SPEECH ENCRYPTION

ABSTRACT
THE increasing relevance of multimedia applications is placing a great demand on content protection and customer privacy. Communications can be intercepted, especially over wireless links. Since encryption can effectively prevent eavesdropping, its use is widely advocated. Unfortunately, encryption and decryption are computationally demanding, a severe problem in mobile, portable devices, where power consumption needs to be reduced as much as possible. The need for encryption in wireless systems has led to intense activity aimed at reducing the complexity of encryption algorithms. A new and computationally efficient procedure for encrypting band pass signals is explained here. This technique utilizes spectrum modification in conjunction with the sample-masking algorithm. By avoiding the more complex mixing process used in standard bandwidth-minimizing techniques, this new technique reduces the computational burden by a large amount Introduced here is a new technique for encrypting bandpass signals. The technique linearly filters a sample-masked signal, modifying the encryption component of the cryptogram to place its spectrum within the frequency band of the signal. Sample masking is a recognized means of encrypting baseband signals while containing the spectrum of the cryptogram within the signal badwidth. A drawback occurs for bandpass signals, owing to the processing required in the mixing procedures used ordinarily to convert the signal from bandpass to baseband and back. In contrast, the linear filter modifies the encryption component of the cryptogram without processing the signal component, reducing the computational burden of encrypting and decrypting by an order of magnitude.

When the maximum frequency is an integer multiple of the minimum frequency, the filter can output the same information symbols that appear at the input and replace the redundant symbols with symbols that control the passband of the output signal. For voice bandwidths of 300-3000 Hz sampled at 6000 Hz, redundancy symbols (one in every ten or 600 samples/s) are removed, the information symbols (nine in ten or 5400 samples/s) are encrypted, and the redundancy symbols are replaced by bandwidth-controlling symbols. Experimental results of clear voice transmitted over simulated telephone lines validate the technique. .

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INTRODUCTION
The idea of encryption of messages is probably as old as the fact that messages have been sent between humans. A lot of different techniques have been developed over the centuries beginning with Caesar cipher all the way to modern standards like advanced encryption standard (AES). In fact, most of the encryption systems were meant to encrypt text messages; the development of encryption systems for voice is in turn a lot more difficult and mostly not satisfying achievable with analogue techniques. Due to that problem, much pioneering work for many digital capabilities was performed while inventing a system to provide secure voice communications. The implementation of speech encryption dates back to World War II when secure communication was paramount to the US armed forces. During that time, noise was simply added to a speech signal to prevent enemies from listening to the conversations. Noise was added by playing a record of noise in sync with the speech signal and when the speech signal reached the receiver, the noise signal was subtracted out, leaving the original speech signal. In order to subtract out the noise, the receiver need to have the exact same noise signal and the noise records were only made in pairs; one for the transmitter and one for the receiver. Having only two copies of records made it impossible for the wrong receiver to decrypt the signal. To implement the system, the army contracted Bell Laboratories and they developed a system called SIGSALY. With SIGSALY, ten channels were used to sample the voice frequency spectrum from 250 Hz to 3 kHz and two channels were allocated to sample voice pitch and background hiss. In the time of SIGSALY, the transistor had not been developed and the digital sampling was done by circuits using the model 2051 Thyratron vacuum tube. Each SIGSALY terminal used 40 racks of equipment weighing 55 tons and filled a large room. This equipment included radio transmitters and receivers and large phonograph turntables. The voice was keyed to two 16-inch vinyl phonograph records that contained a Frequency Shift Keying (FSK) audio tone. The records were played on large precise turntables in synch with the voice transmission. Attainably the system was broken by the Germans. In the 1940th a far more secure system again by Bell Labs was invented and used which was called Sigsaly (which was just a cover name and not an acronym). This 50 ton heavy, complex system that is shown in Figure below was the first secure voice encryption system.

Sigsaly exhibit at the National Cryptology Museum.

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The development of Sigsaly was a truly groundbreaking development; it contained pioneers work in not less than eight achievements: The first realization of enciphered telephony, The first quantized speech transmission, The first transmission of speech by Pulse Code Modulation (PCM), The first use of companded PCM, The first examples of multilevel Frequency Shift Keying (FSK), The first useful realization of speech bandwidth compression, The first use of FSK-FDM (Frequency Shift Keying-Frequency Division Multiplex) as a viable transmission method over a fading medium, The first use of a multilevel "eye pattern" to adjust the sampling intervals.

