Dani
Member
Is there a way to instruct the server or the client to try and keep the connection rather then throw
STREAM_STATUS.FAILED
upon small disconnections ?
most issues are on mobile (iphone and android) but also on some desktops.
Here is my server setup:
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,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_
https.address =192.168.100.24
wss.address =192.168.100.24
ws.port =8080
wss.port =443
wss.keystore.password =
wss.cert.password =
rtmp.port =1935
rtmfp.port =1935
keep_alive.algorithm =HIGH_LEVEL
keep_alive.peer_interval =2000
keep_alive.server_interval =5000
keep_alive.probes =10
video_reliable =partial
audio_reliable =partial
audio_frames_per_packet =6
burst_avoidance_count =100
flush_audio_interval =80
flush_video_interval =0
streaming_video_decoder_fast_start=false
webrtc_cc2_twcc =false
STREAM_STATUS.FAILED
upon small disconnections ?
most issues are on mobile (iphone and android) but also on some desktops.
Here is my server setup:
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,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_
https.address =192.168.100.24
wss.address =192.168.100.24
ws.port =8080
wss.port =443
wss.keystore.password =
wss.cert.password =
rtmp.port =1935
rtmfp.port =1935
keep_alive.algorithm =HIGH_LEVEL
keep_alive.peer_interval =2000
keep_alive.server_interval =5000
keep_alive.probes =10
video_reliable =partial
audio_reliable =partial
audio_frames_per_packet =6
burst_avoidance_count =100
flush_audio_interval =80
flush_video_interval =0
streaming_video_decoder_fast_start=false
webrtc_cc2_twcc =false