Internet long distance telephone service

Long distance communications service between two communications systems is established by using a wide area packet switched network, for example the Internet, to transport signaling data and digitized communication traffic. Each communications system uses an interface server to encapsulate communication traffic and signaling data into data packets suitable for transport over the wide area packet switched network. The interface server accesses a routing and administration database to determine a destination address of a destination interface server based on the area code of the called number. Upon receiving the destination address and a prescribed bandwidth from the routing and administration database, the telephony server inserts the destination address to the data packets for a destination server for a second communications system. The packets are then output to the Internet, and subsequently routed to the destination server serving as an interface for the second communications system. Routing of packets is preferably performed using reserved virtual paths to guarantee quality of service.

TECHNICAL FIELD 
The present invention relates to arrangements for public telecommunications 
systems to provide long distance telephone service over the Internet. 
DESCRIPTION OF THE RELATED ART 
Attention recently has been directed to implementing voice telephone 
service over the worldwide network now commonly known as the Internet. The 
Internet had its genesis in U.S. Government funded research which made 
possible national internetworked communication systems (called ARPA). This 
work resulted in the development of network standards as well as a set of 
conventions for interconnecting networks and routing information. These 
protocols are commonly referred to as TCP/IP. The protocols generally 
referred to as TCP/IP were originally developed for use only through 
ARPANET and have subsequently become widely used in the industry. TCP/IP 
is flexible and robust, in effect, TCP takes care of the integrity and IP 
moves the data. Internet provides two broad types of services: 
connectionless packet delivery service and reliable stream transport 
service. The Internet basically comprises several large computer networks 
joined together over high-speed data links ranging from ISDN to T1, T3, 
FDDI, SONET, SMDS, OT1, etc. The most prominent of these national nets are 
MILNET (Military Network), NSFNET (National Science Foundation NETwork), 
and CREN (Corporation for Research and Educational Networking). In 1995, 
the Government Accounting Office (GAO) reported that the Internet linked 
59,000 networks, 2.2 million computers and 15 million users in 92 
countries. It is presently estimated that the growth of the Internet is at 
a more or less annual doubling rate. 
Referring to FIG. 1 there is shown a simplified diagram of the Internet. 
Generally speaking the Internet consists of Autonomous Systems (ASs) which 
may be owned and operated by Internet Service Providers (ISPs) such as 
PSI, UUNET, MCI, SPRINT, etc. Three such AS/ISPs are shown in FIG. 1 at 
10, 12 and 14. The Autonomous Systems (ASs) are linked by Inter-AS 
Connections 11, 13 and 15. Information Providers (IPs) 16 and 18, such as 
America Online (AOL) and Compuserve, are connected to the Internet via 
high speed lines 20 and 22, such as T1/T3 and the like. Information 
Providers generally do not have their own Internet based Autonomous 
Systems but have or use Dial-Up Networks such as SprintNet (X.25), DATA 
and TYMNET. 
By way of current illustration MCI is both an ISP and an IP, Sprint is an 
ISP, and MicroSoft (MSN) is an IP using UUNET as an ISP. Other information 
providers, such as universities, are indicated in exemplary fashion at 24 
and are connected to the AS/ISPs via the same type connections here 
illustrated as T1 lines 26. Corporate Local Area Networks (LANs), such as 
those illustrated in 28 and 30, are connected through routers 32 and 34 
and links shown as T1 lines 36 and 38. Laptop computers 40 and 42 are 
representative of computers connected to the Internet via the public 
switched telephone network (PSTN) are shown connected to the AS/ISPs via 
dial up links 44 and 46. 
The Information Providers (IPs) constitute the end systems which collect 
and market the information through their own servers. Access providers are 
companies such as UUNET, PSI, MCI and SPRINT which transport the 
information. Such companies market the usage of their networks. 
In simplified fashion the Internet may be viewed as a series of routers 
connected together with computers connected to the routers. In the 
addressing scheme of the Internet an address comprises four numbers 
separated by dots. An example would be 164.109.211.237. Each machine on 
the Internet has a unique number which constitutes one of these four 
numbers. In the address the leftmost number is the highest number. By 
analogy this would correspond to the ZIP code in a mailing address. At 
times the first two numbers constitute this portion of the address 
indicating a network or a locale. That network is connected to the last 
router in the transport path. In differentiating between two computers in 
the same destination network only the last number field changes. In such 
an example the next number field 211 identifies the destination router. 
When the packet bearing the destination address leaves the source router 
it examines the first two numbers in a matrix table to determine how many 
hops are the minimum to get to the destination. It then sends the packet 
to the next router as determined from that table and the procedure is 
repeated. Each router has a database table that finds the information 
automatically. This continues until the packet arrives at the destination 
computer. The separate packets that constitute a message may not travel 
the same path depending on traffic load. However they all reach the same 
destination and are assembled in their original order in a connectionless 
fashion. This is in contrast to connection oriented modes such as frame 
relay and ATM or voice. 
One or more companies have recently developed software for use on personal 
computers to permit two-way transfer of real-time voice information via an 
Internet data link between two personal computers. In one of the 
directions, the sending computer converts voice signals from analog to 
digital format. The software facilitates data compression down to a rate 
compatible with modem communication via a POTS telephone line. The 
software also facilitates encapsulation of the digitized and compressed 
voice data into the TCP/IP protocol, with appropriate addressing to permit 
communication via the Internet. At the receiving end, the computer and 
software reverse the process to recover the analog voice information for 
presentation to the other party. Such programs permit telephone-like 
communication between Internet users registered with Internet Phone 
Servers. 
The book "Mastering the Internet", Glee Cady and Pat McGregor, SYBEX Inc., 
Alameda, Calif., 1994, ISBN 94-69309, very briefly describes three 
proprietary programs said to provide real-time video and voice 
communications via the Internet. 
Palmer et al. U.S. Pat. No. 5,375,068, issued Dec. 20, 1994 for Video 
Teleconferencing for Networked Workstations discloses a video 
teleconferencing system for networked workstations. A master process 
executing on a local processor formats and transmits digital packetized 
voice and video data, over a digital network using TCP/IP protocol, to 
remote terminals. 
