U.S. patent application number 10/336950 was filed with the patent office on 2003-07-03 for voice and text transmission system.
Invention is credited to Colwell, Kevin, Engelke, Robert, Grittner, Kurt, Havens, Jeffrey, Hofstetter, Dean, McCulley, Mathew, Vitex, Troy.
Application Number | 20030125952 10/336950 |
Document ID | / |
Family ID | 25367483 |
Filed Date | 2003-07-03 |
United States Patent
Application |
20030125952 |
Kind Code |
A1 |
Engelke, Robert ; et
al. |
July 3, 2003 |
Voice and text transmission system
Abstract
A communication system and format is described for use in
assisted telephonic communications, intended to help users who are
hearing impaired use the telephone system. A relay connects a
hearing user with the assisted user. The relay creates a text
message stream containing the words spoken by the hearing user. The
relay then combines the digital characters of the text message with
packets of digitized voice spoken by the hearing user and sends the
combined digital data packets to the station of the assisted user.
The station of the assisted user is capable of separating the voice
from the text and displaying the text for reading by the assisted
user.
Inventors: |
Engelke, Robert; (Madison,
WI) ; Colwell, Kevin; (Middleton, WI) ; Vitex,
Troy; (Madison, WI) ; Havens, Jeffrey;
(Madison, WI) ; Grittner, Kurt; (Madison, WI)
; Hofstetter, Dean; (Verona, WI) ; McCulley,
Mathew; (Madison, WI) |
Correspondence
Address: |
Nicholas J. Seay
Quarles & Brady LLP
P O Box 2113
Madison
WI
53701-2113
US
|
Family ID: |
25367483 |
Appl. No.: |
10/336950 |
Filed: |
January 3, 2003 |
Related U.S. Patent Documents
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Application
Number |
Filing Date |
Patent Number |
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10336950 |
Jan 3, 2003 |
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09876340 |
Jun 7, 2001 |
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6504910 |
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Current U.S.
Class: |
704/260 |
Current CPC
Class: |
H04L 12/66 20130101;
H04M 7/006 20130101; H04M 2201/40 20130101; H04M 3/42391
20130101 |
Class at
Publication: |
704/260 |
International
Class: |
G10L 013/00 |
Claims
We claim:
1. A method for transmitting voice and text of words over a
telephonic connection between a hearing user and an assisted user
through a relay, the method comprising the steps of digitizing the
voice of the hearing user; creating a digital text at the relay
corresponding to the words spoken by the hearing user; combining
the digitized voice and the digital text into combined digital data
packets, at least some of the digital data packets combining at
least one byte of digitized voice data with at least one byte of
text representing a character in the text of the words spoken by
the hearing user; and transmitting the combined packets to the
station of the assisted user over a telephone connection so that
the station can reconstitute both voice and text from the digital
data packets for the assisted user.
2. A method as claimed in claim 1 wherein for each digital data
packet, the first byte is a hexadecimal DA.
3. A method as claimed in claim 1 wherein each data packet includes
within it a format character indicating the format of that
packet.
4. A method as claimed in claim 3 wherein for each digital data
packet, the second byte is the format character indicating the
format of that digital data packet.
5. A method as claimed in claim 3 wherein some of the data packets
contains only digitized voice and some of the data packets combine
digitized voice with a text character, the nature of the data in
each packet indicated by the format character.
6. A method as claimed in claim 3 wherein at least one data packet
is defined to carry call set-up information from the assisted user
to the relay.
7. A method as claimed in claim 3 wherein at least one format of
data packet provides for the transmission of a DTMF command from
the assisted user to the relay to command the relay to transmit a
DTMF tone on the telephone connection to the hearing user.
8. A system for assisting a user in telephonic communications with
a hearing user, the system adapted to communicate with an assisted
user station capable of displaying text for the assisted user, the
system comprising a relay capable of converting spoken voice
received over the first telephonic connection into text; a first
telephonic connection between the hearing user and a relay; a
second telephonic connection between the relay and the station of
the assisted user; the relay programmed to created a series of
digital data packets for transmission to the station of the
assisted user over the second telephonic connection, each of the
digital packets including digitized voice of the hearing users and
at least some of the digital data packets including text of the
words spoken by the hearing user.
