I've always noticed a difference in (MIDI) latency response when I visit other studios using Mac computers.
Currently, when I record using my keyboard controller, the timing of the recordings always process a few nudges earlier but when I connected my Focusrite audio interface on an old 2010 MacBook, installed the driver and fired up Logic Pro 9 - without changing any settings - everything was just snappy! It was like moving from black & white to color.
 
Am I missing something? Is there something wrong with my computer? Is there a setting in SONAR that should or shouldn't be checked?

The actual latency of MIDI messages and soft synth response should be pretty consistent. The big difference when playing soft synths will be outbound audio latency. What's you buffer size in SONAR, and what does it report for outbound latency under Preferences > Audio > Driver Setting? I don't know Logic, but I assume it has a similar configuration menu somewhere that will give you this info for comparison.

Buffer Size: 5.0 msec

Effective latency at 44kHz/stereo: 5.0 msec

Input: 11.4 msec, 501 samples
Output: 16.4 msec, 722 samples
Total Roundtrip: 27.7 msec, 1223 samples

That's a huge amount of round-trip latency (and thus the lag you feel).
Reduce the ASIO buffer size to 64-samples... and your round-trip latency will drop significantly.
 
It's possible to achieve sub 5ms total round-trip latency under Windows (64-sample ASIO buffer size/44.1k).
The audio interface driver's hidden safety-buffer is the X-factor.
If your audio interface uses a large hidden safety-buffer, it'll have high round-trip latency.
If your audio interface uses a small hidden safety-buffer, it'll have low round-trip latency.
Your particular unit offers decent round-trip latency (not particularly high... not super low).
 

By the way, my audio interface is a Focusrite Scarlett 18i8.

Thanks for the info, Jim.

What should my roundtrip latency look like on average?

You should be able to get a 10ms round trip latency or maybe better than that. To an extent it depends on how many tracks you are input echoing in the DAW and the cpu load plugins are causing (and some plugins create quite substantial latency fhemselves).

It will also mean making sure the PC is optimally configured for real time audio. Things like the various C sleep states (controlled in the BIOS), cpu core parking and things like drivers hogging too much Windows/hardware time (a different kind of latency on the PCI bus itself caused by delayed procedure calls) can all cause dropouts or crackles if latency is too low. Wifi adaptors are a major source of trouble unless disabled in Window's device manager while you're using the DAW software.

I can get a 32 sample buffer at 44.1KHz on the PC in my sig, so long as I don't load up the project too heavily. 64 samples is more stable though. That's a 5 or 6ms round trip latency. I can also get the same on my MacBook Pro which is quite a bit less powerful and needed no tweaking of anything to get latency that low. Apple's core audio handles things better than Windows, but a well configured PC can equal a Mac. It just requires some fine tuning.

BTW from what I know, that effective latency at the top only regardsWDM. Your ASIO setting is too high. If you contact Focusrite, they may send you the new driver(it was in beta when I got it and may still be). It gives you 3 choices Recording  Balanced Mixing with the ability to change the sample rate. If it was me, and it was.......I'd get it. The old focusrite driver, while stable, was a kludge to configure correctly. Fortunately we usually have leeway and more than one setting will work well.

charlyg
BTW from what I know, that effective latency at the top only regardsWDM. Your ASIO setting is too high. If you contact Focusrite, they may send you the new driver(it was in beta when I got it and may still be). It gives you 3 choices Recording  Balanced Mixing with the ability to change the sample rate. If it was me, and it was.......I'd get it. The old focusrite driver, while stable, was a kludge to configure correctly. Fortunately we usually have leeway and more than one setting will work well.
 

I have the latest driver (1.8) from Focusrite.
What settings are you currently using with your Focusrite soundcard?
Also, my Playback Timing Master is set to the Focusrite's 'SP/DIF L' but I don't use any other gear besides my audio interface, so it's currently set to 'Internal' as the sync source. Is this fine?

My version is 3.1.10.221 It is not publicly available yet, but if you email them......
 
I can get to 48k 64 samples but I don't trust it. 5.4 in 5.4 out 10.8 rountrip
I do just fine with 48k 128 samples for  6.7  6.7  13.5
I can get under 10 rt but depending on the size of the project, I may get some dropouts. Anything much over 15 roundtrip and it's real easy to hear the latency.
 
I don't know much about timing master, but mine is set to the 2i2..

thechurchboi
Buffer Size: 5.0 msec

Input: 11.4 msec, 501 samples
Output: 16.4 msec, 722 samples
Total Roundtrip: 27.7 msec, 1223 samples

 
That's quite a bit of bus and A/D/A conversion latency. Even if you had no buffering at all, your round-trip would be 27.7ms - 2 x 5ms = 17.7ms, which would be a little bothersome with input-monitored instruments, and just tolerable with soft synths that are only subject outbound latency.
 
If the latency is a lot better on your Mac with comparable buffer sizes, it would seem that the USB system and interface drivers are working better there.  SONAR is at the mercy of you Windows hardware performance.