burak guder
Member
hi
after a while audio sync deteriorates over time. page refleshing problem solved
Pls help me
My configuration :
#server ip
ip =xxxx
ip_local =xxxxx
#webrtc ports range
media_port_from =31001
media_port_to =32000
#codecs
codecs =opus,alaw,ulaw,g729,speex16,g722,mpeg4-generic,telephone-event,h264,vp8,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
#websocket ports
ws.port =8080
wss.port =8443
rtp_force_synchronization=true
disable_manager_rmi=false
#disable_rest_auth=false
#disable_drop_aac_frame=true
rest_access_control_allow_origin=*
rest_access_control_allow_headers=content-type,x-requested-with
rest_access_control_allow_methods=POST
after a while audio sync deteriorates over time. page refleshing problem solved
Pls help me
My configuration :
#server ip
ip =xxxx
ip_local =xxxxx
#webrtc ports range
media_port_from =31001
media_port_to =32000
#codecs
codecs =opus,alaw,ulaw,g729,speex16,g722,mpeg4-generic,telephone-event,h264,vp8,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
#websocket ports
ws.port =8080
wss.port =8443
rtp_force_synchronization=true
disable_manager_rmi=false
#disable_rest_auth=false
#disable_drop_aac_frame=true
rest_access_control_allow_origin=*
rest_access_control_allow_headers=content-type,x-requested-with
rest_access_control_allow_methods=POST