COMMUNICATIONS NETWORK SWITCHING AND TRANSMISSION ARCHITECTURE

Telecom network architecture in 2004 is very much a child of many years of growth and technical evolution driven by telephone calls. Telephone calls require 64 Kbs and are point-to-point connections used for short periods. Telephone service economics where the longer the distance between two points of a call costs more money influenced most of the traffic into local and regional calling patterns. These regional calling patterns led to hierarchical network architecture with high capacity switch sites in fewer numbers piecing together circuits to enable regional and national telephone calls.

Figure 1 is an illustration of the hierarchical nature of the telephone network.

 

Figure 1: Classical Telecom Hierarchical Architecture

The access tandems and certain designated transmission facilities are generally referred to as the access network. These switches formed the technical and economic basis for a philosophy the FCC dubbed equal access. Next time you pay a telephone bill, look at the term FCC access charge. Nope, the money doesn’t go to the FCC. It goes into the owner of the access facilities. And according to FCC regulations, the owner of the access facilities in each LATA must afford equal access to any inter-exchange carrier (IXC) willing to arrange for a connection between their facilities and the local exchange carrier’s (LEC) facilities.

On the other side of that equation, equal access means the LEC must offer their subscribers the ability to select their long distance, or IXC.

The higher levels in the hierarchy vary, depending on the size and extent of the individual long distance carriers. AT&T operates much as it did before divestiture. Others, such as MCI and Sprint, have flatter or fewer levels in their switching hierarchy. All the long distance carriers have gateways to international networks and their own shares in international transmission facilities. Piecing together two or more SONET links end to end creates an extension of the physical boundaries of the network with the addition of more links (Figure 2).

 

Figure 2: SONET Transmission Segment

All the layers are extended across the interface. Each layer becomes an extension of itself each time a SONET transmission facility is connected to another. Linking two or more in series along a path or route, extends the boundary or domain as defined in the facility specification. Figure 2 shows a block diagram of SONET transmission infrastructure linked to provide continuous bandwidth that could be used to provision an E-1/T-1/J1 or E3/DS3/J3 private line. Such a facility could be used as a link in a data communications network or to connect two routers in an inter-network or intra-network link.

Another method of linking facilities is the mid-span meet, commonly used by carriers to link or interface between networks. This is depicted in Figure 3 and shows links between LECs and IXCs.

 

Figure 3: Linked SONET Transmission Facility

A variation on the light wave terminal equipment (LTE) is the add-drop multiplexer (ADM). SONET transmission capacity is built around the 51 Mbs STS. An LTE breaks out all the STS in the optical carrier (OC) and makes them available. If the LTE is used at a location where all STS are not terminated in local routing/switching equipment, then any transit streams must be connected directly to other LTE facing other routes. The ADM breaks out and terminates a limited quantity of STS from the optical carrier. For example, the office in which the terminal is located might only require 1 or 2 STS from an OC-3, OC-12, or higher rate transmission facility. Using an ADM would be less costly in terms of capital investment and operations expense.

CIRCUIT SWITCHING | Network Technology And Methodology

Circuit switching and routing is the basis for all domestic and international telephone or voice grade, dial-up traffic. The circuit switching function is distributed between end-office switching systems and network switching systems. End office switching systems may be a private branch exchange (PBX) physically located on subscriber premises, or a partition in the telephone company’s nearest office, commonly referred to as centrex service. Network switching systems include the local serving central office and any other systems facilitating a path for a telephone call. Nowadays, these systems range in size from a few thousand to hundreds of thousands of ports capable of handling millions of calls per hour.

Circuit switching in functional terms is nothing more than connection of one transmit-receive pair on one side of a switch to a transmit-receive pair on another port on the same path or route, or a different path or route, sometimes called the other side of the switch. Tandem switches are nothing more than transit points that link up network or inter-network transmission facilities. For example, each of the 200+ local access and transport areas (LATA) in the United States has a minimum of one tandem switch, which acts as the transit point between the access and transport networks used by long distance carriers to carry calls from one LATA to another.

