CA1273697A - Multiplexed digital packet telephone system - Google Patents

Abstract

MULTIPLEXED DIGITAL PACKET TELEPHONE SYSTEM
ABSTRACT
An efficient system for simultaneously conveying a large number of telephone conversations over a much smal-ler number of relatively low-bandwidth digital communication channels is disclosed. Each incoming telephone speech signal is filtered, periodically sampled, and digitized.
An efficient computational speech compression algorithm is applied to transform a sequence of digitized speech samples into a much shorter sequence of compression vari-ables. The compression variable sequence is further pro-cessed to construct a minimum-length bit string, and an identifying header is appended to form a packet. Only a few packets containing information on representative background noise are generated during pauses in speech, thereby conserving digital bandwidth. The packets are queued and transmitted asynchronously over the first avail-able serial digital communication channel. Numerical feed-back to the compression algorithm is employed which results in the packet size being reduced during periods of high digital channel usage. Packet header information is uti-lized to establish a "virtual circuit" between sender and receiver. At the ultimate receiving terminal, the packet header is employed to route the packet to the circuitry servicing the appropriate telephone channel. The packeting procedure is thereupon reversed. The header is stripped from the packet and the minimum length bit string is expand-ed to reproduce the original sequence of compression vari-ables. By means of an appropriate computational inverse speech compression algorithm, approximations to the original digital samples are synthesized. An intentional delay is thereupon introduced to build a backlog pool of samples to buffer against gaps in speech due to statistical fluctua-tions in packet arrival times. Digital samples from this pool are periodically outputted to a D/A converter and its analog output is appropriately filtered to approximately reproduce the original frequency-limited telephone signal.
Representative background noise is synthesized from informa-tion contained in previously received background packets during pauses in speech.

Description

~736~7 1 Technical Field The invention disclosed herein pertains to multi-channel telephonic communications. In particular, it pertains to a system which utilizes a small number of s relatively low-bandwidth digital communication channels to asynchronously convey a much larger number of simul-taneous telephone conversations in a digital packet format.
Background Art Techniques for communicating telephone conversa~
tions in digital format have become commonplace in recent years. The telephone signal is usually filtered to limit its bandwidth and is then sampled at a rate that is at least twice the frequency of its highest frequency compo-nent. The repetitive samples are applied to an A/D con-verter to obtain digital ~epresentations of the analog samples. Although volume compression and expansion may be utilized to increase the system's dynamic range, it is generally conceded that: at least eight bits are required for each digital sample in order that the quantization noise be held to an acceptable minimum. Thus, the generally accepted digital bandwidth required to digitally communicate a telephone signal that is frequency-limited to 4 kilohertz, is at least 2 times ~ times 8, or 64 kilobits per secondO
Synchronous time-division multiplexing (TDM) is generally employed to simultaneously carry a plurality of digitized telephone conversations over a single digital channel. With synchronous TDM, each digital telephone channel occupies a precise "time slot" in a high bit-rate serial data sequence. Synchronous TDM accommodates a fixed .

1 number of telephone channels and requires that a "time slot" be assigned to each channel whether or not the channel is being used. Such inflexibility can be very wasteful of digital bandwidth. To accommodate ~0 uni.directional telephone channels by conventional synchronous TDM would require a digital bandwidth of approximately 40 times 64 kilobits, or 2.56 megabits per second.
It is desirable for economic reasons to increase the digital efficiency of a telephone system so that a large number of telephone channels can be accommodated by a given digital kandwidlh. The digital ef~iciency of conventional TDM can be increased by the use of digital speech interpolation ~DSI) techniques. With DSI, a "time slot" is only assigned to a particular telephone channel during periods of actual speech and is dynamically re-assigned to another telephone chann~l during a speech pause. Since pauses are known to occupy about 60 percent of a typical speech pattern, DSI can theoretically increase digital efficiency by about: a factor of 2.5. Because of statistical variations in speech patterns, however, a factor of 1.5 is more typically o~)tained in practice. In addi-tion, DSI efficiency is further reduced by the fact that a "chan-nel assignment table" must be transmitted each TDM cycle to permit the receiver to unambiguously identify the current occupant of each "time slot".
Instead of synchronously transmitting individual digital samples of speech in TDM "time slots", longer sequences o~ samples can be assembled and transmitted asyn-chronously as packets - each packet being identified by a packet "header". As with DSI TDM, the digital efficiency ~.~73~i ~7 1 of a multiplexed packet telephone system can be increased by a factor of about 1.5 by suppressing speech pauses and transmitting only packets containing samples obtained during spurts of actual speech. Because of the asynchronous nature of packet transmission, however, such systems have required that a "time stamp" be appended to each sequence of actual speech samples so that the appropriate speech pauses could be correctly re-inserted at the receiving terminal. Such "time stamping" increases packet overhead and seriously complicates the speech reconstruction process. An example of a multiplexed speech tr~nsmission system employing "time stamped" packets of digitized speech samples has been dis-closed in Flanagan U.S. Patent No. 4,100,377.
Recent advances :in technology have provided means for new approaches to increasing the digital efficiency of multiplexed digital telephone systems. In particular, the recent development of high-pe.formance microprocessors capable of many complex calculations per second has made possible the real-time imp]ementation of powerful, but computationally intensive, speech ccmpression algorithms.
A variety of such computational algorithms are presently available and are well-known to those of ordinary skill in the art. Such algorithms typically rely upon the princi-ples of time-domain harmonic scaling, transform coding, linear or adaptive predictive coding, sub-band coding, or combinations thereof.
All such computational speech compression algo-rithms share one common feature. They can effectively transform relatively large groups of digitized speech samples into much smaller sequences, or "frames", of digital ~.

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1 compression variables at the transmitting end of a circuit for transmission efficiency, and then expand the frames at the receiving end to approximate the original larger groups of speech samples ~Jhile still maintaining acceptable speech fidelity Using techniques that are known to those of ordinary skill in the art, compression ratios - defined as the number of bits in a frame of digital compression variables divided by the number of bits in the correspondin~
group of digital speech samples - of less than .25 can presently be routinely achieved with computational algo-rithms which still maintain telephone-quality speech~
The real-time implèmentation of such computational speech compression algorithms in a multiplexed digital telephone system has, however, not heretofore been accomplished.
Summary c,f the Invention The telephone system herein disclosed is a highly efficient system for conveying a large number of simul-taneous telephone conversations (e~g., 40) over a much smaller plurality ~e.g., 4) of relatively low-bandwidth digital channels (e.g., 56 kb/sec) in digital packet for-mat. Low-bandwidth digital channels of the type employed herein have heretofore been used only for data communication since even a small number of digi-tized telephone signals have traditionally required a much wider bandwidth.
In the present invention, each incoming telephone speech signal is filtered, periodically sampled, and digi~
tized. An efficient computational speech compression algo-rithm is applied to successive groups of digitized speech samples; each group being representative of a speech inter-val ranging in duration from about 15 milliseconds to about 3j~37 1 50 milliseconds. This computational algorithm transforms each group of samples into a much smaller sequence, or "frame", of digital speech compression variables. The compression ratio of the txansformation, defined to be the ratio of the number of bits in a frame to the number of bits in the corresponding group of digital samples, is less than 0.25.
Each frame of digital compression variables is ~urther processed to construct a minimum-length bit string, and an identifying header is appended to each string to form a packet. Only a few E)ackets containing information on representative backgrouncl noise are generated during pauses in speech in order to conserve digital bandwidth.
Each packet is queued, along with similar packets derived from other similar telephone channels, and trans-mitted asynchronously over the first available one of a number of serial digital co~munication channels operating at, e.g., 56 kilobits per second. ~umerical feedback to the microprocessor implemen~ing the computational speech compression algorithm is employed which results in the packet size being dynamically reduced during periods of high digital channel usage. Packet header information is utilized to establish a "virtual circuit" between sender and receiver. The packets may be transmitted from the sender to the receiver via inte mediate terminals along the "virtual circuit". The packets are requeued at each intermediate terminal, along with the packets generated at the intermediate terminal, and are asynchronously retransmitted from the intermediate terminal over the first available one o~ a plurality of serial digital channels.

