Hi,
mine it’s a general question about the parameters you have to pay attention to when you ingest a webrtc source from a, say, cellphone and you have to transcode it to a mpeg ts stream.
There are tons of parameters at almost every level: application, vhost, transcoder and sometimes it is very difficult to make things clear. I mean:
As vhost is concerned, there are buffer settings for incoming and outgoing rtp streams, buffer for net connections, and so on.
For the stream target (srt) there is latency, udp buffer settings,… But unfortunately there is no explanation (unless a generic “sets udp buffer in bytes”).
Since there are so many parameters, is there somewhere a general guide, some rules of thumb you have to follow to have a smooth streaming?
For example for cellphones, the incoming webrtc stream may vary a lot, depending on the net conditions. How can you address this problem?
I know this is a bit general question but anyway… Any help is really appreciated!
Thank you very much,
Davide
