CA2315745A1 - Computer-based multifunction personal communication system with caller id - Google Patents

Abstract

An interface (1650) for personal communications systems is described which provides rapid identification of a caller and limited access based on a variety of parameters obtained from caller identification information encoded by the telephone company. The incoming telephone call information is compared to a preprogrammed access matrix to determine if the caller is authorized to access the personal communications system connected to the interface. A telephone call screening method and apparatus which incorporates the encoded caller identification is also described.

Description

The present invention relates to communications systems and in particular to computer assisted digital communications including data, fax, 10 digitized voice and caller identification information.
A wide variety of communications alternatives are ctuxently available to telecommunications users. For example, facsimile transmission of printed master is available through what is commonly referred to as a stand-15 alone fax machine. Alternatively, fax-modem communication systems are currently available for personal computer users which combine the operation of a facsimile machine with the word processor of a computer to transmit documents held on computer disk, Modem communication over telephone lines in combination with a personal computer is also known in the art where 20 file transfers can be accomplished from one computer to another. Also, simultaneous voice and modem data transmitted over the same telephone line has been accomplished in several ways.
The increased accessibility provided by telephone services and modems raises problems for controlling access to corriputer systems.
25 Computer security systems have been developed which incorporate password programs to control access. These programs often monitor the number of times a particular user has logged onto a system. Systems which restrict access by limiting the number of attempted accesses in a given time period or by limiting the number of attempted accesses for a given password enable 30 unauthorized users to "tie up" the system while they attempt to gain unauthorized access. When the system is tied up, authorized users may be prohibited from accessing the system due to repeated unsuccessful attempts by unauthorized users. In addition, such systems fail to guarantee that the unauthorized user will not gain access by guessing a correct password.
35 Another personal communications system access problem is gracefully restricting access to the personal communications system depending on the date or time of day. For example, the system operator of a BBS might want to restrict modem communications with the BBS betvveen the hours of 8:00 a.m and 6:00 p.m to leave time for system maintenance duties.
Therefore, there is a need in the art for an access control system for a persona( communications system which quickly rejects unauthorized users, and, preferably denies access before the unauthorized user has an opportunity to illegally enter the system There is a firrther need for an access control system which screens callers without the use of a password system Finally, there is a need in the art for a personal communications access system which screens calls based on date and time.
S~r~of the Inventi~
The present invention solves the aforementioned problems and shortcomings of the existing art and solves other problems not listed above which will become apparent to those skilled in the art upon wading and understanding the present specification and claims.
The present disclosure describes a complex computer assisted communications system which contains multiple inventions. The subject of the present multiple inventions is a personal communications system which includes components of software and hardware operating in conjunction with a personal computer. The user interface control software operates on a personal computer, preferably within the Microsoft Windows~ environment. The software control system communicates with hardware components linked to the software through the personal computer serial communications port. The hardware components include telephone communication equipment, digital signal pr~oessors, and hardvvare to enable both fax and data communication with a hardware components at a remote site connected through a standard telephone line. The functions of the hardware components are controlled by control software operating within the hardware component and from the software components operating within the personal computer.
The major fimctions of the present system are a telephone fimction, a voice mail fimction, a fax manager function, a mufti-media mail function, a show and tell fimction, a terminal fimction and an address book fimcxion. These fimctions are described in fimher detail in U.S. Patent Application Serial Number 08/002,467 filed January 8, 1993 entitled "COMPUTER BASED MULTIFUNCTION PERSONAL
COMMUNICATIONS SYSTEM".
5 The hardware componatts of the present system include circuitry to enable digital data communication and facsimile communication over standard telephone lines.
The present disclosure also describes a system for personal communications system access control using a caller ID interface ("CID
10 interface"). Many standard telephone carriers are encoding caller )D
information which may be received before answering the telephone. One embodiment of the present invention decodes the incoming caller )D
information and compares the present caller's ide<ttification infon~nation with a preprogtarnmed access matrix to determine if access to the modem is 15 appropriate. The callers' identification information can be recorded and statistically tracked regardless of whether the callers are authorized and regardless of whether each call is answered.
In one embodiment of the present invention, the caller ID
interface incorporates a ring detector, off hook circuit, do holding circuit, 20 caller )D decoder, relay switching circuit, memory, and processor. The ring detector cit~cuit is used to enable the caller ID decoder after the first ring since most caller B7 carriers encode the caller ID inforn~ation using fi~equency shin keying t<ansmission after the first telephone ring and before the second telephone ring. 'Ihe caller ID decode is connected to the telephone line 25 (without answering the call) using the relay switching circuit between the first and second telephone ring to receive the incoming caller ID information. The off hook circuit is used to hang up on an unwanted caller before actually answering the telephone.
In one embodiment of the present invention the caller 1D
30 interface acquires information about incoming calls by decoding the incoming caller ID information and storing it in memory. Statistical tracking of callers is performed on the stored caller ID information if desired by the personal camrtmnications system owner. Another embodiment of the present invention screens access by comparing a preprogrammed access matrix to details of the call such as the caller's name, caller's phone number, the time and date the call is made, and the numbs of previous accesses by that caller in a predefined time frame. A variety of preprogtarnmed criteria are utilized to control access to the personal communications system. For example, in one embodiment, saeetming by name and telephone number is performed on an inclusive (or exclusive) basis by preprogcamining the caller ID interface with the names or telephone numbers of the callers with (or without) ass privileges. The incoming call details obtained from the caller 1D information are then compared to the inclusive (or exclusive) calls list to determine if the callers are authorized to access the personal communications system. In an alternate embodiment of the present invention the caller ID interface hangs up on an unauthorized caller, preventing the unauthorized caller even brief access.
therefore, the present invention solves the deficiencies of the prior art by providing an apparari~s and method for rapid database creation of incoming calls using caller ID information. One embodiment of the present invention also quickly rejects unauthorized callers, and may hang up on them instantly, rather than allow access to the modem. The screening process of the present invention need not use a password for caller authorization, since the caller 1D information can be used to screen out unwanted callers. Yet another embodimelmt of the present invention saeelms incoming calls based on date and time.
j~ ' tion of t~
In the drawings, where like numerals describe like components throughout the several views, Figure 1 shows the telecommunications environment within which the present may operate in several of the possible modes of communication;
Figure 2 is the main menu icon for the software components operating on the personal computer;

Figure 3 is a block diagram of the hardware components of the P
Figure 4 is a key for viewing the detailed elearical schematic diagrams of Figures SA-lOC to facilitate understanding of the interconnect 5 between the drawings;
Figures SA-SC, 6A-6C, 7A-7C, 8A-8B, 9A-9C and l0A lOC
are detailed electrical schematic diagrams of the circuitry of the hardware corr~ponents of the present system;
Figure 11 is a signal flow diagram of the speech compression 10 algorithm;
Figure 12 is a detailed function flow diagram of the speech compression algorithm;
Figure 13 is a detailed function flow diagram of the speech decompression algorithm;
15 Figure 14 is a detailed function flow diagram of the echo cancellation algorithm;
Figure 15 is a detailed fimction flow diagram of the voiceldata multiplexing function;
Figure 16 is a general block diagram showing one embodiment 20 of a caller ID interface for a personal communications system;
Figure 17A is a schematic diagram of one embodiment of a caller m interface for a personal communications system;
Figure 17B is a schematic diagram of an alternate embodim~t of a caller ll'~ interface for a personal communications system;
25 Figure 18 is a block diagram showing the multiple data message format and single data message fom~at used in standard caller 1D
encoded transmissions;
Figure 19 is a flowchart showing one embodiment of a caller ID message fom~at recognition scheme;
30 Figure 20 is a flowchart of the general operation of one embodiment of the present invention; and Figure 21 is a flowchart showing one possible implementation of a scaeening mode algorithm.
In the following detailed description, references made to the accompanying drawings which form a part hereof and in which is shown by S way of illustration specific embodicr~t in which the invention may be practiced. These embodiments are described in sufficient detail to enable those skilled in the art to practice and use the ~irnention, and it is to be understood that other embodiments may be utilized in that electrical, logical, and stnlchual changes may be made without departing firm the spirit and 10 scope of the present invention. The following detailed description is, therefore, not to be taken in a limiting sense in scope of the present invention as defined by the appended claims Figure 1 shows a typical arrangement for the use ~of the present system Personal computer 10 is pinning the software components of the 15 present system while the hardware components 20 include the data communication equipment and telephone headset. Hardware compon~ts 20 communicate over a standard telephone line 30 to one of a variety of remote sites. One of the remote sites may be equipped with the present system including hardware components 20a and software components pinning on 20 personal computer l0a In one alternative use, the local hardware components 20 may be communicating over standard telephone line 30 to facsimile machine 60. In another alten~ative use, the present system may be communicating over a standard telephone line 30 to another personal computer 80 through a remote modem 70. In another alternative use, the 25 present system may be communicating over a standard telephone line 30 to a standard telephone 90. Those skilled in the art will readily recognize the wide variety of communication interconnections possible with the present system by reading and understanding the following detailed description.
The ornamental features of the hardware components 20 of 30 Figure 1 are claimed as part of Design Patent Application Number 29/001368, filed November 12, 1992 entitled "Telephone/Modem case for a Computer-Based Multifunction Personal Communications System" assigned to the same assignee of the present inventions.
The present inventions are embodied in a commercial product by the assignee, MultiTech Systems, Inc. The soRwace component operating 5 on a personal corrrputer is sold under the commercial trademark of MultiFsPCSTM personal communications software while the hardware component of the present systerrr is sold ands the commercial name of MultiModemPCS'TM, Intelligent Personal Communications System Modem In the preferred embodiment, the software component nrns under Mrcrosoft~
10 Wrndows~ however those skilled in the art will readily recognize that the present system is easily adaptable to run under any single or multi-user, single .
or mufti-window operating system.
The present system is a multifimction communication system which includes hardware and software components. The system allows the 15 user to connect to remote locations equipped with a similar system or with modems, facsimile machines or standard telephones over a single analog telephone line. The software component of the present system includes a number of modules which are described in more detail below.
Figure 2 is an example of the Wrndows~ based main menu 20 icon of the present system operating on a personal computer. The fimctions listed with the icons used to invoke those functions are shown in the preferred embodiment. Those skilled in the art will readily recognize that a wide variety of selection techniques may be used to invoke the various firnctions of the present system. The icon of Figure 2 is part of Design Patent Application 25 Number 29/001397, filed November 12, 1992 entitled "Icons for a Computer Based Multifimction Personal Communications System" assigned to the same assignee of the present inventions.
The telephone module allows the system to operate as a conventional or sophisticated telephone system. The system converts voice 30 into a digital signal so that it can be transmitted or stored with other digital data, like computer information. The telephone function supports PBX and Centrex features such a call waiting, call forwarding, caller )D and three-way calling. This module also allows the user to mute, hold or record a conversation. The telephone module enables the handset, headset or hands-fi~ae speaker telephone operation of the hardvvare component. It includes on-screetr push button dialing, speed-dial of stored numbers and digital recording 5 of two-way c~vecsations.
The voice mail portion of the present system allows this system to operate as a telephone ansvv~eering machine by storing voice messages as digitized voice files along with a time/date voice stamp. The digitized voice files can be saved and sent to one or more destinations immediately or at a 10 later time using a queue scheduler. The user can also listen to, forward or edit the voice messages which have been received with a powerful digital voice editing component of the present system This module also creates queues for outgoing messages to be sent at preselected times and allows the users to create outgoing messages with the voice editor.
15 The fax manager portion of the present system is a queue for incoming and outgoing facsimile pages. In the preferred embodiment of the present system, this fimction is tied into the Windows "print" command once the present system has been installed This feature allows the user to create faxes from any Windows~ based document that uses the "print" command.
20 The fax manager function of the present system allows the user to view queued faxes which are to be sent or which have been received. This module creates queues for outgoing faxes to be sent at preselected times and logs incoming faxes with time/date stamps.
The multi-media mail fimction of the present system is a utility 25 which allows the user to compose documents that include text, graphics and voice messages using the message composer function of the present system, described more filly below. The multi-media mail utility of the present system allows the user to schedule messages for transmittal and queues up the messages that have been received so that can be viewed at a later time.
30 The show and tell function of the present system allows the user to establish a data over voice (DOS communications session. When the user is transmitting data to a remote location similarly equipped, the user is able to tally to the person over the telephone line while concrarecrtly transferring the data 'Ibis voice over data function is accomplished in the hardware components of the present system. It digitizr~s the voice and transmits it in a dynamically changing allocation of voice data and digital data 5 multiplexed in the same transmission. The allocation at a given moment is selected depending on the amount of voice digital information required to be transferred. Quiet voice intervals allocate greater space to the digital data transmission.
The terminal function of the present system allows the user to 10 establish a data communications session with another computer which is equipped with a modem but which is not equipped with the present system.
This feature of the present system is a Windows~ based data communications prograrrr that reduces the need for issuing "AT" commands by providing menu driven and "pop-up" window alternatives.
15 The address book function of the present system is a database that is accessible from all the other functions of the present system. This database is created by the user inputting destination add~ses and telephone numbers for data communication, voice mail, facsimile transmission, modem communication and the like. The address book function of the present system 20 may be utilized to broadcast communications to a wide variety of recipients.
Multiple linked databases have separate address books for different groups and different destinations may be created by the users. The address book fiulction includes a textual search capability which allows fast and efficient location of specific addresses as described more fully below.

