Broadband Network
Content
Broadband networks The ideal telecommunication network has the following characteristics: broadband, multi-media, multi-point, multi-rate and economical implementation for a diversity of services (multi-services) [1][2]. The Broadband Integrated Services Digital Network (B-ISDN) provides these characteristics in today's networks. Asynchronous Transfer Mode (ATM) is a target technology for meeting these requirements and is widely deployed as a broadband network [2]. Modern communication services Society is becoming more informationally and visually oriented. Personal computing facilitates easy access, manipulation, storage, and exchange of information, and these processes require reliable data transmission. The means or media for communicating data are becoming more diverse. Communicating documents by images and the use of high-resolution graphics terminals provide a more natural and informative mode of human interaction than do voice and data alone. Video teleconferencing enhances group interaction at a distance. High-definition entertainment video improves the quality of pictures, but requires much higher transmission rates. These new data transmission requirements may require new transmission means other than the present overcrowded radio spectrum [3][4]. A modern telecommunications network (such as the broadband network) must provide all these different services (multi-services) to the user. Differences between traditional (telephony) and modern communication services Conventional telephony communicates using: • • • the voice medium only, connects only two telephones per call, and uses circuits of fixed bit-rates.
In contrast, modern communication services depart from the conventional telephony service in these three essential aspects. Modern communication services can be: • • • multi-media, multi-point, and multi-rate.
Multi-media A multi-media call may communicate audio, data, still images, or full-motion video, or any combination of these media. Each medium has different demands for communication quality, such as: • • • bandwidth requirement, signal latency within the network, and signal fidelity upon delivery by the network.
The information content of each medium may affect the information generated by other media. For example, voice could be transcribed into data via voice recognition, and data commands may control the way voice and video are presented. These interactions most often occur at the communication terminals, but may also occur within the network Multi-point A few examples will be used to contrast point-to-point communications with multi-point communications. Traditional voice calls are predominantly two party calls, requiring a point-to-point connection using only the voice medium. To access pictorial information in a remote database would require a point-to-point connection that sends low bit-rate queries to the database and high bit-rate video from the database. Entertainment video
applications are largely point-to-multi-point connections, requiring one-way communication of full motion video and audio from the program source to the viewers. Video teleconferencing involves connections among many parties, communicating voice, video, as well as data. Offering future services thus requires flexible management of the connection and media requests of a multi-point, multi-media communication Multi-rate A multi-rate service network is one which flexibly allocates transmission capacity to connections. A multimedia network has to support a broad range of bit-rates demanded by connections, not only because there are many communication media, but also because a communication medium may be encoded by algorithms with different bit-rates. For example, audio signals can be encoded with bit-rates ranging from less than 1 kbit/s to hundreds of kbit/s, using different encoding algorithms with a wide range of complexity and quality of audio reproduction. Similarly, full motion video signals may be encoded with bit-rates ranging from less than 1 Mbit/s to hundreds of Mbit/s. Thus a network transporting both video and audio signals may have to integrate traffic with a very broad range of bit-rates [3][5]. A single network for multiple services [edit] Traditional networks Traditionally, the various services mentioned above were carried via separate networks: voice on the telephone network, data on computer networks or local area networks (LANs), video teleconferencing on private corporate networks, and television on broadcast radio or cable networks.These networks are largely engineered for a specific application and are not suited to other applications. For example, the traditional telephone network is too noisy and inefficient for bursty data communication. On the other hand, data networks which store and forward messages using computers have very limited connectivity, usually do not have sufficient bandwidth for digitised voice and video signals, and suffer from unacceptable delays for the real-time signals. Television networks using the radio or the cable medium are largely broadcast networks with minimum switching facilities [2][3]. [edit] Benefits of a single network for multiple services It is desirable to have a single network for providing all these communication services in order to achieve the economy of sharing. This economy motivates the general idea of an integrated services network. Integration avoids the need for many overlaying networks, which complicates network management and reduces flexibility in the introduction and evolution of services. This integration is made possible with advances in broadband technologies and high speed information processing [2][3]. [edit] Fibre optics for broadband networks and MSO While there are multiple network structures capable of supporting broadband services, an ever increasing percentage of broadband and MSO providers are opting for fibre optic network structures to support both present and future bandwidth requirements. CATV (cable television), HDTV (high definition television), VoIP (voice over internet protocol), and broadband internet are some of the most common applications now being supported by fibre optic networks, in some cases directly to the home (FTTh – Fibre To The Home). These types of fibre optic networks incorporate a wide variety of products to support and distribute the signal from the central office to an optic node, and ultimately to the subscriber (end-user). [edit] Broadband Traffic [edit] Types of traffic carried by the network Modern networks have to carry integrated traffic consisting of voice, video and data. The Broadband Integrated Services Digital Network (B-ISDN) satisfies these needs [7]. The types of traffic supported by a broadband network can be classified according to three characteristics [6]:
• • •
Bandwidth is the amount of network capacity required to support a connection. Latency is the amount of delay associated with a connection. Requesting low latency in the Quality of Service (QoS) profile means that the cells need to travel quickly from one point in the network to another. Cell-delay variation (CDV) is the range of delays experienced by each group of associated cells. Low cell-delay variation means a group of cells must travel through the network without getting too far apart from one another.
