Does it make sense converting a file to a higher audio bitrate?

Yes, it might actually make sense if you are being forced to change formats.

If you have a file with 95kbps in a highly efficient format, to retain the same quality, a relatively inefficient format as mp3 needs a higher bitrate.

Of course you will never get anything back that was lost in the first place. On the contrary, encoding as mp3 will reduce the quality further. Every lossy format uses other means to reduce the amount of data that is stored, by (simplified) throwing away "unneeded" parts of the data. Round trip through a bunch of different formats and there won't be much left ...

So if you want to stay as close a possible to the quality your file has now, you should chose a higher bitrate. 320kbps are probably wasted space, but for mp3 something in the order between 128 and 192 is needed to maintain - or at least come close to - the quality of a more efficient 95kbps file.

In the general case this will not usually result in higher quality audio. The basic reason being that you cannot manufacture sounds that aren't there in the original file.

In the best case the only result will be, as you suggest, larger files.

In the worst case the files could even be of worse quality as the second lossy encoder is tying to encode the output from a previous lossy encoder. You will be encoding noise as well as real data.

There might be benefits in recoding at higher bitrate if you have a lossless source and are converting to a lossy output. This would minimize any degradation in the lossy output.

If you can it's far better to back to the original source and re-encode it at the higher bitrate you require.

By increasing the bitrate you won't have an higher sound quality.

Think about it this way: when it was converted from the original media (let's say a CD) it was compressed to fit the "content" in a smaller "box", and by doing so an amount of data has been lost (you may want to read about lossy and lossless formats). If you subsequently increase the bitrate, you are just making the "box" bigger, but the "content" is always the same.

First it's correct that you don't get more information from up sampling. But combining up sampling with a low pass (or interpolation) filter will get you a smoother curve. Passing this to the stereo should result in less noise produced from the stereo trying to reproduce the noise given by the original low sampling rate.

The important factor here is that you know something your stereo doesn't. Your stereo does not know noise from signal. It thinks that what you feed it is what you want. But you know the difference. You know you don't want the shape of the original signal, but a smoother version. So you can up sample and make a smooth curve, before feeding it to your stereo.

So this is not a case of adding more information, but reducing noise coming from low sample rate.

You can't "improve" the signal by re-encoding the output into another lossy format (mp3 etc.). It will always be worse than the original.

If you must re-encode it, the best result you can achieve is the same quality by choosing a losless codec like FLAC or ALAC. Or even uncompressed formats like WAV.

If there's no other source for your file you should keep the version you have.