U.S. patent application number 11/287861 was filed with the patent office on 2006-04-13 for internet telephony network and methods for using the same.
Invention is credited to Brian Fowler.
Application Number | 20060077971 11/287861 |
Document ID | / |
Family ID | 32849745 |
Filed Date | 2006-04-13 |
United States Patent
Application |
20060077971 |
Kind Code |
A1 |
Fowler; Brian |
April 13, 2006 |
Internet telephony network and methods for using the same
Abstract
A telephony network for enabling the origination and termination
of telephone calls between one or more subscriber terminals
connected to a public packet network with an open addressing
protocol and one or more non-subscriber terminals connected to a
PSTN network. Subscribers to the telephony network have E.164
numbers mapped to their current IP address and published by a
mapping server so other subscribers and non-subscribers can call
the subscriber. The mapped information is published on either a
HTML or XML page for direct use over the Internet by other
subscribers or for use by a gateway device. The gateway device
enables non-subscribers on the PSTN to directly call subscribers.
Calls between subscribers completely by-pass the PSTN. Subscribers
can also call anyone with a phone number whether they are a
subscriber or not.
Inventors: |
Fowler; Brian; (Sunrise,
FL) |
Correspondence
Address: |
FRIED, FRANK, HARRIS, SHRIVER & JACOBSON
1001 PENNSYLVANIA AVE., N.W., SUITE 800
WASHINGTON
DC
20004-2505
US
|
Family ID: |
32849745 |
Appl. No.: |
11/287861 |
Filed: |
November 28, 2005 |
Related U.S. Patent Documents
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Application
Number |
Filing Date |
Patent Number |
|
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10366372 |
Feb 14, 2003 |
6999458 |
|
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11287861 |
Nov 28, 2005 |
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Current U.S.
Class: |
370/352 |
Current CPC
Class: |
H04L 29/1216 20130101;
H04L 61/157 20130101; H04L 65/103 20130101; H04M 15/56 20130101;
H04L 61/10 20130101; H04L 29/12018 20130101; H04M 2215/202
20130101; H04L 29/06027 20130101; H04L 65/104 20130101; H04M 7/128
20130101; H04L 65/1069 20130101 |
Class at
Publication: |
370/352 |
International
Class: |
H04L 12/66 20060101
H04L012/66 |
Claims
1. A telephony network for selectively enabling the origination and
termination of telephone calls between a first subscriber terminal
having a first unique terminal identifier and a first unique
address and connected to a public packet network with an open
addressing protocol, and a second subscriber terminal having a
second unique terminal identifier and a second unique address and
connected to said public packet network, and a non-subscriber
terminal having a telephone number and connected to a PSTN network,
comprising: a gateway connected to said public packet network and
said PSTN network, said gateway being selectively operative to be
accessed over said public packet network by said first subscriber
terminal and said second subscriber terminal, said gateway having a
physical connection to said PSTN network that enables said gateway
to be accessed by said non-subscriber terminal over said PSTN
network; and a mapping server connected to said public packet
network, said mapping server being selectively operative to be
accessed over said public packet network by said first subscriber
terminal, said second subscriber terminal and said gateway, said
mapping server being selectively operative to receive said first
unique terminal identifier and said first unique address from said
first subscriber terminal, said mapping server being operative to
receive said second unique terminal identifier and said second
unique address from said second subscriber terminal, said mapping
server being selectively operative to provide said first unique
address to said second subscriber terminal in response to a first
request containing said first unique terminal identifier from said
second subscriber terminal, said mapping server being selectively
operative to provide said first unique address to said gateway in
response to a second request containing said first unique terminal
identifier from said gateway; said gateway being selectively
operative to enable voice communication between said second
subscriber terminal and said first subscriber terminal after said
gateway receives said first unique address in response to said
first request and said gateway establishes a connection between
said second subscriber terminal and said first subscriber terminal
over said public packet network, said gateway being selectively
operative to enable voice communication between said first
subscriber terminal and said non-subscriber terminal after said
gateway receives said first unique address in response to said
second request, and said non-subscriber terminal establishes a
connection to said gateway over said PSTN network, and said gateway
establishes a connection to said first subscriber terminal over
said public packet network, said gateway being selectively
operative to enable voice communication between said non-subscriber
terminal and said first subscriber terminal after said gateway
receives said telephone number from said first subscriber terminal,
and said first subscriber terminal establishes a connection to said
gateway over said public packet network, and when said gateway
establishes a connection to said non-subscriber terminal over said
PSTN network using said telephone number.
