PACKET SWITCHING

One only need examine history over the past couple of hundred years to see that older communications models and methods bear resemblance to some of the current crop of fast growing methods. For example it’s not difficult to see the similarity between smoke signals and telegraph messages. If one takes the smoke signal model and imagines a sender sending a message to a receiver and the receiver repeating the message to another smoke sender and so on, it’s easy to see the resemblance to packet forwarding characteristics of the Internet protocol or email.

If you have any interest at all in the Internet and have done any reading on the subject, you’re aware that it’s based on packet switching. Packet switching depends on some basic functional elements including transmission links, and a switching engine called a router.

There are a lot of similarities between cell-based switching and packet-based switching, and there are some differences. Packet switching is simply making decisions about where to send the packet at hand. Packets are like cells in the sense that they must be opened, intelligence found about where they are headed and where they have been, and then switched and/or routed. The instructions are just inside the packet with a few other tidbits of information.

One of the fundamental differences between packet switching and cell switching is at the heart of most of the ambiguity and hand-wringing that occurs when considering the routing-switching—layer 2-layer 3 solution. It’s really quite simple. Cells, PDH streams, and PPP (HDLC) are layer 2 functions. What separates these techniques from packet techniques is a time base. Packets, at least IP datagrams that make up user datagram protocol and Telecommunications protocol over Internet packets, have no reference or relationship to any clocking, timing, or basic synchronizing intelligence. They are just out there somewhere in the Ether. T1/E1, PPP/HDLC, ATM, and Ethernet all have clocking and synchronizing information embedded in the stream.

Packet switching is nothing more than switching and/or routing at packet borders, or between packets after the details in the packet header have been opened and read. Only after the entity has been opened and read can it be routed or switched to a second port. Many times the mail system is used as a metaphor for packet switching. It’s a pretty good metaphor, but with some subtle differences. First, the packet entity must be opened and read. It does not have an outside and an inside unless the payload has been encrypted, or otherwise sealed and secured in some way. One of the fundamental flaws in the Internet everyone knows about and experiences every day is simple courtesy and security. In addition to the payload and addressing information, there are other significant details inside the packet entity exposed for any and everyone to see and do with as they please. These other details have to do with all kinds of fun things that can muck up the overall machinery such as administrative control of the routing machines.

A view that says the Internet has evolved from prior well-known methods and technology wouldn’t be difficult to contend, but would likely be more difficult to defend. Many modern IP network designers seem blissfully unaware that the Internet is critically dependent on an underlying transmission infrastructure they simply refer to as the network layer. Very few of them have a clue about the importance of network clocking and timing. Many think packet over SONET/SDH isn’t a big deal because it’s done all the time (over PDH), which for the most part goes over SONET/SDH any way. Very few realize that HDLC, or PPP framing, is as rigid and fixed as T1, E1, T3, or E3. A few understand the details of packet-over SONET/SDH. The ones that do understand this fundamental know that Internet architecture includes layer 1 and layer 2 and is not, as the rest of their esteemed colleagues contend, self-healing.

Throw in all the mumbo jumbo about connection-oriented and connectionless protocols and mumble solution in between every fifth use of the word router or whatever else can be thought of, but the basics remain the same. That is, something on the premises, or at the network access point, contains or establishes intelligence that tells the network how to set up a connection between two or more points, and thereby transmit and receive information through the network. Anyone can play around with semantics all day about dumb terminals and smart networks, or at the other extreme, they call intelligent terminals and dumb networks. At the end of the day, what’s important is effective and efficient use of limited resources.

If confusion reigns, stop and ask a couple basic questions: ‘‘Is it circuit, cell, or packet?’’ ‘‘What is being shared?’’ Is it time, bandwidth, or both? What are the interface, bitrate, and active protocols on the facility? What is supposed to be done with it? What did the customer ask for? What is being delivered? Is it broke? With a little patience and perseverance, confusion will soon stop raining, the clouds will pass, and matters will clear up as you climb up or down the stack of bits and bytes.

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.

CHANNEL RATE VS. MULTIPLEX RATE

One of the major points of confusion, both inside and outside the Telecom industry is the misunderstanding and misuse of channel interface and multiplex rate terms. In publication after publication, the PDH hierarchy is explained by simply calling out the various multiplex levels or signal rates without delineating the difference between the rates that specify channel rates through the network, and the rates resulting from multiplexing.

Channel rates apply to channel interface, which is the point of network access and egress. Most of the PDH multiplex aggregate rates and channel rates covered by global and national standards. Basically, the standards are structured toward use in three regions: Europe and the rest of world (E), except North America (T) and Japan (J). Multiplex levels generally are designated with a DS, meaning digital signal (level) and channel interface and bandwidth with a T, E, or J.

