Jitter Buffer Confusion

I am trying to implement jitter buffers on IAX trunks and SIP channels.

I’m not getting the results I’m looking for. For instance, I entered the config below in sip.conf and my understanding is that it should introduce a 2.5 second delay to all conversations but it does not seem to make any noticeable audible difference if this is turned on or not.

jbenable=yes
jbforce=yes
jbmaxsize=2500
jbimpl=fixed
jblog=yes

Also I am unclear as to weather the jbenable=yes should be placed on the configuration for the outgoing (tx) channel or in the configuration for the incoming channel.

The example config file says:
“Jitterbuffer implementation, used on the receiving side of a SIP channel”

But this (asterisk.org/node/48317) says “You may also set jitterbuffer configuration in sip.conf for SIP channels. However, it is very important to understand that this configuration applies to outbound channels, only.”

So which is it?

Beyond forum advice I am looking for a paid consultant who has an in depth understanding and plenty of experience with configuring a jitter buffer.

I am using Asterisk 1.4.21