Make sure you have your driver set to an ASIO driver.
Boydie is right about the latency, and yes you reduce latency by reducing buffer size. This is usually done within your audio interface settings. Sometimes you can do it in software, but its really best to do it in the hardware control panel as it will introduce instability a lot of the time (CRASH!). A workable latency stems from a buffer size of about 256 samples and no higher. If you drop to this level and your start getting audio drop outs, cracks and pops, you will need to up it, or check for a newer driver version.
A word about monitoring. There are two types:
Direct monitoring. This is where you monitor your audio directly through your interface. So your audio goes into your interface and is - well the best way to think about it is that it is "duplicated". One version is sent to your software where it is recorded and another is sent to your output for monitoring.
Indirect Monitoring. This is where the audio signal goes into your interface, is converted to digital, is passed to your software where it records, is then passed through any effects and then to your output. This introduces all sorts of time delays before it hits the headphones, as it needs to be processed by the CPU through the software and this makes it sound well out from what you have played.
You can usually mix between the two of these. Its best to use direct monitoring when recoding and switch to indirect when you hit play.
There are also three types of latency you should be aware of - input latency, output latency and plug in latency, but that's a topic for another day.
