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rtsp 生成 hls 拉流代理流,延时比较严重,有个7秒左右,请问能怎么通过调整配置进行优化? #3526

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CxistHans opened this issue May 10, 2024 · 1 comment

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@CxistHans
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咨询的功能模块

  • 拉流代理流 验证、config配置优化。

咨询的具体内容和问题
rtsp 生成 hls 拉流代理流,延时比较严重,有个7秒左右,请问能怎么通过调整配置进行优化?
尝试过 go2rtc 进行代理流,同一个摄像头,go2rtc 只有一秒的延时。
从使用者的角度,麻烦提供点建议和调整,非编码方向。

全局配置:

[api]
apiDebug=1
defaultSnap=./www/logo.png
downloadRoot=./www
secret=admin3pre
snapRoot=./data/snap/

[cluster]
origin_url=
retry_count=3
timeout_sec=15

[ffmpeg]
bin=/usr/bin/ffmpeg
cmd=%s -re -i %s -c:a aac -strict -2 -ar 44100 -ab 48k -c:v libx264 -f flv %s
log=./log/ffmpeg/ffmpeg.log
restart_sec=0
snap=%s -i %s -y -f mjpeg -frames:v 1 -an %s

[general]
check_nvidia_dev=1
enableVhost=0
enable_ffmpeg_log=1
flowThreshold=1024
maxStreamWaitMS=15000
mediaServerId=cxist_pre
mergeWriteMS=0
resetWhenRePlay=1
streamNoneReaderDelayMS=20000
unready_frame_cache=100
wait_add_track_ms=3000
wait_track_ready_ms=10000

[hls]
broadcastRecordTs=0
deleteDelaySec=120
fastRegister=0
fileBufSize=65536
segDelay=0
segDur=2
segKeep=0
segNum=3
segRetain=5

[hook]
alive_interval=10.0
enable=0
admin_params=secret=admin3
on_flow_report=
on_http_access=
on_play=
on_publish=
on_record_mp4=
on_record_ts=
on_rtp_server_timeout=
on_rtsp_auth=
on_rtsp_realm=
on_send_rtp_stopped=
on_server_exited=
on_server_keepalive=
on_server_started=
on_shell_login=
on_stream_changed=
on_stream_none_reader=
on_stream_not_found=
retry=1
retry_delay=3.0
stream_changed_schemas=rtsp/rtmp/fmp4/ts/hls/hls.fmp4
timeoutSec=10

[http]
allow_cross_domains=1
allow_ip_range=::1,127.0.0.1,172.16.0.0-172.31.255.255,192.168.0.0-192.168.255.255,10.0.0.0-10.255.255.255
charSet=utf-8
dirMenu=0
forbidCacheSuffix=
forwarded_ip_header=
keepAliveSecond=30
maxReqSize=40960
notFound=<html><head><title>404 Not Found</title></head><body bgcolor="white"><center><h1>您访问的资源不存在!</h1></center><hr><center>ZLMediaKit(git hash:edca662/2024-04-05T22:11:50+08:00,branch:master,build time:2024-04-05T14:13:43)</center></body></html>
port=80
rootPath=./www
sendBufSize=65536
sslport=443
virtualPath=home,./www/webassist;api,./www/swagger;rtc,./webrtc

[multicast]
addrMax=239.255.255.255
addrMin=239.0.0.0
udpTTL=64

[protocol]
add_mute_audio=0
auto_close=0
continue_push_ms=15000
enable_audio=1
enable_fmp4=0
enable_hls=1
enable_hls_fmp4=0
enable_mp4=0
enable_rtmp=0
enable_rtsp=0
enable_ts=0
fmp4_demand=0
hls_demand=1
hls_save_path=./www
modify_stamp=2
mp4_as_player=1
mp4_max_second=3600
mp4_save_path=./data
paced_sender_ms=0
rtmp_demand=0
rtsp_demand=0
ts_demand=0

[record]
appName=Record
enableFmp4=1
fastStart=0
fileBufSize=65536
fileRepeat=0
sampleMS=500

[rtc]
datachannel_echo=1
externIP=
max_bitrate=0
min_bitrate=0
port=8000
preferredCodecA=PCMA,PCMU,opus,mpeg4-generic
preferredCodecV=H264,H265,AV1,VP9,VP8
rembBitRate=0
start_bitrate=0
tcpPort=8000
timeoutSec=15

[rtmp]
directProxy=1
enhanced=0
handshakeSecond=15
keepAliveSecond=15
port=1935
sslport=0

[rtp]
audioMtuSize=600
h264_stap_a=1
lowLatency=0
rtpMaxSize=10
videoMtuSize=1400

[rtp_proxy]
dumpDir=
gop_cache=1
h264_pt=98
h265_pt=99
opus_pt=100
port=10000
port_range=30000-35000
ps_pt=96
rtp_g711_dur_ms=100
timeoutSec=15
udp_recv_socket_buffer=4194304

[rtsp]
authBasic=0
directProxy=1
handshakeSecond=15
keepAliveSecond=15
lowLatency=0
port=554
rtpTransportType=-1
sslport=0

[shell]
maxReqSize=1024
port=0

[srt]
latencyMul=4
pktBufSize=8192
port=9000
timeoutSec=5
@xia-chu
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xia-chu commented May 13, 2024

这个是技术常识啊 HLS就是延时高啊 这是这个方案的技术特征决定的

@xia-chu xia-chu closed this as completed May 13, 2024
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