The Human Voice

For the development and understanding of a voice scrambling system it is existentially to know some basic parameters of the voice itself. In fact the theory is quite more complex. The human voice is simply a sound or audio signal which is generated by a human being to communicate with one and another. Therefore, humans use their vocal folds to modulate the air stream, coming from the lungs, into vibrations which make, with the rest of the human sound forming system (like: mouth hole, tong etc), a sound. This sound is treated by some different effects to make a variety of sounds like vowels or consonants. Hence to the difference of the anatomic of every human the voice of two people never sound the same. This is due to the different fundamental frequencies (often also referred as pitch of the voice). These fundamental frequencies are mainly generated by the vocal cords which create through their opening and closing a periodic signal with the fundamental frequency. The frequency distribution for less bandwidth consumption the frequency used in telephony ranges from approximately 300 Hz to 3400 Hz, the fundamental frequency of most speech falls in this range, so that there are always harmonic series in the voice signal to give the listener the feeling of hearing the fundamental tone. With some guard bands added it can be used at a sample rate of 8 kHz.

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DIGITAL ENCRYPTION STANDARDS

Since IT security became a growing market in the past, there are a lot of different standards to choose from. Here is a list of some famous ones: DES (Data Encryption Standard) AES (Advanced Encryption Standard) / Rijndael FEAL (Fast Data Encryption Algorithm) IDEA (International Data Encryption Algorithm) Safer (Secure and Fast Encryption Routine) RC5 (Rivests Code 5) and RC6 (Rivests Code 6)

Digital encryption has the big advantage that there are little possibilities for cryptanalysis compared to analogue scrambling. Except some advanced mathematical approaches, there are only brute force attacks available which are usually not in polynomial time2 computable tasks. But on the other side because encryption is a digital process, it can be critical to ensure that the encrypted data arrives correctly at the receiving end if the transmission channel is not sufficiently reliable. If the data is corrupted, it will not be decrypted correctly causing degradation in voice quality. Furthermore, if data are lost (or added), the encryptions may lose synchronization and communication will be lost. Speech and audio signals are an essential component of most multimedia applications. Not only speech services are the basis of the huge wireless telephony industry, but speech is also the most important component of advanced audiovisual services such as videoconferencing and news broadcasting. The benefits of partial encryption of speech signals could thus be very significant. At the heart of the technique is a linear filter that modifies the spectrum of a baseband sequence by inserting additional terms to make the sequence band pass. The filter modifies the encryption component of the cryptogram, placing its spectrum within the frequency band of the signal while leaving the signal component of the cryptogram unchanged. The origin of the signal is irrelevant to the technique. The signal may be analog or digital. We may also begin with a base band signal, then shift its spectrum to produce a band pass signal. Experimental results for encrypting voice, with a spectrum extending between 300 and 3000 Hz over a simulated telephone line (a 300-3000 Hz band pass channel), have validated the technique.

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SAMPLE MASKING
The sample-masking algorithm is an encryption technique that allows the high degree of security attainable by ciphering techniques and retains the spectral width of the original analog signal without expensive bandwidth-reduction equipment. The technique overlays a masking sequence on the sampled version of the original analog signal. The masking sequence, consisting of a pseudo random sequence with amplitudes uniformly distributed over an interval (- P, + P), is added modulo 2P to the sequence of signal samples having amplitudes limited to the same interval or a subinterval. The resulting cryptogram consists of uniformly distributed terms that are independent and uncorrelated and that appear to be pure noise terms to a listener who does not possess the key. The modulus addition may be performed on either analog or digital sequences. However, a straightforward way to obtain a pseudo random sequence of N-bit terms is to group the output sequence of a pseudo random binary generator into N-bit subsequences to form a pseudorandom sequence with N-bit terms. The signal sequence is digitized to N bits and the two sequences are added digitally modulo 2N. If a baseband analog signal, containing no frequency components above a frequency B, is sampled at its Nyquist rate 2 B, then the principal spectrum of the cryptogram will contain no frequencies above B. However, if a band pass analog signal of spectral width B, having no frequency components above W ( W > B), 'is sampled at a rate 2W to avoid aliasing, then the principal spectrum of the cryptogram has width W, which is greater than. that of the original signal: A standard procedure for avoiding bandwidth spreading when encrypting a band pass signal is to translate the signal down to baseband where the signal can be sampled at a rate 2B (at a rate B in each channel if quadrature channels are used). Encryption is performed at baseband and the cryptogram translated back up to occupy the same bandwidth B that the original signal occupied.

