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Patent 2385478 Summary

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Claims and Abstract availability

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(12) Patent Application: (11) CA 2385478
(54) English Title: SYSTEM AND METHOD FOR CALL ROUTING IN AN INTEGRATED TELECOMMUNICATIONS NETWORK HAVING A PACKET-SWITCHED NETWORK PORTION AND A CIRCUIT-SWITCHED NETWORK PORTION
(54) French Title: SYSTEME ET PROCEDE D'ACHEMINEMENT D'APPELS DANS UN RESEAU DE TELECOMMUNICATIONS INTEGRE AYANT UNE PARTIE DE RESEAU A COMMUTATION PAR PAQUETS ET UN PARTIE DE RESEAU A COMMUTATION DECIRCUIT
Status: Deemed Abandoned and Beyond the Period of Reinstatement - Pending Response to Notice of Disregarded Communication
Bibliographic Data
(51) International Patent Classification (IPC):
  • H04W 92/02 (2009.01)
  • H04L 61/4588 (2022.01)
  • H04W 8/02 (2009.01)
  • H04W 40/02 (2009.01)
  • H04W 80/10 (2009.01)
  • H04W 88/16 (2009.01)
(72) Inventors :
  • XIANG, ZENG-JUN (Canada)
(73) Owners :
  • TELEFONAKTIEBOLAGET LM ERICSSON (PUBL)
(71) Applicants :
  • TELEFONAKTIEBOLAGET LM ERICSSON (PUBL) (Sweden)
(74) Agent: ERICSSON CANADA PATENT GROUP
(74) Associate agent:
(45) Issued:
(86) PCT Filing Date: 2000-08-29
(87) Open to Public Inspection: 2001-03-29
Examination requested: 2002-03-18
Availability of licence: N/A
Dedicated to the Public: N/A
(25) Language of filing: English

Patent Cooperation Treaty (PCT): Yes
(86) PCT Filing Number: PCT/SE2000/001656
(87) International Publication Number: WO 2001022766
(85) National Entry: 2002-03-18

(30) Application Priority Data:
Application No. Country/Territory Date
09/399,918 (United States of America) 1999-09-21

Abstracts

English Abstract


A system and method for IP-based call routing in an integrated
telecommunications network having a packet-switched network portion (e.g., a
Voice-over-Internet Protocol (VoIP) network portion (14) and one or more
circuit-switched network (CSN) portions (12) such as a PSTN (50) or a radio
telephony network. A mobile Switching Center (MSC (24)) serving one or more
mobile subscribers (28) is provided with an Internet Protocol (IP)-
Interworking Unit interface (30) towards the VoIP network portion. The radio
telephony network also includes a Location Server (LS(18)) containing mapping
information between routing numbers (e.g., Temporary Location Directory
Numbers or TLDNs), called party numbers (B-numbers) and IP addresses of
entities to which a call can be routed over an IP trunk from the MSC. A
querying mechanism is provided in the MSC for interrogating the LS based upon
a routing number, a called party number, or both, provided to the MSC. The MSC
obtains an IP address from the LS which is used for effectuating the IP trunk.
A plurality of Bearer Independent Call Control (BICC) messages and a plurality
of Integrated Services Digital Network (ISDN) User Part (ISUP) messages are
transmitted among the various nodes of the integrated telecommunications
network, e.g., one or more MSCs with the IP interface, a Local Exchange of the
PSTN, etc. for establishing the IP trunk. Where an IP trunk is not available,
a circuit-switched path (e.g., a Synchronous Transfer Mode (STM) trunk), is
utilized in the call path. The IP trunk is implemented using Real-time
Transfer Protocol (RTP) and Session Description Protocol (SDP) to convey the
voice payload associated with the call.


French Abstract

L'invention concerne un système et un procédé d'acheminement d'appels par Internet dans un réseau de télécommunications ayant une partie de réseau à commutation par paquets (p. ex. une partie de réseau voix sur IP (VoIP) (14)) et une ou plusieurs parties de réseau par commutation de circuit (CSN) (12), telles que RTPC (50) ou un réseau de radiotéléphonie. Un centre de commutation mobile (CCM (24)) desservant un ou plusieurs abonnés mobiles (28) comprend une interface d'unité d'interfonctionnement IP (30) à la partie de réseau VoIP. Le réseau de radiotéléphonie comporte également un serveur de localisation (LS (18)) contenant des informations de mappage entre des numéros d'acheminement (p. ex. des numéros d'appels de localisation temporaire ou TLDN), des numéros de partie appelée (numéros B) et des adresses IP d'entités auxquelles un appel peut être acheminé sur une ligne IP d'un CCM. Un mécanisme d'interrogation est compris dans le MMC pour interroger le LS sur la base d'un numéro d'acheminement, un numéro de partie appelée ou les deux fournis au MMC. Ce dernier obtient une adresse IP du LS qui est utilisée pour établir la ligne IP. Plusieurs messages de contrôle d'appels indépendant du support (BICC) et plusieurs messages du sous-système utilisateur du réseau numérique à intégration de services (ISDN) sont transmis parmi les différents noeuds du réseau de télécommunications intégré, p. ex. un ou plusieurs CCM ayant l'interface IP, un central du RTPC, entre autres, afin d'établir la ligne IP. Lorsqu'une ligne IP n'est pas disponible, une voie à commutation de circuit (p. ex. une ligne à mode de transfert synchrone (STM)), est utilisée dans la voie d'appel. La ligne IP est implantée au moyen du protocole de transfert en temps réel (RTP) et du protocole de description de session (SDP) afin de transmettre la charge utile vocale associée à l'appel.

Claims

Note: Claims are shown in the official language in which they were submitted.


-16-
WHAT IS CLAIMED IS:
1. A call routing method for routing a call originated by a PSTN phone
to a mobile station (MS) (28) disposed in an integrated telecommunications
network
(10) having a packet-switched network (PSN) portion and a wireless circuit-
switched
network (CSN) portion (12B) including a Location Server (LS) (18) containing
mapping information between routing number information and Internet Protocol
(IP)
address information, the method comprising the steps of:
receiving, in a Local Exchange (LE) disposed in a PSTN portion (50),
a call initiation message from the PSTN phone, the call initiation message
including
the MS's number;
performing a number analysis on the MS's number by the LE;
routing the call by the LE to a home gateway Mobile Switching Center
(MSC) of the MS via an Inter-Exchange Carrier (IXC);
determining, in the home gateway MSC, if the MS is located in a
visited area, wherein the home gateway MSC includes an Internet Protocol (IP)
interface towards the PSN portion (14) and a querying means (25) to query the
LS
(18);
if so, querying, by the home gateway MSC, a Home Location Register
(HLR) (16), for location information of a serving MSC (24) that currently
serves the
MS in the visited area, wherein said querying step is effectuated by sending
an ANSI-
41 LOCREQ message from the home gateway MSC to the HLR (16);
responsive to the querying step by the home gateway MSC,
interrogating the serving MSC (24) by the HLR (16) for a routing number with
respect
to the call originated by the PSTN phone;
responsive to the interrogating step by the HLR (16), returning the
muting number by the serving MSC (24) to the HLR (16);
forwarding the routing number to the home gateway MSC by the HLR
(16);
thereafter, querying, by the home gateway MSC, the LS (18) for a
mutable Internet Protocol (IP) address;

