New Member
You say that your software supports microphone autoleveling. However, the API clearly displays that the microphone level is set manually. Is there a way of auto regulation of the micro like in skype for better speach quality?
Also, what is audio_reliable? The docs say nothing about this parameter.


Staff member
In WebRTC AGC is on by default.
That is, you don't need any manual sound adjustment.
Though, Flash doesn't support AGC, and to make it function we transcode the sound at the server side. That's why we don't use it.
audio_reliable=partial - this is a RTMFP mechanism to limit the number of retransmits - packages resent to reduce latency. By default is on.
The main parameter in

Allows to bypass RTMFP limitations and approach the UDP by completely removing retransmits in the Flash-to-server direction. This parameter helps increasing quality on lost packets.