Webrtc as rtmp republishing with different rtmp url and same stream name issue

[Max]

Hello
You have server-side transcoding VP8 > H.264
Try to avoid transcoding enabling H.264 as priority codec.
WCS_HOME/conf/flashphoner.properties
Find

codecs=opus,alaw,ulaw,g729,speex16,mpeg4-generic,g722,telephone-event,vp8,h264,flv

Replace with

codecs=opus,alaw,ulaw,g729,speex16,mpeg4-generic,g722,telephone-event,h264,vp8,flv

Restart WCS

service webcallserver restart

Stable video bitrate is important for streaming.
You can tune server-side bitrate settings according bitrate docs:
https://docs.flashphoner.com/displa...ement+when+capturing+WebRTC+stream+in+browser

 

[Karan Bhansali]

Hi,
Now everything works well, we just found lip sync issue on few sessions, is there any provision to avoid sync issue(audio video) before sending the stream.

 

[Max]

Hello.
In latest WCS builds sync issues may occurs when packets are lost or dropped when publishing stream. In most cases it depends on channel quality from publisher to server. So we recommend to check channel quality as described in this doc, then choose the optimal resolution for your channel and stabilize video publishing bitrate.