While the PSTN varies in its implementation from country to country, a number of common denominators exist. The PSTN is a collection of digital switching nodes that are interconnected by trunks. The network topology is usually a hierarchical structure, but it often incorporates some degree of mesh topology. The topology provides network access to residential and business subscribers for voice and data services. VoIP began another evolution of the PSTN architecture. The PSTN is a large infrastructure that will likely take some time to completely migrate to the next generation of technologies; but this migration process is underway.

• Soft switches

• H.323

• Session Initiation Protocol (SIP)

Each of these VoIP architectures use VoIP-PSTN gateways to provide some means of communication between the traditional PSTN networks and VoIP networks. These gateways provide access points for interconnecting the two networks, thereby creating a migration path from PSTN-based phone service to VoIP phone service. The core network interface connections for VoIP into the PSTN are the trunk facilities that carry the voice channels and the signaling links that carry SS7 signaling. PRI is also commonly used for business to network access. Figure 5-13 shows the interconnection of VoIP architectures to the PSTN using signaling gateways and trunking gateways. Chapter 14, “SS7 in the Converged World,” discusses these VoIP technologies in more detail.

The PSTN existed long before SS7. The network’s general structure was already in place, and it represented a substantial investment. The performance requirements mandated by the 800 portability act of 1993 was one of the primary drivers for the initial deployment of SS7 by ILECs in the United States. IXCs embraced SS7 early to cut down on post-dial delay which translated into significant savings on access/egress charges. Federal regulation, cost savings, and the opportunity to provide new revenue generating services created a need to deploy SS7 into the existing PSTN.

SS7 was designed to integrate easily into the existing PSTN, to preserve the investment and provide minimal disruption to the network. During SS7′s initial deployment, additional hardware was added and digital switches received software upgrades to add SS7 capability to existing PSTN nodes. In the SS7 network, a digital switch with SS7 capabilities is referred to as a Service Switching Point (SSP). When looking at the SS7 network topologies in later chapters, it is important to realize that the SSP is not a new node in the network.

Instead, it describes an existing switching node, to which SS7 capabilities have been added. Similarly, SS7 did not introduce new facilities for signaling links, but used timeslots on existing trunk facilities. PSTN diagrams containing End Offices and tandems connected by trunks represent the same physical facilities as those of SS7 diagrams that show SSP nodes with interconnecting links. The introduction of SS7 added new nodes, such as the STP and SCP; however, all of the switching nodes and facilities that existed before SS7 was introduced are still in place.

The following section focuses on these areas of the CO:

• The Main Distribution Frame

• The Digital Switch

• The Switching Matrix

• Call Processing

Main Distribution Frame

Incoming lines and trunks are terminated on the Main Distribution Frame (MDF). The MDF provides a junction point where the external facilities connect to the equipment within the CO. Jumpers make the connections between the external facilities and the CO equipment, thereby allowing connections to be changed easily. Line connections from the MDF to the digital switching equipment terminate on line cards that are designed to interface with the particular type of line being connected—such as POTS, ISDN BRI, and Electronic Key Telephone Set (EKTS) phone lines. For analog lines, this is normally the point at which voice encoding takes place. Trunk connections from the MDF are terminated on trunk interface cards, providing the necessary functions for message framing, transmission, and reception.

The Digital Switch

The digital switch provides a software-controlled matrix of interconnections between phone subscribers. A handful of telecommunications vendors produce the digital switches that comprise the majority of the modern PSTN; Nortel, Lucent, Siemens, Alcatel, and Ericsson hold the leading market share. While the digital switch’s basic functionality is common across vendors, the actual implementation is vendor dependent. This section provides a general perspective on the functions of the digital switch that are common across different implementations.

All digital switches are designed with some degree of distributed processing. A typical architecture includes a central processing unit that controls peripheral processors interfacing with the voice channels. Redundancy is always employed in the design to provide the high reliability that is expected in the telephony network. For example, the failure of one central processing unit results in the activation of an alternate processing unit.

