Automated Invoice Validation System

Automated invoice validation requires a file from the telephone company. Most, if not all, telephone company–provided billing information is assembled in standard formats and made available on various transport media or transported across a network connection. If no validation system exists, or for whatever reason it’s desirable to design and build a new one, the process involves getting a file and proceeding to extract the required information from it. The key fields used to match to department allocation are the location and telephone numbers. Information in these fields must be unique. It is highly likely your organization already has unique location names or numbers. In and of itself, the unique name is not sufficient for full allocation. The telephone number making and receiving calls is unique to the telephone company, and they have their own counterpart to the location name, which is a physical port on a switch or access facility into the nearest wire center.

In addition to the key field data, detailed unit count and pricing for all fixed monthly charges and all variable or usage based charges should be extracted and carried over into a file in the organizational accounting system. In addition to the charges for service, taxes and other fees such as access charges and universal service fees should be extracted and maintained in the file. Essentially what has to be done is to take this file with the current charges and perform all the calculations done by the telephone company and re-create the same information, but instead of using the reference pricing they used, use a separate reference database.

Figure 1 is a conceptual diagram of the invoice validation system.

 

Figure 1: Telephone Billing File Interface to Enterprise Accounting System

It must be implemented in the organization’s information technology (IT) infrastructure or built separately and integrated with the other systems.

Note the dotted line from vendor pricing. It will become solid after completion of the standard cost development in steps 4 through 8. Resist temptation to jump too far too fast until you’re confident the following functions are being performed:

• Import the billing file
• Extract the desired data
• Compare the data from the reference file with the billing file and highlights exceptions
• Output a list of invoices with amounts due by department, location, and user name satisfactory for feeding the cost allocation process
• Output a file with taxes and desired details required for allocation and further possible tax treatment
• Output a file with any or all fees in a format acceptable for use by the accounting and reporting systems

This will provide a preliminary look at the difference between manual invoice validation and automated validation. So far the only expenditure is the billing tape (should be under $50 for each telephone company, $100 for one local and one long distance) and the value of the time of the programmer(s). When the bill data of the first service provider(s) is successfully imported into the accounting system, incremental service providers, locations, and service should require less time and occur smoothly. If there are issues with the second, be sure to go back and resolve any common issues with the first one. Over time and with additional service providers, adding more billing into the system will become routine.

Asynchronous Transfer Mode (ATM) System

Asynchronous transfer mode (ATM) is a packet data transmission and switching system that transfers information by dividing all types of data into small fixed length packets of data (53 byte cells). The ATM system uses high-speed transmission (usually 155 Mbps or above) and is a connection-based system. When an ATM circuit is established, a patch through multiple switches is setup and remains in place until the connection is completed. ATM service was developed to allow one communication medium (high-speed packet data) to provide for voice, data, and video service.

As of the 1990’s, ATM has become a standard for high-speed digital backbone networks. ATM networks are widely used by large telecommunications service providers to interconnect their network parts (e.g., DSLAMs and Routers). ATM aggregators operate networks that consolidate data traffic from multiple feeders (such as DSL lines and ISP links) to transport different types of media (voice, data, and video).

The ATM switch rapidly transfers and routes packets to the pre-designated destinations. To transfer packets to their destination, each ATM switch maintains a database (called a routing table). The routing table instructs the ATM switch to which channel to transfer the incoming packet to and what priority should be given to the packet. The routing table is updated each time a connection is setup and disconnected. This allows the ATM switch to forward packets to the next ATM switch or destination point without spending much processing time.

The ATM switch also may prioritize or discard packets that it receives based on network availability (congestion). The ATM switch determines the prioritization and discard options by the type of channels and packets within the channels that are being switched by the ATM switch.

Figure 1, shows a functional diagram of an ATM packet switching system. This diagram shows that there are three signal sources going through an ATM network to different destinations. The audio signal source (signal 1) is a 64 kbps voice circuit. The data from the voice circuit is divided into short packets and sent to the ATM switch 1. ATM switch 1 looks in its routing table and determines the packet is destined for ATM switch 4 and ATM switch 4 adapts (slows down the transmission speed) and routes it to it destination voice circuit. The routing from ATM switch 1 to ATM switch 4 is accomplished by assigning the ATM packet a virtual circuit identifier (VCI) that ATM switch can understand (the packet routing address). This VCI code remains for the duration of the communication. The second signal source is a 384 kbps Internet session. ATM switch 1 determines the destination of these packets is ATM switch 4 through ATM switch 3. The third signal source is a 1 Mbps digital video signal from a digital video camera. ATM switch 1 determines this signal is destined for ATM switch 4 for a digital television. In this case, the communication path is through ATM switches 1, 2, and 4.


