Note: Descriptions are shown in the official language in which they were submitted.
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TRANSMISSION OF AUDIO OR MULTIMEDIA INFORMATIONAL MESSAGES TO USERS
BACKGROUND OF THE INVENTION
FIELD OF THE INVL'NTION
Th:, invention relates generally to telephony and more particularly to devices
and methods that integrate telephony and other messaging systems.
BACKGROUND
Information exchange plays a vital role in modern society and one that is of
growing importance. For decades the telephone system has played a central role
in
information exchange.
Increasingly, the functions of traditional telephony are converging with other
means for information exchange, such as computer networks. One result of this
is that
~5 users of equipment that integrates telephone and other functions have a
richer set of
communications possibilities available to them. Another result is that
providers of
networking services face increasing competition from businesses in many
markets. A
third result is that those who do not have the financial ability to purchase
devices and
services that provide integrz.ed communications are at an even greater
disadvantage in
our society than in prior times.
Accordingly, it would be desirable for there to be an invention that provided
a
means for network service provides to enhance their revenues. Further, it
would be
desirable for there to exist a system that enabled access to telecommunication
services
to be subsidized. Still further, it would be desirable for a solution to these
problems to
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be able to take advantage of the increased functions of telecommunication
devices,
such as personal computers.
Conventional systems are lacking in features desirable for achieving these,
and
other, capabilities of the invention. For instance, conventional voice mail
systems
provide a means for messages to be provided to a telephone user; some even
provide a
means for users to retrieve messages based on user preferences. However, such
systems are not suitable for network service providers to develop a revenue-
generating
service model that allows a sponsor's criteria to be consulted when
determining
whether and/or which informational messages) should be sent to a subscriber.
By
l0 providing a means for sponsors to customize the transmission of
informational
messages to aural communication device users, versions of the present
invention
provide features desirable to sponsors that are lacking in conventional
systems and
therefore provide increased revenue opportunities for network service
providers.
Other conventional systems operate by a user dialing a "1-800" number to
begin interacting with the system; the user then typically navigates a menu of
options,
enters an identification number, is transmitted messages, and thereafter may
begin to
make their desired call. Such systems present unnecessary inconvenience to the
user.
It will be appreciated that from a sponsor's perspective, increa;.7~d user
time, effort,
and confusion in interacting with a system for transmitting informational
messages
may foster negative associations in the user's mind with the sponsor's goods,
services,
etc. It is therefore desirable that user frustration be minimized; accordingly
features
of the present invention can archive a sponsor's goals of transmitting
informational
messages to aural communication device users without a user having to actively
remember and dial a "1-800" to enter the system. Such "1-800" number-type
systems
effectively demand that a user complete two telephone calls in order to
participate in
the system. Particularly for telecommunications systems that cater to busy
customers,
such as cellular systems, this additional time spent to make what is, in
effect, an
additional call, may be a barner to wide adoption of the system. Still
further, such "1-
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800" number systems lack the desirable feature of the present invention by
which an
informational message may be transmitted to a user of a cellular communication
device during the interval of time after the user has entered the number to
which they
wish to be connected but before the communications path is established.
The aforementioned capabilities and other objects are obtained by a method,
apparatus, and system for the transmission of audio or multimedia
informational
messages, such as advertisements or the like, to aural communication device
users.
SUMMARY
An illustrative aspect of the present invention involves practicing a method
for
l0 providing at least one informational message to an aural communication
device user
in connection with an outgoing communication. One such method includes:
detecting
a first aural communication device is in use; retrieving a first informational
message ;
transmitting the first informational message to the first aural communication
device
for perception by the user; and providing an indication to the first aural
15 communication device that the first aural communication device is available
for
connection. to a second aural communication device. i he aural communication
device
may be, for instance, a conventional phone, a computer-telephony system, a
packetized voice communication device (e.g. a Voice over IP phone), a computer
programmed to provide a Voice over IP client, or a wireless communication
device,
20 e.g. a cellular phone. The informational message may be transmitted before
or after
(or both) a dialing indication is provided to the aural communication device.
Similarly, when a communication is incoming, informational messages may be
transmitted to a user of the aural communication device after they place their
aural
communication device 'off hook' but before the incoming call is connected. The
25 informational message may be in audio form, textual form, or multimedia
form
according to the situation and the capabilities of the aural communication
device.
Informational messages used witrr some versions of the invention may be stored
in
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compressed form and be decompressed before transmission; similarly,
informational
messages used with some versions of the invention may be stored in text form
and be
converted to an audio form before transmission.
Another illustrative aspect of the invention is that the informational
messages
sent to the aural communication devices can be controlled based on a profile
of a user,
a profile of a sponsor, or both. A user's profile may include, for instance
positive or
negative preferences, e.g., indications of the desire to be provided
informational
messages of a particular character or of the desire to not be provided
informational
messages of a particular character. A sponsor's profile may include, for
instance,
1o demographic, psychographic, geographic, or socio-economic preferences,
e.g.,
indications of particular attributes of subscribers to be targeted for
transmission of
informational messages.
Still another illustrative aspect of the invention is identity verification of
a
user. Versions of the invention may prompt a user to identify himself or
herself by,
for instance, a personal identifier such as a PIN, or by a voice print. User
identification information may be used in conjunction with user or sponsor
profiles)
to provide a more customized selection of informational messages for the user.
Reliable user identification may also operate with versions of the invention
which
invite commercial transactions and the user's identification used in
conjunction with
payment arrangements.
Another aspect of some versions of the present invention provides interactive
informational messages ("feedback"). Feedback messages may present a user with
selectable options. The user may select from the selectable options and have
an
output generated automatically. The output may be another message, possibly an
electronic mail message, or another informational message. In some versions of
the
invention, the automated output may be a coupon that is sent to the user. The
coupon
may be seat to a merchant or other party for redemption by the user. In some
versions
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a selectable feedback option rr~ay be to be connected to another party such as
a
sponsor, merchant, etc. for additional information, electronic commercial
transactions,
etc. In other versions of the invention a selectable feedback option may be to
establish a connection to a network resource, such as a web site, of a
sponsor,
merchant, etc.
A still further illustrative aspect of the present invention involves
practicing an
apparatus for providing at least one informational message to a user of an
aural
communication device. An illustrative apparatus is configured for connection
to at
least one aural communication device and a data network, said apparatus and
to includes: a processor; a memory comunicatively coupled with the processor;
and a
network interface communicatively coupled with the processor. The network
interface is configured for communication over the data network. The
illustrative
apparatus further includes a subscriber line interface communicatively coupled
with
the processor and a message agent, the message agent adapted to retrieve at
least one
15 informational message for storage in said memory. The at least one message
may be
communicated to the subscriber line interface under the control of the
processor.
Such an apparatus provides the functions of a telephone system supporting
several
phone lines and an client operable with an Internet-type network through which
informational messages may be conveniently and flexibly retrieved thought
client-
20 server communications with a message server. Such an apparatus may, for
instance,
be placed at a customer's premises or collocated with a central office switch.
Another aspect of a version of the present invention involves configuring a
central office switching system. An illustrative systems for providing
informational
messages to a user of aural communication device may include: a central
switching
25 system, the switching system routing voice telecommunications between at
least a
first aural communication device and a second aural communication device. The
central switch is further configured for communicative coupling with a data
network,
and the system further includes a message server, the message server
configured for
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client-server communications with the switching system via said data network.
The
message server stores an informational message and the system further includes
a
message manager, the message manager configuring the switching system to
request
the informational message from the message server. The illustrative system
further
includes a call manager, the call manager configuring the switching system to
perform
steps comprising: receiving the informational message from said message
manager;
and transmitting the informational message to the first aural communication
device.
Aspects of other versions of the invention may operate in a wireless
communications network and provide a method for providing a textual, image, or
to multimedia informational message in conjunction with an incoming
communication
for an aural communication device. The aural communication device may have an
associated identifier. Practicing an illustrative method according to the
invention
includes: receiving an indication of the incoming communication for the aural
communication device; retrieving an informational message from a message
server;
15 establishing a data communication connection to the aural communication
device;
transmitting the informational message to the aural communication device; and
thereafter establishing a connection to the aural communication device for the
incoming communication. Retrieving an informational messar ~.: i_vrom a
message
server may comprise: providing the identifier of the aural communication
device to a
2o message server; selecting an informational message responsive to the
identifier; and
retrieving the informational message selected responsive to the identifier.
Selecting
an informational message responsive to the identifier may also comprise:
querying a
subscription profile, the subscription profile comprising profile information
for a
subscriber, the subscriber associated with the identifier of the aural
communication
25 device; selecting an informational message responsive to the profile
information; and
retrieving the informational message selected responsive to the profile
information.
Other illustrative aspects of the invention operable with cellular wireless
communications provide a system for providing an informational message to a
user of
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a cellular aural communication device having an associated identifier. The
illustrative
system includes: a mobile switching system; a subscriber register, the
subscriber
register communicatively coupled with the mobile switching center, the
subscriber
register storing subscriber information, the subscriber register configured to
receive a
subscriber validation request from the mobile switching system and communicate
a
validation response to the mobile switching system, the validation response
comprising a subscriber identifier; and a message server configured for client-
server
communication with the mobile switching system, the message server storing
said
informational message, wherein, in response to the initiation of an outgoing
1o communication to a first phone number from the cellular aural communication
device,
the mobile switching system submits a validation request to the subscriber
register,
the mobile switching system receives a validation response from the subscriber
register, the mobile switching system communicates a subscriber identifier to
the
message server, the message server communicates the informational message to
the
mobile switching system for transmission to the aural communication device.
Aspects of other versions of the invention may operate in a packet voice
network in which the invention provides a method for providing informational
messages to a user of aural communication device based on, for instance, voice
over
IP technology. Practicing an illustrative system includes: a voice over IP
gateway
2o resident in a central office or with head-end equipment in a cable network.
The IP
gateway routes voice communications between at least a first aural
communication
device, such as an IP phone, and a second aural communication device, such as
a plain
old telephone or an IP phone. The IP gateway is further configured for
communicative coupling with a data network. Such a system further includes a
message server, a message manager, and a call manager. The message server is
configured for client-server communications with said gateway system via said
data
network, the message server further storing an informational message. The
message
manager configures the gateway system to request the informational message
from the
message server. The call manager configures the gateway system to perform
steps
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comprising: receiving said informational message from the message manager; and
transmitting said informational message to the first aural communication
device.
As will be further appreciated by this disclosure versions of the present
invention provide a flexible integration of telephony and internetworking
systems for
the provision of informational messages to users of av~ral communication
devices.
Features of the invention allow rich messaging capabilities to be provided to
users of
computer telephony systems. Further, when informational messages used with the
invention are advertisements or the like, versions of the invention provide a
desirable
means for networking service provides to increase their revenue by selling the
opportunity to provide advertisements to aural communication device users.
Such
versions of the invention provide a service method and system which could
greatly
expand the access to aural com.~nunication technology for those of more modest
means as service provides may subsidize the cost of providing telephony
equipment
and/or services with revenues gained from the provision of informational
messages to
aural communication device users.
BRIEF DESCRIPTION OF THE DRAWINGS
These and other features, aspects, and advantages of the present invention
will
become better understood with regard to the following description, appended
claims,
and accompanying drawings where:
Fig. lA depicts a exemplary network architecture in which the
invention may operate in accor~'ance with an illustrative
embodiment;
Fig. 1B depicts a exemplary public switched telephone network in
which the invention may operate in accordance with an
illustrative embodiment;
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Fig. 1 C depicts additional features of the exemplary network
architecture in which the invention may operate in accordance
with an illustrative embodiment;
Fig. 1D depicts an exemplary cellular network architecture in which
the invention may operate in accordance with an illustrative
embodiment;
Fig. lE depicts an exemplary network architecture including an
interactive voice response system with which the invention
may operate in accordance with an illustrative embodiment;
to Fig. 2 depicts a block diagram of a hardware architecture in
accordance with an illustrative embodiment;
Fig. 3 depicts a block diagram of a software architecture in
accordance with an illustrative embodiment;
Fig. 4 depicts a flow diagram of a method for outgoing
communication message transmission in accordance with an
illustrative embodiment;
Fig. SA depicts a flow diagram of a method for communicating an
AudioMT message in accordance with an illustrative
embodiment;
2o Fig. SB depicts a flow diagram of a method for communicating a
Feedback AudioMT message in accordance with an illustrative
embodiment;
Fig. 6A depicts a flow diagram of a method for communicating an
AudioTM message in accordance with an illustrative
embodiment;
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Fig. 6B depicts a flow diagram of a method for communicating a
Feedback AudioTM message in accordance with an illustrative
embodiment;
Fig. 7 depicts a flow diagram for a method for communicating an
AudioICM message in accordance with an illustrative
embodiment;
Fig. 8A depicts a flow diagram for a method for communicating an
AudioTM message to a VoIP client or Etherphone in
accordance with an illustrative embodiment;
1o Fig. 8Fs depicts a flow diagram for a method for communicating an
AudioICM message to a VoIP client or Etherphone in
accordance with an illustrative embodiment;
Fig. 9 depicts a flow diagram of a method for communicating an
AudioTM message in accordance with an illustrative cellular
embodiment;
Fig. 10 depicts a flow diagram of a method for communicating a
TextICM message in accordance with an illustrative cellular
embodiment;
Fig. 11 depicts a flow diagram of a method for communicating
2o informational messages to a PC & phone in conjunction with
an outgoing call in accordance with an illustrative
embodiment;
Fig. 12 depict a flow diagram of a method for communicating
informational messages to a PC & phone in conjunction with
an incoming call in accordance with an illustrative
embodiment;
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Fig. 13A depicts a flow diagram of a method for communicating an
PopupM message to a VoIP client in conjunction with an
outgoing call in accordance with an illustrative embodiment;
Fig. 13B depicts a flow diagram of a method for communicating a
PopupM message to a VoIP client in conjunction with an
incoming call in accordance with an illustrative embodiment;
Fig. 14 depicts a flow diagram of a method for communicating a
feedback PopupM message to a computer/phone in
conjunction with an outgoing communication in accordance
with an illustrative embodiment;
Fig. 15 depicts a flow diagram of a method for communicating a
feedback PopupM message to a computer/phone in
conjunction with an incoming communication in accordance
with an illustrative embodiment;
Fig. 16 depicts a flow diagram of a method for alias dialing in
accordance with an illustrative embodiment.
DETAILED DESCRIPTION OF THE INVENTION
NOTATIONS AND NOMENCLATURE
2o AudioMT An audio message that may be sent to an "ear piece" of an aural
communication device before dial tone.
AudioTM An audio message that may be sent to an "ear piece" of an aural
communication device after or during dial tone.
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AudioICM An audio message that may be sent to an "ear piece" of an aural
communication device after an incoming call has been received.
TextICM A Textual message that may be sent to a text display of an aural
communication device before an incoming call; may also include
image data.
Ring Back Tone A special tone sent to a call originator that a destination
aural
communication device is ringing.
to
SLIC Subscriber line interface circuit.
CPE A customer premise equipment such as router, phone, key
telephone system, FAD, modem, or broadband Integrated Access
Device (IAD) etc. These devices may be used to connect
subscribers to service providers' networks.
PopupM A window or other display element in a computer system
comprising a multimedia message.
Subscriber A user who subscribes to a particular service provided by a service
provider, e.g. a cellular phone subscriber, a Internet subscriber.