[edit] Requirements of the different types of traffic The different types of traffic found in a broadband network (with examples) and their respective requirements are summarized in Table 1. Table 1: Network traffic types and their requirements [6] TRAFFIC TYPE Constant Variable Available EXAMPLE Voice, guaranteed circuit emulation Compressed video Data REQUIRED BANDWIDTH Minimal Guaranteed Not guaranteed Low Variable Widely variable Low Variable CELL-DELAY VARIATION LATENCY
Digital switches Digital switches work by connecting two or more digital circuits together, according to a dialed telephone number. Calls are set up between switches using the Signalling System 7 protocol, or one of its variants. In U.S. and military telecommunication, a digital switch is a switch that performs time division switching of digitized signals.[10] This was first done in a few small and little used systems. The first product using a digital switch system was made by Amtelco. Prominent examples include Nortel DMS-100, Lucent 5ESS switch, Siemens EWSD and Ericsson AXE telephone exchange. With few exceptions, most switches built since the 1980s are digital, so for practical purposes this is a distinction without a difference. This article describes digital switches, including algorithms and equipment. A digital exchange (Nortel DMS-100) used by an operator to offer local and long distance services in France. Each switch typically serves 10,000-100,000+ subscribers depending on the geographic area Digital switches encode the speech going on, in 8000 time slices per second. At each time slice, a digital PCM representation of the tone is made. The digits are then sent to the receiving end of the line, where the reverse process occurs, to produce the sound for the receiving phone. In other words, when you use a telephone, you are generally having your voice "encoded" and then reconstructed for the person on the other end. Your voice is delayed in the process by a small fraction of one second — it is not "live", it is reconstructed — delayed only minutely. (See below for more info.) Individual local loop telephone lines are connected to a remote concentrator. In many cases, the concentrator is co-located in the same building as the switch. The interface between remote concentrators and telephone switches has been standardised by ETSI as the V5 protocol. Concentrators are used because most telephones are idle most of the day, hence the traffic from hundreds or thousands of them may be concentrated into only tens or hundreds of shared connections. Some telephone switches do not have concentrators directly connected to them, but rather are used to connect calls between other telephone switches. These complex machines (or a series of them) in a central exchange building are referred to as "carrier-level" switches or tandems. Some telephone exchange buildings in small towns now house only remote or satellite switches, and are homed upon a "parent" switch, usually several kilometres away. The remote switch is dependent on the parent switch for routing and number plan information. Unlike a digital loop carrier, a remote switch can route calls between local phones itself, without using trunks to the parent switch. Telephone switches are usually owned and operated by a telephone service provider or carrier and located in their premises, but sometimes individual businesses or private commercial buildings will house their own switch, called a PBX, or Private branch exchange. The switch's place in the system Telephone switches are a small part of a large network. The majority of work and expense of the phone system is the wiring outside the central office, or the outside plant. In the middle 20th century, each subscriber telephone number required an individual pair of wires from the switch to the subscriber's phone. A typical central office may have tens-of-thousands of pairs of wires that appear on terminal blocks called the main distribution frame or MDF. A component of the MDF is protection: fuses or other devices that protect the switch from lightning, shorts with electric power lines, or other foreign voltages. In a typical telephone company, a large database tracks information about each subscriber pair and the status of each jumper. Before computerization of Bell System records in the 1980s, this information was handwritten in pencil in accounting ledger books. To reduce the expense of outside plant, some companies use "pair gain" devices to provide telephone service to subscribers. These devices are used to provide service where existing copper facilities have been exhausted or by siting in a neighborhood, can reduce the length of copper pairs, enabling digital services such as ISDN or DSL. Pair gain or digital loop carriers (DLCs) are located outside the central office, usually in a large
neighborhood distant from the CO. DLCs are often referred to as Subscriber Loop Carriers (SLCs), after a Lucent proprietary product. DLCs can be configured as universal (UDLCs) or integrated (IDLCs). Universal DLCs have two terminals, a central office terminal (COT) and a remote terminal (RT), that function similarly. Both terminals interface with analog signals, convert to digital signals, and transport to the other side where the reverse is performed. Sometimes, the transport is handled by separate equipment. In an Integrated DLC, the COT is eliminated. Instead, the RT is connected digitally to equipment in the telephone switch. This reduces the total amount of equipment required. Several standards cover DLCs, including Telcordia Technologies Generic Requirements documents GR-8-CORE and GR-303-CORE. Switches are used in both local central offices and in long distance centers. There are two major types in the Public switched telephone network (PSTN):
1. Class 4 telephone switches designed for toll or switch-to-switch connections. 2. Class 5 telephone switches or subscriber switches, which manage connections
from subscriber telephones. Since the 1990s, hybrid Class 4/5 switching systems that serve both functions have become common. Another element of the telephone network is time and timing. Switching, transmission and billing equipment may be slaved to very high accuracy 10 MHz standards which synchronize time events to very close intervals. Time-standards equipment may include Rubidium- or Caesium-based standards and a Global Positioning System receiver. Switch design Long distance switches may use a slower, more efficient switch-allocation algorithm than local central offices, because they have near 100% utilization of their input and output channels. Central offices have more than 90% of their channel capacity unused. While traditionally, telephone switches connected physical circuits (e.g., wire pairs), modern telephone switches use a combination of space- and time-division switching. In other words, each voice channel is represented by a time slot (say 1 or 2) on a physical wire pair (A or B). In order to connect two voice channels (say A1 and B2) together, the telephone switch interchanges the information between A1 and B2. It switches both the time slot and physical connection. To do this, it exchanges data between the time slots and connections 8000 times per second, under control of digital logic that cycles through electronic lists of the current connections. Using both types of switching makes a modern switch far smaller than either a space or time switch could be by itself. The structure of a switch is an odd number of layers of smaller, simpler subswitches. Each layer is interconnected by a web of wires that goes from each subswitch, to a set of the next layer of subswitches. In most designs, a physical (space) switching layer alternates with a time switching layer. The layers are symmetric, because in a telephone system callers can also be callees. A time-division subswitch reads a complete cycle of time slots into a memory, and then writes it out in a different order, also under control of a cyclic computer memory. This causes some delay in the signal. A space-division subswitch switches electrical paths, often using some variant of a nonblocking minimal spanning switch, or a crossover switch.