Figure 3 is a block diagram of the hardware components of the present system corresponding to reference number 20 of Figure 1. These components form the link between the user, the personal computer nmning the softvvare component of the present system and the telephone line interface.
30 As will be more fully described below, the interface to the hardware components of the present system is via a serial communications port connected to the personal computer. The interface protocol is well ordered and defined such that other software systems or programs running on the personal computer may be designed and implemented which would be capable of controlling the hardware components shown in Figure 3 by using the control and communications protocol defined below.
5 In the preferred embodiment of the present system three alternate telephone interfaces are available: the telephone handset 301, a telephone headset 302, and a hands-free microphone 303 and speaker 304.
Regardless of the telephone interface, the three alternative interfaces connect to the digital telephone coder-decode (CODEC) circuit 305.
10 The digital telephone CODEC circuit 305 interfaces with the voice control digital signal processor (DSP) circuit 306 which includes a voice control DSP and CODEC. This circuit does digital to analog (D/A) conversion, analog to digital (A/D) conversion, coding/decoding, gain control and is the interface between the voice control DSP circuit 306 and the 15 telephone interface. The CODEC of the voice control circuit 306 transfers digitized voice information in a compressed format to multiplexor circuit 310 to analog telephone line interface 309.
The CODEC of the voice control circuit 306 is actually an integral component of a voice control digital signal processor integrated 20 circuit, as described more fully below. The voice control DSP of circuit controls the digital telephone CODEC circuit 305, performs voice compression and echo cancellation.
Multiplexor (MIJX) circuit 310 selects between the voice control DSP circuit 306 and the data pump DSP circuit 311 for transmission 25 of information on the telephone line through telephone line interface circuit 309.
The data pump circuit 311 also includes a digital signal processor (DSP) and a CODEC for communicating over the telephone line interface 309 through MCJX circuit 310. The data pump DSP and CODEC of 30 circuit 311 performs functions such as modulation, demodulation and echo cancellation to communicate over the telephone line interface 309 using a plurality of telecommunications standards including FAX and modem protocols.
The main controller circuit 313 controls the DSP data pump circuit 311 and the voice control DSP circuit 306 through serial input/output and clock timer control (SIO/C'TC) circuits 312 and dual port RAM circuit 5 308 respectively. The main controller circuit 313 communicates with the voice control DSP 306 through dual port RAM circuit 308. In this fashion digital voice data can be read and written simultaneously to the memory portions of circuit 308 for high speed communication be<we~ the user (through interfaces 301, 302 or 303/304) and the personal computer connected 10 to serial interface circuit 315 and the remote telephone connection connected through the telephone line attached to line interface circuit 309.
As described more fully below, the main controller circuit 313 includes, in the preferred embodiment, a microprocessor which controls the functions and operation of all of the hardvvare components shown in Figure 3.
15 The main controller is connected to RAM circuit 316 and an prograrnn~able and electrically erasable read only memory (PEROM) circuit 317. The PEROM circuit 317 includes non-volatile memory in which the executable control programs for the voice control DSP circuits 306 and the main controller circuits 313 operate.
20 The RS232 serial interface circuit 315 communicates to the serial port of the personal computer which is running the software components of the present system. The RS232 serial interface circuit 315 is connected to a serial input/output circuit 314 with main controller circuit 313. SIO
circuit 314 is in the preferred embodiment, a part of SIO/CTC cinvit 312.

Referring once again to Figure 3, the multiple and selectable functions described in conjunction with Figure 2 are all implemented in the hardware components of Figure 3. Each of these functions will be discussed in rum.
30 The telephone fimction 115 is implemented by the user either selecting a telephone number to be dialed from the address book 127 or manually selecting the number through the telephone menu on the personal compute. The telephone number to be dialed is downloaded from the personal cotrqnrter over the serial interface and received by main controller 313. Main cornrolla 313 causes the data pump DSP circuit 311 to seize the telephone line and transmit the DTMF tones to dial a number. Main 5 controller 313 configures digital telephone CODEC circuit 305 to enable either the handset 301 operation, the microphone 303 and speaks 304 operation or the headset 302 operation. A telephone connection is established through the telephone line interface circuit 309 and communication is enabled The users analog voice is transmitted in an analog fashion to the digital 10 telephone CODEC 305 where it is digitized. The digitized voice patters are passed to the voice control circuit 306 where echo cancellation is accomplished, the digital voice signals are reconstructed into analog signals and passed through multiplexor circuit 310 to the telephone line interface ci~uit 309 for analog transmission over the telephone line. The incoming 15 analog voice from the telephone connection through telephone connection circuit 309 is passed to the integral CODEC of the voice control circuit 306 where it is digitized. The digitized incoming voice is then passed to digital telephone CODEC circuit 305 where it is reconverted to an analog signal for transmission to the selected telephone interface (either the handset 301, the 20 microphone/speaker 303/304 or the headset 302). Voice Control DSP circuit 306 is programmed to perform echo cancellation to avoid feedback and echoes between transmitted and received signals, as is more fully described below.
In the voice mail function mode of the present system, voice messages may be stored for later transmission or the present system may 25 operate as an answering machine receiving incoming messages. For storing digitized voice, the telephone interface is used to send the analog speech patterns to the digital telephone CODEC circuit 305. Circuit 305 digitizes the voice patterns and passes them to voice control circuit 306 where the digitized voice patterns are digitally corr~pressed. The digitized and compressed voice 30 patterns are passed through dual port ram circuit 308 to the main controller circuit 313 where they are transferred through the serial interface to the persona( computer using a packet protocol defined below. The voice patterns are then stored on the disk of the personal computer for latrs use in mufti-media mail, f~ voice mail, as a pre-recorded ansvv~ering machine message or for later predetermined transmission to other sites.
For the present system to operate as an answering machine, the hardware components of Figure 3 are placed in answer mode. An incoming telephone ring is detected through the telephone line interface circuit 309 and the main controller circuit 313 is alerted which passes the infonwation off to the personal computer through the RS232 serial interface cin:uit 31 S. The _ telephone line interface circuit 309 seizes the telephone line to make the telephone carinecxion. A pre-recorded message may be sent by the personal computer as corrrpressed and digitized speech through the RS232 interface to the main controller circuit 313. The corr>pressed and digitized speech from the personal computer is passed from main controller circuit 313 through dual port ram circuit 308 to the voice control DSP circuit 306 where it is unco~ and converted to analog voice patterns. These analog voice patterns are passed through multiplexor circuit 310 to the telephone line interface 309 for transmission to the caller. Such a message may invite the caller to leave a voice message at the sound of a tone. The incoming voice messages are received through telephone line interface 309 and passed to voice control circuit 306. The analog voice patterns are digitized by the integral CODEC of voice control circuit 306 and the digitized voice patterns are compressed by the voice control DSP of the voice control circuit 306.
The digitized and compressed speech patterns are passed through dual port ram circuit 308 to the main controller circuit 313 where they are transferred using packet protocol described below through the RS232 serial interface 31 S
to the personal computer for storage and later retrieval. In this fashion the hardware components of Figure 3 operate as a transmit and receive voice mail system for implementing the voice mail function 117 of the present system.
The hardware components of Figure 3 may also operate to facilitate the fax manager function 119 of Figure 2. In fax receive mode, an incoming telephone call will be detected by a ring detect circuit of the telephone line interface 309 which will alert the main controller circuit 313 to the incoming call. Main controller circuit 313 will cause line interface circuit 309 to seize the telephone line to receive the call. Main controller circuit will also concutrently alert the operating programs on the personal compute through the RS232 interface using the packet protocol described below. Once 5 the telephone line interface seizes the telephone line, a fax carrier tone is transmitted and a return tone and handshake is received from the telephone line and detected by the data pump circuit 311. The reciprocal transmit and receipt of the fax tones indicates the imminent receipt of a facsimile transmission and the main controller circuit 313 configures the hardware 10 compon~ts of Figure 3 for the receipt of that information. The ner~ssaiy handshaking with the remote facsimile machine is accomplished through the data pump 311 under control of the main controller circuit 313. The incoming data packets of digital facsimile data are received over the telephone line interface and passed through data pump circuit 311 to main controller 15 circuit 313 which forwards the information on a packet basis (using the packet protocol described more fully below) through the serial interface circuit 315 to the personal computer for storage on disk. Those skilled in the art will readily recognize that the FAX data could be transferred from the telephone line to the personal computer using the same path as the packet transfer 20 except using the normal AT stream mode. Thus the incoming facsimile is automatically received and stored on the personal computer through the hardware components of Figure 3.
A facsimile transmission is also facilitated by the hardware components of Figure 3. The transmission of a facsimile may be immediate 25 or queued for later transmission at a predetermined or preselected time.
Control packet information to configure the hardware comp<ments to send a facsimile are sent over the RS232 serial interface between the personal computer and the hardware components of Figure 3 and are received by main controller circuit 313. The data pump circuit 311 then dials the recipient's 30 telephone number using DTMF tones or pulse dialing over the telephone line interface circuit 309. Once an appropriate connection is established with the remote facsimile machine, standard facsimile handshaking is accomplished by the data pump circuit 311. Once the facsimile connection is established, the digital facsimile picture information is received through the data packet protocol transfer over serial line interface circuit 315, passed through main controller circuit 313 and data pump circuit 311 onto the telephone line 5 through telephone line interface circuit 309 for receipt by the remote facsimile machine.
The operation of the multi-media mail function 121 of Figure 2 is also facilitated by the hardware components of Figure 3. A multimedia transmission consists of a combination of picture infomration, digital data and 10 digitized voice information. For example, the type of multimedia information transferred to a remote site using the hardvvare components of Figure 3 could be the multimedia format of the MicroSoft~ Multimedia Wave~ format with the aid of an Intelligent Serial Interface (ISI) card added to the personal computer. The multimedia may also be the type of multimedia information 15 assembled by the software component of the present system which is described more fully below.
The multimedia package of information including text, graphics and voice messages (collectively called the multimedia document) may be transmitted or received through the hardware components shown in Figure 3.
20 For example, the transmission of a multimedia document through the hardvvare components of Figure 3 is accomplished by transferring the multimedia digital information using the packet protocol described below over the RS232 serial interface betwe~ the personal corrrputer and the serial line interface circuit 315. The packets are then transferred through main controller 25 circuit 313 through the data pump circuit 311 on to the telephone line for receipt at a remote site through telephone line interface circuit 309. In a similar fashion, the multimedia documents received over the telephone line from the remote site are received at the telephone line interface circuit 309, passed through the data pump circuit 31 I for receipt and forwarding by the 30 main controller circuit 313 over the serial line interface circuit 315.
The show and tell function 123 of the present system allows the user to establish a data over voice communication session. In this mode of operation, full duplex data transmission may be accomplished simultaneously with the voice communication berweerr both sites. This mode of operation assumes a like configured remote site. The hardware components of the present system also include a means for sending voice/data over cellular links.
5 The protocol used for transmitting multiplexed voice and data include a supervisory packet described more fully below to keep the link established through the cellular link This s~ervisory packet is an acknowledgement that the link is still up. The sr~pervisory packet may also contain link information to be used for adjusting various link parameters when needed. This 10 supervisory packet is sent every second when data is not being sent and if the packet is not acknowledged after a specified number of attempts, the protocol would then give an indication that the cellular link is down and then allow the modem to take action. The action could be for example; change speeds, retrain, or hang up. The use of supervisory packets is a novel method of 15 maintaining inherently intermittent cellular links when transmitting multiplexed voice and data.
The voice portion of the voice over data transmission of the show and tell function is accomplished by receiving the user's voice through the telephone interface 301, 302 or 303 and the voice information is digitized 20 by the digital telephone circuit 305. The digitized voice information is passed to the voice control circuit 306 where the digitized voice information is compressed using a voice compression algorithm.described more fully below.
The digitized and compressed voice inforzration is passed through dual port RAM circuit 308 to the main controller circuit 313. Dwing quiet periods of 25 the speech, a quiet flag is passed from voice control circuit 306 to the main controller 313 through a packet trursfer protocol described below by a dual port RAM circuit 308.
Simultaneous with the digitizing compression and packetizing of the voice information is the receipt of the packetizsd digital information 30 from the personal computer over interface line circuit 315 by main controller circuit 313. Main controller circuit 313 in the show and tell function of the present system must eil<iciently and ei~'ectively combine the digitized voice information with the digital information for transmission over the telephone line via telephone line interface circuit 309. As described above and as described more fully below, main controller circuit 313 dynamically changes the amount of voice information and digital infom~ation transmitted at any 5 given period of time depending upon the quiet times during the voice transmissions. For example, during a quiet momern where there is no speech infon~nation being transmitted, main controller circuit 313 ensures that a higher volume of digital data infon~nation be transmitted over the telephone line interface in lieu of digitized voice infom~ation.
10 Also, as described more fully below, the packets of digital data transmitted over the telephone line interface with the transmission packet protocol described below, requires 100 percent accuracy in the transmission of the digital data, but a lesser standard of acc~uacy for the transmission and receipt of the digitized voice infoctt>ation. Since digital information must be 15 transmitted with 100 percent acc;~aacy, a corrupted packet of digital information received at the remote site must be re-transmitted. A
retransmission signal is communicated back to the local site and the packet of digital information which was corrupted during transmission is retransmitted If the packet transmitted contained voice data, however, the remote site uses 20 the packets whether they were comtpted ar not as long as the packet header was intact. If the header is comtpted, the packet is discarded. Thus, the voice information may be corrupted without requesting retransmission since it is understood that the voice information must be transmitted on a real time basis and the corruption of any digital information of the voice signal is not 25 critical. In contrast to this the transmission of digital data is critical and retransmission of com>pted data packets is requested by the remote site.
The transmission of the digital data follows the CCITT V.42 standard, as is well known in the industry and as described in the CCITT Blue Book, volume VIII entitled Data Communication over the Telephone Network, 30 1989. The voice data packet information also follows the CCITT V.42 standard, but uses a different header fom~at so the receiving site recognizes the difference between a data packet and a voice packet. The voice packet is distinguished from a data packet by using undefined bits in the header (80 hex) of the V.42 standard. The packet protocol for voice over data transmission during the show and tell function of the present system is described more fully below.
5 Since the voice over data communication with the remote site is full-duplex, incoming data packets and incoming voice packets are received by the hardware components of Figure 3. The incoming data packets and voice packets are received through the telephone line interface circuit 309 and passed to the main controller circuit 313 via data pump DSP circuit 311. The 10 incoming data packets are passed by the main controller circuit 313 to the serial interface circuit 315 to be passed to the personal computer. The incoming voice packets are passed by the main controller circuit 313 to the dual port RAM circuit 308 for receipt by the voice control DSP circuit 306.
The voice packets are decoded and the compressed digital inforn~ation therein 15 is uncompressed by the voice control DSP of circuit 306. The unco~
digital voice inforn~ation is passed to digital telephone CODEC circuit 305 where it is reconverted to an analog signal and retransmitted through the telephone line interface circuits. In this fashion full-duplex voice and data transmission and reception is accomplished through the hardware components 20 of Figure 3 doting the show and tell functional operation of the present system Terminal operation 125 of the present system is also supported by the hardware components of Figure 3. Terminal operation means that the local personal computer simply operates as a "dumb" terminal including file 25 transfer capabilities. Thus no local processing takes place other than the handshaking protocol required for the operation of a dumb terntinal. In terminal mode operation, the remote site is assumed to be a modem connected to a personal computer but the remote site is not necessarily a site which is configured according to the present system In terminal mode of operation, 30 the command and data information from personal computer is transferred over the RS232 serial interface circuit 31 S, forwarded by main controller cit~cuit 313 to the data pump circuit 311 where the data is placed on the telephone line via telephone line interface circuit 309.
In a reciprocal fashion, data is received from the telephone line over telephone line interface circuit 309 and simply forwarded by the data pump circuit 311, the main controller circuit 313 over the serial line interface 5 circuit 315 to the personal computer.
As described above, and more fully below, the address book function of the present system is primarily a support function for providing telephone numbers and addresses for the other various functions of the present system 10 T>erailed Electrical ~c~j~
The detailed electrical schematic diagrams comprise Figures SA-C, 6A-C, 7A-C, 8A-B, 9A-C and l0A-C. Figure 4 shows a key on how the schematic diagrams may be conveniently arranged to view the passing of signals on the electrical lines between the diagrams. The electrical 15 connections between the electrical schematic diagrams are through the designators listed next to each wire. For example, on the right side of Figure SA, address lines AO-A19 are attached to an address bus for which the individual electrical lines may appear on other pages as AO-A19 or may collectively be connected to other schematic diagrams through the designator 20 "A" in the circle connected to the collective bus. In a like fashion, other elecorical lines designated with symbols such as RNGL on the lower left-hand side of Figure SA may connect to other schematic diagrams using the same signal designator RNGL.
Beginning with the electrical schematic diagram of Figure 7C, 25 the telephone line connection in the preferred embodiment is through connector J2 which is a standard six-pin modular RJ-11 jack In the schematic diagram of Figure 7C, only the tip and ring connections of the first telephone circuit of the R,1-11 modular connector are used Fen-ite beads FB3 and FB4 are placed on the tip and ring wires of the telephone line connections 30 to remove any high frequency or RF noise on the incoming telephone line.
The incoming telephone line is also overvoltage protected through SII7ACTbR R4. The incoming telephone line may be full wave rectified by the full wave bridge comprised of diodes CR27, CR28, CR29 and CR31.
Swig S4 switches betvv~ direct connerxion and fill wave rectified connection depending upon whether the line is a non-powered leased line or a standard telephone line. Since a leased line is a "dead" line with no voltage, 5 the fill-wave rectification is not needed.
Also connected across the incoming telephone line is a ring detect circuit. Optical isolator U32 (part model numbs CNYI ~ senses the ring voltage threshold when it exceeds the breakdown voltages on zener diodes CRl and CR2. A filtering circuit shown in the upper right comer of 10 Figure 7C creates a long RC delay to sense the constant presence of an AC
ring voltage and buffers that signal to be a binary signal out of operational amplifier U25 (part model number TT.082). Thus, the RNGL and J1RING
signals are binary signals for use in the remaining portions of the electrical schematic diagrams to indicate a presence of a ring voltage on the telephone 15 line.
The present system is also capable of sensing the caller >D
information which is transmitted on the telephone line between rings.
Between the rings, optically isolated relays U30, U31 on Figure 7C and optically isolated relay U33 on Figure 7B all operate in the period between 20 the rings so that the FSK modulated caller ID information is connected to the CODEC and data pump DSP in Figures 8A and 8B, as described more fully below.
Referring now to Figure 7B, more of the telephone line filtering circuitry is shown. Some of the telephone line buffering circuitry such as 25 inductor Ll and resistor Rl are optional and are connected for various telephone line standards used around the word to meet local requirements.
For example, Switzerland requires a 22 millihenry inductor and 1K resistor in series the line. For all other countries, the 1 K resistor is replaced with a ohm resistor.
30 Relay U29 shown in Figure 7B is used to accomplish pulse dialing by opening and shorting the tip and ring wires. Optical relay X2 is engaged during pulse dialing so that the tip and ring are shorted directly.