Lewen et al. U.S. Pat. No. 5,341,374, issued Aug. 23, 1994 for 
Communication Network Integrating Voice Data and Video with Distributed 
Call Processing, discloses a local area network with distributed call 
processing for voice, data and video. Real-time voice packets are 
transmitted over the network, for example to and from a PBX or central 
office. 
Hemmady et al. U.S. Pat. No. 4,958,341, issued Sep. 18, 1990 for Integrated 
Packetized Voice and Data Switching System, discloses an integrated 
packetized voice and data switching system for a metropolitan area network 
(MAN). Voice signals are converted into packets and transmitted on the 
network. Tung et al. U.S. Pat. No. 5,434,913, issued Jul. 18, 1995, and 
U.S. Pat. No. 5,490,247, issued Feb. 6, 1996, for Video Subsystem for 
Computer Based Conferencing System, disclose an audio subsystem for 
computer-based conferencing. The system involves local audio compression 
and transmission of information over an ISDN network. 
Hemmady et al. U.S. Pat. No. 4,872,160, issued Oct. 3, 1989, for Integrated 
Packetized Voice and Data Switching System, discloses an integrated 
packetized voice and data switching system for metropolitan area networks. 
Sampat et al. U.S. Pat. No. 5,493,568, issued Feb. 20, 1996, for Media 
Dependent Module Interface for Computer Based Conferencing System, 
discloses a media dependent module interface for computer based 
conferencing system. An interface connects the upper-level data link 
manager with the communications driver. 
Koltzbach et al. U.S. Pat. No. 5,410,754, issued Apr. 25, 1995, for 
Bi-Directional Wire Line to Local Area Network Interface and Method, 
discloses a bi-directional wire-line to local area network interface. The 
system incorporates means for packet switching and for using the Internet 
protocol (IP). 
The known prior art does not disclose an efficient arrangement for 
establishing reliable long distance service via the Internet on a large 
scale. Known telephone-like communications via the Internet require each 
end station to have a TCP/IP address, resulting in an inefficient use of 
addressing resources. Moreover, the packet switched architecture of the 
Internet does not provide guaranteed bandwidth or bounded access 
latencies, resulting in poor quality voice communication over the 
Internet. 
DISCLOSURE OF THE INVENTION 
There is a need to provide long distance telephone service via the Internet 
to users of the public telecommunications network without access to a 
computer and without separate telephone user access to the Internet. 
There is also a need to provide the general public with an economical and 
convenient long distance telephone service via the Internet without 
requiring the possession of computing equipment or familiarity with the 
Internet or its methodology on the part of the user. 
There is also a need to to provide the public with impulse access to the 
Internet for voice communications without requiring maintenance of a 
subscription to an Internet access service. 
There is an additional need to provide the foregoing types of telephone 
service over the Internet via the public telephone network without the 
necessity of reliance on signaling systems of interexchange carriers. 
There is yet another need to provide voice service over public telephone 
systems via the Internet where the use of the Internet is optional to the 
Telco and transparent to the customer. 
There is yet another need to provide voice service over public telephone 
systems via the Internet from telephone to telephone, from telephone to 
computer, from computer to telephone, and from computer to computer. 
There is also a need to provide the foregoing type services with billing 
capabilities based substantially on equipment and methodologies presently 
available in the public switched telephone network. 
These and other needs, as well as the drawbacks identified with respect to 
the known prior art, are resolved by the present invention, where a 
routing and administration database provides the destination address, 
based on at least a portion of a destination number, for a telephone 
server servicing a long distance telephone number and receiving voice 
traffic via the Internet. 
According to one aspect of the present invention, a method of 
telecommunication over a wide area packet switched network comprises 
establishing a communication link between telephony servers serving 
respective telephone systems. The telephony servers each include a 
telephony platform and a wide area packet switched platform enabling the 
telephony servers to transfer signaling data and traffic data from the 
telephone system domain to the wide area packet switched network and vice 
versa. 
The method includes sending, from a calling party, a called number 
corresponding to a called party and including an area code. The called 
number is sent to a first central office connected to a first telephone 
system. The called number is forwarded from the first central office to a 
first telephony server connected to the first telephone system and in 
communication with the wide area packet switched network. A routing and 
administration database identifies a second telephony server in 
communication with the wide area packet switched network and serving the 
called party in a second telephone system. The routing and administration 
database identifies the second telephony server using at least the area 
code. The first telephony server sends the called number to the second 
telephony server via the wide area packet switched network, and a 
communication link is selectively established between the first telephony 
server and the second telephony server according to a prescribed service 
level to establish communication between the calling and called parties. 
The communication link-between the servers minimizes the number of hosts 
on the wide area packet switched network. Hence, a plurality of 
communications links can be established between the two servers for calls 
to and from the respective area codes using the same destination addresses 
on the wide area packet switched network. The servers then use higher 
level protocol to divide and distribute the voice calls throughout the 
respective telephone system. 
Another aspect of the present invention provides a method of 
telecommunication over a wide area packet switched network, where a 
communication link is established via the wide area packet switched 
network according to a prescribed service level. The method includes the 
steps of receiving, in a first telephony server connected to a first 
telephone system, a first data packet via a wide area packet switched 
network. The received first packet is transmitted by a second telephony 
server of a second telephone system, and includes (1) a destination 
address corresponding to the first telephony server, (2) a session 
identifier, and (3) a destination number having an area code served by the 
first telephony server. A condition of the destination number from a first 
central office serving the destination number is determined via a 
signaling communication network of the first telephone system. A second 
data packet carrying the session identifier and condition is then sent 
from the first telephony server to the second telephony server, and a 
communication link is selectively established between the first telephony 
server and the second telephony server according to a prescribed service 
level to establish communication between the destination number and a 
station served by the second telephony server. Signaling communications 
via the wide area packet switched network enable two telephone systems to 
establish telephone calls without the necessity of establishing separate 
landline signaling networks or leasing signaling communication links from 
interexchange carriers. 