9. A system as claimed in claim 8 wherein for each digital data
packet, the first byte is a hexadecimal DA.
10. A system as claimed in claim 8 wherein each packet includes a
format character indicating the format of that particular packet so
that packets of varying format can be transmitted in a single
communication session.
11. A system as claimed in claim 10 wherein for each digital data
packet, the second byte is the format character indicating the
format of that digital data packet.
12. A system as claimed in claim 10 wherein some of the data
packets contains only digitized voice and some of the data packets
combine digitized voice with a text character, the nature of the
data in each packet indicated by the format character.
13. A method as claimed in claim 10 wherein at least one data
packet is defined to carry call set-up information from the
assisted user to the relay, the content of that packet being
indicated by the format character in the packet.
14. A method as claimed in claim 10 wherein at least one format of
data packet provides for the transmission of a DTMF command from
the assisted user to the relay to command the relay to transmit a
DTMF tone on the telephone connection to the hearing user.
15. A system as claimed in claim 8 wherein the station of the
assisted user is a captioned telephone, operating as an analog
telephone but also capable of displaying the text of the words
spoken by the hearing user for the benefit of the assisted
user.
16. A system as claimed in claim 8 wherein the station of the
assisted user is a portable personal interpreted device capable of
providing the assisted user with a visual display of the text
created at the relay from the words spoken by the hearing user.
17. A method capable of transmitting both voice and text of words
over a telephonic connection to facilitate a conversation between
two users, the method comprising the steps of digitizing the voice
of a first of the users; generating a digital text corresponding to
the words spoken by the first of the users; combining the digitized
voice and the digital text into digital data packets, at least some
of the digital data packets having at least one byte of digitized
voice data and at least some of the digital data packets having at
least one byte of text representing a character in the text of the
words spoken by the first user, each packet including a format
character indicating the format of that packet so that packets of
differing format can be sent in a single communication session; and
transmitting the combined packets to the station of the second user
over a telephone connection so that the station of the second user
can supply text to the second user of the words spoken by the first
user.
18. A method as claimed in claim 17 wherein for each digital data
packet, the first byte is a hexadecimal DA.
19. A method as claimed in claim 17 wherein for each digital data
packet, the second byte is the format character indicating the
format of that digital data packet.
20. A system as claimed in claim 17 wherein some of the data
packets contains only digitized voice and some of the data packets
combine digitized voice with a text character, the nature of the
data in each packet indicated by the format character.
21. A method as claimed in claim 17 wherein at least one data
packet is defined to carry call set-up information from the
assisted user to the relay, the content of that packet being
indicated by the format character in the packet.
Description
CROSS-REFERENCE TO RELATED APPLICATIONS
[0001] None.
STATEMENT REGARDING FEDERALLY SPONSORED RESEARCH OR DEVELOPMENT
[0002] None.
BACKGROUND OF THE INVENTION
[0003] The present invention relates to the general field of
telephone communications. In more particular, the invention relates
to systems to assist telephone communications by those persons who
are deaf, hard of hearing, or otherwise have impaired hearing
capability.
[0004] Most modem human communications in both social and business
environments takes place through sound communications. Yet within
modem society there are many persons who have attenuated hearing
capability. To assist those persons in making use of our telephonic
communication system built for the hearing majority, there has been
developed a system of telephone communication which has been
principally used by the deaf community. That system makes use of a
category of device known variously as a telecommunication device
for the deaf (TDD), text telephone (TT) or teletype (TTY). Current
TDDs are electronic devices consisting of a keyboard and a display
as well as a specific type of modem, to acoustically or directly
couple to the telephone line. Modem TDDs permit the user to type
characters into their keyboard, with the character strings then
encoded and transmitted over the telephone line to be displayed on
the display of a communicating or remote TDD device.
