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Refine code
This commit is contained in:
winlin
parent
a342f460e7
commit
6012ac4eb0
10 changed files with 167 additions and 96 deletions
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@ -226,64 +226,134 @@ http_server {
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#############################################################################################
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# the streamer cast stream from other protocol to SRS over RTMP.
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# @see https://github.com/ossrs/srs/tree/develop#stream-architecture
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# MPEGTS over UDP
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stream_caster {
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# whether stream caster is enabled.
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# default: off
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enabled off;
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enabled on;
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# the caster type of stream, the casters:
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# mpegts_over_udp, MPEG-TS over UDP caster.
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# rtsp, Real Time Streaming Protocol (RTSP).
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# flv, FLV over HTTP by POST.
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caster mpegts_over_udp;
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# the output rtmp url.
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# for mpegts_over_udp caster, the typically output url:
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# rtmp://127.0.0.1/live/livestream
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output rtmp://127.0.0.1/live/livestream;
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# the listen port for stream caster.
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# for mpegts_over_udp caster, listen at udp port. for example, 8935.
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listen 8935;
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}
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# RTSP
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stream_caster {
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# whether stream caster is enabled.
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# default: off
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enabled on;
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# the caster type of stream, the casters:
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# rtsp, Real Time Streaming Protocol (RTSP).
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caster rtsp;
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# the output rtmp url.
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# for rtsp caster, the typically output url:
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# rtmp://127.0.0.1/[app]/[stream]
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# for example, the rtsp url:
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# rtsp://192.168.1.173:8544/live/livestream.sdp
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# where the [app] is "live" and [stream] is "livestream", output is:
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# rtmp://127.0.0.1/live/livestream
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# for flv caster, the typically output url:
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# rtmp://127.0.0.1/[app]/[stream]
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# for example, POST to url:
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# http://127.0.0.1:8936/live/livestream.flv
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# where the [app] is "live" and [stream] is "livestream", output is:
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# rtmp://127.0.0.1/live/livestream
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output rtmp://127.0.0.1/live/livestream;
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output rtmp://127.0.0.1/[app]/[stream];
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# the listen port for stream caster.
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# for mpegts_over_udp caster, listen at udp port. for example, 8935.
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# for rtsp caster, listen at tcp port. for example, 554.
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# for flv caster, listen at tcp port. for example, 8936.
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# TODO: support listen at <[ip:]port>
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listen 8935;
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listen 554;
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# for the rtsp caster, the rtp server local port over udp,
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# which reply the rtsp setup request message, the port will be used:
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# [rtp_port_min, rtp_port_max)
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rtp_port_min 57200;
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rtp_port_max 57300;
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}
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# FLV
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stream_caster {
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enabled off;
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caster mpegts_over_udp;
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output rtmp://127.0.0.1/live/livestream;
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listen 8935;
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}
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stream_caster {
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enabled off;
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caster rtsp;
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output rtmp://127.0.0.1/[app]/[stream];
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listen 554;
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rtp_port_min 57200;
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rtp_port_max 57300;
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}
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stream_caster {
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enabled off;
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# whether stream caster is enabled.
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# default: off
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enabled on;
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# the caster type of stream, the casters:
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# flv, FLV over HTTP by POST.
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caster flv;
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# the output rtmp url.
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# for flv caster, the typically output url:
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# rtmp://127.0.0.1/[app]/[stream]
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# for example, POST to url:
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# http://127.0.0.1:8936/live/livestream.flv
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# where the [app] is "live" and [stream] is "livestream", output is:
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# rtmp://127.0.0.1/live/livestream
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output rtmp://127.0.0.1/[app]/[stream];
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# the listen port for stream caster.
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# for flv caster, listen at tcp port. for example, 8936.
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listen 8936;
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}
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# GB28181
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stream_caster {
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# whether stream caster is enabled.
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# default: off
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enabled on;
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# the caster type of stream, the casters:
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# gb28181, Push GB28181 to SRS.
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caster gb28181;
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# the output rtmp url.
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# for gb28181 caster, the typically output url:
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# rtmp://127.0.0.1/live/[stream]
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# where the [stream] is the VideoChannelCodecID.
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output rtmp://127.0.0.1/live/[stream];
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# the listen port for stream caster.
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# for gb28181 caster, listen at udp port. for example, 9000.
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# @remark We can bundle all gb28181 to this port, to reuse this port.
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# User can choose to bundle port in API port_mode or SIP invite_port_fixed.
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listen 9000;
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# If not bundle ports, use specified ports for each stream.
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rtp_port_min 58200;
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rtp_port_max 58300;
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# Whether wait for keyframe then forward to RTMP.
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wait_keyframe off;
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# Max timeout in seconds for RTP stream, if timeout, RTCP bye and close stream.