Digital Transmission Hierachies
There are two hierarchial structures that exist for digital networks: Plesiochronous hierarchies Synchronous hierarchies To further complicate things, North American standards (usually derived from US standards bodies) are different from the International CCITT (now ITU-T) recommendations.
Plesiochronous Hierarchies
In a Plesiochronous hierarchy, the higher level multiplex functions include "bit stuffing" techniques. This allows the input bit streams from I/O channels to use "free-running" clocks. As such, the user's clock rate is propagated (plus a little "Jitter") through the higher level multiplexer. Slip rates requirements between End-User multiplex equipment must still be met, for adequate performance of voice and (particularly) data.
North American Digital hierarchy
The North American Digital hierarchy starts off with a basic Digital Signal level of 64 KBPS (DS0). Thereafter, all facility types are usually referred to as "T x", where "x" is the Digital Signal level within the hierarchy (e.g. T1 refers to the DS1 rate of 1.544 MBPS). Up to the DS3 rate, these signals are usually delivered from the provider on Twisted-Pair or Coaxial cables. North American T1 service providers often refer to the signal interface between the User and the Network as "DS-1" signals. In the case of User to User interfaces, the term "DSX-1" is used to describe those DS1 signals at the "cross-connect" point.
Name ---DS0 DS1 DS1C DS2 DS3 DS4 Rate ----------64 KBPS 1.544 MBPS 3.152 MBPS 6.312 MBPS 44.736 MBPS 274.176 MBPS
International (CCITT) Digital hierarchy
The CCITT Digital hierarchy's basic level is the DS0 rate of 64 KBPS. These signals are usually delivered from the provider on Twisted-Pair or Coaxial cables.
Name ---DS0 E1 Rate ----------64 KBPS 2.048 MBPS
E2 E3 E4
8.448 MBPS 34.368 MBPS 139.264 MBPS
Synchronous Hierarchies
In the later 1980s, synchronous network hierarchies were defined. In Synchronous networks, all multiplex functions operate using clocks derived from a common source.
North American SONET (Synchronous Optical NETwork)
This system is based upon multiples of a fundamental rate of 51.840 MBPS, called STS-1 (Synchronous Transport Signal, Level 1). The facility designators are similar, but indicate the facility type, which is usually Fiber Optic Cable (e.g. OC-3 is an Optical Carrier supporting a STS-3 signal; while OC-12 supports a STS-12 signal, etc). Some typical rates are listed below:
Name Rate ------ --------------STS-1 51.840 MBPS STS-3 155.520 MBPS STS-9 466.560 MBPS STS-12 622.080 MBPS STS-48 2488.320 MBPS STS-192 9953.280 MBPS STS-768 39813.120 MBPS
International SDH (Synchronous Digital Hierarchy)
This system is based upon a fundamental rate of 155.520 MBPS, three times that of the SONET system. This fundamental signal is called STM-1 (Synchronous Transport Module, Level 1). The typical transmission media is defined to be fiber, but the Broadband ISDN specification does define a User-Network Interface (UNI) STM-1 (155.520 MBPS) operating over coaxial cables. Some typical rates within this hierarchy:
Name Rate ------ --------------STM-1 155.520 MBPS STM-3 466.560 MBPS STM-4 622.080 MBPS STM-16 2488.320 MBPS STM-64 9953.280 MBPS STM-256 39813.120 MBPS
Synchronous Digital Hierarchy (SDH)
Introduction The Transmission System is traditionally seen as the link between main
WAN switching centres. These Transmission Systems consist of large bandwidth highways that form the backbone to the network. They typically serve many customers each with their own requirements so the systems have to be reliable, resilient and flexible. Rather than have two wires for every voice or data conversation, Time Division Multiplexing is used. ITU-T G.704 defines 32 channels of 64Kb/s to form 2.048Mb/s where channel 0 is used for framing. You will often see the standard G.703 mentioned with G.704, this is because G.703 defines the unframed physical interface coaxial (75 ohm) or RJ48 (120 ohm) used for the E1/T1 connection at the client premises. Channel 0 is for timing used to synchronise the multiplexers at each end of the link. Channels 1 to 15 and 17 to 31 are for voice or data whilst channel 16 is used for Common Channel Signalling (CCS) or Channel Associated Signalling (CAS). Every 3.91 microseconds 8 bits from one channel is sent down the line followed by 8 bits from the next channel during the next 3.91 microseconds and so on in a round robin fashion throughout all the channels, thus 32 channels are used once every 125 microseconds. The connection at the end is either a 75 ohm coax, 120 ohm coax or a 150 ohm UTP/STP.