Voice grade dial-up service is almost all digital in the United States. However, many analog switches remain in other parts of the world. Where digital switches provide the service, integrated services digital network (ISDN) services—really an access method, not a service—is available. In highly populated areas of many countries, digital subscriber line (DSL) access is available and growing.

Transmission bandwidth available in circuit switched facilities varies from below 64 Kbs (rarely more than 49 Kbs) to 1.536 Mbs. The limitation in analog service is a matter of the ability of a modem to talk to another modem over a local telephone loop. Of course, it doesn’t much matter to voice grade service. After all most, if not all, telephone equipment is bandwidth limited to around 3.5 kHz, which fits easily into 8-kHz sampled PCM.

ISDN and DSL access provide higher capabilities though. ISDN Access is either 144 Kbs, called basic rate interface (BRI), or1.544 Mbs, called primary rate interface (PRI). BRI is channelized into three channels, two bearer or B channels at 64 Kbs, and one delta or data or D channel (16 Kbs) used for signaling and control purposes. PRI access is facilitated with T1 transmission facilities and is channelized into 23 to 64 Kbs B channels and 1 to 64 Kbs D channel. It should be emphasized that the previous explanation is purely in terms of technical capability. Leveraging the bandwidth into variable amounts and getting charged for it on a case-by-case, service-by-service basis is an entirely different matter.

For example, ISDN-based Internet access never achieved large usage because the equipment used by ISPs and their users was limited to BRI rates—64 Kbs at best. And because the ISPs are not the telephone company and have no capability, such as a big digital circuit switch, and have no funds available to buy a big digital circuit switch and therefore no interest in competing with the telephone company, they only offer Internet access service. From the telephone company viewpoint, they simply are prevented from being in the data services—Internet access, or Internet service provider (ISP) business—by current FCC rules and legislation. The telephone company can only sell POTS, ISDN, or private line service. It cannot offer any type of switching other than these services. Some of the independent non–regional Bell operating telephone companies have purchased and operate ATM equipment, but basically they are quite limited in the service they can provide using these or other non-voice service, frame relay, and IP-based switching and routing systems.

DSL access varies according to several factors, the main one of which is the distance between the subscriber premises equipment and nearest central office or wire center. Conceptually and technically, DSL access is intended to be capable of multiple service types such as voice and data. However, implementation reality has driven most service providers to offer only Internet access without any voice service initially. It remains to be seen how long this is likely to continue. The classical telephone companies don’t want to cannibalize their bread and butter—lucrative voice services—and they desperately want to tap into new revenue streams of their up and coming competitors—cable modems and DSL-capable ISPs. Therefore, initial DSL service is limited to Internet access. As the Internet matures—achieves a grade and quality of service capable of supporting voice-over IP—this situation will change. Who knows when, but someday in the future it may be possible to call up the telephone company and ask them to discontinue POTS.

Keep in mind that the main purpose of the switching function is to share use of the transmission function. Also, keep in mind the fact that change in the network is more a direct result of economic pressure than technological or regulatory forces.

ROUTING | Network Technology

Routing and switching mean different things to different people. For example, both terms apply to switching and transmission facilities. Routing in the circuit switched or voice services world means that a call is routed according to service configuration parameters in a PBX or end office switch. For example, least-cost routing is established when the originating switch is programmed to use the least expensive route between the origination point and termination point for a telephone call. Alternate routes may be a point-to-point private line, a virtual private network (VPN), or the public network, where the private line is the lowest and fixed cost, the VPN is the next least expensive, and the public network is the most expensive.

Wholesale routing of traffic occurs when the traffic is moved from one transmission facility to another. For example, traffic normally routed from New York to Atlanta may go direct, but an alternate route may pass through Cincinnati. A national fiber ring might have a southern route and a northern route, as a regional ring might have an eastern route and a western route. Traffic normally on an Intelsat transponder facility might be moved to another route using a transponder on a PanAmSat satellite.

