Finally got everything together and now want to record some guitar but when i monitor it ,"input echo" it is about 2 seconds from what i am actually playing .
 
So how can i bring that down to "as i play i hear"  ? ?
Do i need to bring buffer size down on my Audio interface  ?
What setting to change this , i have it almost perfect now apart from this , I am also using Amplitube 3 as a VST plugin within SONAR .
 
Thanks again 
 
Dan

2 seconds of latency is rather excessive,  something wrong there...
You should be more in the 8 - 10 millisecond range for a "standard" setup or down to 4 - 5 for a well setup rig.
 
What are your system specs? Audio interface?

This activate "hear as I play" select "Input Echo"

To reduce buffers Preferences-> Audio if using ASIO then select that button to open the ASIO applet where you can change them

If not using ASIO then the third page under Preferences Audio will have a slider - left =smaller right = larger. Yo have to find a balance. Too small crackles too large too much delay.

I haven't tried this myself as yet, but if you're using ASIO drivers the first post in the thread below may be useful for you:
 
http://forum.cakewalk.com/Possible-Key-to-SONAR-Performance-Breakthrough-Check-this-Out-m3103263.aspx

Karyn
2 seconds of latency is rather excessive,  something wrong there...
You should be more in the 8 - 10 millisecond range for a "standard" setup or down to 4 - 5 for a well setup rig.
 
What are your system specs? Audio interface?

MSI Z77A-GD65 Motherboard
Intel core i5 3570k (unlocked) 1155 Ivy Bridge CPU
BeQuiet Dark Rock Advanced CPU Cooler
Corsair Vengeance Black low profile 8GB (2x4GB) Memory
Samsung 830 SSD SATA III 256GB 6GBPS 2.5"
Coolermaster 600W Silentpro PSU
EVGA 3GB GTX 780 Ti Superclocked, 28nm, PCIe 3.0 (x16), 7000MHz GDDR5, GPU 980MHz, Boost 1046MHz, Cores 2880, DP/ HDMI
Coolermaster CM 690 II Case
 
And a 
PreSonus AudioBox USB 2x2 USB 2.0

You should have no problems with that setup Dan.   Things to check....
 
Latest driver version from PreSonus.
Make sure you're using ASIO mode.
Make sure AudioBox is NOT set as the default device in Windows sound settings.
Reduce buffer size  (in the driver settings)
AudioBox input is the clock master in Sonar.
Make sure there are no reverb (or similar) plugins that require (and set) huge latency in any tracks , especially on the Master bus.
Try a test using WDM, Sonar should tell you the response time when it does its analysis, but ASIO is the preferred option. 
 
I know it works, my daughter has one.

Karyn
You should have no problems with that setup Dan.   Things to check....
 
Latest driver version from PreSonus.
Make sure you're using ASIO mode.
Make sure AudioBox is NOT set as the default device in Windows sound settings.
Reduce buffer size  (in the driver settings)
AudioBox input is the clock master in Sonar.
Make sure there are no reverb (or similar) plugins that require (and set) huge latency in any tracks , especially on the Master bus.
Try a test using WDM, Sonar should tell you the response time when it does its analysis, but ASIO is the preferred option. 
 
I know it works, my daughter has one.

Buffer size is 512 lowered from 1024 . 
AudioBox input is the clock master in Sonar ? > can you explain how to check this  please

You should be able to get the buffer size down to 128,  certainly 256.
I use a PreSonus FireStudio at 128 or even 64 when I'm tracking live mics.  For editing/mixdown I'll increase it to 512 or more to allow heavy plugins at the expense of added latency.  But during editing it doesn't bother me that there's a half second pause between pressing 'play' and the track playing...  You need the lowest buffer setting you can stand for live mic recording though.
 
If the AudioBox is actually working then the clock master is most likely set correctly, but look in the audio settings in preferences.  I only mentioned it because it's something I've come across before,   it's not a standard known issue.

Sonar does an automatic latency calc to allow for plugins currently in use.  I'm sure there's a way to for it to re-calc, but I'm not at my DAW just now and I can't remember it.             
 
Put simply, plugins like older convolution reverbs (Perfect Space) take a lot of processing and add latency.  Any plugin that has a 'look ahead' option (gates, compressors, limiters,  ALL the FabFilter versions of those) will add however much look ahead you specified as extra latency,  for EACH plugin using it..   Not good for live tracking.

Start at an ASIO buffer 64 or 32 ,if it lets you and make sure you do not have any plugins with hidden buffers. Some plugins are not meant for real-time. as they are meant for mixing.
 
As always, make sure your sound card is up to date with the newest drivers.
 
CJ