Transistors Q2 and Q3 along with the associated discrete resistors cor>prise a holding circuit to provide a current path or ciutent loop on the telephone line to grab the line.
Figure 7A shows the telephone interface connections between the hardware components of the present system and the handset, headset and microphone.
The connecti~rs Tl and T2 for the telephone line firm Figure 7B ace connecxed to transformer TRl shown in the electrical schematic diagram of Figure 8B. Only the AC components of the signal pass through transformer TRl. The connection of signals attached to the secondary of TRI
is shown for both transmitting and_ receiving information over the telephone line.
Incoming signals are buffered by operational amplifiers U27A
and U27B. The first stage of buffering using operational amplifier U27B is used for echo sr>ppression so that the transmitted information being placed on the telephone line is not fed back into the receive portion of the present system The second stage of the input buffering through operational amplifier U27A is configured for a moderate amount of gain before driving the signal into CODEC U35.
CODEC chip U35 on Figure 8B, interface chip U34 on Figure 8A and digital signal processor (DSP) chip U37 on Figure 8A comprise a data pump chip set manufactured and sold by AT&T Mrcuoelectronics. A detailed description of the operation of these three chips in direct conrerxion and ion with one another is described in the publication entitled "AT&T
V.32bisN.32/FAX I~'rgh-Speed Data Pump Chip Set Data Book" published by AT&T Microelectronics, December 1991. This AT&T data pump chip set comprises the core of an integrated, two-wire firll duplex modem which is capable of operation over standard telephone lines or leased lines. The data pump chip set conforms to the telecommunications specifications in CCTIT
recommendations V.32bis, V.32, V.22bis, V.22, V.23, V.21 and is compatible with the Bell 212A and 103 modems. Speeds of 14,400, 9600, 4800, 2400, 1200, 600 and 300 bits per second are supported. This data pump chip set consists of a ROM-coded DSP16A digital siginal processor U37, and interface chip U34 and an AT&T T7525 linear CODEC U35. The AT&T V.32 data pump chip set is available from AT&T Microelectronics.
The chip set U34, U35 and U37 on Figures 8A and 8B perform 5 all A/D, D/A, modulation, demodulation and echo cancellation of all signals placed on or taken from the telephone line. The CODEC U35 performs DTMF tone generation and detection, signal analysis of call progress tones, etc. The transmission of information on the telephone line from CODEC U35 is through buffer U28A, through CMOS switch U36 and through line buffer 10 U25. The CMOS switch U36 is used to switch between the data pump chip set CODEC of circuit 310 (shown in Figure 3) and the voice control CODEC
of circuit 306 (also shown in Figure 3). The signal lines AOUTN and AOUTP conrspond to signals received from the voice control CODEC of circuit 306. CODEC U35 is part of circuit 31 I of Figure 3.
15 The main controller of controller circuit 313 and the support circuits 312, 314, 316, 317 and 308 are shown in Figures 5A-5C. In the preferred embodiment of the present system, the main controller is a 280180 eight-bit microprocessor chip. In the preferred implementation, microcontroller chip U17 is a 280180 microprocessor, part number Z84C01 20 by Zilog Inc. of Campbell, California (also available from Hitachi Semiconductor as part number HD64180Z). The Zilog 280180 eight-bit microprocessor operates at 12 MHz internal clock speed by means of an external crystal XTAL,, which in the preferred embodiment, is a 24.576 MHz crystal. The crystal circuit includes capacitors C4 and C5 which are 20 pf 25 capacitors and resistor R28 which is a 33 ohm resistor. The crystal and support circuitry is connected according to manufacfia~er's specifications found in the Zilog Intelligent Peripheral Controllers Data Book published by Zilog Inc. The product description for the Z84C01 280180 CPU from the Z84C01 280 CPU Product Specification pgs. 43-73 of the Zilog 1991 Intelligent 30 Peripheral Controllers databook.
The 280180 microprocessor in microcontroller chip UI7 is intimately connected to a serial/parallel I/O counter timer chip U15 which is, in the preferred embodiment, a Zilog 84090 CMOS 280 KIO
seriaUparallel/counter/timer integcateci circuit available from Zilog, Inc.
This multi-function I/O chip U15 combines the functions of a parallel input/output port, a serial input/output port, bus control circuitry, and a clock timer circuit 5 in one chip. The Zilog 284090 product specification describes the detailed internal operations of this circuit in the Zilog Intelligent Peripheral Controllers 1991 Handbook available from Zilog, Inc. 284090 CMOS Z80KI0 Product specification pgs. 205-224 of the Zilog 1991 Intelligent Peripheral Controllers databook.
10 Data and address buses A and B shown in Figure SA connect the 280180 mianprocessor in microcontroller U17 with the 280 KIO circuit U15 and a gate array circuit UI9, and to other portions of the electrical schematic diagrams. The gate array U19 includes miscellaneous latch and buffer circuits for the present system which norn~ally would be found in 15 discr~e SSI or MSI integrated cit~cuits. By combining a wide variety of miscellaneous support circuits into a single gate array, a much reduced design complexity and manufacturing cost is achieved A detailed description of the internal operations of gate array U19 is described more fully below in conjunction with schematic diagrams of Figures l0A IOC.
20 The memory chips which operate in conjunction with the 280 microprocessor in microcontroller chip UI7 are shown in Figure 5C. The connections A, B correspond to the connections to the address and data buses, respectively, found on Figure 5A. Memory chips U16 and UI3 are read-only melrrory (ROM) chips which are electrically alterable in place. These 25 prograrr>ir>able ROMs, typically referred to as flash PROMS or Progracnrr~able Erasable Read Only Memories (PEROMs) hold the program code and operating parameters for the present system in a non-volatile memory. Upon power-up, the programs and operating parameters are transferred to the voice control DSP RAM UI2, shown in Figure 9B.
30 In the preferred embodiment, RAM chip UI4 is a pseudostatic RAM which is essentially a dynamic RAM with a built-in refi~esh. Those skilled in the art will readily recognize that a wide variety memory chips may be usod and substituted for pseudo-static RAM U14 and flash PROMS U16 and U13.
Referring once again to Figure 3, the main controller circuit 313 communicates with the voice control DSP of circuit 306 through dual port 5 RAM circuit 308. The digital telephone CODEC circuit 305, the voice control DSP and CODEC circuit 306, the DSP RAM 307 and the dual port RAM 308 are all shown in detailed electrical schematic diagrams of Figures 9A-9C.
Refezting to Figure 9A, the DSP RAM chips U6 and U7 are 10 shown with associated support chips. Support chips Ul and U2 are in the preferred embodim~t part 74HCT244 which are TIL-level latches used to capnu~e data from the data bus and hold it for the DSP RAM chips U6 and U7. Circuits U3 and U4 are also latch circuits for also latching address information to control DSP RAM chips U6 and U7. Once again, the address 15 bus A and data bus B shown in Figure 9A are mufti-wire connections which, for the clarity of the drawing, are shown as a thick bus wire representing a grouping of individual wires.
Also in Figure 9A, the DSP RAMS U6 and U7 are connected to the voice control DSP and CODEC chip U8 as shown split between Figures 20 9A and 9B. DSP/CODEC chip U8 is, in the preferred embodiment, part number l~E~ DSP16C, digital signal processor and CODEC chip manufacriaed and sold by AT&T Nficroelectronics. This is a 16-bit progracnrnable DSP with a voice band sigma-delta CODEC on one chip.
Although the CODEC portion of this chip is capable of analog-to-digital and 25 digital-to-analog signal acquisition and conversion system, the actual D/A
and A/D functions for the telephone interface occur in digital telephone CODEC
chip U12 (corresponding to digital telephone CODEC circuit 305 of Figure 3).
Chip U8 includes circuitry for sampling, data conversion, anti-aliasing filtering and anti-imaging filtering. The progtamnsable control of 30 DSP/CODEC chip U8 allows it to receive digitized voice from the telephone interface (through digital telephone CODEC chip U12) and store it in a digitized form in the dual port RAM chip Ul 1. The digitized voice can then be passed to the main controller circuit 313 where the digitized voice may be transmitted to the personal .computer over the RS232 circuit 315. In a similar fashion, digitized voice stored by the main controller circuit 313 in the dual port RAM Ul l may be transferred through voice control DSP chip U8, 5 comrerted to analog signals by telephone CODEC U12 and passed to the user.
Digital telephone CODEC chip U12 includes a direct telephone handset interface on the chip.
The connections to DSP/CODEC chip U8 are shown split across Figures 9A and 9B. Address/data decode chips U9 and U10 on Figure 10 9A serve to decode address and data information from the combined address/data bus for the dual port RAM chip Ul l of Figure 9B. The interconnection of the DSP/CODEC chip U8 shown on Figures 9A and 9B is described more fully in the WEB DSP16C Digital Signal Processor/CODEC
Data Sheet published May, 1991 by AT&T Ivficroelectronics.
15 The Digital Telephone CODEC chip U12 is also shown in Figure 9B which, in the preferred embodiment, is part number T7540 Digital Telephone CODEC manufactured and sold by AT&T Ivhaoelearonics. A
more detailed description of this telephone CODEC chip U12 is described in the T7540 Digital Telephone CODEC Data Sheet and Addendum published 20 July, 1991 by AT&T Nficroelectronics.
Support circuits shown on Figure 9C are used to facilitate communication betvv~ CODEC chip U12, DSP/CODEC chip U8 and dual port RAM U11. For ale, an 8 kHz clock is used to synchronizre the operation of CODEC U12 and DSP/CODEC U8.
25 The operation of the dual port RAM Ul l is controlled both by DSP U8 and main controller chip U17. The dual port operation allows writing into one address while reading from another address in the same chip.
Both processors can access the exact same memory locations with the use of a contention protocol such that why one is reading the other cannot be writing.
In the preferred embodiment, dual port RAM chip Ul I is part number CYZC131 available from Cyprus Semiconductor. This chip includes built in contention control so that if two processors try to access the same memory location at the sarr~ time, the first one making the request gets control of the address location and the other processor must wait. In the preferred embodiment, a circular buffer is arranged in dual port RAM chip Ul l comprising 24 bytes. By using a circular buffs configuration with pointers into the buffer area, both processors will not have a contartion problem The DSP RAM chips U6 and U7 are connected to the DSP
chip U8 and also connected through the data and address buses to the Zilog miaocontroller U17. In this configuration, the main controller can download the control programs for DSP U8 into DSP RAMS U6 and U7. In this fashion, DSP control can be changed by the main controller or the operating programs on the personal computer, described more firlly below. The control programs stored in DSP chips U6 and U7 originate in the flash PEROM chips U16 and U17. The power-up control routine operating on controller chip U17 downloads the DSP control routines into DSP RAM chips U6 and U7.
The interface between the main controller circuit 313 and the personal computer is through SIO circuit 314 and RS232 serial interface 315.
These interfaces are described more firlly in conjunction with the detailed electrical schematic diagrams of Figure 6A-C. RS232 connection J1 is shown on Figure 6A with the associated control circuit and interface circuitry used to generate and receive the appropriate RS232 standard signals for a serial communications interface with a personal corr>puter. Figure 6B is a detailed elearical schematic diagram showing the generation of various voltages for powering the hardware components of the electrical schematic diagrams of hardware components 20. The power for the present hardware compcments is received on connector JS and controlled by power switch S34. From this circuitry of Figure 6B, plus and minus 12 volts, plus five volts and minus five volts are derived for operating the various RAM chips, controller chips and support circuitry of the present system. Figure C shows the interconnection of the status LED's found on the front display of the box 20.
Finally, the "glue logic" used to support various fimctions in the hardware components 20 are described in conjunction with the detailed electrical schematic diagrams of Figures l0A-lOC. The connections between Figures l0A and lOC and the previous schematic diagrams is made via the labels for each of the lines. For example, the LED status lights are controlled and held active by direct addressing and data cortttol of latches GAl and GA2. For a more detailed description of the connection of the glue logic of 5 Figures l0A lOC, the gate array U19 is shov~~n connected in Figures SA and 5B.
~t_h_e arrj~rr_ .~mt~onern A special packet protocol is used for communication between 10 the hardware components 20 and the personal computer (PC) 10. The protocol is used for transferring different types of information between the two devices such as the transfer of DATA, VOICE, and QUALIFIED
information. The protocol also uses the BREAK as defined in CCITT X28 as a means to maintain protocol synct~roniz~ion. A description of this BREAK
I S sequence is also described in the Statutory Invention Registration entitled "ESCAPE METHODS FOR MODEM COMMUNICATIONS", to Timothy D.
Gunn filed January 8, 1993.
The protocol has two modes of operation. One mode is packet mode and the other is stream mode. The protocol allows mixing of different 20 types of information into the data stream without having to physically switch modes of operation. The hardware component 20 will identify the packet received from the computer 10 and perform the appropriate action according to the specifications of the protocol. If it is a data packet, then the controller 313 of hardware component 20 would send it to the data pump circuit 311. If 25 the packet is a voice packet, then the controller 313 of hardware component 20 would distribute that infon~nation to the Voice DSP 306. This packet transfer mechanism also works in the reverse, where the controller 313 of hardvvare component 20 would give different infom~ation to the computer 10 without having to switch into different modes. The packet protocol also 30 allows commands to be sent to either the main controller 313 directly or to the Voice DSP 306 for controlling different options without having to enter a command state.