2. The telephony network recited in claim 1, wherein said gateway
is a software programmed hardware device.
3. The telephony network recited in claim 2, wherein said hardware
device is an add-on card within an existing telecommunications
system.
4. The telephony network recited in claim 1, wherein said gateway
is further operative to monitor voice communication between said
first subscriber terminal and said non-subscriber terminal for a
call length.
5. The telephony network recited in claim 1, wherein said mapping
server is operative to query said first subscriber terminal to
determine a first call status; said first unique address
corresponds to a primary call destination and an alternative call
destination for said first subscriber terminal, and wherein said
gateway is further operative to request and receive said call
status from said mapping server and to establish a connection
between said second subscriber terminal and either said primary
call destination or said alternative call destination based on said
call status.
6. The telephony network recited in claim 2, wherein said first
subscriber terminal and said second subscriber terminal are
selected from a group consisting of a computer, an IP-enabled
telephone, an IP-enabled PBX, an IP-enabled messaging system, an
IP-enabled handheld device, or an IP-enabled e-mail system.
7. The telephony network of claim 1 wherein at least said first
subscriber terminal is operative to connect to said public packet
network through a wireless connection to a wireless local network
in communication with said public packet network.
8. A process for notifying a user of an IP network of an IP address
from a plurality of IP addresses corresponding to a plurality of
terminals to facilitate telephony calls to at least one of said
terminals from said user over an IP network, comprising the steps
of: obtaining said IP addresses for said terminals; creating a
plurality of files containing said IP addresses and corresponding
to a plurality of users utilizing said terminals, said files being
created on a server connected to said IP network; publishing said
files on said server; updating periodically said files to reflect a
service status for one or more of said terminals; and allowing said
user to access at least part of one of said files.
9. The process recited in claim 8, wherein service status is
updated to a not-in-service status if a telephony call cannot be
connected to said terminal over said IP network.
10. A process of enabling the origination of a telephone call over
a packet network between a first subscriber terminal and a second
subscriber terminal connected to said packet network, comprising
the steps of: sending a signal to a gateway connected to said
packet network when said first subscriber terminal enters telephone
number data corresponding to said second subscriber terminal;
identifying an IP address corresponding to said telephone number
data stored within said gateway based on said signal; creating
mapping data for connecting said first subscriber terminal to said
second subscriber terminal over said packet network utilizing said
IP address; storing said mapping data in a memory of said gateway;
utilizing said mapping data to establish a connection over said
packet network between said first subscriber terminal and said
second subscriber terminal; and removing said mapping data from
said memory when said connection is terminated.
11. A telephony network for enabling the origination and
termination of telephone calls between a first terminal having a
first unique terminal identifier and a first unique address and
connected to a packet network, and a second terminal having a
telephone number and connected to a telephony network, comprising:
a gateway connected to said packet network and said telephony
network, said gateway being operative to be accessed over said
packet network by said first terminal, and said gateway having a
connection to said telephony network that enables said gateway to
be accessed by said second terminal over said telephony network;
and a mapping server connected to said packet network, said mapping
server being operative to be accessed over said packet network by
said first terminal and said gateway, said mapping server being
operative to receive said first unique terminal identifier and said
first unique address from said first subscriber terminal, said
mapping server being operative to provide said first unique address
to said gateway in response to a request containing said first
unique terminal identifier from said gateway, said mapping server
being operative to query said first subscriber terminal to
determine a call status for said first subscriber terminal; said
gateway being operative to enable voice communication between said
first subscriber terminal and said second terminal after said
gateway receives said first unique address in response to said
request, said second terminal establishes a connection from said
second terminal to said gateway over said telephony network, and
said gateway establishes a connection from said gateway to said
first subscriber terminal over said packet network; said gateway
being operative to enable voice communication between said second
terminal and said first subscriber terminal after said gateway
receives said telephone number from said first subscriber terminal,
said first subscriber terminal establishes a connection to said
gateway over said packet network, and said gateway establishes a
connection from said gateway to said non-subscriber terminal over
said telephony network using said telephone number.
12. The telephony network recited in claim 11, wherein said
telephony network is the PSTN and said second terminal is a
PSTN-enabled telephone.
Description
CROSS-REFERENCE TO RELATED APPLICATION
[0001] This application is a continuation of application Ser. No.
10/366,372, filed on Feb. 14, 2003, which is hereby incorporated by
reference in its entirety.
