From 6012ac4eb0c9355df849a420a236b1f5b6e12c6c Mon Sep 17 00:00:00 2001 From: winlin Date: Tue, 31 Mar 2020 22:30:57 +0800 Subject: [PATCH] Refine code --- trunk/conf/full.conf | 128 ++++++++++++++++++++------ trunk/conf/push.gb28181.conf | 96 +++++++++---------- trunk/src/app/srs_app_config.cpp | 20 ++-- trunk/src/app/srs_app_config.hpp | 3 +- trunk/src/app/srs_app_gb28181.cpp | 4 +- trunk/src/app/srs_app_gb28181.hpp | 4 +- trunk/src/app/srs_app_gb28181_sip.cpp | 2 +- trunk/src/app/srs_app_gb28181_sip.hpp | 2 +- trunk/src/protocol/srs_sip_stack.cpp | 2 +- trunk/src/protocol/srs_sip_stack.hpp | 2 +- 10 files changed, 167 insertions(+), 96 deletions(-) diff --git a/trunk/conf/full.conf b/trunk/conf/full.conf index a0b39e7ad..37f15eaa7 100644 --- a/trunk/conf/full.conf +++ b/trunk/conf/full.conf @@ -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 ############################################################################################# diff --git a/trunk/conf/push.gb28181.conf b/trunk/conf/push.gb28181.conf index f7a3779b6..a106016c3 100644 --- a/trunk/conf/push.gb28181.conf +++ b/trunk/conf/push.gb28181.conf @@ -3,8 +3,6 @@ listen 1935; max_connections 1000; daemon off; -pid ./objs/srs28181.pid; -srs_log_file ./objs/srs28181.log; srs_log_tank console; http_api { @@ -16,80 +14,84 @@ stream_caster { enabled on; caster gb28181; - #转发流到rtmp服务器地址与端口 + # 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 + # 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 + # 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 on; + # 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 + # 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 + # 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; + # sip监听udp端口 + listen 5060; - #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 + # 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; - #服务器端域 + # 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 + # 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 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 + # 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 + # 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 + # 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; } } diff --git a/trunk/src/app/srs_app_config.cpp b/trunk/src/app/srs_app_config.cpp index 6f925d068..4b5a4b895 100644 --- a/trunk/src/app/srs_app_config.cpp +++ b/trunk/src/app/srs_app_config.cpp @@ -4310,9 +4310,9 @@ int SrsConfig::get_stream_caster_rtp_port_max(SrsConfDirective* conf) return ::atoi(conf->arg0().c_str()); } -int SrsConfig::get_stream_caster_gb28181_rtp_idle_timeout(SrsConfDirective* conf) +srs_utime_t SrsConfig::get_stream_caster_gb28181_rtp_idle_timeout(SrsConfDirective* conf) { - static int DEFAULT = 30; + static srs_utime_t DEFAULT = 30 * SRS_UTIME_SECONDS; if (!conf) { return DEFAULT; @@ -4323,7 +4323,7 @@ int SrsConfig::get_stream_caster_gb28181_rtp_idle_timeout(SrsConfDirective* conf return DEFAULT; } - return ::atoi(conf->arg0().c_str()); + return (srs_utime_t)(::atoi(conf->arg0().c_str()) * SRS_UTIME_SECONDS); } int SrsConfig::get_stream_caster_gb28181_ack_timeout(SrsConfDirective* conf) @@ -4349,7 +4349,7 @@ int SrsConfig::get_stream_caster_gb28181_ack_timeout(SrsConfDirective* conf) int SrsConfig::get_stream_caster_gb28181_keepalive_timeout(SrsConfDirective* conf) { - static int DEFAULT = 30; + static int DEFAULT = 120; if (!conf) { return DEFAULT; @@ -4386,7 +4386,7 @@ string SrsConfig::get_stream_caster_gb28181_host(SrsConfDirective* conf) string