-17-
determining, in the LS (18), if an 1P address of the serving MSC (24)
exists based on the routing number;
if so, returning the IP address of the serving MSC (24) to the home
gateway MSC by the LS (18);
based on the IP address of the serving MSC (24) received from the LS
(18), establishing an IP trunk between the home gateway MSC and the serving
MSC
(24);
muting the call originated by the PSTN phone to the serving MSC (24)
using a first circuit-switched path between the LE and the home gateway MSC
via the
IXC, in conjunction with the 1P trunk between the home gateway MSC and the
serving
MSC (24), the 1P trunk forming at least a segment of the PSN portion (14);
if the determining step in the LS (18) determines that the serving MSC
(24) does not have an 1P address based on the routing number, ascertaining an
IP
address of a destination MSC which includes an IP interface towards the PSN
portion
(14) and is located closest to the serving MSC (24);
thereafter, forwarding the IP address of the destination MSC to the
home gateway MSC;
based on the IP address of the destination MSC received from the LS
(18), establishing an IP trunk between the Name gateway MSC and the
destination
MSC; and
muting the call originated by the PSTN phone to the serving MSC (24)
using the first circuit-switched path between the LE and the home gateway MSC
via
the IXC, in conjunction with the IP trunk between the home gateway MSC and the
destination MSC, the 1P trunk forming at least a segment of the PSN portion
(14), and
a second circuit-switched path between the destination MSC and the serving MSC
(24).
2. Cancel.
3. The call routing method as set forth in claim 1, wherein the step of
interrogating the serving MSC (24) by the HLR (16) is effectuated by sending
an

-18-
ANSI-41 ROUTREQ message from the HLR (16) to the serving MSC (24).
4. The call routing method as set forth in claim 3, further comprising the
step of paging the MS by the serving MSC (24) for a pre-routing call setup.
5. The call routing method as set forth in claim 3, wherein the step of
querying the LS (18) by the home gateway MSC is effectuated by sending a
Service
Location Protocol (SLP) SERVICEREQ message from the home gateway MSC to the
LS (18).
6. The call muting method as set forth in claim 3, wherein the step of
establishing an IP trunk between the home gateway MSC and the destination MSC
and
the step of establishing IP trunk between the home gateway MSC and the serving
MSC
(24) are effectuated by sending a plurality of Bearer Independent Call Control
(BICC)
messages and a plurality of Integrated Services Digital Network (ISDN) User
Part
(ISUP) messages among at least a subset of the following nodes: the LE, the
IXC, the
home gateway MSC, the serving MSC (24) and the destination MSC.
7. The call routing method as set forth in claim 6, wherein the IP trunk
between the home
gateway MSC and the destination MSC and the IP trunk between the home gateway
MSC and the serving MSC (24) are implemented using Real-time Transfer Protocol
(RTP) and Session Description Protocol (SDP) to convey a voice payload
associated
with the call.
8. The call routing method as set forth in claim 6, wherein the each of the
first and second
circuit-switched paths comprises a Synchronous Transfer Mode (STM) trunk.
9. A call routing method for routing a call originated by a mobile station
(MS) (28) to a PSTN phone disposed in an integrated telecommunications network

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(10) having a packet-switched network (PSN) portion (14) and a wireless
circuit-
switched network (CSN) portion (12B) including a Location Server (LS) (18)
containing mapping information between called party numbers and Internet
Protocol
(IP) addresses, the method comprising the steps of
receiving, in a Mobile Switching Center (MSC) serving the MS, a call
initiation message from the MS, the call initiation message including the PSTN
phone's number, wherein the MSC includes an Internet Protocol (IP) interface
towards
the PSN portion (14) and a querying means (25) to query the LS (18);
performing a number analysis on the PSTN phone's number by the
MSC to determine if the call is a long distance call; and characterized by:
if so, interrogating the LS (18) by the MSC to obtain an IP address of
a destination MSC which includes an IP interface towards the PSN portion (14)
and
is located closest to a Location Exchange (LE) that serves the PSTN phone,
wherein
said interrogating step is effectuated by sending a Service Location Protocol
(SLP)
SERVICEREQ message from the MSC to the LS (18);
responsive to the interrogating step, receiving the IP address of the
destination MSC by the MSC serving the MS;
based on the IP address of the destination MSC received from the LS
(18), establishing an IP trunk between the MS's MSC and the destination MSC;
if the destination MSC and the LE are located in a single coverage area,
routing the call originated by the MS to the LE using the IP trunk between the
MS's
MSC and the destination MSC, in conjunction with a first circuit-switched path
between the LE and the destination MSC; and
otherwise, routing the call using the IP trunk between the MS's MSC
and the destination MSC, in conjunction with a second circuit-switched path
between
the LE and the destination MSC via an Inter-Exchange Carrier (IXC).
10. Cancel
11. The call routing method as set forth in claim 9, wherein the step of
establishing an IP trunk between the MSC and the destination MSC is
effectuated by

-20-
sending a plurality of Bearer Independent Call Control (BICC) messages and a
plurality of Integrated Services Digital Network (ISDN) User Part (ISUP)
messages
among at least a subset of the following nodes: the LE, the IXC, the MSC
serving the
MS, and the destination MSC.
12. The call routing method as set forth in claim 11, wherein the
IP trunk between the MSC serving the MS and the destination MSC is implemented
using Real-time Transfer Protocol (RTP) and Session Description Protocol (SDP)
to
convey a voice payload associated with the call.
13. The call routing method as set forth in claim 12, wherein the
each of the first and second circuit-switched paths comprises a Synchronous
Transfer
Mode (STM) trunk.
14. A call routing method for routing a call originated by a first mobile
station (MS) (28) and terminating at a second MS located in its home area,
each of the
MSs disposed in an integrated telecommunications network (10) which includes a
packet-switched network (PSN) and a circuit-switched network (CSN) portion
(12B)
including a Location Server (LS) (18) containing mapping information between
called
party numbers and Internet Protocol (IP) addresses, the method comprising the
steps
of
receiving, in a Mobile Switching Center (MSC) serving the first MS,
a call initiation message from the first MS, the call initiation message
including the
second MS's number, wherein the first MS's MSC includes an Internet Protocol
(IP)
interface towards the PSN portion (14) and a querying means (25) to query the
LS
(18);
performing a number analysis on the second MS's number by the MSC
serving the first MS to determine if the call is a long distance call; and
characterized
by:
if so, interrogating the LS (18) by the first MS's MSC to obtain an IP
address of a serving MSC (24) that serves the second MS in its home area, the

-21-
interrogating step including sending the second MS's number from the first
MS's
MSC to the LS (18), wherein said interrogating step is effectuated by sending
a
Service Location Protocol (SLP) SERVICEREQ message from the first MS's MSC
to the LS (18) which includes the second MS's number;
determining, in the LS (18), if the IP address of the serving MSC (24)
exists based on the second MS's number;
if so, returning the IP address of the serving MSC (24) to the first MS's
MSC by the LS (18);
based on the IP address of the serving MSC (24) received from the LS
(18), establishing an IP trunk between the first MS's MSC and the serving MSC
(24);
routing the call originated by the first MS to the serving MSC (24)
using the IP trunk between the first MS's MSC and the serving MSC (24), the IP
trunk
forming at least a segment of the PSN portion (14);
if the determining step in the LS (18) determines that the serving MSC
(24) does not have an IP address based on the second MS's number, ascertaining
an
IP address of a destination MSC which includes an IP interface towards the PSN
portion (14) and is located closest to the serving MSC (24);
thereafter, forwarding the IP address of the destination MSC to the
MSC serving the first MS;
based on the IP address of the destination MSC received from the LS
(18), establishing an IP trunk between the first MS's MSC and the destination
MSC;
and
routing the call originated by the first MS to the serving MSC (24)
using the IP trunk between the first MS's MSC and the destination MSC, the IP
trunk
forming at least a segment of the PSN portion (14), and a circuit-switched
path
between the destination MSC and the serving MSC (24).
15. The call routing method as set forth in claim 14, wherein the step of
interrogating the LS (18) by the MSC serving the first MS is effectuated by
sending
a Service Location Protocol (SLP) SERVICEREQ message from the first MS's MSC
to the LS (18) which includes the second MS's number.

