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Description  |
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The present invention relates to telephone communication systems and
methods, being more particularly concerned with integrated audio and
multi-format data terminal communication between a pair of suitably
equipped telephone instruments, with the users conversing in normal
fashion while alpha-numeric data is exchanged as between machines, such as
computers, fax machines, displays as of the LED type, etc., all smoothly
and, if desired, with user transparency (i.e., without knowledge or
awareness of the users).
BACKGROUND
Present-day advances in telephony, while sometimes referred to as
"integrated systems", are actually "integrated" only from the point of
view of the telephone network and in the limited sense that digital and
analog signals are switched by much common equipment; but from the user's
point of view, there is little or no integration in that there is a
telephone either for voice communication, or for a data call or connecting
a computer in a particular way similar to ordinary modem connections. The
ISDN system (Integrated Services Digital Network) of AT&T, for example, as
described in AT&T Technical Journal, January-February, 1986, Vol. 65,
Issue 1, in conjunction with the AT&T Mode 7506 ISDN telephone handset,
selectively operates multiple phones, including with LCD display to show
the number being dialed and/or the number calling the telephone (so-called
ANI or automatic number identification) and, through an RS232 connector at
the back of the phone, enables the plugging in of one or more modems to
enable data transmission totally independent of the communication use of
the telephone. No simultaneity of voice communication and data exchange to
a single telephone number is possible, let alone with user transparency.
On any given phone call, the operation either is just exactly like a phone
having no computer or other data equipment whatsoever, or it is a computer
operation exactly like a modem having no audio phone conversation use
whatsoever. If the user does not possess a computer or a data terminal,
then the user has no means whatsoever to do anything beyond a normal phone
connection with this system. Thus, there is nothing "integrated" in these
systems from the user's point of view.
Underlying the present invention, however, is a very different philosophy
that achieves simultaneously both the function of conducting conversation
and effecting digital transfer of information (such as computer-computer
data exchange, FAX transmission, etc.). Currently, the mode of exchange
(audio, computer data, FAX data, etc.) is usually implicit in the
telephone number being dialed; some numbers are FAX numbers, for example,
and some computer numbers. By means of the present invention, however, a
number of these services are handled by a single telephone number;
automatically and without confusion, and with providing entirely new
services as well. Particularly interesting are those which combine audio
conversation with data transmission simultaneously.
To initiate such a service smoothly and to achieve automatic coordination
of extended telephone services sensible by both humans and machines, it is
essential that the equipment at both ends of the line and the conversing
users be informed of the fact that the telephones are equipped for
specialized services that are to be provided. This is achieved, in
accordance with the invention, by the use of a particular signal, called
the SMARTPHONE.TM. RECOGNITION SIGNAL (SRS) which serves as the basis for
much of this automatic recognition and mode selection. Briefly, an SRS is
a particular in-band, complex, time-varying unique or distinctive audio
signal, chosen with regard to a number of criteria later described. Once
such an SRS has been transmitted, recognized, and responded to, all
parties on the line (whether human user or machine) are aware that this is
as SRS-equipped call. They are then ready to exchange further signals, if
necessary, which can select a number of features and transmission modes as
desired. The use of a later-discussed Digital Signal Processor (DSP), such
as the Texas Instruments TMS320 series or Motorola DSP56000 series, as
described in the Motorola technical data bulletin DSP56001, 1988 (pages
1-60), is a convenient means to make effective use of the flexibility made
possible by the invention.
OBJECTS OF INVENTION
An object of the invention, accordingly, is to provide a new and improved
method of and apparatus for integrated voice (audio) communication
simultaneously with "under voice" user-transparent digital data exchange
between telephone instruments. By the term "under voice", as used herein,
it is intended generically to embrace the concept of absence of noticeable
interference with the voice communication, including, but not limited to,
the use for data transmission of frequency bands below or above the voice
channel and signal compression techniques, and alternatively by converting
the voice to digital form as by a vocoder-type system and integrating with
digital data.
