp
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Expand Up @@ -486,7 +486,7 @@ class SrsConfig
// Get the max udp port for rtp of stream caster rtsp.
virtual int get_stream_caster_rtp_port_max(SrsConfDirective* conf);

virtual int get_stream_caster_gb28181_rtp_ide_timeout(SrsConfDirective* conf);
virtual int get_stream_caster_gb28181_rtp_idle_timeout(SrsConfDirective* conf);
virtual int get_stream_caster_gb28181_ack_timeout(SrsConfDirective* conf);
virtual int get_stream_caster_gb28181_keepalive_timeout(SrsConfDirective* conf);
virtual bool get_stream_caster_gb28181_audio_enable(SrsConfDirective* conf);
Expand All @@ -495,6 +495,11 @@ class SrsConfig
virtual std::string get_stream_caster_gb28181_realm(SrsConfDirective* conf);
virtual bool get_stream_caster_gb28181_print_sip_message(SrsConfDirective* conf);
virtual bool get_stream_caster_gb28181_wait_keyframe(SrsConfDirective* conf);
virtual bool get_stream_caster_gb28181_sip_enable(SrsConfDirective* conf);
virtual bool get_stream_caster_gb28181_sip_auto_play(SrsConfDirective* conf);
virtual int get_stream_caster_gb28181_sip_listen(SrsConfDirective* conf);
virtual bool get_stream_caster_gb28181_sip_invite_port_fixed(SrsConfDirective* conf);


// vhost specified section
public:
Expand Down
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