Hi Niki,
In the 1930's paper by C.E.Shannon, of Bell Labs, titled; 'A mathematical theory of Communication', he showed how the sampling rate of an analog waveform needed to be at twice the highest frequency required to be transmitted.
In the telephone system, the bandwidth is in the 0.3 to 4.0 KHz range and therefore the sampllng rate needs to be 8 Kbits/sec. In modern phone systems the sampling rate for speech telephony is 8k bits/sec. For the bell system, there are 24 channels stacked to give a 1.44Mbits system. The CCITT system samples at the same rate and the 32 channel system runs at 2Mbit/sec.
In your project you need to sample at a rate of twice the highest frequency you wish to transfer through your system. Generally for speech, a bandwidth of say 6 to 7 Khz would be sufficient. Telephone quality would use a sampling rate of 8 Ksamples/ sec.
With audio CD's, the normal sampling rate is 44.1 KHz and this is slightly higher than twice the usual 20 to 20000 Hz bandwidth required for high quality audio. If you look at the various sampling rates for audio recordings made on say iTunes, you can select the sampling rate to suit your fidelity requirements.
hope this helps.
pr.