Still another aspect of the present invention provides a method of 
telecommunication over the Internet, where virtual paths are established 
between servers to maintain a guaranteed quality long distance calls 
between two telephone systems. The method includes establishing a 
dedicated virtual path having a prescribed bandwidth between at least 
first and second telephony servers. The first and second telephony servers 
have respective network addresses specifying points of presence on the 
Internet, and are connected to first and second telephone systems, 
respectively. A routing and administration database stores the prescribed 
bandwidth and, for each of the telephony servers, the network address and 
area codes served within the corresponding telephone system. A call 
request initiated by a calling party within the first telephone network, 
is received at the first telephony server. The call request includes a 
calling party number corresponding to the calling party and a called party 
number, the called party number including an area code. A routing request 
is then sent by the first telephony server to routing and administration 
database that includes the calling party number and the area code of the 
called party number. The routing and administration database outputs a 
bandwidth allocation and the network address of the second telephony 
server in response to the area code supplied by the routing request, 
wherein the routing and administration database provides the bandwidth 
allocation from the prescribed bandwidth. Signaling data packets are then 
sent by the first telephony server to the second telephony server along 
the dedicated virtual path, wherein the signaling data packets include the 
called party number and the bandwidth allocation. A communication link is 
then established between the first telephony server and the second 
telephony server according to the bandwidth allocation to establish 
communication between the calling party and a destination corresponding to 
the called party number. 
Additional objects, advantages and novel features of the invention will be 
set forth in part in the description which follows, and in part will 
become apparent to those skilled in the art upon examination of the 
following or may be learned by practice of the invention. The objects and 
advantages of the invention may be realized and attained by means of the 
instrumentalities and combinations particularly pointed out in the 
appended claims.

BEST MODE FOR CARRYING OUT THE INVENTION 
Overview 
The present invention implements long distance communications service 
between two communications systems by using a wide area packet switched 
network, for example the Internet, to transport signaling data and 
digitized communication traffic. Each communications system uses an 
interface server, referred to as a telephony server, to encapsulate 
communication traffic and signaling data into data packets suitable for 
transport over the wide area packet switched network. 
In the context of using the Internet as the wide area packet switched 
network, the telephony server of one communications system packetizes the 
communication traffic or signaling data into TCP/IP format. The telephony 
server accesses a routing and administration database to determine a 
destination address to the data packets. The routing and administration 
database maintains an inventory of servers on the basis of area codes and 
destination address corresponding to the point of presence (POP) on the 
Internet, and monitors the use of bandwidth between the servers. Upon 
receiving the destination address and a prescribed bandwidth from the 
routing and administration database, the telephony server inserts the 
destination address to the data packets for a destination server for a 
second communications system. The packets are then output to the Internet, 
and subsequently routed to the destination server serving as an interface 
for the second communications system. Routing of packets is preferably 
performed using reserved virtual paths to guarantee quality of service. 
Upon receiving a data packet from the Internet, the destination server 
assembles the data packet to recover the communication traffic or 
signaling data. Using higher-level voice or signaling protocols known in 
the telephony art, the destination server directs the communication 
traffic or signaling data to the appropriate destination using tandem 
trunk lines or signaling communications paths. 
Signaling 
To facilitate understanding of the present invention, it will be helpful 
first to review the architecture and operation of a telephone network 
having Common Channel Interoffice (CCIS) capabilities. The following 
description of signaling in telephone systems will provide a better 
understanding of the use of signaling data by internet telephony servers 
in establishing communication links across the Internet. 
Referring to FIG. 2 there is shown a simplified block diagram of a switched 
traffic network and the common channel signaling network used to control 
the signaling for the switched traffic network. In the illustrated 
example, the overall network actually comprises two separate networks 1 
and 2. As shown, these networks serve different regions of the country and 
are operated by different local exchange carriers. Alternatively, one 
network may be a local exchange carrier network, and the other network may 
comprise an interexchange carrier network. Although the signaling message 
routing of the present invention will apply to other types of networks, in 
the illustrated example, both networks are telephone networks. 
In FIG. 2, a first local exchange carrier network 1 includes a number of 
end office switching systems providing connections local communication 
lines coupled to end users telephone station sets. For convenience, only 
one end office 41 is shown. The first local exchange carrier network 1 
also includes one or more tandem switching systems 43 providing 
connections between offices. As such, the first telephone network 
comprises a series of switching offices interconnected by voice grade 
trunks, shown as in FIG. 2 solid lines. One or more trunks also connect 
the tandem 43 to one or more switches, typically another tandem office 53, 
in the second network 2. 
Each switching office has SS7 signaling capability and is conventionally 
referred to as a signaling point (SP) in reference to the SS7 network. In 
the first network 1, each switching office 41, 43 also is programmed to 
recognize identified events or points in call (PICs). In response to a 
PIC, either office 41 or 43 triggers a query through the signaling network 
to an Integrated Service Control Point (ISCP) 47 for instructions relating 
to AIN type services. Switching offices having AIN trigger and query 
capability are referred to as Service Switching Points (SSPs). The ISCP 47 
is an integrated system, recognized in the art. 
The end office and tandem switching systems typically include programmable 
digital switches with CCIS communications capabilities. One example of 
such a switch is a 5ESS type switch manufactured by AT&T; but other 
vendors, such as Northern Telecom and Siemens, manufacture comparable 
digital switches which could serve as the SPs. 
Within the first network 1, the common channel interoffice signaling (CCIS) 
network includes one or more Signaling Transfer Points (STPs) 45 and data 
links shown as dotted lines between the STP(s) 45 and the switching 
offices. A data link also connects the STP 45 to the ISCP 17. One or more 
data links also connect the STP(s) 45 in the network 1 to those in 
networks of other carriers, for example to the STP 55 in the network 2. 
Although shown as telephones in FIG. 2, the terminal devices can comprise 
any communication device compatible with the local communication line. 
Where the line is a standard voice grade telephone line, for example, the 
terminals could include facsimile devices, modems etc. 
The network 2 is generally similar in structure to the network 1. The 
network 2 includes a number of end office SP type switching systems 51 
(only one shown) as well as one or more tandem switching systems 53 (only 
one shown). The network 2 includes a CCIS network comprising one or more 
STPs 55 and data links to the respective SP type switching offices and to 
the CCIS system of other carriers, networks. 