SUMMARY OF THE INVENTION
[0002] The present invention is generally related to the field of
Internet telephony, and particularly related to an Internet
Protocol ("IP") based telephony system that allows for the
origination and termination of phone calls to and from the Public
Switched Telephone Network ("PSTN"). The present invention involves
the mapping of subscribers' E.164 numbers to their current IP
address and publishing the mapped information in a distributed form
so other subscribers and non-subscribers can call the subscriber
without reliance on a central answering position. The mapped
information is published on either a Hyper Text Markup Language
("HTML") document or eXtended Markup Language ("XML") database for
direct use over the Internet by other subscribers or for use by a
gateway device. The gateway device enables non-subscribers on the
PSTN to directly call subscribers. Calls between subscribers
completely by-pass the PSTN. Subscribers can also call anyone with
a phone number whether they are a subscriber or not.
BACKGROUND OF THE INVENTION
[0003] Traditional telephone service is circuit-switched--every
call is transmitted over dedicated facilities reserved for that
particular call. Long distance calls are transmitted from the
user's phone line over copper wires to a local telephone company's
network switch, which converts the call to digital format and hands
it off to the long distance carrier. The long distance carrier then
routes the call over its network to a local telephone company's
network serving the call recipient. That carrier's local network
switch re-converts the call to analog signals and connects the call
to a dedicated line serving the recipient. As long as the call is
ongoing, a circuit--a dedicated splice of bandwidth--remains open
throughout all three networks involved in transmitting the voice
signals. Both the called party and the calling party pay a fee to
their local telephone company for access to each company's local
network, and the calling party pays an additional fee, usually on a
per minute basis, for use of the long distance carrier's
network.
[0004] Internet-based telephone services, or IP telephony,
presently offer significant benefits over traditional telephone
service. Although users generally pay a fee for Internet access, it
is usually a set amount on a monthly basis regardless of the amount
of bandwidth used. If two people use the Internet to call one
another, they can by-pass both local telephone companies' networks
and the long distance carrier's network, without incurring
additional fees for making such a call. This type of IP telephony,
that does not require any intermediary service (charged at some
price), is extremely inconvenient because it requires both users to
be using the same software at the same time and to know that the
other user is available on-line at the designated time of the call.
Other technologies have been developed to enhance the convenience
of calling, but services utilizing these technologies often require
additional hardware devices to be installed on each user's system
and have introduced other issues, such as service quality, which
will be discussed more fully below.
[0005] Internet service providers are naturally interested in IP
telephony because IP telephony increases the demand for access
services. Customers have been reluctant to use IP telephony,
however, because the quality of such services has not been as high
as standard telephony and there have been significant limitations
on who could be called using IP telephony. Nevertheless, Internet
Telephony Service Providers ("ITSP") are realizing there is a
significant marketing advantage associated with having the ability
to offer IP telephony. Since some IP telephony services are of a
lower quality than traditional telephony, this provides ITSP's a
way to version their service on quality: high quality/high price
and low quality/low price, with variations in between being
developed. By offering multiple versions of their product, ITSPs
can better match their products against their customer's needs and
their willingness to pay. This enables them to extract the full
value of their services from their customers.
[0006] A Voice over IP ("VoIP") system enables the transmission of
telephone calls over an IP data network such as the Internet. A
VoIP system handles a telephone call over most of the network as
just another stream of data. Typically, an IP telephony user dials
a toll-free number to connect the user to an IP telephony gateway.
The gateway is the key element here, as it bridges the public
telephone network and the public or private IP network providing
the service. Once connected to the gateway, the user dials his or
her account number (for billing purposes) and the destination phone
number of the call. The gateway receives telephone signals on one
side, converts them to IP packets, and outputs the packets to
public or private IP networks for routing to the terminating user,
and vice versa. A typical packet includes 10 to 30 milliseconds
worth of conversation. Each packet is coded with the second party's
phone number, and compressed for rapid transmission.
[0007] The packets travel the IP network, passing through routers,
computers that operate like switches by reading the addresses on
each packet and assigning them to appropriate transmission lines,
to arrive at a gateway that decompresses them and converts the
packets back into a voice transmission signal. The gateway then
passes the call to the local phone network, which delivers it to
the intended party.