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# default: 30
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rtp_idle_timeout 30;
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# Whether has audio.
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# @remark Flash/RTMP only supports 11025 22050 44100 sample rate, if not the audio may corrupt.
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# default: off
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audio_enable off;
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# The exposed IP to receive media stream.
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host 192.168.1.3;
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sip {
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# Whether enable embeded SIP server.
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# default: on
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enabled on;
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# The SIP listen port.
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# default: 5060
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listen 5060;
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# The SIP server ID.
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# default: 34020000002000000001
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serial 34020000002000000001;
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# The SIP server domain.
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# default: 3402000000
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realm 3402000000;
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# The SIP ACK response timeout in seconds.
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# default: 30
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ack_timeout 30;
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# The keepalive timeout in seconds.
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# default: 120
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keepalive_timeout 120;
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# Whether print SIP logs.
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print_sip_message off;
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# Whether play immediately after registered.
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# default: on
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auto_play on;
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# Whether bundle media stream port.
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# default: on
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invite_port_fixed on;
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}
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}
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#############################################################################################
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# SRT server section
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#############################################################################################
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@ -3,8 +3,6 @@
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listen 1935;
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max_connections 1000;
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daemon off;
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pid ./objs/srs28181.pid;
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srs_log_file ./objs/srs28181.log;
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srs_log_tank console;
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http_api {
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@ -16,80 +14,84 @@ stream_caster {
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enabled on;
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caster gb28181;
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#转发流到rtmp服务器地址与端口
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# 转发流到rtmp服务器地址与端口
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# TODO: https://github.com/ossrs/srs/pull/1679/files#r400875104
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# [stream] is VideoChannelCodecID(视频通道编码ID)
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output 127.0.0.1:1935;
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#接收设备端rtp流的多路复用端口
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# 接收设备端rtp流的多路复用端口
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listen 9000;
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#rtp接收监听端口范围，最小值
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# rtp接收监听端口范围，最小值
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rtp_port_min 58200;
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#rtp接收监听端口范围，最大值
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# rtp接收监听端口范围，最大值
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rtp_port_max 58300;
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#是否等待关键帧之后，再转发，
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#off:不需等待，直接转发
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#on:等第一个关键帧后，再转发
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# 是否等待关键帧之后，再转发，
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# off:不需等待，直接转发
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# on:等第一个关键帧后，再转发
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wait_keyframe off;
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#rtp包空闲等待时间，如果指定时间没有收到任何包
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#rtp监听连接自动停止，发送BYE命令
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# rtp包空闲等待时间，如果指定时间没有收到任何包
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# rtp监听连接自动停止，发送BYE命令
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rtp_idle_timeout 30;
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#是否转发音频流
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#目前只支持aac格式，所以需要设备支持aac格式
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#on:转发音频
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#off:不转发音频，只有视频
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#*注意*!!!:flv 只支持11025 22050 44100 三种
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#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
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#同时也会将adts的头封装在flv aac raw数据中
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#这样的话播放器为自动通过adts头自动选择采样频率
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#像ffplay, vlc都可以，但是flash是没有声音，
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#因为flash,只支持11025 22050 44100
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audio_enable on;
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# 是否转发音频流
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# 目前只支持aac格式，所以需要设备支持aac格式
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# on:转发音频
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# off:不转发音频，只有视频
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# *注意*!!!:flv 只支持11025 22050 44100 三种
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# 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
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# 同时也会将adts的头封装在flv aac raw数据中
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# 这样的话播放器为自动通过adts头自动选择采样频率
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# 像ffplay, vlc都可以，但是flash是没有声音，
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# 因为flash,只支持11025 22050 44100
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audio_enable off;
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#服务器主机号，可以域名或ip地址
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#也就是设备端将媒体发送的地址，如果是服务器是内外网
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#需要写外网地址，
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#调用api创建stream session时返回ip地址也是host
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# 服务器主机号，可以域名或ip地址
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# 也就是设备端将媒体发送的地址，如果是服务器是内外网
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# 需要写外网地址，
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# 调用api创建stream session时返回ip地址也是host
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# TODO: https://github.com/ossrs/srs/pull/1679/files#r400917594
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host 192.168.1.27;
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sip {
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#是否启用srs内部sip信令
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#为on信令走srs, off 只转发ps流
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# 是否启用srs内部sip信令
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# 为on信令走srs, off 只转发ps流
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enabled on;
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#sip监听udp端口
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listen 15060;
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# sip监听udp端口
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listen 5060;
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#服务器端编号
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#设备端配置编号需要与该值一致，否则无法注册
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# SIP server ID(SIP服务器ID).
