Method and apparatus of providing bit count integrity and synchronous data transfer over a channel which does not preserve synchronization

The present invention relates generally to a data transfer communication system, more specifically to the transfer of a synchronized data stream over a medium that does not retain the synchronization characteristics of the data stream.

In radiotelephone communication systems, many users communicate over radio channels to connect to other radiotelephone and wired telephone systems. Communication over wireless channels can be one of many multiple access technologies. These multiple access technologies include a time division multiple access system (TDMA), a frequency division multiple access system (FDMA), and a code division multiple access system (CDMA). CDMA technology has many advantages. A typical CDMA system is assigned and incorporated herein by an assignee of the invention, the name issued on February 13, 1990 by K. Gihousen et al., "Satellite or terrestrial repeater." It is described in US Pat. No. 4,901,307 of "Spread Spreading Multi-Access Communication System Using".

The patent just described discloses a multi-access technology in which a large number of mobile phone system users, each with a transceiver, use CDMA spread spectrum communication signals to communicate through satellite repeaters or ground base stations. When using CDMA communication, the frequency spectrum can be reused many times to increase system user capacity.

In a CDMA cellular system, each base station provides coverage in a limited geographic area, and remote units in that coverage are coupled to the public switched telephone network (PSTN) via switches in the cellular system. When the remote unit moves to the scope area of the new base station, the routing selection of this user's call is transferred to the new base station. The base station-remote unit signal transmission path is called a forward link, and the remote unit-base station transmission path is called a reverse link.

In a typical radiotelephone communication system, the remote unit transmitter can use a bocoding system that encodes voice information in a variable rate format. For example, the data rate may be reduced by pausing voice activity. Lower data rates reduce the level of interference to other users caused by remote unit transmission. In the receiver, or especially in connection with the receiver, the bocoding system is used to reconstruct the audio information. In addition to voice information, only non-voice information or a mixture of the two can be transmitted by the remote unit.

If the remote unit generates its own data for transmission, the internal vocabulary will have four different rates, eg about 8,000 bits per second (bps), based on voice activity in a frame of 20 milliseconds (ms). , 4,000 bps, 2,000 bps and 1,000 bps generated from digital samples of coded data. Each frame of bocoder data is formatted with overhead bits as 9,600bps, 4,800bps, 2,400bps, and 1,200bps data frames. The highest rate data frame corresponding to a 9,600 bps frame is called a "full rate" frame, the 4,800 bps data frame is called a "1/2 rate" frame, and the 2,400 bps data frame is called a "1/4 rate" frame. It is said that a 1,200 bps data frame is called a "1/8 rate" frame. The rate information is not included in the data in either the coding process or the frame formatting process. A vocoder suitable for application in this environment was issued on May 9, 1995, and the name transferred to the assignee of the present invention is described in US Pat. No. 5,414,796, "Vocoder". If the remote unit receives data from an external source such as a terminal equipment unit, the remote unit will continue to process the data in this variable speed frame format.

When the first cellular telephone spectrum license was issued by the government, one of the restrictions on the use of spectrum was the inability of carriers to provide dispatching system services. However, due to the many advantages of CDMA systems and the issue of placement and maintenance of dedicated dispatch systems, the government will re-examine this issue. The government itself benefits greatly from such services.

Typical radiotelephone and wired telephone services provide one-to-one (point-to-point) services, while dispatch services provide one-to-many services. Common uses of dispatch services are local police radio systems, taxi dispatch systems, federal secret operations operations communication systems and all military communications systems.

The basic model of the dispatch system consists of the user's broadcasting net. Each broadcast net user monitors a common broadcast forward link signal. If the net user wants to communicate, he presses the push-to-talk (PTT) button. Typically, the user voice of the communication is routed from the reverse link via the broadcast forward link. Ideally, the dispatch system allows terrestrial and wireless access to the system.

If a government agency wishes to use a dispatch service, in addition to the unique privacy provided by the CDMA waveform, this agency may wish to use cryptographic mechanisms to prevent further interference. .. Cryptographic mechanisms are generally operated on the basis of internally generated clocks to generate data at a constant rate.

In order for cryptographic mechanisms to be used with wireless systems, cryptographic mechanism data rate, clock and bit total integrity requirements must be accepted.

