Jitter buffer significantly improved call response


#1

Hi,

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
Implementation: Adaptive
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. :slightly_smiling:


#2

I was hoping to find more information on this topic too.

I’m curious about how asterisk handles or doesn’t handle jitter between two endpoints. If both endpoints have reinvites disabled (i.e. all audio through asterisk) what happens when some of the packets from endpoint 1 arrive at asterisk out of order?

Are they sent to endpoint 2 out of order as well? And then it’s up to endpoint 2’s jitter buffer to sort things out?