In the illustrated example, the second network 2 is not a full AIN type 
network. The switching systems do not have full AIN trigger and query 
capabilities. The network 2 includes a Service Control Point (SCP) 57, but 
the routing tables utilized in that database are more limited than those 
in the ISCP 47. The switching systems 51, 53 can query the SCP 57 for 
routing information, but the range of trigger events are more limited, 
e.g., to 800 number call processing. 
An end office switching system 41 or 51 shown in FIG. 2 normally responds 
to a service request on a local communication line connected thereto, for 
example an off-hook followed by dialed digit information, to selectively 
connect the requesting line to another selected local communication line. 
The connection can be made locally through only the connected end office 
switching system but typically will go through a number of switching 
systems. For example, when a subscriber at station X calls station Y, the 
connection is made through the end office switching system 41, the tandem 
offices 43 and 53 and the end office switching system 51 through the 
telephone trunks interconnecting the various switching offices. 
In the normal call processing, the central office switching system responds 
to an off-hook and receives dialed digits from the calling station. The 
central office switching system analyzes the received digits to determine 
if the call is local or not. If the called station is local and the call 
can be completed through the one central office, the central office 
switching system connects the calling station to the called station. If, 
however, the called station is not local, the call must be completed 
through one or more distant central offices, and further processing is 
necessary. If at this point the call were connected serially through the 
trunks and appropriate central offices between the caller and the called 
party using in-band signaling, the trunks would be engaged before a 
determination is made that the called line is available or busy. 
Particularly if the called line is busy, this would unnecessarily tie up 
limited voice trunk circuit capacity. The CCIS system through the STP's 
was developed to alleviate this problem. 
In the CCIS type call processing method, the originating end office 
switching system, switching system 11 in the present example, suspends the 
call and sends a message through the CCIS network to the end office 
switching system serving the destination telephone line, i.e., to a 
terminating end office 21. The terminating end office determines whether 
or not the called station Y is busy. If the called station is busy, the 
terminating end office 21 so informs the originating end office 11 via 
CCIS message, and the originating end office provides a busy signal to the 
calling station. If the called station Y is not busy, the terminating end 
office 21 so informs the originating end central office 11. A telephone 
connection is then constructed via the trunks and end offices (and/or 
tandem offices) of the network between the calling and called stations. 
For an AIN type service, such as call redirection based on data stored in 
the ISCP 47, the end offices and/or tandems are SSP capable and detect one 
of a number of call processing events, each identified as a `point in 
call` (PIC), to trigger AIN type processing. Specifically, in response to 
such a PIC, a tandem 43 or end office switching system 41 suspends call 
processing, compiles a call data message and forwards that message via 
common channel interoffice signaling (CCIS) links and one or more STPs 45 
to an ISCP 47. If needed, the ISCP 47 can instruct the particular 
switching office to obtain and forward additional information. Once 
sufficient information has reached the ISCP 47, the ISCP 47 accesses its 
stored data tables to translate the received data into a call control 
message and returns the call control message to the switching office via 
the STP 45 and the appropriate CCIS links. The office uses the call 
control message to complete the particular call through the public 
switched network in the manner specified by the subscriber's data file in 
the ISCP 47. 
The SCP 57 offers a similar capability in the network 2, but the range of 
service features offered by that database are more limited. Typically, the 
SCP 57 offers only 800 number calling services with a limited number of 
related call routing options. The triggering capability of the tandem 53 
and end office 51 is limited to 800 number recognition. If the end office 
51 is capable of 800 number recognition and CCIS communication with the 
SCP 57, as shown, then the office 51 launches a CCIS query to the SCP 57 
in response to dialing of an 800 number at a station set Y. The SCP 57 
translates the dialed 800 number into an actual destination number, for 
example the telephone number of station X, and transmits a CCIS response 
message back to end office 51. End office 51 then routes the call through 
the public network to the station X identified by the number sent back by 
the SCP 57, using CCIS call routing procedures of the type discussed 
above. 
SS7 signaling protocol is based on the OSI model. The International 
Standards Organization (ISO) Open Systems Interconnection (OSI) reference 
model specifies a hierarchy of protocol layers and defines the function of 
each layer in the network. FIG. 3 shows the OSI model and the relationship 
thereof to the protocol stack for SS7. The lowest layer defined by the OSI 
model is the physical layer (L1). This layer provides transmission of raw 
data bits over the physical communication channel through the particular 
network. The layer next to the physical layer is the data link layer (L2). 
The data link layer transforms the physical layer, which interfaces 
directly with the channel medium, into a communication link that appears 
error-free to the next layer above, known as the network layer (L3). The 
data link layer performs such functions as structuring data into packets 
or frames, and attaching control information to the packets or frames, 
such as checksums for error detection, and packet numbers. The network 
layer provides capabilities required to control connections between end 
systems through the network, e.g., set-up and tear-down of connections. 
In the OSI model, a transport layer protocol (L4) runs above the network 
layer. The transport layer provides control of data transfer between end 
systems. Above the transport layer, a session layer (L5) is responsible 
for establishing and managing communication between presentation entities. 
For example, the session layer determines which entity communicates at a 
given time and establishes any necessary synchronization between the 
entities. 
Above the session layer, a presentation layer (L6) serves to represent 
information transferred between applications in a manner that preserves 
its meaning (semantics) while resolving differences in the actual 
representation (syntax). A protocol (L7) that is specific to the actual 
application that utilizes the information communicated runs at the top of 
the protocol stack. 
A detailed explanation of the SS7 protocol may be found in Bell 
Communications Research, "Specification of Signaling System Number 7," 
Generic Requirements, GR-246-CORE, Issue 1, December 1994, the disclosure 
of which is incorporated herein in its entirety by reference. A summary 
description of the most relevant aspects of SS7 appears below. 
For SS7, typical applications layer protocols include Transaction 
Capability Application Part (TCAP); Operations, Maintenance, Application 
Part (OMAP); and ISDN User Part (ISDN-UP). TCAP provides the signaling 
protocols for exchange of non-circuit related, transaction-based 
information, typically for accessing databases such as SCPs. For example, 
TCAP specifies the format and content of an initial query message from an 
SSP to an SCP and various response messages from the SCP back to the SSP. 
