fine tuning buffersize

[Gabriel T]

Hello, what would be the buffer parameters to play with to try to reduce latency at start. As finally i noticed that there is 4/5 seconds latency just when a stream start to be published until it start playing, but afterward the latency is not more than 1 second. So i think proper workaround would be or to reduce latency on encoder side, or on player client side. Or a mix of both if its possible to set this setting in client side too.
bref, any tips welcome.
Thanks

 

[Max]

Good day.
Please clarify the following:
1. How do you publish a stream: WebRTC, RTMP, RTSP?
2. If you're using RTMP, do you publish stream from encoder to server or pull from another server?
3. What WCS and WebSDK versions do you use?

 

[Gabriel T]

hello, talking about webRTC, with latest SDK and WCS

 

[Max]

So, the stream is published and played as WebRTC? What is the video resolution and bitrate, is transcoding enabled?

 

[Gabriel T]

resolution is 320*240, i've limited the bitrate server side according to the tuning page recommandations, not sure about transcoding but probably no transcoding

 

[Max]

Good day.
Please collect a report as described here, including server configuration files and stream debug logs, and send to support@flashphoner.com, we will check.