[Outline of Invention] The present invention is an efficient buffering scheme for adapting a transmissive wireless link protocol that services a constant rate bit stream that requires total bit integrity and low latency. The terminal device provides a constant rate bit stream. A wireless link is used to carry a constant rate bitstream to a destination using a variable rate, non-frame-based error-free protocol. At the receiving end, the constant rate bit stream must be reconfigured to maintain bit total integrity. The buffering system must produce a minimum fixed delay that is consistent with the needs of the voice service system.

To achieve these requirements, a field is added to each frame. The length region itself must consist of a small number of bits to minimize the impact of the length region on the link's overall bit carrying capacity. The length region is a modulo value that is incremented in proportion to the size of the corresponding data frame. If a frame is erased, the number of bits contained in that frame can be determined based on the length region of the next non-erased frame received. At the receiving station, the queue receives the frame data and produces a constant rate bit stream. When a loss is received, a number of fill bits is added to the queue equal to the maximum number of bits that may have been contained in the lost frame. These filling bits can be used immediately by the queue. When the next non-erased frame is received, if it is determined that the lost frame contained less than the maximum number of bits, the extra bits will be overwritten by the data contained in the next non-erased frame. To.

[Explanation of preferred embodiments] The features, objectives, and advantages of the present invention will become even more apparent when the detailed description described below is referenced in combination with the drawings.

Figure 1 shows a typical dispatch system. In a preferred embodiment, remote units 10, 20, 22 and 24 may function as both a dispatch unit and a one-to-one telephone. In Figure 1, remote unit 10 is the currently active talker, and remote units 20, 22 and 24 are currently passive listeners. Base stations 30, 32 and 34 provide broadcast forward link channels to remote units 10, 20, 22 and 24. Base station 30 also receives a reverse link signal from an operating remote unit. The Mobile Switch Center (MSC) 38 coordinates the control signals from the base station to the base station. In a preferred embodiment, the control signal is commonly referred to as IS-95, "Mobile Station-Base Station Compatibility Stanndard for Dual-Mode". Wideband Spread Spectrum Cellular Systems) According to TIA / EIA / IS-95, remote units are referred to as mobile stations. Communication manager 40 controls the broadcast network to give priority to requests when two remote units press the Push-to-Talk (PTT) -button at the same time.

Although preferred embodiments illustrate the invention for use in coupling encrypted voice systems to one-to-one or dispatched wireless CDMA systems, general principles can be applied to various parts of the digital environment. .. For example, the same principles can be applied to systems deployed using time division multiple access (TDMA) or other digital transmission techniques. The digital data can be fax or computer data. In general, the invention is broadly applicable to any synchronized data stream transmitted over a medium that does not maintain signal synchronization. Some examples of such systems are asynchronous transfer via synchronous frame format video transmission system, packet data burst carried by synchronous bearer (isochronous system) service, and connection oriented synchronous transfer. Multiplexed voice and data traffic carried by mode (ATM).

FIG. 2 shows a typical embodiment of the present invention. Figure 2 describes a secure one-to-one voice connection between a remote unit and a terrestrial telephone. The same technique can be applied directly to a one-to-one connection between two remote units, or directly to a dispatch system. The remote unit 110 is equipped with a Crypto Lump (clump) 100. Clamp 100 provides a stable flow of data bits to the remote unit 110. In the clamp 100, the clock 102 has a frequency f₁Generates an independent clock that works with. Clock 102 has frequency f₂CDMA clock and frequency f operating on₃May drift against PSTN clocks running on. The clock 102 is used to drive the vocoder 104 and encodes the audio signal received by the speaker and microphone 108. The output of the vocoder 104 is used to drive the encryption / decryption 106. The output of the encryption and decryption 106 is the input to the remote unit 110. The output of the encryption and decryption 106 is typically secure voice, but the CDMA connection with remote unit 110, base station 118 and network connection function 124 is the data output from the standard digital equipment terminal unit. It works with that data as if. The speaker and microphone 108 are shown to be inside the clamp 100, but may be inside the remote unit. In such cases, an audio signal connection between the clamp 100 and the remote unit 110 is required.

One aspect of the encryption and decryption process is that a stable flow of data is generated at the end of the encryption and the stable flow of data must be regenerated at the end of the decryption for the decryption process to work effectively. That is. The decryption process can tolerate errors in the input data and provide valid results, thus not requiring error-free links. The important situation is that the bits going to the decryption process must have same time alignment with each other, as well as coming out of the encryption process. With only one bit of data lost, the decryption process produces scraps rather than valid data. The process of supplying a stable flow of data bits is called synchronization. The process of supplying the same number of bits as leaving the encryption process to the decryption process is called maintaining the integrity of the total number of bits.