-22-
16. The call routing method as set forth in claim 14, wherein the step of
establishing an IP trunk between the MSC serving the first MS and the
destination
MSC and the step of establishing an IP trunk between the first MS's MSC and
the
serving MSC (24) are effectuated by sending a plurality of Bearer Independent
Call
Control (BICC) messages and a plurality of Integrated Services Digital Network
(ISDN) User Part (ISUP) messages among at least a subset of the following
nodes: the
MSC serving the first MSC, the serving MSC (24) serving the second MS, and the
destination MSC.
17. The call muting method as set forth in claim 16, wherein the
IP trunk between the first MS's MSC and the destination MSC and the IP trunk
between the first MS's MSC and the second MS's serving MSC (24) are
implemented
using Real-time Transfer Protocol (RTP) and Session Description Protocol (SDP)
to
convey a voice payload associated with the call.
18. The call routing method as set forth in claim 16, wherein the
circuit-switched path between the destination MSC and the serving MSC (24)
comprises a Synchronous Transfer Mode (STM) trunk.
19. A call routing method for routing a call originated by a first mobile
station (MS) (28) and terminating at a second MS located outside its home
area, each
of the MSs disposed in an integrated telecommunications network (10) which
includes
a packet-switched network (PSN) and a circuit-switched network (CSN) portion
(12B)
including a Location Server (LS) (18) containing mapping information between
routing numbers, called party numbers and Internet Protocol (IP) addresses,
the
method comprising the steps of
receiving, in a Mobile Switching Center (MSC) serving the first MS,
a call initiation message from the first MS, the call initiation message
including the
second MS's number, wherein the first MS's MSC includes an Internet Protocol
(1P)
interface towards the PSN portion (14) and a querying means (25) to query the
LS

-23-
(18);
performing a number analysis on the second MS's number by the MSC
serving the first MS to determine if the call is a long distance call; and
characterized
by:
if so, interrogating the LS (18) by the first MS's MSC to obtain an IP
address of a home gateway MSC that serves the second MS in its home area, the
interrogating step including sending the second MS's number from the first
MS's
MSC to the LS (18), wherein said interrogating step is effectuated by sending
a
Service Location Protocol (SLP) SERVICEREQ message from the first MS's MSC
to the LS (18) which includes the second MS's number;
determining, in the LS (18), the IP address of the home gateway MSC
based on the second MS's number;
returning, by the LS (18), the IP address of the home gateway MSC to
the MSC serving the first MS;
thereafter, sending the second MS's number and the IP address of the
first MS's MSC from the first MS's MSC to the home gateway MSC;
responsive to the second MS's number received from the first MS's
MSC, determining by the home gateway MSC that the second MS is located in a
visited area;
querying, by the home gateway MSC, a Home Location Register (HLR)
(16) of the second MS to obtain the second MS's location;
responsive to the querying step by the home gateway MSC,
interrogating by the HLR (16) a serving MSC (24) that serves the second MS in
the
visited area;
responsive to the interrogating step by the HLR (16), returning a routing
number by the serving MSC (24) to the HLR (16);
forwarding the routing number to the home gateway MSC by the HLR
(16);
thereafter, interrogating the LS (18) by the home gateway MSC to
obtain an IP address of the serving MSC (24) that serves the second MS in the
visited
area, the interrogating step including sending the routing number received
from the

-24-
HLR (16);
determining, in the LS (18), if the IP address of the serving MSC (24)
exists based on the routing number;
if so, returning the IP address of the serving MSC (24) to the home
gateway MSC by the LS (18);
forwarding the IP address of the serving MSC (24) from the home
gateway MSC to the MSC serving the first MS;
based on the IP address of the serving MSC (24) received from the
home gateway MSC, establishing an IP trunk between the MSC serving the first
MS
and the serving MSC (24) which serves the second MS in the visited area;
routing the call originated by the first MS to the serving MSC (24)
using the IP trunk between the first MS's MSC and the serving MSC (24), the IP
trunk
forming at least a segment of the PSN portion (14);
if the determining step in the LS (18) determines that the serving MSC
(24) does not have an IP address based on the routing number, ascertaining an
IP
address of a destination MSC which includes an IP interface towards the PSN
portion
(14) and is located closest to the serving MSC (24);
thereafter, returning the IP address of the destination MSC to the home
gateway MSC by the LS (18);
forwarding the IP address of the destination MSC from the home
gateway MSC to the MSC serving the first MS;
based on the IP address of the destination MSC received from the home
gateway MSC, establishing an IP trunk between the MSC serving the first MS and
the
destination MSC; and
routing the call originated by the first MS to the serving MSC (24)
using the IP trunk between the first MS's MSC and the destination MSC, the IP
trunk
forming at least a segment of the PSN portion (14), and a circuit-switched
path
between the destination MSC and the serving MSC (24).
20. Cancel

-25-
21. The call muting method as set forth in claim 19, wherein the step of
interrogating the LS (18) by the home gateway MSC is effectuated by sending a
Service Location Protocol (SLP) SERVICEREQ message from the home gateway
MSC to the LS (18) which includes the routing number of the serving MSC (24)
received from the HLR (16).
22. The call routing method as set forth in claim 21, wherein the step of
establishing an IP trunk between the first MS's MSC and the destination MSC
and the
step of establishing an IP trunk between the first MS's MSC and the serving
MSC (24)
are effectuated by sending a plurality of Bearer Independent Call Control
(BICC)
messages and a plurality of Integrated Services Digital Network (ISDN) User
Part
(ISUP) messages among at least a subset of the following nodes: the first MS's
MSC,
the home gateway MSC for the second MS, the serving MSC (24) that serves the
second MS in the visited area, and the destination MSC.
23. The call routing method as set forth in claim 22, wherein the
IP trunk between the first MS's MSC and the destination MSC and the IP trunk
between the first MS's MSC and the serving MSC (24) are implemented using Real-
time Transfer Protocol (RTP) and Session Description Protocol (SDP) to convey
a
voice payload associated with the call.
24. The call muting method as set forth in claim 22, wherein the
the circuit-switched path between the destination MSC and the serving MSC (24)
comprises a Synchronous Transfer Mode (STM) trunk.
25. The call routing method as sct forth in claim 24, wherein the
the Synchronous Transfer Mode (STM) trunk involves an Inter-Exchange Carrier
disposed between the serving and destination MSCs.
26. An integrated telecommunications network (10) with a wireless circuit-
switched network (CSN) portion (12B) and a Voice-over-Internet Protocol (VoIP)

-26-
network portion, comprising
a Mobile Switching Center (MSC) serving one or more mobile
subscribers, the MSC having an Internet Protocol (IP)-Interworking Unit
interface
towards the VoIP network portion;
a Location Server (LS) (18) containing mapping information between
routing numbers, called party numbers and IP addresses of entities to which a
call can
be routed over an IP trunk from the MSC; and characterized by:
querying means (25) in the MSC for interrogating the LS (18) based
upon at least one of a routing number and a called party number provided to
the MSC,
in order to obtain an IP address from the LS (18) for effectuating the IP
trunk from the
MSC based on the IP address such that a call involving a mobile subscriber
that is
served by the MSC is routed at least in part over the IP trunk, wherein said
querying
means (25) operates by sending a Service Location Protocol (SLP) SERVICEREQ
message from the MSC to the LS (18).
27. Cancel
28. The integrated telecommunications network (10) as set forth in claim
26, wherein the IP trunk is implemented using Real-time Transfer Protocol
(RTP) and
Session Description Protocol (SDP) to convey a voice payload associated with
the call.
29. The integrated telecommunications network (10) as set forth in claim
26, further includes a PSTN (50) wherein the call is originated by a PSTN
phone
served by a Local Exchange disposed in the PSTN (50).
30. The integrated telecommunications network (10) as set forth in claim
26, further includes a PSTN (50) wherein the call is terminated to a PSTN
phone
served by a Local Exchange disposed in the PSTN (50).