The line and trunk interface cards, mentioned previously, represent the point of entry into the digital switch. These cards typically reside in peripheral equipment that is ultimately controlled by the central processor. Within the digital switch, all voice streams are digitized data. Some voice streams, such as those from ISDN facilities and digital trunks, enter the switch as digital data. Other voice streams, such as the analog phone, enter as analog data but undergo digital conversion at their point of entry. Analog lines interface with line cards that contain codecs, which perform the PCM processing to provide digital data to the switch and analog data to the line. Using the distributed processing architecture, many functions related to the individual voice channels are delegated to the peripheral interface equipment. This relieves the central processor of CPU intensive, low-level processing functions, such as scanning for on/off hooks on each individual line to determine when a subscriber wants to place a call.

The central processing unit monitors information from peripheral processors on call events—such as origination, digit collection, answer, and termination—and orchestrates the actual call setup and release. Information from these events is also used to perform call accounting, billing, and statistical information such as Operational Measurements (OM).

Although the main purpose of the digital switch is to perform call processing, much of its functionality is dedicated to maintenance, diagnostics, and fault recovery to ensure reliability.

Switching Matrix

A modern digital switch can process many voice channels. The actual number of channels it processes varies with the switch vendor and particular model of switch, but they often process tens of thousands of voice channels in a single switch. A number of switches have capacities of over 100,000 connections.

The switch is responsible for many tasks, but one of its primary functions is connecting voice channels to create a bi-directional conversation path between two phone subscribers. All digital switches incorporate some form of switching matrix to allow the connection of voice channels to other voice channels. Once a circuit is set up between the two subscribers, the connection remains for the duration of the call. This method of setting up call connections is commonly known as circuit switching.

There are various methods of synchronizing nodes. One method involves a single master clock source, from which other nodes derive timing in a master/slave arrangement. Another method uses a plesiochronous arrangement, where each node contains an independent clock whose accuracy is so great that it remains independently synchronized with other nodes. You can also use a combination of the two methods by using highly accurate clocks as a Primary Reference Source (PRS) in a number of nodes, providing timing to subtending nodes in the network.

The clocks’ accuracy is rated in terms of stratum levels. Stratums 1 through 4 denote timing sources in order of descending accuracy. A stratum 1 clock provides the most accurate clock source with a free-running accuracy of ±1 x 10 ^-11, meaning only one error can occur in 10¹¹ parts. A stratum 4 clock provides an accuracy of ±32 x 10^-6.

Since the deployment of Global Positioning System (GPS) satellites, each with a number of atomic clocks on-board, GPS clocks have become the preferred method of establishing a clock reference signal. Having a GPS clock receiver at each node that receives a stratum 1-quality timing signal from the GPS satellite flattens the distributed timing hierarchy. If the GPS receiver loses the satellite signal, the receiver typically runs free at stratum 2 or less. By using a flattened hierarchy based on GPS receivers, you remove the need to distribute the clock signal and provide a highly accurate reference source for each node. Figure 5-9 shows an example that uses a stratum 1 clock at a digital switching office to distribute timing to subtending nodes, and also shows an example that uses a GPS satellite clock receiver at each office.

Lines

Lines are used to connect the subscriber to the CO, providing the subscriber access into the PSTN. The following sections describe the facilities used for lines, and the access signaling between the subscriber and the CO.

• The Local Loop

• Analog Line Signaling

• Dialing

• Ringing and Answer

• Voice Encoding

• ISDN BRI

The Local Loop

The local loop consists of a pair of copper wires extending from the CO to a residence or business that connects to the phone, fax, modem, or other telephony device. The wire pair consists of a tip wire and a ring wire. The terms tip and ring are vestiges of the manual switchboards that were used a number of years ago; they refer to the tip and ring of the actual switchboard plug operators used to connect calls. The local loop allows a subscriber to access the PSTN through its connection to the CO. The local loop terminates on the Main Distribution Frame (MDF) at the CO, or on a remote line concentrator.

Remote line concentrators, also referred to as Subscriber Line Multiplexers or Subscriber Line Concentrators, extend the line interface from the CO toward the subscribers, thereby reducing the amount of wire pairs back to the CO and converting the signal from analog to digital closer to the subscriber access point. In some cases, remote switching centers are used instead of remote concentrators.

Remote switching centers provide local switching between subtending lines without using the resources of the CO. Remotes, as they are often generically referred to, are typically used for subscribers who are located far away from the CO. While terminating the physical loop, remotes transport the digitized voice stream back to the CO over a trunk circuit, in digital form.

