U.S. patent application number 10/992821 was filed with the patent office on 2006-05-25 for composite voice applications and services using single sign-on across heterogeneous voice servers.
This patent application is currently assigned to International Business Machines Corporation. Invention is credited to Debanjan Saha, Zon-Yin Shae.
Application Number | 20060109837 10/992821 |
Document ID | / |
Family ID | 36460862 |
Filed Date | 2006-05-25 |
United States Patent
Application |
20060109837 |
Kind Code |
A1 |
Saha; Debanjan ; et
al. |
May 25, 2006 |
Composite voice applications and services using single sign-on
across heterogeneous voice servers
Abstract
A framework is provided to offer composite voice applications
and services. A composite application and service begins from the
user dialing in via phone and ends by the user hanging up the
phone. The composite interactive voice services architecture
includes a session initiation protocol session service unit is in
the loop of session signaling all the time starting from the time
the user first dials in, during the user roaming across various
voice servers, and until the end of the composite service when user
hangs up the phone. This unit accepts a command and login
instruction of the next interactive voice service from the previous
interactive voice service. The unit has knowledge of DTMF sequences
required for the user to login to next interactive voice service.
The session service unit automatically accomplishes a roaming
process such that composite applications and services can be
achieved across various voice servers.
Inventors: |
Saha; Debanjan; (Mohegan
Lake, NY) ; Shae; Zon-Yin; (South Salem, NY) |
Correspondence
Address: |
DUKE. W. YEE
YEE & ASSOCIATES, P.C.
P.O. BOX 802333
DALLAS
TX
75380
US
|
Assignee: |
International Business Machines
Corporation
Armonk
NY
|
Family ID: |
36460862 |
Appl. No.: |
10/992821 |
Filed: |
November 19, 2004 |
Current U.S.
Class: |
370/352 ;
370/401 |
Current CPC
Class: |
H04M 3/493 20130101;
H04M 3/382 20130101; H04L 29/06027 20130101; H04L 65/40 20130101;
H04L 65/4007 20130101; H04L 65/1063 20130101; H04L 65/1006
20130101 |
Class at
Publication: |
370/352 ;
370/401 |
International
Class: |
H04L 12/66 20060101
H04L012/66; H04L 12/56 20060101 H04L012/56 |
Claims
1. A method for providing a composite service, the method
comprising: establishing a first session between a telephone device
and a first interactive voice service; responsive to activation of
an external link, establishing a connection between a service
broker and a second interactive voice service; automatically
logging the telephone device into the second interactive voice
service; and terminating the first session and establishing a
second session between the telephone device and the second
interactive voice service in response to the telephone device
successfully logging into the second interactive voice service.
2. The method of claim 1, wherein the first session and the second
session are realtime transport protocol sessions.
3. The method of claim 2, wherein terminating the first session and
establishing a second session includes: reporting a first realtime
transport protocol port from the telephone device to the service
broker; sending the first realtime transport protocol port to the
second interactive voice service; reporting a second realtime
transport protocol port from the second interactive voice service
to the service broker; and sending the second realtime transport
protocol port to the telephone device.
4. The method of claim 1, wherein the first interactive voice
service and the service broker communicate using a first session
initiation protocol session.
5. The method of claim 4, further comprising: sending login
information for the telephone device and the second interactive
voice service from the first interactive voice service to the
service broker using the first session initiation protocol
session.
6. The method of claim 5, wherein the login information includes at
least one of a user identification and a passcode for the second
interactive voice service.
7. The method of claim 5, wherein the login information includes
parameters for accessing features of the second interactive voice
service.
8. The method of claim 4, wherein the second interactive voice
service and the service broker communicate using a second session
initiation protocol session and wherein automatically logging the
telephone device into the second interactive voice service includes
sending the login information from the service broker to the second
interactive voice service using the second session initiation
protocol session.
9. The method of claim 1, wherein the first interactive voice
service is a personalized voice portal.
10. The method of claim 9, further comprising: monitoring for
activation of a default link; and terminating the second session
and establishing a third session between the telephone device and
the personalized voice portal responsive to the telephone device
activating the default link.