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34~t7 1 The packet header is employed to route the packet to the circuitry servicing the appropriate telephone chan-nel at the ultimate receiving terminal. The packeting procedure is thereupon reversed. The header is stripped from the packet and the minimum-length bit string lS expand-ed to recover the frame of digital speech compression var-iables. Approximations to the original digitized speech samples are synthesized from the recovered speech compres-sion variables by means of an appropriate computational inverse speech compression algorithm. An intentional delay is thereupon introduced to build a backlog pool of samples to buffer against gaps in cpeech due to statistical fluctua-tions in packet arrival times. Digital samples from this pool are periodically outputted to a D/A converter and its analog output is appropriately 'iltered to approximately reproduce the original frequency-limited telephone signal.
Representative background noise is synthesized during pauses in speech from information contained in previously received background information packets.
In the invention disclosed herein, the need for "time stamping" the speech data as in Flanagan U.S. Patent No. 4,100,377 is totally avoided. This desirable result is primarily due to the fact that the maximum speech inter-val represented by an individual packet according to the present inventlon is only about 50 milliseconds - a time that is substantialIy less than the length of a typical speech spurt. In contrast, Flanagan's much longer packets contain at least one speech spurt and may contain several.
Thus, since packets of the present invention are inherently smaller than those of Flanagan, packet delays as well as t~
1 variations therein will also be inherently smaller. Fur-thermore, since a speech spurt comprises many packets, the time interval between the reception of the packet con-taining the last frame at the end of a speech spurt and the reception of the packet containing the first frame at the beginning of the next speech spurt will correspond approximately to the actual length of the pause between the two spurts in real time. Finally, the present invention takes advantage of the facl: that ~he human ear is highly 10 insensitive to small variations in the lengths of the pauses between speech spurts. Thus, no attempt is made to accu-rately reproduce the lengths of these pauses. In fact, as will be discussed more l.ully herein below, speech pauses are treated as flexible parameters in setting up the FIFO
15 buffers which smooth out potential variations in timing of outputted speech samples which could arise from statist-ical fluctuations in speech packet arrival times.
One object of the invention herein disclosed is to provlde a digital tel.ephone system employing the 20 novel implementation of real-time digital signal processing to obtain digital efficiencies greater than those obtained in conventional telephone systems.
Another object of this invention is to provide a multiplexed digital telephone system employing asynchro~
25 nously transmitted data packets to avoid the wasted digital channel capacity associated with synchronous time division multiplexing but without the necessity for "time-stamping"
the packets.
Another object of this invention is to provide 30 a multiplexed digital packet telephone system wherein digi-3~ ~
1 tal efficiency is enhanced by generally transmitting packets only during periods of actual speech and wherein background noise is synthesized at the receiving terminal during pauses in speech by using stored information obtained from occa-sionally transmitted background information packets.
Another object of the present invention is to provide a multiplexed digital packet telephone system where-in speech gaps due to statistical fluctuations in packet arrival times are avoided by sufficiently delaying the outputting of digital speec:h samples at the receiving termi-nal to build a backlog pool of such samples to buffer aqainst such gaps.
Another object of this invention is to provide a multiplexed digital packet telephone system employing numerical feedback to dynamically adjust the computational speech compression algorithm in accordance with the total digital data load in order to accommodate a large number of telephone channels but still reduce the likelihood of packet discards during temporary periods of high data load-ing of the serial digital channels.
Still another object of the present invention is to provide a high-capacity digital multiplexed telephone system which utilizes a plurality of relatively low-band-width serial communication channels of the type that are normally used only for data communication rather than a single hicJher-bandwidth channel in order to derive economic advantages therefrom.
These, and other, objects of the present invention will become apparent from a reading of the detailed descrip-tion herein below together with the appended claims~

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1 Brief Descri~tion of the_Drawings . .
Fi~. 1 is a block diagram of a simple multiplexed digital telephone system comprising three digital telephone terminals located at three different cities in accordance with the present invention.
Fig. 2 is a block diagram of a single digital telephone terminal in accordance with the present invention showing its common bus structure and the interconnection of the terminal's constituent printed circuit boards.
Fig. 3 is a bloc~; diagram of any one of the Speech Processor Boards, SPB-l through SPB-3, identified in Fig.

2.
Fig. 4 is a bloc~; diagram of either of the Packet Multiplexer Boards, PMB-l or PMB-2, identified in Fig.
2.
Fig. 5 is a schematic representation of two packet formats employed in communicating over the digital communi-cation channels in accordance with the present invention.
Detailed Description of the Drawings .
Referring now to Fig. 1, a simple multiplexed digital packet telephone systern in accordance with the present invention is represented in block diagram form.
The system comprises three digi~al telephone terminals, represented generally as 8a, 8b, and 8c, located in three different cities denoted as City A, City B, and City C.
In each city, a plurality of bi-directional telephone lines 10a, 10b, or 10c coming from a Telephone Company Central Office or PBX switchboard interfaces ~ith a Speech Processor (SP) section 12a, 12b, or 12c of the appropriate correspond-ing digital telephone terminal 8a, 8b; or 8c. In addition

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1 to a Speech Processor section, each terminal further com-prises a Packet Multiplexer (PM section 14a, 14b, or 14c and a Vtility Logic (UL) section 16a, 16b, or 16c. Packet Multiplexer sections 14a, 14b, and 14c interface with a small plurality of serial digital channels 18, 20, 22, and 24 interconnecting the cities and carrying digital information in opposite directions at an individual channel rate of, e.g., 56 kilobits per second.
Still referring to ~ig. 1, the implementation of a telephone call from a user in City A to another user in City C may be described l~y the following sequence o~
events:
Circuitry within Speech Processor section 12a of terminal 8a detects "off-hook" status and decodes the dial code inputted on one of the plurality of telephone lines lOa. This dial code, which may be in either pulse-dial (i.e., rotary~dial phone) or DTMF (Dual Tone Multi Frequency - i.e., "Touchtone") format, indicates that the call's recipient is in City C. In response, Speech Processor section 12a constructs appropriate signalling packets which are communicat:ed to Packet Multiplexer section 14a via internal communication path 26a. Packet Multiplexer section 14a, in turn, interchanges signalling packets with Packet Multiplexer section 14b in City B via first available 2~ ones of serial digital channel pluralities 18 and 20 and Packet Multiplexer section 14b interchanges signalling packets with Packet Multiplexer section 14c in City C via first available ones of serial digital channel pluxalities 22 and 24.
Using a standard Link Access Protocol (LAP B) to ensure accuracy, the terminals 8a, 8b, 8c intercommuni-~ 10 -~73~