Other and further objects will be explained hereinafter and are more
particularly defined in the appended claims.
SUMMARY
In summary, however, from one of its viewpoints, the invention embraces a
method of voice and simultaneous user-transparent digital data
communication between a pair of telephones connectable through an ordinary
telephone exchange, that comprises, providing each telephone with special
cooperative digital processing and visual digital display capability;
programming the processing (1) to transmit a distinctive audio recognition
signal from the receiving telephone when connected to the calling
telephone and from the calling telephone back to the receiving telephone
to indicate to the users that both phones are specially equipped for
cooperation, and (2) to transmit such recognition signal between the
calling and receiving telephone digital processing equipments; causing the
visual display of the calling telephone to display digital data
originating from the programming in the receiving telephone processing and
digitally transmitted "under voice" therefrom in a manner and protocol
transparent or imperceptible to the telephone users and during their
continued voice conversation. The invention also provides substantial
advantage with the before-mentioned ISDN type telephone exchange system as
will hereinafter be explained. Preferred and best mode embodiments and
details are later presented.
DRAWINGS
The invention will now be described with reference to the accompanying
drawings,
FIG. 1 of which is a block and circuit diagram of a preferred embodiment
which may provide all these features in a form factor compatible from both
physical and electrical points of view with an ordinary conventional
telephone;
FIG. 2 is a similar diagram illustrating details of the digital processing
systems incorporated into the phones of FIG. 1; and
FIG. 3 is a block diagram showing the incorporation of the invention in an
ISDN type telephone exchange.
DESCRIPTION
Referring to the drawing, a pair of telephones equipped in accordance with
the invention and conventionally connected by a telephone exchange is
shown in FiG. 1 at SP.sub.1, and SP.sub.2. Each instrument terminal is
provided with a means 2 FIG. 2, for generating the SRS signal and a filter
means 4 for recognizing the same, both to produce a distinctive audible
alerting signal to the user and to enable the digital data exchange
apparatus associated with the terminal to recognize the same. Criteria
involved in choosing an appropriate SRS include at least the following;
(1) Pleasing and distinctive sound to humans;
(2) Ease of detection by machine (typically a DSP);
(3) Minimal probability of confusion with signals occurring during ordinary
speech or other modes of operation;
(4) Reasonable duration; long enough to be easily distinguishable but not
so long as to cause excessive line occupancy;
(5) Minimal probability of the SRS itself being confused by ordinary
equipment with signals present early in the call in any conventional
operational mode.
For example, if an ordinary FAX call is made with such an SRS telephone
apparatus, it is undesirable that an SRS signal be interpreted by the FAX
modem as a FAX modem answer signal. The exact considerations here depend
on conventions adopted as to which party (calling, called, or both)
transmits SRS signals, what are considered to be triggering events (line
polarity reversal, etc.), and what time delays are employed. Some
flexibility in these regards may be built into the equipment, so as to
enable operation with whatever conventional equipment might be
encountered.
The timeline of a typical call using SRS equipment at both ends might look
like this:
(1) The user at SP.sub.1 dials the number at keyboard 10, FIG. 1, in the
usual fashion and connection is made through the "PHONE EXCHANGE" to the
receiving phone SP.sub.2 ;
(2) The called phone SP.sub.2 answers and immediately sends an SRS by the
generator 2 in its DSP chip 6, FIG. 2, along the "SIGNAL TO PHONE LINE"
path 20, ultimately back to calling phone SP.sub.1 as later explained;
(3) The calling phone SP.sub.1 detects the SRS at later-referenced
recognition module 4, FIG. 2, and replies with an SRS reply from its SRS
signal generator 2 in its DSP chip 6. The SRS reply may be a signal
distinct from the original SRS, although this is not absolutely necessary.
(4) The called phone SP.sub.2 recognizes the SRS reply in its recognition
module 4, FIG. 2.
