Distributed voice processing system

Multiplex communications – Pathfinding or routing – Combined circuit switching and packet switching

Reexamination Certificate

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Details

C370S357000, C370S389000, C370S395430, C370S442000

Reexamination Certificate

active

06724751

ABSTRACT:

BACKGROUND OF THE INVENTION
The invention relates to a distributed voice processing system, and more particularly to such a system comprising at least first and second components connected by an isochronous communications link.
In recent years telephone networks have started to provide increasingly sophisticated services, and this trend is likely to continue, for example with speech dialling (where the caller simply speaks in the name of a person who it is desired to call), and so on. In order to implement such advanced services, so-called Intelligent Peripherals (IPs) are required to provide the extra capability over and above the routing and billing functions of a traditional telephone network. These IPs are normally located at Service Nodes (SN) within the network. IPs can also be used at private installations, especially those which handle a large volume of incoming and/or outgoing calls.
IPs are used to provide enhanced services such as voice services, voice response, voice mail, fax mail, and other applications. Most IPs are based on voice processing units (or voice response units), and are usually restricted in capacity to handling typically between 100 and 200 channels. However, it is often required to have Service Nodes that support hundreds or even thousands of telephony channels, so that multiple voice processing units must be used. This requires an architecture suitable for combining the multiple voice processing units into a service node.
An example of a system having multiple voice processing units is disclosed in U.S. Pat. No. 5,029,199, which describes a voice messaging system for use in the telephone network or at a large private installation. This system includes multiple voice processing units, a digital switch and a master control unit. The voice processing units have a digital T
1
trunk connection to the switch for telephony channels (or an E
1
connection where appropriate), whilst the voice processing units, digital switch, and master control unit are also connected together via a data control bus, such as an Ethernet. Notification of an incoming call initially arrives at the master control unit, which selects an appropriate voice processing unit to handle the call (based for example on current loading of the units, or the called or calling number), and sends an appropriate instruction to the switch. Then, when the incoming call itself arrives at the switch, it is routed to the selected voice processing unit. If necessary, the selected voice processing unit can retrieve the appropriate greeting or a stored message belonging to the caller from another voice processing unit over the data control bus.
A somewhat similar system is described in U.S. Pat. No. 5,394,460, in which multiple voice processing systems are connected by a Fibre Distributed Data Interface (FDDI). This system does not have a master control unit—each voice processing unit can handle calls in a suitable manner for any subscriber to the system, including retrieving, where necessary, a subscriber greeting and messages from the other voice processing unit(s) over the FDDI link. “Conversant
1
voice system: Architecture and Applications”, by R Perdue and E Rissanen, in AT&T Technical Journal, September/October 1986, Vol 65/5, p34-47 describes a system where multiple subsystems provide eg voice response, isolated word recognition etc, are connected to a switch so that they can be shared between calls. A line interface unit is interposed between the switch and the telephone network. The subsystems and line interface unit are also attached via a general-purpose interface bus, which runs an application to control overall system operation, invoking the facilities of subsystems when necessary. This system effectively operates as a single, multi-function IP, and so is limited in its overall call processing capabilities.
An example of the use of an IP installation is illustrated in
FIG. 1
, which acts as a Service Node
10
within the telephone network. Thus calls are routed to the service node from within the telephone network
5
via multiple trunk lines
15
, such as T
1
digital trunk lines, which each carry twenty-four individual telephony channels. The incoming trunk lines arrive at a switch
20
, which has a set of line interface (LI) units
22
, such that each incoming trunk line is terminated by a corresponding LI unit. Usually this switch is a programmable, time-division multiplex (TDM) switch, which allows host-controlled routing of incoming calls to a selected voice processing unit, and host-controlled bridging between calls. The line interface units can be used to perform a variety of functions, such as analog to digital conversion (not necessary if T
1
trunks are used), signalling, DTMF detection, and so on.
The back end of the switch is connected to multiple voice processing units
50
A,
50
B via a further set of digital trunk lines
45
, which typically are also T
1
trunks. A pair of line interface units are provided at opposite ends of each internal digital trunk line
45
, the first line interface unit
24
being at the back end of the switch, and the second line interface unit
52
being attached to the voice processing unit to which the trunk is connected. The voice processing units
50
contain multiple voice resources
55
, which can be used for example to play voice prompts, perform voice recognition, FAX-back etc, depending on the desired service. The line interface units
52
are connected to the voice resources
55
via a TDM bus
54
. Often this connection is hardwired so that calls on a particular line interface unit are always directed to the same voice resource. Overall control of a call at the voice processing unit is provided by one or more applications
60
running on the
15
voice processing unit. The application determines, for example, which voice prompts to play, and in which order.
The switch routes the incoming call from trunk line to an appropriate voice processing unit under the control of a Call Manager
30
(also termed the host). The Call Manager is connected to the switch
20
by a first control interface
35
(typically provided either by a LAN connection or some switch-dependent hardware link), and to the voice processing units
50
by a second control interface
36
(typically a LAN connection). The Call Manager uses the second control interface to communicate with the voice processing units
50
, via a call manager component
62
on each voice processing unit.
The routing decision of the Call Manager can be based on various criteria. For example, the caller may have personal information (eg a voice mail message or greeting) stored on a particular voice processing unit, or may require a special service (eg use of specialised voice recognition hardware). This routing is often performed based on Automatic Number Identification and/or Dialled Number Identification Service (ANI/DNIS) information, the former representing the calling number, the latter the called number. This ANI/DNIS information (if available) is supplied to the Call Manager from the telephone network via switch
20
over control interface link
35
. Other information might also be employed to establish the routing, such as time of day, and/or current loading of the different end units. This latter information can be obtained by the Call Manager directly from the different call manager components on their respective voice processing units. The second control interface
36
also allows the Call Manager to pass information (such as ANI/DNIS) via the call manager component
62
to the application
60
on the voice processing unit which is to receive a particular call, so that this information is available if desired in order to process the call.
Once the Call Manager has instructed the switch of a suitable destination for the incoming call, the switch effects this routing by completing the appropriate internal connections. This involves routing the call from a channel on an incoming trunk
15
through to an available channel on an appropriate one of the internal trunk lines
45
for con

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