Plesiochronous Digital Hierarchy (PDH)
As bandwidth demand grew the technology called Plesiochronous Digital Hierarchy (PDH) was developed by ITU-T G.702, whereby the basic primary multiplexer 2.048Mb/s trunks were joined together by adding bits (bit stuffing) which synchronised the trunks at each level of the PDH. 2.048Mb/s was called E1 and the hierarchy is based on multiples of 4 E1s. • • • • E2, 4 x E1 - 8Mb/s E3, 4 x E2 - 34Mb/s E4, 4 x E3 - 140Mb/s E5, 4 x E4 - 565Mb/s
The E3 tributaries are faster than the E2 tributaries, E2 tributaries are faster than the E1 tributaries and so on. These need to be synchronised with other tributaries, so extra bits are added called Justification bits. These tell the multiplexers which bits are data and which are spare. Multiplexers on the same level of the hierarchy remove the spare bits and are synchronised with each other at that level only. Multiplexers on one level operate on a different timing from multiplxers on another level. For instance, the timing between Primary Rate Muxes (combines 30 x 64Kb/s channels into 2.048Mb/s E1) will be different from the timing between 8Mbit muxes (combines up to 4 x 2Mb/s into 8Mb/s).
Inserting and dropping out traffic from different customers can only happen at the level at which the customer is receiving the traffic. This means that if a 140Mb/s fibre is near a particular site and a new customer requires a 2Mb/s link, then a whole set of demultiplexers are required to do this.
Synchronous Digital Hierarchy (SDH)
Management is very inflexible in PDH, so SDH was developed. Synchronous Digital Hierarchy (SDH) originates from Synchronous Optical Network (SONET) in the US. It includes capabilities for bandwidth on demand and is also made up of multiples of E1. STM-1 (155Mb/s) is 63 x E1, STM-4 (622Mb/s) is 4 x STM-1 and STM-16 (2.5Gb/s) is 4 x STM-4. The benefits of SDH are: • • • Different interfaces or different bandwidths can connect (G708, G781). Network topologies are more flexible. There is flexibility for growth.
• •
The optical interface is standard (G957). Network Management is easier to perform (G774 and G784).
Existing PDH can interface into SDH. There are three G transmission series recommendations that are very important: • • • G.707 - SDH Bit Rates G.708 - The SDH Network Node Interface. G.709 - Synchronous Multiplexing structure.
With the exception of 8Mb/s, different PDH outputs are 'mapped' into Containers (C) and then into fixed size Virtual Containers (VC). When the VC is aligned in the Tributary Unit (TU) a Pointer is added which indicates the phase of the particular VC. TU's are then grouped, via Time Division Multiplexing (TDM), into Tributary Unit Groups (TUG). The TUGs are collated into Administrative Units (AU) via more VCs where more pointers are added (these being fixed relative to the frame). The VCs and the pointers are incorporated into the section overhead of the Synchronous Transport Module (STM). One AU forms an STM-1, 4 AUs form an STM-4. You can also get STM-16 and STM-64. Let us follow a 2Mb/s pipe through the hierarchy. The 2Mb/s PDH first enters a container C12 which compensates for the varying speeds via the use of stuff bits (R). Stuff opportunities are identified by S1 and S2 and these are controlled by the control bits C1 and C2 respectively. If the C bits are are 0s then the corresponding S bits contain data and if the C bits are 1s then the S bits are not defined. In the diagram below, O represents Overhead channel bits and I represents Information bits.
To create the VC12 a Path Overhead (POH) is added. The POH uses Bit Interleaved Parity (BIP) to monitor errors. In addition, there are fault indicators, Far End Block Error (FEBE), Remote Fail Indicator (RFI) and Far End Receive Failure (FERF). The Signal Label is normally set at 2 to indicate asynchronous data. A pointer is added to the VC12 which defines the phase alignment of the VC12 and this changes during transmission. Phase variation can be due to Jitter (from regeneration and multiplexing equipment) and Wander (temperature differences within the transmission media). VC12s created by different multiplexers may not be synchronous so the TU adds a pointer at a fixed position within the TU. The value of the pointer indicates the start of the VC12. If the phase of the VC12 changes then the value of the pointer changes such that if data is running faster than the TU then the pointer value is increased and if the data speed is slower then the pointer value is decreased. This difference in speed can be up to one byte per frame in SDH. The following diagram illustrates three TU12s entering a TUG2 at three different times with the VC12 pointers indicating where the POH is for each:
The TU12 is multiplexed into a TUG 2 along with 2 other TU12s. This is achieved by interleving the bytes of each TU12 in turn. Next, seven TUG 2s are byte interleaved into a TUG 3 and then three TUG 3s can be byte interleaved to form the VC4 (see the SDH diagram above). You can see that 3 x 7 x 3 = 63 2Mb/s circuits can be contained in VC 4.
Network synchronization Network synchronization are used to maintain and synchronize timing across a computer network. Network administrators use network synchronization products to manage time-sensitive services such as streaming audio and video, data transfer and encryption, client-server time accuracy, Web conferencing, backup and redundancy, and database communications. Network synchronization products are also used with client computers, printers, and phones with file, printer, and application servers. A network clock is a network synchronization product that use the network time protocol (NTP) standardized in RFC 1305. To synchronize timing across a local area network (LAN), network clocks match NTP with an external source such as a global positioning satellite (GPS) or lowfrequency (LF) time signaling. Additional interfaces for network synchronization include simple network time protocol, daytime protocol (RFC 867), and user datagram protocol/Internet protocol (UDP/IP). Network timers or master clocks are network synchronization products that use a microprocessor and internal network oscillator to synchronize applications that require analog clocking. In simplest terms, oscillators are circuits that use amplification and feedback to generate radio frequency (RF) outputs. Because network timers use microprocessors, they can be manufactured as programmable erasable read-only memory (PROM) or electrically erasable read-only memory (EEPROM). This enables chip designers to program devices that can produce an RF signal after a given timing period. Typically, this timing period begins or ends when a switch actuates the program. Network synchronization products can be categorized into hardware components and software applications. The proper selection of software and hardware depends on considerations such as time sources, synchronization frequency, time models, network models, and installation specifics. Network synchronization products also include software applications that are designed to display and monitor the times on servers, client computers, and domain controllers. These applications provide management tools to monitor synchronization activities on a network, such as data and file transfers, application-to-database communications.