ISDN-UP is the actual call control application protocol of SS7. ISDN-UP 
specifies the procedures for setting up and tearing down trunk connections 
utilizing CCIS signaling. ISDN-UP messages, for example, include an 
Initial Address Message (IAM), an Address Complete Message (ACM) and an 
Answer Message (ANM) 
SS7 specifies an Application Service Part (ASP) for performing the 
functions of the presentation, session and transport layers for the TCAP 
and OMAP protocols. The lower four layers of the SS7 protocol correspond 
to the lower three layers (network, link and physical) of the OSI model. 
The lower three layers of the SS7 protocol, the network layer, the 
signaling link layer and the data link layer, form the Message Transfer 
Part (MTP) of SS7. The MTP is common to messages for all applications and 
provides reliable transfer of signaling messages between network nodes. 
The MTP relays messages between applications running at different nodes of 
the network, effectively like a datagram type service. 
The SS7 network layer (lower portion of L3) routes messages from source to 
destination. Routing tables for the signaling network layer facilitate 
routing based on logical addresses. The routing functionality at this 
layer is independent of the characteristics of particular links. 
The signaling link layer (L2) performs flow control, error correction and 
packet sequence control. The signaling data link layer (L1) is the actual 
physical connection between nodes of the CCIS network. The signaling data 
link layer in CCIS provides full duplex packet switched data 
communications. The signaling data link layer element provides a bearer 
for the actual signaling message transmissions. In a digital environment, 
56 or 64 Kbits/s digital paths carry the signaling messages between nodes, 
although higher speeds may be used. 
At the equivalent of the OSI network layer (L3), the SS7 protocol stack 
includes a Signaling Connection Control Part (SCCP) as well as the network 
layer portion of the MTP. SCCP provides communication between signaling 
nodes by adding circuit and routing information to SS7 messages. The SCCP 
routing information serves to route messages to and from specific 
applications. Each node of the signaling network, including the various 
switching offices and databases in each network, is assigned a 9-digit 
point-code for purposes of addressing signaling messages through the CCIS 
network. Both the SCCP protocol and the MTP processing utilize these point 
codes. 
The SS7 messages traverse the network at all times. The messages themselves 
comprise digital serial messages that come into the STP. FIG. 4 provides a 
graphic illustration of an SS7 message packet. The first byte or octet of 
the message is a flag, which is a zero followed by 6 ones and another 0. 
This constitutes a unique bit pattern in the SS7 protocol. The protocol 
ensures that this particular pattern is not repeated until the next 
message. This provides a flag at the beginning of a new message. A flag at 
the end of a message is also provided usually in the form of the flag at 
the beginning of the next message, i.e., a message usually contains only 
one flag. The message is arranged in 8 bit bytes or octets. These octets 
represent the information carried by the message. The message contains 
both fixed and variable parameters. The Message Transport Part (MTP) of 
the SS7 message is always in the same place. The values change but the MTP 
is always in the same place. 
Octets 2-11 form a routing label as discussed later with regard to FIG. 3. 
Octet 12 contains a signaling link selection (SLS) byte used to select 
specific links and/or determine the extent to which the network can select 
specific links to achieve load sharing. Octet 13 contains a Customer 
Identification Code (CIC) which typically is used to select an 
interexchange carrier. Octet 14 contains a message type indicator, and 
octets 15-N contain the actual message, in the form of fixed parameters, 
mandatory parameters and optional parameters. The length of the mandatory 
parameters field and the optional parameters field are variable. There 
would be 16 other bits that have Cyclic Redundancy Codes (CRCs) in them 
and another flag which would constitute the end of the SS7 message (and 
typically the start of the next message). CRCs constitute a further error 
detection code which is a level 1 function in the protocol. 
FIG. 5 is a graphic illustration of the routing label of the SS7 message 
packet. The first 7 bits of octet 2 constitute the Backward Sequence 
Number (BSN). The eighth bit is the Backward Indicator Bit (BIB) which is 
used to track whether messages have been received correctly. The length of 
an SS7 message is variable, therefore octet 4 contains a message length 
indicator. 
Octet 5 is the Service Information Octet (SIO) This indicates whether it is 
a Fill In Signal Unit (FISU), Link Service Signaling Unit (LSSU) or 
Message Signaling Unit (MSU). MSUs are the only ones used for setting up 
calls, LSSUs are used for alignment, and FISUs are fill in signals. The 
MSU indicator type SIO octet is formatted and encoded to serve as an 
address indicator, as discussed below. 
The routing label includes fields for both destination related addressing 
and point of origin addressing. The destination or `called party` address 
includes octets 6, 7 and 8. Octets 9-11 carry origination point code 
information, for example member, cluster and network ID information. 
In the example shown in FIG. 5, the three octets of the called party 
address contain an actual destination point code (DPC) identified as 
DPC-member, DPC-cluster and DPC-network ID information. In operation, the 
translation tables stored in the STP cause the STP to actually route based 
on the DPC without translating any of the DPC octets into new values. The 
called party address octets (6-8), however, may carry other types of 
called party addressing information and receive different treatment by the 
STP. For example, these octets may carry a global title (GTT) and 
subsystem number (SSN) information. 
To distinguish the types of information carried in octets 6-8, the MSU type 
service information octet (5) contains an address indicator. For example, 
a `1` value in the first bit position in this octet signifies that the 
called party address octets contain a subsystem number, a `1` value in the 
second bit position in this octet signifies that the called party address 
octets contain a signaling point code. The third, fourth, fifth and sixth 
bits of the address indicator serve as the global title indicator and are 
encoded to identify the presence and type of global title value in octets 
6-8. 
Additional details related to transport of signaling data over the Internet 
can be found in commonly-assigned, copending application Ser. No. 
08/710,594, filed Sep. 20, 1996, entitled TELECOMMUNICATIONS NETWORK 
(attorney docket 680-188), the disclosure of which is incorporated in its 
entirety by reference. 