VoIP "VoIP" stands for Voice over IP (Internet Protocol). It refers to a
particular type of packetized voice communication technology that
carries voice over packet oriented networks) using the IP.
VoIP Gateway A Gateway which allows the devices using VoIP technology to
inter-operate with traditional telephones and vice versa.
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Etherphone "Etherphone" refers to a device that behaves like a telephone, but
it
is connected to a packet transport medium. Typically, the Internet
Protocol is used; etherphones typically use a VoIP protocol for
voice communication. Also known as IP phone.
Multimedia message In addition to a standard meaning, the term 'multimedia
message' includes, without limitation, single-media non-audio
messages.
l0 DSLAM Digital Subscriber Line Access Multiplexor.
DESCRIPTION OF FIGURES
Fig. lA depicts a exemplary network architecture 1000 in which illustrative
embodiments invention may operate. An Internet-type network 1100 is shown.
Illustrative embodiments of the invention work with the Internet. Other
networks
using the Internet Protocol ("IP"), either public or prig ate may also be
used, as may
networks which do not use the IP.
Addressable in the Internet-type network 1100 are a first message server 1110
and a second message server 1120. The first message server 1110 and the second
message server 1120 generally function, at least, to store text, audio, or
multimedia
2o messages that may be transmitted in accordance with versions of invention.
Further,
the first message server 1110 and the second message server 1120 may provide
management and control of the messages sent to subscribers. In some
embodiments
of the invention they gather statistics respecting messages transmitted to
subscribers.
The first message server 1110 or the second message server 1120 may be
computing
machinery of a general nature configured to act as a server in client-server
communications. An illustrative server may be a SPARC-based workstation from
SUN MICROSYSTEMS of Mountain View, CA running the SOLARIS operating
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system and the Apache HTTPd server software, e.g., a "web server."
Conveniently,
the first message server 1110 or the second message server 1120 may be located
at a
service providers' Point of Presence. One of skill will recognize that
versions of the
invention may operate with one or several servers.
Further illustrated is a public switched telephone network 1200, commonly
known as "PSTN." Shown as part of the public switched telephone network 1200
are
a first edge switch 1210 and a second edge switch 1240. The first edge switch
1210
and the second edge switch 1240 typically are able to route voice calls to the
public
switched telephone network 1200 and Internet traffic to the Internet-type
network
1100 including, for example, to the first message server 1110 or the second
message
server 1120. As part of the public switched telephone network 1200, the first
edge
switch 1210 is connected with a first central office stnitch 1220 and the
second edge
switch 1243 is connected with a second central office s witch 1230. Exemplary
structures for the first edge switch 1210 or the second edge switch 1240
include, for
example, Intelligent DSL Multiplexors, Multiservice Switches, ATM Edge
Switches,
Cable Modem Head-end Equipment and analogous equipment recognized as suitable
in the art. Structures suitable for the first central office switch 1220 or
the second
central office switch 1230 include, for example, a earner grad~:~ .~~3X switch
and/or
SS7 switch.
Referring now to the first edge switch 1210 as illustrative of the second edge
switch 1240 as well, the first edge switch 1210 is connected to a first CPE
device
1400. The first CPE device 1400 may be an apparatus embodying aspects of the
invention as will be described below. Typically functions of the first CPE
device
1400 in accordance with illustrative embodiments include, access to the public
switched telephone network 1200 and the Internet-type network 1100 and control
of
one or more phone lines.
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Shown connected with the first CPE device 1400 are a first set of aural
communication devices 1500 including a first POTS phone 1510, a first coupled
PC
& phone 1520, and a first VoIP client 1530. The first coupled PC & phone 1520
or
the first VoIP client 1530 may communicate with the first CPE device 1400 via
a first
LAN 1540. Any of the first set of aural communication devices 1500 may be used
with the first CPE device 140~r in accordance versions of the invention as
will be
described below.
Also shown are a second set of aural communication devices 1600 including a
second POTS phone 1610, a second coupled PC & phone 1620, and a second VoIP
1o client 1630. The second coupled PC & phone 1620 and the second VoIP client
1630
may comrr~anicate with a second CPE device 1300 via a second LAN 1640. A first
user may use one of the first set of aural communication devices 1500 to
initiate a call
to, or receive a call from, a second user operating one of the second set of
aural
communication devices 1600. During the period in which the first user and the
is second user operate their respective aural communication devices, either or
both may
be transmitted an informational message by their respective CPE devices in
accordance with versions of the invention as will be described below.
Fig. lA also depicts a set of aural communication devices directly connected
to a central office switch 1700 including a third POTS phone 1710 which also
may be
2o transmitted an informational message in some versions of the invention.
In some versions of the invention, the first CPE device 1400 is an integrated
access device ("IAD") while in others it may be that, for example, the first
central
office switch 1220 has been suitably modified to perform functions and embody
features of the invention. One of skill in the art having the benefit of this
disclosure
25 will recogr_ize numerous structures of programmed or programmable logic
capable of
being suitably configured to perform functions according to versions of the
invention.
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Some versions of the invention could readily operate with VoIP Gateways, and
it is contemplated that the invention will be independent of the underlying
protocols
used with the VoIP devices and VoIP Gateways Fig. 1C depicts additional
features
of the exemplary network architecture in which the invention may operate in
accordance with illustrative embodiments. For clarity of organization, some
features
of the exemplary network architecture 1000 described in connection with Fig.
lA are
also depicted in Fig. 1 C. Fig. 1 C further illustrates a first VoIP gateway
1800 and a
second VoIP gateway 1825 that are communicatively connected with the first
central
office switch 1220 and the second central office switch 1230, respectively.
Connected
between the VoIP Gateway and the subscribers may be a transport medium
multiplexor, such as a DSLAM, Modem Banks, or Cable Head-end equipment.
Alternatively, these functions may be integrated in the VoIP Gateway. For
convenience, a DSLAM is used in this illustrative embodiment, as the
particular
transport medium multiplexor is not fundamental to the illustrative
embodiment. A
first DSLAM 1850 and a second DSLAM 1875 are also depicted. The first DSLAM
1850 may communicate with the first Vole gateway 1800 and the first message
server
1110; the second DSLAM 1875 may communicate with the second VoIP gateway
1825 and the second message server 1120.
Illustrative VoIP embodiments may be appreciated with reference to Fig. 1C
by one of skill in the art who, also having the benefit of this disclosure,
will
apprehend how various other embodiments could be practiced. A converting
computer 1555 is depicted as well as a POTS phone connected through a
converting
computer I 550. The converting computer 1555 may be configured to convert the
signaling of a POTS phone to VoIP transmissions for communication to the first
VoIP
gateway 1800 via the first DSLAM 1850.
Also depicted is a first VoIP client programmed computer 1560. The first
VoIP client programmed computer 1560 may be connected to a first DSL equipment
1565 and t?:e first DSL equipment 1565 may connect to the first DSLAM 1850.
The
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first DSL equipment 1565 may be a DSL modem, or suitable customer premises
equipment configured to perform the functions of a DSL modem. A user may use a
speaker and microphone operatively coupled with the first VoIP client
programmed
computer 1560 to engage in VoIP communications according to many software and
communications protocol configurations. Some embodiments use the NetMeeting
application from Microsoft Corp. of Redmond, WA, U.S.A. that may employ the
H.323 protocol, many other suitable applications are commercially available.
Additionally depicted is a VoIi' PC & phone 1570 connected to the first DSL
equipment 1565. Elements of the VoIP PC & phone 1570 may be used separately or
to in conjunction.
In some embodiments, a VoIP phone 1665 is connected to a second DSL
equipment 1660. The second DSL equipment 1660 is, in turn, connected to the
second DSLAM 1875. In other embodiments, a POTS phone 1670 is connected to a
VoIP converter 1675 that converts the signaling to VoIP transmissions. The
VoIP
converter 1675 may then be connected to the second DSL equipment 1660. A
second
VoIP programmed computer 1680 is also depicted and illustrates that various
embodiments of the invention may be employed with one, or plural, parties to a
aural
communication communicating with VoIP systems.
Fig. 1D depicts an exemplary cellular network architecture 1900 in which the
2o invention may operate in accordance with an illustrative embodiment. A
mobile unit
1905 is shown. The mobile unit 1905 may be, for instance, a conventional
cellular
phone that supports text communication, personal digital assistant or computer
with
cellular communication capabilities, or other cellular-type communication
device that
supports text communication.
Several base transceiver stations 1903 provide a connection interface for the
mobile unit 1905. The base transceiver stations 1903 are controlled by a base
station
controller 1910 ("BSC") that is responsible for call hand over operations.
Further
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depicted is a mobile switching center 1920 ("MSC") that is responsible for
setting up,
managing, and clearing connections, as well as routing calls to an appropriate
cell site.
The mobile switching center 1920 also provides linkage to a signal transfer
point 1980
("STP") that may use SS7 protocols to provide signaling information required
to
establish circuit connections and disconnection with the public switched
telephone
network 1200, as well as database retrieval from, and data sharing with,
another
entity. The mobile switching center 1920 communicates with cell sites through
the
base station controller 1910.
The mobile switching center 1920 further communicates with a home local
1 o register 1930 ("HLR"), a visitor location register 1950 ("VLR"), and an
operation and
maintenance center 1940 comprising an equipment identity register 1960 ("EIR")
and
an authentication center 1970. The home local register 1930 is a database that
stores
the subscriber information for all subscribers within the home service area of
the
service provider. The visitor location register 1950 stores information about
visiting
15 subscribers who are not in their home service area including, for example,
roaming
number information, so that subscribers may use their cellular phones while in
another
city. The equipment identity register 1960 stores the identification serial
number of
all cellular telephones activated within the coverage area whilw the
authentication
center 1970 stores a security key embedded into cellular phones. Both the
equipment
2o identity register 1960 and the authentication center 1970 are used for user
authentication, activation and detection, and other operational and
maintenance
purposes.
Further depicted in Fig. 1D is a message server 1990 such as the first message
server 1110 or the second message server 1120. The message server 1990 may be
25 collocated with the home local register 1930 so that when a user places a
call from the
mobile unit 1905 an identifier, such as the serial number in a cellular phone
or phone
identification, is transmitted from the mobile unit 1905 to one of the base
transceiver
stations 1903. This information is then passed to the mobile switching center
1920.
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When the message server 1990 is collocated with the home local register 1930,
The
mobile swi aching center 1920 may also communicate v~ith the message server
1990
after the mobile switching center 1920 validates the calling phone number with
the
home local register 1930. Operation of aspects of the invention that may
employ the
exemplary cellular network architecture 1900 is discussed in greater detail
below in
connection with Fig. 9 and Fib. 10.
Various embodiments of the invention provide a telephone switching systems
such as, for instance, key telephone switching systems, PBX systems, VoIP
Gateways
or the like. In accordance with various embodiments of the invention, a
conforming
1o telephone switching system may operate, for instance, with a PSTN, a
Cellular
network, or a Cable Network. One embodiment illustrative of various features
of the
invention is a device that has the characteristics of a key telephone system,
an Internet
access muter, and a voice over IP gateway. However, to one of skill in the
art, the
teachings ~Sf this disclosure set fouth other embodiments as well; and various
15 alterations, modifications, omissions of elements are also within the grasp
of one
skilled in the art having the benefit of this disclosure. As one of skill in
the art will
recognize from the following descriptions, an apparatus with hardware and
software
architecture such as that of the illustrative versions of the invention
described below
may be an IAD. In addition to 8 telephone connections, and an Ethernet LAN
2o connection towards the customer side the IAD may have a T1/E1 interface or
a HDSL
interface to a network. The apparatus may act as a small key telephone system,
providing all the necessary functions of conventional plain-old telephone
usage and a
router for access to the Internet-type network 1100.
As noted, in some embodiments the first CPE device 1400 is an IAD and Fig.
25 2 illustrates a block diagram of a hardware architecture of an illustrative
IAD
embodying aspects of the invention. The hardware arc.iitecture 2000 includes a
microprocessor subsystem 2100, a memory subsystern 2200, a network interface
subsystem 2300, a set of subscriber interfaces 2400, and a set of user
interfaces 2500.
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The microprocessor subsystem 2100 comprises a MPC860 microprocessor
2110, an address bus 2120, a data bus 2130, a memory controller 2140 and an
DMA
controller 2150. The microprocessor subsystem 2100 typically also includes
conventional supporting circuitry including, for example, a power, reset,
debug port,
and clock circuits (not shown). The MPC860 microprocessor 2110 is a
commercially
available MPC860 microprocessor from Motorola. Conveniently, the MPC860
includes an integrated Ethernet Media Access Controller ("MAC"). Embodiments
that use the MPC860 conveniently configure the data bus 2130, the memory
controller
2140, and the DMA controller 2150 according to Motorola's SAMBA Reference
1o Design Board for the MPC860 processor.
The memory subsystem 2200 is communicatively coupled with the
microprocessor subsystem 2100. The memory subsystem 2200 comprises an DRAM
subsystem 2210 and a flash memory subsystem 2220. The flash memory subsystem
2220 may be implemented with four Advanced Micro Devices 29F800 (1M x 8) flash
memory devices arranged as a 32-bit wide memory bank. The MPC860
microprocessor 2110 may access the memory subsystem 2200 through a glueless
interface consisting of signals from the address bus 2120 and the data bus
2130, and
control signals from the memory controller 2140. These signals are connected
to the
flash memory devices through a set of 3.3V-SV voltage translation buffers (not
2o shown), to alleviate signal loading on the MPC860 microprocessor 2110. The
MPC860 microprocessor 2110 accesses to the flash memory subsystem 2220 are
controlled by chip-select CSO of the memory controller 2140.
The network interface subsystem 2300 is communicatively coupled with the
microprocessor subsystem 2100. The network interface subsystem 2300 includes a
field programmable gate array 2310 and a set networking ports 2350 that
includes an
HDSL port interface 2320, a T1/E1 port interface 2330.
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The field programmable gate array 2310 manages multiplexing and routing of
voice and data streams (or combined voice and data streams) over one or more
of the
set networking ports 2350. The T1/E1 port interface 2330 may be implemented
with
a Dallas Semiconductor DS2152 Enhanced T1 Transceiver device available from
Dallas Semiconductor of Dallas, Texas, U.S.A.. The DS2152 may be configured to
support a variety of formats such as D4, ESF, and SLC-96. An external
processor
such as the MPC860 microprocessor 2110 accesses the DS2152's internal control
and
status registers through its parallel bus interface (not shown). The interface
to the
MPC860 microprocessor 2110's bus may be in the Motorola non-multiplexed mode.
l0 In alternative embodiments in which an E1 interface is desired, the T1/El
port
interface 2330 may be implemented with a Dallas Semiconductor DS2154.
A suitable structure for the HDSL port interface 2320 is an HDSL chip set
from Metalink that comprises one MtH2430BL, two MtH2410AL and two MtH2445
processors that may be designed according to the Metalink reference design
(all
available from Metalink, Ltd. of Tel Aviv, Israel).
In some embodiments an ADSL port interface may be added to the set
networking ports 2350. Many suitable ADSL chip sets are commercially
available,
one example is from Motorola. One of skill will recognize many other
commercially
available suitable structures and designs.
In some embodiments a Cable Modem interface may be added to the set
networking ports 2350 (or optionally in lieu of the HDSL port interface 2320).
A
suitable Cable Modem interface may be implemented with, for instance, the
BCM330-(OAMLINK chipset from Broadcom Corp. of Irvine, California.