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# 设备端配置编号需要与该值一致，否则无法注册
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serial 34020000002000000001;
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#服务器端域
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# SIP server domain(SIP服务器域)
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realm 3402000000;
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#服务端发送ack后，接收回应的超时时间，单位为秒
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#如果指定时间没有回应，认为失败
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# 服务端发送ack后，接收回应的超时时间，单位为秒
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# 如果指定时间没有回应，认为失败
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ack_timeout 30;
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#设备心跳维持时间，如果指定时间内(秒）没有接收一个心跳
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#认为设备离线
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keepalive_timeout 30;
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# 设备心跳维持时间，如果指定时间内(秒）没有接收一个心跳
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# 认为设备离线
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keepalive_timeout 120;
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#日志打印是否打印sip信息
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#off:不打印
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#on:打印接收或发送sip命令信息
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# 日志打印是否打印sip信息
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# off:不打印
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# on:打印接收或发送sip命令信息
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# TODO: https://github.com/ossrs/srs/pull/1679/files#r400929300
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print_sip_message off;
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#注册之后是否自动给设备端发送invite
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#on: 是 off 不是，需要通过api控制
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# 注册之后是否自动给设备端发送invite
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# on: 是 off 不是，需要通过api控制
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auto_play on;
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#设备将流发送的端口，是否固定
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#on 发送流到多路复用端口 如9000
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#off 自动从rtp_mix_port - rtp_max_port 之间的值中
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#选一个可以用的端口
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# 设备将流发送的端口，是否固定
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# on 发送流到多路复用端口 如9000
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# off 自动从rtp_mix_port - rtp_max_port 之间的值中
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# 选一个可以用的端口
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invite_port_fixed on;
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}
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}
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@ -4310,9 +4310,9 @@ int SrsConfig::get_stream_caster_rtp_port_max(SrsConfDirective* conf)
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return ::atoi(conf->arg0().c_str());
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}
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int SrsConfig::get_stream_caster_gb28181_rtp_idle_timeout(SrsConfDirective* conf)
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srs_utime_t SrsConfig::get_stream_caster_gb28181_rtp_idle_timeout(SrsConfDirective* conf)
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{
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static int DEFAULT = 30;
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static srs_utime_t DEFAULT = 30 * SRS_UTIME_SECONDS;
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if (!conf) {
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return DEFAULT;
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@ -4323,7 +4323,7 @@ int SrsConfig::get_stream_caster_gb28181_rtp_idle_timeout(SrsConfDirective* conf
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return DEFAULT;
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}
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return ::atoi(conf->arg0().c_str());
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return (srs_utime_t)(::atoi(conf->arg0().c_str()) * SRS_UTIME_SECONDS);
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}
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int SrsConfig::get_stream_caster_gb28181_ack_timeout(SrsConfDirective* conf)
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@ -4349,7 +4349,7 @@ int SrsConfig::get_stream_caster_gb28181_ack_timeout(SrsConfDirective* conf)
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int SrsConfig::get_stream_caster_gb28181_keepalive_timeout(SrsConfDirective* conf)
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{
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static int DEFAULT = 30;
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static int DEFAULT = 120;
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if (!conf) {
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return DEFAULT;
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@ -4386,7 +4386,7 @@ string SrsConfig::get_stream_caster_gb28181_host(SrsConfDirective* conf)
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string SrsConfig::get_stream_caster_gb28181_serial(SrsConfDirective* conf)
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{
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static string DEFAULT = "";
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static string DEFAULT = "34020000002000000001";
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if (!conf) {
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return DEFAULT;
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@ -4407,7 +4407,7 @@ string SrsConfig::get_stream_caster_gb28181_serial(SrsConfDirective* conf)
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string SrsConfig::get_stream_caster_gb28181_realm(SrsConfDirective* conf)
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{
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static string DEFAULT = "";
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static string DEFAULT = "3402000000";
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if (!conf) {
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return DEFAULT;
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@ -4428,7 +4428,7 @@ string SrsConfig::get_stream_caster_gb28181_realm(SrsConfDirective* conf)
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bool SrsConfig::get_stream_caster_gb28181_audio_enable(SrsConfDirective* conf)
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{
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static bool DEFAULT = true;
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static bool DEFAULT = false;
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if (!conf) {
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return DEFAULT;
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@ -4481,7 +4481,7 @@ bool SrsConfig::get_stream_caster_gb28181_wait_keyframe(SrsConfDirective* conf)
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bool SrsConfig::get_stream_caster_gb28181_sip_enable(SrsConfDirective* conf)
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{
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static bool DEFAULT = false;
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static bool DEFAULT = true;
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if (!conf) {
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return DEFAULT;
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@ -4503,7 +4503,7 @@ bool SrsConfig::get_stream_caster_gb28181_sip_enable(SrsConfDirective* conf)
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bool SrsConfig::get_stream_caster_gb28181_sip_auto_play(SrsConfDirective* conf)
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{
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static bool DEFAULT = false;
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static bool DEFAULT = true;
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if (!conf) {
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return DEFAULT;
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@ -4525,7 +4525,7 @@ bool SrsConfig::get_stream_caster_gb28181_sip_auto_play(SrsConfDirective* conf)
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int SrsConfig::get_stream_caster_gb28181_sip_listen(SrsConfDirective* conf)
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{
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static int DEFAULT = 0;
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static int DEFAULT = 5060;
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if (!conf) {
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return DEFAULT;
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@ -500,7 +500,7 @@ public:
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// Get the max udp port for rtp of stream caster rtsp.