Description

Note: Descriptions are shown in the official language in which they were submitted.


CA 02385478 2002-03-18
WO 01/22766 PCT/SE00/01656
SYSTEM AND METHOD FOR CALL ROUTING IN AN INTEGRATED
TELECOMMUNICATIONS NETWORK HAVING A PACKET
SWITCHED NETWORK PORTION AND A CIRCUIT-SWITCHED
NETWORK PORTION
BACKGROUND OF THE INVENTION
Technical Field of the Invention
The present invention relates to integrated telecommunication systems and,
more particularly, to a system and method for routing long-distance calls in
an
integrated telecommunications network having a packet-switched network portion
(for
example, a network using Internet Protocol (IP)) that is coupled to circuit-
switched
network portions such as a wireless telephony network portion, a Public
Switched
Telephone Network (PSTN), or both.
Description of Related Art
Coupled with the phenomenal growth in popularity of the Internet, there has
been a tremendous interest in using packet-switched network (PSN)
infrastructures
(e.g., those based on IP addressing) as a replacement for the existing circuit-
switched
network (CSN) infrastructures used in today's telephony. From the network
operators'
perspective, the inherent traffic aggregation in packet-switched
infrastructures allows
for a reduction in the cost of transmission and the infrastructure cost per
end-user.
Ultimately, such cost reductions enable the network operators to pass on the
concomitant cost savings to the end-users. One of the well-known advantages of
the
IP-based networks with respect to voice transmission is that considerable
savings in
long distance charges may be realized for calls that are to be routed over
multiple
geographic regions such as, for example, Local Access and Transport Areas
(LATAs).
Some of the market drivers that impel the existing Voice-over-IP (VoIP)
technology are: improvements in the quality of IP telephony; the Internet
phenomenon;
emergence of standards; cost-effective price-points for advanced services via
media-
rich call management, et cetera. One of the emerging standards in this area is
the well-
known H.323 protocol, developed by the International Telecommunications Union

CA 02385478 2002-03-18
WO 01/22766 PCT/SE00/01656
-2-
(ITU) for multimedia communications over packet-based networks. Using the
H.323
standard, devices such as personal computers can inter-operate seamlessly in a
vast
inter-network, sharing a mixture of audio, video, and data across all forms of
packet-
based network portions.
The H.323 standard defines four major types of components for forming an
inter-operable network: terminals, gateways, gatekeepers and Multipoint
Control Units
(MCUs). In general, terminals, gateways and MCUs of an H.323-based network are
referred to as "endpoints." Gateways are typically provided between networks
(or
network portions) that operate based on different standards or protocols. For
example,
one or more gateways may be provided between a packet-switched network portion
and a circuit-switched network portion. Terminals are employed by end-users
for
accessing the network or portions thereof, for example, for placing or
receiving a call,
or for accessing multimedia content at a remote site.
The gatekeeper is typically defined as the entity on the network that provides
address translation and controls access to the network for other H.323
components.
Usually, a gatekeeper is provided with the address translation capability for
a specified
portion of the network called a "zone." Typically, a zone comprises all
terminals,
gateways, and MCUs (that is, all endpoints) managed by a single gatekeeper.
Accordingly, a plurality of gatekeepers (sometimes referred to as a
"gatekeeper cloud")
may be provided for managing the entire network, each gatekeeper being
responsible
for a particular zone. In addition to address translation, gatekeepers may
also provide
other services to the terminals, gateways, and MCUs such as bandwidth
management
and gateway location.
Those of ordinary skill in the art should appreciate that although the current
VoIP networks offer rudimentary location services, they are not adequate for
the
mobility management required of a wireless network. In part, this deficiency
is due
to the condition that the gatekeeper which provides for call routing services
and the
registration of other H.323 entities within the VoIP network is typically
unaware of
conventional telecommunications terminals. While this condition is not a
problem for
fixed wireline telephones in terms of providing savings in long distance
charges (one
of the most important economic motivations behind IP-based call routing),
calls

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involving mobile stations (MSs) may still require establishing long distance
circuit-
switched trunks from one Mobile Switching Center (MSC) to another for routing.
For
example, when a call to a mobile subscriber is received in a gateway MSC
(GMSC),
it queries a Home Location Register (HLR) of the mobile subscriber for the
location
of the serving MSC. The HLR, in turn, queries the serving MSC {or, visited MSC
or
VMSC) for a Temporary Location Directory Number (TLDN) for routing the call to
the mobile subscriber. The TLDN is then passed to the GMSC for routing the
call
using circuit-switched trunks. Since the mobile station is no longerpresent in
its home
area, the GMSC-VMSC call leg can be a long distance call between two
neighboring
regions such as LATAs, two LATAs geographically separated from each other, or
across a continent. Clearly, routing such long distance call segments over CSN
portions defeats the rationale behind the use of VoIP network portions in
integrated
telecommunications networks having CSN portions.
WO 9933250 teaches a method and apparatus for providing voice
communication between a mobile station and a mobile radio network. If the
mobile
station can operate in voice mode, a circuit switched communication on a
traffic
channel is established between the mobile station and the network. If not, the
incoming voice call is routed to a voice gateway that converts the call to
data packets
and routes the packets through an IP communication network to a packet gateway
of
the mobile radio network. The packets are forwarded to the destination mobile
station
using a packet data service.
WO 9908890 generally describes a system for use within a company capable
of routing voice calls over an IP network or over a PSTN. The system can route
a call
from a first location to a second location via the IP network, and from the
second
location to the destination via the PSTN.
It should also be understood that there are situations in some integrated
telecommunications networks where long distance call segments are used even
when
the mobile subscriber is not roaming. For example, when the MSC that services
the
call originating mobile station (MS) and the MSC serving the home area of the
terminating MS (that is, home gateway MSC) are situated in two different
regions
(e.g., LATAs), and the terminating MS is located in its home area, the call
path still
AMENDED SHEET
CA 02385478 2002-03-18

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involves a CSN-based inter-MSC long distance trunk. Again, the economic
advantages of a VoIP network are not achieved in such situations. Similarly,
calls
between a mobile station and a wireline phone served by a Local Exchange (LE)
of the
PSTN are also typically muted over CSN-based long distance trunks if di~'erent
geographic regions are involved. Providing for IP-based call routing in such
situations
also gives rise to savings in long distance charges.
Based on the foregoing, it can be readily appreciated that there is an acute
need
for a solution that provides IP-based call routing so that the benefits of
integrating
VoIP network portions with CSN portions in an integrated telecommunications
network are realized. The present invention provides such a solution.
SUM1MARY OF THE INVENTION
AMENDED SHEET
CA 02385478 2002-03-18