[0005] Most TDD communication is conducted in an idiosyncratic code
specific to the community of TDD users. This code, known as Baudot,
evolved historically at a time when many telecommunication devices
for the deaf were based on mechanical or electromechanical devices
rather than the current technology based on digital electronic
components. Accordingly, the Baudot protocol was constructed for a
set of constraints which are no longer relevant to present date
devices. The original Baudot protocol was a unidirectional or
simplex system of communication conducted at 45.5 Baud. The
conventional Baudot character set was a character set consisting of
5 bit characters and the system encodes the bits of those
characters in a two-tonal system based on carrier tones of 1400 and
1800 Hertz.
[0006] The system of TDD communications is widely used and in fact
has become indispensable to the deaf community throughout the
industrialized world. Deaf persons extensively communicate with
their neighbors and with other deaf and hearing people remotely,
using the TDD system. In addition, systems have been developed to
facilitate the exchange of communication between the deaf community
and hearing users who do not have access to or utilize a TDD
device. In the United States, telephone companies have set up a
service referred to as a "relay." A relay, as the term is used
herein, refers to a system of voice to TDD communication in which
an operator, referred to as a "call assistant," serves as a human
intermediary between a hearing user and a deaf person. Normally the
call assistant wears a headset that communicates by voice with the
hearing user and also has access to a TDD device which can
communicate to the deaf user using a TDD appropriate protocol. In
normal relay operations in the prior art, the call assistant types
at a TDD keyboard the words which are voiced to her by the hearing
user and then voices to the hearing user the words that the call
assistant sees upon the display of his or her TDD. The call
assistant serves, in essence, as an interpreting intermediary
between the deaf person and the hearing person to translate from
voice to digital electronic forms of communication.
[0007] To facilitate and modernize the systems available for
providing telecommunication services for the deaf, efforts have
been made to both update the techniques for providing assistance to
the hearing impaired as well as providing services to users who are
modestly hearing impaired but not deaf. In U.S. Pat. No. 5,909,482,
a relay is described which uses a re-voicing technique and a speech
recognition engine to greatly improved the speed of services
provided by a relay. This patent also discloses a small portable
device, called a personal interpreter, which make possible
providing location independent and instantaneously available
interpreting services to the deaf. In U.S. Pat. No. 6,075,842,
methods and devices for providing text enhanced telephony are
described in which a text stream is provided along with voice in
telephone communications with hard of hearing users. The text
stream is used to provide the assisted user with a visual
representation of the text of what is said by the other person in a
communication session, so as to gently assist a person with some
hearing deficiency in using the telephone. The full specification
of U.S. Pat. Nos. 5,909,482 and 6,075,842, as well of that of each
other patent referred to in this document, is incorporated herein
by reference.
BRIEF SUMMARY OF THE INVENTION
[0008] The present invention is summarized in a method for
transmitting voice and text of words over a telephonic connection
between a hearing user and an assisted user through a relay, the
method including the steps of digitizing the voice of the hearing
user; creating a digital text at the relay corresponding to the
words spoken by the hearing user; combining the digitized voice and
the text into combined digital data packets, each packet including
a format character indicating the type of format for that packet,
at least some of the digital data packets combining at least one
byte of digitized voice data with at least one byte of text
representing a character in the text of the words spoken by the
hearing user; and transmitting the combined packets to the station
of the assisted user over a telephone connection so that the
station can reconstitute both voice and text from the digital data
packets for the assisted user.
[0009] The present invention is also summarized in a communication
system using that method to communicate voice and text of the words
spoken by the voice to a station used by an assisted user.
[0010] The present invention is intended to create a flexible
communication protocol, using minimal overhead, which is capable of
sending voice and the text for the words spoken by that voice, in
digital form over common telephonic communication linkages.
[0011] Other objects, advantages and features of the present
invention will become apparent from the following specification
when taken in conjunction with the accompanying drawings.
BRIEF DESCRIPTION OF THE SEVERAL VIEWS OF THE DRAWINGS
[0012] FIG. 1 is a schematic diagram of a system using the method
and protocol of the present invention, showing particular details
of the exemplary relay used in the system.
[0013] FIG. 2 is a schematic diagram of a captioned telephone for
use in the system of FIG. 1.
DETAILED DESCRIPTION OF THE INVENTION
[0014] The communication protocol of the present invention is
intended to facilitate voice and text communications between
hearing persons and assisted persons by a relay that intermediates
the call. Since the full implementation of the protocol is most
appropriately done by a relatively sophisticated relay, the
construction of such a relay and some details about the device at
the assisted users end will be described first.