11. A method, in a first interactive voice service, for providing a
composite service, the method comprising: establishing a first
session with a telephone device responsive to the telephone device
logging into the first interactive voice service; responsive to
activation of an external link, establishing a second session with
a service broker; and sending login information for the telephone
device and a second interactive voice service to the service broker
using the second session.
12. The method of claim 11, wherein the first session is a realtime
transport protocol session.
13. The method of claim 11, wherein the second session is a session
initiation protocol session.
14. An apparatus for providing a composite service, the apparatus
comprising: a gateway device that converts protocols and data
signals between a public switch telephone network and session
initiation protocol; a service broker, wherein the service broker
establishe a first session between a telephone device and a first
interactive voice service, establishes a connection between a
service broker and a second interactive voice service responsive to
activation of an external link, automatically logs the telephone
device into the second interactive voice service, terminates the
first session, and establishes a second session between the
telephone device and the second interactive voice service in
response to the telephone device successfully logging into the
second interactive voice service.
15. The apparatus of claim 14, wherein the first session and the
second session are realtime transport protocol sessions.
16. The apparatus of claim 15, wherein the service broker
terminates the first session and establishes a second session by
reporting a first realtime transport protocol port from the
telephone device to the service broker, sending the first realtime
transport protocol port to the second interactive voice service,
reporting a second realtime transport protocol port from the second
interactive voice service to the service broker, and sending the
second realtime transport protocol port to the telephone
device.
17. The apparatus of claim 14, wherein the first interactive voice
service and the service broker communicate using a first session
initiation protocol session.
18. The apparatus of claim 17, wherein the service broker receives
login information for the telephone device and the second
interactive voice service from the first interactive voice service
using the first session initiation protocol session.
19. The apparatus of claim 17, wherein the second interactive voice
service and the service broker communicate using a second session
initiation protocol session and wherein the service broker
automatically logs the telephone device into the second interactive
voice service by sending the login information from the service
broker to the second interactive voice service using the second
session initiation protocol session.
20. The apparatus of claim 14, wherein the first interactive voice
service is a personalized voice portal.
21. The apparatus of claim 20, wherein the service broker monitors
for activation of a default link, terminates the second session,
and establishes a third session between the telephone device and
the personalized voice portal responsive to the telephone device
activating the default link.
Description
BACKGROUND OF THE INVENTION
[0001] 1. Technical Field
[0002] The present invention relates to data processing and, in
particular, to interactive voice services. Still more particularly,
the present invention provides a method, apparatus, and computer
program product for providing composite voice applications and
services using single sign-on across heterogeneous voice
servers.
[0003] 2. Description of Related Art
[0004] Interactive voice servers (IVS) are services that can be
accessed from all phone devices by a public switched telephone
network (PSTN) or voice over Internet protocol (VoIP) Internet
(wired or wireless). In general, these servers interact with users
only by voice and dual tone multi frequency (DTMF) signals, also
known as touchtone signals, or equivalent. This is especially
important for mobile users that can access the services by cell
phones to perform critical business functions while away from the
office, for example.
[0005] It is generally required for users to type in a user
identification (ID) and/or a passcode by pushing the phone's keypad
to log in an IVS after dialing in to the service. The passcode may
be transported to the IVS via DTMF industrial standard.
Consequently, every time a user dials into a different voice
server, the user must go through the DTMF login process again.
Currently, an IVS only provides services accessible from its own
server, and cannot provide access to the services provided by
another IVS.
[0006] As a result, each user must remember a user ID and passcode
for each service. Furthermore, the user must dial into each service
separately and perform a DTMF sequence to log into each service.
This results in an inconvenience for users who wish to access
several services, especially when the user accesses these services
within a short period of time. There is added frustration when the
services are somewhat related to one another and the user must
remember information gained from one service to perform a task with
another service.
SUMMARY OF THE INVENTION
[0007] The present invention recognizes the disadvantages of the
prior art and provides a framework to provide composite voice
applications and services. A composite application and service
begins from the user dialing in via phone and ends by the user
hanging up the phone. During this process, users will be able to
access an integrated and much more powerful voice services in a
user controllable sequence from multiple interactive voice servers.