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In one aspect, the present invention is directed to an integrated
telecommunications network having a packet-switched network portion (e.g., a
Voice-over-Internet Protocol (VoIP) network portion) and one or more circuit-
switched network (CSN) portions such as a PSTN or a radio telephony network. A
Mobile Switching Center (MSC) serving one or more mobile subscribers is
provided
with an Internet Protocol (IP)-Interworking Unit interface towards the VoIP
network
portion. The radio telephony network also includes a Location Server (LS)
containing
mapping information between routing numbers (e.g., Temporary Location
Directory
Numbers or TLDNs), called party numbers (B-numbers) and IP addresses of
entities
to which a call can be routed over an IP trunk from the MSC. A querying
mechanism
is provided in the MSC for interrogating the LS based upon a routing number, a
called
party number, or both, provided to the MSC. The MSC obtains an IP address from
the
LS which is used for effectuating the IP trunk. A plurality of Bearer
Independent Call
Control (BICC) messages and a plurality of Integrated Services Digital Network
(ISDI~ User Part (ISUP) messages are transmitted among the various nodes of
the
integrated telecommunications network, e.g., one or more MSCs with the IP
interfaces,
a Local Exchange of the PSTN, etc. for establishing the IP trunk. Where an IP
trunk
segment is not available or possible, a circuit-switched path such as, e.g, a
Synchronous Transfer Mode (STM) trunk, is used for completing the call routing
path.
In another aspect, the present invention is directed to several embodiments of
an IP-based long distance call routing method. In one embodiment, the call
routing
method relates to routing a call originated by a PSTN phone to an MS disposed
in the
integrated telecommunications network comprising the infrastructure as set
forth
above. In another embodiment, the call routing method relates to routing a
call from
an MS to a PSTN phone served by a Local Exchange. In yet another embodiment,
the
call routing method of the present invention is directed to routing a call
originated by
an MS to another MS that is located in its home area. In a still further
embodiment,
the call routing method relates to routing an MS-originated call to an MS that
is
located outside its home area.
BRIEF DESCRIPTION OF THE DRAWINGS

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A more complete understanding of the present invention may be had by
reference to the following Detailed Description when taken in conjunction with
the
accompanying drawings wherein:
FIG. 1A depicts a functional block diagram of an integrated
telecommunications network provided in accordance with the teachings of the
present
invention;
FIGS. 1B and 1C depict two scenarios, respectively, of a routing scheme for
MS-to-MS calls wherein the called MS is located in a home system of an
integrated
telecommunications network;
FIGS. 2A and 2B depict a flow chart of a call routing method for MS-to-MS
calls wherein the called MS is located in a home system;
FIGS. 3A and 3B depict two scenarios, respectively, of a routing scheme for
MS-to-MS calls wherein the called MS is roaming in a visited system of an
integrated
telecommunications network;
FIGS. 4A - 4D depict a flow chart of a call routing method for MS-to-MS calls
wherein the called MS is roaming;
FIGS. 5A and SB depict two scenarios, respectively, of a routing scheme for
PSTN-to-MS calls in an integrated telecommunications network;
FIGS. 6A - 6C depict a flow chart of a call routing method for PSTN-to-MS
calls in an integrated telecommunications network;
FIGS. 7A and 7B depict two scenarios, respectively, of a routing scheme for
MS-to-PSTN calls in an integrated telecommunications network; and
FIGS. 8A - 8C depict a flow chart of a call routing method for MS-to-PSTN
calls in an integrated telecommunications network.
DETAILED DESCRIPTION OF EMBODIMENTS
In the drawings, like or similar elements are designated with identical
reference
numerals throughout the several views, and the various elements depicted are
not
necessarily drawn to scale. Refernng now to FIG. 1 A, depicted therein is a
functional
block diagram of an integrated telecommunications network 10 provided in
accordance with the teachings of the present invention. It should be
appreciated that

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the integrated telecommunications network 10 is provided herein in order to
exemplify
the network-level infrastructure used in the various call routing scenarios
described in
greater detail hereinbelow.
The integrated telecommunications network 10 comprises a PSN portion 14
such as, for example, an H.323-based Voice-over-IP (VoIP) portion, that is
coupled
to a plurality CSN portions including, for example, one or more wireless
telephony
network portions (e.g., WL-CSN portions 12A and 12B) and a PSTN portion S0.
It should be readily apparent that the wireless CSN portions of the integrated
telecommunications network 10 may be realized in any known radio telephony
technology, for example, a Time Division Multiple Access (TDMA), et cetera.
The
WL-CSN portion 12A is shown in greater detail. A Home Location Register (HLR)
16 is provided for maintaining a subscriber profile or record associated with
a mobile
subscriber / mobile station (MS) 28. A Radio Base Station (RBS) 26 is included
as
part of the cellular infrastructure that comprises the WL-CSN portion 12A, in
order
to provide radio access services to the MS 28. A serving system 20, comprising
a
Visitor Location Register (VLR) 22 and a Mobile Switching Center (MSC) 24, is
also
included as a switching node therewith.
In accordance with the teachings of the present invention, an IP-Interworking
Unit (IWU) is provided as a hardware/firmware platform for interfacing and
interworking between the switching node (i.e., the MSC/VLR combination in this
exemplary embodiment) of the WL-CSN portion 12A and the PSN portion 14.
Preferably, the IP-IWU 30A is provided as an IP interface to the MSC 24, and
includes
appropriate media gateway (MGW) functionality for carrying voice traffic
(i.e.,
payload) over the IP-based PSN portion 14.
In addition, a Location Server (LS) 18 is provided as an entity within the WL-
CSN portion 12A that operates as a query-able database containing, preferably,
mappings between CSN-based routable numbers (e.g., a called party's number
(i.e.,
the B-number) or a Temporary Location Directory Number or TLDN) and an IP-
network address of a signaling endpoint (e.g., an MSC having the IP-IWU
interface)).
Preferably, the switching node (e.g., the MSC/VLR combination 24/22 or the MSC
24
separately) includes a hardware/software/firmware-based LS-query function 25
that

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facilitates interrogation by the MSC of the LS 18. Further, the LS is
preferably
configured in such a way that it returns a unique IP-network address of a
"destination
MSC" which, in some instances, may be an MSC that serves a called MS. On the
other hand, if the MSC associated with the called MS does not have an IP
interface
(i.e., not IP-addressable), the LS is configured so as to return the IP
address of an MSC
that is located closest thereto. With respect to PSTN calls, the destination
MSC is the
terminating IP signaling point connected to a Local Exchange (LE) disposed in
the
PSTN 50.
Based on the foregoing, it should be readily appreciated that the provision of
a database as set forth above in conjunction with a query function provided in
an IP-
capable MSC facilitates efficient call routing over an IP network without
having to
utilize the H.323 protocol, etc.
Set forth below in greater detail are a plurality of exemplary call routing
scenarios for routing calls in an integrated telecommunications network such
as the
network described hereinabove. More specifically, the following scenarios are
provided:
(A) MS-to-MS call routing where the called MS is in its home system;
(B) MS-to-MS call routing where the called MS is roaming;
(C) PSTN-to-MS call routing; and
(D) MS-to-PSTN call routing.
Furthermore, each of the scenarios is provided in two exemplary embodiments,
depending on certain conditions as will be described hereinbelow.
In the ensuing portions of the Detailed Description, various arrangements of
an integrated telecommunications network are presented that are appropriate
for the
different call routing schemes, in addition to the necessary message flows
wherein
signaling message paths are shown in broken lines and payload-bearing paths
are
depicted using solid lines.
(A) MS-TO-MS CALL ROUTING WHERE THE CALLED MS IS IN ITS HOME
SYSTEM