[0008] One of the key challenges emerging from the integration
between circuit switched (traditional phone) and packet switched
(IP) networks is how to address calls that pass from one network
service to another. Currently, it is possible to originate calls
from IP address-based networks to other networks, however it is
difficult to terminate calls from other networks, such as the PSTN,
to IP address-based networks. Instead, calls from the PSTN are
typically being terminated on the PSTN. For example, in PCT
Application Number PCT/US99/29171, published as International
Publication Number WO 00/41383, by Ranalli et al., a system is
described for resolving a PSTN number to an IP address for voice
communication between two simultaneous users over the Internet, but
which system lacks an IP-enabled gateway for completing calls from
an Internet user to a PSTN phone or from a PSTN phone to an
Internet user. Hence, a calling party must use a standard terminal
device, such as a phone, for connecting to another user on the
PSTN. Therefore, there is a need for an addressing system across
both PSTN and public packet networks that can allow a PSTN caller
to access a subscriber on a public packet network and vice
versa.
[0009] Although intermediate gateways exist, they present another
problem in prior VoIP systems. As noted above, in some systems, a
user is required to call a gateway, which then calls the
recipient's number. This results in over dialing and may cause some
calls to be blocked at the receiving end. Further, in other prior
systems, the mapping data for logged on subscribers is not
published in a public forum that can be accessed by other
subscribers and gateways connected to PSTN terminals. Rather, the
providers of such systems distribute a hardware device to
subscribers only, who use that device to pull mapping data for
other subscribers from a database server. Therefore there is a need
for a system that handles the entire call without a central
answering position (an over dialing gateway) and which allows any
user, even if through an intermediary, easy public access to
subscriber mapping data. The present invention allows subscribers
to call other subscribers and completely bypass the PSTN local
office by using a call agent browser to connect to a gateway device
and to send mapping data and receive mapping data from other users
via distributed servers. The call agent can be an applet or any
specifically enabled Internet browser. The call agent can also
support soft-phone software applications, provide a user profile,
and other user information.
BRIEF DESCRIPTION OF THE DRAWING
[0010] FIG. 1 is a block diagram illustrating the interface between
the present invention and the Public Switch Telephone Network;
[0011] FIG. 2 is a block diagram illustrating the operation of a
gateway in accordance with the present invention and calls between
different originators and recipients;
[0012] FIG. 3 illustrates a sample user profile in accordance with
the present invention;
[0013] FIG. 4 is a block diagram illustrating the operation of a
call between subscribers in accordance with the present
invention;
[0014] FIG. 5 is a block diagram illustrating the operation of a
call originating from a user connected to an IP address-based
network and terminating at a PSTN terminal in accordance with the
present invention; and
[0015] FIG. 6 is a block diagram illustrating the operation of a
call originating from a PSTN terminal and terminating at a user of
IP address-based network in accordance with the present
invention.
DETAILED DESCRIPTION OF THE INVENTION
[0016] The present invention allows for the origination and
termination of phone calls to and from any PSTN or IP enabled voice
communication terminal by mapping public E.164 numbers or
identifications ("IDs") for subscribers to their current Internet
IP address. Mapped ID and address information for each subscriber
is then published to an HTML document or an XML database for direct
use by other logged on IP telephony subscribers or by a gateway
that enables calls to and from non-subscribers. The gateway enables
non-subscribers to directly call subscribers without a PSTN
connection to the subscriber. Subscribers completely by-pass
charges associated with the local PSTN and can call anyone with a
unique identifier such as a phone number regardless of whether they
on the PSTN.
[0017] The Internet is a data network that carries data in a
digital form in specially formatted packets. This is contrasted
with the PSTN, which is a circuit switched network. The Internet
uses the TCP/IP protocol, which enables any TCP/IP device or
network of devices to connect to the network and function as though
they were part of a single network. In order to locate any specific
point, or host, on the Internet, a unique IP address is used for
each host. A host can have a dedicated IP address, as is frequently
the case with large bandwidth access customers, or a dynamically
generated IP address, as is the case with any dial-up user. Since
IP addresses are numerical, and therefore difficult to remember, or
dynamic, and therefore constantly changing, and because many users
are connected to different hosts all the time, many users are also
assigned more permanent types of addresses, such as a Simple Mail
Transport Protocol ("SMTP") e-mail address or a Voice Profile for
Internet Mail ("VPIM") voicemail system address. There are many
other types of address mechanisms available on the Internet.