Analog Line Signaling

Currently, most phone lines are analog phone lines. They are referred to as analog lines because they use an analog signal over the local loop, between the phone and the CO. The analog signal carries two components that comprise the communication between the phone and the CO: the voice component, and the signaling component.

The signaling that takes place between the analog phone and the CO is called in-band signaling. In-band signaling is primitive when compared to the out-of-band signaling used in access methods such as ISDN; see the “ISDN BRI” section in this chapter for more information. DC current from the CO powers the local loop between the phone and the CO. The voltage levels vary between different countries, but an on-hook voltage of –48 to –54 volts is common in North America and a number of other geographic regions, including the United Kingdom.

Dialing

When a subscriber dials a number, the number is signaled to the CO as either a series of pulses based on the number dialed, or by Dual Tone Multi-Frequency (DTMF) signals. The DTMF signal is a combination of two tones that are generated at different frequencies. A total of seven frequencies are combined to provide unique DTMF signals for the 12 keys (three columns by four rows) on the standard phone keypad. Usually, the dialing plan of the CO determines when all digits have been collected.

Ringing and Answer

To notify the called party of an incoming call, the CO sends AC ringing voltage over the local loop to the terminating line. The incoming voltage activates the ringing circuit within the phone to generate an audible ring signal. The CO also sends an audible ring-back tone over the originating local loop to indicate that the call is proceeding and the destination phone is ringing. When the destination phone is taken off-hook, the CO detects the change in loop current and stops generating the ringing voltage. This procedure is commonly referred to as ring trip. The off-hook signals the CO that the call has been answered; the conversation path is then completed between the two parties and other actions, such as billing, can be initiated, if necessary.

Voice Encoding

An analog voice signal must be encoded into digital information for transmission over the digital switching network. The conversion is completed using a codec (coder/decoder), which converts between analog and digital data. The ITU G.711 standard specifies the Pulse Coded Modulation (PCM) method used throughout most of the PSTN. An analog-to-digital converter samples the analog voice 8000 times per second and then assigns a quantization value based on 256 decision levels. The quantization value is then encoded into a binary number to represent the individual data point of the sample.

Two variations of encoding schemes are used for the actual quantization values: A-law and m-Law encoding. North America uses m-Law encoding, and European countries use A-law encoding. When voice is transmitted from the digital switch over the analog loop, the digital voice data is decoded and converted back into an analog signal before transmitting over the loop.

The emergence of voice over IP (VoIP) has prompted the use of other voice-encoding standards, such as ITU G.723, G.726, and ITU G.729. These encoding methods use algorithms that produce more efficient and compressed data, making them more suitable for use in packet networks. Each encoding method involves trade-offs between bandwidth, processing power required for the encoding/decoding function, and voice quality. For example, G.711 encoding/decoding requires little processing and produces high quality speech, but consumes more bandwidth. In contrast, G.723.1 consumes little bandwidth, but requires more processing power and results in lower quality speech.

ISDN BRI

Although Integrated Services Digital Network (ISDN) deployment began in the 1980s, it has been a relatively slow-moving technology in terms of number of installations. ISDN moves the point of digital encoding to the customer premises. Combining ISDN on the access portion of the network with digital trunks on the core network provides total end-to-end digital connectivity. ISDN also provides out-of-band signaling over the local loop. ISDN access signaling coupled with SS7 signaling in the core network achieves end-to-end out-of-band signaling. ISDN access signaling is designed to complement SS7 signaling in the core network.

There are two ISDN interface types: Basic Rate Interface (BRI) for lines, and Primary Rate Interface (PRI) for trunks. BRI multiplexes two bearer (2B) channels and one signaling (D) channel over the local loop between the subscriber and the CO; this is commonly referred to as 2B+D. The two B channels each operate at 64 kb/s and can be used for voice or data communication. The D channel operates at 16 kb/s and is used for call control signaling for the two B channels. The D channel can also be used for very low speed data transmission. Within the context of ISDN reference points, the local loop is referred to as the U-loop. It uses different electrical characteristics than those of an analog loop.