Internet Long Distance Architecture 
FIG. 6 is a block diagram illustrating the architecture of a 
telecommunications system using a wide area packet switched network such 
as the Internet. The telecommunications system includes a plurality of 
switched telecommunications networks 62a, 62b, and 62c operating in 
different geographical regions. For example, each telecommunications 
network 62 may be a public switched telephone network such as a Regional 
Bell Operating Company (RBOC), or a private communication network having a 
limited service area. Each network 62 has at least one assigned number 
code, such as an area code, that uniquely identifies service areas of that 
network. Each network 62 also includes a plurality of interconnected 
switching systems 41 serving customer premises terminals 64 via local loop 
connections 66. As described above with respect to FIG. 2, each network 62 
also includes trunk lines 68 and signaling lines 70 that support the 
interoffice signaling for the particular network. 
Each telephone system 62 also includes an Internet telephony server (ITS) 
72 that provides an interface between the corresponding telephone system 
62 and the wide area packet switched network 74, for example the Internet. 
The ITS 72a is typically connected to a local central office 41 via a 
standard voice grade line or trunk connection 68, for example a T-1 or T-3 
connection. Alternatively the hardware associated with the ITS 72a may be 
situated at the central office 41 and associated with the switching 
system. 
The ITSs 72 include signaling capabilities, for example SSP capabilities, 
and are connected into the CCIS network as indicated by the links 70 to 
the illustrative STP 76. The SSPs serving the corresponding ITS 72 are 
inter-connected with the central office SSPs and CCIS network. The ITSs 
may be linked for signaling purposes by conventional F links. The Internet 
Modules are connected to the Internet 74 by T1/T3 trunks 78. 
The system 60 also includes a routing and administration server (RAS) 80 
that includes a routing and administration database for managing call 
routing translations and user access permissions. The RAS 80 is shown as 
an Internet node having a dedicated virtual path 82, described below. The 
routing and administration database stores records for every area code/NNX 
served by a telephony system 62, along with the network address for the 
corresponding ITS 72. FIG. 10A is a diagram illustrating the stored 
records of the routing and administration database of the RAS 80 stored in 
a translation table 90. The translation table 90 stores for each area code 
and central office code (NNX) the IP address of the corresponding ITS 72, 
also referred to as the ITS address. The routing and administration 
database in the RAS 80 thus stores all area codes serviced by a given 
telephone system 62a, as well as the Internet address identifying the 
point of presence (POP) for the serving ITS 72a. Hence, the RAS 80 serves 
as a pointer to identify a destination Internet telephony server 72 based 
on the area code of the called station. If a telephone system 62 includes 
a plurality of ITSs 72 within a selected area code, then the translation 
table 90 provides the unique IP address based on the area code and central 
office code being accessed. 
For example, the ITS 72c processes a telephone call for called party 64a 
initiated by the calling party 64c by sending a routing request to the RAS 
80. The routing request will include the area code of the called party 
64a. The RAS 80 accesses the internal translation table 90 to determine 
the ITS address corresponding to the area code of the called party. If the 
destination telephone network has a plurality of internet telephony 
servers within an area code, the RAS 80 may send to the ITS 72c a 
signaling message requesting the central office code (NNX) as well. Once 
the RAS 80 has sufficient information to identify the specific ITS 72a 
serving the called party 64a, the RAS 80 sends the IP address of the ITS 
72a serving the specified area code to the ITS 72c. The ITS 72c in 
response sends signaling and/or voice traffic to the ITS 72a by outputting 
data packets having the IP address of the ITS 72a as a destination 
address. Once received by the ITS 72a, the signaling and/or voice traffic 
is recovered from the payload of the data packets and processed by 
upper-layer protocol to establish the communication link between the 
calling station 64c and the called station 64a via the Internet. 
A particular aspect of the disclosed embodiment is the use of dedicated 
virtual paths established in the Internet 74 to maintain a prescribed 
service level, i.e., quality of service, for the calling party. 
Specifically, the Internet 74 includes a plurality of routers 84 (R) that 
route data packets along available paths 86 based on known algorithms. As 
known in the art, the separate packets that constitute a message may not 
travel the same path 86 depending on traffic load. However they all reach 
the same destination and are assembled in their original order in a 
connectionless fashion. 
In order to provide guaranteed service quality during long distance 
telephone calls via the Internet, the data packets can be transported on 
dedicated virtual paths at a minimum guaranteed bandwidth and latency, for 
example 28.8 kbps per telephone call in each direction. The disclosed 
embodiment establishes dedicated virtual paths 88 for large-scale 
transport of packets carrying long distance traffic to different telephone 
systems 62. Specifically, selected routers 84' reserve a predetermined 
amount of bandwidth, for example, twenty percent of the overall capacity, 
for virtual paths for use by the RAS and the ITSs 72 in transporting voice 
and signaling data. FIG. 11 is an example of an internal matrix table 92 
in one of the routers 84', where the router 84' receiving a data packet 
from a source node (i.e., another router) outputs the received data packet 
to a predetermined destination node based on the destination IP address in 
the data packet. As shown in FIG. 11, the router reserves a 51.8 MB/s 
(OC-1) path between source and destination nodes IP1 and IP2 for packets 
having a destination address corresponding to the ITS (B) 72b. Hence, 
assuming a router 84' has a capacity of switching up to 466.56 MB/s 
(OC-9), the router can reserve one virtual path at 51.8 MB/s (OC-1), 
another path at 44.7 MB/s (DS-3), and a third virtual path at 155.5 MB/s 
(OC-3) between two nodes. 
Hence, a complete virtual path having a predetermined bandwidth between two 
ITSs 72 can be established by forming a sequence of routers, each having 
predetermined path segments for transporting data packets along the 
virtual path to the next router or node. The virtual path is initially 
arranged by contracting with the Internet service provider controlling 
each router 84' in the desired virtual path. The ISP will then program the 
router 84' and any associated Autonomous System (AS) with the table 92 to 
guarantee the desired bandwidth along the virtual path. 
Once the sequence of routers has been established, the end-to-end virtual 
path (POP(1) to POP(2)) is stored as a virtual path lookup table 94 in the 
RAS 80 database, along with the total available bandwidth, shown in FIG. 