Packet mode is made up of 8 bit asynchronous data and is identified by a beginning synchronization character (O1 hex) followed by an ID/IrI ct~tac~r and that followed by the information to be sent. In addition to the ID/LI character codes defined below, those skilled in the art will readily 5 recognize that other ID/LI character codes could be defined to allow for additional types of packets such as video data, or alternate voice compression algorithm packets such as Codebook Excited Linear Ptrdictive Coding (CELP) algorithm, GS1V>; RPE, VSELP, etc.
Stream mode is used when large amounts of one type of packet 10 (VOICE, DATA, or QUALIF>EID) is being sent. The transmitter tells the receiver to eater stream mode by a unique command. Thetrafter, the transmitter tells the receiver to terminate stream mode by using the BREAK
command followed by an "AT' type command The command used to terminate the stream mode can be a command to enter another type of stream 15 mode or it can be a command to enter back into packet mode.
Currently there are 3 types of packets used: DATA, VOICE, and QUALIFIED. Table 1 shows the common packet parameters used for all three packet types. Table 2 shows the three basic types of packets with the subtypes listed.

TABIE 1: Pgcket Pgrametets 25 1. Async~u~onous transfer 2. 8 bits, no parity 3. Maximum packet length of 128 bytes - iDentifier byte = 1 - InFotmation = 127 30 4. SPEED
- variable from 9600 to 57600 - default to 19200 TABLE 2: Packet Types 5 1. Data 2. Voice 3. Qualified:
a COMMAND
b. RESPONSE
10 c. STATUS
d FLOW CONTROL
e. BREAK
f. ACK
g. NAK
15 h. STREAM
A Data Packet is shown in Table 1 and is used for normal data 20 transfer between the controller 313 of hardvvare component 20 and the computer 10 for such things as text, file transfers, binary data and any other type of information presently being sent through modems. All packet transfers begin with a synch character O1 hex (synchronization byte). The Data Packet begins with an ID byte which specifies the packet type and 25 packet length. Table 3 describes the Data Packet byte structure and Table 4 describes the bit strut of the )D byte of the Data Packet. Table 5 is an example of a Data Packet with a byte length of 6. The value of the LI field is the actual length of the data field to follow, not counting the ID byte.

TABLE 3: Data P~dcet Byte Strucdue byte 1 - O I h (sync byte) 35 byte 2 - 1D/LI (ID byte/length indicator) bytes 3-127 - data (depending on LI) O1 m ; '' p ; __ _ i.
40 SYNC ' IS data I~ data : data ~ data ~,~ data , TABLE 4: m Byte of Data Packet 5 Bit 7 identifies the type of packet Bits 6 - 0 contain the LI or length indicator portion of the ID byte i '~ 0 LI (Length Indicator) = 1 to 127 20 TABLE 5: Dad Packet Example L,I (lalgth indicator) = 6 0l 06 Ii li i: ', SYNC , ID data L' data i, data ' data ' data data j The Voice Packet is used to transfer compressed VOICE
messages between the controller 313 of hardware component 20 and the computer 10. The Voice Packet is similar to the Data Packet except for its length which is, in the prefen~ed embodiment, currently fixed at 23 bytes of data. Once again, all packets begin with a synchronization character chosen in the prefen~ed embodiment to be Ol hex (OlI-~. The ID byte of the Voice Packet is completely a zero byte: all bits are set to zero. Table 6 shows the m byte of the Voice Packet and Table 7 shows the Voice Packet byte structure.
TABLE 6: m Byte of Voice Paclaet , -i 0 I LI (Length Indicator) = 0 TABLE 7: Voice Packet Byte Stnicdne LI (length indicator) = 0 23 bytes of data O1 i 00 ', ' SYNC '. ID data ~ data ~ data data data -_ _ _. L
The Qualified Packet is used to transfer commands and other non-data/voice related infom~ation between the controller 313 of hardware component 20 and the computer 10. The various species or types of the Qualified Packets are described below and are listed above in Table 2. Once again, all packets start with a synchronization character chosen in the preferred embodiment to be OI hex (OlI-~. A Qualified Packet starts with two bytes where the first byte is the ID byte and the second byte is the QUAL>FTP~t type identifier. Table 8 shows the ID byte for the Qualified Packet, Table 9 shows the byte structure of the Qualified Packet and Tables 10-12 list the Qualifier Type byte bit maps for the three types of Qualified Packets.
TABLE 8: )(D Byte of Qualified Packet I I
i 1 j LI ~L~gth Indicator) = 1 to 127 The Length Identifier of the ID byte equals the amount of data which follows including the QUAhIFIER byte (QUAL byte + DATA). If LI
= 1, then the Qualifier Packet contains the Q byte only.

TABIE 9: Qual~er Pgcket Byte Stnrcdue of '' s5 !~ Stmt, SYNC ~ ID i': BYTE '' data ; data data ' data I:
_ _ The bit maps of the Qualifier Byte (QUAL BYTE) of the Qualified Packet are shown in Tables 10-12. The bit map follows the pattern _ whereby if the QUAL byte = 0, then the command is a break. Also, bit 1 of the QUAL byte designates ack/nak, bit 2 designates flow control and bit 6 designates stream mode command. Table 10 describes the Qualifier Byte of Qualified Packet,.Group 1 which are immediate commands. Table 11 describes the Qualifier Byte of Qualified Packet, Group 2 which are stream mode commands in that the command is to stay in the designated mode until a BREAK + INIT comrr~and string is sent. Table 12 describes the Qualifier Byte of Qualified Packet, ~.~oup 3 which are infon~naiion or status commands.
TABLE 10: Qualifier Byte of Qualified Picket: Group 1 x x x x x x x x 0 0 0 0 0 0 0 0 = break 0 0 0 0 0 0 1 0 = ACK
0 0 0 0 0 0 1 1 =NAK
0 0 0 0 0 1 0 0 = xoff or stop sending data 0 0 0 0 0 1 0 1 = xon or res~une sending data 0 0 0 0 1 0 0 0 = cancel fax TABLE 11: Qualifier Byte of Qualified Packet: (soap 2 x x x x x x x x 0 1 0 0 0 0 0 1 = stream command mode 10 0 1 0 00 0 = stream data 0 1 0 00 1 = stream voice 0 1 0 01 0 = stream video 0 1 0 01 1 = stream A

0 1 0 01 0 = stream B

15 0 1 0 01 1 = stream C

The Qualifier Packet indicating stream mode and BREAK
20 attention is used when a large of amount of infom~ation is sent (voice, data...) to allow the highest throughput possible. This command is mainly intended for use in DATA made but can be used in any one of the possible modes. To change from one mode to another, a break-init s~ce would be given. A
break "AT...<cr>" type command would cause a change in state and set the 25 serial rate from the "AT' command.
TABLE 12: Qualifier Byte of Qualified Packet: (soup 3 x x xx xx x x 1 0 00 00 0 0 = corntnmlds 35 1 0 00 00 0 1 = responses 1 0 00 00 1 0 = status 40 ('.eilttlar ~nr~rvicnnr Park In order to determine the status of the cellular link, a supervisory packet shown in Table 13 is used Both sides of the cellular link will send the cellular supervisory packet every 3 seconds. Upon receiving the cellular supervisory packet, the receiving side will acknowledge it using the 45 ACK field of the cellular supervisory packet. If the sender does not receive an aclmowledgement within one second, it will repeat sending the cellular supervisory packet up to 12 times. After 12 attempts of sending the cellular supervisory packet without an aclmowledgement, the sender will disconnect the line. Upon receiving an acknowledgement, the sender will restart its 3 second timer. Those skilled in the art will readily recognize that the timer values and wait times selected here may be varied without departing from the spirit or scope of the present invention.
TABIE 13: Cellular Supervisory Packet Byte Strucdae SF ID ~, LI ~ - I- Vita data _ - , data The Speech Compression algorithm described above for use in transmitting voice over data accomplished via the voice control circuit 306.
Referring once again to Figure 3, the user is talking either through the handset, the headset or the microphonelspeaker telephone interface. The analog voice signals are received and digitized by the telephone CODEC
circuit 305. The digitized voice information is passed from the digital telephone CODEC circuit 305 to the voice control circuits 306. The digital signal processor (DSP) of the voice control circuit 306 is programmed to do the voice compression algorithm. The DSP of the voice control circuit 306 compresses the speech and places the cod digital representations of the speech into special packets described more fully below. As a result of the voice compression algorithm, the compressed voice information is passed to the dual port ram circuit 308 for either forwarding and storage on the disk of the personal computer via the RS232 serial interface or for multiplexing with conventional modem data to be transmitted over the telephone line via the telephone line interface circuit 309 in the voice-over-data mode of operation Show and Tell function 123.

l~ritt~m To multiplex high-fidelity speech with digital data and transmit both ova the over the telephone line, a high available bandwidth would normally be required. In the present invention, the analog voice infom~ation 5 is digitized into 8-bit PCM data at an 8 kHz sampling rate producing a serial bit stream of 64,000 bps serial data rate. This rate cannot be transmitted over the telephone line. With the Speech Compression algorithm described below, the 64 kbs digital voice data is compressed into a 9500 bps encoding bit stream using a fixed-point (non-floating point) DSP such that the compressed 10 speech can be transmitted over the telephone line multiplexed with asynchronous data. This is accomplished in an efficient manner such that enough machine cycles remain during real time speech compression to allow to allow for echo cancellation in the same fixed-point DSP.
A silence detection function is used to detect quiet intervals in 15 the speech signal which allows the data processor to substitute asynchronous data in lieu of voice data packets over the telephone line to efficiently time multiplex the voice and asynchronous data transmission. The allocation of time for asynchronous data transmission is constantly changing depending on how much silence is on the voice channel.
20 The voice compression algorithm of the present system relies on a model of human speech which shows that human speech contains rediuidancy inherent in the voice patterns. Only the incremental innovations (changes) need to be transmitted The algorithm operates on 128 digitized speech samples (20 milliseconds at 6400 Hz), divides the speech samples into 25 time segments of 32 samples (5 milliseconds) each, and uses predicted coding on each segment. Thus, the input to the algorithm could be either PCM data sampled at 6400 Hz or 8000 Hz If the sampling is at 8000 H~ or any other selected sampling rate, the input sample data stream must be decimated fiom 8000 Hz to 6400 Hz before procxssing the speech data. At the output, the 30 6400 Hz PCM signal is interpolated back to 8000 Hz and passed to the CODEC.
With this algorithm, the cturent segment is predicted as best as possible based on the past recreated segmertu and a difference signal is determined The difference values are compared to the stored difference values in a lookup table or code book, and the address of the closest value is sent to the remote site along with the predicted gain and pitch values for each 5 segment. In this fashion, the entire 20 milliseconds of speech can be represented by 190 biu, thus achieving an effective data rate of 9500 bps.
To produce this compression, the present system includes a unique Vector Quarttization (VQ) speech compression algorithm designed to provide maximum fidelity with minimum compute power and bandwidth. The 10 VQ algorithm has two major componenu. The first section reduces the dynamic range of the input speech signal by removing short term and long term redundancies. This reduction is done in the waveform domain, with the synthesized part used as the reference for determining the incremental "new"
content. The second section maps the residual signal into a code book 15 optimized for preserving the general spectral shape of the speech signal.
Figure 11 is a high level signal flow block diagram of the speech compression algorithm used in the present system to compress the digitized voice for transmission over the telephone line in the voice over data mode of operation or for storage and use on the personal computer. The 20 transmitter and receiver componenu are implemented using the programmable voice control DSP/CODEC circuit 306 shown in Figtae 3.
The DC removal stage I 101 t~oceives the digitized speech signal and removes the D.C. bias by calculating the long-term average and subtracting it from each sample. This ensut~es that the digital samples of the 25 speech are centered about a zero mean value. The pre-emphasis stage 1103 whitens the spectral contest of the speech signal by balancing the extra energy in the low band with the reduced energy in the high band.
The system finds the innovation in the current speech segment by subtracting 1109 the prediction from reconstructed past samples 30 synthesized from synthesis stage 1107. This process requires the synthesis of the past speech samples locally (analysis by synthesis). The synthesis block 1107 at the transmitter performs the same function as the synthesis block 1113 at the receiver. When the reconstructed previous segm~t of speech is sr.ibttacxed from the present segment (before prediction), a difference term is produced in the form of an error signal. This residual error is used to find the best match in the code book 1105. The code book 1105 quantizes the error 5 signal using a code book generated from a representative set of speakers and environments. A minimum mean squared error match is determined in segments. In addition, the code book is designed to provide a quantization error with spearal rolloff (higher quantization error for low fiuquerrcies and lower quantization error for higher fimquencies). Thus, the quantization noise 10 spectrum in the reconstructed signal will always tend to be smaller than the underlying speech signal.
The channel corresponds to the telephone line in which the compressed speech bits are multiplexed with data bits using a packet format described below. The voice bits are sent in packets of 5 frames each, each 15 frame corresponding to 20ms of speech in 128 samples. The size of the packets depends upon the ype of compression used Three compression algorithms are described which will be called 8K, 9.6K and 16K The 8K and 9.6K algorithms results in a 24 byte packet while the 16K algorithm produces a packet of 48 bytes for each 20 ms speech segment.
20 Each flame of 20ms is divided into 4 sub-blocks or segments of Sms each. In each sub-block of the data consists of a plurality of bits for the long term predictor, a plurality of bits for the long term predictor gain, a plurality of bits for the sub-block gain, and a plurality of bits for each code book entry for each Sms. The bits for the code book entries consists of four 25 or five table entries in a 256 long code book of 1.25 ms duration. In the code book block, each 1.25ms of speech is looked up in a 256 word code book for the best match. The table entry is transmitted rather than the actual samples.
The code book entries are pre-computed from representative speech segments, as described more fully below.
30 On the receiving end 1200, the synthesis block 1113 at the receiver performs the same fwrction as the synthesis block 1107 at the transmitter. The synthesis block 1113 reconstructs the original signal from the voice data packets by using the gain and pitch values and code book address corresponding to the error signal most closely matched in the code book The code book at the receiver is similar to the code book 1105 in the transmitter.
Thus the synthesis block recreates the original pre-ernphasizad signal. The 5 de-emphasis stage 1115 inverts the pre~errrphasis operation by restoring the balance of original speech signal.
The complete speech compression algorithm is summarized as follows:
a) Digitally sample the voice to produce a PCM sample bit 10 stream sampled at 16,000 samples per second, 9600 samples per second or 8,000 samples per second b) Decimate the sampled data to produce a common sampling rate of 8,000 samples per second from all of the 15 actual sample rates.
c) Remove any D.C. bias in the speech signal.
d) Pre-emphasize the signal.