In this architecture, the control signal to establish session is
separated from the voice data path for scalability. The composite
interactive voice services architecture includes a session
initiation protocol session service unit is in the loop of session
signaling all the time starting from the time the user first dials
in, during the user roaming across various voice servers, and until
the end of the composite service when user hangs up the phone. This
unit accepts a command and login instruction of the next
interactive voice service from the previous interactive voice
service. The unit has knowledge of DTMF sequences required for the
user to login to next interactive voice service. The session
service unit automatically accomplishes a roaming process such that
composite applications and services can be achieved across various
voice servers.
BRIEF DESCRIPTION OF THE DRAWINGS
[0008] The novel features believed characteristic of the invention
are set forth in the appended claims. The invention itself,
however, as well as a preferred mode of use, further objectives and
advantages thereof, will best be understood by reference to the
following detailed description of an illustrative embodiment when
read in conjunction with the accompanying drawings, wherein:
[0009] FIG. 1 illustrates an example interactive voice services
environment in which exemplary aspects of the present invention may
be implemented;
[0010] FIG. 2 is a block diagram illustrating a framework for
composite applications and services in accordance with an exemplary
embodiment of the present invention;
[0011] FIG. 3 illustrates an example composite interactive voice
service in accordance with a preferred embodiment of the present
invention;
[0012] FIG. 4 illustrates a session initiation protocol
infrastructure in accordance with an exemplary embodiment of the
present invention;
[0013] FIG. 5 is an example timing diagram illustrating a single
sign on between two interactive voice services in accordance with a
preferred embodiment of the present invention;
[0014] FIGS. 6A-6F are block diagrams illustrating an example
bridge-and-roll operation in accordance with a preferred embodiment
of the present invention; and
[0015] FIG. 7 is a flowchart illustrating the operation of a
framework for composite applications and services in accordance
with an exemplary embodiment of the present invention.
DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENT
[0016] Interactive voice servers (IVS) are very services that can
be accessed from telephone devices by a public switch telephone
network (PSTN) or voice over Internet Protocol (VoIP) connection.
FIG. 1 illustrates an example interactive voice services
environment in which exemplary aspects of the present invention may
be implemented. In general, these servers interact with users only
by voice and dual tone multi frequency (DTMF) signals or
equivalent. This is especially important for mobile users that
access the services by cellular telephones.
[0017] In the example shown in FIG. 1, a user of telephone device
104 accesses interactive voice services through PSTN/Internet
infrastructure 102. The interactive voice services may include, for
example, home appliance control 111, voice enabled calendar 112,
voice message box 113, conferencing server 114, help desk 115, and
gaming portal 116. However, more or fewer interactive voice
services and various other services may be available through
PSTN/Internet infrastructure 102. It is generally required for
users to type in a user ID and/or passcode by pushing buttons on a
keypad to log into an IVS right after dial-in. The passcode is
transported to IVS via DTMF signals or equivalent. Consequently, in
current systems, every time a user dials into a different voice
server, the user must go through the DTMF login process again. An
IVS currently only provides services accessible from its own server
and cannot provide access to services that provided by another
IVS.
[0018] In most interactive voice applications, it is required for
users to roam and compose the individual services from more than
one IVS. In these cases, users are asked to type in a passcode
again for each service, potentially causing the user to remember
and enter a different passcode for each service. In accordance with
a preferred embodiment of the present invention, a framework is
provided, which allows composite applications and services. A
composite application or service begins from the user dialing in
via a telephone device and ends by the user hanging up the
telephone device. During this process, users are able to access
integrated and much more powerful voice services in a user
controllable sequence from multiple IVSs. In this framework, the
control signal to establish session is separated from the voice
data path for the scalability.