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First, the network arrangements for two exemplary embodiments are described.
The MS-to-MS call routing method is then explained in greater detail.
Refernng now to FIG. 1B, depicted therein is an integrated
telecommunications network 100 for effectuating a routing scheme for an MS-to-
MS
call in accordance with the teachings of the present invention. Three
geographic
regions, regionl 102A, region2 102B and region3 102C, form a coverage area of
the
network 100. Region3 102C is not involved in the call routing scenario
contemplated
herein and accordingly, will not be described in this section.
A call-originating party, MS 1 108A, is located in regionl 102A, and is served
byMSCl/VLRI 104A. AnRBSl 106Aprovides radio access services to MS1 108A.
A call- terminating party, MS2 108B, is located in region2 (home area for MS2
in this
exemplary scenario), and is served by MSC2/VLR2 104B. Also, an RBS2 106B is
included for providing radio access to MS2 108B.
MSC 1 and MSC2 are provided with a suitable IP-IWU as described above.
Also, a Location Server (LS 112) is provided within the network 100. Because
MSC1/VLRl and MSC2/VLR2 are located in two different geographic areas, the
MS1-MS2 call is a long distance call. For illustrative purposes, MSC2 is
treated as
both the home gateway MSC and serving MSC of MS2. A signaling path 114 is
provided between MSCI and LS 112. Also, another signaling path 116 is provided
between MSC1 and MSC2. An IP trunk path 118 is established therebetween for
routing the voice payload associated with the call.
FIG. 1 C depicts the network 100 in a form that is essentially identical to
the
network arrangement described above, except that the called party, that is MS2
108B,
is served by an MSC (MSC3/VLR3 104) that has no direct IP-IWU interface. MSC2
operates as the destination MSC, the MSC with IP interface that is closest to
the
serving MSC (i.e., MSC3). Accordingly, an additional signaling path 124 and a
circuit-switched trunk such as a Synchronous Transfer Mode (STM) trunk 122
(e.g.,
T1 or El) are established between the destination MSC and the serving MSC. The
STM trunk 122 is used in conjunction with the IP path 118 for transporting the
voice
payload.
FIGS. 2A and 2B depict a flow chart that describes a call routing method for

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the two exemplary network arrangements set forth above. It should be
appreciated that
the various messages depicted in FIGS. 1B and 1C are used in setting forth the
steps
of the flow chart. Accordingly, FIGS. 1 B and 1 C may again be referred to in
connection with the flow chart shown in FIGS. 2A and 2B.
After MSC1 receives a call initiation from MSlincluding MS2's B-number
(step 202), MSC1 performs a number analysis on the B-number (step 204) to
determine if the call to MS2 is a long distance call (decision block 206). If
it is not a
long distance call, the call maybe completed using conventional local call
termination
procedures (step 208).
If MSC1 determines that the call involves a different region (i.e., a long
distance call), it interrogates the Location Server by sending a Service
Location
Protocol (SLP) message (SERYICEREQ), together with the B-number of the called
party (step 210) to query the IP address of the destination MSC (MSC is IP-
capable
and its IP address is provided in the LS's database), which can also be the
serving
MSC (as illustrated in FIG. 1B). The destination MSC is used as a transit MSC
(as
illustrated in FIG. 1C) in the case where serving MSC (MSC3 in this case) is
not IP
capable. The Location Server, in response, returns the IP address of MSC2,
which is
provided as the IP-capable MSC, via a servicereq message to MSC1 (steps 214
and
224). Depending on whether the serving MSC for the called MS is IP-capable or
not,
two scenarios emerge.
Where MSC2 is the IP-capable serving MSC for MS2, MSC 1 sends a Bearer
Independent Call Control (BICC) message called Initial Address Message (IAM)+
to
MSC2, including the B-number and the IP address of MSC1 (step 216). The IAM+
message is essentially a modified N-ISDN User Part (N-ISUP) IAMmessage,
provided
to effectuate the signaling as set forth herein. Upon receiving the IAM+
message, a
BICC message called Address Complete Message (ACM)+ is sent back to MSC 1 to
by MSC2 (step 218). Thereafter, a BICC message called Answer Message (ANM)+
is sent by MSC2 to MSC1 (step 220). Subsequently, the IP trunk 118 is
established
between MSC1 and MSC2 via Real-time Transfer Protocol (RTP) and Session
Description Protocol (SDP) to convey the voice payload (step 222) associated
with the
call. These actions are shown in a consolidated step 222.

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Where MSC2 is only the IP-capable destination MSC because the serving
MSC (MSC3) is not IP-capable, an IAM+ message is also initially sent from MSCl
to MSC2, after receiving the result from the LS (step 226). Subsequently, an
ISUP
IAMmessage is forwarded by MSC2 to MSC3 (step 228). MSC3 then sends anACM
message to MSC2 as an acknowledgment of the IAMmessage (step 230). Thereafter,
upon receiving the ACM message, MSC2 initiates a BICC ACM+ message to MSC 1
(step 232). The ISUP ANMmessage is then sent by MSC3 to MSC2 (step 234), which
is forwarded by MSC2 to MSC 1 by sending the BICC ANM+ message (step 236). The
STM trunk 122 is thereby established between MSC2 and MSC3. The IP trunk 118
is subsequently established between MSC 1 and MSC2 via RTP and SDP to convey
the
voice payload as shown in the consolidated step 238.
(B) MS-TO-MS CALL ROUTING WHERE THE CALLED MS IS ROAMING
FIGS. 3A and 3B depict the relevant network arrangements for effectuating
MS-to-MS call routing where the called MS is roaming. It is clear that the
network
arrangements shown herein are similar to those described above. Accordingly,
only
the salient features of FIGS. 3A and 3B are set forth herein.
The called MS 108B is no longer located in its home area, that is, region2
102B. Rather, it is now located in region3, being served by MSC4NLR4 104D. An
RBS4 106D is included in the infrastructure of the region for providing radio
access
services to MS2. MSC2 is still provided as the home gateway MSC of MS2.
Whereas
in FIG. 3A, each of the call-originating MSC (MSC 1), home gateway MSC (MSC2),
and the serving MSC (MSC4) have IP interfaces, FIG. 3B depicts the scenario
where
the serving MSC (MSC4) does not have an IP interface and, accordingly, has to
connect to a destination MSC (MSC3 in this exemplary embodiment; MSC3 is the
IP-
capable MSC that is geographically closest to MSC4) via an Inter-Exchange
Carrier
(IXC) S l OB that is disposed between region2 and region3. Consequently, the
network
arrangement in FIG. 3B depicts STM trunks (path 170 and path 172) for the IXC
connection. FIG. 3A, on the other hand, illustrates a direct IP connection 162
between
the originating MSC and serving MSC for call routing.

