I have a Windows 10 64 bit with an AMD 8 core processor @ 4700 Mhz with 16 GB RAM. I use a Scarlett 2i2 as my DAC. 
 
I'm trying to simply record any instrument and vocal and be able to wear the headphones and hear my live feed in real time with the playback as I record. There is always a delay. I've adjusted the settings in the ASIO panel. Increased latency compensation, messed with the kernel buffers and ASIO buffer size. So far nothing has gotten it just right. It's gotten better, but if I go too far I lock up the program and even sometimes the entire machine. 
 
Thoughts?

You should be able to enable "direct monitoring" on the 2i2.  That way you hear the audio you are recording before it is sent to the PC with essentially zero latency. I'm sure someone else with a 2i2 will be able to tell you exactly how to set that up if you can't find out but it should be in the Focusrite manual somewhere.
 
AMB
 

From the 2i2 manual:
 
"The Scarlett 2i2 is fitted with a “Direct Monitoring” option, which overcomes this problem. Setting the front panel DIRECT MONITOR control to ON will route your input signals directly to the Scarlett 2i2’s headphone and main monitor outputs. This enables you to hear yourself with zero latency – i.e., in “real time” – along with the computer playback. The input signals to your computer are not affected in any way by this setting. When Direct Monitoring is set to ON, ensure that your DAW software is not set to route its input (what you are currently recording) to its output. If it is, you will hear yourself “twice”, with one signal audibly delayed as an echo."
 
AMB
 

Direct monitoring is the obvious answer, but asking a question like this it would be good to offer the information
needed to get a more detailed picture of the issue.
What is your ASIO latency set to? Do you have FX enabled while recording (compressors or reverb)? Are there hungry-or-and-many  VSTs running in the project outside the tracks you're recording? Are you using Scarlets own ASIO driver, not ASIO4ALL?  Is the Scarlett the only soundcard selected anywhere in SONARs settings?
 
Your gear should easily handle the task, but you say lowering buffer/latency halts the machine. So, what kind of values for ASIO latency or buffer sizes you're using when the machine halts??
 

If your PC and Sound Card can handle low latency, just set it to its lowest ASIO buffer setting or WDM setting. 
 
CJ

Some processors, particularly limiters, compressors, and maximizers, have a "look-ahead" function so they can be aware of transients and act appropriately before the transients actually happen. Since they can't actually look into the future, everything else gets delayed. That could be part of your problem. For example, just having one insert of a Waves L3 Multimaximizer in the master bus will make doing vocals (when monitoring through the computer) more or less impossible.
 
Hit "E" to bypass all effects and see if that makes a difference.