At this point, the equipment at both ends is aware that appropriately
equipped SRS terminals are in use and that they are ready to exchange
further signals according to a particular protocol which corresponds with
the particular SRS which has been exchanged. Furthermore, any humans
listening in on the line have heard the distinctive SRS signal and are
aware of this situation. This has been done in a fully compatible manner,
in that no special directory number or other action was required. This
scheme does not interfere with the provision of ordinary services by
ordinary equipment without change (provided the SRS has been chosen
properly). Developments from this point depend on the protocol adopted.
Typical services which might be selected include:
Transmission of alpha-numeric data simultaneously with ordinary
conversation. This data might appear on an LCD display 8, FIG. 1, so as to
be visible to the user while the conversation takes place. This would be
particularly valuable in the case of travel reservations, appointments,
etc..
Further exchange of configuration information can take place, in either a
machine-to-machine or machine-to-machine-and-human mode. Useful
information to exchange might include: the existence and size of LCD
displays; available computer connections and baud rates; transmission
standards supported; existence of FAX capability; ability to support
demultiplexing protocols; availability of specific devices such as
printers and authentication modules.
Each of the telephone terminals SP.sub.1, SP.sub.2, etc. that is to
communicate in accordance with the invention is provided with special
digital signal processing equipment (DSP), above-mentioned, and display
equipment, such as LCD display(s), and optionally with auxiliary computer,
FAX, message recording, printing and other facilities, the existence of
which will be communicated between the user phone terminals as available
for use.
In accordance with this invention, a user can dial all the same numbers
normally dialled and in the normal way. The user doesn't have to push any
additional buttons or even be aware of the existence of this device or
treat it in any way any differently than the user of an ordinary voice
communications telephone. However, if it so happens that the number
dialled is a facility that is also fitted out with a "smart" phone, then a
number of additional features may be triggered and these are triggered in
a smooth, transparent way without the user being aware of the same, though
another important feature of the system is that the triggering may also,
if desired, be made apparent to the user.
One of the interesting features residing in the provision of an LCD display
on the SBS-equipped phone is that, with the appropriate equipment at the
far end of the connection, data can then be transferred back to the user
to show up on this LCD display, rather than having to pass that
information over the audio channel at all. And the user may, while this is
going on, continue to talk in the normal way of using the telephone
without intrusion ("under voice"). Conceptually, within this phone box, a
number of modules are provided to effect the above-described
functions--some of them being the ordinary phone analog such as, for
example, the headset H, FIG. 1, and the line conditioning electronics. As
previously mentioned, a digital signalling processing module is employed
having within it the digital signal processing chip 6 itself which as
previously stated may be a Motorola 56001 with ordinary EIA serial ports
connected with an RS 232 type of interface, so-labelled. Also within the
phone, earlier described LCD display 8 is provided, interfaced with the
DSP chip 6. Alternatively, the implementation of this type of control may
be effected with a separate computer on the order of an 8751 Intel single
chip computer supplemental to the DSP chip 6, dependent upon how much
memory is needed, etc.--such serving as the hardware that would be
required to implement the various data exchange and reproduction
functions. The conventional touch tone dial, hook switch and other
conventional parts of an ordinary phone SP.sub.1, of course, interface to
the DSP chip 6 as later more fully discussed. The digital signal
processing chip 6 implements the SRS recognition filter 4 in software,
FIG. 2, with the incoming data in digitized form being applied to this
filter continuously at a certain sample rate of some thousands of times
per second. The output of the filter 4 is zero most of the time; but if
this particular SRS signal comes along, then it is recognized and the
output becomes "one". The digital processing chip 6 also implements the
SRS generator--again through software. Basically when the logic of the
program dictates that it is time to apply this signal, it adds the SRS
signal to the outgoing digital signal which is then converted to analog by
the upper "codec" 9 (coder-decoder), FIG. 1. The codec may be of the
serial 13 - bit linear type (A/D & D/A--analog-to-digital and
digital-to-analog) such as the Motorola MC145402 described in its Sept.