[0015] Shown in FIG. 1 is a schematic view of the relay which can
intermediate such a call. At 12 is indicated the normal telephone
of the hearing user. The telephone 12 connects through a telephonic
connection 14 to a DAA and hybrid circuit 16 at the relay. It is
envisioned that the hearing user's telephone 12 can be any of the
devices generally thought of by laypersons as a telephone,
including but not limited to land line telephones, cellular
telephones, PCS devices and audio links over the internet. At any
event, what is received at the DAA and hybrid circuit 16 of the
relay is a voice signal. The voice from the DAA and hybrid circuit
15 is connected as the input to a codec 18. Codecs have become
industry standard devices that convert an analog signal, such as
analog voice, into digital data. Commercially available codec
integrated circuits, such as those from Texas Instruments and
Analog Devices, can convert both analog voice signals into digital
data and do the reverse, i.e. reconstruct an analog voice signal
from digital data representing voice. The output digital signal
from the codec 18 is then connected to a circuit or software
designed to cancel echo on the telephone line, as indicated at
20.
[0016] Another type of communication circuit in common use today is
referred to as a voice coding and decoding circuit or "vocoder." A
vocoder is a type of digital signal processing chip or an algorithm
implemented by a processor specifically designed to transform
digital data carrying voice to compress the data for transmission.
There are several common standards for vocoders so that telephones
from different manufacturers using digital transmission formats can
communicate with each other. Such formats include GSM, G.729, and
G. 723. One preferred format for the relay of the present invention
is G. 729, which encodes speech into 8000 bits per second with an
audio quality comparable to a long distance telephone line. This
format, and the 8000 bit per second data rate, enables the
communication protocol of the present invention to be used even
over cellular telephone connections. The vocoder is indicated at
22, and is capable of both encoding, or compressing, and decoding,
or de-compressing, the digital data stream carrying voice.
[0017] Indicated at 24 is the UVT formatter, which is actually
implemented electronically by a specially programmed microprocessor
or digital signal processor. The UVT formatter 24 combines the
digital data stream representing voice, from the vocoder 22 with a
digital data stream carrying text from a call assistant computer,
indicated at 26. The call assistant computer 26 is a general
purpose digital computer preferably equipped with a speech
recognition software package. The cell assistant wears a headset 28
connected to transmit the voice of the hearing user to the ear of
the call assistant. The call assistant repeats, or "re-voices," the
words spoken by the hearing user into a microphone of the headset
28 that is connected to the computer 26. The speech recognition
engine in the call assistant computer 26 recognizes the voice of
the call assistant and translates that voice into a text stream.
The output of the call assistant computer 26 is thus a digital data
stream carrying text which is provided to the UVT formatter 24 as
well. The UVT formatter is constructed to combine the digital data
stream carrying voice with the digital data stream carrying text,
using the UVT format described below. The UVT formatter is also
capable of doing the reverse, that is, separating the digital data
stream carrying voice from the digital data stream carrying text,
using information contained in the UVT protocol to make that
separation.
[0018] The output of the UVT formatter 24 is connected to a modem
30, in this case illustrated as an industry standard V.32bis format
modem. The output of the modem 30 is connected through another
codec 32 to a hybrid circuit 34 and DAA at the output of the relay.
The DAA and hybrid circuit 34 is connected, by any form of
telephonic connection 36 to an assisted user station 38. The
assisted user station 38 can be a personal interpreter of the
general type as shown in U.S. Pat. No. 5,974,116, or a captioned
telephone of the general type as shown in U.S. Pat. No. 6,075,842,
or any other device intended to assist the assisted user in the
communication session by providing text to the assisted user to
help that user understand the words spoken by the hearing user.
[0019] Whether the protocol of the present invention, here
sometimes referred to as "UVT," is used with a personal interpreter
or a captioned telephone, the assisted user station needs to be
able to receive and process the type of packets sent by the relay.