Figure 1: Asynchronous Transfer Mode (ATM) Systems

Frame Relay System

Frame relay is a packet-switching technology provides dynamic bandwidth assignment. Implementations of frame relay in 2002 allowed for dynamic bandwidth allocation up to 45 Mbps. Frame relay systems offer dynamic data transmission rates through the use of varying frame sizes. The frame relay system is a connection based switching system. Switches are initially programmed to create a logical path (virtual connection) from the entry point to the exit point.

Frame relay systems are a simple bearer (transport only) technology and do not offer advanced error protection or retransmission. This reduces the time for packet switching (reduced transmission delay time). It is up to the sender and receiver of frame relay data to ensure the integrity of the data. When used in systems that have good digital communication systems, frame relay provides reliable data communication service.

The first frame relay standard I.122 was defined in 1988 by the International Telecommunications Union (ITU). The current frame relay specification standards include the ITU I.233 and American National Standards Institute (ANSI) T1.606.

The key components of a frame relay system include frame relay access device (FRAD) frame relay network devices (FRND) and frame relay switches. The FRAD converts end user data into protocol data unit (PDU) variable length packets. The FRAD communicates to the FRND over an access line (e.g., fractional T1/E1 or ISDN line).

The FRND is a packet switch that also operates as a gateway to the frame relay network. The FRND passes frames it receives from the FRAD to other frame relay switches that forward packets toward their destination network. Frame relay switches have buffer memory that allows them to hold and prioritize packets before they are retransmitted. Packet switches can selectively discard packets if network congestion occurs. The FRAD and FRND provide information about the priority of the frames (e.g., non-essential discard eligibility) and status of the system (e.g., network congestion notification).

The frame relay system uses a discard eligibility (DE) flag system to indicate the essential nature of the packet’s data. The DE flag(s) allow systems to selectively discard data packets or frames that are non-essential. This process allows some data transmission systems to send more data than is agreed to (dynamic bandwidth). If the network is not congested, it may allow the extra packets of data to reach their destination.

Congestion notification is a control flag signaling system that is used to indicate status of network congestion in a data network. Congestion notification allows data communication devices that are connected to the data network to send or delay the sending of data dependent on the status of the network.

The frame relay system uses both forward and backward congestion notification. Forward explicit congestion notification (FECN) indicates to upstream switching devices that data that is being transmitted through congested switches and it is likely that some of the remaining packets may be discarded. The upstream switch can then change the data discard priority level accordingly. Backward explicit congestion notification (BECN) indicates to the sending (downstream) switching devices that congestion is occurring and packets that are received may be discarded. The sending switch can then change the priority of packet discarding and send and indication to other switches indicating network congestion. This should eventually reduce the amount of data end-users are sending into the network.

The frame service provider usually agrees to provide the frame relay service at certain data transmission rate (service level). The frame relay system may provide a committed information rate (CIR) and a maximum burst information rate (BIR).

Figure 1 shows a frame relay system. This diagram shows a local area network (LAN) in San Francisco is connected to a LAN in New York. A virtual path is created through the frame relay network so data can rapidly pass through each frame relay switch as its path is previously established. When data is to be transferred through the LAN (e.g., a large image file), the data file passes through a FRAD that is the gateway to the frame relay network. The FRAD divides the data file from the LAN into variable length data frames. The FRAD sends and receives control commands to the frame relay network that allows the FRAD to know when and if additional data frames can be sent.


Figure 1: Frame Relay Systems

Optical Transmission System

Optical transmission systems use strands of glass or plastic fiber to transfer optical energy between points. For most optical transmission systems, the transmitting end-node uses a light amplification through stimulated emission of radiation (LASER) device to convert digital information into pulsed light signals (amplitude modulation). The light signals travel down the fiber strand by bouncing (reflecting) off the sides of the fiber (called the cladding) until they reach the end of the fiber. The end of the fiber is connected to a photo-detector that converts these light pulses back into their electrical signal form.

Synchronous optical transmission systems used a specific frame structure and the data transmission through the transmission line is synchronized to a precise clock. This eliminates the signaling overhead requirement for framing or timing alignment messages. The basic frame size used in optical transmission systems is 125 usec frames.

Optical transmission systems are characterized by their carrier level (OCx) where the basic carrier level 1 is 51.84 Mbps. Lower level OC structures are combined to produce higher-speed communication lines. There are different structures of OC used in the world. The North American optical transmission standard is called synchronous optical network (SONET) and the European (world standard) is synchronous digital hierarchy (SDH).