Returning to Figure 2, the clamp 100 produces a stable flow of encrypted data bits. Such encrypted data is typically generated at 4800 bits per second. In contrast to the stable flow of bit output from clamp 100, remote unit 110 produces variable rate frame data according to IS-95. According to IS-95, the remote unit 110 produces a frame of data for 20 milliseconds (ms). A frame of data can take one of four different rates, approximately 8,000 bits per second (bps), 4,000 bits per second (bps), 2,000 bits per second (bps) and 1,000 bits per second (bps). However, it depends on the rate at which the data is generated and received. Each frame data is formatted with overhead bits as 9,600 bps, 4,800 b, ps, 2,400 bps and 1,200 bps data frames. The highest rate data frame corresponding to a 9,600 bps frame is called a "full rate" frame, the 4,800 bps data frame is called a "1/2 rate" frame, and the 2,400 bps data frame is called a "1/4 rate" frame. It is said that a 1,200 bps data frame is called a "1/8 rate" frame.

The bit count pay load for a full-rate frame is 160 bits. The total number of bits in one half-rate frame The payload is 80 bits. If Clamp 100 is generating data at 4,800 bps, it will generate 96 bits between each 20 ms frame interval. Therefore, the remote unit 110 produces dithering for a combination of full-rate and 1/2-rate data frames and adapts to the output of the clamp 100. For each frame, the remote unit 110 adds one length area. The length region itself must consist of a small number of bits to minimize the effect of the length region on the overall bit carrying capacity of the link. The length region is a modulo index generated by adding the current flume length (ie, the number of bits) to the value of the length region sent in the previous frame. index). In addition, other CDMA control signals, control and overhead bits are added to the frame. There, the frame is convolutionally coded. The encoded bits are interleaved. The interleaved bits are orthogonally Walsh-encoded and diffused with a mask of pseudo-random noise PN code. The masked spread signal is offset quadrature shift key (OQPSK) modulated in the I and Q channel spread sequences and transmitted from antenna 112 over radio link 120.

The base station 118 receives the signal of the remote unit from the wireless link 120 via the antenna 114. Base station 118 excludes OQPSK modulation and spreading masks. Base station 118 Walsh decodes the unspread signal and deinterleaves the signal. There, the signal is decoded so that it is decoded by the Viterbi decoder and transmitted from the base station 118 to the network connection function (IWF) 124.

The IWF124 provides the functionality required for the clamp 100 to make a network connection with the secure telephone unit 130. The physical implementation may include a pool of modems. The IWF124 outputs pulse code modulation (PCM) data to the public switched telephone network (PSTN) 128. The PSTN128 conveys PCM-encoded data to a secure telephone unit (STU) 130. The data flow is deciphered and devocoded in the STU130, and the audio signal is output to the final listener. The link from STU130 to Clamp 100 works much like the link above.

The IS-95 radio protocol is for carrying voice signals. Due to the nature of the audio signal, it does not require a fully reproduced copy of the original digitized audio signal because it is a long-distance communication conversation of comprehensible quality. Therefore, if an excessive error occurs for one frame, the frame can simply be lost. When the number of such disappearances is kept to a minimum, the resulting audio impact is minimized. Therefore, the IS-95 link does not provide error-free communication by nature.

If the IS-95 link is used to provide a data connection that requires error-free data transmission, an additional protocol layer may be added to detect frame loss. Based on the detection of frame loss, the receiving station can request the re-transmission of the frame. However, such a mechanism is unacceptable when dealing with synchronous data connections that carry voice data. Error detection and frame iterative operations cause delays in the system. In a synchronous system, the maximum delay introduced by such a system would have to be permanently inserted by the buffering mechanism. Such delays are unacceptable in voice systems. Because they are large enough to be detected by the end user.

Since the IS-95 radio interface does not provide synchronous data transmission with bit total integrity, the present invention provides a new protocol stack layer on top of the existing protocol stack. Figure 3 shows a protocol stack with a newly added adaptive layer for the Transparent Radio Link Protocol (TRLP). The adaptive layer can be used in connection with any user traffic that has a constant bitstream, whether the bitstream is generated by secure voice terminal equipment or by any synchronous data source. The adaptive layer uses a buffering mechanism and the associated lost data replacement algorithm for lost data in the transparent RLP layer to restore traffic synchronism and bit total integrity. The advantage of the present invention is to minimize the real-time queue delay incurred by the data.