[0019] FIG. 2 is a block diagram illustrating a framework for
composite applications and services in accordance with an exemplary
embodiment of the present invention. A user of telephone device 204
accesses interactive voice services through session initiation
protocol (SIP) service broker 220. SIP is an Internet protocol (IP)
telephony signaling protocol that is primarily used for voice over
IP calls, although SIP can also be used for video or any media
type. SIP is a text-based protocol that is based on hypertext
transport protocol (HTTP) and multipurpose Internet mail extensions
(MIME), which makes it suitable and very flexible for integrated
voice-data applications. The interactive voice services may
include, for example, home appliance control 111, voice enabled
calendar 112, voice message box 113, conferencing server 114, help
desk 115, and gaming portal 116. However, more or fewer interactive
voice services and various other services may be available through
SIP service broker 220.
[0020] SIP service broker 220 is in the loop of SIP session
signaling all the time starting when the user first dials in,
during the user roaming across various voice servers, and until the
end of the composite service when user hang up telephone device
204. SIP service broker 220 accepts the command and log-in
instruction of the next IVS from the previous IVS. SIP service
broker 220 has knowledge of the DTMF sequences required for the
user to login to next IVS. SIP service broker 220 then accomplishes
a roaming process from one IVS to another such that a composite
application or service is achieved across various voice
servers.
[0021] SIP service broker 220 provides a single service entry point
with personalized interactive voice selection. Personalized voice
portal 222 is an IVS that provides access to other interactive
voice services. Personalized voice portal 222 stores the user ID
and passcode information for each IVS that the user will access.
For example, the user of telephone device 204 logs into SIP service
broker 220, which initially provides access to personalized voice
portal 222. When the user first access personalized voice portal
222, the user enters a user ID and/or passcode using DTMF signals
or equivalent. Personalized voice portal 222 then authenticates the
user ID or passcode and performs an interactive service that allows
the user to access other IVSs.
[0022] In a particular example, the user of telephone device 204
may indicate that she wishes to access voice enabled calendar 212.
SIP service broker 220 then establishes a session with voice
enables calendar 212 and performs the DTMF sequences to log the
user into IVS 212. SIP service broker 220 then ends the session
with personalized voice portal 222 and connects telephone device
204 to IVS 212. The user may then determine that she wishes to
access conferencing server 214. SIP service broker than performs a
rendezvous operation 250 to transfer the session from IVS 212 to
IVS 214. After SIP service broker 220 performs rendezvous operation
250, telephone device 204 is connected to conferencing server 214.
The details of the rendezvous operation will be described in
further detail below.
[0023] The composite application and service framework of the
present invention enables roaming among services across various
interactive voice servers with only a single user dial-in session.
Each IVS can provide links for the user to jump to external
services, similar to the way Web pages provide links to jump to
other Web pages in other Web servers. Each IVS can provide a
default link to return to personalized voice portal 222. For
example, when a user enters a "*0" DTMF signal, this may indicate
that the user wishes to terminate the session with the current IVS
and return to personalized voice portal 222. SIP service broker 220
may intercept every DTMF signal sent from telephone device 204 for
navigation among interactive voice services. Enterprises or
carriers may outsource individual voice service components and then
combine these services to form a composite application or service.
Similarly, individual voice service providers may team together to
provide more powerful services or applications.
[0024] FIG. 3 illustrates an example composite interactive voice
service in accordance with a preferred embodiment of the present
invention. In this example, a first IVS is an interactive voice
response (IVR) service 310 that allows a user to access calendar
314 through an application 312 for meeting entries. Application 312
may be written in voice extensible markup language (VoiceXML or
VXML). VXML is an extension to extensible markup language (XML)
that defines voice segments and enables access to the Internet via
telephones and other voice-activated devices. A second IVS is a set
of conference servers 330 that provide a conference service but
require the user to provide a DTMF conference passcode to
enter.
[0025] The user first dials into SIP service broker 320 using
telephone device 304 to establish a SIP session between telephone
device 304 and SIP service broker 320. The user may log into the
SIP service broker using a user ID and/or passcode within the SIP
session. The user may then instruct SIP service broker 320 to
establish a session with IVR 310. SIP service broker 320 then
establishes a SIP session between SIP service broker 320 and IVR
310. In the SIP session, SIP service broker 320 logs into IVR 310
using a user ID and/or passcode on behalf of the user.