Signals are applied to and are extracted from optical transmission systems using an optical add/drop multiplexer (OADM). The OADM is a network element that provides access to all or some subset synchronous transport signal (STS) line signals contained within an optical carrier level N (OC-N). The process used to direct a data signal or packet to a payload of an optical signal is called mapping. The mapping table is contained in the OADM. A copy of the OADM mapping is kept at other locations in the event of equipment failure. This allows the OADM to be quickly reprogrammed.

Synchronous digital hierarchy (SDH) is an international digital transmission format used in optical (fiber) networks standardized that is similar (but not identical) to SONET. SDH uses standardized synchronous transmission according to CCITT standards G.707, G.708, and G.709. These standards define data transfer rates, defined optical interfaces, and signal structure formats.

Some of the key differences between SONET and SDH include differences in overhead (control) bits and minimum transfer rates. The first level available in the SONET system is OC1 and is 51.84 Mbps. The first level in the SDH system starts at STM-1 and has a data transmission rate of 155.52 Mb/s. SONET also multiplexes synchronous transport signal level 1 (STS-1s) to form multiple levels of STS. The SDH system divides the channels into multiple DS0s (64 kbps channels). This is why the overhead signaling structures are different.

Figure 1 shows the optical standards for both SONET and SDH. This table shows that the first common optical level between SONET and SDH is OC3 or STS-1. STS-x and STM-x are the standards that specify the electrical signal characteristics that are input to the respective optical encoding/multiplexing processes.


Figure 1: Optical Transmission Systems

System : Passive Optical Network (PON)

Passive Optical Network (PON)
A passive optical network (PON) combines, routes, and separates optical signals through the use of passive optical filters that separate and combine channels of different optical wavelengths (different colors). The PON distributes and routes signals without the need to convert them to electrical signals for routing through switches.

PON networks are constructed of optical line termination (OLT), optical splitters and optical network units (ONUs). OLTs interface the telephone network to allow multiple channels to be combined to different optical wavelengths for distribution through the PON. Optical splitters are passive devices that redirect optical signals to different locations. ONU’s terminate or sample optical signals so they can be converted to electrical signals in a format suitable for distribution to a customer’s equipment. When used for residential use, a single ONU can server 128 to 500 dwellings. In 2001, most PON’s use ATM cell architecture for their transport between the provider EO or point of presence (POP) and the ONU (in some case even to the user workstation). When ATM protocol is combined with PON system, it is called ATM passive optical network (APON).

Figure 1 shows an APON that locates ONUs near residential and business locations. This network uses ATM protocol to coordinate the PON. ONU interfaces are connected via fiber to an OLT located at the provider’s EO or POP. Each ONU multiplexes user channels (between 12 and 40) into an optical frequency spectrum allocated to that ONU. Up 32 ONU’s can share access to a single PON using the features of dense wave division multiplexing (DWDM). Some newer PON’s use high density wave division multiplexing (HDWDM). Use of HDWDM increases the number of ONU’s per PON from 32 to 64. This diagrams shows that a PON that uses HDWDM can support approximately 2500 residential customers.


Figure 1: Passive Optical Network (PON)

System : Digital Loop Carrier (DLC)

Digital Loop Carrier (DLC)
Digital loop carrier (DLC) is a high efficiency digital transmission system that uses existing distribution cabling systems to transfer digital information between the telephone system (central office) and a telephone or other communication device. There are two types of DLC: universal digital loop carrier (UDLC) and integrated digital loop carrier (IDLC).

The UDLC is a system that consists of RDTs and central office terminals (COTs). Optical systems such as synchronous optical network (SONET) can transfer signals transparently through the COT to the RDT. The RDT provides an interface between the digital transmission line (e.g., DS1) and the customer’s access line. The RDT can dynamically assign time slots from the communication line to customer access lines.

Integrated digital loop carrier (IDLC) is a digital line interface that has been re-engineered to integrate within a switch (usually as card) and shares the internal bus structure of the switch. This function (or card) is called an integrated digital terminal (IDT). Using the IDT, the switch can directly communicate with a remote digital terminal (RDT) that is closer to the end customer using an efficient multi-channel communication line. The RDT provides an interface between the high-speed digital transmission line (e.g., DS1) and the customer’s access line. The RDT can dynamically assign time slots from the communication line to customer access lines. Because customer access lines are not used at the same time, an RDT that interfaces to a DS1 line (24 channels) usually provides service to 96 customer access lines.