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FIGS. 4A - 4D depict a flow chart for the call routing scenarios described
above. Once again, FIGS. 3A and 3B may be referred to for locating appropriate
signaling messages referenced in the flow chart. Further, it should be
appreciated that
most of the steps effectuated in this flow chart are similar to the steps
described in the
flow chart of FIGS. 2A and 2B. Accordingly, a concise description of the call
routing
method for each of the scenarios is set forth below without explicitly
refernng to the
reference numerals of the flow chart shown in FIGS. 4A - 4D.
Where the serving MSC (MSC4) is provided with the IP interface, the call
routing steps are as follows. After MSC 1 receives a call initiation from MS 1
including
MS2's B-number, MSC 1 performs B-number analysis to determine if the call is a
long
distance call. If so, MSC 1 interrogates LS by sending the SLP SER VICEREQ
message
including the B-number to query the IP address of the destination MSC. Since
MSC2
is provided to be the home gateway MSC of MS2, the SLP servicereg message
(transmitted back to MSC 1 by LS) includes the IP address of the home gateway
MSC,
i.e., MSC2. A BICC IAM+ message is then sent by MSC 1 to MSC2, including the B-
number and MSC1's IP address.
Upon receiving the IAM+ message, MSC2/VLR2 checks its record and
determines that MS2 is not in its home system, i.e., MS2 has roamed out. An
ANSI-
41 LOCREQ is transmitted by MSC2 to HLR to query the location of the serving
MSC
that currently serves MS2. Upon receiving the LOCREQ message, HLR verifies the
active services and queries the serving MSC4/VLR4 in region3 by transmitting
an
ANSI-41 ROUTREQ message. The pre-routing call setup is done by MSC4 by means
of paging MS2. The serving MSC4/VLR4 replies with an ANSI-41 routreq message
containing the routing number (TLDN). Thereafter, HLR sends the answer message
locreq including the TLDN to MSC2.
An SLP SERVICEREQ message including the TLDN of MSC4 is then sent by
MSC2 to LS to query the IP address of MSC4. The returned servicereq message
from
LS contains the IP address of MSC4. Thereafter, a BICC IAM+ message is sent by
MSC2 to MSC4. After receiving the IAM+ message, a BICC ACM+ message is sent
to MSC2 by MSC4 to acknowledge the IAM+ message. A BICC ACM+ message is
then forwarded by MSC2 to MSC1 as an acknowledgment. Afterwards, a BICC

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ANM+ message is sent to MSC2 byMSC4, which is subsequently forwarded to MSC 1
by MSC2. The direct IP trunk between MSC 1 and MSC4 is then established via
RTP
and SDP to convey the voice payload associated with the call.
Regarding the case where the serving MSC does not have an IP address, the
call routing process is essentially similar to the above up to the SLP
SERVICEREQ
message sent by MSC2 to LS to query the IP address of MSC4. Since in this
exemplary embodiment none of the MSCs in region3 are provided with a direct IP
connection, a destination MSC (i.e., MSC3) is found in region2. Thereafter, an
STM
trunk is established via the IXC between MSC3 and MSC4, in addition to the IP
trunk
between MSC1 and MSC3, for the purpose of call routing. It should, however, be
understood that in other variations of the present invention, a destination
MSC may
be provided within region3, thereby obviating the need for the IXC.
(C) PSTN-TO-MS CALL ROUTING
FIGS. 5A and SB depict network arrangements for effectuating two exemplary
embodiments of a PSTN-to-MS call routing scheme in accordance with the
teachings
of the present invention. Once again, only the essentials are set forth herein
because
the components of the network arrangements are similar to the components
described
above.
A PSTN phone 504 served by a Local Exchange (LE1) 508 is provided as the
call-originating entity in regionl 102A. MS 1 108A is provided as the call-
terminating
party and, in this exemplary embodiment, is roaming out of its home system
(MSC2/VLR2 104B) provided in region 2. MSC3/VLR3 104C is provided as the
serving system for MS 1 in the visited region, region3 102C. An IXC 51 OA is
provided
between regionl and region2 for establishing an STM trunk (first circuit-
switched
trunk path involving trunk segments 564 and 568) between LE1 and MSC2.
In the scenario exemplified in FIG. 5A, the serving MSC (MSC3) is provided
with an IP interface (i.e., IP-IWLJ interface). However, in the scenario
illustrated in
FIG. 5B, the serving MSC (MSC4) is not IP-capable and, therefore, a separate
destination MSC (MSC3) is provided in FIG. 5B. Accordingly, another STM trunk

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(second circuit-switched trunk path 582) is established between MSC3 and MSC4
for
the scenario illustrated in FIG. 5B. While the destination MSC (IP-capable MSC
that
is closest to the serving MSC) is provided within region3 (where the serving
MSC is
also located), one of ordinary skill should understand that in some other
exemplary
embodiments of the present invention, the destination MSC may be outside the
region
of the serving MSC, thereby necessitating the use of another IXC.
FIGS. 6A - 6C depict a flow chart for a call routing scheme for the network
arrangements set forth above. The steps provided in the flow chart are
believed to be
self explanatory and, in large part, similar to the steps of the call routing
methods
already set forth hereinabove. Accordingly, a concise description of the flow
chart is
provided below.
With respect to the case where the serving MSC is provided to be IP-capable,
an exemplary embodiment of the PSTN-to-MS call routing scheme is as follows.
After LE 1 receives a call initiation from the PSTN phone including MS 1's B-
number,
LE1 performs a number analysis and routes the call to MS1's home gateway MSC
(MSC2) via IXC by sending an IAMmessage. After receiving the call, an ANSI-41
LOCREQ message is sent by MSC2 to HLR to query the location of the serving
MSC.
Upon receiving the LOCREQ message, HLR verifies the active services and
queries
the serving MSC3/VLR3 with an ANSI-41 ROUTREQ message. The pre-routing call
setup is done by MSC3 by means of paging MS1.
The serving MSC3 replies with an ANSI-41 routreq message which includes
the routing number (i.e., TLDN). Upon receiving the routreq message, HLR sends
the
locreq message including the TLDN to MSC2. Thereafter, MSC2 sends an SLP
SERVICEREQ message containing the TLDN of the serving MSC3 to LS in order to
query the IP address of MSC3. A servicereq message is sent back to MSC2 from
LS
with MSC3's IP address. A BICC IAM+ message is then sent by MSC2 to MSC3. In
response, a BICC ACM+ message is returned byMSC3 to MSC2 in order to
acknowledge the IAM+ message. Upon receiving the ACM+ message, MSC2 sends
an ISUP A CMmessage to LE 1 via IXC to acknowledge the IAMmessage sent by LE
1.
Afterwards, a BICC ANM+ message is sent to MSC2 by MSC3. MSC2 then sends an
ISUP ANM message to LE1 via IXC. Thereafter, an STM trunk between LE1 and

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MSC2 is established, in addition to an IP trunk between MSC2 and MSC3 via RTP
and SDP.
With respect to the scenario where the serving MSC does not have an IP
interface, the call routing process is essentially similar up to the
servicereq message
from LS which now includes the IP address of the destination MSC (i.e., MSC3).
As
can be seen in FIG. 5B, additional ISUP and BICC messaging is done in order to
effectuate an STM trunk between MSC3 and MSC4. The call leg between MSC2 and
MSC3 is still routed over an IP trunk via RTP and SDP.
(D) MS-TO-PSTN CALL RouTING
FIGS. 7A and 7B depict two network arrangements for effectuating MS-to-
PSTN call routing. In FIG. 7A, an IP-capable MSC (MSC3) 104C within region3 is
used as a destination MSC to route the call via the IP network to the called
PSTN
phone 586 (served by LE3 584). In FIG. 7B, since there are no IP-capable MSCs
in
region3, MSC2 in region 2 (which is the closest IP-capable MSC to LE3) is used
as
a destination MSC to route the call via the IP network to the called PSTN
phone 586.
FIGS. 8A - 8C depict a flow chart for an exemplary embodiment of the MS-to-
PSTN call routing scheme for the network arrangements set forth above. Again,
only
a concise account thereof is set forth below.
After MSC1 receives a call initiation from MS1 together with the PSTN
phone's B-number, MSC 1 performs a number analysis to determine if the call is
a long
distance call. If so, MSC 1 interrogates LS by sending an SLP SER VICEREQ
message
including the B-number to query the IP address of the destination MSC if the
call is
IP-routable. Since the B-number is a PSTN number and LE has no direct IP
connection, a servicereq message including the IP address of MSC3 (closest MSC
with IP capability in region3 with respect to LE3) is sent back to MSC1 from
LS. A
BICC IAM+ message is then sent by MSC 1 to MSC3, including the B-number and
the
IP address of MSC 1. Upon receiving the IAM+ message, MSC3/VLR3 sends an IAM
message to LE3. An ACM message is sent back by LE3 to MSC3 as an
acknowledgment to the IAMmessage. After receiving the A CMmessage, MSC3 sends

