Audio out of sync

[Gabor Nagy]

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?

 

[Max]

Hello.
In most cases the reason for out of sync is packets losses while publishing a stream. So you can:
1) Switch to TCP transport for WebRTC streaming, see this doc for details. This should help with packets losses but can increase whole latency.
2) Test publishers channel bandwith as described here, then adjust optimal bitrate and resolution for publisher to escape packets losses.

rtp_force_synchronization=true

This parameter is not actual anymore, you can remove it from config

 

[Gabor Nagy]

We did both, switched to TCP and kept a fixed bitrate but the issue still persist. Worth noting, audio out of sync happens all the time on mobile (on both android and iOS).

Any other suggestions to solve the problem?

 

[Max]

What is stream source?

1. Could you please provide step by step instructions how to reproduce the issue.
Example:
1. Stream from Chrome 73, Windows 10
2. Play on Android Chrome 73.
3. Out of sync.

2. Did you test with your own application?
Please test with our demo sample first:
https://demo.flashphoner.com/client...ming/two_way_streaming/two_way_streaming.html
Do you have any sync issues with our demo server?