1 cate and agree to establish a "virtual circuit" between the appropriate one of the plurality of telephone lines lOa in City A and another one of the plurality of telephone lines lOc interfacing terminal 8c at City C. Thereafter, for the duration of the call, all signalling and speech information communicated between these two telephone lines is digitized and coded into packets which are identified by particular address codes that are unique to this "virtual circuit".
Any such packet qoing from City A to City C is transferred via path 26a flom Speech Processor section 12a to Packet Multiplexer section l~a where it is queued along with other similar packets and transmitted asynchro-nously to City B over the i~irst available one of a plurality of serial digital channels 18. At City B, the packet is transferred over internal path 28, requeued, and retransmit-ted asynchronously to City C over the first available one of a plurality of serial digital channels 22. Upon arrival at City C, the packet is passed from Packet Multiplexer section 14c to Speech Processor section lOc via internal path 26c. The original speech or signalling information is then reconstructed from the digital information contained in the packets and the reccnstructed speech is appropriately transferred to the predetermined one of the plurality of telephone lines lOc established by the "virtual circuit".
Speech or signalling information traveling in the reverse direction takes the opposite path. The speech or signalling information is appropriately digitized and coded into packets identified by a unique address code at Speech Processor section 12c at City C. Each such packet 3~?7 l is then asynchronously transferred to the corresponding Speech Processor section 12a at City A via internal path ~6c, the first available one of a plurality o~ serial digi-tal channels 24, internal path 28, the first available one of a plurality of serial digital channels 20, and inter-nal path 26a. The speech or signalling information is thereupon reconstructed and appropriately transferred to the agreed upon one of the plurality of telephone lines lOa established by the "virtual circuit".
Those skilled in the art will appreciate, from the foregoin~ explanation, that the Link Access Protocol could just as well have been used to ensure accuracy of messages establishing a "virtual circuit" between a user in City A and a user in City B or between a user in City B and a user in City C. Such two~erminal "virtual cir-cuits" would, of course, utilize internal communication path 26b rather than inter~al communication path 28 within digital telephone terminal 8b at City B. Those skilled in the art will also appreciate that the sequence of events described above can be extended and applied to larger sys-tems comprising more than three digi-al telephone terminals.
Fig. 2 is a block diagram of a single digital telephone terminal such as one identified as either 8~, 8b, or 8c in k'ig. l. Fig. 2 discloses a Speech Processor section 12, a Packet Multiplexer section 14, and a Utility Logic section 16 all interconnected by a Common Bus 40.
Common Bus 40 comprises any standard multi-processor bus structure utili~ing the "Master-Slave" concept and may, for example, comprise the Multibus (TM~ structure developed by the Intel Corporation of 3065 Bowers Avenue, Santa Clara, California.

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1 Speech Processor section 12 ln Fig. 2 comprises a plurality of Speech Pr~cessor Boards, SPB-l, SPB-2, etc., identified individually with the numeral 30. Each Speech Processor Board 30 interfaces ~Jith at most two of the total plurality of telephone lines 10 which interface with Speech Processor section 12. Thus, e.g., at least twenty Speech Processor Boards 30 would be required to interface with forty telephone lines 10. In additional to interfacing with two telephone lines, each Speech Processor Board 30 interfaces as a "Slave" to Common Bus 40. Because of its "Slave" status, a Speech Processor Board 30 cannot ini-tiate a Common Bus transaction but can only respond to transac-tions initiated by a Commor, Bus "Master".
As also seen in E'ig. 2, Packet Multiplexer section 14 comprises a plurality of Packet Multiplexer Boards, PMB-l, PMB-2, etc., identified indi~idually as 32. Each Packet Multiplexer Board 32 interfac2s with a group of from one to four outgoing serial digital channels 34 and with a group of from one to four incoming serial digital channels 36 as well as to Common Bus 40. The unidirectional serial digital channels comprising groups 34, 36 may, for example, operate at an individual data rate of 56 kilobits per second. Such relatively low-bandwidth channels are generally utilized for data communication only. (The groups of channels identified as 34, 36 in Fig. 2 correspond to the channels identified in Fig. 1 as 18, 20, 22, 24.) For a simple bidirectional system employing only two digital telephone terminals, one Packet Multiple~er Board 32 inter-facing four outgoing and four incoming 56 kilobit/sec digi-tal channels would normally be employed in each terminal , ~ 3~

1 for each twenty Speech Processor Boards 30 ser~icing forty bidirectional telephone lines.
~ Packet Multiplexer Board 32 serves as a bus "Master" and initiates all Common Bus transactions on Common Bus 40. Although only one Packet Multiplexer Board 32 can serve as "Master" at any given time, all such boards 32 have "Master" capability and can assume control of Common Bus 40 by means of a standard prioritized bus exchange protocol. In a terminal comprising a plurality of Packet Multiplexer Boards 32, control of Common Bus 40 is time-shared among the members of that pluralityO It will be appreciated that the internal data paths denoted 26a, 26b, and 26c in Fig. 1 actually represent transactions between a Speech Processor Board 30 and a Packet Multiplexer Board 32 via common bus 40, and that internal data path 28 actually represents a transaction between two Packet Multiplexer Boards 32 via l~ommon ~us 40.
Fig. 2 further discloses a Utility Logic section 16 comprising a single Utility Logic Board 38. Utiiity Logic Board 38 interfaces ~lS a "Slave" to Common Bus 40 and can therefore not initiate Common Bus transactions.
It comprises conventional hardware to provide the Speech Processor Boards 30 and the Packet Multiplexer Boards 32 with services not found elsewhere in the system. Such services include converting 15 volts dc to 12 volts dc for use by the Packet Multiplexer Boards 32; generating a 40 megahertz clock signal for the Speech Processor Boards 30; providing a time-of-day clock; providing a "watchdog"
timer to reset the system and sound an external alarm in the event of a software malfunction; and providing "broad-3~
1 cast" logic to permit a Packet Multiplexer Board 32 to address all of the plurality of Speech Processor Boards 30 by means of a single command. Since such conventional hardware is well-known to one of ordinary skill in the art, Utility Logic Board 38 will not be discussed further herein.
Referring now to Fig. 3, a Speech Processor Board 30 is depicted in block diagram format. Each SPB 30 inter-faces with two bidirectional telephone lines lOa and lOb by means of Telephone InterEaces 42a and 42b. Telephone Interfaces 42a and 42b comp:rise impedance matching and filtering circuitry as well as amplifiers to modify the incoming and outgoing audio levels and circuitry to detect DTMF (i.e., "Touch~one") di(~its.
Data manipulation within SPB 30 is under the control of three microprocessor denoted MPU 44, DSP-l 46a, and DSP-2 46b. MPU 44 is a general purpose microprocessing unit which controls all of the input/output operations of SPB 30. It may for example be a type M68000 microproces-sor manufactured by Motorola IncorpGrated of 3501 Ed ~lue-stein Boulevard, Austin, Texas. DSP-l 46a and DSP-2 46b are special purpose numerical microprocessors utilized for digital signal processing tasks and must b~ capable of performing complex numerical calculations with high throughput. They may, for example, be type TMS320 micropro-cessors manufactured by Texas Instruments Incorporated, of P.O. Box 5012, Dallas, Texas. Each such processor serves one telephone channel and processes information flowing in both directions.
Each speech processor board SPB 30 includes random access memory that is divided into six distinct memory 3~i~3 7 1 areasO Common RAM 48 interfaces both MPU 44 and Common Bus 40. This memory area is addressable for both read and write operations by both MPU 44 and by a Packet Multi-plexer Board 32 serving as bus "Master". Common RAM 48 thus serves as an exclusive "gateway" for data interchanges between SPB 30 and a Master PMB 32. It also serve% as a storage area for the program of MPU 44. A second memory area 50, a portion of RAM 48, is addressable for read and write operations by MPU 44. This memory area is used as a working area by MPU 44 during execution of its program.
Third and fourth memory are~s, denoted Voice RAM-l 52a and Voice RAM-2 52b are addressable for both read and write operations by MPU 44 and by DSP-l 46a or DSP-2 46b, respec-tively. These two memory areas are used by DSP-l 46a and DSP-2 46b as working areas during processing of both out-goiny and incoming packets Eor the ~wo voice channels.
Finally, memory areas 54a and 54b are addressable for both read and write operations by DSP-l 46a and by DSP-2 46b, respectlvely, but by MPU 44 for write operations only.
These two areas serve as storage areas for the programs of DSP-l 46a and DSP-2 46b, respectivelyO During system initialization, programs for MPU 44, DSP-l 46a and DSP-2 46b are copied from PROM memory in a Master PMB 32 to Common RAM 48 in SPB 30 by means oi a "broadcast" transaction on Common Bus 40. The programs for DSP-l and DSP-2 are thereupor transferred to the appropriate program storage areas, 54a, and 54b, respectively, by action of MPU 44.
MPU 44 recognizes six hardware interrupts. A
Master PMB 32 can interrupt MPU 44 via Common Bus 40 and Master Interrupt line 56. The digital signal processors 3~
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1 DSP-l 46a and DSP-2 46b can each interrupt MPU 44 via inter-rupt lines 58a and 58b, respectively. In addition, a set of three timers 60 interrupts MPU 44 via a set of three interrupt lines 62 to provide timing references. One of these timing interrupts occurs periodically to identify a sampling interv~l of 167 microseconds. The other two timers provide interrupts to MPU 44 at one and three milli-second intervals, respectively, and are used in the imple-mentation of less frequent]y occurring signalling tasks.
MPU 44 transfers data to and from Telephone Inter-faces 4~a and 42b by means of MPU I/O Ports 64. Telephone signalling information is transferred directly to and from Telephone Interfaces 42a ar,d 42b via signalling lines 66a and 66b, respectively. Fi~tered voice information coming from Telephone Interfaces 42a and 42b is digitized by A/D
converter 68 and the resulting digital samples are trans-ferred to MPU I/O Ports 64. Digital samples of voice infor-mation going to Telephone Interfaces 42a and 42b are trans-ferred from MPU I/O Ports 64 to D/A converter 70 and the resultant analog signals are then communlcated to Telephone Interfaces 42a and 42b for filtering and level shifting.
In operation, "virtual circuits" are first estab-lished between a Speech Processing Board 30 and one or two other Speech Processing Boards 30 at remote locations in the system. Thereafter, the first SPB 30 accepts voice signals on telephone lines 10a and 10b, converts them to packets of digital information, and places the packets in Common RAM 48 for transmission by a Master Packet Multi~
plexer Board 32. At the other ends of the "virtual cir-~cuits", identical Speech Processing Boards 30 are receiving .