[0026] When the login is complete, SIP service broker 320 instructs
IVR 310 to establish a realtime transport protocol (RTP) session
between telephone device 304 and IVR 310. RTP is an Internet
protocol (IP) that supports realtime transmission of voice and
video. RTP is widely used for IP telephony and video streaming. An
RTP packet rides on top of user datagram protocol (UDP) and
includes timestamp and synchronization information in its header
for proper reassembly at the receiving end. UDP is a protocol
within the TCP/IP protocol suite that is used in place of TCP when
a reliable delivery is not required, as is the case with Internet
telephony and other realtime transmissions, for instance. UDP is
widely used for realtime audio and video traffic where lost packets
are simply ignored, because there is no time to retransmit. The
user may then use telephone device 304 to access calendar 312 using
application/VXML 312 via the RTP session with IVR 310.
[0027] While the user accesses calendar 312, the user may discover
that a telephone conference is scheduled using the second IVS.
Therefore, the user must establish a communications session with
one of conference servers 330. IVR 310 may be modified to provide a
link to jump to the second IVS. Thus, the user may enter a button
sequence using telephone device 304 to instruct IVR 310 to transfer
the session to the second IVS. IVR 310 then sends the user ID and
conference ID for the session with conference servers 330 through
its SIP session with SIP service broker 320. Service broker 320
then establishes a SIP session with one or more of conference
servers 330. In the SIP session, SIP service broker 320 logs into
one or more of conference servers 330 using the user ID and
conference ID on behalf of the user. When the login is complete,
SIP service broker 320 instructs conference servers 330 to
establish a RTP session between telephone device 304 and conference
servers 330. The user may then access the service provided by
conference servers 330 without having to dial separate numbers and
perform multiple logins. The end composite service is very powerful
enterprise application. A user may dial in to find out about
current meeting and then join the conference automatically.
[0028] FIG. 4 illustrates a session initiation protocol
infrastructure in accordance with an exemplary embodiment of the
present invention. In order to access the composite service, a user
of telephone device 404 dials into SIP service broker unit 420,
which itself can be reached globally using a pre-specified
telephone number. Telephone device 404 communicates with SIP
service broker 420 either by dynamically SIP registration to SIP
register server or by static dial plan configuration in the SIP
infrastructure. The unit in turn dials in to IVS A 412 on behalf of
the user. As a result, SIP service broker unit 420 will have two
SIP control sessions, one to IVS A 412, and the other to user
404.
[0029] IVS A 412 then establishes an RTP session with telephone
device 404. The RTP session directs flow of voice packets between
telephone device 404 and IVS A 412 for scalability. SIP service
broker unit 420 is not in the voice data path. This increases the
scalability of the architecture.
[0030] User 404 may access the voice services provided by this IVS
A 412 at this point. There are a few ways that the user can
navigate and roam the services among the IVSs. First, each IVS may
provide links for users to jump to external services similar to the
manner in which web pages provide links to jump to other web pages
in other web servers. As another example, each IVS may provides a
"default link" to jump back to the default personalized voice
portal. For example, whenever a user pushes a "*0" DTMF signal,
this may instruct the IVS to terminate the session and return to
the personalized voice portal. As yet another example, SIP service
broker unit 420 may intercept every DTMF signal sent from the user
for navigation.
[0031] DTMF signals may only be sent after the call session is
connected. After the session between IVS A 412 and telephone device
404 is established, the user may follow voice response instructions
to enter a user ID and/or password from the key pad of telephone
device 404. This creates a DTMF sequence from phone 404 to IVS A
412 via the RTP path. If the password accepted, then the user may
access the services provided by IVS A 412 by voice through the RTP
path. IVS A 412 may provide links for allowing the user to jump to
other IVSs to continue the composite services. The link might
provide the phone number of the next IVS, such as IVS B 414, the
required DTMF sequences, or the desired service ID. If only desired
service ID is available, SIP service broker unit 420 may optimally
allocate an IVS server to dial in for the services.