23, 1987, technical data bulletin.
Typically there are several design choices possible. Let it be assumed, for
example, that it is desired to digitize the headset H as by using the
lower codec 11, FIG. 1, designed to perform this function and to interface
in a very clean way to the digital processing chip 6. From the point of
view of the program, inside the digital processing chip 6, there are thus
two audio sources and two sinks. One source is the signal from the
microphone M of the handset at 22 in FIG. 1, following the path through
codec 11 into the later-referenced high speed serial port HSP of the DSP
chip 6, shown at the "SIGNAL FROM MICROPHONE" line M' in FIG. 2, feeding
the upper talking path unit 13. The other source is the incoming signal
from the line L.sub.1, FIG. 1, through line buffer 5 and codec 9 into the
HSP port of the DSP chip 6 at the point represented at "SIGNAL FROM PHONE
LINE" 24 in FIG. 2. The sinks are the earphone of the headset H and the
outgoing signal on the line. The latter originates at the upper line 20
from the lower talking path unit 13 and the SRS signal generator 2 at the
"SIGNAL TO PHONE LINE" of the DSP chip 6, FIG. 2, applied to the codec 9,
FIG. 1, and through line buffer 5 to the line L.sub.1. The signal
originating at the lower talking path 13 along the lower line H' therefrom
at the "SIGNAL TO EARPHONE" region, is fed out from the HSP port of the
DSP chip 6 through codec 11 of FIG. 1 to the earphones H of the headset.
The port 26 of the DSP chip 6 is the RS232 port, connected through a line
amplifier buffer chip 3, precisely in the same fashion as an ordinary
analog modem for connecting to a computer, although extended services may
also be provided through this port as later described. In connection with
the before-described DSP chip very high speed serial path HSP, this is
designed to interface to a number of kinds of chips including, for
example, the before-mentioned codecs, connecting to both the input and the
output in a high speed serial path, with the codecs providing an A-to-D
and a D-to-A conversion.
The lower codec 11 is thus shown connected to both the headset H and the
microphone M. The upper codec 9 is connected through buffer 5 to the phone
line L.sub.1 from the telephone exchange. Through the port L.sub.2, for
example, a conventional analog FAX machine or answering machine may be
connected in a smooth fashion; that is, when appropriate signals are
recognized in the DSP chip 6, such activates the by-pass switch 12, FIG.
1, to allow the signal in the line L.sub.1 to connect directly to the port
L.sub.2, by-passing the system. The system described herein, therefore, is
sufficient for providing all the different features before discussed with
proper programming and without the necessity for major additional
hardware.
The program consists of two portions. The first is signal processing
running repeatedly in time at a particular digital sample rate consistent
with the Nyquist theorem, typically 8000 times per second (executions).
This signal processing module has the two basic inputs of the phone and
the microphone and two outputs which are basically the phone line and
earphone as previously traced. Within the signal processing module there
are a number of different program segments that are being
executed--different ones at different times, including the
before-mentioned simulating of the operation of analog filters.
Classically analog filters are made up with capacitors and inductors and
resistors and they can be configured in a number of forms. With the
digital signal processing module 6, the effect of different filters can be
simulated in a dynamic kind of arrangement. Basically, there is the main
signal flow in order that conversation can occur on the phone; namely,
reading the inputs applied through input buffer 5 from phone line L.sub.1,
applying filters to the input signals, mixing the signal coming from the
microphone M with some appropriate portion of side tone. The talking path
maintenance module 13, FiG. 2, supports the conversation function by
appropriately interconnecting the line L.sub.1 and the headset H as
earlier traced. Also in the audio rate loop of the DSP chip 6, FIG. 2, is
the modem recognition module 14 which automatically recognizes to which
international standard an applied modem signal conforms. Once recognized,
modem implementation modules 15 exchange data in conformance with
appropriate international standards as later described. The talking path
maintenance module 13 can operate either in a fully conventional mode or
with the data "under-voice" mode where the data signal is filtered out.