Conveniently, in embodiments comprising a Cable Modem interface, a device so
configured may communicate with Cable head-end equipment. When configured for
communication with Cable head-end equipment features of the invention may
conveniently be embodied in ,a 'set-top box'.
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The set of user interfaces 2500 comprises a set of subscriber interfaces 2400,
a
RS-232 port 2510, an ethernet transceiver 2430, and a UTP port 2440. The set
of
subscriber interfaces 2400 comprises a pair of QSLACs 2410, a set of four
SLICs
2420, and a set of four DTMF detectors 2450. The set of user interfaces 2500
may
provide the function of a "plain old telephone system" (POTS) interface, an
Ethernet
LAN interface, and an RS-232 interface.
The POTS interface function may be implemented by the pair of QSLACs
2410 which may be Advanced Micro Devices Am79Q02 QSLAC. Each of the pair of
QSLACs 2410 may drive a set of four SLICs 2420. The set of four SLICs 2420 may
to be Advanced Micro Devices Am7920 SLICs (available from Advanced Micro
Devices, Sunnyvale, California, U.S.A). The set of four SLICs 2420 are
connected to
the set of four DTMF detectors 2450 and a power supply (not shown). Each of
the set
of four SLICs 2420 provides DC current to send voice signals to a telephone
(not
shown) connected with a two-wire line. Further each of the set of four SLICs
2420
15 manages the TIP and Ring leads of the telephone line interface, detects the
off hook
status of a connected telephone, and works with the power supply and a ringing
relay
to switch tile ringing signal on or off to the connected t::lephone. Each of
the pair of
QSLACs 2410 processes digital PCM voice data into analog _:aA~zals and inputs
them
to one or more of the set of four SLICs 2420. In the transmit path, the analog
output
20 of one of the set of four SLICs 2420 is processed by the QSLAC device and
output in
serial digital format to the PCM interface.
The Ethernet LAN interface may be implemented with a LXT970 10/100
ethernet transceiver 2430 available from Level One Communications of
Sacramento,
California, U.S.A. Conveniently the ethernet transceiver 2430 may be
interfaced with
25 the integrated MAC of the MPC860 microprocessor 2110. The ethernet
transceiver
2430 communicates through the UTP port 2440. The design of the LXT970's analog
interface to the UTP port 2440 may be as in the LXT970's databook. The LXT970
ethernet transceiver 2430 interfaces to the MPC860 microprocessor 2110 via its
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Media Independent Interface, ~MII). The MPC860 microprocessor 2110 MII signals
are implemented on its Port D pins and four other pins that are defined as
spare pins
on the MPC860 microprocessor 2110.
The RS-232 port 2510 may be centered on a SMC1 communication port of the
MPC860 microprocessor 2110. The SMC1 port may ue used as a low-speed serial
link for a resident ROM monitor. The design of this port comprises a Maxim
3232
RS-232 transceiver that converts SMCI's transmit and receive signals, to RS-
232
levels.
Also depicted is a digital signal processor 2600 communicatively coupled with
1o the microprocessor subsystem 2100. The digital signal processor 2600 may
operate
with versions of the invention to provide digital audio signal generation for
particular
messages.
Fig. 3 depicts a block diagram of a software architecture 3000 that may
operate wi~i1 the hardware architecture 2000 in an illustrative version of the
invention .
15 The software architecture 3000 includes a system initialization routine
3010, a system
configuration routine 3020, a Key/PBX module 3100, an Internet communication
module 3200, a message server agent 3030, a message manager 3040, and a VoIP
manager 3070.
The system initialization routine 3010 performs conventional system
2o initialization functions. The system configuration routine 3020 allows a
user or
network operators to config~;re parameters used in conjunction with operation
of the
device such as the phone line numbers, an IP address for the device, an access
speed
to the network, etc.
In addition the system configuration routine 3020 allows configuration of a
25 subscription profile 3050. The subscription profile 3050 may specify, for
example, a
service type for users associated with the aural communication device to which
a
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device embodying aspects of the invention may communicate informational
messages.
The subscription profile 3050 may control, for example, a number of times
which an
informational message will be transmitted to an aural communication device.
The
subscription profile 3050 may also comprise a profile for a sponsor or for a
user
associated with an aural communication device to which the device transmits
informational messages. The system configuration routine 3020 may configure
the
subscription profile 3050 in addition to conventional parameters used in
conjunction
with operation of the device.
The Internet communication module 3200 includes a set of WAN device
1o drivers 3430. The set of WAN device drivers 3430 include both synchronous
or
asynchronous communications protocols. The set of WAN device drivers 3430
communicate with a set of low level device drivers 3500 that drive the
chipsets of the
T1/E1 port interface 2330, the HDSL port interface 2320, or Cable Modem
interface.
The set of WAN device drivers 3430 passes data to a frame relay services
module
15 3410 and a PPP / ML PPP services module 3420. The frame relay services
module
3410, the PPP / ML PPP services module 3420, and an Ethernet device driver
3440
provide link layer functions. A set of link support layers 3400 supports the
link layer
function and provides data to network layer modules such as an IP muter module
3360 and afi STP module 335(1. The STP module 3350 implements the Spanning
2o Tree Bridging Protocol that provides an IEEE 802-compliant MAC layer
bridging
engine containing both port state control and frame filtering/forwarding. The
IP
muter module 3360 implements the functions of a conventional Internet Protocol
muter. Suitable implementations are widely available commercially including
from
RouterWare of Newport Beach, CA, U.S.A. A TCP module 3300 provides transport
25 layer functions and several routing protocol modules, such as an OSPF
module 3310,
an EGP module 3320, an BGP module 3330, and an IGRP module 3340 exchange
routing information from the IP muter module 3360 with gateways in other
systems.
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A socket layer 3260 provides an IP address and "port" for application layer
modules to communicate with lower layers. Exemplary application layer modules
include an SNMP agent 3250, a DHCP module 3240, a TFTP/BOOTP module 3230,
a Telnet handler 3220, and an HTTP handler 3210 that operate conventionally.
The HTTP handler 3210 operates with a message server agent 3030. The
message server agent 3030 communicates with a message server such as the first
message server 1110 or the second message server 1120. In one embodiment of
the
invention, when an IAD is to be installed, an installer provides the IAD with
the URL
of a message server. Upon the IAD powering up and establishing a valid network
1o connection, the IAD communicates an HTTP Open Connection Request to the
message server. The message server transfers one or more informational
messages, to
the IAD where they are stored, for example in the memory subsystem 2200. The
message server agent 3030 may also periodically communicate an HTTP Open
Connection Request to the message server and next communicate an HTTP Post
Request to transfer logs andlor statistics gathered by tl~e IAD relating to
the
informational messages transmitted by the IAD to various aural communication
devices. In some embodiments, the message server agent 3030 may initiate a
request
to open a connection with the message server responsive to status of the
subscription
profile 3050. The message server agent 3030 may then transfer one or more
2o informational messages to the IAD.
Turning now to the Key/PBX module 3100 of the software architecture 3000,
the Key/PBX module 3100 includes a call management module 3110, a Key/PBX
driver module 3150, a dialing plan manager 3120, and a signal conversion
module
3130.
The Key/PBX driver module 3150 works with hardware described in
connection with Fig. 2. A set of device drivers 3180 operate with the pair of
QSLACs
2410, the set of four~SLICs 2420, and the set of four DTMF detectors 2450.
When the
CA 02369634 2001-10-09
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set of device drivers 3180 detect an "off hook" condition for a phone
connected to one
of the set of four SLICs 2420. the set of device drivers 3180 coordinates
operation of
a ringing circuit, assembles DTMF tone digits, and generates various tones.
The
dialing plan manager 3120 communicates with an analog signal handler 3170 and
a
voice payload handler 3160.
The analog signal handler 3170 communicates with the set of four SLICs 2420
and the pair of QSLACs 2410 to provide analog signaling handling including,
for
instance, Key Telephone System ("KTS") signaling, and Off Premise Extension
("OPX") signaling. The analog signal handling typically includes handling of
"off
hook" status of a connected analog aural communication device, such as the
first
POTS phone 1510, ringing of the phone, dialing tone generation, and ring back
tone
generation. In addition, the analog signal handler 3170 operates with the set
of four
DTMF detectors 2450 to translate the DTMF tones to dialing digits.
The voice payload handler 3160 programs the pair of QSLACs 2410 and
assembles PCM information stream into packetized form. In addition, the voice
payload handler 3160 may provide fitter buffering of the voice payload based
on an
estimated round trip delay. In some versions, the pair of QSLACs 2410 operates
with
the digital signal processor 2600 to provide echo cancellation :n the audio
pathway.
In other versions, an echo cancellation chipset such as TEC03264 from Lucent
2o Technologies of Murray Hills, New Jersey may provide echo cancellation. As
will be
described further below, some versions of the messages are decompressed and/or
undergo text-to-speech conversion. In embodiments when decompression or text-
to-
speech conversions is required, the voice payload handler 3160 coordinates the
decompression or text-to-audio conversion with the digital signal processor
2600 and
the microprocessor subsystem 2100.
The dialing plan manager 3120 provides conventional dialing management
functions, including, assembling the number of dialing digits, managing an
inter-digit
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time-out timer, managing the total duration of dialing time, and managing the
maximum number of dialing digits allowed.
The call management module 3110 generally manages the state of the
incoming and outgoing calls and provision of informational messages. For
instance,
during an incoming call in the analog case, a user picky up a handset that
triggers the
analog signal handler 3170 to indicate that the phone is off hook. This
information is
passed to the call management module 3110 via the dialing plan manager 3120.
The
call management module 3110 may then consult a message manager 3040 with the
line identif cation of the user's line to determine if an informational
message should
1o be communicated for perception by the user (as will be described in detail
below).
If a call is an extension call to another analog line under the control of the
system, the call management module 3110 generates a signal with the line
identification to the analog signal handler 3170 to indicate there is a call
for that
particular line. This triggers the analog signal handler 3170 to exert ringing
to the line.
When a user answer the phone, the analog signal handler 3170 detects this
phone as
"off hook," and informs the call management module 3110 that the call has been
answered. The call management module 3110 then cor_nects the two lines and
puts
the path to voice payload transfer mode.
The call management module 3110 additionally may consult a directory
2o database 3080 to determine how the call should be switched based on portion
(or all)
of the dialing digits. In some embodiments, the directory database 3080 is an
internal
pre-configured table that is used to route the phone calls based on hierarchy
digits.
However, this is not fundamental, and in other embodiments routing information
could resides on a server accessed through a directory protocol including, for
instance,
the Lightweight Directory Access Protocol ("LDAP"). (When LDAP is used, a
suitable protocol stack should be integrated, as one of skill will
appreciate).
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An IAD according to some embodiments of the invention supports TDM
based access to PSTN including, for instance, T1 through the T1/E1 port
interface
2330. In these embodiments, the call management module 3110 may check, for
instance, the directory database 3080 to determine if the call is should be
routed to the
PSTN. If the call should be routed to the PSTN via the Tl the T1/E1 port
interface
2330 using, for instance, CAS signaling, the call management module 3110
allocates
an appropriate time slot (if available) for the call and converts the analog
call
signaling to CAS signaling. Thereafter, the call management module 3110 places
the
call with the dialing digits as the phone number to the PSTN.
1o An additional aspect of some versions of the invention is routing of calls
over
the Internet-type network 1100 as a Voice over IP ("VoIP") call. When VoIP
call
routing is used, the call management module 3110 communicates with a VoIP
manager 3070. The call management module 3110 passes the dialing digits to the
VoIP manager 3070.
The VoIP manager 3070 then initiates the call in accordance with a suitable
VoIP standard. In some embodiments of the invention the H.323 VoIP standard is
used as promulgated by the International Telecommunication Union ("ITU") of
Geneva, Switzerland. If H.323 is desired, an H.323 protocol stack available
from
Radvision of Tel Aviv, Israel, may be used. The particular VoIP protocol used
is not
2o fundamental to the present invention and it is contemplated that the
invention will
operate with VoIP protocols other than H.323 (now known or later created)
according
to the particular commercial and/or technical needs of the situation. During a
VoIP
call, the call management, module 3110 may maintain the call status state
throughout
the call, and interact with the VoIP manager 3070 appropriately.
The VoIP manager 3070 additionally manages the conversion of internal call
control signaling to signaling format in compliance with a voice over IP
standard. In
some embodiments a Gatekeeper (as proposed in the ITU-T H.323 specification)
is
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WO 00/62518 PCT/US00/08715
not implemented. The directory database 3080 may be used in place of the
Gatekeeper as a simple database for the address conversion from dialing digits
to IP
address format; LDAP access may also be used.
In some embodiments providing VoIP, the 6.711 audio codec is used. In
addition, some embodiments employ the User Datagram Protocol ("UDP") network
layer for the speech payload transfer over the Internet-type network 1100.
The message manager 3040 may look at the line identification, which is
passed to it by the call management module 3110, and check with the
subscription
profile 3050 to determine whether an AudioMT should be sent to that particular
line
to in the case of a call initiated through a phone connected to the IAD. In
the event an
incoming call is received from the PSTN to a line connected to the IAD, the
call
management module 3110 will determine the line identification based on the
incoming call information. (E.g. in the case of DID, a specific number is
associated
with a specific line. If it is rotary, then the call management module 3110
will assign
15 the call to a particular line that is free.) The line identif cation is
sent to the message
manager 3040, which in turn checks with the subscription profiles to determine
whether an AudioICM should be sent to that line. If needed, the message
manager
3040 will retrieve the message and pass it to the call management module 3110
which
in turn passes it to the voice p;~.yload handler 3160.
2o The call management module 3110 implements message transmission features
in various embodiments of the invention. Aspects of illustrative embodiments
will be
described in connection with Fig. 4 through Fig. 16 below. Briefly here, the
call
management module 3110 operates with a message manager 3040 to coordinate the
transmission of informational messages to users of aural communication devices
25 under control by an IAD.
As will be described in detail below, an apparatus according to illustrative
versions of the invention may be, for example, the first CPE device 1400 and
may
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access message servers such as the first message server 1110 to retrieve
informational
messages v.-hich may be stored, for example in the DRAM subsystem 2210, for
later
communication to, for example, one of the first set of aural communication
devices
1500 under its control. Call management as described below allows the
apparatus to
transmit informational messages, including advertisements or the like, to
aural
communication devices under its control.
ILL USTRATIVE EMBODIMENTS
Features of various versions of the invention will now be provided for further
illustration. One of skill in the art will appreciate that the following
embodiments are
to illustrative of the invention and that various modifications, omissions,
and alterations
may be made without departin j from the scope and spirit of the invention as
set forth
in the appended claims. Reference will be made to Fig. lA-Fig. 1D in
describing
illustrative embodiments with the understanding that the illustrative
embodiments are
not limited to the configurations of Fig. lA-Fig. 1D.
15 1. PHONE CONNECTED TO PUBLIC SWITCH
INFRASTRUCTURE THROUGH A CPE
In some embodiments of the invention the first POTS phone 1510 is connected
to the public switched telephone network 1200 with the first CPE device 1400.
The
first CPE device 1400 may be an IAD with hardware and software architectures
as
2o described in connection with Fig. 2 and Fig. 3.
LA. OUTGOING COMMUNICATION
HANDLING
For purposes of illustration of outgoing communication handling, a first user
operating the first POTS phone 1510 is assumed to be making an outgoing
CA 02369634 2001-10-09
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communication to a second user operating one of the second set of aural
communication devices 1600.
