TheProf

Wav's don't get many rave reviews because essentially you're copying the contents of the CD to your hard drive. While this maintains nearly perfect audio quality, it also takes up the maximum amount of space (10 MB/min as a general rule of thumb) and also is usually void of any artist/title/album data.

OGG is a good format, but then again, if your iPOD doesn't support it, no point bringing it into the discussion. And it is unreasonable to assume that you'll be encoding the wav file into both mp3 for your iPOD and ogg for your computer, might as well stick with mp3.

When you're talking about quality, you have 3 main variables
1) the steps involved in extracting the data from the CD
2) the steps involved in encoding the extracted data into mp3
3) what speakers will you be playing the media through

Details:
1) As many people have mentioned, Exact Audio Copy (EAC) is the way to go. The program will keep trying for hours to get the data off a CD, even if other programs say it can't. I've used it to pull data off scratched CDs and while it took 5 hours, it did it. It also has all the usual goodies such as looking up artist/title/album information from online databases and importing the information into the software, etc.

2) There are many encoders available. Which encoder you want to use depends on what target mp3 you want to create. Some encoders work better at high bitrate, others at low, some handle VBR, others are only good at CBR, and others can do ABR. Some forms of hardware can't handle VBR/ABR, and I'm not sure about the iPOD, but this is something you'll want to check. It used to be considered that 128 kbit/sec was what everyone used, but now most people are encoding to 192 for better quality. Generally going up to 256 kbit/sec is the max you want - above that you're just using up a lot of space for not much increase in sonic gain.
Most people recommend LAME for encoding their mp3s, and it is quite good. (Check out this comparison link). To make your life easier, as LAME is a command line program, either get a front-end for it (Like RazorLame) or just point EAC to the directory where LAME is installed, and EAC will rip and encode everything for you. Lame does have additional settings you can use to tweak the resulting mp3 (highpass/lowpass filters, quality values, and many more) but normally you can just use the preset ones and you should be fine.

3) A lot of the above discussion is dependent on what you will be playing the output on. A pair of bud earphones that come standard with an mp3 player will barely be able to produce the quality needed to tell the difference between 128 and 192 kbit/sec. Playing them through a 5.1 stereo system however, and you'll start to notice things at the high and low ends of the spectrum - muffling, softer pitches, and so on.

I'd recommend you take some of your favorite music tracks that you like to listen to, rip and encode using a few different methods, and then listen to them the way you'd normally be listening to your full collection. Whichever one is most pleasing to your ear, go with that.

Final points - as people said correctly, you cannot reliably go from lossy -> original. Upon encoding the lossy format the data is actually removed from the stream. The computer will try to recreate it, if you rip a WAV file, encode it to mp3, decode it back to wav, and then load that and the original wav file into an audio editor, you'll notice a difference in the waveforms. And if you really want to see/hear the difference, invert one of the forms and subtrack the waveforms. The resulting form is what was lost.

And for your own ease of listening, make sure you normalize your data as your rip and encode. It saves having to readjust the volume everytime you change albums!

Good luck.