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1 these packets of digital information in their Common RAMs -48, converting the numbers back into analog signals, and applying these analog signals to the appropria-te ones of their interfaced telephone lines lOa or lOb. The first SPB 30 is said to be analyzing speech while the second two SPBs 30 are said to be synthesizing speech. Simulta-neously with these operations, the second SPBs 30 are also analyzing speech signals received on their own telephone lines while the first SPB 30 is synthesizing the results.
Thus, each SPB 30 is capabl~ of both analyzing and synthez-ing speech for two telephone "virtual circuits" simulta-neously.
In addition to the tasks of speech analysis and synthesis, each SPB 30 is aLso capable of performing signal-ling functions. For each oE its two incominc~ telephone lines lOa and lOb, the SPB 30 identifies on-hooks, off-hooks, dial pulses, valLd DTMF digits, and WINKs.
It encodes these events into specia] signalling packets and places these packets in Common ~AM 48 for transmission by a PMB 32 to the SPBs 30 at the other ends of the "virtual circuits." The first SPB 3(~ also receives signalling pack-ets in its Common RAM 48 that were encoded by the remote SPBs 30. It thereupon decocles each such packet's contents and performs the signalling function identified therein on the appropriate telephone line, lOa or lOb.
Still referring to Figure 3, the tasks of input-ting data ~or speech analysis and outputting data resulting from speech synthesis are periodically initiated every 167 microseconds by virtue of MPU 44 receiving an interrupt from a hardware Timer 60 via an interrupt line 62. MPU 44 3~

1 thereupon collects a digital sample prepared by the A/D
circuitry G8 of the voice signal coming from each Telephone Interface, 42a, and 42b, and transfers same via MPU I/O
Ports 64 to Voice RAM-l 52a and Voice RAM-2 52b, respec-tively, for analysis. During this same routine, ~PU 44 transfers a diyital sample of synthesized speech from each of the Voice RAMs, 52a and 52b, to D/A circuitry 70 via MPU I/O Ports 64 for conversion to analog signals to be communicated to Telephone Interfaces 42a and 42b, respec-tively.
In between the times that MPU 44 is accessing Voice RAMs 52a and 52b, Voice RAMS 52a and 52b are each employed as working areas by the two Digital Signal Proces-sors, DSP-l 46a and DSP-2 46b, respectively, during imple-mentation of speech analysis and synthesis algorithms stored in their respective program memories 54a and 54b.
During implemental:ion of speech analysis, DSPs 46a and 46b read the digital samples stored by MPU 44.
The DSPs periodically estimate the background noise power levels from the magnitudes of the stored digital samples.
Samples representing input power levels sufficiently larger than background are assumed to describe speech and are processed accordingly. For such samples, the DSPs perform speech compression computations on successive groups and store the resultant compression variables back in the same Voice RAM, 52a or S2b, respectively. When a DSP has proces-sed an appropriate number of samples, which may range from about 60 to about 300, it either sets a flag in its respec-tive Voice RAM or it interrupts MPU 44 via interrupt line 58a or 58b to inform MPU 44 that a complete frame of speech compression variahles has been prepared. In .

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1 response, MPU 44 packs the frame into a minimum-length bit string ~nd writes this string into Common RAM 48.
MPU 44 then appends a packet header to the data string and sets a flag in Common RAM 48 to indicate to the Master PMB 32 that a packet is ready for collection.
Telephone speech patterns are characterized by finite spurts of speech separated by discrete pauses.
When the power level represented by the inputted samples has approached the estimated background level sufficiently closely, the frame preparation sequence is terminated.
However, one last frame identified as compressed background noise is prepared by the DSP. Like the preceding speech frames, this background rame is packed into a mini-mum-length bit string and written, along with a packet header, in Common RAM for collec~ion by Master PMB 32.
Thus, each spurt of packetized speech will end with a back-ground noise packet. During pauses in speech, speech pack-ets are not normally prepared. However, special background noise packets containing updates of compressed background noise information are peri~dically generated at a very slow rate such as, for example, once every two seconds.
Although MPU 44 is not directly involved in the calculation of speech compression variables, certain proces-sing options can be controlled by MPU 44 through its a~ility to write into DSP program memories 54a and 54b. This fea-ture is utilized in the present system to provide numerical feedback from Master PMB 32 to the various SPBs 30 in the terminalO A Master PMB 32 periodically broadcasts a number to the Common RAMs 48 of the various SPBs 3~. This number 1 indicates the current size of a moving average of the lengths of queues of packets awaiting transmission on the serial digital channels. At each SPB 30, the MPU 44 period-ically reads this number and modifies the DS~ programs stored in RAM areas 54a and 54b accordingly. Smaller num-bers cause the DSPs 46a and 46b to produce larger frames which more accurately describe the voice signal. Larger numbers have the opposite effect and cause the DSPs to produce smaller frames thereby reducing the length of the transmission queues. ~his !lumerical feedback technique dynamically adjusts the computational speech compression algorithm in accordance with the current total digital data load. It permits servicing a large number of telephone lines while still greatly reducing the likelihood of having to discard packets during periods of abnormally high data loading.
The speech synthesis procedure begins with a Master PMB 32 writing a speech packet into an input queue area in Comrnon RA~ 48 and setting a flag to indicate its arrival to MPU 44. There are two such input queues, one serving each telephone channel, and the packet is placed in the appropriate queue. MPU 48 periodically checks flags in Voice RAMs 52a and 52b to see whether either DSP desires more input data for speech synthesis. If data is desired and a packet is available in Common RAM 44, MPU 44 expands the packet data into a frame of speech compression var-iables, writes this frame into an input buffer in the appro-priate Voice RAM 52a or 52b, and sets a flag in the Voice RAM to inform the appropriate DSP 46a or 46b of the data frame's availability.