e) Find the innovation in the current speech segment by subtracting the prediction from reconstructed past samples.
This step requires the synthesis of the past speech samples locally (analysis by synthesis) such that the residual error is fed 25 back into the system.
f) Quantize the error signal using a code book generated from a representative set of speakers and environments. A
minimum mean squared error match is determined in Sms 30 segments. In addition, the code book is designed to provide a quantization error with spectral rollo$' (higher quantization error for low frequencies and lower quantization error for higher fi~equencies). Thus, the quantization noise specwm in the reconst<ucted signal will always tend to be smaller than the 35 underlying speech signal.
g) At the transmitter and the receiver, reconstruct the speech from the quantized error signal fed into the inverse of the fimdion in step (e) above. Use this signal for analysis by 40 synthesis and for the output to the reconstruction stage below.
h) Use a de-emphasis filter to reconstruct the output.

The major advantages of this approach over other low-bit-rate algorithms are that there is no need for any complicated calculation of reflection coefficients (no matrix inverse or lattice filter computations).
Also, the quantization noise in the output speech is hidden ands the speech signal 5 and there are no pitch tracking artifacts: the speech sounds "natural", with only minor increases of background hiss at lower bit-rates. The computational load is reduced significantly cored to a VSELP algorithm and variations of the present algorithm thus provides bit rates of 8, 9.6 and 16 Kbids, and can also provide bit rates of 9.2kbits/s, 9.Skbits/s and many other rates. The 10 total delay through the analysis section is less than 20 milliseconds in the preferred embodiment. The present algorithm is accomplished completely in the waveform domain and there is no spectral information being computed and there is no filter computations needed.
15 The speech compression algorithm is described in greater detail with reference to Figures 12 through 15, and with reference to the block diagram of the hardware components of the present system shown at Figure 3.
The voice compression algorithm operates within the programmed control of the voice control DSP circuit 306. In operation, the speech or analog voice 20 signal is received through the telephone interface 301, 302 or 303 and is digitized by the digital telephone CODEC circuit 305. The CODEC for circuit 305 is a companding p-law CODEC. The analog voice signal finm the telephone interface is band-limited to about 3,000 Hz and sampled at a selected sampling rate by digital telephone CODEC 305. The sample rates in 25 the preferred embodiment of the present invention are 8kbJs, 9.6kbJs and l6kb/s. Each sample is encoded into 8-bit PCM data producing a serial 64kb/s, 76.8kb/s or 128kb/s signal, respectively. The digitized samples are passed to the voice control DSP/CODEC of circuit 306. There, the 8-bit p-law PCM data is converted to 13-bit linear PCM data. The 13-bit 30 representation is necessary to accurately represent the linear version of the logarithmic 8-bit p-law PCM data. With linear PCM data, simpler mathematics may be performed on the PCM data.

The voice control DSP/CODEC of circuit 306 correspond to the single integrated circuit U8 shown in Figures 9A and 9B as a WEB DSP16C
Digital Signal Prooessot/CODEC from AT&T lVficroelect<onics which is a combined digital signal processor and a linear CODEC in a single chip as 5 described above. The digital telephone CODEC of circuit 305 corresponds to integrated circuit U12 shown in Figure 9(b) as a T7540 companding P-law CODEC.
The sampled and digitized PCM voice signals from the telephone p-law CODEC 305 shown in Figure 3 are passed to the voice 10 control DSP/CODEC circuit 308 via direct data lines clocked and synchronized to a clocking &~equency. The sample rates in CODEC 305 in the preferred embodiment of the present invention are 8kb/s, 9.6kb/s and l6kb/s.
The digital samples are loaded into the voice control DSP/CODEC one at a time through the serial input and stored into an internal queue held in RAM, 15 converted to linear PCM data and decimated to a sample rate of 6.4bb/s. As the samples are loaded into the end of the queue in the RAM of the voice control DSP, the samples at the head of the queue are operated upon by the voice compression algorithm The voice compression algorithm then produces a greatly compressed representation of the speech signals in a digital packet 20 form. The corripressed speech signal packets are then passed to the dual port RAM circuit 308 shown in Figure 3 for use by the main controller circuit 313 for either transferring in the voice-over-data mode of operation or for transfer to the personal computer for storage as corrrpressed voice for functions such as telephone answering machine message data, for use in the mufti-media 25 documents and the like.
In the voice-over-data mode of operation, voice control DSP/CODEC circuit 306 of Figtue 3 will be receiving digital voice PCM data from the digital telephone CODEC circuit 305, compressing it and transferring it to dual port RAM circuit 308 for multiplexing and transfer over the 30 telephone line. This is the transmit mode of operation of the voice control DSP/CODEC circuit 306 corresponding to transmitter block 1100 of Figure 11 and corresponding to the compression algorithm of Figure 12.

Concrnretit with this transmit operation, the voice control DSP/CODEC circuit 306 is receiving cod voice data packets from dual port RAM circuit 308, uncompressing the voice data and transferring the uncornptessed and reconstructed digital PCM voice data to the digital 5 telephone CODEC 305 for digital to analog conversion and eventual transfer to the user through the telephone interface 301, 302, 304. This is the receive mode of operation of the voice control DSP/CODEC circuit 306 corresponding to t~aceiv~ block 1200 of Figure 11 and corresponding to the decompression algorithm of Figure 13. Thus, the voice-control DSP/CODEC
10 circuit 306 is processing the voice data in both directions in a full-duplex fashion.
The voice control DSP/CODEC circuit 306 operates at a clock frequency of approximately 24.576MHz while processing data at sampling rates of approximately BKHz in both directions. The voice 15 compression/decompression algorithms and packetiration of the voice data is accomplished in a quick and efficient fashion to ensure that all procxssing is done in real-time without loss of voice infornration. This is accomplished in an efficient manner such that arough machine cycles remain in the voice control DSP circuit 306 druing real time speech compression to allow real 20 time acoustic and line echo cancellation in the same fixed-point DSP.
In pcogcatrnried operation, the availability of an eight-bit sample of PCM voice data from the lt-law digital telephone CODEC circuit 305 causes an intemtpt in the voice control DSP/CODEC circuit 306 where the sample is loaded into intecrrat registers for processing. Once loaded into an 25 internal register it is transferred to a RAM address which holds a queue of samples. The queued PCM digital voice samples are converted from 8-bit lt-law data to a 13-bit linear data fornrat using table lookup for the conversion.
Those skilled in the art will readily recognize that the digital telephone CODEC circuit 305 could also be a linear CODEC.
30 ~p~?~jlT~j~
The sampled and digitized PCM voice signals from the telephone lt-law CODEC 305 shown in Figure 3 are passed to the voice control DSP/CODEC circuit 308 via direct data lines clocked and synctunnized to a clocking frequency. The sample rates in the preferred embodiment of the present invaltion are 8kb/s, 9.6kb/s and l6kb/s. The digital samples for the 9.6K and 8K algorithms are decimated using a digital 5 decimation process to produces a 6.4K and 6K sample rate, respectively. At the 16K sampling rate for the 16K algorithm, no decimation is needed for the voice compression algorithm Referring to Figure 11, the decimated digital samples are shown as speech entering the transmitter block 1100. The transmitter block, of 10 course, is the mode of operation of the voice-control DSP/CODEC circuit 306 operating to receive local digitized voice infomlation, compress it and packetize it for transfer to the main controller circuit 313 for transmission on the telephone line. The telephone line connected to telephone line interface 309 of Figure 3 cArresponds to the channel 1111 of Figure 11.
15 A frame rate for the voice compression algorithm is 20 milliseconds of speech for each compression. 'Ibis correlates to 128 samples to process per frame for the 6.4K decimated sampling rate. When 128 samples are accumulated in the queue of the internal DSP RAM, the compression of that sample frame is begun.

The voice-control DSP/CODEC circuit 306 is programmed to first remove the DC component 1101 of the incoming speech. The DC
removal is an adaptive function to establish a center base line on the voice signal by digitally adjusting the values of the PCM data. This corresponds to 25 the DC removal stage 1203 of the software flow chart of Figure 12. The formula for removal of the DC bias or drift is as follows:

a(n) = s(n) - s(n-1) + oc * x (n-1) where a =

and where n = sample number, s(n) is the current sample, and x(n) is the sample with the DC bias removed The removal of the DC is for the 20 millisecond frame of voice which amounts to 128 samples at the 6.4K decimated sampling rate which corresponds to the 9.6K ALGORITHM The selection of a is based on empirical observation to provide the best result.
S Referring again to Figure 12, the voice compc~ession algorithm in a control flow diagram is shown which will assist in the understanding of the block diagram of Figure 11. Figure 14 is a simplified data flow description of the flow chart of Figure 12 showing the sample rate decimator 1241 and the sample rate incrementor 1242. The analysis and compression begin at block 1201 where the 13-bit linear PCM speech samples are accumulated until 128 samples (for the 6.4K decimated sampling rate) representing 20 milliseconds of voice or one frame of voice is passed to the DC removal portion of code operating within the programmed voice control DSP/CODEC circuit 306. The DC removal portion of the code described above approximates the base line of the frame of voice by using an adaptive DC removal technique.
A silence detection algorithm 1205 is also included in the programmed code of the DSP/CODEC 306. The silence detection function is a sunvnation of the square of each sample of the voice signal over the frame.
If the power of the voice frame falls below a preselected threshold, this would indicate a silent frame. The detection of a silence frame of speech is important for later multiplexing of the V-data (voice data) and C-data (asynchronous computer data) described below. Dining silent portions of the speech, the main controller circuit 313 will transfer conventional digital data (C-data) over the telephone line in lieu of voice data (V-data). The formula for computing the power is PWR = ~ x (a) * x (n) n=0 where n is the sample number, and x (a) is the sample value If the power PWR is lower than a preselected threshold, then the present voice frame is flagged as containing silence. The 128-sample silent frame is still processed by the voice compression algorithm; however, 5 the silent flame packets are discarded by the main controller circuit 313 so that asynchronous digital data may be transferred in lieu of voice data The rest of the voice compression is operated upon in segments where there are four segments per frame amounting to 32 samples of data per segment. It is only the DC removal and silence detection which is accomplished over an 10 entire 20 millisecond frame.
The pre-emphasis 1207 of the voice compression algorithm shown in Figure 12 is the next step. The sub-blocks are first passed through a pre-emphasis stage which whitens the spectral content of the speech signal by balancing the extra energy in the low band with the reduced energy in the 15 high band. The pre-emphasis essentially flattens the signal by reducing the dynamic range of the signal. By using pre-emphasis to flatted the dynamic range of the signal, less of a sip~al range is required for compression making the compression algorithm operate more efficiently. The formula for the pre-emphasis is x(n)=x(~-p*x(n-1) wherep=0.5 and where n is the sample munber, x (r~ is tt>e saicgtle 25 Each segment thus amounts to five milliseconds of voice which is equal to 32 samples. Pre-emphasis then is done on each segment. The selection of p is based on empirical observation to provide the best result.
The next step is the long-term prediction (LTP). The long-term prediction is a method to detect the innovation in the voice signal. Since the 30 voice signal contains many redundant voice segments, we can detect these redundancies and only send infonriation about the changes in the signal from one segment to the next. This is accomplished by comparing the speech samples of the current segment on a sample by sample basis to the reconst<ucted speech samples from the previous segments to obtain the innovation inforntation and an indicator of the error in the prediction.
The long-term predictor gives the pitch and the LTP-Gain of the sub-block which are encoded in the transmitted bit stream In order to 5 predict the pitch in the ctarent segment, we need at least 3 past sub-blocks of reconstructed speech. This gives a pitch value in the range of IvIIN_PITCH
(32) to MAX PITCH (95). This value is coded with Orbits. But, in order to accotnrrmdate the cotr>pressed data rate within a 9600 bps link, the pitch for segments 0 and 3 is encoded with 6 bits, while the pitch for segments 1 and 2 10 is encoded with 5 bits. When performing the prediction of the Pitch for segments 1 and 2, the correlation lag is adjusted around the predicted pitch value of the previous segment. This gives us a good chance of predicting the correct pitch for the current segment even though the entire range for prediction is not used The computations for the long-term correlation lag 15 PITCH and associated LTP gain factor ~i j (where j = 0, 1, 2, 3 corresponding to each of the four segments of the frame) are done as follows:
For j = ruin pitch .... mao~ibch, first perform the following computations between the current speech samples x(n) and the past 20 reconstructed speech samples xYr~

S=. (~) _ ~ x (i) * x' (i + MAX PITCH-, j) 5,~. ~) _ ~ x' (i + ~~~,qX PITCH j) * x' (i+tl~lqX PITCH; j) r-o The Pitch j is chosen as that which maximizes S~
g~.