Such modules would implement international standards such as V.22 (1200
baud bidirectionally), V.22 bis (2400 baud), V.29 (9600 baud as for FAX
machines), and V.32 (9600 baud bidirectionally as for interactive
computers) as described, for example, in International Telecommunication
Union Red Book "Data Communication Over the Telephone Network", Geneva
1945, pages 64-93, 203-214 and 221-238.
FIG. 2, as previously noted, illustrates the SRS recognition module 4 which
consists of an elaborate set of recognition and conditioning filters
chosen to match the selected SRS signal and discriminated with a high
degree of reliability from other possible signals such as voice, modem
signals, etc.
There is also the control aspect of the system in addition to the signal
processing aspect and there exists an implementation choice as to whether
to implement the control section in a completely separate single chip
computer, such as an Intel 8751, well-designed to do this, or to avoid
that separate chip altogether and implement it by means of interrupt
routines that would be running on the DSP chip 6 itself.
The configuration set-up function allows the user to specify what mode of
operation is desired and what external equipment is connected. The LCD
display function selects data to appear upon the LCD 8. For example, as
one dials the number, it would appear on the LCD for the user to see; and
if nothing else is going on, it might just as well display the date and
time. If in the middle of a call it is recognized that the user is on the
SRS-equipped phone, any message which has been received from the far end
can be displayed on the display.
The RS 232 section, FIG. 1, interfacing through the line buffer 3 and
supplemented by the G-Bus controller lines 13' can operate in several
modes, depending upon the selected configuration. The G-Bus consists of
the RS232 interface plus additional controller lines 13' that can connect
with a wide variety of external devices (printer, authentication module,
multiple computer interfaces and interfaces to consumer devices) which
proceed to function smoothly because of the provisions for automatically
notifying the remote party of their existence and characteristics. The
simplest is to serve as a conventional computer modem at some particular
baud rate.
The SRS signal generator 2, FIG. 2 when activated by the control logic
plays out onto the line L.sub.1 an SRS signal of appropriate form, via
line 20, as before described. Each such SRS signal defines a time zero
situation for the initiation of a particular protocol, underlying the
smooth operation at the heart of the present invention.
The first step of a call, of course, is going to be the user pick up of the
phone and the switch hook activation at 16, FIG. 1. The switch hook is
shown interfaced by means of an I/0 pin A at the DSP module 6, and by
means of conventional operation, the call is processed through the stages
of obtaining dial tone and dialing the desired number into the central
exchange, as is well known.
The user thus dials a number, the remote phone exchange does its normal
operation and at some point it completes the call. When the called party
at SP.sub.2 lifts the switch hook at that phone, the SRS generator 2 at
that phone is activated and sends an SRS distinctive "bong" to the calling
phone SP.sub.1. This is detected by the SRS recognition filter 4 at the
phone SP.sub.1 which replies with its SRS reply "bong" via its SRS signal
generator 2. This, in turn, is detected by the SRS recognition circuit 4
at phone SP.sub.2. Since these two phones are now aware of the fact that
they are both SRS-equipped phones, etc., operation may proceed in a data
"under voice" mode. Thus, the users may carry on their conversation, while
meanwhile the exchange of data signals under the control of the respective
DSP chips occurs, transparent or imperceptible to the users. In
particular, SRS phone protocol may be initiated by which the phones advise
each other of their available respective configurations, display sizes,
printers and other external devices and other capabilities.
As an example, a patient calls the doctor's office using the usual
telephone number and procedure. Immediately as the phones are answered,
both the receptionist and the calling patient are advised of the existence
of SRS services by the SRS "bong" and reply "bong". Once the patient's
appointment is arranged and the date and time thereof entered by the
receptionist on her computer, this would automatically appear on the
patient's LCD display 8. Moreover, the receptionist's computer would be
automatically advised if a printer is available on the patient's SRS
telephone, and, if so, will print out the appropriate appointment slip
automatically on the patient's printer. This system can interface
naturally with an automated calendar system that the patient might
maintain, and which can enable later cancellation or modification without
human intervention. The invention thus introduces the possibility for
smooth multi-party automatic calendar updating or similar functions.