In the data world, routing becomes more of a technical issue because of the underlying network technology. From a classical perspective, data networks were built using leased or private line facilities provided by carriers on a 24/7 basis at fixed prices. For an enterprise with a headquarters located data center and several field operations, private or leased lines are used to connect computer terminals to the data center. Depending on the number and geographic locations, all the field offices may be connected directly to the data center in a home run arrangement. However, if two or three of the remote operations were physically close to one another, it may make sense to hub them into one common location, aggregate traffic, and connect to the data center over a common facility. In this case the hub becomes a routing point. All traffic from the other nearby locations is routed through the hub to the data center.

Data communications routing can become very complex and confusing because of the proliferation of various flavors of ATM, Ethernet, frame relay, and IP techniques, to say nothing of classic SNA, X.25, HDLC/SDLC, BISYNC, and others.

In circuit switching, routing intelligence is in the user’s head and the network, and is used to tell the network how to route a call or make a connection. The same basic principle applies to ATM, frame relay, and IP networks as well. In the circuit switched network, the routing intelligence resides in the common channel interoffice signaling system and its configuration software. In ATM, the intelligence is included in each cell header and distributed across switching machine configurations. In Ethernet, frame, and IP networks, it’s in the packet headers and distributed across configuration parameters in switching machines swooned over and lovingly called routers—except frame relay, which is typically a meshed, point-to-point arrangement and therefore has limited connection capability.

BASIC FRAMING | Network Technology

Framing bits are also applied when a basic 1.544 Mbs stream is multiplexed with another stream or additional streams into higher order aggregate signal. Additional bits are added in specific timeslots and designated as framing bits to enable the receiving equipment to recover the original clock and separate the payload, first into the next lower order bitstream and then ultimately down through the multiplex hierarchy to the original 1.544 Mbs payload and 24 individual 64 Kbs voice channels or DS0 signals.

This makes the multiplexing bit oriented. That is, each stream is multiplexed into a specific pattern based on individual bits where each bit in each frame has a specific (theoretical) timing relationship to the same timeslot in peer bit streams. Because of the fact that each of the original 1.544 Mbs bit streams is generated from a clock that runs in the real world, and may not be precisely on the same frequency as any of the others generating the T1 signals being multiplexed, and because the timing of the signals being multiplexed may change due to propagation delay variation in the transmission media, the resulting aggregate signal multiplexing is said to be plesiochronous, meaning almost or nearly synchronous, but not asynchronous or non-synchronous. Multiplexing of signals from disparate clocks that are almost or nearly synchronous requires another technique called bit stuffing.

Bit stuffing is exactly what it’s name implies, adding or ‘‘stuffing’’ bits into a multiplexed stream to raise the speed, or number of bits per unit of time, so there are enough bits to fill the timeslots in the higher order channel. For example, when 4 DSI signals are multiplexed to make up a DS2 signal, one of the signals is sent at exactly, 1,544,000Bps, one has 364Bps, the third gets 314Bps, while the fourth gets 414Bps, making an aggregate for the DS2 of 6,306,272Bps.

When 7 DS2 signals are used to build up a DS3, still more bits are added to each DS2 to enable the network to accommodate the disparate nature of the various DSI clocks and multiplexing operation.

The T-carrier concept originated in the United States, but was followed in due course in other countries. Initially designed for four-wire media, it found its way to coaxial cable, wireless, and optical fiber media. An international version of the DS1 is called an E1. It uses the same 8 Kbs sampling and 64 Kbs DS-0 channel rates, and 125-microsecond framing. However, 30 timeslots are placed in the 125-microsecond frame, resulting in a payload of 240 bits per frame, a payload rate of 1.920 Mbs, and a total channel rate of2.048 Mbs.

