Dmitry Chistyakov
Member
Добрый день, подскажите, плиз. Есть ситуация - сервер лег под нагрузкой в 300 человек, т.е. тупо перестал отображать стрим у всех кто был свыше. При это загрузка проца в среднем 20-30% на каждый камень из 24, оперативы забито было гиг 6 из 80.
от конфиги сервера, подскажите, что могло к такому привести?
от конфиги сервера, подскажите, что могло к такому привести?
#Config
ws.port =8080
wss.port =8443
#File will be located in conf directory
wss.keystore.file =wss.jks
wss.keystore.password =password
wss.cert.password =password
rtmp.port =1935
rtmfp.port =1935
#keep_alive_algorithm may be INTERNAL, NONE, HIGH_LEVEL
keep_alive.algorithm =HIGH_LEVEL
keep_alive.peer_interval =20000
keep_alive.server_interval =50000
keep_alive.probes =10
keep_alive_streaming_sessions_enabled =true
send_receive_buffer_size = 1600
#Reliability: on, partial, off
video_reliable =partial
audio_reliable =partial
audio_frames_per_packet =6
burst_avoidance_count =100
flush_audio_interval =80
flush_video_interval =80
ws.port =8080
wss.port =8443
#File will be located in conf directory
wss.keystore.file =wss.jks
wss.keystore.password =password
wss.cert.password =password
rtmp.port =1935
rtmfp.port =1935
#keep_alive_algorithm may be INTERNAL, NONE, HIGH_LEVEL
keep_alive.algorithm =HIGH_LEVEL
keep_alive.peer_interval =20000
keep_alive.server_interval =50000
keep_alive.probes =10
keep_alive_streaming_sessions_enabled =true
send_receive_buffer_size = 1600
#Reliability: on, partial, off
video_reliable =partial
audio_reliable =partial
audio_frames_per_packet =6
burst_avoidance_count =100
flush_audio_interval =80
flush_video_interval =80
port_from =30000
port_to =31000
media_port_from =31001
media_port_to =32000
waiting_answer =60
user_agent =Flashphoner/1.0
balance_header =balance
cost_header =cost
video_enabled =true
domain =
outbound_proxy =
outbound_port =
log_level =5
enable_context_logs =false
rtp_activity_detecting =true,60
sip_msg_listener =com.flashphoner.sdk.sip.ChangeCallIdListener
call_record_listener =com.flashphoner.server.client.DefaultCallRecordListener
dtmf =rfc2833
auto_login_url =/usr/local/FlashphonerWebCallServer/conf/account.xml
get_callee_url =/usr/local/FlashphonerWebCallServer/conf/callee.xml
#codecs =opus,alaw,ulaw,g729,speex16,g722,mpeg4-generic,telephone-event,h264,vp8,flv,mpv
codecs =opus,alaw,ulaw,g729,speex16,g722,mpeg4-generic,telephone-event,vp8,h264,flv,mpv
codecs_exclude_sip =mpeg4-generic,flv,mpv
codecs_exclude_streaming =flv,telephone-event
codecs_exclude_sip_rtmp =opus,g729,g722,mpeg4-generic,vp8,mpv
on_record_hook_script =on_record_hook.sh
#rtmp_transponder_stream_name_prefix =rtmp_
webrtc_cc_min_bitrate=30000
webrtc_cc_max_bitrate=400000
webrtc_cc2 = true
webrtc_cc2_сс = false
webrtc_cc2_bitrate_overuse_event_threshold=0.05
webrtc_cc2_min_remb_bitrate_bps =100000
#streaming_video_decoder_fast_start=false
record_flash_published_streams =true
#audio_incoming_buffer_size =10 # def is 50
#video_incoming_buffer_size =10 # def is 50
port_to =31000
media_port_from =31001
media_port_to =32000
waiting_answer =60
user_agent =Flashphoner/1.0
balance_header =balance
cost_header =cost
video_enabled =true
domain =
outbound_proxy =
outbound_port =
log_level =5
enable_context_logs =false
rtp_activity_detecting =true,60
sip_msg_listener =com.flashphoner.sdk.sip.ChangeCallIdListener
call_record_listener =com.flashphoner.server.client.DefaultCallRecordListener
dtmf =rfc2833
auto_login_url =/usr/local/FlashphonerWebCallServer/conf/account.xml
get_callee_url =/usr/local/FlashphonerWebCallServer/conf/callee.xml
#codecs =opus,alaw,ulaw,g729,speex16,g722,mpeg4-generic,telephone-event,h264,vp8,flv,mpv
codecs =opus,alaw,ulaw,g729,speex16,g722,mpeg4-generic,telephone-event,vp8,h264,flv,mpv
codecs_exclude_sip =mpeg4-generic,flv,mpv
codecs_exclude_streaming =flv,telephone-event
codecs_exclude_sip_rtmp =opus,g729,g722,mpeg4-generic,vp8,mpv
on_record_hook_script =on_record_hook.sh
#rtmp_transponder_stream_name_prefix =rtmp_
webrtc_cc_min_bitrate=30000
webrtc_cc_max_bitrate=400000
webrtc_cc2 = true
webrtc_cc2_сс = false
webrtc_cc2_bitrate_overuse_event_threshold=0.05
webrtc_cc2_min_remb_bitrate_bps =100000
#streaming_video_decoder_fast_start=false
record_flash_published_streams =true
#audio_incoming_buffer_size =10 # def is 50
#video_incoming_buffer_size =10 # def is 50