[0018] IP Telephony subscribers may gain access to the Internet
through a number of different means. A roaming subscriber using a
wireless device (e.g. 802.11 a/b/g protocol compatible devices)
could log onto the Internet from multiple locations. A subscriber
relying on fixed access could only connect to the Internet through
a wire-line, a cable-modem connection, etc. In accordance with the
present invention, calls to an IP Telephony subscriber may
originate from the PSTN through wireline or wireless telephones, or
from other IP Telephony subscribers, provided the identity of the
called number and a corresponding IP address is known to the
calling party, which will be further discussed below in greater
detail.
[0019] So as to technically accommodate the growing number of
telephone users around the world, and increased interest in
Internet telephony, the International Telecommunications Union
("ITU") has adopted a number of standards or protocols. One such
protocol is E.164, which provides a uniform means for identifying
any telephone number in the world to any telephony user in the
world. This protocol has been widely adopted, along with other
protocols, such as H.323 and H.450, which collectively operate to
improve the quality and functionality of IP telephony. For purposes
of the present invention, however, the E.164 protocol is most
important. There are six different components to an IP telephony
network that supports the E.164 protocol within the context of the
preferred embodiment of the present invention:
[0020] (1) E.164. E.164 is the ITU-T recommendation for Global
Switched Telephone Network ("GSTN") numbering. E.164 is a sixteen
digit numbering scheme that provides a unique telephone number for
every subscriber in the world. The E.164 address provides a
globally unique, language independent identifier for resources on
public telecommunication networks. E.164 numbers are used to
identify ordinary phones, fax machines, pagers, and data
modems.
[0021] (2) IP Address. An IP address is a 32-bit number that
identifies the host that is sending or receiving information sent
in packets across the Internet. When an Internet user attempts to
locate a website by entering a Universal Resource Locator ("URL")
request, the user is also sending the IP address for the host that
the user is using at the time so HTML pages from the website can be
returned to that user. The URL request includes an Internet domain
name that is translated into a corresponding IP address to locate
the host that is storing the requested website. E-mail messages and
many other different types of messages sent or received by hosts
must include IP addressing information in order operate. An IP
address has two parts: an identifier of a particular network on the
Internet; and an identifier of the particular host (the host can be
a server or a workstation) within that network.
[0022] (3) File Transfer Protocol ("FTP"). FTP is a standard
Internet protocol that represents the simplest manner of exchanging
files between computers on the Internet. Like the Hypertext
Transfer Protocol ("HTTP"), discussed below, which enables the
transfer of displayable Web pages and related files, FTP is an
application protocol that uses the Internet's TCP/IP protocols. FTP
enables the transfer of specific files on one computer to a
specific location on another computer. FTP is commonly used to
transfer website files from their creator to the computer that acts
as their server for everyone on the Internet. And ISP might, for
example, operate an FTP-enabled server for providing IP address
information and other information to its customers
[0023] (4) The Hypertext Transfer Protocol. HTTP is the set of
rules for exchanging files on the World Wide Web. Relative to the
TCP/IP suite of protocols (which are the basis for information
exchange on the Internet), HTTP is an application protocol.
Essential concepts that are part of HTTP include (as its name
implies) the idea that files contain references to other files
whose selection will elicit additional transfer requests. Any Web
server machine contains, in addition to the HTML and other files it
can serve, an HTTP daemon, a program that is designed to wait for
HTTP and FTP requests and handle them when they arrive.
[0024] (5) Hypertext Markup Language. HTML is the set of markup
symbols, codes or words that are inserted in a file intended for
display on a World Wide Web page. The markup tells a requesting
browser how to display website pages, words and images for the user
of that browser. Each individual markup code is referred to as an
element. Some elements come in pairs that indicate when some
display effect is to begin and when it is to end.
[0025] (6) IP Telephony Terminal. This term is generally used in to
refer to a set of facilities for managing the delivery of voice
information using IP. VoIP is one means of delivering voice data
using the Internet Protocol. In general, VoIP refers to the process
of sending voice information in digital form in discrete packets
over the Internet, rather than sending voice information in analog
form (or analog converted to digital and then converted back to
analog form) in the traditional circuit-committed protocols of the
PSTN.
[0026] While the preferred embodiment of the present invention
relies on existing E.164 numbering management systems, the
operation of the present invention is not dependent on these
systems. At the present time, the ITU Telecommunication Standards
Board assigns country/service codes, and within a geographic area,
a national or regional numbering administration has the
responsibility of managing the numbering plan for the countries
assigned to that area. To have the E.164 number allocation process
managed by the existing numbering plan administrations can expedite
the allocation process, leading to an earlier deployment of
Internet telephony services.
