I'm using JVB with audio mixing and I'm trying to use Opus and ULAW with different packetization times.
So this is the situation:
* JVB transcodes audio between Opus and Ulaw;
* Opus is used with 60ms ptime;
* Ulaw is used with the default 20ms ptime;
Sometimes the audio is fine, sometimes is choppy and almost not intelligible.
When I have audio issues I see in the log a lot of "Packets dropped because full" in the RTPSourceStream statistics after the call.
My assumption is that the jitter buffer requires some changes to work properly when different ptimes are used.
I tried to disable the adaptive jitter buffer and use a higher value for net.java.sip.communicator.impl.neomedia.RECEIVE_BUFFER_LENGTH but still I have this issue.
Could someone give me some hints to which parts I have to modify to fix this issue? Actually I'm quite lost among FMJ's classes :).
Thanks in advance,