In contrast, modern communication services depart from the conventional telephony service in these three essential aspects. Modern communication services can be: • • • multi-media, multi-point, and multi-rate.
Multi-media A multi-media call may communicate audio, data, still images, or full-motion video, or any combination of these media. Each medium has different demands for communication quality, such as: • • • bandwidth requirement, signal latency within the network, and signal fidelity upon delivery by the network.
The information content of each medium may affect the information generated by other media. For example, voice could be transcribed into data via voice recognition, and data commands may control the way voice and video are presented. These interactions most often occur at the communication terminals, but may also occur within the network Multi-point A few examples will be used to contrast point-to-point communications with multi-point communications. Traditional voice calls are predominantly two party calls, requiring a point-to-point connection using only the voice medium. To access pictorial information in a remote database would require a point-to-point connection that sends low bit-rate queries to the database and high bit-rate video from the database. Entertainment video
applications are largely point-to-multi-point connections, requiring one-way communication of full motion video and audio from the program source to the viewers. Video teleconferencing involves connections among many parties, communicating voice, video, as well as data. Offering future services thus requires flexible management of the connection and media requests of a multi-point, multi-media communication Multi-rate A multi-rate service network is one which flexibly allocates transmission capacity to connections. A multimedia network has to support a broad range of bit-rates demanded by connections, not only because there are many communication media, but also because a communication medium may be encoded by algorithms with different bit-rates. For example, audio signals can be encoded with bit-rates ranging from less than 1 kbit/s to hundreds of kbit/s, using different encoding algorithms with a wide range of complexity and quality of audio reproduction. Similarly, full motion video signals may be encoded with bit-rates ranging from less than 1 Mbit/s to hundreds of Mbit/s. Thus a network transporting both video and audio signals may have to integrate traffic with a very broad range of bit-rates [3][5]. A single network for multiple services [edit] Traditional networks Traditionally, the various services mentioned above were carried via separate networks: voice on the telephone network, data on computer networks or local area networks (LANs), video teleconferencing on private corporate networks, and television on broadcast radio or cable networks.These networks are largely engineered for a specific application and are not suited to other applications. For example, the traditional telephone network is too noisy and inefficient for bursty data communication. On the other hand, data networks which store and forward messages using computers have very limited connectivity, usually do not have sufficient bandwidth for digitised voice and video signals, and suffer from unacceptable delays for the real-time signals. Television networks using the radio or the cable medium are largely broadcast networks with minimum switching facilities [2][3]. [edit] Benefits of a single network for multiple services It is desirable to have a single network for providing all these communication services in order to achieve the economy of sharing. This economy motivates the general idea of an integrated services network. Integration avoids the need for many overlaying networks, which complicates network management and reduces flexibility in the introduction and evolution of services. This integration is made possible with advances in broadband technologies and high speed information processing [2][3]. [edit] Fibre optics for broadband networks and MSO While there are multiple network structures capable of supporting broadband services, an ever increasing percentage of broadband and MSO providers are opting for fibre optic network structures to support both present and future bandwidth requirements. CATV (cable television), HDTV (high definition television), VoIP (voice over internet protocol), and broadband internet are some of the most common applications now being supported by fibre optic networks, in some cases directly to the home (FTTh – Fibre To The Home). These types of fibre optic networks incorporate a wide variety of products to support and distribute the signal from the central office to an optic node, and ultimately to the subscriber (end-user). [edit] Broadband Traffic [edit] Types of traffic carried by the network Modern networks have to carry integrated traffic consisting of voice, video and data. The Broadband Integrated Services Digital Network (B-ISDN) satisfies these needs [7]. The types of traffic supported by a broadband network can be classified according to three characteristics [6]:
• • •
Bandwidth is the amount of network capacity required to support a connection. Latency is the amount of delay associated with a connection. Requesting low latency in the Quality of Service (QoS) profile means that the cells need to travel quickly from one point in the network to another. Cell-delay variation (CDV) is the range of delays experienced by each group of associated cells. Low cell-delay variation means a group of cells must travel through the network without getting too far apart from one another.