The data "under voice" mode of operation represents a base mode from which
transitions into any of several pure digital modes are possible. Such
digital modes allow data to be sent much faster--typically 9600 or more
baud, by temporarily interrupting the voice path. While during such
transmission, the analog voice signal may not be available, it may be
restored when the transmission is complete.
The invention, in all events, provides for automatic configuring and well
beyond what has been done before, with a cornerstone of the SRS signal
concept and appropriate protocols and configurations. Ultimately, the
invention provides the user with a multi-function device that provides
some functionality which has been conventionally available, such as a
modem, etc., and also some functionality which is new, such as allowing an
LCD to be updated for the user at the same time as the call is in
progress. Based on this, the invention has opened the door to a whole
series of other progressions, including simultaneous transparent data
exchange and demultiplexing as in causing the LCD and a printer to all
work together and to function smoothly.
As before stated, the invention can also be employed in connection with an
ISDN telephone exchange as well as the conventional exchanges. For
example, the invention may be used in a standard ISDN reference point or
interface 5, FIG. 3, described, for example, in the Geneva 1985
International Telecommunication Union publication Integrated Services
Digital Network (ISDN), commencing at page 128 (Standard I 411). This
standard interface 5 specifies that audio information is transferred by
means of a so-called B channel of, labelled "ISDN DIGITAL B CHANNEL", of
8000 bytes/second or 64K bits/second (64 KB). In the SRS-equipped ISDN
vesion of FIG. 3, this digital signal is introduced to the HSP port of the
DSP chip 6, FIG. 1, instead of the codec 9. This allows the audio rate
loop of the DSP chip 6, FIG. 1, to function as before described. Some
additional functionality is, however, made possible by the capabilities of
the ISDN telephone interface. Within the ISDN network there exist limited
capabilities for matching certain characteristics of the calling and
called telephones. These can be significantly extended by means of the SRS
concept of the present invention. Phone connections that are not wholly
contained within the ISDN network receive no benefit from any ISDN
matching features. The SRS technique overcomes this by operating smoothly
with SRS telephone and has the additional feature of advising the users.
Certain features for determining the characteristics of the digital path
are provided by this ISDN standard, but these are effective only for calls
within the ISDN network; they do not inform users on the line; and, in
many cases, they operate so as to deny service if the channel is not of a
particular type, as opposed to adapting smoothly to the characteristics of
the channel which is available; and existing telephone company tariffs and
procedures do not envision the possibility of providing data and voice
simultaneously to a single number--but the invention solves all these
problems, including obviating the necessity for any revising of telephone
company procedures.
It is desirable for the SRS-equipped system to be able to automatically
determine that the call has been completed within the ISDN network, if in
fact that is the case. This is accomplished by a so-called SRS-ISDN
digital signal, the first part of which is exactly the SRS in digitized
form. Following that, however, is a "digital signature", consisting of
just a few bytes containing some sharp transitions. After the SRS has been
recognized, the DSP chip 6 examines the digital data just following the
now-identified SRS for the exact digital signature. If the digital
signature is found, it means the call has been completed by a fully
digital (i.e. ISDN) path. If analog processing has been involved at any
stage of the call path, the digital signal will be corrupted (since it
represents frequency components well beyond the normal telephone
bandwidth), and the system will operate in the analog SRS mode.
If the existence of a digital path has been detected, the logical operation
of the SRS protocol (configuration communications, demultiplexing, etc.)
is not affected; however, improved data rate is easily possible (the
before-mentioned B channel having the said basic data rate of 64 KB).
Probably, however, some form of "modulation" | | |
