System and method for enhanced internet service connections

Telephonic communications – Special services – Service trigger

Reexamination Certificate

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Details

C379S221090, C379S900000

Reexamination Certificate

active

06519333

ABSTRACT:

BACKGROUND
1. Field of the Invention
The present invention relates generally to telecommunications systems. More particularly, the present invention relates to an advanced intelligent network system providing enhanced Internet service connections.
2. Background of the Invention
Over the last ten years, use of the Internet has grown rapidly. A large segment of this growth stems from an increase in individual dial-up subscribers. These dial-up subscribers use the public switched telephone network (“PSTN”) to establish connections to their Internet Service Providers (“ISPs”).
FIG. 1
is a schematic diagram illustrating how these dial-up subscribers, or users, connect to their ISPs using PSTN
10
. To support multiple connections, ISPs must maintain numerous telephone lines connected to modems. Rather than advertising a different telephone number for each telephone line, ISPs generally advertise a limited number of telephone access numbers. Each telephone access number corresponds to one or more telephone lines. These telephone lines may be made up of, e.g., individual plain old telephone service (“POTS”) lines, one or more T
1
lines, or Primary Rate Integrated Services Digital Network (“PRI”) lines. For simplicity, the figures and discussion herein show the connection to be made up of PRI lines
21
, as shown in FIG.
1
.
PRI lines
21
lead to ISP
20
where they are connected to multi-line hunt group (“MLHG”)
22
as shown in FIG.
1
. MLHG
22
is a modem pool allowing multiple simultaneous connections and is controlled by access server
23
. MLHG
22
takes incoming subscriber calls and routes them to the first open modem in the modem pool. When caller
30
dials the telephone access number for ISP
20
(using computer
31
, modem
32
and subscriber line
33
), PSTN
10
processes the call like any other call. That is, the call is routed between caller
30
and the called party (in this case, ISP
20
) through one or more switches. If the ISP's lines are all busy, or “off-hook,” i.e., there are no voice communication paths available, the caller gets a busy signal, which is provided by PSTN
10
. On the other hand, if lines are available, the ISP's switch terminates the call and it is the ISP's responsibility to answer the call, verify the user authorization to access the ISP's system, and set up the caller's connection to the Internet.
When a call reaches ISP
20
via PRI lines
21
and MLHG
22
, access server
23
answers the call and determines whether the caller is a valid ISP subscriber. If the caller is a valid subscriber, then access server
23
must determine which services the caller should have access to. Access server
23
queries caller
30
for information such as a username and password for use in validating caller
30
and determining caller
30
's authorized services. The dialog between caller
30
and access server
23
is usually performed automatically between access server
23
and communications software operating on computer
31
.
Generally, ISPs use centralized servers to store and manage their subscriber databases. Remote Authentication Dial-In User Service (“RADIUS”) server
24
, having database
24
a
, shown in
FIG. 1
, is functionally connected to access server
23
and provides this centralized management. Thus, access server
23
collects username and password information from caller
30
and passes it on to RADIUS server
24
. After RADIUS server
24
verifies caller
30
's username and password, it provides access server
23
with configuration information specific to caller
30
. Access server
23
uses the configuration information to provide the authorized services to caller
30
. Access servers and RADIUS servers are described in more detail in commonly assigned U.S. patent application, Ser. No. 09/133,299, which is incorporated herein by reference in its entirety. Additional information on access servers and RADIUS servers may be found in Rigney et al.,
Remote Authentication Dial
-
In User Service
(
RADIUS
), Network Working Group, January 1997, or in Rigney et al.,
RADIUS Accounting
, Network Working Group, April 1997.
It is well known in the art that not all subscribers connect to their ISPs at the same time. Additionally, not all subscribers connect every day, nor do they connect for the same length of time each session. For this reason, it is not practical or realistic for ISPs to provide a 1:1 ratio of lines to subscribers. ISPs must pay their local telephone service providers for each telephone line maintained. Instead, ISPs have developed formulas to determine the appropriate number of telephone lines required. In general, a telephone line to user ratio of at least 1:10 provides an acceptable level of service. However, as Internet usage continues to grow, it is becoming more difficult to predict the telephone line requirements for an ISP.
In the conventional system described above, all callers are given equal priority within telephone network
10
and by ISP
20
. That is, all calls are handled on a first-come, first-served basis. If the ISP has an open telephone line, the call is terminated and the ISP answers the call, regardless of whether or not the caller is a valid ISP subscriber. If the caller is a valid ISP subscriber, the caller gains access to the ISP's resources. Otherwise, RADIUS server
24
instructs access server
23
to disconnect the call. On the other hand, if the ISP does not have any open telephone lines, all callers, even valid ISP subscribers, are denied access because no calls can be connected to the ISP for verification.
FIG. 2
shows Service Switching Points (“SSPs”)
240
,
250
and
260
connected to MLHGs
222
a
,
222
b
and
222
c
via PRI lines
241
,
251
and
261
, respectively. SSP
240
hosts telephone access number 222-444-1000, SSP
250
hosts access number 222-555-1000 and SSP
260
hosts access number 222-666-1000. In conventional systems, when caller
230
attempts to connect to ISP
220
by dialing telephone access number 222-444-1000, SSP
211
sends a call setup message to SSP
240
. The call setup message is transmitted via Common Channel Signaling System 7 (“SS7”) network
213
. SSP
240
determines whether any lines are available going into MLHG
222
a
. If there are no lines available, i.e., all lines in PRI
241
are “off-hook,” caller
230
receives a busy signal.
ISP
220
has two additional telephone access numbers and corresponding MLHGs which caller
230
may use to obtain access.
FIG. 2
shows each telephone access number residing on individual SSPs. However, as would be apparent to those skilled in the art, an SSP can support multiple telephone access numbers. In conventional systems, if caller
230
's initial attempt to access ISP
220
results in a failed connection, caller
230
will have to redial either the same telephone access number or one of the additional numbers. Of course, caller
230
must be aware of the additional numbers and must reconfigure the communications software on computer
231
to dial the additional numbers.
Even if caller
230
is aware of and tries the other telephone access numbers there is little assurance that a line will be available and caller
230
's efforts may be wasted. For example, suppose caller
230
makes another attempt to connect to ISP
220
, this time by dialing 222-555-1000. As before, if there are no available lines going into MLHG
222
b
, the call is not terminated and caller
230
receives a busy signal. Even if a line is available and the call is terminated, i.e., connected, a subscriber will not have a successful connection if the ISP does not answer the call. As noted above, if the call is not successful, caller
230
will have to hang up and make another attempt to connect to the ISP. In this example, on caller
230
's third attempt, the telephone access number used is 222-666-1000. As described above, SSP
211
sends a call setup message to SSP
260
. In this example, at least one voice channel is available in PRI lines
261
going into MLHG
222
c
. In this case

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