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virtual int get_stream_caster_rtp_port_max(SrsConfDirective* conf);
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virtual int get_stream_caster_gb28181_rtp_idle_timeout(SrsConfDirective* conf);
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virtual srs_utime_t get_stream_caster_gb28181_rtp_idle_timeout(SrsConfDirective* conf);
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||||
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);
|
||||
|
|
@ -514,7 +514,6 @@ public:
|
|||
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:
|
||||
// Get the vhost directive by vhost name.
|
||||
|
|
|
|||
|
|
@ -1,7 +1,7 @@
|
|||
/**
|
||||
* The MIT License (MIT)
|
||||
*
|
||||
* Copyright (c) 2013-2020 Winlin
|
||||
* Copyright (c) 2013-2020 Lixin
|
||||
*
|
||||
* Permission is hereby granted, free of charge, to any person obtaining a copy of
|
||||
* this software and associated documentation files (the "Software"), to deal in
|
||||
|
|
@ -749,7 +749,7 @@ srs_error_t SrsGb28181RtmpMuxer::do_cycle()
|
|||
}
|
||||
|
||||
srs_utime_t now = srs_get_system_time();
|
||||
srs_utime_t duration = (now - recv_stream_time) / (1000*1000);
|
||||
srs_utime_t duration = now - recv_stream_time;
|
||||
|
||||
SrsGb28181Config config = gb28181_manger->get_gb28181_config();
|
||||
if (duration > config.rtp_idle_timeout){
|
||||
|
|
|
|||
|
|
@ -1,7 +1,7 @@
|
|||
/**
|
||||
* The MIT License (MIT)
|
||||
*
|
||||
* Copyright (c) 2013-2020 Winlin
|
||||
* Copyright (c) 2013-2020 Lixin
|
||||
*
|
||||
* Permission is hereby granted, free of charge, to any person obtaining a copy of
|
||||
* this software and associated documentation files (the "Software"), to deal in
|
||||
|
|
@ -296,7 +296,7 @@ class SrsGb28181Config
|
|||
{
|
||||
public:
|
||||
std::string host;
|
||||
int rtp_idle_timeout;
|
||||
srs_utime_t rtp_idle_timeout;
|
||||
bool audio_enable;
|
||||
bool wait_keyframe;
|
||||
std::string output;
|
||||
|
|
|
|||
|
|
@ -1,7 +1,7 @@
|
|||
/**
|
||||
* The MIT License (MIT)
|
||||
*
|
||||
* Copyright (c) 2013-2020 Winlin
|
||||
* Copyright (c) 2013-2020 Lixin
|
||||
*
|
||||
* Permission is hereby granted, free of charge, to any person obtaining a copy of
|
||||
* this software and associated documentation files (the "Software"), to deal in
|
||||
|
|
|
|||
|
|
@ -1,7 +1,7 @@
|
|||
/**
|
||||
* The MIT License (MIT)
|
||||
*
|
||||
* Copyright (c) 2013-2020 Winlin
|
||||
* Copyright (c) 2013-2020 Lixin
|
||||
*
|
||||
* Permission is hereby granted, free of charge, to any person obtaining a copy of
|
||||
* this software and associated documentation files (the "Software"), to deal in
|
||||
|
|
|
|||
|
|
@ -1,7 +1,7 @@
|
|||
/**
|
||||
* The MIT License (MIT)
|
||||
*
|
||||
* Copyright (c) 2013-2020 Winlin
|
||||
* Copyright (c) 2013-2020 Lixin
|
||||
*
|
||||
* Permission is hereby granted, free of charge, to any person obtaining a copy of
|
||||
* this software and associated documentation files (the "Software"), to deal in
|
||||
|
|
|
|||
|
|
@ -1,7 +1,7 @@
|
|||
/**
|
||||
* The MIT License (MIT)
|
||||
*
|
||||
* Copyright (c) 2013-2020 Winlin
|
||||
* Copyright (c) 2013-2020 Lixin
|
||||
*
|
||||
* Permission is hereby granted, free of charge, to any person obtaining a copy of
|
||||
* this software and associated documentation files (the "Software"), to deal in
|
||||
|
|
|
|||
Loading…
Add table
Add a link
Reference in a new issue