SrsConfig::get_stream_caster_gb28181_serial(SrsConfDirective* conf) { - static string DEFAULT = ""; + static string DEFAULT = "34020000002000000001"; if (!conf) { return DEFAULT; @@ -4407,7 +4407,7 @@ string SrsConfig::get_stream_caster_gb28181_serial(SrsConfDirective* conf) string SrsConfig::get_stream_caster_gb28181_realm(SrsConfDirective* conf) { - static string DEFAULT = ""; + static string DEFAULT = "3402000000"; if (!conf) { return DEFAULT; @@ -4428,7 +4428,7 @@ string SrsConfig::get_stream_caster_gb28181_realm(SrsConfDirective* conf) bool SrsConfig::get_stream_caster_gb28181_audio_enable(SrsConfDirective* conf) { - static bool DEFAULT = true; + static bool DEFAULT = false; if (!conf) { return DEFAULT; @@ -4481,7 +4481,7 @@ bool SrsConfig::get_stream_caster_gb28181_wait_keyframe(SrsConfDirective* conf) bool SrsConfig::get_stream_caster_gb28181_sip_enable(SrsConfDirective* conf) { - static bool DEFAULT = false; + static bool DEFAULT = true; if (!conf) { return DEFAULT; @@ -4503,7 +4503,7 @@ bool SrsConfig::get_stream_caster_gb28181_sip_enable(SrsConfDirective* conf) bool SrsConfig::get_stream_caster_gb28181_sip_auto_play(SrsConfDirective* conf) { - static bool DEFAULT = false; + static bool DEFAULT = true; if (!conf) { return DEFAULT; @@ -4525,7 +4525,7 @@ bool SrsConfig::get_stream_caster_gb28181_sip_auto_play(SrsConfDirective* conf) int SrsConfig::get_stream_caster_gb28181_sip_listen(SrsConfDirective* conf) { - static int DEFAULT = 0; + static int DEFAULT = 5060; if (!conf) { return DEFAULT; diff --git a/trunk/src/app/srs_app_config.hpp b/trunk/src/app/srs_app_config.hpp index 801f24ca1..cdeb861aa 100644 --- a/trunk/src/app/srs_app_config.hpp +++ b/trunk/src/app/srs_app_config.hpp @@ -500,7 +500,7 @@ public: // 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_idle_timeout(SrsConfDirective* conf); + virtual srs_utime_t 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); @@ -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. diff --git a/trunk/src/app/srs_app_gb28181.cpp b/trunk/src/app/srs_app_gb28181.cpp index 39b1714bf..29373ae16 100644 --- a/trunk/src/app/srs_app_gb28181.cpp +++ b/trunk/src/app/srs_app_gb28181.cpp @@ -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){ diff --git a/trunk/src/app/srs_app_gb28181.hpp b/trunk/src/app/srs_app_gb28181.hpp index 8eff26b32..812f390fb 100644 --- a/trunk/src/app/srs_app_gb28181.hpp +++ b/trunk/src/app/srs_app_gb28181.hpp @@ -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; diff --git a/trunk/src/app/srs_app_gb28181_sip.cpp b/trunk/src/app/srs_app_gb28181_sip.cpp index 64a8512b6..72a55685c 100644 --- a/trunk/src/app/srs_app_gb28181_sip.cpp +++ b/trunk/src/app/srs_app_gb28181_sip.cpp @@ -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 diff --git a/trunk/src/app/srs_app_gb28181_sip.hpp b/trunk/src/app/srs_app_gb28181_sip.hpp index 17e4313cb..9206e66c9 100644 --- a/trunk/src/app/srs_app_gb28181_sip.hpp +++ b/trunk/src/app/srs_app_gb28181_sip.hpp @@ -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 diff --git a/trunk/src/protocol/srs_sip_stack.cpp b/trunk/src/protocol/srs_sip_stack.cpp index 1b5d49812..e24e743c2 100644 --- a/trunk/src/protocol/srs_sip_stack.cpp +++ b/trunk/src/protocol/srs_sip_stack.cpp @@ -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 diff --git a/trunk/src/protocol/srs_sip_stack.hpp b/trunk/src/protocol/srs_sip_stack.hpp index 26f712081..006ecbba3 100644 --- a/trunk/src/protocol/srs_sip_stack.hpp +++ b/trunk/src/protocol/srs_sip_stack.hpp @@ -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