Fig. 4 depicts a flow diagram of an 'outgoing communication message
transmission' method 4000 in accordance with an illustrative embodiment. The
first
CPE device 1400 may implement aspects of the 'outgoing communication message
transmission' method 4000. Processing initiates at a'start' terminal 4025 and
one of
the set of four SLICs 2420 generates an 'indication the first POTS phone 1510
is 'off
hook' 4050. The set of device drivers 3180 receive a signal form one of the
set of four
SLICs 2420 that is passed to an "off hook' detection' process 4075 in the
analog
signal handler 3170 that detects the signal and provides an indication thereof
to the
call management module 3110..
Next, an 'AudioMT to first user' decision process 4125 invol ves the call
management module 3110 consulting the message manager 3040 to determine
whether an AudioMT should be sent to the first POTS phone 1510 at this time.
The
message manager 3040 examines the line identification, which is passed to it
by the
call management module 3110, and may consult a subscription profile 3050
associated with the first POTS phone 1510 or the AudioMT to determine whether
to
send the AudioMT.
The subscription profile 3050 may comprise records commonly used by
2o advertisers or marketers in customizing their messages including, for
example, a
frequency with which the AudioMT is to be communicated, a class or group of
users
to which the AudioMT is to be communicated, and temporal limits on when the
AudioMT should be communicated. The temporal limits may indicate particular
times during the day when the AudioMT should or should not be communicated,
and
may also indicate the expiry of a 'lifespan' associated with the AudioMT.
If the AudioMT should be communicated to the first aural communication
device, the 'AudioMT to first user' decision process 4125 exits through its
'yes'
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branch to enter an'AudioMT retrieval' process 4150. The'AudioMT retrieval'
process
4150 is performed by the message manager 3040 and retrieves the AudioMT from
the
DRAM subsystem 2210.
Process flow continues to a'feedback AudioMT' decision process 4175
performed by the message manager 3040. Some embodiments of the invention
operate with informational messages that offer feedback or interactive
("feedback")
options to a user. In some embodiments, the informational messages convey to
the
user feedback options. For example, a user may hear, "Press star-1 to learn
more;
press start 2 to receive a coupon for this item, or press star-3 to speak with
an operator
to participate in a promotion . . . " Some embodiments include an indication
in a
header portion of the informational message indicating whether the
informational
message will present feedback options to the user. The 'feedback AudioMT'
decision
process 4175 detects whether the AudioMT retrieved in the 'AudioMT retrieval'
process 4150 indicates that it will present feedback options. If so, the
'feedback
AudioMT' decision process 4175 exits through its 'yes' branch and process flow
continues to a'feedback AudioMT communication' process 4300 that will be
described below in connection with Fig. 5B. If the AudioMT retrieved in the
'AudioMT retrieval' process 4150 does not indicate that it will present
feedback
options, tr~e 'feedback AudioMT' decision process 4175 exits through its 'no'
branch
and process flow continues to an'AudioMT communication' process 4200 that will
be
described below in connection with Fig. 5A.
From either the 'AudioMT communication' process 4200 or the 'feedback
AudioMT ~.ommunication' process 4300 process flow continues to a'remaining
messages' decision process 4400 that consults the message manager 3040 and
determines whether one or more additional AudioMTs should be sent. If
additional
AudioMTs should be sent, the 'remaining messages' decision process 4400 exits
through its 'yes' branch and process flow returns to the 'AudioMT retrieval'
process
4150 for another iteration. When no additional AudioMTs should be sent, the
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'remaining messages' decision process 4400 exits through its 'no' branch and
process
flow continues to the 'outgoing call handling' process 4425. The call
management
module 3110 performs the 'outgoing call handling' process 4425 and instructs
the
analog signal handler 3170 to provide an indication that the first POTS phone
1510 is
available fer connection to one of the second set of aural communication
devices
1600, typically a conventional dial tone. Ingress digits are typically then
received
from the first POTS phone 1510 and handled by the call management module 3110
to
initiate a call.
Process flow continues to an'AudioTM to first user' decision process 4450
1o that involves the call management module 3110 consulting to the message
manager
3040 deterrnine whether to send an AudioTM to the firat POTS phone 1510 in a
fashion analogous to that described above in connection with the 'AudioMT to
first
user' decision process 4125. If the AudioTM should be communicated to the
first
aural communication device, the 'AudioTM to first user' decision process 4450
exits
15 through its 'yes' branch to enter an'AudioTM retrieval' process 4475. The
message
manager 3040 performs the 'AudioTM retrieval' process 4475 that retrieves the
AudioTM from the DRAM subsystem 2210. Process flow continues to a'feedback
AudioTM' decision process 4500 that (similar to the 'feedback AudioMT'
decision
process 4175) will detect if the AudioTM is a feedback AudioTM. If so, the
'feedback
2o AudioTM' decision process 4500 exits though its 'yes' branch to enter a
'feedback
AudioTM communication' process 4600; if not, the 'feedback AudioTM' decision
process 4500 exits through its 'no' branch and process flow enters an 'AudioTM
communication' process 4700. The 'AudioTM communication' process 4700 and the
'feedback AudioTM communication' process 4600 will be described in detail
below in
25 connection with Fig. 6A and Fig. 6B, respectively.
Process flow continues to a'remaining messages' decision process 4800 that
consults the message manager 3040 and determines whether one or more
additional
AudioTMs should be sent. If additional AudioTMs should be sent, the 'remaining
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messages' decision process 4800 exits through its 'yes' branch and process
flow
returns to the 'AudioTM retrieval' process 4475 for another iteration. When no
additional AudioMTs should be sent, the 'remaining messages' decision process
4800
exits through its 'no' branch and process flow continues to the 'call
establishment
handling' process 4825.
The 'call establishment handling' process 4825 performs conventional steps to
establish t:~e communication path to one of the second set of aural
communication
devices 1600 familiar to one of skill in the art. Briefly, the dialing plan
manager 3120
assembles the number of dialing digits and passes them the call management
module
3110. The call management module 3110 then checks the directory database 3080
to
see how to route the call. Assuming, for example, the communication is to
route
through the T1/E1 port interface 2330 to the public switched telephone network
1200,
the call management module 3110 sets up a logical path and sends out a
connection
request through the call management module 3110. The call management module
3110 determines an appropriate digital interface signaling protocol. If CAS
signaling,
the call management module 3110 assigns a time slot (DSO) and invokes signal
conversion to initiate the AECD bit according to the connection signaling
protocol. If
PRI ISDN interface, the call management module 3110 perfonris according to the
PRI
ISDN specification. If the remote is not busy, the call management module 3110
2o maintains the logical path, and establishes the voice payload transfer
state. During the
communication, the call management module 3110 passes back and forth the voice
payload between the calling and called party. This typically includes the
handshakes
(such as hearing the remote phone is ringing, the remote handset is picked up -
off
hook) before the speech path is fully connected. If the remote is busy, the
call
management module 3110 generates a busy tone to the first POTS phone 1510. If
the
number is invalid the call management module 3110 generates a fast busy tone.
Process flow completes through an'end' terminal 4900.
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The 'AudioMT communication' process 4200 will now be described in greater
detail. Fig. 5A depicts a flow diagram of a method communicating an AudioMT in
accordance with an illustrative embodiment. The 'AudioMT communication'
process
4200 initiates at 'start' terminal 4205 and process flow continues to a
'compressed
message' decision process 4220 performed by the voice payload handler 3160.
The
'compressed message' decision process 4220 exits through its 'yes' branch if
the
AudioMT is in compressed form and a 'decompression' process 4230 decompresses
the AudioMT. Process flow continues to a 'text to audio' decision process 4240
that
exits through its 'yes' branch if the AudioMT is in textual form and should be
to converted to audio. A 'text to audio conversion' process 4250 performs the
conversion. The 'decompression' process 4230 and the 'text to audio
conversion'
process 4250 may operate conventionally with the digital signal processor 2600
to
perform their functions. Process flow continues to an 'AudioMT transmission'
process 4260 that transmits the AudioMT in audio form to the first POTS phone
1510
for perception by the first user. Process flow continues to a "messages sent'
flag
setting' process 4280 that sets an internal flag indicating that a message has
been sent
and process flow completes through an 'end' terminal 4290.
The 'feedback AudioMT communication' process 4300 will now be described
in more detail with reference to Fig. 5B. Some embodiments of the invention
support
2o interactive informational messages (termed "feedback"). Process flow
initiates at a
'start' terminal 4305 and continues to a'compressed message' decision process
4315
performed by the voice payload handler 3160. The 'compressed message' decision
process 4315 exits through its 'yes' branch if the AudioMT is in compressed
forth and
a'decompression' process 4320 decompresses the AudioMT. Process flow continues
to a'text to audio' decision process 4325 that exits through its 'yes' branch
if the
AudioMT is in textual form and should be converted to audio. A 'text to audio
conversion' process 4330 performs the conversion. The 'decompression' process
4320
and the 'text to audio conversion' process 4330 may operate conventionally
with the
digital signal processor 2600 to perform their functions. Process flow
continues to a
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'feedback .4udioMT transmission' process 4335 that begins transmitting the
AudioMT
in audio fo-m to the first POTS phone 1510 for perception by the first user.
Next, a 'AudioMT feedback indication' decision process 4340 detects if the
message sent to the user requires a feedback from the user. If so, then the
user is
prompted to provide a feedback indication. For instance, an AudioMT may offer
one
or more feedback options along with associated indications the user may make
to
select among the one or more feedback options. Generally, the type, quality,
and
nature of feedback options are only limited by the capabilities of the aural
communication device being used. Illustrative versions of the invention are
described
1o below; one of skill in the art will appreciate various modifications,
alterations and
adaptations of the illustrative embodiments that still lie within the spirit
and scope of
the invention as set forth in the appended claims.
In some illustrative versions, prompting for feedback input is similar to that
found with conventional voice mail systems, e.g. "press 1 for to speak with an
operator to learn more; press 2 to participate in a promotion; press 3 to
receive a
catalog; press 4 to proceed to make your original call . . . " Advertisers or
other
providers of informational messages may customize the particular message and
feedback option prompting to their liking.
When a user provides the dialing sequence for a feedback option, one of the
2o set of four DTMF detectors 2450 detects the tone sequence, converts it to
binary and
provides the binary sequence to the call management module 3110 . The 'AudioMT
feedback indication' decision process 4340 exits through its 'yes' branch and
the call
management module 3110 couples the information with a line identity, and other
information such as a message identifier, IAD identifier to form a feedback
indication
message. Process flow continues to a'feedback indication storage' process 4350
that
passes the feedback indication message to the message server agent 3030 for
communication to the message server. Such information may be logged by message
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servers, used to generate usage profiles of subscribers, used to determine
pricing of
informational messages, and other information collation, and customization
operations with which one of skill in the art is familiar.
Process flow continues to a'call redirection' decision process 4355. Some
AudioMTs may prompt the user to have a call placed ("redirected call") to a
third
party including, for instance, a sponsoring entity. Some embodiments place the
redirected call upon receipt of the user feedback indication while others
place the
redirected call after the user completes their outgoing call.
The call management module 3110 upon receiving the user feedback selection
to determines if the selection is for a redirected call. In some embodiments,
the
AudioMT also includes one or more fields comprising selections and associated
redirected call numbers. In other embodiments, the call management module 3110
may receive a response message from the message server comprising the number
to
dial for the redirected call. Still other embodiments nse "alias dialing".
With alias
15 dialing the user may prompted to use a word, mnemonic, acronym, etc.
created by the
letters found on telephone keypad, e.g. "press star ITX to speak with the ITX
Networks operator" where "star ITX" corresponds to the "*", "4", "8", "9" keys
on a
conventional telephone keypad. The alias may correspond to the number that
should
be dialed to complete the redirected call.
2o In still other embodiments, alias dialing may be used whenever the user
begins
using their aural communication device. In these embodiments, when the user
places
their aural communication device "off hook" and begins to enter a dialing
sequence,
the call management module 3110 in an IAD, CPE, or central office switch can
detect
an alias from the dialing sequence and place a call to a target number for
which the
25 alias is word, mnemonic, acronym etc. The conversion from the alias name to
the
target telephone number may be a simple table look up, or a directory call
conversion
or a message exchange with the message server. This feature allows a feedback
37
and the 'text to
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message to prompt a user, for instance, "Dial star ITX at any time to reach
ITX
Networks" and the user may pick up their aural communication device at some
time
and use this alias to reach ITX Networks. Further description of alias dialing
is
provided below with reference to Fig. 16.
Returning to Fig. SB, the 'call redirection' decision process 4355 exits
through
its 'yes' branch if the call management module 3110 determines that the
selected
feedback involves a call being placed to a third party. A'redirected call'
process 4360
executes in which the call management module 3110 places the call.
If the 'feedback AudioN(T transmission' process 4335 determines that the
to selected feedback does not involve a call being placed to a third party and
process
flow continues to an 'automated output' decision process 4365.
The 'automated output' decision process 4365 determines whether the selected
feedback option involves automated output generation. If so process flow
continues
to an 'output generation' process 4370. The 'output generation' process 4370
involves
15 the call management module 3110 posting to the message server the feedback
selection and also comprises the message server generating the automated
output.
In some embodiments, the automated output may be axn email message sent to
an address associated with the user of the aural comm;mication device. The
message
may be, for instance, a promotion, coupon, or other information inviting a
commercial
2o transaction. The automated output may be mail sent by conventional postal
methods,
in which case the message server would provide an indication to, for instance,
a
human operator to send the mail. In still other embodiments, the automated
output
may be a coupon redeemable at a merchant. The message server may provide the
information necessary to prepare the coupon directly to the merchant for the
user to
25 pick up from the merchant and redeem. The user may be provided (via their
aural
communication device, electronic mail, or otherwise) with an identification
number or
personal code to identify himself or herself at the merchant in order to
redeem the
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coupon . This feature of some embodiments of the invention allows the great
volumes
of conventional commercial mail in paper form sent to residences to be
substituted
with coupon provision according to embodiments disclosed herein. In still
other
embodiments, the automated output is not sent to the user but may be sent to
another
person. In yet other embodiments, the automated output may be a next
informational
message sent to the user's aural communication device. The next informational
message may prompt the user to select from additional feedback options.
From the 'output generation' process 4370 process flow continues to an 'output
communication' process 4375 in which the output from the 'output generation'
process
to 4370 is communicated to the appropriate recipient. From the 'output
communication'
process 4375 or the 'redirected call' process 4360 process flow continues to a
'remaining feedback' decision process 4380. The feedback options presented to
a user
may provide for a next feedback option selection after the user has selected a
first
feedback option and completed a first feedback process. If the 'remaining
feedback'
15 decision process 4380 detects that a next feedback option has been
selected, it exits
through its 'yes' branch and a'next feedback message selection' process 4385
executes. The 'next feedback message selection' process 4385 involves the call
management module 3110 consulting a message server for the next feedback
message.
Process flow continues to a'next feedback message communication' process 4390
in
2o which the next feedback message is transmitted to the aural communication
device
and process flow returns to the 'feedback indication storage' process 4350 for
another
iteration.
When the 'remaining feedback' decision process 4380 exits through its 'no'
branch, indicating no remaining feedback options are to be presented, or if no
25 feedback indication is required during the 'AudioMT feedback indication'
decision
process 4340, process flow continues to a'messages sent flag setting' process
4395
that sets an internal flag that the AudioMT has been sent and process flow
completes
through an 'end' terminal 4399.
