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Merge GB28181
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@winlinvip
winlinvip committed Mar 31, 2020
2 parents df6274d + 6012ac4 commit e19b927
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128 changes: 99 additions & 29 deletions trunk/conf/full.conf
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Original file line number Diff line number Diff line change
Expand Up @@ -226,64 +226,134 @@ http_server {
#############################################################################################
# the streamer cast stream from other protocol to SRS over RTMP.
# @see https://github.com/ossrs/srs/tree/develop#stream-architecture

# MPEGTS over UDP
stream_caster {
# whether stream caster is enabled.
# default: off
enabled off;
enabled on;
# the caster type of stream, the casters:
# mpegts_over_udp, MPEG-TS over UDP caster.
# rtsp, Real Time Streaming Protocol (RTSP).
# flv, FLV over HTTP by POST.
caster mpegts_over_udp;
# the output rtmp url.
# for mpegts_over_udp caster, the typically output url:
# rtmp://127.0.0.1/live/livestream
output rtmp://127.0.0.1/live/livestream;
# the listen port for stream caster.
# for mpegts_over_udp caster, listen at udp port. for example, 8935.
listen 8935;
}

# RTSP
stream_caster {
# whether stream caster is enabled.
# default: off
enabled on;
# the caster type of stream, the casters:
# rtsp, Real Time Streaming Protocol (RTSP).
caster rtsp;
# the output rtmp url.
# for rtsp caster, the typically output url:
# rtmp://127.0.0.1/[app]/[stream]
# for example, the rtsp url:
# rtsp://192.168.1.173:8544/live/livestream.sdp
# where the [app] is "live" and [stream] is "livestream", output is:
# rtmp://127.0.0.1/live/livestream
# for flv caster, the typically output url:
# rtmp://127.0.0.1/[app]/[stream]
# for example, POST to url:
# http://127.0.0.1:8936/live/livestream.flv
# where the [app] is "live" and [stream] is "livestream", output is:
# rtmp://127.0.0.1/live/livestream
output rtmp://127.0.0.1/live/livestream;
output rtmp://127.0.0.1/[app]/[stream];
# the listen port for stream caster.
# for mpegts_over_udp caster, listen at udp port. for example, 8935.
# for rtsp caster, listen at tcp port. for example, 554.
# for flv caster, listen at tcp port. for example, 8936.
# TODO: support listen at <[ip:]port>
listen 8935;
listen 554;
# for the rtsp caster, the rtp server local port over udp,
# which reply the rtsp setup request message, the port will be used:
# [rtp_port_min, rtp_port_max)
rtp_port_min 57200;
rtp_port_max 57300;
}

# FLV
stream_caster {
enabled off;
caster mpegts_over_udp;
output rtmp://127.0.0.1/live/livestream;
listen 8935;
}
stream_caster {
enabled off;
caster rtsp;
output rtmp://127.0.0.1/[app]/[stream];
listen 554;
rtp_port_min 57200;
rtp_port_max 57300;
}
stream_caster {
enabled off;
# whether stream caster is enabled.
# default: off
enabled on;
# the caster type of stream, the casters:
# flv, FLV over HTTP by POST.
caster flv;
# the output rtmp url.
# for flv caster, the typically output url:
# rtmp://127.0.0.1/[app]/[stream]
# for example, POST to url:
# http://127.0.0.1:8936/live/livestream.flv
# where the [app] is "live" and [stream] is "livestream", output is:
# rtmp://127.0.0.1/live/livestream
output rtmp://127.0.0.1/[app]/[stream];
# the listen port for stream caster.
# for flv caster, listen at tcp port. for example, 8936.
listen 8936;
}

# GB28181
stream_caster {
# whether stream caster is enabled.
# default: off
enabled on;
# the caster type of stream, the casters:
# gb28181, Push GB28181 to SRS.
caster gb28181;
# the output rtmp url.
# for gb28181 caster, the typically output url:
# rtmp://127.0.0.1/live/[stream]
# where the [stream] is the VideoChannelCodecID.
output rtmp://127.0.0.1/live/[stream];
# the listen port for stream caster.
# for gb28181 caster, listen at udp port. for example, 9000.
# @remark We can bundle all gb28181 to this port, to reuse this port.
# User can choose to bundle port in API port_mode or SIP invite_port_fixed.
listen 9000;
# If not bundle ports, use specified ports for each stream.
rtp_port_min 58200;
rtp_port_max 58300;
# Whether wait for keyframe then forward to RTMP.
wait_keyframe off;
# Max timeout in seconds for RTP stream, if timeout, RTCP bye and close stream.
# default: 30
rtp_idle_timeout 30;
# Whether has audio.
# @remark Flash/RTMP only supports 11025 22050 44100 sample rate, if not the audio may corrupt.
# default: off
audio_enable off;
# The exposed IP to receive media stream.
host 192.168.1.3;