[0032] Telephone device 404 may communicate with SIP service broker
420, IVS A 412, and/or IVS B 414 using public switch telephone
network 406 or Internet/intranet 402. SIP service broker 420 may
communicate with IVS A 412 and/or IVS B 414 using Internet/intranet
402, although, alternatively, SIP service broker 420 may also
communicate with IVSs through PSTN 406. An SIP media gateway is a
gateway between PSTN signaling and the IP world. SIP media gateway
422 converts protocols and data signals between PSTN and SIP such
that SIP devices in the IP world can communicate with phone devices
in PSTN world.
[0033] FIG. 5 is an example timing diagram illustrating a single
sign on between two interactive voice services in accordance with a
preferred embodiment of the present invention. When the user of
telephone device 504 dials in and accesses ISV A 512, SIP service
broker 520 establishes session control 1 with telephone device 504
and establishes session control 2 with IVS A 512 (step 1).
Telephone device 504 establishes RTP voice interaction with IVS A
512 (step 2).
[0034] When the user requests an external service link, ISV A 512
sends the link parameters to SIP service broker 520 to indicate the
user's service request (step 3). SIP service broker 520 terminates
the ISV A session by sending a "BYE" signal (step 4). SIP service
broker 520 then stops the user's old RTP session and request and
prepares a new RTP session by sending a RE-INVITE without session
description protocol (SDP) (step 5). SDP describes the media to be
used in the SIP session.
[0035] Telephone device 504 reports its RTP port for the new
session (step 6). SIP service broker 520 then dials in IVS B 514
and reports the new RTP port by INVITE (step 7). IVS B 514 report
its RTP port for the session (step 8). SIP service broker 520 sends
an acknowledgement (ACK) to IVS B 514 to complete the call leg for
IVS B 514 (step 9).
[0036] Note that, at this moment, the second call leg has
completed, but not the first call leg. Thus, telephone device 504
will not start sending RTP packets. This is important because, if
telephone device 504 starts to send RTP packets, it will corrupt
the DTMF sequences due to the continuity requirements of RTP packet
sequences. SIP service broker 520 then sends required DTMF
sequences to automatically login to IVS B 514 (step 10). SIP
service broker 520 then completes the call first leg and enable
telephone device 504 to send RTP packets by sending ACK (step 11).
If an IVS supports login by SIP control signaling, SIP service
broker 520 can also use the corresponding SIP signaling to login
that IVS.
[0037] FIGS. 6A-6F are block diagrams illustrating an example
bridge-and-roll operation in accordance with a preferred embodiment
of the present invention. More particularly, with reference to FIG.
6A, a user of telephone device 604 connects to IVS A 612 by
establishing a connection with public branch exchange (PBX)/SIP
gateway 610 using time division multiplexing (TDM). TDM is
technology that transmits multiple signals simultaneously over a
single transmission path. Each lower-speed signal is time sliced
into one high-speed transmission. For example, three incoming 1,000
bps signals can be interleaved into one 3,000 bps signal. The
receiving end divides the single stream back into its original
signals. SIP service broker 620 then establishes a SIP connection
with telephone device 604 and with IVS A 612 and then establishes
an RTP session between telephone device 604 and IVS A 612.
[0038] With reference to FIG. 6B, SIP service broker 620
establishes a SIP connection with IVS B 614 and prepares a new RTP
session by sending an INVITE without SDP to IVS 614 and sending a
RE-INVITE without SDP to PBX/SIP gateway 610. This disconnects the
first session, forms a second connection to IVS B 614, and
terminates and cleans up the RTP residue in PBX/SIP gateway 610.
Turning to FIG. 6C, PBX/SIP gateway 610 sends a "200 OK" signal
with SDP(A) to SIP service broker 620. This is to tell the SIP
service broker 620 the media connection properties described in SDP
that PBX/SIP gateway 610 would like to use for this SIP session.
IVS B 614 sends a "200 OK" signal with SDP(B) to SIP service broker
620. This is to tell SIP service broker 620 the media connection
properties described in SDP that IVS B 614 would like to use for
this SIP session.