[0027] To operate IP telephony services in accordance with the
present invention, IP telephony service providers will need to
obtain blocks of E.164 numbers from numbering plan administrations
for their subscribers. An IP telephony subscriber with a specific
E.164 number will ordinarily subscribe to the service of one IP
telephony service provider, such as their ISP. The service provider
or a trusted third party provider will maintain the data related to
that subscriber, including the IP address at which the subscriber
can currently be reached, as will be described in further detail
below in reference to FIG. 1.
[0028] FIG. 1 illustrates the interface between the IP telephony
system of the present invention and the PSTN. Telephones 102 could
be any type of conventional rotary or push-button phone. The same
basic principles apply to wireless phone connecting to the PSTN
through a cellular receiver/transmitter station (not shown), but
only the more traditional phones 102 will be discussed with regard
to FIG. 1. Telephone calls originating or terminating from wireline
or wireless telephones 102 must pass through the PSTN local office
104. If a telephone call originated from telephone 102, the local
office 104 would establish a contact with the telephone 102, look
for the dialed number in a database, and then switch the call
according to the instructions associated with the dialed number. In
this case, the dialed number is handed over to the gateway 108,
which sends an HTTP request over the Internet 112 to Internet
Telephony Home Server ("ITHS") 114 to retrieve the IP address
mapping data for the IP telephony subscriber 116 assigned to the
E.164 ID dialed by the telephone 102. There may be only one ITHS
114, or multiple ITHS distributed around the IP telephony network.
The ITHS is described in more detail with reference to FIG. 2
below. The gateway 108 could be specially designed software
programmed hardware device, or an add-on card within an existing
telecommunication system. On a PBX-class level, for example, a
Dialogic Communications Corporation (Franklin, Tenn.) MultiTech
Systems MultiVOIP 800 card mounted within a True Data Technology
Inc. (Carlsbad, Califonia) Televantage station or trunk could be
utilized. On a carrier-class level, a programmable Brooktrout, Inc.
(Needham, Mass.) TR2020 VoIP gateway could be utilized. The IP
address of the subscriber 116 may be an IP address for a computer
station, an IP-telephone, an IP mobile telephone, an IP-PBX, a SMPT
address of a unified messaging system, a SMPT address of an e-mail
system, or any similar type of address.
[0029] If the subscriber 116 is logged on to the Internet 112 and
has indicated that she is available to receive telephone calls, a
session will be established over the Internet 116 between the
gateway 108 connected to the PSTN user and the subscriber 116,
during which packets of information will be transmitted back and
forth between the subscriber 116 and the gateway 108. The gateway
108 will be described in greater detail with reference to FIG. 2
below. If the subscriber 116 is not logged on at the time of the
call, the call might be routed to an alternative call destination,
such as an email server or a voicemail box or to some other
IP-enabled device pursuant to the subscriber's directions.
[0030] FIG. 2 illustrates the operation of a gateway in accordance
with the present invention, with respect to three different types
of originating calls, a call from a wired telephone 202, a call
from a cellular telephone 206, and a call from another subscriber
208, in each case where the subscriber call recipient is connected
to a Digital Data Service ("DDS"). As noted above with respect to
FIG. 1, a call from a wired telephone 202 or cellular telephone 206
must be connected to the gateway 108 by some type of local or
cellular service, but in order to simplify FIG. 2, the telephones
202 and 206 are shown connected directly to the gateway 108. As
shown in FIG. 2, devices located on the right side of the gateway
108 are connected to the PSTN, and devices located on the left side
of the gateway 108, but to the left of the DDS, are connected to
the Internet or a similar type of IP network.
[0031] Also, as described above with reference to FIG. 1, a
subscriber must be logged on to the Internet to receive calls and
have registered with the IP telephony network, i.e., indicated a
readiness to receive calls, in order for a session to be created.
The registration process may be initiated by the subscriber's IP
telephony terminal (either a computer running a soft phone/IP
telephony application, or a computer connected to a regular
telephone and running an IP telephony application) or it may be
requested by the IP telephony network, upon incoming call arrival,
by employing a paging mechanism. Either way, the IP telephony
application will typically open a Transfer Control Protocol ("TCP")
socket to a File Transfer Protocol ("FTP") server (not shown) which
resides on the subscriber's ISP network, in order to obtain the
correct local IP address for that subscriber's terminal. If the IP
address can be obtained and authenticated with the FTP server, the
subscriber will then upload the file to a web server of the ISP
provider. A service directory such as the ITHS 114 will then
download the uploaded subscriber information from the FTP server
using the HTTP protocol. The ITHS will then authenticate the IP
address and publish the information. The information contained
within the file includes an E.164 ID of the subscriber's terminal.