[edit] Requirements of the different types of traffic The different types of traffic found in a broadband network (with examples) and their respective requirements are summarized in Table 1. Table 1: Network traffic types and their requirements [6] TRAFFIC TYPE Constant Variable Available EXAMPLE Voice, guaranteed circuit emulation Compressed video Data REQUIRED BANDWIDTH Minimal Guaranteed Not guaranteed Low Variable Widely variable Low Variable CELL-DELAY VARIATION LATENCY
Digital switches Digital switches work by connecting two or more digital circuits together, according to a dialed telephone number. Calls are set up between switches using the Signalling System 7 protocol, or one of its variants. In U.S. and military telecommunication, a digital switch is a switch that performs time division switching of digitized signals.[10] This was first done in a few small and little used systems. The first product using a digital switch system was made by Amtelco. Prominent examples include Nortel DMS-100, Lucent 5ESS switch, Siemens EWSD and Ericsson AXE telephone exchange. With few exceptions, most switches built since the 1980s are digital, so for practical purposes this is a distinction without a difference. This article describes digital switches, including algorithms and equipment. A digital exchange (Nortel DMS-100) used by an operator to offer local and long distance services in France. Each switch typically serves 10,000-100,000+ subscribers depending on the geographic area Digital switches encode the speech going on, in 8000 time slices per second. At each time slice, a digital PCM representation of the tone is made. The digits are then sent to the receiving end of the line, where the reverse process occurs, to produce the sound for the receiving phone. In other words, when you use a telephone, you are generally having your voice "encoded" and then reconstructed for the person on the other end. Your voice is delayed in the process by a small fraction of one second — it is not "live", it is reconstructed — delayed only minutely. (See below for more info.) Individual local loop telephone lines are connected to a remote concentrator. In many cases, the concentrator is co-located in the same building as the switch. The interface between remote concentrators and telephone switches has been standardised by ETSI as the V5 protocol. Concentrators are used because most telephones are idle most of the day, hence the traffic from hundreds or thousands of them may be concentrated into only tens or hundreds of shared connections. Some telephone switches do not have concentrators directly connected to them, but rather are used to connect calls between other telephone switches. These complex machines (or a series of them) in a central exchange building are referred to as "carrier-level" switches or tandems. Some telephone exchange buildings in small towns now house only remote or satellite switches, and are homed upon a "parent" switch, usually several kilometres away. The remote switch is dependent on the parent switch for routing and number plan information. Unlike a digital loop carrier, a remote switch can route calls between local phones itself, without using trunks to the parent switch. Telephone switches are usually owned and operated by a telephone service provider or carrier and located in their premises, but sometimes individual businesses or private commercial buildings will house their own switch, called a PBX, or Private branch exchange. The switch's place in the system Telephone switches are a small part of a large network. The majority of work and expense of the phone system is the wiring outside the central office, or the outside plant. In the middle 20th century, each subscriber telephone number required an individual pair of wires from the switch to the subscriber's phone. A typical central office may have tens-of-thousands of pairs of wires that appear on terminal blocks called the main distribution frame or MDF. A component of the MDF is protection: fuses or other devices that protect the switch from lightning, shorts with electric power lines, or other foreign voltages. In a typical telephone company, a large database tracks information about each subscriber pair and the status of each jumper. Before computerization of Bell System records in the 1980s, this information was handwritten in pencil in accounting ledger books. To reduce the expense of outside plant, some companies use "pair gain" devices to provide telephone service to subscribers. These devices are used to provide service where existing copper facilities have been exhausted or by siting in a neighborhood, can reduce the length of copper pairs, enabling digital services such as ISDN or DSL. Pair gain or digital loop carriers (DLCs) are located outside the central office, usually in a large
neighborhood distant from the CO. DLCs are often referred to as Subscriber Loop Carriers (SLCs), after a Lucent proprietary product. DLCs can be configured as universal (UDLCs) or integrated (IDLCs). Universal DLCs have two terminals, a central office terminal (COT) and a remote terminal (RT), that function similarly. Both terminals interface with analog signals, convert to digital signals, and transport to the other side where the reverse is performed. Sometimes, the transport is handled by separate equipment. In an Integrated DLC, the COT is eliminated. Instead, the RT is connected digitally to equipment in the telephone switch. This reduces the total amount of equipment required. Several standards cover DLCs, including Telcordia Technologies Generic Requirements documents GR-8-CORE and GR-303-CORE. Switches are used in both local central offices and in long distance centers. There are two major types in the Public switched telephone network (PSTN):
1. Class 4 telephone switches designed for toll or switch-to-switch connections. 2. Class 5 telephone switches or subscriber switches, which manage connections
from subscriber telephones. Since the 1990s, hybrid Class 4/5 switching systems that serve both functions have become common. Another element of the telephone network is time and timing. Switching, transmission and billing equipment may be slaved to very high accuracy 10 MHz standards which synchronize time events to very close intervals. Time-standards equipment may include Rubidium- or Caesium-based standards and a Global Positioning System receiver. Switch design Long distance switches may use a slower, more efficient switch-allocation algorithm than local central offices, because they have near 100% utilization of their input and output channels. Central offices have more than 90% of their channel capacity unused. While traditionally, telephone switches connected physical circuits (e.g., wire pairs), modern telephone switches use a combination of space- and time-division switching. In other words, each voice channel is represented by a time slot (say 1 or 2) on a physical wire pair (A or B). In order to connect two voice channels (say A1 and B2) together, the telephone switch interchanges the information between A1 and B2. It switches both the time slot and physical connection. To do this, it exchanges data between the time slots and connections 8000 times per second, under control of digital logic that cycles through electronic lists of the current connections. Using both types of switching makes a modern switch far smaller than either a space or time switch could be by itself. The structure of a switch is an odd number of layers of smaller, simpler subswitches. Each layer is interconnected by a web of wires that goes from each subswitch, to a set of the next layer of subswitches. In most designs, a physical (space) switching layer alternates with a time switching layer. The layers are symmetric, because in a telephone system callers can also be callees. A time-division subswitch reads a complete cycle of time slots into a memory, and then writes it out in a different order, also under control of a cyclic computer memory. This causes some delay in the signal. A space-division subswitch switches electrical paths, often using some variant of a nonblocking minimal spanning switch, or a crossover switch.