SPECTRUM MODIFICATION
The spectrum modifier technique is described by two techniques: DEPT. OF ECE, PESSE. 5

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By taking samples at an average sampling rate equal to twice the spectral width of the signal. A band pass signal sampled at this rate must, in general, be sampled at non uniform intervals to prevent aliasing. For 'any arbitrary sequence of information symbols with an average symbol rate 2B, there exists a continuous waveform with its spectrum lying entirely within the frequency interval ( W, W+ B) where W is any arbitrary frequency. The sequence can be reconstructed by sampling the waveform at some determined set of (possibly non uniform) sampling instants. The first principle is applied to the signal sequence and the second to the encryption sequence. One filter simultaneously reconstructs the original signal sequence and shapes the frequency spectrum of the encryption sequence. In the following analysis, the analog signal is considered to be voice which has been limited to the frequency interval 300-3000 Hz and is to be transmitted through a channel with a pass band of 300-3000 Hz, such as a voice-grade telephone line. The technique, however, is quite general. When the maximum frequency is an integer multiple of the minimum frequency, the technique may be used to shift the frequency spectrum of any baseband signal to a band pass signal. (A generalization of this technique transforms the spectrum of an arbitrary signal into a spectrum confined to an arbitrary frequency interval or set of intervals equal in bandwidth to that of the original spectrum.) The analog voice signal is sampled at a 6000 Hz rate, but every tenth sample is omitted or deleted. The reduced-voice sequence has non uniform sampling intervals and an average sampling rate of 5400 Hz, twice the voice bandwidth. It is encrypted according to the sample-masking algorithm by modular addition with a sequence of encryption symbols. Consequently, this sequence has an average symbol rate of 5400 Hz. The encrypted sequence (cryptogram) is then passed through a linear filter, called the spectrum-modifier filter, which inserts terms that modify the spectrum of the cryptogram at those instants where voice samples have been deleted, the sequence thus lies in the frequency interval 300-3000 Hz, and, augmented by the inserted terms, has uniform sampling intervals and a sampling rate of 6000 Hz. Since the filter is linear, each inserted term may be considered as the sum of a voice component and an encryption component. The voice components of the inserted terms are those values discarded from the voice sequence, and the encryption components are exactly those needed to put the spectrum of the encryption sequence in the frequency range 300-3000 Hz. Because the voice sequence is completely encrypted before passing through the spectrum-modifier filter and because this filter is linear, the security of the cryptogram is in no way compromised by the filter processing.

THE SPECTRUM-MODIFIER FILTER

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To understand the operation of the spectrum-modifier filter, consider Fig. 1. The cryptogram input to the filter follows two paths: one leading directly to the positive input terminal of an adder, the other leading to an ideal low-pass filter with a 300 Hz cutoff frequency. The output of the low-pass filter has no frequency component above 300 Hz (Fig. 2); thus, its output sampling rate can be compressed to 600 Hz without aliasing. Nine zeros are then padded in between each pair of adjacent samples, returning the sampling rate to 6 kHz without modifying the 0-300 Hz spectrum in the 6 kHz sequence. Thus, the spectrum below 300 Hz of the sequence at the adder's negative terminal input is identical to that at the positive terminal input, and the filter output has no spectral components below 300 Hz. The linear character of the spectrum-modifier filter allows us to follow the voice and encryption components of the cryptogram separately through the filter. We also invoke linearity to consider the voice sequence (reduced by deleting every tenth term) as the sum of two sequences-the original voice sequence and a second sequence with amplitudes equal but opposite in sign to the voice samples at the deletion instants and with zero amplitude elsewhere. Consider the original voice sequence passing through the spectrum-modifier filter. It appears at the adder's positive input terminal and at the low-pass filter input. Since it has no spectral components below 300 Hz, the output of the low-pass filter and the input to the negative terminal are the zero sequence. The output of the filter due to the original voice-input component is, therefore, the original voice-input component. Consider the second sequence, which has nonzero terms only in every tenth position of the sequence. From the property of low-pass filter every tenth output term is identical to the corresponding input term. The switch sets the remaining terms equal to zero, so that the input to the negative terminal of the adder is identical to the positive terminal input. The filter output is, in this case, the zero sequence. The total filter output when the reduced-voice sequence is entered into the filter is, therefore, the original voice.