Figure 3 shows a protocol stack with an adaptive layer of transparent wireless link protocol. Clamp 100 provides a constant stream of data bits using the interface defined in Japan Electronics and Information Technology Industries Association (EIA / TIA) Document 232-E. The remote unit 110 uses the same protocol to receive data. The APP layer is a standard modem AT command processing layer. The layer labeled AL is the adaptive layer. For reverse links, the adaptive layer within the remote unit 110 transforms a constant rate bitstream into a series of octets, which are sent to the TRLP layer. In addition, the adaptive layer has a frequency f₁Clamp 100 clocks operated by and frequency f₂Provides synchronization between clocks in the remote unit 110 operated by. The IS-95 layer provides data and signaling for wireless link interfaces, including the coding, interleaving, spreading and OQPSK modulation briefly described above.

At base stations 118 and IWF128, the IS-95 layer removes the IS-95 operation and outputs either a data frame or the loss of each frame of data it intends to receive. TRLP receives the frame data and outputs the octet of the data. The adaptive layer receives the incoming set of data octets and frame loss indications and creates a constant rate bitstream.

The adaptive layer is composed of the queue 150 shown in FIG. 4, which buffers the output of TRLP and generates a constant rate bitstream. The queue 150 does not start outputting a constant rate bitstream until the buffer prefill X region 154 and the buffer prefill Z region 156 are full of bits. Obviously, the buffer prefill X region 154 introduces an unavoidable fixed delay. The buffer prefill X region 154 has a frequency f at clamp 100.₁Clock operating at and frequency f at STU130₁Occupies any clock drift between clocks operating in. The size of the buffer prefill X area 154 is determined by the system specifications. For example, in a preferred embodiment, the minimum permissible time between synchronous resets is set to 10 minutes. A synchronous reset occurs when the buffering queue runs out of data and cannot generate a constant rate bitstream, and the system must be reset and the buffer refilled to continue the operation. As the minimum allowed time between synchronous resets increases, the size of the buffer must increase, and so does the fixed delay. The size of the buffer prefill X region 154 is calculated based on the maximum drift between the STU 130 and the clamp 100 clocks. The buffer prefill X region 154 stores the maximum number of bits possible that two units can drift relative to each other over a 10 minute interval. The actual size of the buffer prefill X region 154 is irrelevant to the operation of the present invention. The number of bits stored in the buffer prefill X area 154 increases or decreases during system operation.

Octets of data from buffer prefill Z region 156 are sent to buffer prefill X region 154 at approximately the same bps rate as bits are sent from buffer prefill X region 154 to a constant rate bitstream. Bit transmission from buffer prefill Z region 156 to buffer prefill X region 154 is CDMA equipment clock frequency f₂Is based on. The buffer prefill Z region 156 is further prefilled with data until the data is transmitted at the constant rate bitstream output, thus introducing a fixed delay into the system. To introduce the minimum fixed delay possible, the size of the buffer prefill Z region 156 is equal to a small value. One convenient value is the average number of bits transmitted to the buffer prefill X region 154 before another dataset arrives from the TRLP layer. In a preferred embodiment, an average of 96 bits are sent from each frame of queue 150. Therefore, in a preferred embodiment, the size of the buffer prefill Z region 156 is 12 octets.

In one alternative embodiment, the size of the buffer prefill Z region 156 may be reduced to zero. If the connection is first set up between clamp 100 and STU130, dialing tone instructions are sent from the source unit to the receiving unit. When the receiving unit responds, the two units exchange a series of training tones. When both the transmitting unit and the receiving unit are ready to transmit data, each unit sends a carrier wave to the other unit. The first moment when data can be sent from one unit is the moment when that unit detects a carrier wave from the other unit. As soon as the modem in the IWF124 detects the carrier wave, the protocol commands the IWF124 to produce a constant rate bitstream. If no frame is yet available, a fill bit must be added to buffer prefill X region 154 and buffer prefill Z region 156 to satisfy the protocol. However, if the carrier wave is not provided to the IWF124 until the first non-erased frame arrives, the buffer Z region 158 can immediately start outputting the received data at a constant rate. Thus, before the buffer Z region 158 is empty, the next data frame or disappearance display is available and the buffer prefill Z region 156 may be deleted.