If the connection to the ITHS 114 is acknowledged, a host file for
all, or at least a subset of subscribers, within the ITHS 114 will
be updated with the IP address and E.164 ID of the subscriber's
terminal and the subscriber's status will be updated to In-Service.
If for any reason the call cannot be connected to the subscriber,
the subscriber's status will be updated to Not-In-Service.
[0032] The ITHS 114 actively operates to maintain current data on
all subscribers. For example, ITHS 114 will routinely use the HTTP
protocol to retrieve an HTML document including mapping data for
other subscribers. The ITHS 114 will determine the IP address and
E.164 ID for each subscriber, as well as the time they connected to
the ITHS 114. The ITHS 114 will then attempt to connect to each
subscriber. If the connection is acknowledged, the host file will
be updated with the IP address and E.164 ID of the subscriber and
the subscriber's status will be updated to In-Service. If the
connection fails, the subscriber's status will be updated to
Not-In-Service.
[0033] Although only two status states are described above, a
subscriber's IP telephony terminal essentially has one of three
different states. Each state has a certain level of readiness
depending on the IP network to which the IP telephony terminal is
connected. When in an "idle" state, no calls can be delivered to
the IP telephony terminal of a subscriber. This can occur because
the terminal is powered off or it is powered on but not connected
to the IP network. When in this state, no dynamic IP address can be
assigned to the terminal and no IP telephony calls can be delivered
to it. When in a "standby" state, the IP telephony terminal is
powered on and connected to the IP network and has an IP address
assigned to it, but the IP telephony application is not running. In
this state, no calls can be delivered to the IP telephony terminal.
In the "ready" state, the IP telephony terminal is connected to the
IP network, has an IP address assigned to it, and the IP telephony
application has been invoked. Under such circumstances, the
terminal is ready to receive an incoming IP telephony call.
[0034] Hence, once the terminal has established an IP network
connection with the ITHS 114, thereby becoming part of the IP
telephony network, in either the standby or ready state, the
subscriber's IP telephony application, which for purposes of the
present invention is referred to as a "client connector,"
essentially causes the IP address assigned to the terminal and the
corresponding E.164 ID for that terminal to be published, as an
HTML document or XML database, within the ITHS 114. Although there
could be only a single ITHS 114, for scalability purposes, it would
be preferable to have multiple ITHS distributed throughout the IP
telephony network.
[0035] Since the mapping data is published to documents that can be
read by any Internet browser, the information can be easily
accessed via any enabled browser. The client connector also checks
the status of other subscribers by checking to see if other
subscribers have published their IP address and E.164 ID, and then
updates the local file used by the client connector with their
current IP addresses and E.164 IDs. Once the appropriate
information has been published the IP telephony terminal is able to
place and receive real-time voice calls between other published
agents using easily recognizable telephone numbers rather then
cryptic IP address.
[0036] For multiple subscribers sharing a common IP address, such
as in a company or office building, the ITHS 114 might be connected
to a DDS 218 that publishes mapping data for multiple subscribers
and stores either static or dynamic data for each individual IP
telephony subscriber. The gateway 108, as well as the various
subscribers, also requests mapping data from the ITHS 114 for
address resolution. Therefore, the main function of the ITHS 114 is
to find and provide the IP address of the terminating IP telephony
subscriber, when subscribers are in the ready or standby state, to
any device requesting such information. When a call is to be
delivered to a subscriber, such as subscriber 220 or 222, the ITHS
114 is consulted to determine the IP address for the called IP
telephony subscriber. When a user is in the standby state, calls
can be routed to an email server through the SMTP server 224, a
voicemail box through the VPIM server 228 connected to the DDS 218,
or any other similar addressable, IP-enabled device. The location
of the SMTP server 224 and VPIM server 228 are for illustration
purposes only. Each server could be accessible directly over the
Internet, connected to the DDS, or accessible in any of a number of
other manners.
[0037] Each ITHS 114 may include identical information, or
different information depending on their physical location within
the network and/or the subscribers served. The ITHS 114 could be
configured as a stand-alone server connected to one or more
gateways, as a partitioned portion of a server performing other
functions, or even as an individual subscriber's terminal. If the
function of the ITHS was configured within subscriber's terminals,
the IP telephony network would largely operate in the same manner
as the distributed information networks used to share music files.