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The 'AudioTM communication' process 4700 will now be described in greater
detail. Fig. 6A depicts a flow diagram of a method communicating an AudioTM in
accordance with an illustrative embodiment. Process flow initiates at a
'start'
terminal 4705 and continues to a 'compressed message' decision process 4715.
If the
AudioTM is compressed, a 'compressed message' decision process 4715 exits
through its 'yes' branch and a 'decompression' process 4720 decompresses the
AudioTM. Process flow continues to a 'text to audio' decision process 4725
that exits
through its 'yes' branch if the AudioTM is in text form and should be
converted to
audio. A 'text to audio conversion' process 4730 performs the conversion. The
'text
to audio conversion' process 4730 and the 'decompression' process 4720 may
operate
conventionally with the digital signal processor 2600 to perform their
functions.
Process flow continues to a 'AudioTM transmission' process 4740 that begins
transmission of an AudioTM in audio form to the first POTS phone 1510 for
perception by the first user.
Next, a 'destination ring back tone detected' decision process 4745 exits
through its 'yes' branch if the ring back tone has been detected from one of
the second
set of aural communication devices 1600. If so, a 'set Ring Back Tone flag'
process
4750 sets a flag indicating that the ring back tone has been detected. Process
flow
continues to a 'destination answered' decision process 4755 that exits through
its 'yes'
2o branch if the call has been answered by one of the second set of aural
communication
devices 1600.
If the 'destination ar~;~wered' decision process 4755 exits through its 'yes'
branch, a 'set Call Answered flag' process 4760 sets a flag indicating that
the call has
been answered. Process flow continues from here to a 'set Messages Sent flag'
process 4770 that sets a flag indicating that a message has been sent and
process flow
completes through an 'end' terminal 4790.
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If the 'destination answered' decision process 4755 exits through its 'no'
branch, process flow continues to a 'Ring Back Tone flag set' decision process
4765
that determines if the flag indicating that a ring back tone has been detected
is set. If
so, the 'Ring Back Tone flag set' decision process 4765 exits through its
'yes' branch
and a mixed AudioTM and Ring Back Tone transmission' process 4780 acoustically
mixes the ring back tone and the AudioTM for transmission to the first POTS
phone
1510 and process flow returns to the 'AudioTM transmission' process 4740. If
the
'Ring Back Tone flag set' decision process 4765 determines that the flag
indicating
that a ring back tone has been detected is not set, it exits through its 'no'
branch and
l0 process flow continues to a 'time expired' decision process 4775. The 'time
expired'
decision process 4775 determines whether a time limit in which the line is
permitted
to ring has expired. No particular time limit is fundamental to the features
of the
invention. If the time limit has not expired, process flow continues to the
'AudioTM
transmission' process 4740 to continue transmission of the AudioTM. When the
time
limit expires, the 'time expired' decision process 4775 exits through its
'yes' branch
and process flow completes through the 'end' terminal 4790.
The 'feedback AudioTM communication' process 4600 will now be described
in more detail with reference to Fig. 6B. The 'feedback AudioTM communication'
process 4600 is similar is the 'feedback AudioMT communication' process 4300.
2o Process flc w initiates at a 'start' terminal 4605 and continues to a
'compressed
message' decision process 4615 performed by the voice payload handler 3160.
The
'compressed message' decision process 4615 exits through its 'yes' branch if
the
AudioTM is in compressed form and a'decompression' process 4620 decompresses
the AudioTM. Process flow continues to a'text to audio' decision process 4625
that
exits through its 'yes' branch if the AudioTM is in textual form and should be
converted to audio. The 'text to audio conversion' process 4630 performs the
conversion. The 'decompression' process 4620 and the 'text to audio
conversion'
process 4630 may operate with the digital signal processor 2600. Process flow
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continues to a 'feedback AudioTM transmission' process 4635 that transmits the
AudioTM in audio form to the first POTS phone 1510 for perception by the first
user.
Unlike the 'feedback AudioMT communication' process 4300, in the 'feedback
AudioTM communication' process 4600, when the 'feedback AudioTM transmission'
process 4635 executes and user feedback processing occurs, the dial tone to
the aural
communication device conveniently is suppressed. This is indicated by a'dial
tone
suspension' block 4637.
Process flow within the 'dial tone suspension' block 4637 is similar to that
in
the 'feedback AudioMT communication' process 4300.
A 'feedback indication' decision process 4640 detects whether the message
sent to the user requires a feedback from the user, then the user will be
prompted for
the feedback. When a user provides the dialing sequence for a feedback option,
one
of the set of four DTMF detectors 2450 detects the tone sequence, converts it
to
binary and provides the binary sequence to the call management module 3110 .
The
'feedback indication' decision process 4640 exits through its 'yes' branch and
the call
management module 3110 couples the information with a line identity, and other
information such as a message identifier, IAD identifier to fo~v~3 a feedback
indication
message. Process flow continues to a'feedback indication storage' process 4650
that
passes the feedback indication message to the message manager 3040 for
2o communication to the message server. Such information may be logged by
message
servers, used to generate usage profiles of subscribers, used to determine
pricing of
informational messages, and other information collation, and customization
operations with which one of skill in the art is familiar. Process flow
continues to a
'call redirection' decision process 4655 that exits through its 'yes' branch
if the call
management module 3110 determines that the selected feedback involves a call
being
placed to a third party. A 'redirected call' process 4660 executes in which
the call
management module 3110 places the call.
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If the 'call redirection' decision process 4655 determines that the selected
feedback does not involve a call being placed to a third party process flow
continues
to an 'automated output' decision process 4665. The 'automated output'
decision
process 4665 determines whether the selected feedback option involves
automated
output generation. If so process flow continues to an 'output generation'
process 4670.
The 'output generation' process 4670 involves the cali management module 3110
posting to the message server the feedback selection and also comprises the
message
server generating the automated output. The automated output may be, for
instance,
of the type described above in connection with the 'feedback AudioMT
1o communication' process 4300.
From the 'output generation' process 4670 process flow continues to an 'output
communication' process 4675 in which the automated output generated by the
'output
generation' process 4670 is :;ommunicated to the appropriate recipient. From
the
'output communication' process 4675 or the 'redirected call' process 4660
process
flow continues to a'remaining feedback' decision process 4680. The feedback
options
presented to a user may provide for next feedback option selection after
having
selected a first feedback option and completed a first feedback process. If
the
'remaining feedback' decision process 4680 detects that a next feedback option
has
been selected, it exits through its 'yes' branch and a 'next feedback message
selection'
2o process 4685 executes. The 'next feedback message selection' process 4685
involves
the call management module 3110 consulting a message server for the next
feedback
message. Process flow continues to a'next feedback message communication'
process
4690 in which the next feedback message is transmitted to the aural
communication
device and process flow returns to the 'feedback indication storage' process
4650 for
another iteration.
When the 'remaining feedback' decision process 4680 exits through its 'no'
branch, indicating no remaining feedback options are to be presented, or if no
feedback indication is required during the 'feedback indication' decision
process 4640,
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process flow continues to a'messages sent flag setting' process 4695 that sets
an
internal flag that the AudioMT has been sent. Process flow continues to a
'dial tone
generation' process 4697 where the call management module 3110 issues a
command
to provide dial tone to the user. The user may then enter a dialing sequence
and
process flow completes through an'end' terminal 46~J9.
The various feedback and options and embodiments discussed above in
connection with the 'feedback ~.udioMT communication' process 4300 are equally
operable with the 'feedback AudioTM communication' process 4600. That
description
is applicable to the 'feedback .AudioMT communication' process 4300 and is
provided
1o by referring the reader to the discussion above.
l.B. INCOMING COMMUNICATION
HANDLING
For purposes of illustration of incoming communication handling a second
user operating the second POT'S phone 1610 is assumed to be receiving an
incoming
15 communication from a first user operating one of the first set of aural
communication
devices 1500.
Fig. 7 depicts a flow diagram of an AudioICM communication method 7000.
The second CPE device 1300 implements aspects of AudioICM communication
method 7000 in this illustrative embodiment.
20 Process flow initiates at a 'start' terminal 7010 and continue] to receive
an
'incoming call' data block 7020 indicating that an incoming communication has
been
received from the public switched telephone network 1200. Next, a 'ring and
ring
back tone generation' process 7030 activates one of tl~~ set of four SLICs
2420 in the
second CPE device 1300 to ring the second POTS phone 1610. The 'ring and ring
25 back tone generation' process 7030 may also generate a ring back tone to
the one of
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the first set of aural communication devices 1500 initiating the incoming
communication to indicate than the receiving phone is ringing.
Next, a 'call answered' decision process 7040 exits through its 'no' branch if
the call has not been answered by the second POTS phone 1610 and a 'ring
limit'
decision process 7050 determines if a number of allowable rings has been
exceeded.
If not, the 'ring limit' decision process 7050 exits through its 'no' branch
and process
flow returns to the 'call answered' decision process 7040. If the allowable
number of
ring has been exceeded, the 'ring limit' decision process 7050 exits through
its 'yes'
branch to enter a 'set No Answer flag' process 7060. The 'set No Answer flag'
process 7060 sets a flag indicating the call was not answered and the AudioICM
communication method 7000 completes through an 'end' terminal 7070.
If the call is answered by the second POTS phone 1610, the 'call answered'
decision process 7040 exits through its 'yes' branch and processing continues
to an
'AudioICM to second user' decision process 7080. The 'AudioICM to second user'
decision process 7080 may consult a subscription profile to determine whether
an
AudioICM should be sent as has been described above in connection with
AudioMTs
and AudioTMs. If the AudioICM should not be transmitted, the 'AudioICM to
second user' decision process 7080 exits through its 'no' branch and the
AudioICM
communication method 7000 completes through the 'end' terminal 7070.
2o If an AudioICM is to be transmitted, the 'Aud.ioICM to second user'
decision
process 7080 exits through its 'yes' branch and a 'retrieve AudioICM' process
7090
retrieves the AudioICM from the memory subsystem 2200. Next, if the AudioICM
is
compressed, a 'compressed message' decision process 7100 exits through its
'yes'
branch and a 'decompression' process 7110 decompresses the AudioICM. Process
flow continues to a 'text to audio' decision process 7120 that exits through
its 'yes'
branch if the AudioICM is in text form and should be converted to audio. A
'text to
audio conversion' process 7130 performs the conversion. The 'decompression'
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process 7110 and the 'text to audio conversion' process 7130 may operate
conventionally with the digital signal processor 2600 to perform Their
functions.
Process flow continues to an 'AudioICM transmission' process 7140 that
transmits
the AudioICM in audio form to the second POTS phone 1610 for perception by the
second user and begin a next iteration.
Process flow continues to a 'remaining messages' decision process 7150. If a
next AudioICM is not to be transmitted the 'remaining messages' decision
process
7150 exits through its 'no' branch and the AudioICM communication method 7000
completes through the 'end' terminal 7070. If a next message is to be
transmitted the
'remaining messages' decision process 7150 exits through its 'yes' branch and
a
'continue Ring Back Tone' process 7160 executes. In some versions, the
'continue
Ring Back Tone' process 7160 provides the Ring Back Tone to the originator of
the
incoming communication. In some other versions, the 'continue Ring Back Tone'
process 71 GO sends a call connected message to the switching system to
indicate that
the incoming communication has been accepted and thereafter a message may be
sent
to the first user indicating the first user that his or her call is connecting
with the
second user. The message indicating that his or her call is connecting with
the second
user may be substituted by, or accompanied with, an informat=~~nal message
such as
has been described above in connection with Part LA. From the 'continue Ring
Back
2o Tone' process 7160 process flow returns to the 'retrieve AudioICM' process
7090 to
retrieve the next AudioICM.
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2. VOICE OVER IP CLIENT, ETHERPHONE,
OR REGULAR PHONE CONNECTED
THROUGH CONVERTER TO FUNCTION
AS A VOICE OVER IP PHONE
CONNECTED THROUGHA VOIP
GATEWAY
2.A. GATEWAYAT CUSTOMER
PREMISES
In some embodiments of the invention the first VoIP client 1530 is connected
1o to the Internet-type network 1100 through a gateway located at a customer's
premises.
In these embodiments a conventional gateway is modified to include programmed
or
programmable logic for the transfer of message files from a message server to
the
VoIP gatev: ay (and suitable memory for storage of the transferred files). In
addition,
the call management module 3110 is modified to provide suitable incoming and
outgoing communication handling of AudioT'M and AudioICM messages. Handling
of AudioMT messages is analogous to the processes described in connection with
Fig.
4, Fig. SA, and Fig. 5B.
Fog purposes of illustration of outgoing communication handling a first user
operating the first VoIP client 1530 is assumed to be sending an outgoing
2o communication to a second user operating one of the second set of aural
communication devices 1600.
Some embodiments of the invention handle VoIP communication transport
(both incoming and outgoing). Some of these embodiments may comprise the
hardware architecture 2000 of Fig. 2 and the software architecture 3000 of
Fig. 3 in an
IAD. Others of these embodiments may comprise software configuring
conventional
central office switching hardware or PBX systems to perform functions of (all
or part
of) the hardware architecture 2000 and the software architecture 3000.
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Some VoIP handling embodiments use the ITU H.323 protocol, although as
noted abo~,~e, the particular protocol is not fundamental. One of skill art,
having the
benefit of this disclosure, will appreciate how features and advantages of the
invention
may be obtained with other VoIP protocols as well. For illustration, H.323
Gatekeepers will be not be discussed as they, too, are not fundamental to
practicing
the invention.
Some VoIP handling embodiments perform functions of conventional VoIP
Gateway communication handling. In some embodiments of the invention, ITU-T
Q.931 is the ISDN user network interface layer 3 specification for basic call
control.
1o When a Gateway is implemented between an IP infrastructure and the public
switched
telephone network 1200, the H.225 call signaling may be converted to the
appropriate
Q.931 signaling when a PRI ISDIvT digital interface is used: Such an
implementation
may use T1 CAS signaling to interface with PSTN as per the T1 specification
(AT &
T PUB 43801).
15 When an endpoint (a first user) initiates a call, call setup takes place
using the
call control messages defined in H.225. A receiving endpoint sends an Alerting
message to indicate that the called party (a second user) has been alerted of
an
incoming call. In the case of a Gateway, the receiving endpoint sends the
Alerting
message when it receives a ring indication from the PSTN. In accordance with
the
2o current H.323 specification, if the endpoint can respond to a Setup message
with a
Connect, Call Proceeding, or Release Complete within 4 seconds, it need not
required
to send the Alerting message. An end-point sending the Setup message can
expect to
receive either an Alerting, Connect, Call Proceeding, or Release Complete
message
within 4 seconds after successful transmission.
25 With reference to Fig. 3, the VoIP manager 3070 receives the Call Setup
Message from a Transport Service Access Point. In some embodiments, the
Transport
Service Access Point is to the Internet-type network 1100 via a dedicated
Frame Relay
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connection. The IP packets may be encapsulated in the Frame Relay packet
according
to RFC 1490. This then goes through the data protoco~ stack and becomes an IP
packet, i.e. a Call Setup Message to the VoIP manager 3070. The VoIP manager
3070
may then interact with the call management module 3110 to determine if there
is a
POT line available for this incoming call. If there is a line available for
the incoming
call, the VoIP manager 3070 sends a Call Proceeding message to the client
endpoint.