Another major difference and significant improvement of E1 over T1 structure is increased overhead. From the start, this was a troubling characteristic of T1, not because it was too much, but because it wasn’t enough. There was never a standard method, nor enough bits to deal with the many overhead requirements for voice service. Besides, when data transport came on the scene, US DS0 channels could reliably deal with only 56 Kbs instead of the entire 64 Kbs bandwidth. So the designers of E1 digital facilities added 2 to 64 Kbs channels providing 128 Kbs. Adding these two timeslots in the 125-microsecond frame resulted in 256 bits in each frame, 240 for payload and 16 for overhead.

BASIC NETWORK ELEMENTS AND FUNCTIONS | Network Technology And Methodology

Communication network architecture (yes, including the Internet) includes six critical functional elements, or capabilities: clocking, multiplexing, routing, signaling, switching, and transmission.

Clocks control basic timing in digital networks. Digital networks simply wouldn’t work without accurate, consistent, long-term, stable clocking and timing mechanisms. The basic clocking scheme used to maintain timing and synchronization in networks is not much different than it was when first conceived in the 1950s, except it’s significantly more accurate and much less expensive, especially at the higher levels of accuracy and precision.

Multiplexing enables two or more signals to share time and/or bandwidth of a common facility. Multiplexing gains greater use of a limited resource. Multiplexing was a key characteristic of early analog telephone systems. Analog multiplexing shares frequency spectrum instead of time. Multiplexing can be active or passive. Active multiplexing involves electronic circuitry, while passive multiplexing, sometimes referred to as combining and filtering, requires no power supply, and attenuates the signals being combined.

Demultiplexing simply reverses the multiplex process. The multiplexing techniques used in classical T-carrier networks are active at the bit level. Timing differences between signals generated by disparate clocks running within frequency tolerance specification limits, along with a variation in propagation delay of the transmission path require the use of bit stuffing techniques to avoid clock slips and errors in transmission.

Routing in its broadest context applies to multiple ways to get from here to there, or connect point A to point B. A router or routing switcher in a broadcast facility is a drastically different beast than a router that can pass Internet packets from one port to another. Routing telephone calls and configuring private line connections play an important part in the global communications network today and are likely to remain so well into the future.

Signaling is the mechanism whereby customers, subscribers, and users (through equipment) communicate with the network to setup and tear down a connection, or configure it for initial use, or reconfigure it for different use (i.e., change the default service configuration). Signaling is also a process whereby network elements communicate with each other in response to commands from users for service, or the owner for changes in configuration or service capability. Successful signaling depends on a logical addressing or numbering scheme whereby all the elements in and outside the network carry a unique identification label.

Switching has been around since someone had a hunch that telephone service could take a cue from the railroads and get more use from fewer telephone lines by installing a switching point somewhere in the service area. From automatic switch-over when a transmission backbone segment fails, to provisioning private lease lines, to telephone service, data communications, audio and video conferencing, content creation, distribution, and delivery, modern communications networks simply wouldn’t do what they do so well without it. Switching concepts include circuit switching, cell switching, and packet switching.

Transmission is the act of propagating energy or moving information from point A to point B. In the context of communications networking, the term includes sending and receiving. If the heart of the network is the clocking system, transmission is analogous to the arteries and capillaries carrying oxygen from the lungs to the brain and other important organs. Modern communications network transmission seems to have started when someone figured out that a direct current voltage applied to one end of a pair of wires could be detected at the other end as long as the conductive characteristics of the path are intact. Without the underlying transmission facilities, today’s IP would be of no more value than Samuel B. Morse’s telegraph code without a baseband electrical signal transmission facility. Successful transmission requires a viable medium. Electrical transmission works well on copper wire. Radio transmission moves easily through free space, where electrical current doesn’t travel well. Light waves move through transparent glass, but opaque objects block them.

