I have a small LAN network where I am running a Jain-Sip softphone on two user pc's. These softphones are connected through Asterisk(Trixbox). Although the phones do work in providing an audio conversation, there is a long delay(about 20 seconds) in the initial RTP session setup. I have tried a few values for the buffer length including setting it to zero. I assumed this would drastically reduce the delay but there was no change. I also tried a number of values for the minimum threshold and this did not change the amount of delay either. Would anyone have an idea of why this delay is occurring and possibly how to reduce it?
Any advice would be greatly appreciated,