Telephonic communications – Plural exchange network or interconnection – With interexchange network routing
Reexamination Certificate
2000-03-30
2002-05-14
Hong, Harry S. (Department: 2642)
Telephonic communications
Plural exchange network or interconnection
With interexchange network routing
C370S352000, C370S395510, C370S395600, C370S410000, C370S467000
Reexamination Certificate
active
06389130
ABSTRACT:
FIELD OF THE INVENTION
The invention relates to internetworking of multiple services using an asynchronous transfer mode (ATM) network. More particularly, the invention relates to use of dynamic ATM bearer trunking to support multiple services.
BACKGROUND OF THE INVENTION
Common Channel Signaling System No. 7 (SS7 or C7) is a global standard for telecommunications defined by the International Telecommunications Union (ITU) to define procedures and protocols by which network elements of public switched telephone networks (PSTNs) exchange information to provide call setup, routing and control. The ITU definition of SS7 allows for regional variations such as, for example, the American National Standards Institute (ANSI) and Bell Communications Research (Bellcore) standards in North America and the European Telecommunications Standards Institute (ETSI) standard used in Europe.
SS7 provides a framework in which telephone networks provide basic call setup, management, and tear down, wireless services, local number portability, enhanced call features (e.g., call forwarding, calling party name
umber information, three-way calling), etc. SS7 messages are exchanged between network elements over bi-directional channels called signaling links. These messages are communicated out-of-band on dedicated channels rather than in-band on voice channels. Out-of-band signaling provides faster call setup times, more efficient use of voice circuits, improved control over fraudulent network usage, and other advantages, compared to in-band signaling.
FIG. 1
is a block diagram of a Common Channel Signaling System No. 7 (SS7) network. In general, each signaling point in an SS7 network is uniquely identified by a numeric point code. Point codes are carried in signaling message between signaling points to identify the source and destination points for the message. Signaling points use routing tables to select an appropriate signal path for a message.
Service switching points (SSPs), such as SSPs
130
and
132
, are switches that originate, terminate, or relay calls. SSPs are typically located in end offices that are coupled to multiple telephones or other devices that use telephone service. Telephones, such as telephones
110
,
112
,
120
and
122
, are coupled to SSPs via local connections. Facsimile machines, modems and other devices can also be coupled to SSPs
130
and
132
. An SSP sends signaling messages to other SSPs to setup, manage, and release voice circuits required for a call. An SSP can also send a query message to a service control point (SCP), such as SCPs
170
and
172
, which acts as a database for certain types of calls, for example, 1-800/888 calls. The SCP sends a response to the originating SSP with routing information for the dialed number.
Traffic between SSPs can be routed by signal transfer points (STPs), such as STPs
140
,
142
,
150
,
152
,
160
and
162
. STPs operate to route incoming messages to an outgoing signal link based on routing information contained in an SS7 message. In other words, STs operate as network hubs and eliminate the need for direct links between signaling points. An STP can perform global title translation to determine a destination signaling point based on digits present in the signaling message.
Signaling links between signaling points are logically organized by link type according to the purpose of the link. Access (“A”) links
180
connect a signaling end point (e.g., an SCP or SSP) to an STP. Only messages originating from or destined to the signaling end point are transmitted by an access link. Bridge (“B”) links
182
connect an STP to another STP. Typically, a group of four bridge links interconnect peer (or primary) STPs (e.g., the STPs from one network to STPs of another network).
Cross (“C”) links
184
connect STPs performing identical functions into a mated pair. A cross link is used only when an STP has no other route available to a destination signaling point due to, for example, a link failure. Diagonal (“D”) links
186
connect secondary (e.g., local or regional) STP pairs in a quad-link configuration.
Extended (“E”) links
188
connect an SSP to an alternate STP. Extended links provide an alternate signaling path if an SSP's primary STP cannot be reached via an access link. Fully associated (“F”) links
190
can be used to connect two signaling end points (e.g., SSPs and SCPs). Fully associated links are generally not used in networks with STPs.
When a party initiates a call, the call is held at SSP servicing the caller. For example, if the party initiates the call from telephone
110
, the call is held at SSP
130
. SSP
130
then transmits the information necessary to locate the called location and determines if the called location is busy or available to accept the call. If the called party is telephone
112
, SSP
130
can directly determine whether telephone
112
is busy.
If the call destination is a telephone that is not coupled to SSP
130
, call information is routed through network
100
to the appropriate SSP. If, for example, the call destination is telephone
120
or telephone
122
, SSP
130
routes call information to SSP
132
. The call information can be routed directly between SSP
130
and SSP
132
by a fully associated link, if present. Otherwise, the call information can be routed, for example, to STP
142
via an access link to STP
152
via a diagonal link to STP
162
via a bridge link to SSP
132
via an access link. SSP
132
determines whether the destination telephone is available to receive the call and returns the appropriate information to SSP
130
.
If the destination telephone is available to receive the call, a trunk is established through network
100
from the call source to the call destination to establish a talk path and the call is established. When the call is completed, the trunk is torn down (call tear down) and the call is terminated. Connections through network
100
are established for each call in a similar manner.
Current PSTNs are based on designs and hardware from the 1970s. As the usage of these PSTNs changes because of, for example, Internet access and related activities, traditional PSTNs have become less optimal and other networking protocols have been used for specific purposes. For example, many telephone companies maintain both PSTNs and asynchronous transfer mode (ATM) networks for supporting various services. However, maintaining multiple networks with multiple protocols and hardware components is more time consuming and more expensive than maintaining a single network type. What is needed is a single network that can efficiently support multiple types of network services.
SUMMARY OF THE INVENTION
A switching device having a distributed architecture is described. The switching device includes a system controller coupled to multiple service modules. Each of the service modules has a first interface to receive and transmit telephone calls, a second interface to receive and transmit data according to a first protocol, and a third interface to transmit and receive data according to a second protocol. Telephone calls received via the first interface and data received via the second interface are converted to the second protocol and routed to an external device. Data received from the third interface is converted to one of the first protocol and a telephone protocol and routed to the second interface, if converted to the first protocol, and routed to the first interface if converted to the telephone protocol.
REFERENCES:
patent: 6169735 (2001-01-01), Allen, Jr. et al.
patent: 6181786 (2001-01-01), Detampel, Jr. et al.
patent: 6205214 (2001-03-01), Culli et al.
Alleman Andrew P.
Shenoda George
Columbia IP Law Group, PC
Hong Harry S.
Oresis Communications
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