HISTORICAL BACKGROUND SUMMARY | Network Technology And Methodology

Between around 1960 and 1980, the public switched telephone network underwent rapid and dramatic change from developments in solid-state digital technology. Initially, the diode and transistor were single function devices, but it didn’t take long for them to be packaged into containers and branded integrated circuits. Computers—large, slow data processing machines and systems—were not immune to the same technological turmoil. Consequently, computers and their terminals migrated across the scientific landscape into office territory. Connections between the computers changed significantly as Teletype machine controllers turned into timeshare terminals. Someone figured out a way to convert the digital signal between the timeshare terminal and the computer from digital to analog, and reverse the process at the other end; devices made with modulator and demodulator techniques extended acronym territory with the term MODEM. All of a sudden the analog telephone network could connect timeshare terminals and computers as well as the Teletype network could. Originally, telephone networks were analog. Modems allowed telephone networks to be used to support computer communications.

As this initial impact from transistors and integrated circuit electronics enabled faster and faster computers, it had a similar effect on network technology. Bell Labs started working on digital transmission technology in the 1960s. The objective was to double voice channel capacity of a single trunk line from 12 simultaneous conversations to 24. This technology had tremendous value in large cities where the potential return was superior compared to digging up the street and burying more conduit.

Throughout the 1970s and 1980s, the long distance switching and transmission network underwent a conversion from analog to digital. Mini-computers replaced many mainframes; mainframes became faster and computer traffic grew. Data communications became full-time jobs for communications-savvy engineers and technicians.

A significant computer standard, developed in the late 1970s, remains in wide use today. The open systems interconnect (OSI) stack defines a hardware section beneath a software section in a total of seven layers, bottom to top. The OSI stack makes a good framework for communications networks, including the Internet. Figure 1 shows the two-section, seven-layer stack with a brief explanation about what it represents and how it is applied. 

Figure 4-1: OSI Stack: Hardware and Software Sections

When the OSI stack was introduced, computers were just beginning to change from stand-alone islands into distributed processing systems connected by data networks. The basic idea behind the stack concept is that each layer interfaces and interacts or communicates with the one immediately above and below, except, of course, the bottom and top layers for obvious reasons. If each layer successfully accomplishes its functions, then the system it’s applied to should operate top to bottom. Attempts to map Internet and Telecom functions or processes to the stack are made from time to time, but in isolated ways such as a reference to layer 2 switching,or layer 3 routing, or even layer 2/3 switching or routing. These references seem to be more of a way to characterize a particular switching or routing function in terms of the OSI stack, rather than applying the OSI stack to communications networks in general. Furthermore, it would seem to be useful in analyzing and structuring or designing networks capable of carrying disparate, converged traffic types on a common access or transport facility.

DESIGN CONSIDERATIONS AND CRITERIA

As with any project, content transport network design and construction is not successful without some amount of performance criteria. Content transport networks are different than ordinary voice and data networks. Ordinary voice and data networks don’t typically do content transport well. On the other hand, design and build a network capable of transporting valuable program content, and voice and data can come along for the ride. Below is a list of considerations and criteria that can be used when preparing to undertake a content transport network project. The following list is not to be taken literally, nor is it exhaustive or all-inclusive:

§  Access, switching, and transport elements

§  Access and transport facilities can be terrestrial, satellite, or a combination

§  Switch facilities will be time (TDM), cell (ATM), or packet (IP)

§  Service availability is full-time 24/7 or shared

§  Availability, reliability, robustness, grade, and quality of service

§  Capital and operating cost

§  Geographical or physical coverage includes local (LAN), metropolitan (MAN), regional, national, and global turf (WAN)

§  LANs may have single or multiple segments covering a room, floor, building or group of buildings in a campus arrangement

§  A MAN typically involves third party telco or ISP service and uses standard telephone facilities, such as E1/T1, E3/DS3

§  WAN extends LAN and MAN to wider geographic areas not covered by local telephone companies and ISPs

Content transport networks can be built or bought, but practical realization is a combination of buying equipment and the rights to use facilities and services.

