Telephonic communications – Audio message storage – retrieval – or synthesis – Interaction with an external nontelephone network
Reexamination Certificate
2001-12-18
2004-11-09
Bui, Bing (Department: 2642)
Telephonic communications
Audio message storage, retrieval, or synthesis
Interaction with an external nontelephone network
C370S352000, C370S356000, C379S088130, C379S088180, C379S265090
Reexamination Certificate
active
06816579
ABSTRACT:
BACKGROUND OF THE INVENTION
1. Field of the Invention
The present invention relates to call processing, and is more particularly related to establishing a voice call over a packet switched network via a voice response unit.
2. Discussion of the Background
The popularity and convenience of the Internet has resulted in the reinvention of traditional telephony services. These services are offered over a packet switched network with minimal or no cost to the users. IP (Internet Protocol) telephony, thus, have found significant success, particularly in the long distance market. In general, IP telephony, which is also referred to as Voice-over-IP (VOIP), is the conversion of voice information into data packets that are transmitted over an IP network. Users also have turned to IP telephony as a matter of convenience in that both voice and data services are accessible through a single piece of equipment, namely a personal computer. The continual integration of voice and data services further fuels this demand for IP telephony applications.
With the growing acceptance of IP telephony among the millions of consumers, service providers are cognizant of the impact that these users have on network capacity (e.g., switch sizing, line capacity) as well as network resources (e.g., peripheral voice processing devices). A valuable network resource is the voice response unit (VRC), which provides announcement and interactive voice response functions. These functions have become essential for the expedient treatment of voice calls, especially in call center applications and operator assistance. Because VRU ports are expensive it is desirable to ensure efficient use of such ports.
FIG. 8
illustrates a conventional IP telephony system. In this system
800
, an end office
801
houses a switch
803
and a VRU
805
; the switch
803
communicates with the VRU over a release line trunk (RLT). Switch
803
serves user
807
to a public switch telephone network (PSTN)
809
. The VRU
805
is not functionally integrated With the IP network
815
. That is, the VRU
805
works primarily in conjunction with the switch
803
within the PSTN realm. Using plain old telephone service (POTS), a calling party
807
can place a telephone call over PSTN
809
to a called party
811
or
813
.
The PSTN
809
is connected to an IP (Internet Protocol) network
815
, thereby enabling communication among the voice stations
807
,
811
, and
813
, which are connected to the public switch telephone network
809
, and the personal computers
817
and
819
, which are attached to the IP network
815
. Attention is now drawn to transmission of voice calls over the IP network
815
.
Four possible scenarios exist with the placement of a VOIP call: (1) phone-to-phone, (2) phone-to-PC (3) PC-to-phone, and (4) PC-to-PC. In the first scenario of phone-to-phone call establishment, voice station
807
is switched through PSTN
809
by switch
803
to a VOIP gateway (not shown), which forwards the call through the IP network
815
. The packetized voice call is then routed through the IP network
815
, exiting the IP network
815
at an appropriate point to enter PSTN
809
and terminates at voice station
811
. Under the second scenario, voice station
807
places a call to personal computer (PC)
817
through switch
803
to PSTN
809
. This voice call is then switched by the PSTN
809
to a VOIP gateway (not shown), which forwards the voice call to PC
817
via IP network
815
. The third scenario involves PC
817
placing a call to voice station
813
, for example. Using a voice encoder, PC
817
introduces a stream of voice packets into IP network
815
that are destined for a VOIP gateway (not shown). A VOIP gateway (not shown) converts the packetized voice information into a POTS electrical signal, which is circuit switched to voice station
813
. Lastly, in the fourth scenario, PC
817
establishes a voice call with PC
819
. In this case, packetized voice data is transmitted from PC
817
via IP network
815
to PC
819
, where the packetized voice data is decoded.
As indicated above, a network resource that permits the efficient processing of voice calls is a VRU
805
.
FIG. 9
shows a conventional call path that is established by switch
803
to VRU
805
. RLT links
901
connect switch
803
to VRU
805
, consuming two ports of each of these network components
803
and
805
. RLT links
901
enable the release of a call back to switch
803
from VRU
805
. This releasing functionality allows the VRU
805
to be dropped from the voice call without impacting the call completion between call originator
903
and call terminator
905
.
For explanatory, purposes, it is assumed that a VRU
805
is needed to assist with call processing from call originator
903
(i.e., calling party) to call terminator
805
(i.e., called party). Call originator
903
places a voice call to switch
803
using port
1
. In turn the switch
803
switches the call out of port
3
to port
1
of VRU
805
. Once this call is established with VRU
805
, the VRU
805
prompts the call originator
903
, for example, to collect digits regarding account codes or billing information in order to authorize and validate the call originator
903
. After this process, the VRU
805
loops the voice call back to the switch
803
via port
2
over RLT
901
over RLT
901
into port
4
of switch
803
. Switch
803
then switches the call out of port
2
to call terminator
905
. The RLT links
901
permits the VRU
805
to drop out of the call when the call is completed between call originator
903
and call terminator
905
. This release mechanism occurs over the PSTN
809
. Such a mechanism is important because it frees up the VRU
805
to process other calls; in addition, the switch
803
frees up two of its ports. An equivalent functionality is desirable in an IP telephony system.
Based on the foregoing there is a clear need for improved approaches for call processing with respect to use of network resources.
There is also a need to increase the integration of voice services over a data network.
There is a further need to minimize the cost of network operation.
Based on the need to efficiently employ network resources, an approach for optimizing the use of VRU in an IP telephony environment is highly desirable.
SUMMARY OF THE INVENTION
According to one aspect of the invention, a method is provided for processing a voice call over a packet switched network between a call originator and a call terminator. The method comprises establishing a first call-leg between the call originator and a voice response unit (VRU) using a menu router that provides call control services according to a signaling protocol. The method also includes establishing a second call-leg between the VRU and the call terminator based upon the signaling protocol. The method further includes binding the first call-leg and the second call-leg to complete the voice call between the call originator and the call terminator, and releasing the voice call from the VRU based upon the signaling protocol. Under this approach, network resources are efficiently utilized, resulting in reduction of network operation costs.
According to another aspect of the invention, a communication system for processing a voice call over a packet switched network comprises a call originator that is configured to initiate and to receive the voice call over the packet switched network. A menu router performs call control services relating to the voice call. A voice response unit (VRU) processes a call setup request from the call originator. A call terminator is configured to process the voice call. The call originator, the call terminator, menu router, and the VRU communicate using a common protocol. The call originator establishes a first call-leg with the VRU via the menu router. The VRU establishes a second call-leg with the call terminator and drops from the voice call upon binding the first call-leg and the second call-leg. The above arrangement advantageously provides greater integration of voice services over a pa
Cannon Matthew
Donovan Steven R.
Hearty John
Bui Bing
MCI Inc.
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