Digital Transmission Hierachies
There are two hierarchial structures that exist for digital networks: Plesiochronous hierarchies Synchronous hierarchies To further complicate things, North American standards (usually derived from US standards bodies) are different from the International CCITT (now ITU-T) recommendations.
Plesiochronous Hierarchies
In a Plesiochronous hierarchy, the higher level multiplex functions include "bit stuffing" techniques. This allows the input bit streams from I/O channels to use "free-running" clocks. As such, the user's clock rate is propagated (plus a little "Jitter") through the higher level multiplexer. Slip rates requirements between End-User multiplex equipment must still be met, for adequate performance of voice and (particularly) data.
North American Digital hierarchy
The North American Digital hierarchy starts off with a basic Digital Signal level of 64 KBPS (DS0). Thereafter, all facility types are usually referred to as "T x", where "x" is the Digital Signal level within the hierarchy (e.g. T1 refers to the DS1 rate of 1.544 MBPS). Up to the DS3 rate, these signals are usually delivered from the provider on Twisted-Pair or Coaxial cables. North American T1 service providers often refer to the signal interface between the User and the Network as "DS-1" signals. In the case of User to User interfaces, the term "DSX-1" is used to describe those DS1 signals at the "cross-connect" point.
Name ---DS0 DS1 DS1C DS2 DS3 DS4 Rate ----------64 KBPS 1.544 MBPS 3.152 MBPS 6.312 MBPS 44.736 MBPS 274.176 MBPS
International (CCITT) Digital hierarchy
The CCITT Digital hierarchy's basic level is the DS0 rate of 64 KBPS. These signals are usually delivered from the provider on Twisted-Pair or Coaxial cables.
Name ---DS0 E1 Rate ----------64 KBPS 2.048 MBPS
E2 E3 E4
8.448 MBPS 34.368 MBPS 139.264 MBPS
Synchronous Hierarchies
In the later 1980s, synchronous network hierarchies were defined. In Synchronous networks, all multiplex functions operate using clocks derived from a common source.
North American SONET (Synchronous Optical NETwork)
This system is based upon multiples of a fundamental rate of 51.840 MBPS, called STS-1 (Synchronous Transport Signal, Level 1). The facility designators are similar, but indicate the facility type, which is usually Fiber Optic Cable (e.g. OC-3 is an Optical Carrier supporting a STS-3 signal; while OC-12 supports a STS-12 signal, etc). Some typical rates are listed below:
Name Rate ------ --------------STS-1 51.840 MBPS STS-3 155.520 MBPS STS-9 466.560 MBPS STS-12 622.080 MBPS STS-48 2488.320 MBPS STS-192 9953.280 MBPS STS-768 39813.120 MBPS
International SDH (Synchronous Digital Hierarchy)
This system is based upon a fundamental rate of 155.520 MBPS, three times that of the SONET system. This fundamental signal is called STM-1 (Synchronous Transport Module, Level 1). The typical transmission media is defined to be fiber, but the Broadband ISDN specification does define a User-Network Interface (UNI) STM-1 (155.520 MBPS) operating over coaxial cables. Some typical rates within this hierarchy:
Name Rate ------ --------------STM-1 155.520 MBPS STM-3 466.560 MBPS STM-4 622.080 MBPS STM-16 2488.320 MBPS STM-64 9953.280 MBPS STM-256 39813.120 MBPS
Synchronous Digital Hierarchy (SDH)
Introduction The Transmission System is traditionally seen as the link between main
WAN switching centres. These Transmission Systems consist of large bandwidth highways that form the backbone to the network. They typically serve many customers each with their own requirements so the systems have to be reliable, resilient and flexible. Rather than have two wires for every voice or data conversation, Time Division Multiplexing is used. ITU-T G.704 defines 32 channels of 64Kb/s to form 2.048Mb/s where channel 0 is used for framing. You will often see the standard G.703 mentioned with G.704, this is because G.703 defines the unframed physical interface coaxial (75 ohm) or RJ48 (120 ohm) used for the E1/T1 connection at the client premises. Channel 0 is for timing used to synchronise the multiplexers at each end of the link. Channels 1 to 15 and 17 to 31 are for voice or data whilst channel 16 is used for Common Channel Signalling (CCS) or Channel Associated Signalling (CAS). Every 3.91 microseconds 8 bits from one channel is sent down the line followed by 8 bits from the next channel during the next 3.91 microseconds and so on in a round robin fashion throughout all the channels, thus 32 channels are used once every 125 microseconds. The connection at the end is either a 75 ohm coax, 120 ohm coax or a 150 ohm UTP/STP.
Plesiochronous Digital Hierarchy (PDH)
As bandwidth demand grew the technology called Plesiochronous Digital Hierarchy (PDH) was developed by ITU-T G.702, whereby the basic primary multiplexer 2.048Mb/s trunks were joined together by adding bits (bit stuffing) which synchronised the trunks at each level of the PDH. 2.048Mb/s was called E1 and the hierarchy is based on multiples of 4 E1s. • • • • E2, 4 x E1 - 8Mb/s E3, 4 x E2 - 34Mb/s E4, 4 x E3 - 140Mb/s E5, 4 x E4 - 565Mb/s
The E3 tributaries are faster than the E2 tributaries, E2 tributaries are faster than the E1 tributaries and so on. These need to be synchronised with other tributaries, so extra bits are added called Justification bits. These tell the multiplexers which bits are data and which are spare. Multiplexers on the same level of the hierarchy remove the spare bits and are synchronised with each other at that level only. Multiplexers on one level operate on a different timing from multiplxers on another level. For instance, the timing between Primary Rate Muxes (combines 30 x 64Kb/s channels into 2.048Mb/s E1) will be different from the timing between 8Mbit muxes (combines up to 4 x 2Mb/s into 8Mb/s).