Live, streaming content requires continuous, uninterrupted connections with an equal amount of bandwidth. That’s the theory; however, in practice it’s always prudent to leave just a tad of headroom. So how much is a tad? Practicality drives such in the form of how the service provider divides up the bandwidth and sells it. For example, a 10 Mbs ATM or IP network facility likely won’t be precisely 10 Mbs. These animals usually break out in increments of octal numbers. So somewhere around 10 Mbs will be something like 10240000. If that is your choice of network transport channel, then the compression system output bitrate should be set at some number less than the channel rate. This parameter is also a victim of practical circumstances as well because these devices commonly have to deal with octal numbers. So a tad in practice happens to be the difference between the highest speed the encoder can be set at, and the channel rate. (See Appendix II for an example of calculating payloads and matching channel rates.)

Non–real-time content can be transported using continuous, uninterrupted connections, but it can also be carried on discontinuous bandwidth connections, usually at lower cost and improved utilization of the facilities. Be aware that realization of lower cost is dependent on obtaining use of facilities and services at unit prices based on time used and type of bandwidth occupied for each session or transmission just like the old fashioned long distance telephone call.

Standard network performance and characteristics must be understood before they can be applied to content transport networks. The next few paragraphs provide an introduction to time division multiplexing (TDM), ATM, and IP network technology.

TDM technology characteristics and performance are the standard cell and packet based network performance should be measured against. If a standard for TDM is required, then use wire, fiber, or another passive conductor of known performance. The characteristics of interest include available channel bandwidth, bit error rate, and jitter. However, in cell and packet networks, bit errors cause cell and packet loss or impairment, as can jitter.

ATM transport technology offers 5 classes of service. Constant bit rate (CBR), variable bit rate—real-time (VBR-rt), variable bitrate— non–real-time (VBR-nrt), unspecified bit rate, and available bit rate. While it may change in the future, ATM CBR is currently the only ATM class of service capable of transporting high-quality, high bit rate content in real time.

Packet-switched networks are inherently chaotic unless specifically configured to deal with continuous signal, or mixed-signal traffic and class-of-service. Packet networks are either Ethernet or IP. (Several packet or packet-like techniques exist; however, they only support content transport as a file transfer, not real time.)

In general there are two types of IP technology and methodology: Ethernet and Internet. The IEEE 802.1 standard defines Ethernet. Internet or more precisely, IP is defined in RFC 791. Ethernet transport of IP is defined in RFC894.

Ethernet architecture is built around shared media in the form of common set of cabling where the information is carried in packets, and the device such as a workstation or server listens or monitors the buss before attempting to establish a connection or session. The way the process works, end-to-end, has the sender and all the receivers constantly listening or monitoring the buss. A session is kicked off after a sender sends an initial transmission to all stations using a unique address. If the initial transmission has a valid destination address, that is an actual receiver connected to and listening to the buss, it responds with an acknowledgement. After the sender receives the acknowledgement, then and only then do the two computers establish a connection and carry on with the session using their unique address information.

IP networks, the Internet in particular, behave in similar fashion as Ethernet.

All these types of transport work well for moving files, including hypertext markup language—coded pages, fixed images, and other static objects. Uncongested networks may even support low volume continuous signals such as produced by voice or telephone service over IP, and even ‘‘work okay’’ with higher bandwidth continuous signals. Make no mistake about it though, unstructured networks cannot be relied on for transport of continuous signal, high bit rate, valuable content such as audio, video, closed captioning, control, or other signals associated with, or embedded in, program content.

Reliable, predictable, safe, and secure content transport requires network connections with sufficient bandwidth, grade, and quality of service (GOS, QOS). Even non–time-sensitive or non–real-time transport—so-called FTP—should be planned and implemented with care because of the size of the files and the time required to move them have significant economic implications.

Obtaining sufficient bandwidth, GOS, and QOS is a matter of specifying and configuring LAN, MAN, and WAN network resources.