Fig. 8A depicts a flow diagram for a method for communicating an AudioTM
message to a VoIP client or Etherphone in accordance with an illustrative
embodiment. Process flow initiates at an 'start' terminal 8100 in response to
an
to incoming call signal from the first VoIP client 1530 being detected by the
gateway.
Call establishment occurs as described above.
Next, a 'send AudioTM to first user' decision process 8200 determines if an
AudioTM is to be sent to the first user. If not, the 'send AudioTM to first
user'
decision process 8200 exits through its 'no' branch and process flow completes
through an 'end' terminal 8500.
If an AudioTM is to be sent, process flow continues to the 'send Call
Accepted' process 8300 sends a signal from the gateway to the first VoIP
client 1530
to indicate that the call signal has been accepted. The signal may be, for
instance, a
Connect Message according to the H.225 protocol. Processing continues to an
2o 'AudioTM communication' process 8400. The 'AudioTM communication' process
8400 then executes in a fashion analogous to the 'AudioTM communication'
process
4700 or the 'feedback AudioTM communication' process 4600 of Fig. 4.
Processing
completes through an 'end' terminal 8500.
Incoming communication handling will now be described. For purposes of
illustration of incoming communication handling a second user operating the
second
VoIP client 1630 is assumed to be receiving an incoming communication from a
first
user operating one of the first set of aural communication devices 1500.
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Fig. 8B depicts a flow diagram for a method for communicating an AudioICM
message to a VoIP client or Etherphone in accordance with an illustrative
embodiment. Process flow initiates at a 'start' terminal 8600 and continues to
a 'send
AudioICM to second user' decision process 8700 that determines if an AudioICM
should be sant to the second user. If not, the 'send AudioICM to second user'
decision process 8700 exits through its 'no' branch and processing completes
through
an 'end' terminal 8950.
If an AudioICM should be sent, the 'send AudioICM to second user' decision
process 8700 exits through its 'yes' branch to a "Call Accepted' received'
decision
1o process 8800. The "Call Accepted' received' decision process 8800
determines if a
'Call Accepted' signal has been received from the second VoIP client 1630.
When
the 'Call Accepted' signal is received, the "Call Accepted' received' decision
process
8800 exits through is 'yes' branch and process flow continues to an 'AudioICM
communication' process 8900. The 'AudioICM communication' process 8900 then
executes in a fashion analogous to the AudioICM communication method 7000 of
Fig. 7. Processing completes through an 'end' terminal 8500.
2.B. G.ATEWAYAT CENTRAL OF~~CE
In some embodiments, the VoIP gateway is not located at a customer's
premises and may be located, for instance, at a central office. Reference may
be made
2o to Fig. 1C for an exemplary network topology in which such embodiments may
be
deployed. In accordance with such embodiments, the aural communication device
may be, fog instance, a computer programmed to provide a VoIP client such as
the
second VoIP programmed computer 1680, an IP phone such as the VoIP phone 1665,
or a conventional phone connected through a converter and working as an IP
phone,
such as the POTS phone 1670 and the VoIP converter 1675.
Generally, when the VoIP gateway is at a central office, practicing versions
of
the invention is similar to that described above in connection with VoIP
gateways at
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customer premises. Differences may be illustrated in connection with the
second
VoIP programmed computer 1680 as an illustrative embodiment.
The second VoIP programmed computer 1680 may be a conventional PC
running the WINDOWS operating system from Microsoft and NetMeeting client
software. The second VoIP programmed computer 1680 may have a built-in modem
(analog or digital) for access to the Internet-type network 1100. The second
VoIP
programmed computer 1680 may also operate with a conventional phone going
through a converter and then connected to the second VoIP programmed computer
1680 through, for instance, an Ethernet interface, Universal Serial Bus, IEEE
1394
1o serial bus, PCI bus, or other suitable means. Speakers and a microphone are
operatively coupled with the second VoIP programmed computer 1680 to act as
the
ear and mouth pieces of a phone. The NetMeeting software converts audio and
voice
into H.323 compliant IP format. This information is communicated through the
modem and to the second DSLAM 1875 or a Remote Access System ("RAS"), e.g. a
conventional modem bank or Cable Head-end system (not shown). The second
DSLAM 1Q75, RAS, or Cable Head-end communicates the information to the second
VoIP gateway 1825 that may follow the H.323 protocol to determine routing. If
routing to the public switched telephone network 1200 is needed, the second
VoIP
gateway 1825 performs necessary conversion between VoIP signaling and PSTN
voice.
In this situation, programmed instructions may configure the second VoIP
gateway 1825 to embody aspects of, and perform methods according to, versions
of
the invention illustrated and described herein. Typically this may involve, at
least,
providing capability to transfer messages to and from message servers and
local
storage for messages retrieved from message servers. AudioMT, AudioTM, and
AudioICM transmission may be provided as described above with reference to the
processes ~n Fig. 4-Fig. 7 by suitably modifying the call management module
3110 as
one of skilin the art will readily appreciate having the benefit of this
disclosure.
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However, when a simulated dial tone is not provided by the second VoIP
gateway 1825, AudioMT is not supported. AudioTM and AudioICM transmission are
provided as described above in connection with embodiments where the VoIP
gateway is located on a customer's premises.
In some embodiments, the invention may operate with speaker verification
system. The speaker verification system may include a phone control
interactive
voice response ("IVR") system and a voice/key server. One illustrative
embodiment
may be understood with reference to Fig. lE where IVR systems are shown on the
side of the VoIP gateways of Fig. 1C (omitted in Fig. lE for clarity). Shown
in Fig.
to lE are a first phone/key IVR 1250, a second phone/key IVR 1260, a first
voice/key
server 1150, a second voice/key server 1160. Suitable structures for these
systems are
commercially available and the PhoneKey system from ITT Industries of San
Diego,
California and its partner Buytel of Dublin, Ireland, may be used. The first
phone/key
IVR 1250 communicates with the first central office switch 1220, the first
message
switch 127U, and the first voice/key server 1150. The second phone/key IVR
1260
analogously communicates with the second central office switch 1230, the
second
message sv~itch 1280, and the second voice/key server 1160. In operation, when
a
user of, for instance the VoIP phone 1665, initiates a call, the second
central office
switch 1230 will route the call to the second phone/key IVR 1260 for speaker
2o verification based on the subscription profile for the line associated with
the VoIP
phone 1665. The second phone/key IVR 1260 prompts the user to verify his or
her
identificat?on by speaking a particular phrase, e.g., the user's name, a
personal
identification number, or other suitable phrase. The second phone/key IVR 1260
then
consults subscriber records accessible through the second voice/key server
1270 to
determine if the spoken particular phrase can be verified as associated with a
particular subscriber, i.e. the "voice print" matches. If so, validation
information is
returned from the second phone/key IVR 1260 to the second central office
switch
1230 and the call continues as if it were originated from a phone connected to
the
public switched telephone network 1200. In alternative embodiments, a user may
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prompted to identify themselves by entering an identifier such as personal
identification number ("PIN") rather than with an IVR system.
As described above with reference to Fig. lE, the second phone/key IVR
1260, for instance, is shown connected to the second central office switch
1230, i.e.,
on the PS iN side of the VoIP gateway. Alternatively, VoIP gateways, as shown
in
Fig. 1C, may be configured to detect if a caller is initiating a call from a
line
associated with a particular class of subscribers. The VoIP gateway could then
route
the call to a particular location in the public switched telephone nerivork
1200;
thereafter the user's identification could be verified using one of the above-
described
l0 means.
Whatever the means of user identification verification, once the user's
identification is verified, additional features of the invention may be
conveniently
deployed. For instance, a more personalized messaging experience may be
provided
to the user by selection of informational messages responsive to demographic,
psychographic, or socio-economic information in the user's subscription
profile.
Further, user identification verification facilitates the consummation of
voice-based
commercial transactions as the known user's identification more reliably
secures
payment of a price. In some embodiments a feedback message may prompt a user
to
enter into ,~. commercial transaction, the user's identification could be
verified using
2o means described above, and the price the user agrees to pay in the
commercial
transaction could be added to the user's periodic bill from the service
provider. One
of skill in the art will appreciate that these features of the invention are
not limited to
embodiments using VoIP communications.
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3. AURAL COMMUNICATION DEVICE
CONNECTED TO PBX OR KEY
TELEPHONE SYSTEM BEFr7RE
CONNECTED TO CPE
In some embodiments of the invention one of the first set of aural
communication devices 1500 is connected to a PBX or Key Telephone System
before
it is connected to the first CPE device 1400 (through a gateway). In these
embodiments the first CPE device 1400 does not have direct control over the
first
POTS phone 1510 and accordingly does not support transmission of AudioMT
1o messages. Other features of previously described embodiments may still
operate
when the first POTS phone 1510 is connected to a PBX or Key Telephone System
before it is connected to the first CPE device 1400.
In some embodiments when the first POTS phone 1510 is connected to a PBX
or Key Telephone System, the signaling format may reed to be modified. For an
analog trunk type connection, E&M signaling format is suitable. In the case of
a
digital trunk connection, suitable signaling formats include, for example, T1
(or E1)
running ISDN PRI and an appropriate signaling format supported by the PBX; in
the
case of T1, this may be CAS.
4. PHONE DIRECTLY CONNECTED TO THE
CENTRAL OFFICE SWITCH
Some embodiments of the invention involve a phone, such as the third POTS
phone 1710 directly connected to a central office switch, such as the first
central
office swit:;h 1220.
In some embodiments software operating the central office switch is modified
to implement the functions of the call management module 3110, the message
server
agent 3030, and the message manager 3040 as has been previously described. In
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addition, a network connection between the central office switch and a message
server
is established. The message server may be on a public network such as the
Internet or
may be on a private network. The precise location of the message server in a
network
topology is not fundamental.
In a modern PSTN network, a typical network architecture is shown as in Fig.
1B. The public switched telephone network 1200 may rely on SS7 as the basic
infrastructure needed for a service switching point 1210 ("SSP"), which
provides the
local access as well as an ISDN interface for a signaling transfer point 1220
("STP").
The signaling transfer point 1220 provides packet switching of message-based
l0 signaling protocols for use in the network and for a service control point
1230
("SCP"), v=~hich provides access to a network database (not shown). The
service
control point 1230 communicates with a service management system 1240 ("SMS"),
that provi~es a human interface to the database, as well as the capability to
update the
database when needed. An intelligent peripheral 1250 may also be added. The
intelligent peripheral 1250 may provide resource management of devices such as
voice response units, voice announcers, and DTMF sensors for caller-activated
services.
An additional feature of a modern PSTN is service-independent Advanced
Intelligent Networks (AIN) capabilities. With AIN capabilities, service
providers are
able to load service logic in a SCP. The service logic typically allows for
triggering
capabilities to be programmed in order to route the calls of the service's
subscribers.
Many different triggering conditions can be programmed including, for
instance, an
off hook' condition or a number of digits collected. Illustrative versions of
the
invention r.~ay operate with systems that route calls in this fashion.
In one illustrative version, the triggering condition could be an 'off hook'
condition, e.g, a subscriber has taken his or her aural communication device
off hook.
In response to this triggering condition, calls from this subscriber's line
could be
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routed to a particular switch that performs functions analogous to those
described
above in connection with Fig. ~ through Fig. 7. In this illustrative version,
when a
subscriber takes his or her phone off hook to place a call, the off hook
condition is
detected and this triggers the routing of the call to the particular switch.
The
subscriber may then be transmitted informational messages) under the control
of the
particular switch in a fashion analogous to that described above in connection
with
Fig. 2 through Fig. 7. In another version, the service providers could trigger
on a
single digit key such as '* ' or '#' to offer the service of similar
functionality. In still
others, another provider could trigger on multiple digits for the service.
1o Another illustrative embodiment would be where the central office switch is
programmed to route calls originating from subscribers to the device by
appending a
routing code to the calls. The appended routing code triggers the routing of
calls to
the device. One example of a suitable routing code are conventional Primary
Interexchange Carner ("PIC") codes, although one of skill in the art will
appreciate
15 others.
In one version of this illustrative embodiment, the central office switch
automatically appends the PIC code. Conventionally, central office switches
use
routing codes, such as PIC codes, to route long distance calls based on the
subscriber's predetermined long-distance earner. In this version, a
subscription
2o service could be established. The subscription service could offer, for
instance,
subsidized telecommunication products or services in exchange for the
subscriber's
receiving informational messages from a sponsor. A central office switch could
be
configured to append an appropriate PIC code associated with the service as
the
default routing code for subscribers in a fashion analogous to default routing
codes for
25 long-distance calls. Calls from subscribers to the service could then be
routed to a
particular switch which has been configured to implement the functions for and
features of transmission of infoumational messages as described in this
specification.
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In another version, the routing code is entered by the user of the aural
communication device. Conv,;ntionally, central office switches also use
routing codes
entered by the user, to route calls for particular long distance earners. For
instance,
Telecom*ITSA of Arlington, Virginia, offers a service commercially available
in the
United States by which a user dialing "10 10 321 ", will have their call
routed to
Telecom*USA for long-distance calling. This conventional method of routing
calls
could be used to route calls to the particular switch ve~hich has been
configured to
implement the functions for arid features of transmission of informational
messages as
described in this specification.
1o In ~ till another version, the operator assistance code could be entered by
the
user of the aural communication device. This assistance code could be used by
the
central office switches conventionally to route calls from the caller to the
particular
switch which has been configured to implement the functions for and features
of
transmission of informational messages as described in this specification.
15 The aforementioned embodiments in which calls are routed to a particular
switch provide a convenient architecture for commercial adoption of versions
of the
invention: the particular switch practicing features of the invention may be
acquired
by the operator of a central office switch (the particular switch may be
collocated with
existing central switching equipment or remotely disposed therefrom), existing
2o routing methods may be used to then route subscriber's calls to the
particular switch,
and the particular switch may provide informational messages to subscribers.
In the
aforementioned embodiments, the functions described above in connection with
Fig. 2
through Fig. 7 may be implemented. They may be implemented, for instance
through
software that configures the central office switch to implement the functions.
25 Alternatively, the particular switch may be an IVR system designed to
perform
functions analogous to those described above in connection with Fig. 2 through
Fig.
7. However, as one of skill in the art will appreciate, the call density is
typically high
in this type of embodiment. Accordingly, suitable interfaces should be used.
Some
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embodiments could use multiple T1/E1, others could use DS3; others could use
still
higher bandwidth interfaces.
In some variations of the aforementioned embodiments, an IVR system could
be employed for verification o.f a subscriber. Again making reference to Fig.
lE, a
first message switch 1270 and a second message switch 1280 will be noted.
Referring
to the first message switch 1270 as also illustrative of the second message
switch
1280, it can be appreciated that the first message switch 1270 may be flexibly
integrated with existing central switching hardware such as the first central
office
switch 1220 that could route calls to the first message switch 1270, i.e., the
first
1o message switch 1270 may be the particular switch described above. When a
user of,
for instance the third POTS phone 1710, initiates a call, the call could be
routed by the
first central office switch 1220 to the first phone/key IVR 1250 for user
verification.
The user may verify himself or herself by voice print matching in conjunction
with the
first phone/key IVR 1250, by entering personal identification information from
a
phone keypad, or through other means. If the voice print matches or the
personal
identification information validates, the system notifies the first message
switch 1270
of the caller's identification and a voice channel on which the call is to be
carned out.
