Karan Bhansali
Member
Hi,
flashphoner properties config
ip =52.74.204.163
ip_local =172.31.0.82
port_from =30000
port_to =30999
media_port_from =31000
media_port_to =31999
profiles =420020
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 =false,0
auto_login_url =/usr/local/FlashphonerWebCallServer/conf/account.xml
get_callee_url =/usr/local/FlashphonerWebCallServer/conf/callee.xml
codecs =opus,alaw,ulaw,g729,speex16,mpeg4-generic,g722,telephone-event,h264,vp8,flv
speex_in_policy =8,false,8,4
ptime = 20
force_vp8_to_h264 = true
vp8_max_rtp_packet_size = 1400
rtp_send_buffer_size = 65536
rtp_receive_buffer_size = 65536
rtp_packet_cache_size = 100
client_dump_level = 0
stun_thread_pool_size = 3
stun_max_threads = 2^31-1
stun_server = stun1.l.google.com:19302
rtp_activity_video = true
flash_policy.port = 843
rtmp_transponder_stream_name_prefix =
video_streamer_generate_seq = true
client_mode=false
rtmp_transponder_full_url=true
rtc_ice_add_local_component=true
keep_extended_logs_max_days= 7
aac_bitrate = 64000
video_filter_enable_fps=true
video_filter_fps=30
#The part below is moved from server.properties...
#Config
ws.port =8080
wss.port =8443
#File will be located in conf directory
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 =2000
keep_alive.server_interval =5000
keep_alive.probes =10
#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 =0
Webrtc audio is not in sync with live video
flashphoner properties config
ip =52.74.204.163
ip_local =172.31.0.82
port_from =30000
port_to =30999
media_port_from =31000
media_port_to =31999
profiles =420020
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 =false,0
auto_login_url =/usr/local/FlashphonerWebCallServer/conf/account.xml
get_callee_url =/usr/local/FlashphonerWebCallServer/conf/callee.xml
codecs =opus,alaw,ulaw,g729,speex16,mpeg4-generic,g722,telephone-event,h264,vp8,flv
speex_in_policy =8,false,8,4
ptime = 20
force_vp8_to_h264 = true
vp8_max_rtp_packet_size = 1400
rtp_send_buffer_size = 65536
rtp_receive_buffer_size = 65536
rtp_packet_cache_size = 100
client_dump_level = 0
stun_thread_pool_size = 3
stun_max_threads = 2^31-1
stun_server = stun1.l.google.com:19302
rtp_activity_video = true
flash_policy.port = 843
rtmp_transponder_stream_name_prefix =
video_streamer_generate_seq = true
client_mode=false
rtmp_transponder_full_url=true
rtc_ice_add_local_component=true
keep_extended_logs_max_days= 7
aac_bitrate = 64000
video_filter_enable_fps=true
video_filter_fps=30
#The part below is moved from server.properties...
#Config
ws.port =8080
wss.port =8443
#File will be located in conf directory
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 =2000
keep_alive.server_interval =5000
keep_alive.probes =10
#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 =0
Webrtc audio is not in sync with live video