Another factor that is often completely missed is the fact that telecom networks are dominated by voice traffic. The nature of voice traffic caused the network to evolve into circuit switched 64 Kbs channels. These channels are multiplexed into higher order bit streams. An OC192 transmission facility configured to handle only voice traffic contains 192-DS3 equivalents, each capable of 672 64 Kbs channels for a total of 129,024 unique, independent segments of bandwidth. An STM64 configured to handle only voice traffic contains 122,880 bandwidth segments. What about using some of this bandwidth to carry something of a different nature than a 64 Kbs voice? It is entirely possible, and that’s where network channel interface comes into play. From a network transmission facility perspective, a bit is a bit. The transmission network doesn’t know or care if the bit carries voice, data, or anything else, including errors, as long as it meets network clocking and time duration parameters. What this means is that the transmission network will accept access and egress or drop and insert segments of bandwidth that just happen to be sized and structured to fit in place of 24 (T1), 32 (E1), 480 (E3), 672 (T3) voice grade equivalent, or 64 Kbs channels.

Another point of confusion is unchannelized or clear channel rates. This type facility is used for non-voice channel applications such as Internet access, wholesale encrypted traffic (encryption applied to a T1 or group of individual channels rather than individually encrypting each channel), asynchronous transfer mode (ATM) network, physical convergence layer protocol (PLCP) access, compressed program content over E3 or DS3, and other applications that require contiguous bandwidth that cannot operate over 64 Kbs channelized facilities. IP routers are likely to have a PDH wide access network interface at T1/E1/J1 or E3/J3/DS3.

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.

CLOCKING, TIMING, AND SYNCHRONIZATION | Network Technology

Regardless of the type of transmission media (wire, radio, or fiber), bits are sent and bits are received. Every receive port in the network has to deal with an incoming serial bit stream that includes clocking and payload bits. Clocking and data, also called payload, have their timing and phasing relationships established at the point of creation. Along the way, the serial bit stream may be multiplexed with additional serial bit streams, cross-connected to a different carrier, or switched at the circuit, cell, or packet layer. Yes, this is layer 1 and 2 of the OSI stack. Separating clock information from payload, or mucking around with the time relationship between signal transitions amounts to errors. If, for whatever reason, a network element loses its synchronization reference and wanders outside holding limits, anything and everything using it as a synchronization reference is out of time with the larger network, converting valuable data to invaluable trash.

Clocking and data recovery are a critical function. Considering the clocking concept from the perspective of a receive port on a network element or the receive end of a transmission path gives the ability to look backward toward the source and forward toward other network elements and facilities dependent on the clock for proper operation and delivery of the associated payload.

If there is a single point in the entire end-to-end, top-to-bottom process that is the most critical in moving digitized information through a network, it has to be at each and every receive point in the network. Receiving the bitstream intact and then extracting clocking information must be done before the payload can be extracted. Even if the receiver is clocked externally, clock and data signals in the incoming stream must be separated and defined. Payload framing depends on timing relationships, and guess what, timing depends on clocking. Serial data streams just aren’t real if they don’t have a mechanism delineating framed data and clock signals. Understanding clock and data recovery as a stand-alone function is one of the keys to understanding digital communications networks. Figure 1 is a block diagram of a typical clock and data recovery function found in almost every network element.

Figure 1: Clock and Data Recovery Functional Block Diagram

The functions in Figure 1 include receiving light or radio waves from the media, detecting and converting it to electrical voltage transitions. This requires a lens or optical interface in the case of light waves. The equivalent in radio waves would be an antenna. The purpose of either is to detect and focus the received energy into a signal containing the time variations between and during presence and absence of signal. The signal is then passed to a detector, which converts the signal into a series of transitions representing the serial bitstream originally sent by the transmitter.

The raw bitstream is then fed to a clock extraction circuit and a buffer. The clock is extracted and used for two purposes: one is a clock signal representing the original clock used when the data was created or multiplexed, and the other drives a data recovery circuit. The data recovery circuit extracts and re-clocks the data to restore as precisely as possible the transitions between 1s and 0s making up the data. At this point, the data is simply just that. We don’t know anything about framing or channelization and nothing about protocols, byte size, packing, or error correction. These are all to be determined with additional processing, and take place through the upper layers of the ISO stack.

The clocking signal is representative of the original clock sent by the transmitter. It also has a lot of unknowns, but the important point to recognize is that this signal represents timing from another node or remote site outside the physical boundaries of the receiving equipment site. This signal could be important in the overall scheme of things or it could be irrelevant. For example, it could be used as a timing reference for the entire site. If that were the case, then many other considerations become important. If it is used as a timing reference for the entire site, there is no question that it will continue being proliferated to other sites because it becomes the clock that turns up as the receive clock after clock and data recovery at other sites receive signals from the instant site.