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. Consider the encryption sequence passing through the spectrum-modifier filter. At the output of the low-pass filter is a sequence whose spectrum is identical to that of the input sequence below 300 Hz. It also has no spectral components above 300 Hz, so that there is no aliasing of the spectrum at the output of the switch. Therefore, the spectrum of the sequence at the adders negative terminal input is identical to the adders positive terminal input spectrum below 300 HZ. Thus, the filter output has no spectral components below 300 Hz. The total output of the spectrum-modifier filter, then, consists of two components: the original voice sequence and the original encryption sequence augmented by terms that confine its spectrum to within the frequency limits of the voice spectrum. Because it retains the input terms at the output of the spectrum-modifier filter, the receiver has no additional computational burden other than to regenerate the augmenting terms of the encryption sequence. For the example employed here, only every tenth output of the low-pass filter (Fig. 1) is used. The remaining 90 percent of the low-pass filter outputs need not be calculated. DEPT. OF ECE, PESSE. 8

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TRANSMITTER AND RECEIVER CONFIGURATIONS

In a typical transmitter configuration (Fig. 3) where analog voice encrypted by the sample-masking algorithm is transmitted through a 300-3000 Hz band pass channel, the voice spectrum is limited to the frequency range 300-3000 Hz and sampled at a rate R = 6000 Hz. The voice samples, converted to an N-bit digital number, are added modulo 2N to an encryption sequence consisting of N-bit pseudorandom terms. The cryptogram is passed through the filter, which modifies the input sequence by changing every one-tenth term so that the spectrum of the output sequence has no components below 300 Hz. The cryptogram is then converted to an analog waveform with a frequency spectrum in the range 300-3000 Hz, ready for transmission through the channel. The modulus addition may be thought of as the linear addition of three sequences: a voice sequence, an encryption sequence, and a modulus sequence (the terms of the modulus sequence consist of the values 0, k 2N). The modulus sequence is added to the linear voice-plus-encryption sum to keep the sum of the three terms within the amplitude range .

Of the three cryptogram components, the modulus sequence is the most critical for the receiver configuration (Fig. 4). Voice is an analog signal, which can still be quite intelligible after undergoing the severe distortion produced by, say, a voice-grade telephone line; the encryption sequence is generated in the receiver so that no information transfer through the channel is required by the encryption component. However, the modulus sequence, essentially a digital signal, must be recovered from the received signal in the receiver in order to remove it from the cryptogram.

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A disadvantage of the sample-masking algorithm is the relative difficulty of decrypting the cryptogram after transmitting it through a less-than-ideal channel. The receiver configuration suggests a means of easing the removal of the encryption and modulus components from a cryptogram distorted by the channel. Instead of an equalizer, typically employed to remove the distortion from the cryptogram, we use filters that replicate the channel impulse response to distort the encryption and modulus sequences generated in the receiver in the same way that the encryption and modulus components are distorted by the channel. The distorted replicas of the encryption and modulus sequences are then used to remove the corresponding components from the cryptogram, leaving clear voice. For the same filter complexity, a channel-replicating filter removes significantly more of the encryption and modulus sequences than does a channel equalizer. After the analog signal out of the channel is digitized, the encryption component is removed by subtracting out a replica of the distorted encryption component, leaving the voice and modulus components of the cryptogram. The value of the voice component lies in. the interval

If a voice-plus-modulus term of an undistorted cryptogram is equal to or larger than + 2N-1, the modulus component must be +2N; if the term is less than 2N-1, the modulus term is -2N; if the term is between 2N-1 and + 2N-1, the modulus component is zero. In order to use this simple algorithm to recover the modulus sequence, an equalizer must often remove much of the distortion from the cryptogram to decrease the error rate when recovering the modulus sequence. To further reduce the error rate caused by distortion and channel noise in the cryptogram, voice amplitudes may be limited to some fraction, say 75 percent, of the maximum allowable value of the encryption terms.

