I am currently in the fine tuning phase of PBX, users has beenexperiencing delay and cracking sound at remove office and especially noticeable with conference call.
I noticed jitter while capturing with tcpdump command, so I forced Jitter Buffer from Settngs -> Asterisk SIP Settings -> Chan SIP -> Jitter Buffer Settings.
Jitter Buffer: Enabled
Force Jitter Buffer: Yes
Jitter Buffer Size: 200 (jbmaxsize) 1000 (jbresyncthreshold)
Interesting thing to note was that forcing jitter buffer drastically improved delay which we were experienced, and I hear less delay when I call *43 to hear my looped back voice.
I am hoping the setting to resolve issues remote office users are experiencing, I am gathering more statistic data right now but I am curious to know why forcing jitter buffer drastically improved the call response.
Any commend would be highly appreciated.