Sip Jitter Buffer?

I am using asterisk:
version: SVN-group-rtpjitterbuffer-r30963

jbenable = yes
jbforce = yes
jbmaxsize = 600
jbresyncthreshold = 1000
jbimpl = fixed
jblog = yes

I am using xlite which I believe supports jitter buffer. When I set jbmaxsize should that make my voice delay? What should I do to test jitter buffer with sip? Everything sounds good but I want to make sure it is really working. Is there something in the console I can do to see if its working? If I need to do something else please tell me. I once I get it working. I am going to do some major testing.

Linux Adminstrator
Feel free to call me.