Since the amplitude of an undistorted voice-plus-modulus term cannot lie in either of the intervals DEPT. OF ECE, PESSE. 10

SPEECH ENCRYPTION given below (-2N-1 -Q, -2N-1 +Q) or ( 2N-1 -Q, 2N-1 +Q) Where Q is 25 percent of 2N-1, an error can possibly occur only if the distortion on a term changes its value by an amount Q or greater. The algorithm for determining the modulus components does not recover the terms generated by the spectrum-modifier filter. They are regenerated in the receiver by passing the recovered modulus sequence with every tenth value missing through a spectrum-modifier circuit. The completed modulus sequence is distorted by a channel-replicating filter, and the distorted sequence is subtracted from the cryptogram from which the encryption sequence has already been removed. The remaining clear-voice sequence is then converted back to an analog signal. It should be stressed here that, in general, to recover data passed through a channel requires a channel equalizer. But precisely because voice is the signal, we need only to strip off the encryption and modulus components in the receiver leaving clear voice having, ideally, the same distortion it would have had if the clear voice alone had been transmitted through the channel. Nor does the intended receiver have to recover the encryption sequence: it already has a copy of the original encryption sequence.. It need only recover the modulus component, which is a binary sequence. Thus, the problem of trying to equalize a 2N level signal to the point where the magnitude of each term of the sequence is within one-half a quantization level of its true value is avoided. The problem is reduced to the more tractable one of equalizing the modulus sequence to within Q quantization levels of its true value, as explained above. For these same reasons, the channel-replicator technique mitigates the problem of recovering speech from a cryptogram sent over a nonlinear channel. However, the difficult problem of evaluating the performance of this technique on nonlinear real channels is not dealt with in this paper. An experiment was designed to subjectively compare (by listening) clear voice transmitted over a simulated telephone line to decrypted voice transmitted over the same line. The root-meansquare values of the encryption and modulus residual sequences in the decrypted voice were measured to determine the performance of the replicating filter and equalizer. The speech was converted to a 16-bit digital sequence at a 12 kHz sampling rate. Then the speech spectrum was limited to 3 kHz by a low-pass filter (Fig. 5) and the sampling rate was compressed to 6 kHz.

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A high-pass filter (Fig. 6) removed spectral speech components below 300 Hz, limiting the speech spectrum to the frequency range 300-3000 Hz.

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To take full advantage of the 16 bit resolution, we normalized the amplitudes of the terms in the sequence by multiplying each of them by a factor to make the magnitude of the largest term 75 percent of the maximum value 215 - 1 of the 16 bit (15 bit plus sign bit) terms. After encryption, the sequence was normalized at the output of each filter by a factor that made the magnitude of the largest term equal to 100 percent of the maximum allowable amplitude 215 - 1. The 16 bit resolution of the speech sequence and the lengths of the filters are not optimal values for use with the spectrum-modifier filter. They are conservatively chosen and determined in part by the design and simulation tools used. The computer-simulated transmitter and receiver configurations are shown in Figs. 3 and 4, respectively. The impulse response of the channel, also computer simulated, is the digitized impulse response of a commercial voice grade telephone line simulator. The simulator models only the linear distortion of a telephone line. To test the performance of the technique on an actual telephone line would require us to take into account both the noise and the nonlinear characteristics of the line. The ideal low-pass filter in the spectrum-modifier filter is approximated by a weighted sin x/x filter truncated to a length of 599. To compensate for a delay of 300 sampling intervals in the real filter compared to that of the ideal filter, a similar delay was required before the input to the positive adder terminal. To avoid a duplication of delay lines, the two parallel paths were combined into one filter (Fig. 7). Because there was no interval separating components of the principal spectrum from those of the first-order spectrum at a 6 kHz rate, the decrypted-voice sequence was interpolated to produce a 12 kHz sequence before passing it to the digital-to-analog converter. This lessened the requirements on the analog post filters for removing the higher order spectra. In order to use practically achievable tap-weight values for the equalizer and replicators, the values used were obtained in a separate computer program which simulated filters that automatically determine the tap values, using the stochastic update method to obtain the minimum mean-square error output. The values represent the equivalent of 1 s worth of real-time determination of the tap weight values.