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1 DSPs 46a and 46b continually process any frames of speech co~pression variables found to be available in their input buffers. By means of an appropriate computa-tional inverse speech compression algorithm, the~ synthesize approximations to the original digitized samples of speech and write these approximations into output FIFO buffers in their respective Voice RAMs. The approximated digital samples are periodically removed from these output FlFOs and transferred to the app~opriate D/A circuitry 70 by MPU ~4 during the same Timer interrupt routine that is used to bring digitized sa~ples of speech from A/D cir-cuitry 68 into Voice RAM for analysis. Thus, one approxi-mal;ed digital sample of synthesized speech is removed from the output FIFO buffer of each Voice RAM, 52a and 52b, every 167 microseconds.
During speech pauses, no data will be available for processing in a DSP's input buffer in Voice RAM. During these periods, a DSP continually fills its output FIFO buf-fer with approximated digit~l samples of representative background noise synthesize~ from data received earlier in periodically updated background noise information pack-ets.
When speech compression variables first appear in the input buffer indicating the beginning of a spurt of speech, the DSP synthesizes the appropriate digital samples but does not place them at the "head of the line"
in the output FIFO buffer. Instead, an intentional delay is in-troduced by placing the beginning sample a fixed number of samples back from the sample that is next in line to be outputted. This number may, for example, be 300 samples, 3~
1 which corresponds to a time delay of 50 milliseconds.
Thus, even after the beginning samples of the speech spurt have been synthesized, MPU 44 will continue to output back-grcund noise samples to D/A circuitry 70 for a fixed length OL time equal to this Voice RAM output FIFO buffer delay.
As a speech spurt progresses, this buffer delay will random-ly shrink and grow as a result of statistical variations in packet arrival times. Unless a packet is lost or dis-carded, however, the nominal buffer delay will remain con-stant. This nominal buffer delay is chosen to be large enough to buffer against the occurrence of gaps in speech due to packet delays under worst case data loading condi-tions. At the transmitting PMB 32, a new packet will be discarded rather than queued if it cannot be transmitted within the receive buffer's delay period.
The end of the speech spurt is recognized by DSP 46a or 46b by the background noise information packet that terminates the sequence of speech packets. Upon re-ceiving a background noise ~rame in ;ts input buffer in Voice RAM, the DSP follows the synthesized speech samples in its output buffer with synthesized background noise samples and then continues ~o fill its output buffer with synthesized background nois~ samples for the duration of the pause.
The DSP will recognize that one or more packets have been lost or discarded if there is a gap in speech data available to the DSP that is not preceded by a back-ground noise information packet. In this case, the DSP
re-outputs samples synthesized from the last speech informa-tion frame received. If another packet becomes available ?~

1 before the repeated frame has been converted to analog, the speech synthesis resumes normally. Otherwise, the DSP fills its output buffer with as many background noise samples as are requ red to recover the desired nominal buffer delay. Although this procedure will produce a short "glitch" in the synthesized speech, a terminal's ratio of telephone line connections to total digital channel capacity is so chosen that at full data loading, the occur-ence of "glitches" due to discarded packets falls wi-thin industry standards for acceptable speech clipping.
Fig. 4 depicts a block diagram of the principal data and control paths of either of the Packet Multiplexer Boards (PMBs) 32. All operations on a particular PMB 32 are under the control of Microprocessor Unit (MPU) 72.
MPU 72 is a general purpose microp.ocessor and may, for example, be a type 80186 microprocessor manufactured by the Intel Corporation of 30(j5 Bowers Avenue, Santa Clara, California.
MPU 72 interfaces directly to Local Bus 74 which, in turn, interfaces to Common Bus 40 through Bus Interface Logic 76. Bus Interface Logic 76 permits MPU 72 to ser~e as a Common Bus Master and initiate read and write transac-tions to the Common RAMs 48 located on the various Speech Processor Boards 30. It also permits an MPU 72 to execute a Bus Exchange Protocol with another MPU 72 located on a different Packet Multiplexer Board 32 in order to acquire or relinquish control of Common Bus 40. In addition to Bus Interface Logic 76, Common Memory 78 also interfaces both Local Bus 74 and Common Bus 40. Accordingly, Common Memory 78 can be accessed for read and write operations by both the resident MPU 72 on the same board as Common Memory 78 and by a remote MPU 72 located on a different 3~
. .
1 Packet Multiplexer Board 32 and having control of Common Bus 40.
Programmable Read Only Memory (PROM) 80 and RAM
82 are addressable only by the MPU 72 resident on the same board. PROM 80 contains the program of MPU 72, and RAM
82 is used for temporary storage of flags and da-ta during execution of that program. In addition, at least one PMB
32 in the terminal will have the programs of MPU 44 and DSP 46a and 46b of the various SPBs 30 stored in its PROM
80. As described earlier, these programs are transferred to the various SPBs 30 via Common Bus 40 during initiali~a-tion of the terminal. Operator monitoring and control of PMB 32 is provided through CRT Interface 84 which inter-faces with Local Bus 74.
MPU 72 controls data flow- to the outgoing serial digital channels 34 as well as datcl flow rrom the incoming serial digital channels 36 This ccntrol is exercised by the programming of Serial Communications Controller 86, Outgoing DMA (Direct Memory Access) Controller 88, and Incoming DMA Controller 90. The performance of the DMA programming tasks takec place during interrupts that are coordinated by InterruFt Controller 92 along with Pro-grammable Timer 94. Serial data flowing from Serial Commun-ications Controller 86 to the outgoing serial digital chan-nels 34 passes through Serial Transmitter (TX) Interface96. Serial data flowing from the incoming serial digital channels 36 to Serial Communications Controller 86 passes through Serial Receiver (RX) Interface 98.
Still referring to Fig. 4, events leading to the transmission of speech packets over a serial digital .

1 communication channel 34 may be described by the following sequence:
MPU 72 routinely polls the output buffer flags in the Common RAMs 48 of the various SPBs 30 via Common Bus 40. When it finds a packet that is ready for collec-tion, it copies the packet into Common Memory 78 and resets flags in Common RAM 48 to free the buffer area for re-use by the MPU 44.
The packet, now residing in Common Memory 78, is logically attached to the shortest one of the four queues serving the four outgoing serial communication channels.
The use of separate queues facilitates channel operations.
As those skilled in the art will appreciate however, a single queue serving all four channels could also be employed. If all queues have reache~ a maximum size, the packet is simply discarded since it would not reach the receiver within the time delay period allowed by the receiver's output buffer. If this is not the case, however, the packet is queued for transmission.
The actual transmission of the queued packets is performed as the result of a byte~by-byte direct memory access transfer from Common Memory 78 to Serial Communica-tions Controller 86 under the control of Outgoiny DMA Con-troller 88. DMA Controller 88 serves all four outgoing channels. Each Channel is separately programmed by MPU
72 for the starting address and number of bytes of the next packet in the channel's queue. Once a DMA channel is programmed and unmasked, the DMA controller takes over and transfers consecutive bytes from Common Memory 78 to Serial Communication Controller 86 via local bus 74. The ~73~
. .