The first message switch 1270 could then consult the user's su'rrscription
profile to
determine whether one or more informational messages should be sent
transmitted to
the user. As this illustrative example assumes the user is initiating a call,
the
informational messages would be AudioMT or AudioTM (with or without feedback
options). The first message switch 1270 (configured for communication with the
first
message server 1110) can then transmit one or more informational messages
retrieved
from the first message server 1110 to the user. After the transmission of the
informational message(s), the user's call proceeds (in the case of an
AudioMT(s), an
additional dial tone may be provided to the user after the transmission of the
informational message(s)).
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In still other embodiments an intelligent peripheral such as the intelligent
peripheral l 250 may be used to implement features of the invention.
Connection to a
message server such as the first message server 1110 is made with a digital
facility
such as that typically found in a central office. Typically, a central office
switch, such
as the first central office switch 1220 has a connection to a digital
facility. The digital
facility allows signaling information between central office switches to be
consolidated and sent through its own network apart from the voice network.
This
method is used in ISDN and SS7 today. In addition, this method of signaling is
capable of sending and receiving messages, including, for instance, AudioTM,
l0 AudioICM including information retrieved from a remote database.
As will be appreciated by one of skill in the art having the benefit of this
disclosure, a connection to, fcr instance, the first message server 1110 may
be
obtained by having the first message server 1110 act as an intelligent
peripheral such
as the intelligent peripheral 1250. The first message server 1110 would
provide
support for a protocol appropriate for communication with the central office
switch
(currently, SS7 could be used). The first message server 1110 may then either
be
directly connected to the central office switch through digital connection or
through
the service control point 1230. Software running on a central office switch or
a SCP
may then be modified to implement functions according to the invention as
described
herein.
For instance, when a central office switch det;,cts that the subscriber's
aural
communication device is off hook, the switch could be configured by programmed
instructions to assemble the dialing digits and communicate them to the first
message
server 1110 and to communicate the subscriber's line information to the
intelligent
peripheral 1250. If an AudioMT, an AudioTM, or an AudioICM should be sent to
the
subscriber, it will retrieve the message from the message server Intelligent
Peripheral,
and then send it to the subscriber.
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In addition, all or portions of a wireless communications network may also be
integrated with the public switched telephone network 1200. When a wireless
network is integrated with the public switched telephone network 1200, users
of
wireless mobile units may be transmitted informational messages as has been
described above with reference to a phone directly connected to a central
office
switch. Other versions of the invention operate with other configurations of a
wireless
network as illustrated below.
5. WIRELESS
As noted, some embodiments of the invention operate with wireless networks.
to Illustrative embodiments are described below in the context of a cellular
wireless
network with the understanding that one of skill in the art will comprehend
principles
of the invention in this description applicable in other types of wireless
networks (also
within the scope of the invention).
S.A OUTGOING COMMUNICATION
is Fig. 9 depicts a flow diagram of a method for communicating an AudioTM in
accordance with an illustrative cellular embodiment. Process flow of a
'cellular
AudioMT communication' method 9000 initiates at a 'start' terminal 9010 and
continues to a 'call initiation' data block 9020. The 'call initiation' data
block 9020 is
received by the base transceiver stations 1903 and comprises an identification
of the
2o mobile unit 1905 and a serial number. A 'phone ID transmission' process
9030
passes this information to the mobile switching center 1920 and a 'calling
phone
authentication' process 9040 validates the calling phone number with the home
local
register 1930 and the visitor location register 1950 and authenticates the
identity of
the mobile unit 1905.
25 Process flow continues to a 'message server consultation' process 9050. The
'message server consultation' process 9050 is performed by the mobile
switching
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center 1920 and includes the mobile switching center 1920 transmitting mobile
unit
1905 subscriber identification information to the message server 1990. The
message
server 1990 executes an 'AudioTM to cellular' decision process 9060 to
determine
whether AudioTM should be sent to the mobile unit 1905 at this time.
If the AudioTM should not be sent, the 'AudioTM to cellular' decision process
9060 exits trwough its 'no' branch and process flow coiainues to a 'call
placement'
process 9240 that initiates a conventional call connection sequence performed
by the
mobile switching center 1920.
If the AudioTM should be sent, the 'AudioTM to cellular' decision process
l0 9060 exits through its 'yes' branch and process flow continues to an 'an
AudioTM
retrieval' process 9070 that retrieves the AudioTM from the message server
1990 for
sending to the mobile unit 1905. Process flow continues to a 'feedback
AudioTM'
decision process 9080 that consults the AudioTM to determine whether the
AudioTM
will offer feedback options. If not, an 'AudioTM transmission' process 9090
15 transmits the AudioTM to the mobile unit 1905, a 'messages sent flag
setting' process
9230 sets a flag indicating that the AudioTM was sent and process flow
continues to
the 'call placement' process 9240.
If the AudioTM, will offer feedback options, the 'feedback AudioTM'
decision process 9080 exits through its 'yes' branch and process flow
continues to an
20 'feedback AudioTM transmission' process 9100 that transmits the feedback
AudioTM
to the mobile unit 1905. The message may offer any of the feedback options
previously discussed. An 'AudioTM feedback indication' decision process 9110
awaits an indication from the mobile unit 1905 of the selection of a feedback
option.
If no indication of selection of a feedback option is received within a
suitable
25 predetermined time limit, e.g., 5 seconds, the 'AudioTM feedback
indication' decision
process 9110 exits through its 'no' branch and process flow continues to the
'messages sent flag setting' process 9230.
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If an indication of the selection of a feedback option is received, the
'AudioTM feedback indication' decision process 9110 exits through its 'yes'
branch
and process flow continues to a 'feedback indication storage' process 9120
that posts
the selected feedback option to the message server 1990. Next a 'call
redirection'
decision process 9130 determines if the particular selected feedback option
involves
redirecting the user of the mobile unit 1905 to complete a call to another
entity before
completion of the call from the 'call initiation' data block 9020. If so, the
'call
redirection' decision process 9130 exits through its 'yes' branch and a
'number
storage' process 9140 stores the number from the 'call initiation' data block
9020
1o while a 'redirected call' process 9150 executes. The 'redirected call'
process 9150
involves the mobile switching center 1920 consulting the message server 1990
for the
number to which a call should be placed given the particular feedback
selection. In
some embodiments, a user of the mobile unit 1905 may use an alias dialing
feature as
described above (in connection with the 'feedback AudioMT communication'
process
4300) and below (in connection with Fig. 16) and the message server 1990 may
map
the dialed alias to the appropriate number to call. During the 'redirected
call' process
9150 the user of the mobile unit 1905 may carry on a conventional phone
conversation with the entity associated with the user's feedback option
selection.
From the 'redirected call' process 9150 process flow continue: ~o a 'number
retrieval'
2o process 9160 in which the mobile switching center 1920 retrieves the
originally-called
number from the 'call initiation' data block 9020 that was stored by the
'number
storage' process 9140.
If the particular selected feedback option does not involve redirecting the
user
of the mobile unit 1905 to complete a call to another entity, the 'call
redirection'
decision process 9130 exits through its 'no' branch and process flow continues
to an
'automated output' decision process 9170 that determines if an automated
output
should be sent in response to the feedback selection. If so, the 'automated
output'
decision process 9170 exits through its 'yes' branch and process flow
continues to an
'output generation' process 9180 that consults the message server 1990 for the
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particular type of automated output that should be generated. Any of the types
of
automated output discussed above may be generated. Process flow continues to
an
'output communication' process 9190 that communicates the automated output to
a
suitable receiving entity.
From the 'output communication' process 9190 or the 'number retrieval'
process 9160 process flow continues to a 'remaining feedback' decision process
9200
that determines if the user has indicated an additional feedback option
selection. The
user may he presented with such an option by the 'output communication'
process
9190 prompting them with an audio message of additional feedback options or
the
to AudioTM retrieved by the 'an AudioTM retrieval' process 9070 may indicate
that any
of a menu of feedback options could be selected at this point of the process
flow. If
the 'remaining feedback' decision process 9200 detects that an additional
feedback
option has been selected, it exits through its 'yes' branch and a 'next
feedback
message selection' process 9210 retrieves the appropriate next message.
Process flow
continues to a 'next feedback message communication' process 9220 that
communicates the next message to the mobile unit 1905 and process flow returns
to
the 'feedback indication storage' process 9120 for the next iteration.
If the 'remaining feedback' decision process 9200 does not detect that an
additional feedback option has been selected, it exits through its 'no' branch
and
2o process flow continues to the "messages sent flag setting' process 9230 and
the 'call
placement' process 9240. Processing completes through an 'end' terminal 9250.
S.B INCOMING COMMUNICATION
Fig. 10 depicts a flow diagram of a method for communicating a TextICM
message in accordance with an illustrative embodiment. The 'TextICM
communication' method 10000 may operate with elements of a suitable cellular
network topology including, for instance, the exemplary cellular network
architecture
1900 discussed above in connection with Fig. 1D. Process flow initiates at
a'start'
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terminal 10100 when a user is placing a call to the mobile unit 1905. An
'incoming
call' input block 10200 comprising the identification and serial number of the
mobile
unit 1905 as well as the called number is communicated to the mobile switching
center 1920 that validates that called number with suitable lookup in the home
local
register 1930 and the visitor location register 1950. Process flow continues
to a
'TextICM' decision process 10300 that includes the mobile switching center
1920
providing a query to the message server 1990 to determine whether a TextICM
should
be communicated to the mobile unit 1905. The query to the message server 1990
may
include, for instance, identification information from the mobile unit 1905
such as the
1o serial number, and the identification information may be used by the
message server
1990 in conjunction with profile information to generate or select a TextICM
responsive to the particular identification information. As one of skill in
the art will
appreciate, demographic, pshychographic, and profile information may be used
in
generating or selecting the TextICM.
If the TextICM should not be sent the 'TextICM' decision process 10300 exits
through its 'no' branch and process flow continues to an 'incoming call
handling'
process 10800 that completes the placement of the incoming call to the mobile
unit
1905. If the TextICM should be sent to the mobile unit 1905, the 'TextICM'
decision
process 10300 exits through its 'yes' branch and process flow continues to
a'TextICM
2o retrieval' process 10400 in which the TextICM is retrieved from the message
server
1990. Process flow continues to a 'text connection establishment' process
10500 in
which the mobile switching center 1920 establishes a data communication
connection
to the mobile unit 1905. Next, a'TextICM transmission' process 10600 transmits
the
TextICM to the mobile unit 1905 where it may be perceived by the user thereof.
A
'text connection termination' process 10700 terminates the text connection and
process
flow continues to the 'incoming call handling' process 10800 where the
incoming call
may be conventionally completed with the mobile unit 1905. Process flow
completes
through an 'end' terminal 10900.
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User verification, as described above with reference to Fig. lE, could also be
employed with cellular embodiments. To employ speaker verification in a
cellular
embodiment, a speaker verification system, such as the PhoneKey system from
ITT
Industries and Buytel, can be installed between the mobile switching center
1920 and
the public switched telephone network 1200. The process logic would then be
similar
to that described above with reference to the third POTS phone 1710.
It will be understood by one of skill in the art having the benefit of this
disclosure That the principles of the invention are not limited to messages of
the type
refereed to above as a TextICM. Rather, this type of message has been used as
the
1o basis of this description as it what the current generation of wireless
communication
devices in broad distribution can support. It is contemplated that as wireless
communication device technology evolves it will support, for instance video
and other
media. It is contemplated that one of skill in the art having the benefit of
this
disclosure will be able to apply the principles of the invention to operate
with such
15 devices, and such application is contemplated as within the scope and
spirit of the
present invention.
6. MULTIMEDIA MESSAGES
Some embodiments of the invention involve the transmission of non-audio or
multimedia messages.
2o Suitable systems for such embodiments include, for example, a programmed
general-purpose computer, or special-purpose computing machinery, that may be
situated proximate to a conventional telephone. An office or home environment
are
typical situations where suitable systems may be found. The computer may be
used to
provide an enhanced message transmission system.
25 In an illustrative embodiment, a personal computer is situated near a
phone,
for example the first coupled PC & phone 1520. The phone and the computer are
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connected with an IAD such as the first CPE device 1400. The computer is
connected
to the IAD via a LAN such as the first LAN 1540. The IAD has a hardware and
software architecture as described in connection with Fig. 2 and Fig. 3 with
additional
logic for sending a signal ("SIGicm") to the computer via the LAN when the IAD
receives an indication that there is an incoming phone call for the phone line
associated with the computer.
An IAD application runs on the computer that enhances the types of messages
that may be transmitted in conjunction with use of the phone. In the
illustrative
embodiment, the IAD application provides conventional computer telephony
to functions including, for example, name based dialing, voice recognition
dialing,
multiple call forwarding and screening functions, etc., although the presence
of these
functions are not fundamental to the invention and may be omitted in
alternative
embodiments. The computer may also comprise a microphone and speakers that can
operate as the mouthpiece and earpiece of a conventional telephone so that the
15 computer may be used to send and receive telephone calls. The IAI)
application
includes a driver to detect the SIGicm from the IAD. Further, the IAD
application can
generate pop-up-type windows or banners, and play multimedia, audio, or non-
audio
messages under the control of the IAD.
Upon installation of the IAD application, the I~.,D application sends a
message
2o to the IAI7 indicating that the computer is running the IAD application.
The IAD then
extracts a network address, such as an IP address, application software
version,
relevant host hardware and software configuration information, etc. from the
computer. Then, the IAD sends informational messages in audio, non-audio, or
multimedia form for storage on the computer.
25 Operation of the computer, phone, and IAD to provide informational messages
in connection with outgoing communication will now be described. For this
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discussion it is assumed that a first user of a first PC & phone is making an
outgoing
call.
Fig. 11 depicts a flow diagram of a method for communicating informational
messages to a PC & phone in conjunction with an outgoing call 11000 in
accordance
with an illustrative embodiment. Process flow initiates at a 'start' terminal
11010 and
receives an indication a first PC c~ phone is 'off hook" 11020. This may
result from
the first user picking up the handset of the phone or initiating a call though
the
computer. The 'off hook' condition is detected by an "off hook' detection'
process
11030 and process flow continues to an 'AudioMT to first user' decision
process
l0 11050 that determines if the first user should receive an AudioMT message.
If so, the
'AudioMT to first user' decision process 11050 exits through its 'yes' branch
and
process flow continues to a 'AudioMT communication' process 11050 that
communicates the AudioMT for perception by the first user. From the 'AudioMT
communication' process 11060, or if the 'AudioMT to first user' decision
process
11050 exits through its 'no' branch, process next flows to a 'message to
computer
display' decision process 11070 that determines whether a PopupM should be
sent to
the computer display.
If the PopupM should not be sent to the computer display, process flow
continues to a 'dial tone to first user' process 11090. If the PopupM should
be sent to
2o the computer display, the 'message to computer display' decision process
11070 exits
through its 'yes' branch and a 'send message to display' process 11080 sends a
signal
to the IAD application running on the computer to present the PopupM on the
computer display. Process flow then continues to the 'dial tone to first user'
process
11090 which provides a dial tone to the aural communication device of the
first user
and an outgoing call handling' process 11100 that accepts dialing digits from
the first
user's aural communication device and initiates the outgoing call.
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Process flow continues to an 'AudioTM to firs': user' decision process 11130.
