Technically, this can work and produce equivalent results. But there’s an efficiency caveat.
First, you bring up the excellent point that the order of the filters in Shotcut does not necessarily have to be the same order that their parameters are configured. For instance:
Filter order in Shotcut filter panel:
Low Cut/High Pass → Parametric EQ → Compressor → Normalize: Two Pass
Order that filters are configured:
Low Cut/High Pass → Compressor → Normalize: Two Pass → Parametric EQ
Using this sequence, you are correct that any boosts made by the EQ filter would get controlled by the Compressor and be less likely to clip. True point.
Note that Parametric EQ is still configured last because it’s inefficient (a waste of time) to configure it before Normalize, only to configure it again after Normalize due to the perception differences of being at a different volume level.
This brings us to the caveat. In the filter sequence above, any changes made to the EQ filter are going to cause volume changes that affect the threshold and makeup gain of the compressor. That means changing the EQ now requires the Compressor to be updated to reflect the new incoming volume level. Depending on what the Compressor is doing, a change might need to be rippled to Normalize as well.
The basic issue is that the EQ filter will likely be the filter that changes the most as you hunt for your perfect sound. The more likely a filter is to change, the nicer it is to put it at the end of the chain. If EQ is last, you can alter it all day long without having to go back and update the Compressor or Normalize filters to compensate for the new EQ values.
So, I personally use Low Cut/High Pass → Compressor → Normalize: Two Pass → Additional EQ from the perspectives of both highest audio quality and most efficient workflow (least amount of rippled rework). It is not the only way to do things, of course. But it’s efficient and effective on many sources.
You also made a good point about the potential of a big EQ boost to cause clipping if EQ is done last. Yes, technically, this is a possibility. However, this is also why YouTube and other platforms set their LUFS targets between -14 and -16 LUFS. Broadcast television goes as low as -23 LUFS, and cinema goes even lower. These targets mean you’re allowed to have significant headroom before clipping without being perceived as quieter than everyone else’s audio. As in, if there is an audio track whose average volume level is -14 LUFS, and an EQ change causes that track to clip, then something alarming has just happened with that EQ:
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That audio track may have crazy wide dynamic range that needs to be shrunk.
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That EQ change may have been ridiculously drastic. If a source has to be altered by more than 10 dB of EQ, then either the microphone is not capturing the sound they want at all in the first place, or they’re trying to stretch the sound into something totally foreign, like doing sound effects work to create alien voices.
Generally speaking, normal EQ changes will not be drastic enough to overrun -14 LUFS of headroom. That’s what the headroom is there for… room to play. Even if EQ does cause a drastic enough change that clipping happens, all that needs to be done is go back to the Normalize filter and drop it a few dB. (Watching the “I” loudness meter can provide an exact number.) Now the boosted EQ will fit inside the additional headroom we just freed up. That’s a much simpler change than having to redo the Compressor threshold to account for the boosted EQ coming into it. Nobody wants to tweak a compressor any longer than they absolutely have to. That thing is complicated and touchy. 
Good call. Personally, for speech, I like the Low Cut/High Pass rolloff filters because they drop volume all the way to minus infinity instead of just shelving (stopping) at a certain volume. The infinity rolloff will block more low-rumble background noise than a shelf will. Granted, for non-solo musical sources, a shelf can often sound better than an infinity rolloff.
This is usually done with a half-billion quick fade-ins and fade-outs at clip split points where breathing happens. Or with keyframed volume ducking. It’s ugly.
Sometimes, a Gate filter can detect that your voice volume dropped to nearly nothing and close the sound down so that breathing isn’t enough to trip the gate and get through. However, a lot of pro timelines still use fade-in/fade-out because it’s manual and fully controlled. As in, a Gate kinda does whatever it wants based on the incoming sound. If you’re still tweaking your sound and changing volume levels, then those changes will mess with your gate settings too. The only way to know if the Gate is responding properly is to listen to the entire track in real-time to verify it’s right. And then you can’t modify the volume settings anymore without risk of messing up a working Gate. Pros don’t have time for a full-track manual review, and can’t sacrifice their editing flexibility. So they often stick with the ugly but reliable fade-in/fade-out method.
That said, if your audio is super constant and clean, you might be able to get away with a gate filter.
On VER C, it was comfortable enough on my speakers that the bass could maybe even be boosted ever so slightly. It didn’t have the punch and presence that VER A did. So, removing even more bass might make it too weak to compete with background music. Just my thoughts based on external speakers.
If it sounds better, then yes. If it doesn’t, then no. This is the only iron-clad rule of audio engineering. 
Generally no to both. There are some voiceover guys with massive voices that can dip slightly into 80 Hz, but the fundamental for mere mortal guys is higher. If somebody did a Low Cut/High Pass at 70 Hz, they would probably be solid for 100% of humanity. (I like how “probably” and “solid” and “100%” are in the same sentence. Only a Sith deals in absolutes.)
Also for speech, there is generally nothing of interest above 10 kHz. It will mostly (if anything) be the hiss that results from sharp “sss” words. Few people need more “sss” in their speech. The meat of sibilance is around 5-6 kHz, so I could imagine some EQ control being done in that range.
To your point, yes, it makes sense to roll off everything below and above that range to reduce stray noises. A High Shelving filter from the parametric EQ is probably sufficient for the 10+ kHz stuff rather than a dedicated High Cut filter, since there is unlikely to be any significant noise there in a halfway-decent room. Air conditioners don’t hang out in that higher range, for instance.