The key advantages to DLC carrier systems are the cost effective transmission and the ability to rapidly add, delete, or change customer services without having to dispatch an installation technician. The DLC system offers improved efficiency through the use of existing distribution cabling systems. DLC systems also offer the ability to extend the range of access lines from the central office to the end customer as the RDT effectively operates as a repeater.

An RDT is divided into three major parts: digital transmission facility interface, common system interface, and line interface. The digital transmission interface terminates the high-speed line and coordinates the signaling. The common system interface performs the multiplexing/de-multiplexing, signaling, insertion, and extraction. The line interface contains digital to analog conversions (if the access line is analog) or digital formatting (if the line is digital).

DLC initially allowed 40 analog telephone connections to be extended to the remote neighborhoods using a device called an SLC-40. Later an SLC-96 (known as a “slick 96”) was put into service that allowed 96 voice grade analog circuits to be extended from the CO on just ten (10) pairs thus reclaiming 86 pairs per installation. Still in use the SLC-96 has allowed the LEC’s to conserve much of their installed outside copper infrastructure.

Unfortunately, DLC systems are not transparent to other systems such as DSL systems. Although it is possible to install digital subscriber line network equipment (co-locate) along with RDT equipment, the RDT equipment housings and power supplies were not originally designed to hold additional equipment.

Figure 1 shows the deployment of an integrated digital loop carrier (IDLC) application in a local telephone distribution network. This diagram shows that a switching system has been upgraded to include an IDT and an RDT has been located close to a residential neighborhood. The IDT dynamically connects access lines (actually digital time slots) in the switching system to time slots on the communications line between the IDT and RDT. The RDT can connect to up to 96 residential telephone lines. When a call is to be originated, the RDT connects (locally switches) the residential line to one of the available channels on the DS1 interconnection line. The IDT communicates with the RDT using the GR-303 standard.


Figure 1: Integrated Digital Loop Carrier (IDLC)

System : Digital Subscriber Line (DSL)

Digital Subscriber Line (DSL)
Digital subscriber line is the transmission of digital information, usually on a copper wire pair. Although the transmitted information is in digital form, the transmission medium is usually an analog carrier signal (or the combination of many analog carrier signals) that is modulated by the digital information signal.

A DSL network is composed of several key parts; this includes a local access line provider, DSL access provider, backbone network aggregator, ISP provider, and other media providers. DSL services can be provided by a single service provider or may result from the combination of processes from different service providers. The communication network can be divided into several parts; local access lines (copper), voice communications network (PSTN), high-speed digital subscriber line (DSL), aggregator (interconnection), Internet service provider (ISP) and content provider (media source). These network parts and the service providers who operate them, must interact to provide most DSL services.

The physical parts of a DSL network include a subscriber access device, network access lines and digital subscriber line access module (DSLAM). There are many configuration options for a DSL network. They vary from a simple end-user’s modem bridge that connects a single end-user’s computer to the DSL network to complex multi-channel, asynchronous transfer mode (ATM) systems that connect routers and set-top boxes.

Figure 1 shows the functional parts of DSL network. This diagram shows that end user equipment adapts, or converts analog and digital signals to a high-speed DSL transmission signal via a DSL modem (an ATU-R for an ADSL system). The copper wire carries this complex DSL signal to a DSL modem at that connects to the central office (an ATU-C for an ADSL system) where it is converted back to its analog and digital components. The analog POTS portion of the signal (if any) is routed to the central office switching system. The high-speed digital portion is routed to a digital subscriber line access module (DSLAM). The DSLAM combines (concentrates) the signals from several ATU-Cs and converts and routes the signals to the appropriate service provider network.


Figure 1: DSL Network Diagram

System : Integrated Digital Services Network (ISDN)

Integrated Digital Services Network (ISDN)
A structured all digital telephone network system that was developed to replace (upgrade) existing analog telephone networks. The ISDN network supports for advanced telecommunications services and defined universal standard interfaces that are used in wireless and wired communications systems.

ISDN provides several communication channels to customers via local loop lines through a standardized digital transmission line. ISDN is provided in two interface formats: a basic rate (primarily for consumers) and high-speed rate (primarily for businesses). The basic rate interface (BRI) is 144 kbps and is divided into three digital channels called 2B + D. The primary rate interface (PRI) is 1.54 Mbps and is divided into 23B + D. The digital channels for the BRI are carried over a single, unshielded, twisted pair, copper wire and the PRI is normally carried on (2) twisted pairs of copper wire.

