There is no default sampling rate for ifft because ifft does not actually sample anything. If the original fft is calculated correctly (with the correct frequency vector) the time vector for ifft can be obtained from the length of the fft and the Nyquist frequency.
If the absolute value of the fft from 0 Hz to the Nyquist frequency is available, it is necessary that the phase information is also available, so that the complex fft can be reconstructed. The second ‘half’ of the fft is then the complex conjugate of the first ‘half’ of the fft, concatenated onto the end of it. Then you can take the ifft of it. That will be the length of the original signal, and the sampling interval will be 
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