In a preferred embodiment, the nominal position of the write pointer 160 is at the boundary between the buffer prefill Z region 156 and the buffer Z region 158. As the data moves from TRLP to buffer Z region 158 in the form of actual and filled data, the data is transferred as octets from buffer prefill Z region 156. The position of the write pointer 160 moves to indicate where the next octet from TRLP should be placed. As will be described later, the write pointer 160 can be moved within the stack pointer range consisting of both the buffer prefill Z area 156 and the buffer Z area 158. Since the size of the buffer X area 154 is variable, the write pointer 160 does not enter the buffer prefill X area 154.

When the disappearance indication is received, the size of the lost frame is unknown. To protect the integrity of the total number of bits, the filler bits must be added to the queue 150 to display the lost bits. According to the present invention, the maximum number of bits that a lost data frame can contain is added to queue 150. In a preferred embodiment, the full rate frame carries 160 bits, so 160 filling bits are added to the queue 150. These bits are immediately moved through queue 150 and output to a constant rate bitstream as needed. If the lost frame is not a full-rate frame, some of the bits added to the queue 150 may still be in the queue 150 when the next non-lost frame is received.

As mentioned above, each frame has a length region. This length region indicates the number of bits of the current frame with respect to the immediately preceding frame. Using the data rates of the preferred embodiments, Table 1 shows an exemplary scheme for providing such a modulo length region. In the specific example in Table 1, the length region is only 4 bits. That is, it takes a value from 0 to 15. The value of the length region is added for each frame based on the frame rate. When a full-rate frame is transmitted, the value in that area is incremented by 8. When a 1/2 rate frame is transmitted, the value in that area is incremented by 4. When a 1/4 rate frame is transmitted, the value in that area is incremented by 2. When a 1/8 rate frame is transmitted, the value in that area is incremented by 1.<tables num="1"><img file="JP4634407B2_D0001.tif" /></tables>

According to this procedure, assuming that the first value of the length region is 0, the first column in Table 1 shows the rate of the data transmitted by the frame. The second column represents the value of the corresponding length region. Therefore, since the value of the transmitted first frame is 1/4 rate, the value of the length region is 2. Since the next frame is 1/2 rate, the value is added by 4 and takes the value of 6. The second 1/2 rate frame that follows increases the value by another 4 to a value in the length region of 10. A 1/8 rate frame increases the value in the length region to 11. Subsequent full-rate frames increase the value beyond the maximum 4-bit value of 15, so the modulo result of adding 8 is the value 3 in the length region. A 1/8 rate frame increases the value in the length area to 4, and the last full rate frame increases the value in the length area to 12.

It should be noted that when a frame disappears, so does the value in the corresponding length region. Now suppose that the sequence shown in Table 1 was transmitted, but the first full-rate frame was lost, as shown in Table 2.<tables num="2"><img file="JP4634407B2_D0002.tif" /></tables>

It should be noted that the values in the length region corresponding to the 1/8 rate frame following the disappearance are the same. It should also be noted that the number of bits in a 1/8 rate frame is known because the corresponding bits are obtained to add to the queue 150. Therefore, the rate of lost frames (and the number of filled bits to be added) is the sum of the last correctly received value before disappearance and the value added by the first correctly received frame after disappearance. Can be determined by subtracting from the first correctly received length region after disappearance. If the result is negative, the modulo value 16 is added to the result. For example, in the example shown in Table 2,: Last correctly received value before disappearance = 11; The value added by the first correctly received frame = 1; The sum of these two = 12; First correctly received length region after disappearance = 4; 4-12 = -8; and Since the result is negative, add 16 and the answer is 8.

Since the result is 8, it can be seen that the lost frame was a full-rate frame. If the frame is a 1/2 rate frame, the result is 4. If the frame is a 1/4 rate frame, the result is 2. And if the frame is a 1/8 rate frame, the result is 1. Even when the disappearance is received two or more times in a row, the same method can be used. The actual rate of each lost frame is not important. Only the total number of lost bits needs to be determined.

The 4-bit length region described above is highly constrained in use. This is because if a full-rate frame is received twice in a row, the value of the length region will not be known. It is possible to achieve similar results without departing from the scope of the present invention by using a myriad of more complex schemes. For example, in a real system, it may consist of 7 bits or more in order to correspond to the maximum number of continuous disappearances expected. 7 bits are needed to accommodate the reception of 6 consecutive full-rate frame loss. IS-95 defines an 8-bit return link processor sequence area. The value of the length region may be used instead of the return link processor sequence region.