1 consecutive bytes are converted to serial format by Serial Communications Controller 86 and transmitted over the appro-priate channel by means of Serial TX Interface 96. When the pre-programmed number of bytes has been transferred, Serial Communications Controller 86 appends a special flag byte, OllllllO, to the serial data stream to denote the packet's end. In addition, MPU 72 receives an End of Pro-cess (EOP) interrupt from D~IA Controller 88 via Interrupt Controller 92. During the resulting interrupt-level rou-tine, MPU 72 consults tables in RAM 82 to identify the starting address of the next: packet in the channel's trans-mission queue and then reprograms DMA Controller 88 accord-ingly. The transmission process will continue in this m~nner as long as packets are available in the channel's transmission queue. When pcckets are no longer available, the channel reverts to "idle" status~ An "idle" channel continuously transmits flag bytes, ~illlllO, generated by Communications Controller 86.
While MPU 72 services the transmission queues, it periodically broadcasts a number to the Common RAMs 48 on the Speech Processor ~oards 30 indicating the current size of a moving average of the transmission queue lengths.
As described previously, MPUs 44 utilize this information to providé numerical feedback which dynamically adjusts the courseness of the computational speech compression algorithm in accordance with the current total data load.
Speech packets are transmitted over the serial digital channels without check bytes since the delay caused by retransmitting a packet containing an error would be prohibitively long for communicating speech. Data errors ~ ~ ~3 ~ ~7 .. .
1 due to transmission problems may therefore eause oeeasional impairments of the received speech. Such occasional impair-ments are of minor impor-tanee and will not ordinarily be noticed, however. As described more fully below, the system mixes speech p~ckets with data paekets containing control and signalling information. These data packets include a CRC code for error checking and are retransmitted accord-ing to a standard data transmission protocol (LAP-B) if found to be in error.

Continuing to refer to Fig. 4, the sequenee of events related to the reeeption of speeeh packets on a serial digit~l eommunieation ehannel 36 will now be de-tailed:
Paekets arriving on one of the four serial ehan-nels 36 pass through the Serial Reeeiver Interface 98 to the Serial Communieations Controller 86. The serial data ! is thereupon eonverted to parallel format and transferred byte-by-byte to an input buffer area of common memory 78.
This transfer is a direet memory access transfer under the control of Incoming DMA Controller 90.
When Serial Communications Controller 86 receives the speeial flag byte which had been inserted to denote the end of a paeket, an interrupt signal is generated and communicated to MPU 72 via Interrupt Controller 92. As a result of this action, the program of MPU 72 transfers eontrol to an interrupt-level routine.
Upon entering the interrupt-level rou~ine, MPU

72 consults tables in RAM 82 to determine the starting address of the next available buffer in Common Memory 78.

It then programs the appropriate channel of Incoming DMA

1 Controller 90 accordingly to prepare to transfer the next incoming packet into this new buffer area.
Following the programming of Incoming DMA Control-ler 90, MPU 72 examines the header of the newly arrived packet in Common Memory 78. If this header identifies the packet as a speech packet, MPU 72 transfers the packet via Common Bus 40 to the appropriate telephone channel buffer in Common RAM 48 on the appropriate Speech Processor Board 30 as identified by the address in the packet header.
A flag is then set in Common RAM 48 to notify MPU 44 of the packet's arrival. MPU 44 routinely inspects the buffer flags in Common RAM 48 looking for fresh packet arrivals.
In this fashion, speech packets will continue to be transferred to the appropriate SPBs 30 as long as they continue to arrive on serial digital communications channels 36.
~eferring now to Fig. 5, the packet format as it is communicated serially over a digital communication channel is disclosed. Each packet begins with a Packet Header and ends with a terminating Flag Byte (01111110).
As disclosed above, this F]ag Byte is automatically appended to the end of the packet by Serial Communications Controller 86.
Two packet formats are disclosed in Fig. 5; a format appropriate to transmission of a frame of digital compression variables, and a format appropriate to the transmission of all other types of digital data. The former is referred to as a Speech Packet and the latter is referred to as a Data Packet. The Packet l~eader of both packet formats begins with an eight-bit Type Byte, and this Type .

73~
1 Byte identifies whether the ensuing packet is a Speech Packet or a Data Packet. For both packet formats, the Type Byte conveys only four bits of digital information, and the other four bits are utilized as checkbits to ensure accuracy of the four information bits.
If the four information bits of the Type Byte represent a binary number between 0 and 3 - that is, if the higher order two of these four bits are zeros - the packet is identified as a Speech Packet. In this case, the Packet Header is two bytes, or sixteen bits long.
A Speech Packet Header contains a ten-bit address field consisting of an eight~bit Address Byte along wi-th the two lower order bits of the Type Byte information field.
Thus, there are 1024 unique digital codes available for establishing virtual circuits bet~-een speech sources and corresponding speech sinks.
If the four infor~ation bits of the Type Byte represent a binary number b_tween ~ and 15, the pac]cet is identified as a Data Packet, and the type of data being conveyed is identified by the value of this binary number.
Thus, provision has been made for communciating up to twelve different types of data. At the present time, three differ-ent types of data have been differentiated:
-Telephone Signalling or Line Control ~ata.
Data Packets of this type convey signalling information such as on-hooks, off-hooks, and dialing codes from a Speech Processing Board 30 at one location to a corresponding Speech Processing Board 30 at a second location for appropriate interfacing to a telephone line.

~ ~3~
, 1 -Multiplexer Control ~ata. Data Packets of this type convey messages between a Packet Multiplexer Board 32 at one location and a Packet Multiplexer Board 32 at a second location. They are used for a variety of control purposes includin~
the establishment of virtual circuits linking sources t:o corresponding sinks.
-User Data. Data Packets of this type convey data provided in digital format by a system user data source for deliver~
in digital format to a system user data sink.
Continuing to refer to Fig. 5, a Data packet Header is seen to be four by-tes long~ In addition to the Type Byte, the Packet Header contains a single Packet Con-trol Byte (PCB) and a two-byte Address field. The two-byte Address field identifies the destination of individual Data Packets by any one of up to 65~35 unique digital codes. The Packet Control Eyte contains information for data link level control and conforms with the Link Access Protocol (LAP-B) defined in the X.25 standard~ As those skilled in the art will appreciate, the purpose of this byte is to provide means for retransmission of a Data Packet if it is found to be in error. Immediately following the data in the packet is a two-byte Cyclic Redundancy Check (CRC~ code. These bytes provide means for checking a received Data Packet to determine whether the data has been corrupted in transmission. Thus, the Data Packet Format provides both means for detecting data transmission .

.
.

~73~j~3~

1 errors and means for implementing retransmission when such errors are detected.
The system as disclosed herein above inputs tele-phone speech signals in analog format and outputs telephone speech signals in analog format. As those skilled in the art will appreciate, however, the input and output formats could also be digital. In fact, certain simplifications can be accomplished when telephone signals are interfaced to a digital telephone terminal in multiplexed serial digi-tal format rather than in analog format since the operations of filtering, sampling, analog to digital conversion, and digital to analog conversion are thereby avoided. Further-more, as those skilled in the art will appreciate, disital data other than speech data can also be assembled into packets and communicated from a data source to a data sink.
Thus, the system disclosed herein is broadly capable of simultaneously communicating a mixture of analog telephone speech signals, digital teLephone speech signals, and digi-tal data signals from a plurality of appropriate sources at a first location to cor;-esponding ones of a plurality of appropriate sinks at a second location in digital packet format.