Shown in FIG. 2 is a captioned telephone device 100 with that
capability. In FIG. 2 the interior components of the captioned
telephone device 100 are illustrated in block diagrams indicating
the digital logic components from which the device may be
constructed. It is preferred, however, that the components within
the dotted lines in FIG. 2, labeled as DSP software, actually be
implemented in the form of a software routine operating a digital
signal processing integrated circuit to perform the functions of
the illustrated blocks. In the captioned telephone 100, the
telephonic connection to the relay is indicated at 102. The input
telephonic signal connects to a DAA and hybrid 104 and then to a
codec 106 to digitize input signals. Following the codec 106 is
modem 108, the output of which connects to a UVT formatter 110. The
UVT formatter 110 operates to separate the digitized voice signals
from the digital text signals. The digitized text signals are
transferred from the UVT formatter to a visually readable display
112 on which the text can be displayed for the assisted user. The
digitized voice signal is transferred from the UVT formatter 110 to
a vocoder 114, compatible in format to the vocoder used in the
relay of FIG. 1, in this case using format G.729A. The output of
the vocoder 114 is connected through an acoustic echo control 116
to another codec 118 which reconstructs the analog voice signal for
delivery to the handset 120 of the assisted user. In summary, the
assisted station decompressed the digital data stream and separates
the digital text data from the data representing digitized voice.
The text is displayed on the display 112 and the voice is
reconstituted into analog and played on the speaker in the handset
120. The assisted user thus receives both the voice of the hearing
user and is provided a text display of the words spoken by the
hearing user.
[0020] In the basic operation of the system illustrated in FIG. 1,
the hearing user at telephone 12, and that person's voice is
converted into digital form in the relay, that digital form being a
direct digitalization of the hearing person's voice. At the same
time, or at very nearly the same time, the call assistant computer
creates a text data stream, also in digital form, of the text of
the words spoken by the hearing person. The UVT protocol,
implemented by the UVT formatter 24 in the system of FIG. 1
provides a methodology to efficiently combine the two digital data
streams, one for voice one for text. The two digital data streams
are combined in a method that is convenient to create and transmit
and convenient as well to separate at the receiving end. The
problem in combining the two digital data streams is that the
digitized voice tends to require much more data to transit than the
text data stream. It is inconvenient, however, to interrupt the
transmission of voice to transmit the needed text data. It is also
important that the total data transmitted not exceed the carrying
capacity of any of the forms of telephonic connections supported by
the network. The solution described here is to continually send
formatted information packets carrying the digitized voice to the
remote station and then, in addition and as needed, some of the
data packets are flagged to carry a portion of the digital text
message data stream in the same packet. Since the data transmission
requirements for the text message are so small in comparison to
that of the digitized voice, single bytes of text message are
combined with multi-byte portions of digitized voice data in these
specialized data packets. It is a unique attribute of the data
packets described here that the packets carry digitized voice and
at the same time carry text data for the words contained in the
speech of the voice. In other words, digitized voice and digitized
text for the words spoken by that voice are carried in common
digital data packets. The text and voice may be delivered
simultaneously or near simultaneously. The text can be delivered to
the assisted user as the text stream is created by the computer of
the call assistant, in which case the text stream may lag the
corresponding voice signal by a brief delay. In that event, the
text character may not travel in the same packet as the voice for
the word of which the character is a part. As an alternative, it is
possible to slightly delay the transmission of the voice of the
hearing user through the relay so that the text and corresponding
voice are transmitted to the assisted user at approximately the
same time.
[0021] To accomplish these unique objectives, a new protocol for
voice and text transmission has been designed. This format uses a
single format of data packet, sent as a digital data packet, but
the packets do not all carry the same type of content. Some packets
carry only digitized voice, while other packets carry both
digitized voice and digital text data. The packet header is used to
indicate the type of packet, and as long as the designation of the
packet header remains constant, later revisions of the format
permit other later packet specifications to be defined. Other
packet types include software upgrade data, user preferences for
system set-up, settings for parameters of devices or for
configurations, and error code information. Typically, at the
initiation of the communication session, special packets are
transmitted between the communicating devices. Such special packets
can be used to identify the version number of the protocol used by
the communicating devices, information on the type of connection
and speed, information of the type of data in following data
packets, error correcting formats or codes, device status such a
processor of memory resources available, or information on the
downloading of software upgrades to devices.
