Gabor Nagy
New Member
Hi,
We are running WCS 5.2.43, and experiencing audio out of sync. This is the config we are running:
#server ip
ip =x.x.x.x
ip_local =y.y.y.y
#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
rtmp_transponder_stream_name_prefix =
rtmp_transponder_full_url=true
webrtc_cc_min_bitrate=300000
webrtc_cc_max_bitrate=1000000
rtp_force_synchronization=true
disable_drop_aac_frame=true
use_fdk_aac=true
Any advice?
We are running WCS 5.2.43, and experiencing audio out of sync. This is the config we are running:
#server ip
ip =x.x.x.x
ip_local =y.y.y.y
#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
rtmp_transponder_stream_name_prefix =
rtmp_transponder_full_url=true
webrtc_cc_min_bitrate=300000
webrtc_cc_max_bitrate=1000000
rtp_force_synchronization=true
disable_drop_aac_frame=true
use_fdk_aac=true
Any advice?