Claims (31)

The embodiments of the invention in which an exclusive property or privilege is claimed are defined as follows:

1. A method for communicating speech signals from a first location to a second location over a digital communication medium comprising the steps of:
providing a speech signal of predetermined bandwidth in analog signal format at said first location;
periodically sampling said speech signal at a predetermined sampling rate to provide a succession of analog signal samples;
representing said analog signal samples in a digital format thereby providing a succession of binary digital samples;
dividing said succession of binary digital samples into groups of binary digital samples arranged in a temporal sequence;
transforming at least two of said groups of binary digital samples into corresponding frames of digital compression variables by means of a computational speech compression algorithm whereby the compression ratio of the number of binary bits in each of said frames to the number of binary bits in the corresponding group of binary digital samples is less than one;

appending an identifying header to each of said frames of digital compression variables to provide a succession of speech information packets;
transmitting said speech information packets over said digital communication medium;
receiving said speech information packets at said second location;
removing said identifying header from received ones of said speech information packets to recover said frames of digital compression variables at said second location;
successively synthesizing groups of binary digital words at said second location from said frames of digital compression variables by means of a computational inverse speech compression algorithm, said groups of synthesized binary digital words being approximate representations of said groups of binary digital samples;
arranging said groups of synthesized binary digital words into a queue in a temporal order generally corresponding to said temporal sequence of said groups of binary digital samples;
periodically removing the synthesized binary digital word at the head of said queue at said predetermined sampling rate;
representing said periodically removed words in analog signal format thereby providing a sequence of discrete analog signal values; and filtering said sequence of discrete analog signal values to approximately reproduce said speech signal of predetermined bandwidth in analog signal format at said second location.

2. A method for communicating speech signals as in claim 1 including the steps of:
determining the average power level of said speech signal provided at said first location;
discontinuing the step of transforming groups of binary digital samples into corresponding frames of digital compression variables when said average power level becomes less than a predetermined value, thereby suspending transmission of said speech information packets pursuant to a pause in said speech signal;
substituting special binary digital words for said synthesized binary digital words at said second location when receipt of said information packets at said second location is suspended pursuant to a pause in said speech signal.

3. A method for communicating speech signals as in claim 1 wherein said digital communication medium is a serial digital communication medium, and said steps further comprise:

formatting said speech information packets in a serial digital format at said first location;
formatting said speech information packets in a parallel digital format at said second location.

4. A method as claimed in claim 1 wherein said first location includes a plurality of speech sources, said second location includes a plurality of speech sinks, each of said speech sources presenting a speech signal for transmission to a predetermined one of said speech sinks, said method for communicating speech signals further comprising the steps of:
establishing respective virtual circuits between respective ones of said speech signal sources and respective predetermined corresponding ones of said speech signal sinks;
assigning at least one circuit identifying code to each of said virtual circuits; and appending respective ones of said circuit identifying codes to said packets whereby packets originating from respective ones of said sources are routed to respective predetermined ones of said speech signal sinks.

5. A method for communicating speech signals as in claim 4 including the steps of accumulating said speech information packets in a transmission queue at said first location and dynamically adjusting said compression ratio in inverse relation to the approximate current length of said transmission queue.

6. A method for communicating speech signals as in claim 4 wherein said digital communication medium is a serial digital communication medium, and said steps further comprise:
formatting said speech information packets in a serial digital format at said first location; and formatting said speech information packets in a parallel digital format at said second location.

7. A method for communicating speech signals as in claim 4 wherein said digital communication medium comprises a plurality of digital communication channels, further including the steps of:
accumulating said speech information packets in a plurality of transmission queues at said first location, each of said queues associated with a predetermined one of said channels; and periodically identifying the shortest one of said transmission queues and accumulating said speech information packets by assigning each speech information packet, as it is made available for accumulation, to the transmission queue when identified as the shortest transmission queue.

8. A method for communicating speech signals as in claim 7 including the step of dynamically adjusting aid compression ratio in inverse relation to the approximate current average length of said plurality of transmission queues.

9. A method for communicating speech signals as in claim 8 including the steps of:
determining the individual average power levels of said speech signals presented at said speech sources;
discontinuing the step of transforming groups of binary digital samples derived from one of said speech sources into corresponding frames of digital compression variables when the average power level of the speech signal presented at said one of said speech sources is less than a predetermined value, thereby suspending transmission of speech information packets from said one speech source pursuant to a pause in speech at said speech source;
substituting special binary digital words for said synthesized binary digital words at said second location when receipt of said speech information packets at said second location is suspended pursuant to a pause in speech at said first location.

10. A method for coupling a plurality of asynchronous digital information packet sources and sinks to a plurality of digital transmitters and receivers each of said transmitters operably coupled to a corresponding one of said receivers, comprising the steps of:
detecting the presence of digital information packets at at least one of said digital information packet sources;
transferring said digital information packets from said one digital information packet source to a transmission local memory;
encoding said digital information packets received from said one packet source with routing information for routing to a predetermined corresponding one of said packet sinks;
arranging said information packets into a plurality of packet transmission queues within said transmission local memory, each queue being operably coupled with a corresponding one of said digital transmitters;
determining the respective lengths of said transmission queues;
appending successive ones of said digital information packets to the shortest one of said transmission queues;
detecting the availability of said digital transmitters for transmission of said information packets;
transferring the digital information packet at the head of one of said transmission queues from said local memory to the corresponding one of said digital transmitters when said corresponding one of said digital transmitters is available for transmission of information packets;
transmitting said digital information packets from said one of said digital transmitters to the corresponding one of said receivers;
transferring said digital information packets from said corresponding one of said digital receivers to a receiver local memory;
examining respective ones of said digital information packets in said receiver local memory; and routing said digital information packets from said receiver local memory to the respective digital information packet sinks identified by the routing information of each respective information packet.

11. A method as claimed in claim 10 wherein said step of transferring digital information packets from said local transmission memory to one of said digital transmitters and said step of routing digital information packets from said digital receivers to said local receiver memory comprise direct memory access transfers.

12. A method as claimed in claim 10 wherein said digital transmitters are serial digital transmitters and said receivers are serial digital receivers further comprising the steps of:

converting digital information from parallel digital format to serial digital format at respective ones of said plurality of serial digital transmitters; and converting digital information from serial digital format to parallel digital format at respective ones of said serial digital receivers.

13. A method as claimed in claim 10 further including the step of periodically reporting the current average length of said transmission queues to said digital information packet sources whereby the rate at which said digital information packets are presented at said digital information packet sources can be inversely adjusted in accordance with said average length of said transmission queues.

14. A method for communicating information signals from a first location to a second location over a digital communication medium comprising the steps of:
presenting an information signal as a succession of binary digital samples;
dividing said succession of binary digital samples into groups of binary digital samples arranged in a temporal sequence;
transforming at least two of said groups of binary digital samples into corresponding frames of digital compression variables by means of a computational compression algorithm whereby the compression ratio of the number of binary bits in each of said frames to the number of binary bits in the corresponding group of binary digital samples is less than one;
appending an identifying header to each of said frames of digital compression variables to provide a succession of data packets;
transmitting said data packets over said digital communication medium;
receiving said data packets at said second location;
removing said identifying header from received ones of said data packets to recover said frames of digital compression variables at said second location;
successively synthesizing groups of binary digital words at said second location from said frames of digital compression variables by means of a computational inverse compression algorithm, said groups of synthesized binary digital words being approximate representations of said groups of binary digital samples;
arranging said groups of synthesized binary digital words into a queue in a temporal order generally corresponding to said temporal sequence of said groups of binary digital samples;

periodically removing the synthesized binary digital word at the head of said queue at said predetermined sampling rate to approximately reproduce said information signal at said second location.