[0022] Each call to the relay service begins with an exchange of
special packets that allow the captioned telephone or personal
interpreter to provide the relay service with the information
necessary for the relay to automatically set-up the desired type of
service and, for captioned telephone calls, to complete the
connection to the hearing person. When the data communication
connection (e.g. V.32bis) is established, the relay service sends a
special packet called a Request for Call Set-up Information packet.
The captioned telephone or personal interpreter device responds
with a special packet in return, here called the Call Set-up
Information Block packet. The Call Set-up Information Block packet
includes the service type requested by the user (e.g. captioned
telephone or personal interpreting), a user identification number,
a user password in some cases, and, for a captioned telephone call,
the telephone number of the hearing party. In relays as operated in
the prior art, the user had to type in information of this type, in
response to inquiries from the call assistant, to complete the call
set-up. This protocol permits the call to be set up automatically
in a fraction of the time previously required.
[0023] Another type of special packet is used to transmit DTMF
control signal to the relay equipment. A captioned telephone user
may dial, using the relay service, a telephone number that connects
to an automated attendant or voice response unit. These automated
attendant type devices prompt the caller to enter their choice by
pressing the number buttons on the their touch-tone telephone, i.e.
"press 1 for sales." Such devices and voice mail systems are
commonly encountered in telephone usage today. In current relays, a
TDD user must type instructions to the relay call assistant to
convey their intention, e.g. "press 1," and the call assistant then
manually presses a key to produce the DTMF signal on the second
line. The delays involved when the TDD user types to the call
assistant and the call assistant manually selects the digit often
exceeds the time permitted by the automated systems for the user to
make a selection. In this event, the traditional relay call
assistant must dial back the telephone number and wait for the
system to reach the point to enter the user's choice. The problem
can result in multiple calls and long time delays for prior art
relay users.
[0024] The DTMF special packet in the present format provides the
captioned telephone user with the functionality of a traditional
touch tone telephone in the digital captioned telephone
environment. The captioned telephone connects to the relay service
using a data communications protocol (e.g. V.32bis) which does not
permit the captioned telephone to emit the DTMF tones directly onto
the telephone line. The captioned telephone could generate the DTMF
tone signal and transmit that signal to the vocoder. Such a signal
would be carried as digital data to the relay equipment where it
would be converted back to tones. However, most vocoders cannot
produce a precise enough set of frequencies or loud enough signals
to meet network DTMF standards. To avoid that limitation, the
present protocol permits the transmission of the special DTMF
packet that instructs the relay to produce the DTMF signal directly
on the telephone line to the other party. To use this feature, the
captioned telephone user simply presses the numerical dial of his
or her telephone, and the captioned telephone then generates a
command packet to the relay instructing it to produce the correct
DTMF tones on the second telephone line automatically without
interaction with the call assistant.
[0025] The embodiments of FIGS. 1 and 2 are illustrated with
telephonic connections between the hearing person and the relay and
between the relay and the assisted user. It is specifically
contemplated that the manner of actual telephonic connection
between these parties, and the number of telephone lines or
telephone line equivalents that are used is not important. The term
telephonic connection, as used here, is intended to apply to actual
dedicated connections through the telephone system, such as
land-lines or analog cellular connections. The term is also
intended to encompass other types of connections that can serve as
telephonic connections in the lay sense of the term, such as
digital cellular telephone service, PCS service and communication
over the internet using IP protocol. The term telephone line here
is intended to encompass both traditional twisted pair physical
telephone lines as well as any type of channels or software sockets
that provide an equivalent connection between users of the
telephone system. So while the use of the protocol described here
is particularly intended to make possible transmission of text and
voice using a minimal number of telephone lines, it is envisioned
that the protocol may be used using two or more telephone lines or
over other higher bandwidth forms of interconnection. Examples of
such higher bandwidth connections include ISDN or DSL telephone
connections, or other standards that provide a bandwidth more than
the equivalent of one telephone line. Another example is the use of
the present or future internet system, such as the present IP
format, that permits communication sessions analogous to present
telephone system sessions through computer to computer
linkages.
[0026] One specific multiple line arrangement is specifically
contemplated whereby the assisted user receives a telephone call
from a hearing user over a telephone line, and then that assisted
user conferences to the relay to obtain text assistance for the
call. The assisted user would communicate with the relay in the UVT
format described here. The connection to the relay could be by a
second telephone line which carries voice to the relay and
transcribed text back to the assisted user, using the UVT format.