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PROCEDURE AND RESULTS

A 5 s speech segment was digitized, filtered to remove spectral components outside the frequency range of 300-3000 Hz, and normalized to take full advantage of its 16 bit resolution. The 6 kHz sequence was then processed in three ways. The unencrypted speech was passed through the simulated telephone line, converted back to an analog waveform, and recorded. The sequence was encrypted, sent through the line, and decrypted. The resulting clearvoice sequence was converted to an analog waveform and recorded. The sequence was encrypted and sent through the line, but not decrypted. The received cryptogram was converted to an analog waveform and recorded.

None of several listeners who compared the two clear voice recordings noticed any degradation in the voice obtained from the cryptogram compared to that obtained from the clear-voice transmission. This is consistent with the measurement of the encryption and modulus residues, which were 31 dB below the voice signal. The undecrypted cryptogram, which is what an eavesdropper would hear, sounded like pure noise. The presence or absence of voice in cryptogram could not be detected.

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MATHEMATICAL MODEL
The presence of voice and encryption components through the cryptogram through spectrum modifier filter is modeled .The switch shown in Fig. 1 is replaced by the equivalent combination of a sampling-rate compressor & sampling-rate expander, as in Fig. 8.

The compressor generates an output sequence consisting of every Lth term of its input sequence; the expander generates' an output sequence by padding in L - 1 zeros between each adjacent pair of terms of its input, sequence. Because the low-pass filter has an non zero frequency response of bandwidth 2/L such that

there is no spectrum-aliasing caused by the sampling-rate compressor, 'and its combination with the ideal low-pass filter forms a decimator. Consider an arbitrary sequence {x(n)} at the input to the spectrum-modifier filter.' The spectrum of the sequence out of the decimator due to the input sequence {x( n )} is

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The spectrum

of the sequence

out of the sampling rate expander is

That is, the spectrum between '- /L and + /L of the input sequence {x 3(n)} to the adder's negative. Terminal is identical to the input sequence { x ( n ) } to the positive terminal. Therefore, the output sequence {y(n)}. Of the spectrum modifier filter has no spectral component below /L. For the case of voice in the frequency range 300-3000 Hz sampled at 6000 Hz, L = 10 and the filter-output sequence contains no spectral components below 300 Hz. Consider next a sequence {x(n)} at the input to the spectrum-modifier filter such that a term x ( n ) = 0 if n/L is not an integer. The output sequence x1(n) of the low-pass filter with impulse response

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SPEECH ENCRYPTION That is, every Lth output is identical to its corresponding input. The compressor samples these terms and the expander inserts L - 1 zeros between each adjacent pair of terms to regenerate the sequence at the low-pass filter input. The inputs to the adder terminals are identical, so that the spectrum-modifier output due to an input with nonzero terms only in every Lth position is the zero sequence.

CONCLUSIONS:
The technique described in this paper confines the spectrum of an arbitrary sequence of rate R with the frequency band (W,nW) where n and W satisfy the inequality

Unique to this technique is its ability to retain the values of the original sequence is modified by inserting additional terms which put the spectrum within the desired frequency band. The required computational rate is lower because the original spectrum need not be modified. A comparison was made between the computational rates using the new technique and using a standard mixing technique for encrypting voice. The new technique required one-tenth the rate of the standard mixing technique mostly because only one-tenth of the terms in the output sequence needed to be computed.

APPLICATIONS:
-MILITARY APPLICATIONS - PRIVACY FOR CONSUMERS USING WIRED AND WIRELESS MEDIUMS

REFERENCES:
1) IEEE JOURNAL ON SELECTED AREAS IN COMMUNICATIONS, VOL. SAC-2, NO. 3, MAY 1984.An Efficient Technique for Sample-Masked Voice TransmissionRONALD J. COSENTINO, MEMBER, IEEE, AND STEPHEN J. MEEHAN 2)E. R. Brunner, Efficient scrambling techniques for speech signals, in ZCC Conf. Rec., vol. 1, 1980, pp. 16.1.1-16.1.6. 3)A. Kohlenberg, Exact interpolation of band-limited functions, J. Appl. Phys., vol. 24, Dec. 1953. 4) R. J. Cosentino and S. J. Meehan, Secure voice-bandwidth modem, presented at the 1982 Carnahan Conf. Security Technol., Lexmgton, KY, May 12, 1982.

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