sip {
# Whether enable embeded SIP server.
# default: on
enabled on;
# The SIP listen port.
# default: 5060
listen 5060;
# The SIP server ID.
# default: 34020000002000000001
serial 34020000002000000001;
# The SIP server domain.
# default: 3402000000
realm 3402000000;
# The SIP ACK response timeout in seconds.
# default: 30
ack_timeout 30;
# The keepalive timeout in seconds.
# default: 120
keepalive_timeout 120;
# Whether print SIP logs.
print_sip_message off;
# Whether play immediately after registered.
# default: on
auto_play on;
# Whether bundle media stream port.
# default: on
invite_port_fixed on;
}
}

#############################################################################################
# SRT server section
#############################################################################################
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99 changes: 99 additions & 0 deletions trunk/conf/push.gb28181.conf
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@@ -0,0 +1,99 @@
# push gb28281 stream to SRS.

listen 1935;
max_connections 1000;
daemon off;
srs_log_tank console;

http_api {
enabled on;
listen 1985;
}

stream_caster {
enabled on;
caster gb28181;

# 转发流到rtmp服务器地址与端口
# TODO: https://github.com/ossrs/srs/pull/1679/files#r400875104
# [stream] is VideoChannelCodecID(视频通道编码ID)
output 127.0.0.1:1935;

# 接收设备端rtp流的多路复用端口
listen 9000;

# 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;

# 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
# on:转发音频
# off:不转发音频，只有视频
# *注意*!!!:flv 只支持11025 22050 44100 三种
# 如果设备端没有三种中任何一个，转发时为自动选择一种格式
# 同时也会将adts的头封装在flv aac raw数据中
# 这样的话播放器为自动通过adts头自动选择采样频率
# 像ffplay, vlc都可以，但是flash是没有声音，
# 因为flash,只支持11025 22050 44100
audio_enable off;

# 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
# TODO: https://github.com/ossrs/srs/pull/1679/files#r400917594
host 192.168.1.27;

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 5060;

# SIP server ID(SIP服务器ID).
# 设备端配置编号需要与该值一致，否则无法注册
serial 34020000002000000001;

# SIP server domain(SIP服务器域)
realm 3402000000;

# 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;

# 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 120;

# 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
# TODO: https://github.com/ossrs/srs/pull/1679/files#r400929300
print_sip_message off;

# 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__ {
}
4 changes: 2 additions & 2 deletions trunk/configure
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Original file line number Diff line number Diff line change
Expand Up @@ -224,7 +224,7 @@ MODULE_DEPENDS=("CORE" "KERNEL")
ModuleLibIncs=(${SRS_OBJS_DIR} ${LibSSLRoot})
MODULE_FILES=("srs_protocol_amf0" "srs_protocol_io" "srs_rtmp_stack"
"srs_rtmp_handshake" "srs_protocol_utility" "srs_rtmp_msg_array" "srs_protocol_stream"
"srs_raw_avc" "srs_rtsp_stack" "srs_http_stack" "srs_protocol_kbps" "srs_protocol_json"
"srs_raw_avc" "srs_rtsp_stack" "srs_sip_stack" "srs_http_stack" "srs_protocol_kbps" "srs_protocol_json"
"srs_protocol_format")
if [[ $SRS_RTC == YES ]]; then
MODULE_FILES+=("srs_stun_stack")
Expand Down Expand Up @@ -277,7 +277,7 @@ 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_coworkers" "srs_app_hybrid" "srs_app_gb28181" "srs_app_gb28181_sip")
if [[ $SRS_RTC == YES ]]; then
MODULE_FILES+=("srs_app_rtc" "srs_app_rtc_conn" "srs_app_dtls" "srs_app_audio_recode" "srs_app_sdp")
fi
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