The 'AudioTM to first user' decision process 11130 determines if an AudioTM
should be sent to the first user. If the AudioTM should not be sent to the
first user
process flow continues to a 'call establishment and handling' process 11150
that
completes establishing the first user's call. If the AudioTM should be sent to
the first
user a 'AudioTM communication' process 11140 executes in a manner analogous to
that described in connection with Fig. 6A and process flow continues through
the 'call
establishment and handling' process 11150 and completes through the 'end'
terminal
11120.
1o Operation of the computer, phone, and IAD to provide infor~~national
messages
in connection with incoming communication will now be described. For this
discussion it is assumed that a second user of a second PC & phone is
receiving an
incoming ~~all from a first user of a first aural communication device.
Fig. 12 depicts a method for communicating informational messages to a PC
& phone in conjunction with an incoming call 12000. Process flow initiates at
a
'start' terminal 12010 and continues to receive an 'incoming call' data block
12020
indicating that an incoming call has been received. Process flow continues to
a 'ring
and ring back tone' process 12030 that sends a ring signal to the second PC &
phone
and a ring back tone to the first aural communication device. Next, a 'display
2o message' decision process 12040 determines if a PopupM should be sent to
the
computer display. If a message should be sent to the computer display, the
'display
message' decision process 12040 exits through its 'yes' branch and an 'message
display' process 12050 sends the PopupM to the IAD application running on the
computer to initiate the display of the message. From the 'message display'
process
12050, or if the 'display message' decision process 12u~40 exits through its
'no'
branch, process flow continues to an 'answered via PC' decision process 12060.
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The 'answered via PC' decision process 12060 determines if the incoming call
has been answered with the computer. If so, process flow continues to a 'set
'Answered by PC' flag' process 12070 that sets a flag indicating the incoming
call has
been answered with the computer and process flow completes through an 'end'
termina112080.
If the incoming call was not answered with the computer, process flow
continues ~o an 'answered by handset' decision process 12090. The 'answered by
handset' decision process 12090 determines if the incoming call has been
answered
with the phone handset. If the incoming call was not answered with the phone
1o handset, process flow continues to a 'ring limit' decision process 12100.
The 'ring
limit' decision process 12100 determines if a limit on the number of rings has
been
exceeded. If the ring limit has been exceeded, process flow continues to an
'set 'No
Answer' flag' process 12110 that sets a flag indicating that the incoming call
was not
answered and process flow completes through the 'end' terminal 12080.
If the incoming call was answered with the phone handset, the 'answered by
handset' decision process 12090 exits through its 'yes' branch to enter an
'AudioICM
to second user' decision process 12120. If an AudioICM is not to be
transmitted, the
'AudioICM to second user' decision process 12120 exits through its 'no' branch
and
process flow completes through the 'end' terminal 12080.
2o If an AudioICM is to be transmitted, the 'AudioICM to second user' decision
process 12120 exits through its 'yes' branch and a 'retrieve AudioICM' process
12130
retrieves the AudioICM from the memory subsystem 2200. Next, if the AudioICM
is
compressed, a 'compressed message' decision process 12140 exits through its
'yes'
branch and a 'decompression' process 12150 decompresses the AudioICM. Process
flow continues to a 'text to audio' decision process 12160 that exits through
its 'yes'
branch if the AudioICM is in text form and should be converted to audio. A
'text to
audio conversion' process 12170 performs the conversion. Process flow
continues to
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a 'send AudioICM to second user' process 12180 that transmits the AudioICM in
audio form to the second phone for perception by the second user.
Process flow continues to a 'remaining messages' decision process 12190. If
no messages remain for transmission, the 'remaining messages' decision process
12190 exits through its 'no' branch and a 'set 'Message Sent' flag' process
12210 sets
a flag indicating that the AudioICM was sent and process flow completes
through the
'end' terminal 12080. If there is a remaining message for transmission, the
'remaining messages' decision process 12190 exits through its 'yes' branch and
process flcw continues to a 'continue ring back tone' process 12200. As was
1o described above in connection with the 'continue Ring Back Tone' process
7160 of
Fig. 7, the 'continue ring back tone' process 12200 of Fig. 12 may provide the
Ring
Back Tone to the originator of the incoming communication. In other versions,
the
'continue ring back tone' process 12200 sends a call connected message to the
switching system to indicate that the incoming communication has been accepted
and
thereafter a message may be sent to the first user indicating the first user
that his or
her call is connecting with the second user. In other versions the message
indicating
that his or her call is connecting with the second user may be sent before the
call
connected message is sent to the switching system. In either eG~sc:, the
message
indicating that his or her call is connecting with the second user may be
substituted
by, or accompanied with, an informational message such as has been described
above.
Process flow returns to the 'retrieve AudioICM' process 12130 to retrieve the
remaining message and begin the next iteration.
6.A. VOICE OVER IP CLIENT RUNNING
ON A PC CONNECTED TO AN IAD
In some embodiments of the invention where the computer is running the IAD
application, the computer is also running an application that allows the
computer to
act as a VoIP client.
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Fig. 13A depicts a flow diagram for a method for communicating a PopupM
message to a VoIP client in cor_junction with an outgoing call in accordance
with an
illustrative embodiment. Process flow initiates at an 'start' terminal 13010
and
continues to a 'VoIP call initiated' data block 13020 that is detected by the
IAD and
indicates that a VoIP call has been initiated by the VoIP client running on
the
computer. Process flow continues to a 'send PopupM' decision process 13030
that
determines whether a PopupM should be sent. If the PopupM should be sent,
process
flow continues to a 'signal PC' process 13040 where the IAD communicates a
signal
to the IAD application running on the computer to display the PopupM.
1o From the 'signal PC' process 13040, or if the 'send PopupM' decision
process
13030 exits through its 'no' branch, process flow continues to a 'continue
VoIP call'
process 13050 that continues the VoIP conventionally. Process flow completes
through an 'end' terminal 13060.
Fig. 13B depicts a flow diagram for a method for communicating a PopupM
message to a VoIP client in conjunction with an incoming call in accordance
with an
illustrative embodiment. Process flow initiates at a 'start' terminal 13070
and
continues to a 'VoIP call received' data block 13080 that is detected by the
IAD and
indicates that an incoming VoIP call has been received. Process flow continues
to a
'send PopupM' decision process 13090 that determines whether a PopupM should
be
sent. If the PopupM should be sent, process flow continues to a 'signal PC'
process
13100 whe:e the IAD communicates a signal to the IAD application running on
the
computer to display the PopupM.
From the 'signal PC' process 13100, or if the 'send PopupM' decision process
13090 exits through its 'no' branch process flow continues to an 'incoming
call
indication' process 13110 where the IAD passes a Call Setup message to the
VoIP
client running on the computer. Process flow continues to a 'continue VoIP
call'
process 13120 that continues the VoIP conventionally. Process flow completes
through an 'end' terminal 13130.
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6.B. FEEDBACK MULTIIrIEDIA
MESSAGES
Some embodiments provide feedback multimedia messages. In Fig. 14 and
Fig. 15 described below, to illustrate structures performing functions in
accordance
with an illustrative version of the invention, headings are provided in the
figures
indicating the structures performing the processes below. While described
below in
the context of an "HTTP Server" and "IAD/PBX" it will be understood that these
structures are illustrative and not limiting. Further a "Computer/Phone" as
used
below refers to a device, such as an integrated computer and telephone,
capable of
l0 performing ordinary telephonic communications and the function of a general
purpose
programmable computer.
Fig. 14 depicts a flow diagram for a method for communicating a feedback
PopupM message to a computer/phone in conjunction with an outgoing
communication in accordance with an illustrative embodiment. An 'outgoing
feedback PopupM' method 14000 initiates at a 'start' terminal 14010 and
process
flow continues to receive an 'off hook' input block 14020 indicating that the
computer/phone has detected that a user desires to make a call. The 'off hook'
input
block 14020 is passed to the IAD/PBX and received by a 'detects off hook'
process
14030.
2o Process flow continues in the IAD/PBX to a 'send PopupM' decision process
14040 that determines whether a PopupM should be sent to the computer/phone.
The
'send PopupM' decision process 14040 may consult a profile in local or remote
data
store as has been previously illustrated.
If the PopupM should be sent, the 'send PopupM' decision process 14040
exits though its 'yes' branch and process flow continues to a 'PopupM
communication' process 14050 where the IAD/PBX retrieves the PopupM and
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transmits it to the computer/phone where a 'PopupM display' process 14060
presents
it to the user.
Sorr_e embodiments allow feedback to take place with a browser application
running on the computer/phone. Others allow feedback to take place with a call
being
placed to a third party. The 'PopupM display' process 14060 may prompt the
user for
either of the previously-identified types of feedback. For instance. t:he
PopupM could
provide a banner that comprises hyperlinks that, when selected by the user,
are
followed by a browser application. The hyperlinks may be to a sponsor's web
site or,
more generally, to any network resource addressable in the Internet-type
network
1 o 1100. Alternatively the PopupM could have a sound file prompt the user to
make
such a selection. Still further, the PopupM may use a banner or sound file to
prompt
the user to select an item that will result in a call being placed to a third
party, for
instance, a sponsor, other merchant, etc. Both the Uniform Resource Identifier
in the
hyperlink and the phone number of the third party to be called are examples of
resource identifiers. More generally, other embodiments of the invention may
use
other resource identifiers.
From the 'PopupM display' process 14060 process flow continues to a
'resource identifier selection typing' process 14070 that determines a type of
resource
selected by the user. If the user does not select a feedback option within a
2o predetermined time, process flow continues to an 'outgoing call handling'
process
14120 in which the IAD/PBX accepts a dialing sequence and establishes the
user's
outgoing call.
If the resource identifier selected by the user is a phone number of a third
party
to be called, the 'resource identifier selection typing' process 14070 passes
this
information to the IAD/PBX and a 'RI call handling' process 14140 executes in
which
the user completes the call to the third party. Process flow then continues to
the
'outgoing call handling' process 14120.
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If the resource identifier selected by the user is Uniform Resource
Identifier,
the 'resource identifier selection typing' process 14070 passes this
information to a
browser application running on the computer/phone which sends a request
message
for the identified resource. A ' uniform resource identifier request' input
block 14090
is received by the server from which the identified resource is available and
process
flow continues to a 'response gen;,ration' process 14100 where the server
generates
the response to be sent. The response may be one or more HTML pages, or other
World Wide Web resource. Process flow continues co a 'browsing' process 14110
in
which the user may browse World Wide Web resources while they complete their
1o call. Process flow continues to the 'outgoing call handling' process 14120
where the
IAD/PBX receives the number the user desires to call and establishes the
connection.
Process flow continues to an 'outgoing call' process 14130 in which the user
carnes
on his or her outgoing call. The 'outgoing call' process 14130 is shown within
the
'browsing' process 14110 to indicate that user may operate the browser
application
during the 'outgoing call' process 14130. When the 'outgoing call' process
14130
completes process flow completes through an 'end' terminal 14150.
Fig. 15 depicts a flow diagram for a method for communicating a feedback
PopupM message to a computer/phone in conjunction with an ~~a6;oming
communication in accordance with an illustrative embodiment. An 'incoming
feedback PopupM' method 15000 initiates at a 'start' terminal 15010 and
process
flow continues to receive an 'incoming call' input block 15020 whereupon a
'detects
incoming call' process 15030 executes. The 'incoming call' input block 15020
is
passed to the computer/phone and an 'incoming call indication' process 15040
provides a indication that there is an incoming call, e.g. a conventional ring
or
simulated ring.
Process flow continues back to the IAD/PBX where a 'send PopupM' decision
process 16050 determines whether a PopupM should be sent to the
computer/phone.
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The 'send PopupM' decision process 15050 may consult a profile in local or
remote
data store as has been previously illustrated.
If the PopupM should be sent, the 'send PopupM' decision process 15050
exits though its 'yes' branch and process flow continues to a 'PopupM
communication' process 15060 where the IAD/PBX retrieves the PopupM and
transmits it to the computer/phone where a 'PopupM display' process 15070
presents
it to the user.
In contrast to the 'outgoing feedback PopupM' method 14000 when a call is
incoming, preferred embodiments of the invention do not provide the user with
the
to option of placing a call to a third party, although this feature could be
implemented by
one of skill having the benefit of this disclosure. The call to the third
party could be
placed after completion of the incoming call.
From the 'PopupM display' process 15070 process flow continues to a
'resource identifier selection typing' process 15080 that determines a type of
resource
15 selected by the user. If the user does not select a feedback option within
a
predetermi:ied time, process flow continues to an 'incoming call handling'
process
15120 in which the IAD/PBX establishes the connection for the user's incoming
call.
If the resource identifier selected by the user is a Uniform Resource
Identifier,
the 'resource identifier selection typing' process 15080 passes this
information to a
2o browser application running on the computer/phone which sends a request
message
for the identified resource. A 'uniform resource identifier request' input
block 15090
is received by the server from which the identified resource is available and
process
flow continues to a 'response generation' process 15100 where the server
generates
the response to be sent. The response may be one or more HTML pages, or other
25 World Wide Web resource. Process flow continues to a 'browsing' process
15110 in
which the user may browse World Wide Web resources while the incoming call is
completed. Process flow completes through an 'end' terminal 15140.
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7. ALIAS DIALING
The alias feature will now be descried in more detail. Fig. 16 depicts a flow
diagram an 'alias dialing' method 16000 in accordance with an illustrative
embodiment. Process flow initiates at a 'start' terminal 16010 and continues
to an
'off hook detection' data block 16020 that contains a signal that a user's
aural
communication device is "off hook." When the user enters a dialing sequence a
'dialing sequence' input block 16030 is received and is passed to the call
management
module 3110. Next a 'recognized alias' decision process 16040 determines
whether
the dialing sequence corresponds to a recognized alias. In some embodiments,
an
1o alias may be recognized by having a "star" before and after the digits,
e.g., "*489*",
in other embodiments an alias may be recognized by a "star" before the digits
coupled
with a delay of suitable length, e.g. 5 seconds after entry of the last digit.
If an alias is
not recogr_ized, the 'recognized alias' decision process 16040 exits through
its 'no'
branch and process flow completes though an 'end' terminal 16070. If the call
management module 3110 recognizes an alias, the 'recognized alias' decision
process
16040 exits through its 'yes' branch and process flow continues to a 'target
retrieval'
process 16050. In the 'target retrieval' process 16050, the call management
module
3110 sends the dialing sequence from the 'dialing sequence' input block 16030
to an
alias server, which may be, for example, a message server such as the first
message
2o server 1110 or the second message server 1120. The alias server will
translate the
alias name into a corresponding phone number, generates, and communicates a
response message comprising the phone number. When the alias server is also a
message server, an informational message may also be returned. The call
management module 3110 receives the response message via the message server
agent
3030. Process flow continues to a 'target calling' process 16060 where the
call
management module 3110 places the call to the phone number received in the
response. When an informational message is also received, the 'target calling'
process 161160 may also transmit the informational message. In some
embodiments
the informational message may indicate the party being called, e.g. "Now
dialing ITX
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Networks," and the user will be connected to the target of the alias. The user
may
then used their aural communication device conventionally with the target of
the alias.
Process flow completes through the 'end' terminal 16070.
All patents, patent applications, documents, standards, protocols, and draft
protocols referred to herein are incorporated herein by this reference in
their entirety.
Although the present invention has been described in terms of illustrative
embodiments, one skilled in the art will understand that various modifications
and
alterations may be made without departing from the scope of the invention.
Accordingly, the scope of the invention is not to be limited to the particular
to embodiments discussed herein, but should be defined only by the appended
claims
and equival°nts thereof.
77