Inserting and dropping out traffic from different customers can only happen at the level at which the customer is receiving the traffic. This means that if a 140Mb/s fibre is near a particular site and a new customer requires a 2Mb/s link, then a whole set of demultiplexers are required to do this.
Synchronous Digital Hierarchy (SDH)
Management is very inflexible in PDH, so SDH was developed. Synchronous Digital Hierarchy (SDH) originates from Synchronous Optical Network (SONET) in the US. It includes capabilities for bandwidth on demand and is also made up of multiples of E1. STM-1 (155Mb/s) is 63 x E1, STM-4 (622Mb/s) is 4 x STM-1 and STM-16 (2.5Gb/s) is 4 x STM-4. The benefits of SDH are: • • • Different interfaces or different bandwidths can connect (G708, G781). Network topologies are more flexible. There is flexibility for growth.
• •
The optical interface is standard (G957). Network Management is easier to perform (G774 and G784).
Existing PDH can interface into SDH. There are three G transmission series recommendations that are very important: • • • G.707 - SDH Bit Rates G.708 - The SDH Network Node Interface. G.709 - Synchronous Multiplexing structure.
With the exception of 8Mb/s, different PDH outputs are 'mapped' into Containers (C) and then into fixed size Virtual Containers (VC). When the VC is aligned in the Tributary Unit (TU) a Pointer is added which indicates the phase of the particular VC. TU's are then grouped, via Time Division Multiplexing (TDM), into Tributary Unit Groups (TUG). The TUGs are collated into Administrative Units (AU) via more VCs where more pointers are added (these being fixed relative to the frame). The VCs and the pointers are incorporated into the section overhead of the Synchronous Transport Module (STM). One AU forms an STM-1, 4 AUs form an STM-4. You can also get STM-16 and STM-64. Let us follow a 2Mb/s pipe through the hierarchy. The 2Mb/s PDH first enters a container C12 which compensates for the varying speeds via the use of stuff bits (R). Stuff opportunities are identified by S1 and S2 and these are controlled by the control bits C1 and C2 respectively. If the C bits are are 0s then the corresponding S bits contain data and if the C bits are 1s then the S bits are not defined. In the diagram below, O represents Overhead channel bits and I represents Information bits.
To create the VC12 a Path Overhead (POH) is added. The POH uses Bit Interleaved Parity (BIP) to monitor errors. In addition, there are fault indicators, Far End Block Error (FEBE), Remote Fail Indicator (RFI) and Far End Receive Failure (FERF). The Signal Label is normally set at 2 to indicate asynchronous data. A pointer is added to the VC12 which defines the phase alignment of the VC12 and this changes during transmission. Phase variation can be due to Jitter (from regeneration and multiplexing equipment) and Wander (temperature differences within the transmission media). VC12s created by different multiplexers may not be synchronous so the TU adds a pointer at a fixed position within the TU. The value of the pointer indicates the start of the VC12. If the phase of the VC12 changes then the value of the pointer changes such that if data is running faster than the TU then the pointer value is increased and if the data speed is slower then the pointer value is decreased. This difference in speed can be up to one byte per frame in SDH. The following diagram illustrates three TU12s entering a TUG2 at three different times with the VC12 pointers indicating where the POH is for each:
The TU12 is multiplexed into a TUG 2 along with 2 other TU12s. This is achieved by interleving the bytes of each TU12 in turn. Next, seven TUG 2s are byte interleaved into a TUG 3 and then three TUG 3s can be byte interleaved to form the VC4 (see the SDH diagram above). You can see that 3 x 7 x 3 = 63 2Mb/s circuits can be contained in VC 4.
Network synchronization Network synchronization are used to maintain and synchronize timing across a computer network. Network administrators use network synchronization products to manage time-sensitive services such as streaming audio and video, data transfer and encryption, client-server time accuracy, Web conferencing, backup and redundancy, and database communications. Network synchronization products are also used with client computers, printers, and phones with file, printer, and application servers. A network clock is a network synchronization product that use the network time protocol (NTP) standardized in RFC 1305. To synchronize timing across a local area network (LAN), network clocks match NTP with an external source such as a global positioning satellite (GPS) or lowfrequency (LF) time signaling. Additional interfaces for network synchronization include simple network time protocol, daytime protocol (RFC 867), and user datagram protocol/Internet protocol (UDP/IP). Network timers or master clocks are network synchronization products that use a microprocessor and internal network oscillator to synchronize applications that require analog clocking. In simplest terms, oscillators are circuits that use amplification and feedback to generate radio frequency (RF) outputs. Because network timers use microprocessors, they can be manufactured as programmable erasable read-only memory (PROM) or electrically erasable read-only memory (EEPROM). This enables chip designers to program devices that can produce an RF signal after a given timing period. Typically, this timing period begins or ends when a switch actuates the program. Network synchronization products can be categorized into hardware components and software applications. The proper selection of software and hardware depends on considerations such as time sources, synchronization frequency, time models, network models, and installation specifics. Network synchronization products also include software applications that are designed to display and monitor the times on servers, client computers, and domain controllers. These applications provide management tools to monitor synchronization activities on a network, such as data and file transfers, application-to-database communications.