CA 02385478 2002-03-18
WO 01/22766 PCT/SE00/01656
-15-
a BICC ACM+ message to acknowledge the IAM+ message. An ISUP ANMmessage
is sent thereafter by LE3 to MSC3. Once MSC3 receives the ANM message, it
sends
a BICC ANM+ message to MSC 1. Subsequently, an IP trunk is established between
MSC 1 and MSC3 via RTP and SDP, and a circuit-switched STM trunk (first
circuit-
switched trunk) is established between MSC3 and LE3 for carrying the voice
payload.
Where there are no IP-capable MSCs in region3 (as illustrated in FIG. 7B), the
SLP servicereq message from LS contains the IP address of MSC2, which is the
geographically closest MSC to LE3. A BICC IAM+ message is then sent by MSC1
to MSC2 together with the PSTN phone's B-number and the IP address of MSC1.
After additional ISUP and BICC messaging as shown in FIG. 7B, an STP trunk
(second circuit-switched trunk) is established between MSC2 and LE3 via the
IXC,
in addition to the IP trunk between MSC l and MSC2, for the purpose of call
routing.
Based upon the foregoing, it should be appreciated by those of ordinary skill
in the art that the present solution advantageously provides an IP-based call
routing
scheme for use with an integrated telecommunications network having a PSN
portion
that is coupled to one or more CSN portions (wireless, wireline, or both). It
should be
apparent that the present invention efficiently utilizes the IP "backbone" for
routing
long distance calls by providing a cellular infrastructure entity (i.e., a
Location Server)
that includes a query-able database containing mapping data between mutable
numbers
and IP addresses of entities provided with an IWL1 interface. It should
further be
appreciated that the IP call routing provided herein does not involve the
H.323
protocol. Also, because the Location Server is provided to be a cellular
component
that can interface with IP-capable MSCs, current cellular infrastructures may
be
leveraged to a greater extent in integrated telecommunications networks as the
components can be retrofitted with appropriate IP interfaces etc.
Further, it is believed that the operation and construction of the present
invention will be apparent from the foregoing Detailed Description. While the
method
and system shown and described have been characterized as being preferred, it
should
be readily understood that various changes and modifications could be made
therein
without departing from the scope of the present invention as set forth in the
following
claims.

Representative Drawing
A single figure which represents the drawing illustrating the invention.
Administrative Status

2024-08-01:As part of the Next Generation Patents (NGP) transition, the Canadian Patents Database (CPD) now contains a more detailed Event History, which replicates the Event Log of our new back-office solution.

Please note that "Inactive:" events refers to events no longer in use in our new back-office solution.

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Event History

Description Date
Inactive: IPC from PCS 2022-01-01
Inactive: IPC expired 2022-01-01
Inactive: IPC deactivated 2017-09-16
Inactive: IPC deactivated 2017-09-16
Inactive: IPC assigned 2016-03-02
Inactive: IPC assigned 2016-03-02
Inactive: IPC assigned 2016-03-02
Inactive: IPC assigned 2016-03-02
Inactive: First IPC assigned 2016-03-02
Inactive: IPC assigned 2016-03-02
Inactive: IPC assigned 2016-03-02
Inactive: IPC expired 2009-01-01
Inactive: IPC expired 2009-01-01
Inactive: IPC from MCD 2006-03-12
Application Not Reinstated by Deadline 2004-08-30
Time Limit for Reversal Expired 2004-08-30
Deemed Abandoned - Failure to Respond to Maintenance Fee Notice 2003-08-29
Amendment Received - Voluntary Amendment 2002-12-09
Inactive: Applicant deleted 2002-11-14
Letter Sent 2002-11-14
Inactive: Office letter 2002-10-08
Letter Sent 2002-10-07
Inactive: Delete abandonment 2002-10-07
Inactive: Single transfer 2002-09-27
Appointment of Agent Requirements Determined Compliant 2002-09-26
Inactive: Office letter 2002-09-26
Inactive: Office letter 2002-09-26
Revocation of Agent Requirements Determined Compliant 2002-09-26
Inactive: Cover page published 2002-09-11
Inactive: Courtesy letter - Evidence 2002-09-10
Revocation of Agent Request 2002-09-06
Appointment of Agent Request 2002-09-06
Inactive: Notice - National entry - No RFE 2002-09-06
Inactive: Inventor deleted 2002-09-06
Deemed Abandoned - Failure to Respond to Maintenance Fee Notice 2002-08-29
Application Received - PCT 2002-06-18
National Entry Requirements Determined Compliant 2002-03-18
Request for Examination Requirements Determined Compliant 2002-03-18
All Requirements for Examination Determined Compliant 2002-03-18
Application Published (Open to Public Inspection) 2001-03-29

Abandonment History

Abandonment Date Reason Reinstatement Date
2003-08-29
2002-08-29

Maintenance Fee

The last payment was received on 2002-03-18

Note : If the full payment has not been received on or before the date indicated, a further fee may be required which may be one of the following

  • the reinstatement fee;
  • the late payment fee; or
  • additional fee to reverse deemed expiry.

Please refer to the CIPO Patent Fees web page to see all current fee amounts.

Fee History

Fee Type Anniversary Year Due Date Paid Date
Request for examination - standard 2002-03-18
Basic national fee - standard 2002-03-18
MF (application, 2nd anniv.) - standard 02 2002-08-29 2002-03-18
2002-09-06
Registration of a document 2002-09-27
Owners on Record

Note: Records showing the ownership history in alphabetical order.

Current Owners on Record
TELEFONAKTIEBOLAGET LM ERICSSON (PUBL)
Past Owners on Record
ZENG-JUN XIANG
Past Owners that do not appear in the "Owners on Record" listing will appear in other documentation within the application.
Documents

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Document
Description 
Date
(yyyy-mm-dd) 
Number of pages   Size of Image (KB) 
Representative drawing 2002-09-10 1 7
Description 2002-03-18 16 826
Claims 2002-03-18 11 532
Drawings 2002-03-18 21 395
Abstract 2002-03-18 1 78
Cover Page 2002-09-11 2 63
Reminder of maintenance fee due 2002-09-09 1 110
Notice of National Entry 2002-09-06 1 192
Acknowledgement of Request for Examination 2002-10-07 1 177
Courtesy - Certificate of registration (related document(s)) 2002-11-14 1 109
Courtesy - Abandonment Letter (Maintenance Fee) 2003-10-27 1 176
PCT 2002-03-18 22 969
Correspondence 2002-09-06 1 27
Correspondence 2002-09-06 1 31
Correspondence 2002-09-26 1 19
Correspondence 2002-09-26 1 21
Correspondence 2002-10-07 1 14