15. A method for communicating information signals as in claim 14 including the steps of:
determining the average power level of said information signal provided at said first location;
discontinuing the step of transforming said groups of binary digital samples into corresponding frames of digital compression variables when said average power level becomes less than a predetermined value, thereby suspending transmission of said data packets pursuant to a pause in said information signal;
substituting special binary digital words for said synthesized binary digital words at said second location when receipt of said data packets at said second location is suspended pursuant to a pause in said information signal.

16. A method for communicating information signals as in claim 14 wherein said digital communication medium is a serial digital communication medium, and said steps further comprise :
formatting said data packets in a serial digital format at said first location;

formatting said data packets in a parallel digital format at said second location.

17. A method as claimed in claim 14 wherein said first location includes a plurality of information signal sources, said second location includes a plurality of information signal sinks, each of said sources presenting an information signal for transmission to a predetermined one of said sinks, said method for communicating information signals further comprising the steps of:
establishing respective virtual circuits between respective ones of said sources and respective predetermined corresponding ones of said information signal sinks;
assigning at least one circuit identifying code to each of said virtual circuits; and appending respective ones of said circuit identifying codes to said packets whereby packets originating from respective ones of said sources are routed to respective predetermined ones of said sinks.

18. A method for communicating information signals as in claim 17 including the steps of accumulating said speech information packets in a transmission queue at said first location and dynamically adjusting said compression ratio in inverse relation to the approximate current length of said transmission queue.

19. A method for communicating information signals as in claim 18 wherein said digital communication medium is a serial digital communication medium, and said steps further comprise:
formatting said packets in a serial digital format at said first location; and formatting said packets in a parallel digital format at said second location.

20. A method for communicating information signals as in claim 17, wherein said digital communication medium comprises a plurality of digital communication channels, further including the steps of:
accumulating said data packets in a plurality of transmission queues at said first location, each of said queues associated with a predetermined one of said channels; and periodically identifying the shortest one of said transmission queues and accumulating said data packets by assigning each data packet, as it is made available for accumulation, to the transmission queue then identified as the shortest transmission queue.

21. A method for communicating information signals as in claim 20 including the step of dynamically adjusting said compression ratio in inverse relation to the approximate current average length of said plurality of transmission queues.

22. A method for communicating information signals as in claim 21 including the steps of:
determining the average power level of at least one of said information signal;
discontinuing the step of transforming groups of binary digital samples derived from said one of said information signals into corresponding frames of digital compression variables when the average power level of said one of said information signals is less than a predetermined value, thereby suspending transmission of data packets pursuant to a pause in said one of said information signals;
substituting special binary digital words for said synthesized binary digital words at said second location when receipt of said data packets at said second location is suspended pursuant to a pause in said one of said information signals.

23. A method for communicating information signals as in claim 14, said information signals comprising speech signals, including the steps of:
providing a speech signal of predetermined bandwidth in analog signal format at said first location;

periodically sampling said speech signal at a predetermined sampling rate to provide a succession of analog signal samples;
representing said analog signal samples in a digital format to provide said succession of binary digital samples;
representing said periodically removed words in analog signal format thereby providing a sequence of discrete analog signal values at said second location; and filtering said sequence of discrete analog signal values to approximately reproduce said speech signal of predetermined bandwidth in analog signal format at said second location.

24. A communication system for communicating an information signal over a communication medium comprising:
means for formatting said information signal into successive groups of digital bits;
means for transforming said groups of digital bits into corresponding frames of digital compression variables comprised of digital bits whereby the ratio of the number of bits in a corresponding frame to the number of bits in the group to which the frame corresponds to is less than one;
means for appending an identifying header to said frames of digital compression variables to provide information packets; and means for transmitting said information packets along said communication medium, said communication medium having characteristic bandwidth, said system including means responsive to the average power level of said information signal for inhibiting transmission of information packets along said communication medium when said average power level becomes less than a predetermined value, thereby conserving use of said communication medium bandwidth.

25. A communication system for communicating an information signal over a communication medium, said information signals being presented by a plurality of information signal sources and said communication medium comprising a plurality of communication channels, comprising:
means for formatting said information signal into successive groups of digital binary bits;
means for transforming said groups of digital bits into corresponding frames of digital compression variables comprised of digital bits whereby the ratio of the number of bits in a corresponding frame to the number of bits in the group to which the frame corresponds to is less than one;
means for appending an identifying header to said frames of digital compression variables to provide information packets;

means for transmitting said information packets along said communications medium;
means for accumulating said information packets in transmission queues, each of said queues being operably coupled to an assigned one of said channels;
means for identifying the shortest one of said transmission queues; and means for routing said information packets, as they are made available for accumulation, to the transmission queue then identified as the shortest.

26. A transmission system as claimed in claim 25, including means for dynamically adjusting said compression ratio in inverse relation to the approximate average length of said transmission queues.

27. A communication system for communicating an analog information signal over a communication medium comprising:
means for formatting said information signal into successive groups of digital bits;
means for transforming said groups of digital bits into corresponding frames of digital compression variables comprised of digital bits whereby the ratio of the number of bits in a corresponding frame to the number of bits in the group to which the frame corresponds to is less than one;

means for appending an identifying header to said frames of digital compression variables to provide information packets;
means for transmitting said information packets along said communication medium;
means for periodically sampling said analog information signal to provide a succession of analog signal samples;
means for transforming said analog signal samples into a digital format thereby providing a succession of digital samples.
said means for formatting said information signals into successive groups of digital bits comprising means for dividing said succession of binary digital samples into said groups of digital bits.

28. A communication system as claimed in claim 24, 25 or 27 said means for transforming said groups into corresponding frames of digital compression variables including means for creating digital compression variables by means of a computational compression algorithm.

29. A communication system for communicating an information signal over a communication medium having a communication medium bandwidth, comprising:

means for formatting said information signal into successive groups;
means for transforming said groups into corresponding frames of compression variables thereby defining a compression ratio between a corresponding frame and the group to which the frame corresponds to of less than one;
means for transmitting said frames along said communication medium;
means for dynamically adjusting said compression ratio in inverse relation to the amount of said communication bandwidth being used;
means for appending an identifying header to said frames to provide information packets, said means for transmitting said frames comprising means for transmitting said information packets along said communication medium; and means for accumulating said information packets in a transmission queue for storing said information packets when said communication medium is unavailable for transmission of said information packets, said system including means for detecting the length of said transmission queue as an indication of the bandwidth of said communication medium being used, said means for dynamically adjusting said compression ratio into an inverse relation to the amount of said communication medium bandwidth being used comprising means for adjusting said compression ratio in inverse relation to the length of said transmission queue.

30. A communication system as in claim 29, said communication medium comprised of a plurality of communication channels, said system including means for accumulating said information packets in a plurality of transmission queues, each of said queues being operably coupled to an assigned one of said channels, means for detecting the length of said transmission queues, said means for dynamically adjusting said compression ratio in inverse relation to the amount of said communication medium bandwidth being used comprising means for adjusting said compression ratio in inverse relation to the approximate average length of said transmission queues.

31. A communication system as claimed in claim 29, said communication medium comprised of a plurality of communication channels, and including means for accumulating said information packets in a plurality of transmission queues, each of said queues being operably coupled to an assigned one of said channels, means for identifying the shortest one of said transmission queues, and means for routing said information packets, as they are made available for accumulation, to the transmission queue then identified as the shortest.