10B. The RAS 80 also monitors unused bandwidth by allocating bandwidth for 
each routing request. Hence, the RAS 80 is able to monitor traffic along a 
virtual path to determine whether a data rate in a communication link 
should be changed. If the RAS 80 determines that a virtual path has little 
traffic, then the RAS may specify a higher data rate for the communication 
link. However, if the RAS 80 determines that a large amount of traffic 
exists on the virtual path, then the data rate may be reduced to the 
minimum guaranteed service level stored in the RAS 80 database for the 
calling number, shown in FIG. 10C. 
An alternate arrangement for providing a communication link according to a 
prescribed service level involves using Internet Protocol, version 6 
(IPv6). IPv6 incorporates a variety of functions that make it possible to 
use the Internet for delivery of audio, video, and other real-time data 
that have guaranteed bandwidth and latency requirements. Hosts can reserve 
bandwidth along the route from source to destination. Hosts can specify 
loose or strict routing for each hop along a path. In addition, packets 
are assigned a priority level, ensuring that voice or video transmission 
is not interrupted by lower priority packets. 
As shown in FIG. 6, a group of virtual paths 88 enable transmission of 
signaling and traffic data between the ITSs 72a, 72b and 72c via the 
Internet at prescribed service levels. Signaling information between the 
ITSs 72 and between an ITS 72 and the RAS 80 will typically be given 
highest priority. Service levels for subscribers at calling stations 64 
are typically arranged at different levels, depending on subscriber 
preference and cost. Once a service level for a subscriber is established, 
the guaranteed service level is stored in the RAS 80 database. 
Alternately, an image of the routing and administration database in the 
RAS 80 may be stored in the ITS 72 to reduce access via the Internet. 
FIG. 7 is a block diagram of the ITS 72 of FIG. 6. The ITS 72 includes a 
telephony platform 100 and an Internet server platform 102. The telephony 
platform 100 performs basic telephony functions, including incoming call 
detection (ringing, trunk seizure, etc.), call supervision/progress 
detection (busy tone, disconnect, connect, recorded announcement, 
dialtone, speech, etc.), call origination, DTMF, call termination, call 
disconnect, switch hook flash, etc. 
As shown in FIG. 7, the telephony platform 100 of the ITS 72 includes a 
simplified message desk interface (SMDI) 104 that sends and receives 
signaling data to the CCS signaling network, a digital switch 106 that 
sends and receives communication traffic from the trunk line 68, a master 
control unit (MCU) 108 that controls the overall operations of the ITS 72, 
including controlling the switch 106 to separate data traffic on the trunk 
line 68 into single 64 kb/s data channels 110. The data on each of the 
data channels 110 is compressed by a voice processor unit (VPU) 112 into 
compressed communication data having a data rate of approximately 16 
kbit/s or lower. The compressed communication data may be either voice 
data or other data, for example facsimile data. 
The compressed communication data is output to a local area network (LAN) 
114, for example an Ethernet-based network at 100 Mbit/s. The LAN 114 
carries data signals between the MCU 108 and the voice processing units 
112. The system also includes T1 type digitized audio links 110 between 
the switch 106 and each of the VPU's 112. The LAN 114 transports data 
packets to a packet assembler/disassembler (PAD) 116 that packetizes data 
on the LAN 114 into TCP/IP packets for transport onto the Internet 74. The 
PAD 116 also recovers signaling and communication data from data packets 
received by the router 118. Hence, the PAD 116 receives signaling 
information from the SMDI 104 originated from the signaling network 70, 
and outputs signaling data recovered from data packets received from the 
Internet 74 to the SMDI 104 for subsequent call processing via the 
signaling links 70. 
The ITS 72 also may include a RAS database 120 that is an image of the 
database in the RAS server 80. The RAS database 120 enables translation 
information to be obtained without accessing the RAS 80 via the Internet 
74. In this arrangement, the ITS 72 would monitor its own bandwidth 
allocation as stored in the RAS database 120. 
The router 118 is of the type now generally used in Internet practice. If 
desired, the router 118 may also be connected to a Domain Name Service 
(DNS) server and a Dynamic Host Configuration Protocol (DHCP) server of 
the type conventionally used by Internet Service Providers in existing 
Internet Service. 
Internet Long Distance Call Processing 
An exemplary call using the arrangements of FIGS. 6 and 7 will now be 
described with respect to FIGS. 9A and 9B. The system of FIG. 6 
establishes an Internet connection to link a calling to a called telephone 
without the necessity of either party possessing or using personal or 
office computer equipment. The subscriber in this example uses the POTS 
station 64a to initiate an Internet call to a called party at the POTS 
station 64b in step 120. The caller goes off-hook and dials *82. This 
prefix has been established by the Telco offering the service as a 
predesignated prefix with which the public may initiate an Internet 
telephone call. The dialing of the prefix *82 is followed by the dialing 
of the directory number of the called party at the station 64b including 
the area code. 
The central office switching system responds to the off-hook and receives 
the dialed digits from the calling station in step 122. The central office 
switching system analyzes the received digits and determines from the 
prefix *82 that the call is an Internet call. Responsive to its 
programming it knows that the call must be completed through a remote 
central office and that further processing is necessary. The originating 
central office 41a suspends the call and sends a CCIS query message in 
step 124 to the ITS 72a via the signaling channel 70. 
In response to the query message, the ITS 72a identifies the internet 
telephony server servicing the called party 64b by sending in step 126 a 
routing request, including the number of the calling party 64a and the 
area code of the called party 64b, to the RAS 80 via the Internet 74. 
Alternately, the ITS 72a may access its own internal routing and 
administration database 120, shown in FIG. 7, which is an image of the 
routing and administration database in the RAS 80. The routing and 
administration database (RAS DB) accesses the internal translation tables, 
shown in FIGS. 10A and 10C, and sends a routing response in step 128. The 
routing response includes the identity (e.g., IP address) of the ITS 72b 
serving the called party 64b, the predetermined virtual path between the 
two servers, and the minimum guaranteed service level for the calling 
station 64a. 