Since (3 j is positive, only j with positive S,~ is considered Since the Pifch is encoded with different number of bits for each sub-segment, the value of min~itch and m~~itch (range of the synthesized speech for pitch prediction of the current segment) is computed as follows:
if (seg-number = 0 or 3) {
min~itch = IvBN PITCH
max_pitch = MAX PITCH
if (seg_number = 1 or 2) {
min~itch = prev~itch - 15 if (prev~itch < NNIINN PITCH + 15) min~itch = MIN PITCH
if (prev~itch > MAX PITCH + 15) min~itch = MAX PITCH - 30 max~itch = min~itch + 30 The piev_pibch parameter in the above equation, is the of the pitch of the previous sub-segment. The pitch j is the encoded in 6 bits or 5 bits as:
encoded bits = j - min pitch The LTP-Gain is given by sa- G7 (3 = for Sx,~~)~0 S=x~ G) The vale of the ~i is a normalized quantity between zero and unity for this segment where (3 is an indicator of the correlation between the segments. For example, a perfect sine wave would produce a ~i which would be close to unity since the correlation between the current segments and the previous reconstructed segments should be almost a perfect match so (3 is one.
The LTP gain factor is quantized from a LTP Gain Encode Table. This table is characterized in Table 14. The resulting index (bcode) is transmitted to the far end. At the receiver, the LTP Gain Factor is retrieved from Table 15, as follows:
~iq = dlb tc>b[bcode]
TABLE 14: LTP Ga~ia F.hcode Table 0.1 0.3 0.5 0.7 0.9 bcode= 0 1 2 3 4 5 TABLS 15: LTP C3ain Decode Table (i= 0.0 0.2 0.4 0.5 0.8 1.0 >
lxode=0 1 . 2 3 4 5 After the Long-Term Prediction, we pass the signal through a pitch filter to whiten the signal so that all the pitch effects are removed The pitch filter is given by:
a (n) = x (n) - ~ ' x' (n-~~
where j is the Lag; and (3q is the associated Gain.
Next, the error signal is normalized with respect to the maximum amplitude in the sub-segment for vector-quantization of the error signal. The maximum amplitude in the segment is obtained as follows:
G=MAX(Ie(n)~}
The maximum amplitude (G'~ is encoded using the Gain Encode Table. This table is characterized in Table 16. The encoded amplitude (gcode) is transmitted to the far end At the recxiver, the maximum amplitude is rehieved from Table 17, as follows:
Gq = dlg tc~~gcodeJ
The error signal e(n) is then normalized by r e(n) e(n) _ Gq TABLE 16: Gain F~ode Table G=16 32 64 128 256 512 1024 2048 4096 8192 ->

(gcode) TABLE 17: Gain Decode Table G=16 32 64 128 256 512 1024 2048 4096 8192 (gcode) From the Gain and LTP Gain Encode tables, we can see that we would require 4 bits far gcode and 3 bits for bcode. This results in total of 7 bits for both parameters. In order to reduce the bandwidth of the corr~pressed bit stream, the gcode and bcode parameters are encoded together in 6 bits, as follows:

BGCODE = 6 * gcode + bcod,e The encoded bits for the G and LTP-Gain (~ at the receiver can be obtained as follows:
5 gcode = BGCODE / 6 bcode = BGCODE - 6 * gcode Each segrnetrt of 32 samples is divided into 4 vectors of 8 samples each. Each vector is compared to the vectors stored in the CodeBook and the Index of the Code Vector that is closest to the signal vector is 10 selected. The CodeBook consists of 512 entries (S 12 addresses). The index chosen has the least difference according to the following minimalization formula:
~ ~~ (x ~ - Y;~
;
is where x; = the input vector of 8 samples, and y; = the code book vector of 8 samples The minimization computation, to find the best match between 20 the subsegment and the code book entries is computationally intensive. A
brute force corr~parison may exceed the available machine cycles if real time processing is to be accomplished. Thus, some shorthand processing approaches are taken to reduce the computations required to find the best fit.
The above formula can be computed in a shorthand fashion as follows.
25 By expanding out the above formula, some of the unnecessary temps may be removed and some fixed terms may be pre-computed:
(~-Y~~_(~-Y.)*(~-Y.) - (~z - x;Y~ - xJ'~ + Y z) =(~z_~''+Yz) 30 where x ? is a constant so it may be dropped from the formula, and the value of -'/~ ~y? may be precomputed and stored as the 9th value in the code book so that the only real-time computation involved is the following formula:
5 Min {~ (x; Y; )~
Thus, for a segm~t of 32 samples, we will transmit 4 CodeBook Indexes (9 bits each) corresponding to 4 subsegments of 8 samples 10 each. This means, for each segment, we have 36 bits to transmit.
After the appropriate index into the code book is chosen, the input speech samples are replaced by the con~sponding vectors in the chosen indexes. These values are then multiplied by the Gq to denom~alize the synthesized error signal, e'(n). This signal is then passed through the Inverse 15 Pitch Filter to reintroduce the Pitch effects that was taken out by the Pitch filter. ~ The Inverse Pitch Filter is performed as follows:
Y~n~ = eYnJ + ~a * x~ ~n 'J~
20 where ~3 q is the decoded LTP-Gain from Table 16, and j is the Lag.
The Inverse Pitch Filter output is used to update the synthesized speech buffer which is used for the analysis of the next sub-segment. The update of the state buffer is as follows:
25 x' (k) =x' (k +~V pl'1'CI~
where k = 0, ... , (MAX PITCH - MIN PITCH) - 1 x'~~ =Y~n~
where 1 = MAX PITCH - MIN PITCH, ..., MAX PITCH - 1 The signal is then passed through the deemphasis filter since preemphasis was performed at the beginning of the processing. In the analysis, only the preerrrphasis state is updated so that we properly satisfy the Analysis-by-Synthesis method of performing the compression. In the Synthesis, the output of the deemphasis filter, s' (n), is passed on to the D/A
to generate analog speech. The deernphasis filter is implemented as follows:
s'(n) =Y (n) + P * s' (n -1) where p = 0.5 The voice is reconstnicted at the receiving end of the voice-over data link according to the reverse of the compression algorithm as shown as the decompression algorithm in Figure 13.
If a silence frame is received, the decompression algorithm simply discards the received flame and initialize the output with zeros. If a speech frame is received, the pitch, LTP-Gain and GAIN are decoded as explained above. The error signal is reconsrtucted from the codebook indexes, which is then denormalized with respect to the GAIN value. This signal is the passed through the Inverse filter to generate the neconstrucxed signal. The Pitch and the LTP-Gain are the decoded values, same as those used in the Analysis. The filtered signal is passed through the Daerrrphasis filter whose output is passed on to the D/A to put out analog speech.
The cod frame contains 23 8-bit words and one Orbit word. Thus a total of 24 words. Total number of bits transferred is 190, which corresponds to 9500 bps as shown in Table 18.

7 6 S 4 3 2 1 0 Bit Niunber S S ~s ~4 ~3-. ~z ~i ~ Comp Frame[0]

Vz8 VIg Vo8 P14 P13 Ptz PII PI Comp Frame[I]

Vsg V4a V38 Pz4 Pz3 Pzz Pzl pi Comp Ftame[2]

10 V7g V68 pas p34 p33 p;z p31 p3 Comp Frame[3]

V98 Vg$ B~5 ~4 BG3 ~2 TZl:1B(j Comp_Frame[4]
~' ~' '' ''0 ''0 ~' W W ~1' W ~ W Comp_Frame[5]
s 4 z I
I

V138Vlze BGzsB('h4BCrz3BGzzBGzI BG2 Comp Frame[6]

VI48 ~35 ~34 ~33 ~32 ~31 ~3 ~~ F[~

15 VQ' VQ VQs VQ VQ' VQz VQI VQ Comp Frame[g]
6 LS 8 bits VQ[O]

VQI7VQI6 VQISVQI4VQ13 VQI2VQII VQl CormFrame[9]
-LS 8 bits VQ[ I ]

VQ14VQI4 VQ14VQ14VQI4 VQi4VQ14 VQI4~~ F~le[22]

6 5 4 3 Z 1 ~ g bltS
VQ[ 14]

20 VQISVQu VQISVQISVQIS VQISVQIS VQIS~mP_F~[23]

7 6 s 4 3 z I ~ g bits VQ[ I S]

where BG = Befa/Gain, P = Pitch, VQ = CodeBook Index and S = Spare Bits Table 19 describes the format of the code book for the 9.6K algorithm The code book values in the appendices are stored in a signed floating point format which is converted to a Q22 value fixed point digital format why 5 stored in the lookup tables of the present invention. There are 256 entries in each code book corresponding to 256 different speech segments which can be used to encode and reconstruct the speech Code Book Entries - '/z Sum2 Constant-8 entries 1 entry For the 9.6K algorithm, the code book comprises a table of nine columns and 256 rows of floating point data. The first 8 rows 20 correspond to the 8 samples of speech and the ninth entry is the precomputed constant described above as -'/z E y Z. An example of the code book data is shown in Table 20.

i ao~ e r~ ~~pm ~e lu: Book for the Ln 9 6K Al i m 0.786438 1.1328751.2083751206750gor 3.93769 1.114250 0.937688 0.772062 0.583250 0.609667 1.0191670.9091670.9577500.999833 1.005667 3.36278 0.854333 0.911250 0.614750 1.1507501.4777501.5487501.434750 1.349750 6.95291 1.304250 1.428250 30 0.657000 1.2799091.2047271.335636 1.162000 5.24933 1.132909 1.280818 0.958818 0.592429 0.8975711.1017141.3372861.323571 1.304857 5.6239 1.349000 1.347143 0.325909 0.7741821.0357271.2636361.456455 1.076273 4.628 1.356273 0.872818 35 The code books are converted into Q22 format and stored in PROM memory accessible by the Voice DSP as a lookup table. The table data is loaded into local DSP memory upon the selection of the appropriate algorithm to increase access speed. The code books comprise a table of data in which each entry is a sequential address from 000 to 511. For the 9.6K
algorithm, a 9 X 512 code book is used For the 16K algorithm, a 9 X 512 code book is used and for the 8K algorithm, a 9 X 512 code book is used Depending upon which voice compression quality and compression rate is selected, the corresponding code book is used to encode/decode the speech samples.
C~neration of th_e Code Books The code books are generated statistically by encoding a wide variety of speech patterns. The code books are generated in a learning mode for the above-described algorithm in which each speech segment which the compression algorithm is first exposed to is placed in the code book until 512 entries are recorded. Then the algorithm is continually fed a variety of speech patterns upon which the code book is adjusted As new speech segments are encountered, the code book is searched to find the best match. If the error between the observed speech segment and the code book values exceed a predetermined threshold, then the closest speech segment in the code book and the new speech segment is averaged and the new average is placed in the code book in place of the closest match. In this learning mode, the code book is continually adjusted to have the lowest difference ratio between observed speech segment values and code book values. The learning mode of operation may take hours or days of exposure to different speech patterns to adjust the code books to the best fit.
The code books may be exposed to a single person's speech which will result in a code book being tailored to that particular persons method of speaking. For a mass market sale of this prvoduct, the speech patterns of a wide variety of speakers of both genders are exposed to the code book learning algorithm for the average fit for a given language. For other languages, it is best to expose the algorithm to speech patterns of only one language such as English or Japanese.
Voice Over Data Packet Protocol As described above, the present system can transmit voice data and conventional data concurrartly by using time multiplex technology. The digitized voice data, called V-data caries the speech information. The conventional data is referred to as C-data. The V-data and C-data multiplex transmission is achieved in two modes at two levels: the transmit and receive rrrodes and data service level and multiplex control level. This operation is 5 shown di~cally in Figure 15.
In transmit mode, the main controller circuit 313 of Figure 3 operates in the data service level 1 SOS to collect and buffer data from both the personal computer 10 (through the RS232 port interface 315) and the voice control DSP 306. In multiplex corrt<ol level 1515, the main controller circuit 10 313 multiplexes the data and transmits that data out over the phone line 1523.
In the receive mode, the main controller circuit 313 operates in the multiplex control level 1515 to de-multiplex the V-data packets and the C-data packets and then operates in the data service level 1505 to deliver the appropriate data packets to the correct destination: the personal computer 10 for the C-data 15 packets or the voice control DSP circuit 306 for V-data In transmit mode, there are two data buffers, the V-data buffer 1511 and the C-data buffer 1513, implemented in the main controller RAM
316 and maintained by main controller 313. When the voice control DSP
20 circuit 306 engages voice operation, it will send a block of V-data every ms to the main controller circuit 313 through dual port RAM circuit 308.
Fach V-data block has one sign byte as a header and 23 bytes of V-data The sign byte header of the voice packet is transferred every frame from the voice control DSP to the controller 313. The sign byte header 25 contains the sign byte which identifies the contents of the voice packet.
The sign byte is defined as follows:
00 hex = the following V-data contains silent sound Ol hex = the following V-data contains speech inforn~ation If the main controller 313 is in uansmit mode for V-data/C-data 30 multiplexing, the main controller circuit 313 operates at the data service level to perform the following tests. When the voice control DSP circuit 306 starts to send the 23-byte V-data packet through the dual port RAM to the main corrtroDa circuit 313, the main controller will check the V-data buffer to see if the buffer has room for 23 bytes. If these is su~cient room in the V-data buffer, the main controller will check the sign byte in the header preceding the V-data packet, If the sign byte is equal to one (indicating voice information in the packet), the main controller circuit 313 will put the following 23 bytes of V-data into the V-data buffer and clear the silence counter to zero. Then the main controller 313 sets a flag to request that the V-data be sent by the main controller at the multiplex control level.
If the sign byte is equal to zero (indicating silence in the V
data packet), the main controller circuit 313 will ina~ease the silence counter by 1 and check if the silence counter has reached 5. When the silence counter reaches 5, the main controller circuit 313 will not put the following 23 bytes of V-data into the V-data buffer and will stop increasing the silence counter. By this method, the main controller circuit 313 operating at the service level will only provide non-silence V-data to the multiplex control level, while discarding silence V-data packets and preventing the V-data buffer from being ov~vvcitten.
The operation of the main controller circuit 313 in the multiplex control level is to multiplex the V-data and C-data packets and transmit them through the same channel. At this control level, both types of data packets are transmitted by the HDLC protocol in which data is transmitted in synchronous node and checked by CRC error checking. If a V-data packet is received at the remote end with a bad CRC, it is discarded since 100% acc~uacy of the voice channel is not ensw~d. If the V-data packets were re-sent in the event of com~ption, the real-time quality of the voice transmission would be lost. In addition, the C-data is transmitted following a modem data communication protocol such as CCITT V.42.
In order to identify the V-data block to assist the main controller circuit 313 to multiplex the packets for transmission at his level, and to assist the remote site in recognizing and de-multiplexing the data packets, a V-data block is defined which includes a maximum of five V-data packets. 'Ihe V-data block size and the maximum number of blocks are defined as follows:
The V-data block header = 80h;
The V-data block size = 23;
The maximum V-data block size = 5;
5 The V-data block has higher priority to be transmitted than C-data to the integrity of the real-time voice transmission. Therefore, the main co~oller circuit 313 will check the V-data buffer first to determine whether it will transmit V-data or C-data blocks. If V-data buffer has V-data of more than 69 bytes, a transmit block counter is set to 5 and the main 10 controller circuit 313 starts to transmit V-data from the V-data buffer through the. data pump circuit 311 onto the telephone line. Since the transmit block counter indicates 5 blocks of V-data will be transmitted in a continuous stream, the transmission will stop either at finish the 115 bytes of V-data or if the V-data buffer is empty. If V-data buffer has V-data with number more 15 than 23 bytes, the transmit block counter is set 1 and starts transmit V-data.
This means that the main controller circuit will only transmit one block of V-data If the V-data buffer has V-data with less than 23 bytes, the main controller circuit services the transmission of C-data During the transmission of a C-data block, the V-data buffer 20 condition is checked before transmitting the first C-data byte. If the V-data buffer contains more than one V-data packet, the c~ux~ent transmission of the C-data block will be terminated in order to handle the V-data B~iY~
On the receiving end of the telephone line, the main controller 25 circuit 313 operates at the multiplex control level to de-multiplex received data to V-data and C-data. The type of block can be identified by checking the first byte of the incoming data blocks. Before receiving a block of V
data, the main cor~nller circuit 313 will initialize a receive V-data byte counter, a backup pointer and a temporary V-data buffer pointer. The value 30 of the receiver V-data byte counts is 23, the value of the receive block counter is 0 and the backup pointer is set to the same value as the V-data receive buffer pointer. If the received byte is not equal to 80 hex (80h indicating a V-data packet), the receive operation will follow the current modem protocol since the data block must contain C-data. If the received byte is equal to 80h, the main controller circuit 313 operating in receive mode will process the V~ara.
S For a V-data block received, when a byte of V-data is received, the byte of V-data is put into the V-data receive buffer, the terr~pora~y buffer pointer is in~ased by 1 and the receive V-data counter is dec~sed by 1. If the V-data counter is down to zero, the value of the temporary V-data buffer pointer is copied into the backup pointer buffer. The value of the total V-data counter is added with 23 and the receive V-data counter is reset to 23. The value of the receive block counter is increased by 1. A flag to request service of V-data is then set. If the receive block counter has reached 5, the main controller circuit 313 will not put the incoming V-data into the V-data receive buffer but throw it away. If the total V-data counter has reached its maximum value, the receiver will not put the incoming V-data into the V-data receive buffer but throw it away.
At the end of the block which is indicated by receipt of the CRC check bytes, the main controller circuit 313 operating in the multiplex control level will not check the result of the CRC but instead will check the value of the receive V-data counter. If the value is zero, the check is finished, otherwise the value of the backup pointer is copied back into the current V-data buffer pointer. By this method, the receiver is insured to de-multiplex the V-data from the receiving channel 23 bytes at a time. The main controller circuit 313 operating at the service level in the receive mode will monitor the flag of request service of V-data. If the flag is set, the main controller circuit 313 will get the V-data from the V-data buffer and transmit it to the voice control DSP circuit 306 at a rate of 23 bytes at a time. After sending a block of V-data, it decreases 23 from the value in the total V-data counter.
Negotiation of Voice Compression Ra_rP
The modem hardware component 20 incorporates a modified packet protocol for negotiation of the speech compression rate. A modified supervisory packet is forn~atted using the same open flag, address, CRC, and closing flag fonnatxing bytes which are found in the CCITT V.42 standard data supervisory packet, as is well known in the industry and as is described in the CCITT Blue Book, volume VIII entitled Data Comm~mication over, 5 T~l~ph~le Network 1989 referenced above. In the modified packet protocol embodiment, the set of CCITT standard header bytes (control words) has been extended to include nonstandard control words used to signal transmission of a nonstandard communication command The use of a nonstandard corrrrol word should cause no problems with other data communication terminals, f~
10 example, when communicating with a non-PCS modem system, since the nonstandard packet will be ignored by a non-PCS system Table 21 offers one embodiment of the present invention showing a modified supervisory packet sttucriu~e. It should be noted that Table 21 does not depict the CCITT standard formatting bytes: open flag 15 address, CRC, and closing flag, but such bytes are inherent to using the CCITT standard. The modified supervisory packet is distinguished from a V.42 standard packet by using a nonstandard control word, such as 80 hex, as the header.

