I have been having an issue where my SIP calls to another asterisk box (LAN) takes around 40 seconds before the call is crystal clear. Until then, audio will randomly cut out, you will hear things said 3-4 seconds ago. And then magically it all clears up and the call is perfect. Could anyone tell me what could be causing this strange issue? Also I have noticed “Probation passed - setting RTP source address to xx.x.x.x” on both SIP boxes does the call becomes crystal clear, is there a way to speed this up?