[0039] With reference now to FIG. 6D, SIP service broker 620 sends
an ACK to IVS B 614 to create the RTP session. Next, with reference
to FIG. 6E, SIP service broker 620 sends a passcode to log on to
IVS B 614. Thereafter, as shown in FIG. 6F, SIP service broker 620
sends ACK to PBX/SIP gateway 610 to create the RTP session from the
gateway and to direct the RTP session to the RTP port of IVS B
614.
[0040] FIG. 7 is a flowchart illustrating the operation of a
framework for composite applications and services in accordance
with an exemplary embodiment of the present invention. Operation
begins and the user dials into the SIP service broker (block 702).
Then, the SIP service broker dials the IVS on behalf of the user
(block 704) and establishes a connection for RTP voice interaction
between the user and the IVS (block 706). Next, the user follows
voice response instructions to enter a user ID and/or password
(block 708). The IVS determines whether the password is accepted
(block 710). If the password is not accepted, operation returns to
block 708.
[0041] If the password is accepted in block 710, the user accesses
services of the IVS through the RTP path (block 712). The IVS
determines whether the user activates an external service link
(block 714). If the user does not activate an external service
link, operation returns to block 712. If, however, the user does
activate an external service link, the IVS sends the link
parameters to the SIP service broker (block 716). The SIP service
broker then terminates the IVS session (block 718), stops the RTP
session (block 720), and requests to prepare a new RTP session with
the new IVS (block 722).
[0042] The telephone device reports the RTP port for the new RTP
session (block 724). The SIP service broker then dials the new IVS
and reports the new RTP port to the new IVS (block 726). The new
IVS reports the RTP port for the session (block 728) and the SIP
service broker sends an acknowledgement to the new IVS (block 730).
Thereafter, the SIP service broker sends an acknowledgement with
the RTP port of the new IVS to the telephone device to complete the
old RTP session and to begin the new RTP session (block 732). Then,
operation returns to block 712 to allow the user to access services
of the new IVS through the newly established RTP session.
[0043] Thus, the present invention solves the disadvantages of the
prior art by providing a framework to provide composite voice
applications and services. A composite application and service
begins from the user dialing in via phone and ends by the user
hanging up the phone. During this process, users will be able to
access an integrated and much more powerful voice services in a
user controllable sequence from multiple interactive voice servers.
In this architecture, the control signal to establish session is
separated from the voice data path for scalability. The composite
interactive voice services architecture includes a session
initiation protocol session service unit is in the loop of session
signaling all the time starting from the time the user first dials
in, during the user roaming across various voice servers, and until
the end of the composite service when user hangs up the phone. This
unit accepts a command and login instruction of the next
interactive voice service from the previous interactive voice
service. The unit has knowledge of DTMF sequences required for the
user to login to next interactive voice service. The session
service unit automatically accomplishes a roaming process such that
composite applications and services can be achieved across various
voice servers.
[0044] It is important to note that while the present invention has
been described in the context of a fully functioning data
processing system, those of ordinary skill in the art will
appreciate that the processes of the present invention are capable
of being distributed in the form of a computer readable medium of
instructions and a variety of forms and that the present invention
applies equally regardless of the particular type of signal bearing
media actually used to carry out the distribution. Examples of
computer readable media include recordable-type media, such as a
floppy disk, a hard disk drive, a RAM, CD-ROMs, DVD-ROMs, and
transmission-type media, such as digital and analog communications
links, wired or wireless communications links using transmission
forms, such as, for example, radio frequency and light wave
transmissions. The computer readable media may take the form of
coded formats that are decoded for actual use in a particular data
processing system.
[0045] The description of the present invention has been presented
for purposes of illustration and description, and is not intended
to be exhaustive or limited to the invention in the form disclosed.
Many modifications and variations will be apparent to those of
ordinary skill in the art. The embodiment was chosen and described
in order to best explain the principles of the invention, the
practical application, and to enable others of ordinary skill in
the art to understand the invention for various embodiments with
various modifications as are suited to the particular use
contemplated.
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