Likewise, there may be multiple gateways distributed throughout the
network in a number of fashions, such as at each PSTN local office,
one gateway may serve multiple PSTN local offices, or gateways may
be located within PBX devices and other IP-enabled network
devices.
[0038] As noted above, each subscriber terminal must be operating
IP telephony software, such as the client connecter. The client
connector could be very basic software that largely functions to
enable a subscriber to publish mapped data, or the client connector
could be used to perform a number of additional functions. For
example, the client connector could be used to provide basic soft
phone functionality (a function generally known in the art), or to
create a user profile for a subscriber that provides a simple and
effective mechanism for managing that subscriber's IP telephony
activities. A user profile would also be published to the ITHS
114.
[0039] FIG. 3 illustrates an example of an IP telephony
subscriber's profile 302, which is being housed on the ITHS 114.
Although a variety of information may be included, with respect to
the preferred embodiment of the present invention, an exemplary
record for one subscriber would include: his or her full name 304;
a user name 308; their E.164 number 310; their status 312; a
message indicator 314 for messages they received when in a standby
state; a list of available services 316 for that subscriber (which
might include a number of Class 5 services, such as call waiting,
three-way calling, etc.); an address book 316 that includes names
and corresponding E.164 numbers for other subscribers; billing
information 320 for use by the IP telephony service provider, such
as a home address, and e-mail address or web address; and any type
of other information 324 that might be desired.
[0040] The process of connecting a call originated by one
subscriber to another subscriber in accordance with the preferred
embodiment of the present invention is illustrated with reference
to FIG. 4. When an IP Telephony subscriber 402 enters a telephone
number or other unique identifier into their client connector
software, a signal is sent to the gateway 108 over the Internet
112. The gateway looks up the IP address for the entered number in
the ITHS 114, establishes a connection to the other IP telephony
subscriber 410, and initiates a session between the two
subscribers. When either party terminates the session, the gateway
108 removes the mapping data from its memory, even though the data
may continue to exist within the ITHS. It is important that the
gateway remain connected to both subscribers. In this manner, the
gateway can monitor the length of the call for billing purposes and
be available to perform other functions if needed. For example, if
call waiting or another Class 5 service is enabled, the gateway
will be able to manage the operations necessary to perform that
function.
[0041] Referring now to FIG. 5, the process by which a subscriber
502 can call a non-subscriber 504 is illustrated. When the
subscriber 502 enters a telephone number or other unique identifier
within their client connector software, a signal is sent to a
gateway 108 that causes the gateway 108 to locate the ITHS 114,
which then consults a translation table to map the entered
telephone number to a static IP address. The gateway 108 then
establishes a direct circuit connection to the PSTN local office
104. A session is thereafter established between the subscriber 502
and the gateway 108, which gateway is in turn connected to the
recipient 504 through a local office switch. The gateway 108
receives packets from the subscriber's terminal and converts the
packet data to digital telephony signals and transports those
signals over a digital circuit to the PSTN local office 104. The
local office 104 then converts the digital telephony signals to
analog voice signals and sends those signals to the recipient 504.
Alternatively, the digital to analog conversion could be performed
within the call recipient phone. When the recipient speaks, the
recipient's voice is transmitted in either analog or digital form
to the local office 104, converted to digital signals if
transmitted in analog form, and transmitted to the gateway 108,
where the digital data is converted to IP packets and sent to the
subscriber 502. Once either party terminates the call, the session
is terminated and the gateway removes the mapping information from
its memory.
[0042] A telephone call between a PSTN telephone call originator
602 and an IP telephony subscriber call recipient 612 is
illustrated in FIG. 6. In this illustration, the call originator
602 dials a phone number on their standard phone or cellular phone,
which sends a series of analog or digital signals to the PSTN local
office 104. Based on the dialed number, the local office 104
establishes a direct connection to the gateway 108. If a cellular
phone was used, then the cellular receiver/transmitter station
would process the call and either connect directly to the gateway
108, or establish a connection to the local office 104, which would
then establish a connection with the gateway 108. The gateway 108
would then send an E.164-to-IP-address mapping request to the ITHS
114. The ITHS 114 would identify that the E.164 number corresponds
to an IP telephony subscriber and request the IP address
information for that subscriber. A session would then be
established between the subscriber 612 and the gateway 108, which
is in turn connected to the call originator 602.
[0043] Although the present invention is described above in the
context of a preferred embodiment, it is not limited as such and
could be implemented in any of a number of alternative
embodiments.
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