TA>3~~ 21: Modified 5 Supervisory Packet Structure ~; 80h~i ID ! LI ;'' ACK ~ data ~ data -_ - data , The modified supervisory packet is transmitted by the HI7LC
protocol in which data is transmitted in synchronous mode and checked by CRC error checking. The use of a modified supervisory packet eliminates the need for an escape cornrnand. sent over the telephone line to intearupt data 15 communications, providing an independent channel for negotiation of the compression rate.. The channel may also be used as an alternative means for prograrrzrning standard communications parameters.
The modified supervisory packet is encoded with different function codes to provide an independent communications channel between 20 hardware components. 'Ibis provides a means for real time negotiation and programming of the voice compression rate throughout the transmission of voice data and conventional data without the need for conventional escape routines. The modified supervisory packet is encoded with a function code using several methods. For example, in one embodiment, the function code is 25 embedded in the packet as one of the data words and is located in a predetermined position. In an alternate embodiment, the supervisory packet header itself serves to indicate not only that the packet is a nonstandard supervisory packet but also the compmssion rate to be used between the sites.
In such an embodiment, for example, a different nonreserved header is 30 assigned to each function code. These embodiments are not limiting and other methods known to those skilled in the art may be employed to encode the function code into the modified supervisory packet.
Referring once again to Figure 1, a system consisting of PCS
modem 20 and data terminal 10 are connected via phone line 30 to a second 35 PCS system comprised of PCS modem 20A and data terminal l0A
Therefore, calling modem 20 initializes communication with receiving modem 20A In one embodiment of the present invention, a speech compression corrurrend is sent via a modified supervisory data packet as the request for speech compression algorithm and ratio negotiation. Encoded in the speech compression comrnand is the particular speech compression algorithm and the speech compression ratio desired by the calling PCM modem 20. Several methods for encoding the speech compression algorithm and compression ratio exist. For example, in embodiments where the fimction is embedded in the header byte, the first data byte of the modified supervisory packet could be used to identify the speech compression algorithm using a binary coding . scheme (e.g., OOh for Vector Quantization, Olh for CELP+, 02h for VCELP, and 03h for TrueSpaech, etc.). A second data byte could be used to encode the speech compression ratio (e.g., OOh for 9.5 ICHz, Olh for 16 KHz, 02h for 8KI-1~ etc.). This embodiment of the speech compression comrnand supervisory packet is shown in Table 22.
TABLE 22:
Speech Compression Command Supervisory Packet 80h;,' ID LI AQC ~,Algthmn CRatio~ - .- data Alternatively, as stated above, the fimction code could be stored in a pc~edetem~ined position of one of the packet data bytes. It should be apparent that other fimction code encoding methods could be used. Again, these methods are given only for illustrative purposes and not limiting.
In either case, the receiving PCS modem 20A will recognize the speech compression command and will respond with an acknowledge packet using, for instance, a header byte such as hex 81. The acknowledge packet will alert the calling modem 20 that the speech comp~sion algorithm and speech compression ratio selected are available by use of the ACK field of the supervisory packet shown in Table 22. Receipt of the acknowledge supervisory packet will cause the calling modem 20 to transmit all voice over data information according to the selected speech compression algorithm and compression ratio.
The fiequency of which the speech compression command supervisory packet is transrnitted will vary with the application. For moderate quality voice over data applications, the speech compression algorithm nerd only be negotiated at the initialization of the phone call. For appliartions requiring more fidelity, the speech compc~sion command supervisory packet 5 may be renegotiated during the call to accommodate new patties to the communication which have different speech compression algorithm limitations or to actively tune the speech compression ratio as the quality of the communications link fluctuates.
Therefore, those skilled in the art will recognize that other 10 applications of the speech compression command supervisory packet may be employed which allow for varying transmission rates of the speech compression command supervisory packet and different and more elegant methods of speech compression algorithm and compression ratio negotiation, depending on the available hardware and particular application. Additionally, 15 a number of encoding m~hods can be used to encode the supervisory packet speech compression algorithm and the speech compression ratio, and this m~hod was demonstrated solely for illustrative purposes and is not limiting.
Of course a new supervisory packet may be allocated for use as a means for negotiating multiplexing scheme for the various types of 20 information sent over the communications link. For example, if voice over data mode is employed, there exist several methods for multiplexing the voice and digital data The multiplexing scheme may be selecxed by using a modified supervisory packet, called a multiplex supervisory packet, to negotiate the selection of multiplexing scheme.
25 Similarly, another supervisory packet could be designated for remote control of another hardware device. For example, to control the baud rate or data forn>at of a remote modem, a remote control supervisory packet could be encoded with the necessary selection parameters needed to program the remote device.
30 Those skilled in the art will readily appreciate that there exist numerous other unidirectional and bidinxaional communication and con~ol applications in which the supervisory packet may be used The examples given are not limiting, but are specific embodiments of the present invention offered for illustrative purposes.
Ca_Iler ID Lnterfar$ LL~m Figure 16 shows one embodiment of the present invention in 5 which a personal communications system (PCS) 1600 is the interface between a standard telephone line service 1610 and a computer system 1620 using telephone lines 1630. Caller 1D interface ("CID interface") 1650 provides caller lD fimctionality to personal communications system 1600.
Figure 17A shows one embodiment of the peisonat 10 communications system 1600 with caller ID interface 1650 as shown in Figure 16. Caller ID interface 1650 includes ring detector 1710, off hook circuit 1720, DC holding circuit 1730, caller 1D relays 1740, caller ID decoder 1750, processor 1770, and memory 1780. In this embodiment processor 1770 is the personal communications system processor and memory 1780 is a portion of 15 memory in the personal commimicatians system Alternate embodiments may employ separate processors and memory for the interface without departing from the scope and spirit of the present inverrtion.
Ring detector 1710 si8nals processor 1770 on signal line 1772 when an incoming call is received on telephone lines 1702 and 1704.
20 Processor 1770 signals caller ID relays 1740 on signal line 1774 to decode the caller ID information as transmitters by the telephone company betvveen the first and second telephone rings. Caller 1D relays 1740 route signals on telephone lines 1702 and 1704 to caller ID decoder 1750 via isolation device 1782 when signal 1774 is pulled to a logic "0" state. When signal 1774 is 25 logic low, transistor 1745 conducts and normally op~ optoisolated relays 1742 and 1743 close briefly while normally closed optoisolated relay 1744 opens. 'Ihe switching period only needs to be long enough to receive the frequency shift keying caller ID transmissions between the first and serorxl telephone rings. Rectifier 1794 ensures that the telephone line polarity to the 30 do holding circuit 1730 is consistent regardless of the polarity of the telephone service connections to lines 1702 and 1704.
An access matrix is preprogrammed into the interface memory 17$0 which specifies the parame~s associated with an authorized caller (Further details on the access matrix and screening modes are discussed below.) If the incoming caller ID information and the access matrix pararnetas indicate that the caller is authorized, then processor 1770 answers 5 the telephone call by asserting a logic "0" on iine 1776 and engaging the telephone caarent loop using signal 1 T77 to activate DC holding circuit 1730.
When these circuits are activated caller ID relays 1740 are programmed to pass telephone signals 1732 and 1734 through caller ID decoder 1750 to personal commrurications system electronics 1760 for demodulation and data 10 processing.
If the caller is not authorized access, the caller ID interface 1650 can hang up on the caller by momentarily taking the personal communications system off hook and returning to on-hook by toggling signal line 1776. In this way, the caller ID interface can hang up on an unwanted 15 caller without providing access to the caller.
Referring now to Figures 3, 7B, and 7C, in this embodiment of the present invention telephone line interface 309 includes the caller ID
interface 1650 hardware as shown in Figure 17. Therefore, personal communications system electronics 1760, processor 1770 and memory 1780 of 20 Figure 17 is amount to the hwdvvare shown in Figure 3 excluding telephone line interface 309. In this embodiment:
ring detector 1710 is optical isolator U32 (CNY17) and zener diodes CRl and CR2 of Figure 7C;
caller 117 relays 1742, 1743, and 1744 are relays U30, U31, (of Figure 25 7C) and U33 (of Figure 7B), respectively;
caller ID decoder 1750 is chip set U34, U35, and U37 of Figures 8A
and 8B (U34 is the interface chip 315, U35 is the Code~c 311, and U37 is the DSP/Data Pump 311 as shown in Figure 3);
do holding circuit 1730 is CR19, R73, C71, CR20, CR26, R74, R75, 30 R76, and transistors Q2 and Q3;
off hook circuit 1720 is optoisolated relay U29, transistor Q4, resistor R15, and resistor R69 of Figure 7B;