Sufficient, continuous bandwidth means the network must exhibit bandwidth equal to or greater than the bandwidth of all traffic, not just program content if the network is required to accommodate email, web surfing, network management, and perhaps voice. Insufficient network bandwidth results in denial of service or, at best, delayed service. Program content payload bandwidth is roughly equivalent to the sum of compressed audio, video, and other signals multiplexed into a program stream or included in a file object stored on the system. When more than one real-time stream is present on the interface simultaneously, the aggregate of all the program streams cannot exceed the bandwidth available on the interface points of the sending and receiving systems and the network connecting the systems. In other words, the bandwidth of the sending and receiving systems must equal or, preferably, exceed the aggregate of all traffic.

GOS means the network connecting all workstations and servers must be available to all users within the design limits agreed to or promised to its users. For example, telephone network services use statistical probability based metrics to define and measure GOS level, inside and outside the network. A P.01 GOS means the network is designed and performs, or doesn’t perform, within the limits of probability that the network will enable the user to complete the call in 99 of 100 attempts. This model can be applied to workstations, servers, and a LAN, MAN, WAN or combination of all and will perform satisfactorily 99 of 100 times when someone wants to transfer a file, or set up and use a connection to deliver streaming content originating on a server platform and terminating in one or more peer platforms at other locations or interfaces served by the network.

QOS means that the quality of the connection in terms of bandwidth, bit-error rate (BER), jitter, packet loss or any other parameter the payload may be sensitive to, is of sufficient level to support program content transport between and amongst the service points. The basic model for this category of network is classic TDM facilities found in ANSI/ITU standards-based networks. The acid test of performance is measurement and comparison to TDM private line facilities such as E1/T1, E3/DS3, and OC3/STM1. A good question of network equipment, facilities, and service suppliers is: Can you emulate T1, or DS3, etc.? The right answer is not ‘‘Yes.’’ The right answer is, ‘‘You can expect jitter, packet loss and bit error rate performance of x, y, and z. This compares to TDM emulation performance of x, y, and z.’’ Then you can decide if the differences fit into your required performance and compare one supplier to another.

Ingest, play out, and file transfer of program content as promised in many product and service descriptions require network connections with sufficient bandwidth, GOS, and QOS. Even non–time-sensitive or non–real-time transport—so-called FTP—should be planned and implemented with care because the size of the files and the time required to move them have significant economic implications.

Standard, so-called out-of-the-box or plug-and-play default LAN configuration included with recent generation Microsoft Operating systems (OS; W2000 Workstation & Server; WXP) enable non– real-time or FTP program content transport. Connect Ethernet to a network interface card (NIC) with access to the Internet, install the OS, run the Internet wizard, and voila! Instant success. No further fuss or effort and file transfer across the Internet from one host to another is possible.

These operating systems also permit configuration of an NIC to enable QOS as specified in IEEE 802.1p, a method whereby packets carrying continuous content can be marked and differentiated so LAN segments can isolate and protect the content from the effects of congestion and chaos mentioned above. Ethernet packets mapped to IP enable QOS marking to be passed to the IP network. If the network has differentiated services capability, real-time content transport across the network is possible. Older operating systems (NT 4; 95/98) do not include 802.1p/QOS capability.

Two types of connections are possible, and both may be required by the application. These include Unicast, or point-to-point, and multicast, or point-to-multipoint. These types of connections enable single or multiple deliveries of files or streams, sometimes referred to as objects.

The basic elements of a content transport network include customer premises equipment (CPE), access facilities at each location, and backbone transport in between. The equipment must be selected and configured to support the level and type of traffic. For example, if the traffic is program content only, that’s one set of circumstances. If the network is to carry voice, data, and provide Internet access, that’s another. If the network is to carry multiple types of traffic, the equipment and facilities will have to be structured to accommodate it. Figure 1 shows a general reference architecture capable of supporting voice, data, and content transport.

Figure 1: Premises Equipment Architecture

There are several characteristics of the architecture that should be pointed out and commented on. First, note the presence of a network clock reference and a separate station synchronizing reference. Neither has anything to do with the other and that’s the point. The network clock reference is to make sure the network is stable and jitter-free because it must carry the embedded program clock reference along with the content. After all, if the network isn’t capable of carrying the program clock reference to a satisfactory degree of accuracy, then the content will suffer impairment.