The connection to the relay could also be accomplished by
conferencing in the relay on the line the call was received on and
then arranging a separate telephonic connection between the
assistance device of the assisted user and the relay to transmit
text. While these sorts of connections might not make use of all of
the capabilities of the UVT format, since they may not require
voice and text on the same telephone line (or in the same packets),
it may still be useful to employ the UVT protocol for such
services. Once a relay is set up to communicate in UVT protocol, in
order to support single line calls, and assuming only that the
set-up options of the protocol for the relay are defined to support
multiple line calling arrangements, it may be convenient to use
this same UVT protocol for the multiple line calls. In that event,
text only packets can be defined and sent using the same basic
format as described here for text and voice packets.
[0027] The specification of the UVT data packets begins with the
basic packet header. The standard packet begins with two special
characters, each of one byte (eight bits). The first byte is a
specially designated packet initiation signal, indicated here as
0xda, which is the 8 bits hexadecimal character DA. This first byte
just indicates the start of a packet. The second special character
is a single byte indicating the type or format of data packet. This
is the data packet format type and is indicated in the following
description using the nomenclature 0x09, which indicates a type 9
(hexadecimal format) packet format.
[0028] Thus the structure of a type 9 data packet is as
follows:
1 Speech Speech Start of packet Packet format Text character frame
1 frame 2 0xda 0x09 0nNN 0xNN . . . 0xNN . . . 8 bits 8 bits 8 bits
80 bits 80 bits
[0029] In this representation, the packet start is the special
character, hexadecimal value DA. The second byte is an indication
of the packet type, in this case type 9. That packet type is
defined as a single byte of text data, followed by two frames each
of 80 bits (10 bytes) of digitized speech data. The packet type
also defines the type of compression on the speech data, in this
case G.729A digitized speech standard. The designation 0xNN
indicates any 8 bit value.
[0030] A type 19 UVT data format is defined as follows:
2 Start packet Packet format Speech format Speech frame 0xda 0x19
0xNN 0xNN . . . 8 bits 8 bits 80 bits 80 bits
[0031] A type 19 data packet is defined like data packet nine,
except that a text character is not included. Thus this data packet
is used when no text data needs to be sent, so includes only speech
in format G.729A.
[0032] This UVT protocol is thus able to mix the transmission of
both text and voice data. While there is a connection between the
parties, packets of voice (or sounds) are continuously transmitted.
As the hearing party speaks, the words spoken by the hearing person
are transcribed into text at the relay and the relay combines voice
data and text data for transmission to the assisted user. This
protocol requires very little overhead, as little as two bytes per
packet, one start character and one to indicate the packet type.
This is preferable to the alternative, sending text and voice in
separate packets, since that would require the additional overhead
associated with sending another packet (start of packet and packet
type) for the text characters. This format permits transmission of
voice and text at a steady transmission rate of 9,600 bits per
second. This allows the protocol to be used over analog cellular
telephone systems that currently support only 9600 baud. At this
relatively slow speed, there is simply not enough time to send a
speech packet followed by a text data packet.
[0033] The fact that text and speech packets are combined means
that loss of a packet is not a significant problem. Since each
packet only contains a time period of 0.010 to 0.020 seconds, and
the omission of the speech from such a time period would not
usually be noticed by the hearing users. Since loss of a packet
means loss of only a single text byte, only one character is lost
from the text data transmitted to the assisted user. Assisted users
are accustomed to correcting mentally for informal and erroneous
spellings, and so in most instances the loss of a single character
should not cause serious disruption of the conversation. If the
text was sent in separate packets, the loss of a packet could lose
a significant amount of text information.