Returning to Figure 4, when the frame disappearance indication is passed to the Adaptation Layer, a set of octet-filled characters sufficient to occupy the percentage of full-rate frames is added to queue 150. In a preferred embodiment, 20 octets of AA, which is a hexadecimal (hex) value corresponding to 160 bits of a full rate frame, are added to queue 150. (AA corresponds to the repeating sequence 10101010.) These bits can immediately start feeding to a constant rate bitstream. If another loss is received, a second set of 20 octets of AA, which is a hexadecimal value, is added to queue 150. When the first non-erased data frame is received, the number of lost bits is determined using a method similar to that described above. If the number of lost bits is less than the number of bits added to queue 150, then the next pair of bits received will be such that the extra bits are overwritten by the newly received bits. A queue stack pointer is moved to indicate where it should be placed.

For example, using the numerical examples of preferred embodiments and the examples in Table 1, it is assumed that the sequences shown in Table 3 are received.<tables num="3"><img file="JP4634407B2_D0003.tif" /></tables>

If a loss is received, 160 bits are added to queue 150. If a 1/2 rate frame is received and the value is 10, then using the formula given above, the 1/2 rate frame may have disappeared (4 = 10- (2 + 4)). It is determined. Despite the addition of 160 bits, only 80 filled bits are needed to occupy the lost bits. Therefore, the difference between the number of bits actually added and the number of bits that should be added before the received data bits corresponding to the correctly received 1/2 rate frame are added to the queue 150, The queue write pointer 160 is advanced in the direction of buffer X region 154. In this case, the number of bits that should be added is 80 bits or 10 octets. In this way, the buffer value of the extra filling bits is overwritten by the actual data received.

In other embodiments, instead of adding a set of octet-filled characters sufficient to occupy the percentage of full-rate frames to queue 150, the average number of bits received is added. For example, as described above, an average of 96 bits are transferred frame by frame, and the average number of bits destroyed when one frame is lost is 96. In such a mechanism, when one frame disappears, 12 octets of data are added to queue 150. Once the actual size of the lost frame is determined, the write pointer 160 may move towards the buffer prefill X region 154, back deeper, or towards the buffer Z region 158. If the write pointer 160 moves deeper into the queue 150, it may be necessary to add additional fill bits. In the most common embodiment of the invention, queue 150 has any number of fill bits sufficient to maintain a uniform stream of octets flowing from buffer prefill Z region 156 to buffer prefill X region 154. Can be added. In the most common embodiment according to the present invention, any number of bits can be added as long as the rate at which the bits are added is greater than or equal to the average rate at which the bits are transferred from the queue.

Although preferred embodiments have been described with reference to linear buffers, the ideas of the present invention are directly applicable to circular buffers. Circular buffers use both read and write pointers. One implementation of the circular buffer does not add the actual dummy bits when a loss is received. Instead, all you have to do is adjust the write pointer to the new position.

The idea of the present invention can also be realized as shown in FIG. In FIG. 5, the TRLP octet is input to buffer 200. The filling generator 202 supplies a constant stream of filling bits. Switch 204 selects the output of buffer 200 or the output of fill generator 202 under the control of switch control 206. Only octets that have not been received as lost are input to buffer 200. When the loss is received, the switch control 206 is notified. If the lost frame bits need to be replaced with fill bits, switch 204 connects the fill generator 202 to a constant rate bitstream. Otherwise, switch 204 connects buffer 200 to a constant rate bitstream. The number of bits to be added is determined as described above as before.

The invention can be manufactured or used by providing those skilled in the art with the above description of a preferred embodiment. Various variations of these embodiments will be apparent to those of skill in the art, and the comprehensive principles defined herein are applicable to other embodiments without the use of inventive talent. Therefore, the present invention is not intended to be limited to the embodiments presented herein, and is provided with the broadest scope consistent with the principles and novel features disclosed herein.

<figref num="1">A typical dispatch system is shown.</figref><figref num="2">Demonstrates a one-to-one secure voice connection between a remote unit and a ground line telephone.</figref><figref num="3">A protocol stack with an adaptive layer of transparent wireless link protocol is shown.</figref><figref num="4">Shows the adaptive layer queue that buffers the output of the transparent wireless link protocol to generate a constant velocity bitstream.</figref><figref num="5">It is an alternative embodiment of the linear buffer shown in FIG.</figref>