The ITS 72a then sends in step 130 a signaling message in the form of a 
query message packetized in TCP/IP packets having the IP address of the 
ITS 72b as the destination address. The signaling packets are received via 
the virtual paths 88 by the ITS 72b in step 132 and include a session ID, 
the called number, the calling number, and the requested data transmission 
rate having a minimum data rate corresponding to the prescribed service 
level. The ITS 72b recovers the query message from the payload of the 
TCP/IP packets in step 132, and determines whether or not the called 
station 64b is busy in step 134. 
If the called station 64b is busy, the receiving central office 41b so 
informs the ITS 72b via the signaling network 70, and the ITS 72b returns 
a busy message to ITS 72a in step 136 using signaling packets in TCP/IP 
protocol. The ITS 72a recovers the busy message from the received data 
packets via the Internet 74, and informs the originating central office 
via the signaling network 70 of the busy condition. The originating 
central office provides a busy signal to the calling station. 
If the called station is not busy, the receiving central office 41b busies 
out the called station line 64b by blocking all calls. The receiving or 
destination central office 41b then informs the originating central office 
41a via the ITS servers 72b and 72a and the Internet that the called line 
is available and waiting. Specifically, the ITS 72b in step 138 sends a 
data packet including the session identifier and the available condition 
of the called party 64b to the ITS 72a via the Internet. The ITS 72a 
recovers the signaling information including the session ID and available 
condition from the data packet transmitted by the ITS 72b, and responds in 
step 140 to the query from the originating central office 41a. 
Referring to FIG. 9B, an Internet virtual connection is then established 
between the calling and called stations. The receiving or destination 
central office 41b provides a ringing signal to the called station 64b and 
the originating central office 41a sends ringback tone back through the 
local loop 66 to the calling station 64a in step 142. At the same time, 
the ITS 72a and the ITS 72b establish a two-way communication link on the 
predetermined virtual path at the prescribed service level in step 144. 
Specifically, the initial packets transmitted by each ITS 72 will have 
identification information for the destination switches. Alternately, each 
ITS 72 will use the reserved voice path connections for transmitting voice 
data packets. When the called station 64b goes off-hook in step 146 and 
the Internet virtual connection is completed the conversation via the 
Internet can commence in step 148. 
Each of the ITSs 72a and 72b monitor the communication link to detect a 
disconnect in step 150. If a disconnect condition is detected by one of 
the ITSs 72 in step 150 via a signaling message from the corresponding 
central office 64, then the ITS 72 sends a disconnect message as a 
signaling data packet to the corresponding ITS 72 via the Internet 74 in 
step 152. 
In addition, the ITSs 72a and 72b and the RAS 80 monitor the traffic on the 
established virtual communication path. If any of the ITSs 72a or 72b or 
the RAS 80 detects a substantial increase or decrease in traffic, the 
detecting node outputs a signaling data packet indicating the detected 
change to the corresponding ITSs 72a and/or 72b. If in step 154. a 
signaling data packet is received indicating a detected change in the 
traffic on the virtual communication path 88, the ITS servers 72a and 72b 
in step 156 change the data rate based on the received data rate value in 
the signaling data packet and in accordance with the prescribed service 
level. 
FIG. 8 is a block diagram of an alternate implementation of Internet long 
distance service, where an internet module 96 including a router handles 
routing of low-grade Internet telephone calls using conventional 
compression and routing techniques. For example, the originating central 
office 64 may send a CCIS message to the Internet Module 96 including the 
directory numbers of the calling station and the called station and 
requesting establishment of an Internet connection (or virtual connection) 
between the two. 
The router in the Internet Module 96 may then react to receipt of that CCIS 
signal and request the temporary assignment of Internet addresses for the 
processors associated with the respective central offices. Upon completion 
of the assignment of the addresses module 96 may send a CCIS signal to the 
originating central office advising of that fact. When the originating 
central office receives the message that the addresses have been assigned 
the switching system connects the originating local loop to the Internet 
Module 96. 
The Internet Module router then sends a request for the assignment of 
temporary IP addresses for the two directory numbers to a DHCP server 91. 
The DHCP server hears the message and offers an IP address for each 
directory number for a certain time period which may be determined by the 
router or the server. The router may request a specified time period and 
the DHCP server may decline and offer a longer or shorter period, seeking 
mutual agreement. Upon agreement the addresses are accepted and assigned. 
The originating Internet Module 96 next triggers a CCIS message to a 
destination Internet Module (not shown) which includes the temporary IP 
address assigned to the called directory number and associated processor. 
The transmission of data packets through the Internet using the Internet 
module 96 and the DHCP server 91 does not guarantee bandwidth or a minimum 
latency. Hence, if the Internet module determines that the calling station 
is a subscriber that requests high priority traffic, the Internet module 
96 accesses the RAS 80 instead of the DHCP server 91 in order to obtain a 
predetermined communication path reserved for guaranteed bandwidth and 
latency, as described above with respect to FIG. 6. Hence, the Internet 
module 96 performs the functions of the ITS 72 upon detecting a calling 
station having a prescribed service level that requires a guaranteed 
bandwidth by obtaining the routing information from the RAS 80. 
According to the present invention, routing and administration servers 
provide translation addresses for servers acting as interfaces for public 
telephone networks. The Internet telephone servers are thus able to 
determine the network address of a destination server based on the area 
code of a called station. The servers then establish a communication link 
via the Internet and use higher level protocol to divide and distribute 
voice calls through the respective telephone systems. Hence a plurality of 
communications links can be established between two servers while 
minimizing the number of hosts on the Internet. 
In addition, servers exchanging communications traffic via a wide area 
packet switched network can maintain a guaranteed quality of service by 
reserving predetermined virtual paths throughout the packet switched 
network. The predetermined virtual paths thus ensure a guaranteed 
bandwidth and latency for quality long distance service. 
While this invention has been described in connection with what is 
presently considered to be the most practical and preferred embodiment, it 
is to be understood that the invention is not limited to the disclosed 
embodiment, but, on the contrary, is intended to cover various 
modifications and equivalent arrangements included within the spirit and 
scope of the appended claims.