Naturally, this begs a few questions: ‘‘Where did that clocking signal originate?’’ ‘‘What is its level of accuracy?’’ ‘‘Is it a network clock?’’ Clocking in communications networks is as critical and necessary as synchronizing and time code in television audio and video. Timing in communications networks can be as complicated or as simple as timing in digital audio and video systems. Clock accuracy in communications networks is built around a four-level hierarchy with the most accurate clock at the first level and least accurate at the fourth level. Originally developed by Bell Labs for AT&T’s digital network in the 1960s, adopted and standardized by ANSI, ITU, and other standards-making bodies, the system is referred to as stratum 1 through stratum 4. In the beginning, there was only one stratum 1 clock in operation for the entire network at any given time. This clock was distributed to regional switching and operating centers and used to time and synchronize network elements at that level in the hierarchy and so on to local central offices at the bottom of the hierarchy. Each clock level depended on the one above it for synchronization. If it lost the reference signal from above, the local clock was permitted to operate within a range wider than the higher reference until the higher-level reference was once again available.

As you can imagine, there’s not much of a disturbance when the reference clock disappears. But what about when the reference clock returns or if it is intermittent? Dealing with these situations led to the development of a hold-over specification, which basically requires the clock to remain within, or hold its frequency of operation for a given time and, in the interest of overall stability, not switch back to the higher hierarchy reference clock until after some period of stable operation, and if possible, switch gracefully, sometimes called hitless switching.

Primary reference clock (PRC) products compliant with ANSI and ITU standards are available from many sources. These clock sources can run independently, or they can be referenced and locked to other sources traceable to the World Timing Standard for time-of-day, called Coordinated Universal Time (UTC).^[2] UTC is the result of combining TAI (International Atomic Time) and Universal Time 1 (UT1). TAI is a timing reference derived by averaging outputs from the clocks of approximately 100 countries. These clocks keep the timing relationship to each other within 2 to 3 millionths of a second over a year. UT1 provides a correction to compensate for the difference in solar time and TAI caused by slightly elliptical orbit and polar inclination, both of which affect solar time.

Distributing the PRC to all the elements in early digital network infrastructure was expensive, cumbersome, and complex. However, advances in lowering the cost of clock reference products and the availability of the global positioning system (GPS) and Internet-based references running under network time protocol (NTP) have dramatically reduced cost on all fronts and improved reliability and accuracy of the sources and their references.

Why all the fuss and bother? Essentially, signals derived from multiplexing low-speed digital bit streams into higher-order aggregate bit streams, some of which provide private line, and others that provide circuit-, cell-, and packet-switched services must be synchronized and timed exactly the way digital audio and video signals are timed and synchronized, including SMPTE time code, to bit level accuracy within a frame. The only differences are that Telecom digital signals aren’t subject to switching transitions such as a split screen or cross-fade, and telecom and digital program content run on entirely different time bases. One other point is that this timing accuracy has absolutely nothing to do with the transmission media—it makes no difference if the transmission is satellite or terrestrial radio, optical transmission, or baseband electrical signal transport.

One thing to pay attention to in systems where the bit stream is modulated onto a carrier is the capability of a particular modulator to lock the carrier frequency generator to an external source such as the incoming digital signal or PRC. On the receive end, it may be appropriate to lock the local receiver beat frequency oscillator in the receiver to either a PRC or the incoming digital signal clock. Including these capabilities in a piece of equipment is not without cost; however, those that do provide that capability will be capable of better bit error performance across the transmission facility because of the absence of asynchronous cross talk causing clock slips.

Lastly, be careful not to confuse network primary reference clock with program clock reference in Moving Picture Experts Group (MPEG). The two are completely different animals and have nothing to do with each other. When MPEG2 program streams are transported through networks, the PCR is just another set of bits in the payload. 

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.

LOCAL AREA OR LOCAL AREA NETWORK CONNECTION

LAN as used in this document means specific versions of IEEE 802.1, 802.3, and other standards applicable to network interfaces, referenced in a manufacturer’s product documentation. Terms such as 10BT, 10/100BT, Fast Ethernet, Gigabit Ethernet, and 1000BT are informal references. Where doubt exists, it is recommended that designers rely on direct personal knowledge of formal IEEE or other applicable standards and seek clarification of applicability to a particular manufacturer’s products from a representative knowledgeable about design details of the equipment.

Each standard Ethernet interface is assigned a unique 48-bit identification code by the original equipment manufacturer pursuant to a registry agent defined in IEEE standards. This code is referred to as a MAC address (meaning Media Access Control). The MAC address is inserted into Ethernet packets and enables devices connected to the LAN to establish sessions and route packets around or outside the LAN. When packets are routed outside the LAN, it’s likely the MAC address is translated to an IP address, which becomes the basis for IP network transport. MAC and IP addresses are the functional equivalent of telephone numbers.

Ethernet interfaces enable user access, and connectivity between system components and peer platforms. The minimum required configuration is a simple Ethernet LAN segment. In more complex arrangements, the LAN segment provides access to MAN and WAN network facilities and services. If the network facilities are designed and implemented properly, use of any application on the system by any user with network access and user privileges is possible.

Most operating systems including LAN interface drivers permit manual or automatic, so-called plug-and-play configuration of the network interface. Microsoft Windows 2000 Workstation and Server and Windows XP Professional workstation include real-time communications client software. If the NIC is capable of being configured for QOS, then end-to-end real-time content transport is possible if it is configured. This is the first step in gaining end-to-end real-time content transport across the network between server platforms.