[0034] Larger blocks of text can be sent in a variable length text
only packet. This may be used to transmit prompts to the user's
device during call set-up or at other times when it is not
necessary to carry voice data. Examples of when such a packet might
be used include indicating to the assisted user that the captioning
device is on-line or indicating when the outbound telephone call
has been completed. This type of packet is referred to here as a
type ID (again hexadecimal notation) packet, which is specified as
follows:
3 Start of Packet packet format Sequence number Length Characters
CRC 0xda 0x1d 0xNN 0xNN 0xNN 0xNN 8 bits 8 bits 8 bits 8 bits 8
bits .times. length 8 bits
[0035] The sequence number is simply an ordering of the sequence of
packets which together form a single message. Length refers to the
number of characters in this packet. The notation CRC refers a type
of commonly used error-checking methodology (cyclic redundancy
check) that may be used to conveniently perform error-checking in
this packet type.
[0036] Other types of command and control packet types are used to
initiate the service and for other specialized functions. Examples
of these are described next.
[0037] The packet transmitted by the relay service to the calling
device, to request the transmission of call set-up information, is
referred to here as a type 20 packet. The format of a type 20
packet is as follows.
4 Start of packet Packet format 0xda 0x20 8 bits 8 bits
[0038] When the relay service sends the type 20 packet to the
calling device, the calling device should respond with a type 21
packet. A type 21 packet is intended to provide call set-up
information to the relay service to specify the type of service and
service options to be used on the call. These options can be
implemented in the calling device as pre-selected parameters that
are automatically transferred to the relay when a defined type of
call (e.g. a captioned telephone call) is initiated. The format of
the type 21 packet, or call set-up information block packet, is as
follows:
5 Start Dial of Packet through Service ID packet format number ;
type ; number ; Password CR CRC Oxda 0x21 DT = 1608 0x3b S = 2 0x3b
U = 1234 0x3b P = x456 0x0d 0xNN 2385400 5678 8 bits 8 bits
Variable 8 bits 8 bits 8 bits Variable 8 bits Variable 8 bits 8
bits
[0039] In this format, the call set-up information is provided in 8
bit ASCII characters. Some of the fields are of variable length and
therefore are separated by a special character ";" or 0x3b which is
intended only to serve as an indicator of the end of a variable
length field. A "CR" (0x0d) character indicates the end of all of
the variable length data fields. Each data filed begins with a
filed type indicator, such as the indicator "DT=" indicates that
the characters following make up a dial through telephone number.
This allows the fields to be sent in any order and permits unused
fields to be omitted altogether. It is then also possible to define
new filed types at a later time by selecting a new filed type
indicator and separating the new filed from others by the ";"
indication. A CRC is again used to confirm the accuracy of the data
within the information block by permitting an error check to be
performed.
[0040] A UVT command packet for DTMF information includes the
identification of the DTMF digit tone to be produced, the duration
of the tone, and a CRC. The DTMF digit information indicates to the
relay which DTMF digit signal to produce, the duration indicates
how long the relay equipment should produce the signal and the CRC
again is for error checking. When the captioned telephone user
presses a number button on his or her device during a captioned
telephone call, the captioned telephone device sends a UVT DTMF
command packet to the relay. The relay then generates the DTMF tone
on the telephone line to the other party. This tone will sound to
all users like the normal DTMF tones produced by a conventional
telephone. As long as the user holds down the button, the captioned
telephone device will continue to send such UVT DTMF command
packets to the relay, and the relay will continue to impress DTMF
tones on the telephone line to the other party. Thus, as in
traditional telephone systems, the user can control the length of
time that the DTMF tone is sent. The captioned telephone will
normally select a duration for each DTMF command packet that is
longer than twice the interval between transmission of DTMF command
packets to the relay, so that the DTMF tone continues from the
relay even if a single packet is missed or corrupted in some way.
The format for packet type 1C (again hexadecimal notation) is as
follows:
6 Start of packet Packet format DTMF digit DTMF duration CRC 0xda
0x1c 0xNN 0xNN 0xnn 8 bits 8 bits 8 bits 8 bits 80 bits
[0041] Note that is not required that each packet be in the same
format. Since the identification of packet type travels with the
packet, packets that carry only voice can be interspersed with
packets carrying voice data and text without disruption or
difficulty. This also permits housekeeping packets, about machine
settings or protocols, to be transmitted at the beginning of the
communication session, or during lulls, without creating
confusion.
[0042] It is to be understood that the present invention is not
limited to the embodiment described above, but embraces all such
modified forms thereof as come within the scope of the following
claims.
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