Although there appears to be a single-thread router and network interface, this is purely symbolic, and emblematic of the same level of redundancy as implied in the private branch exchange, LAN router, and Moving Picture Experts Group (MPEG) Codec. Resolving reliability, robustness and network performance concerns may require redundant equipment and facilities, with emphasis on content value and specific traffic levels. The terms and symbols are generic and intentionally chosen to cover several alternatives without stating them implicitly. For example, any new facility design should take a serious look at voice-over IP telephone service. New installations or even replacement/upgrade installations, may find economic advantage in fully integrated voice and data on LAN wiring. And although, not likely, it may be more appropriate to use ATM switching and transport for real-time program content than IP or TDM transport.

On the network side, there are similar issues and concerns; however, they must be addressed with carriers or service providers instead of manufacturers of equipment. As a design exercise, network access, transport, and switching should logically follow the food chain whereby the network facilities support movement of content within and between the creation, distribution, and delivery sections of the model. For example, moving raw, unedited content from a location to an editing facility, or moving finished program material from the post-production facility to a network operations center. And of course there’s the end link, which requires the content to be moved from anywhere else to cable head end, DBS uplink, Internet access facility, or digital television transmitter input.

Figure 2 is a network topology diagram showing the details of how the basic elements fit into an overall architecture serving users located at separate sites, or operating centers.

Figure 2: Reference Architecture

All the various elements must be specified and priced out in detail. CPE is a capital investment. Access and backbone transport is an operating expense and can be provided by third parties, such as Internet service providers (ISPs), ILECs, CLECs, or inter-exchange carriers. Obviously, it is advantageous to deal with a single source for these services. Decisions on the end-to-end solution should only be made after following a due diligence process. Building the simplest of networks is not easy. Scaling start-up or small networks to larger networks becomes geometrically more complex. Churn and change after a network is built, debugged, and operational can be risky and should not be attempted without careful planning and deliberate, task oriented, sequential steps.

Similar to the end-to-end service model, the reference architecture simply lays out the functional components and shows how they relate. The NID or premises equipment interfaces and interoperates with the network to set up and tear down connections, monitor performance, and process alarms. The desired content transport network leverages one or more routing, switching and transport capabilities, depending on requirements and configuration of the access facilities. In situations where multiple types of traffic are converged onto a common access facility, the access facility must be channelized and mapped to the particular transport. For example, voice grade dialup or switched service would have channel capacity sufficient to accommodate peak voice demand on the public switched telephone network (PSTN) or integrated services digital (ISDN) network. However, if the design called for voice-over IP, some amount of bandwidth would be required to accommodate a similar level of voice traffic.

CPE is a router configured to connect to peer routers at the other locations. The router must be sized and have features selected to perform the functions required by the servers. These functions vary and depend on the encoded bit rate or payload of the content and level of traffic.

Another factor for careful consideration is distance between peer devices at other locations. If the distance is short, such as a nearby building, Ethernet could be an option. But outside adjacent buildings within a campus environment it’s likely construction and capital cost will quickly add up to make third-party service providers with IP transport capability attractive.

Access facilities provide basic connectivity between the CPE and the MAN or WAN. Likely alternatives include E3/DS3 or OC3/STM1. Choosing an appropriately sized access facility is a matter of making a conservative estimate of initial traffic level, then monitoring the traffic and adjusting capacity to levels consistent with acceptable utilization and growth plans.

Backbone transport varies based on requirements and usually comes with significant and critical services attached. For example, the access facility is a dumb, point-to-point, TDM unchannelized facility. But routed networks include services such as routing and configuration protocols, IP address provision, service configuration and management that depend on processing functions resident in edge and core routers. It naturally follows that the owner of the core backbone equipment should provide these services.