processor 1770 is main c~noller 313, which is a Zilog 280180 microprocessor, part number Z84C01 by Zilog, Inc. of Campbell, California;
and memory 1780 is the combination of RAM 316 and PEPROM 317.
5 Isolation device 1782 electrically isolates the caller ID interface from the PCS
elecr<onics 1760. Isolati~ device 1782 is typically a transformer, however, alternate embodiments employ and optocoupler device. The detailed operation of this circuitry is discussed above in the section entitled: "Detailed Electrical Schematic Diagrams".
10 An alternative embodiment of the calls ID interface 1652 is described in Figure 17B. The operation of caller ID interface 1652, shown in Figure 17B, is similar to the caller 117 interface 1650 of Figure 17A, however, the caller ID interface 1652 incorporates a single relay 1790 to activate do holding circuit 1730 for pLUposes of answering the telephone call. The on-15 hook condition of caller 1D interface 1652 is characterized by off hook relay 1720 (normally open) being open and relay 1790 being op~. The caller >D
information from telephone lines 1702 and 1704 is decoded after the first ring by closing off hook relay 1720 to pass the ficy shift keying caller ID
signals through capacitor 1792 to caller 1D decoder 1750. Therefore capacitor 20 1792 serves as a do blocking element to create an ac path to caller ID
decoder 1750. Closing off hook relay 1720 connects the ac loop for frequency shift keying demodulation and decoding and the call is not ansvvered as long as relay 1790 remains open. If the caller ID information indicates an authorized caller, then closing relay 1790 creates the off hook condition for connecting 25 the caller to the peaonal communications system 1600. If the caller ID
information in conjunction with the access matrix indicates that the caller is unauthorized a quick hang up can be accomplished by briefly toggling relay 1790 and off hook relay 1720 to answer the call and then hang up.
In yet another embodiment, caller )D decoder 1750 is the Nfitel 30 MT8841 Calling Number Identification Circuit as specified in the Nfitel Ivhcroelectronics Digital/Analog Communications Handbook, Doc. No. 9161-952-007-NA, issue 9 (1993). Processor 1770 is the Zilog 2182 Nficropcncessor as specified in the 2180 Family Ivfiaoprocessors and Peripherals Databook, Doc No. Q2/94 DC 8322-00.
Those skilled in the art will readily recognize that other caller ID decoders and processors may be used without departing from the scope 5 and spirit of the present invention. Alternate embodiments use sophisticated, multifiuxtion decoding devices and data pumps to perform the functions of caller ID decoder 1750. Additionally, processor 1770 may be replaced with combinational logic to control the operation of the caller ID interface.
Finally, alternate relay switching embodiments may be constructed which do 10 not depart from the scope and spirit of the present invention.
Caller ID Encoded 'Transit In standard telephone caller ID systems the caller ID
infbm~aiion is transmitted between the first and second telephone ring. The caller )D information includes a message-type byte, a length byte, and data 15 bytes consisting of date, time, telephone number with area code, telephone owner's name, and check sum byte and is sent using frequency shift keying between the first and second ring. Several industry protocols for caller ID
are being developed by telecommunications vendors, including.
1. Bellcore's single data message frame format and multiple data 20 message frame format as described in Bellcore Technical Reference TR-NWT-000030, Issue 2, October 1992;
2. Rockwell's Calling Number Delivery (CND) as described in Rockwell Application Note, Docmnent No. 29800N73, Order No. 873, October 1991; and 25 3. AT&Ts Caller m as described in the AT&T Ivficroelectronics Modem Designer's Guide, June 3, 1993, Doc. MN92-026DMOS.
Figure 18 summarizes the three caller ID protocols described above. The Bellcore multiple data message frame format 1810 is distinguished from the AT&T or Rockwell single data message frame format 1820 by examining the 30 leading bytes 1811 {OII~ and 1821 (04I-~.
Figure 19 shows an algorithm which distinguishes between the different protocols for proper decoding of the incoming caller ID information in one embodiment of the present invention. The caller ID information is mxrieved from the caller ID decoding hardware 1902 and the first byte is decoded to determine the message type 1904. If the first byte is O1H (1906), then the calls ID protocol is the multiple data message format 1810 of Figure S 18 ( 1910) and the message data is read after skipping the delimiters ( 1912, 1914). If the first byte is 04H (1908) then the caller ID information is in the single data message fon~nat 1820 of Figure 18 (1920) and the message data can be read directly (1922). If the first byte is neither O1H or 04I-1; then the caller ID information is being transmitted by another protocol or an error has 10 been made in the decoding or transmission 1930. Since the above mentioned protocols are the most widely used, the present algorithm provides for automatic switching between formats to ensure that both formats are properly read.
Those skilled in the art would readily recognize that this 15 method could be modified for accommodating furtu~e caller ID message protocols without departing from the scope and spirit of the present invention, and that the protocols presented were not intended in an exclusive or limiting 20 In one embodiment of the present invention the quick hang feature allows the personal communications system to hang up immediately on an unwanted caller by placing the personal communications system off hook for a duration of one second (using off hook cit~cuit 1720), and them place the personal communications system back "on hook" again. The 25 personal communications system is then ready to accept another call. This feature minimizes the amount of time an invalid user can tie up a phone line.
Other dutations are possible without departing finm the spirit and scope of the present invention.

T'irrte of Dav,~~chm iration c'ng alley 1D
In one embodiment of the present invention time synchronization cart be accomplished by using the decoded caller ID
information which identifies the time of day to synchronize an on-board time-5 of-day clock The information available from decoding caller ID information allows the caller B7 interface to screen users by a variety of parameters as specified in a preproaccess matrix. The flowchart shown in Figm~e 10 20 describes the overall operation of the screening function. The program access matrix is programmed by specifying the screening mode and specifying the operative parameters to perform the screening, such as caller name, caller telephone number, time and day frames for receiving calls, and number of accesses (2002). In one embodiment of the present invention the receiving 15 personal communications system opetates in the following modes or combination of modes which will be described further in the "Screening Modes Using Caller 1D" section below:
1. Number Only Mode;
2. Blacklist Mode;
20 3. Day Only Mode;
4. Time Only Mode;
5. Name Only Mode;
6. S Register 50 Mode; and 7. Hybrid Modes 25 The caller )D interface then detects a ring 2004, derndes the caller ID information 2006, and compares the caller ID information with the access matrix (a function of the screening mode, as described below) 2008, and determines whether the caller is authorized to access the persor>al communications system 2010. If the caller is not authorized an exit routine is 30 performed 2020 which may be a number of operations including, but not limited to, a friendly error message and a quick hang up 2022. If the personal communications system is compiling a database of callers, the database can be with the received caller ID information 2024 before waiting for another call 2004. If the caller is authorized, access is enabled 2014 and allowed until the call is terminated 2016. The hang up proced~me 2022 is followed by an database update 2024 before rerinning to the wait state for 5 another call 2004. The step of determining whether access is authorized 2010 is discussed in detail in the below section on Screening Mades.
Screening Modes c'ng Iler 1D
The following modes are used in one embodiment of the present invention to control access to a personal communications system 10 connected to the caller ID interface. The paran~ete~s of each mode become part of the preprogcarnmed access matrix. A number of examples will be offered following a brief description of the various modes of this embodiment:
Number W_n_ly Mode In the number only mode, the personal communications system 15 compares an incoming caller ID number to phone numbers on a "number only" list. Only incoming calls with numbers matching the phone numbers on the list will be ansvrered. The number list is part of the access matrix which is preprogrammed into the caller ID interface merr~ory.
~l~kli~~
20 In the blacklist mode, the personal communications system compares an incoming caller 1D number to a list of caller on a "blacklist."
Any call which matches a phone number on the "blacklist" will be denied access to the device and the incoming call will be terminated immediately using a preprogrammed exit routine, such as the "quick-hang" feature 25 described above. The blacklist and desired exit routine can be tailored depending on the particular blacklisted caller. For example, a BBS might want to quick hang up on a blacklisted abuser of the bulletin board, but only give a "late dues" message to a blacklisted user who is merely late in paying dues. The access matrix contains all of the blacklist parameter.
30 I2av~, In the day only mode, the access matrix is programmed to authorize calls only on specific days.

In the time only mode, the personal communications system only answers calls during a certain preprogrammed times of the day and ignores calls outside of those specified times. For example, this feature 5 eni~ances the secva~ity to a computer network provided by the pn~ent invartion during non-business hours.
jyame On , The name only mode authorizes access only to callers whose names are preprogratnn~d in a name table in the access matrix. This is a . 10 means for inclusively authorizing access to the personal communications system ('Ihe blacklist mode is an exclusive means for authorizing access to the personal communications system.) S Register 50 Mode The S Register 50 mode provides a limited number of accesses 15 by a particular user. The scars matrix is preprowith a predetermined number of calls allowed to a user before that user is black listed This feature is especially useful for electronic bulletin board service operators because it allows them to screen out unwanted users as soon as the S Register number is reached The S Register mode also allows for limiting 20 the number of accesses made by a crew user of the bulletin board, since in one embodiment of the present invention a new user can be assigned a position in the access matrix and subject to a maximum numbs of accesses, similar to the known users. 'Ibis is known as a Temporary Blacklist mode, since after the predetermined number of the user is temporarily blacklisted until 25 the access counter for that user is resex by the system administrator.
Any combination of the above modes provides a specialized access matrix based on each listed user. For example, access authority can be given to Mr. X at only 6:00 to 7:00 p.m. by programming Mr. xs name and 30 the access times as illustrated in the examples below.
Several access matrix examples for a BBS and their associated interpretations are described below for each entry in the table, according to one embodirrtent of the present invention.

~.I~E$ I~' AI~ DAY/ TIME ~5Q
(b) 333-3333 quick hang (b) Mr. J "Illegal Access"
Table 23 shows two examples of the blacklist mode. Any calls from 333-3333 will receive a quick hang immediately on attempt to access the BBS personal communications systemG Additionally, any time Mr.
J attempts to call (from any of his phones), the message "Illegal Access" will be displayed prior to hang up by the BBS.

~ DAY/ ~ $~
Mr. Z
Ms. B
Table 24 shows that only Mr. and Ms. B can access the BBS
personal communications system, regardless of telephone number or day.

».~EB l~ I?AYL~~ ~.~Q EXL'L~?.l~

Table 25 shows that only callers from 123-4567 and 676-8888 can access the BBS personal communications system, regardless of name or time of day.

~Ev $ ~ DAY/ TIME ~ E

Table 26 shows that any caller between 9:00 a.m and 5:00 p.m can access the BBS personal communications system (24 hour time format used in this example).

I?AYI~ ~ E
333-3333 Mr. A M-W/Crl3, 14-15 5 quick hang 444-4444 Ms. B ALL 9-17 555-5555 M-F/9-17 10 "PAY DLJE,S"
Mr. C M W/ 3 Table 27 provides four examples of access matrix entries. In the first line, Mr A. can access the BBS personal communications system fibm Monday to Wednesday and at the times of 6:00 a.m to 1:00 p.m and 2:00 p.m. to 3:00 p.m Mr. A can only access the BBS personal communications system five (5) times before access is denied and the system administrator is notified. Mr. A will get a quick hang up on his sixth attempt to access the BBS personal communications system, and attempts thereafter, until his access register is reset by the system administrator.
Ms. B can access the BBS personal communications system all days of the week, but only between the hours of 9:00 am to 5:00 p.m Ms.
B can access the BBS personal communications system an unlimited number of times.
Any caller finm phone number 555-5555 can access the BBS
personal communications system from Monday to Friday betwe~ the hours of 9:00 a.m to 5:00 p.m A "Pay Dues" message will be displayed to the user before a hang up on the eleventh attempt to aaess the BBS personal communications system, and subsequent attempted accesses. The system administrator must reset the access counter for firture access authorization.
Mr. C can access the BBS personal communications systan finm any of his phone numbers, and can access at any time on Monday through Wednesday. After three accesses, Mr. C's exit routine will be whatever the default exit routine for the BBS happens to be.
The described screening modes are not limiting and not exclusive. One skilled in the art would readily recognize that other modes and variations of these modes are possible without departing from the scope and spirit of the present invention.
The previously described screening modes are not exclusive or limiting to the present invention. Neither is the particular interaction of the screening modes. The following is only one embodiment of a screening algorithm offered to demonstrate the interaction between screening modes in one embodiment of the present invention.
Figure 21 illustrates one embodiment of the authorization process using the above described screwing modes. The caller ID interface waits for a call 2102, and gets the caller ~ID information upon detecting an incoming call 2104, 2106. The caller ID information is verified against the access matrix, in this example the caller's telephone number is verified 2110, 2112. If the number is on the list, then the time of day is verified 2114 and the date is verified 2116 before the personal communications system is allowed to answer the call 2118. The personal communications system is engaged in the call as long as it is connected 2120 and the call is complete after the connection is lost 2122. The personal communications system then quick hangs up 2140 and disconnects 2142 before waiting for the next call 2102. If the time of day or date is invalid the quick hang procedure is automatically initiated. If the telephone number is not on the number list 2110 the user's number of accesses is checked to ensure that the maximum is not 2130 and if the number is not on the ternrporary blacklist 2132 it is added 2136 prior to answering the call 2118. If the number is on the blacklist 2132, a separate S Register 50 for the blacklist is checked 2134 and quick hang is initiated 2140, 2142 if the maximum number of accesses is exceeded, else the register for this caller is incremented 2138 and the call answet~d 2118, 2120, and 2122. The quick hang proced<u~e, 2140 and 2142, is again followed by waiting for the next call 2102.
CONCLUSION
The preset invention was described in terms of a personal communications system interface, however, the methods and apparahis are applicable to a number of data exchange devices. For example, the present imrenxion could be adapted to any system with caller identification infom~atian, including but not limited to applications in the fiber superhighway and similar applications.
Although specific embodiments have been illustrated and 5 described herein, it will be appreciated by those of ordinary skill in the art that any arrangement which is calculated to achieve the same purpose may be substituted for the specific embodiment shown. This application is intended to cover any adaptations or variations of the present inv~tion. Therefore, it is manifestly intended that this invention be limited only by the claims and 10 equivalents thenoof.

Claims (2)

1. A personal communications system interface, connected to a telephone line, for screening incoming telephone calls to personal communications system electronics, the internal personal communications system interface comprising:
a telephone input port for receiving telephone signals into the interface;
a ring detector, connected to the telephone input port, for detecting an incoming call;
an off-hook circuit, connected to the telephone input port, for connecting the personal communications system interface to the telephone line;
a dc holding circuit, connected to the off-hook circuit and the input port, for maintaining a connection with incoming telephone calls;
a decoder for decoding caller identification information and personal communications system data;
a multiplexer, connecting the decoder to the telephone input port and the dc holding circuit, for selecting telephone signals from the telephone input port for caller identification information decoding and from the dc holding circuit for personal communications system data decoding;
a controller, connected to the ring detector, off-hook circuit, dc holding circuit, multiplexer, and decorder, for controlling the internal personal communications system interface and for comparing caller identification to an access matrix for authorization purposes; and a memory device, connected to the controller, for storing the access matrix.

2. A method of screening an incoming call received by a personal communications system, said method comprising:
programming an active matrix in said personal communications system with data to specify a screening mode and operative parameters for said personal communications system;
receiving said incoming call at said personal communications system;
decoding caller ID information associated with said incoming call; and performing an analysis of said caller ID information against said data in said active matrix to determine whether or not said incoming call should have further access to said personal communications system.