The “B” channels operate at 64kb per second digital synchronous rate and the “D” channel is a control channel. The D channel is used to coordinate (signal) the communication with the telephone network. When used on the BRI line, the D channel is 16kbps and when provided on the PRI channel, the D channel is 64 kbps. Because the amount of telephone system control signaling is relatively small, the D channel can also be used for low speed packet data messaging. The 64 kbps “B” channels can be used for voice and data. On the BRI system, the two B channels can be combined for 128 kbps data connection.

ISDN telephone lines exclusively use digital transmission. This requires a customer to replace their analog telephones with ISDN digital telephone equipment if they upgrade to ISDN service. ISDN service is typically provided using modular plugs. These plugs include a RJ45 interface (8 pin) for data equipment (called a BRI-S/T) and the other physical connection type is a two-wire, RJ11 type standard (called the BRI-U).

The maximum distance for a BRI-S/T line is approximately 3,000 feet and the maximum distance for the BRI-U is 18,000 feet. Beyond these distances, the service provider may install repeaters to provide service. However, repeaters are expensive to install and setup.

The ISDN BRI allows the user to change the use of the B channels whenever desired. For example, an ISDN user may be sending data using the two B channels at 128 Kbps. If a voice call comes in or is initiated, the data transmission is not interrupted; but is automatically reduced to one B channel at 64 Kbps. When the voice call ends, the data transmission returns to 128 Kbps on the two B channels.

Figure 1 provides the different interfaces that are available in the integrated services digital network (ISDN). The two interfaces shown are BRI and PRI. These are all digital interfaces from the PSTN to the end customers network termination. 1 (NT1) equipment. devices that are ISDN compatible can directly connect to the NT1 connection. Devices that require other standards (such as POTS or data modems) require a terminal adapter (TA).


Figure 1: Integrated Digital Services Network (ISDN)

Carrier System : Digital Signal Level (DSx)

Digital signal level (DSx) transmission is a hierarchy of digital communication channels and lines that range from 64 kbps to 565 Mbps. Lower level DS structures are combined to produce higher-speed communication lines. There are different structures of DS levels used throughout the world with significant variations between North American and European systems. DSx has been used to represent the digital transmission standards where the “x” denotes which service is under discussion.

Trunk carrier (T Carrier) is often used to describe the DSx level. Trunk carrier uses (Tx) to represent the digital transmission standards where the “x” denotes the multiplexed level of trunk service. Tx is the actual transmission structure where DSx is the digital signal levels. Outside the United States, E Carrier (Ex) is used to represent the transmission carrier.

The following represent the North American DS structure:

Digital Signal 0 (DS0) - is the smallest digital channel operating at 64 kbps. It represents a single digitized analog voice channel;

Digital Signal 1 (DS1) – normally referred as a T-1, it is composed of twenty four voice channels packed into a 193 bit frame and transmitted at 1.544 Mbps. The unframed version, or payload, is 192 bits at a rate of 1.536 Mbps;

Digital Signal 2 (DS2) – normally referred to as a T-2, it is composed of four T-1 frames packed into a higher level frame transmitted at 6.312 Mbps;

Digital Signal 3 (DS3) – normally referred to as a T-3, it is composed of twenty-eight T-1 frames packed into a level frame transmitted at 44.736 Mbps.

Fractional T1 (FracT) - A data transmission rate that is a portion of the total capacity of a T-1 communications line (1.544 Mbps) but greater than a DS0 (64Kbps).

European standards are different from the North American noted above. The European hierarchy is as follows:

Digital Signal 0 (DS0) - is the smallest digital channel operating at 64 kbps. It represents a single digitized analog voice channel;

Digital Signal 1 (DS1) – normally referred as an E-1, it is composed of thirty voice channels and two controls channels. It uses a 256-bit frame and operates at 2.048 Mbps.

Digital Signal 2 (DS2) – normally referred to as a E-2, it is composed of four E-1 frames packed into a higher level frame transmitted at 8.448 Mbps;

Digital Signal 3 (DS3) – normally referred to as an E-3, it is composed of sixteen E-1 frames packed into a level frame transmitted at 34.368 Mbps.

Digital Signal 4 (DS4) – normally referred to as a E-4, it is composed of sixty-four E-1 frames packed into a higher level frame transmitted at 139.268 Mbps;

Digital Signal 5 (DS5) – normally referred to as an E-5, it is composed of two hundred fifty-six E-1 frames packed into a level frame transmitted at 565.148 Mbps.

Digital Signal Level (DSx)