Voice quantization is performed within the ISDN phone (or a Terminal Adapter, if an analog phone is used) and sent to a local bus: the S/T bus. The S/T bus is a four-wire bus that connects local ISDN devices at the customer premises to a Network Termination 1 (NT1) device. The NT1 provides the interface between the Customer Premises Equipment (CPE) and the U-loop.

The NT1 provides the proper termination for the local S/T bus to individual devices and multiplexes the digital information from the devices into the 2B+D format for transmission over the U-loop. Figure 5-6 illustrates the BRI interface to the CO. Only ISDN devices connect directly to the S/T bus. The PC uses an ISDN Terminal Adapter (TA) card to provide the proper interface to the bus.

PSTN Hierarchy in the United States

In the United States, the PSTN is generally divided into three categories:

• Local Exchange Networks

• InterExchange Networks

• International Networks

Local Exchange Carriers (LECs) operate Local Exchange networks, while InterExchange Carriers (IXCs) operate InterExchange and International networks.

The PSTN hierarchy in the United States is also influenced by market deregulation, which has allowed service providers to compete for business and by the divestiture of Bell.

Local Exchange Network

The Local Exchange network consists of the digital switching nodes (EOs) that provide network access to the subscriber. The Local Exchange terminates both lines and trunks, providing the subscriber access to the PSTN.

A Tandem Office often connects End Offices within a local area, but they can also be connected directly. In the United States, Tandem Offices are usually designated as either Local Tandem (LT) or Access Tandem (AT). The primary purpose of a Local Tandem is to provide interconnection between End Offices in a localized geographic region. An Access Tandem provides interconnection between local End Offices and serves as a primary point of access for IXCs. Trunks are the facilities that connect all of the offices, thereby transporting inter-nodal traffic.

InterExchange Network

The InterExchange network is comprised of digital switching nodes that provide the connection between Local Exchange networks. Because they are points of high traffic aggregation and they cover larger geographical distances, high-speed transports are typically used between transit switches. In the deregulated U.S. market, transit switches are usually referred to as carrier switches. In the U.S., IXCs access the Local Exchange network at designated points, referred to as a Point of Presence (POP). POPs can be connections at the Access Tandem, or direct connections to the End Office.

International Network

The International network consists of digital switching nodes, which are located in each country and act as international gateways to destinations outside of their respective countries. These gateways adhere to the ITU international standards to ensure interoperability between national networks. The international switch also performs the protocol conversions between national and international signaling. The gateway also performs PCM conversions between A-law and m-law to produce compatible speech encoding between networks, when necessary.

Service Providers

Deregulation policies in the United States have allowed network operators to compete for business, first in the long-distance market (InterExchange and International) beginning in the mid 1980s, and later in the local market in the mid 1990s. As previously mentioned, LECs operate Local Exchange networks, while IXCs operate the long-distance networks. Figure 5-2 shows a typical arrangement of LEC-owned EOs and tandems interconnected to IXC-owned transit switches. The IXC switches provide long-haul transport between Local Exchange networks, and international connections through International gateway switches.

Over the last several years, the terms ILEC and CLEC have emerged within the Local Exchange market to differentiate between the Incumbent LECs (ILECS) and the Competitive LECs (CLECS). ILECs are the incumbents, who own existing access lines to residences and corporate facilities; CLECs are new entrants into the Local Exchange market. Most of the ILECs in the United States came about with the divestiture of AT&T into the seven Regional Bell Operating Companies (RBOC). The remainder belonged to Independent Operating Companies (IOCs). Most of these post-divestiture companies have been significantly transformed today by mergers and acquisitions in the competitive market. New companies have experienced difficulty entering into the Local Exchange market, which is dominated by ILECs. The ILECs own the wire to the subscriber’s home, often called the “last mile” wiring. Last mile wiring is expensive to install and gives the ILECs tremendous market leverage. The long-distance market has been easier for new entrants because it does not require an investment in last mile wiring.

Pre-Divestiture Bell System Hierarchy

Vestiges of terminology relating to network topology remain in use today from the North American Bell System’s hierarchy, as it existed prior to divestiture in 1984. Telephone switching offices are often still referred to by class. For example, an EO is commonly called a class 5 office, and an AT is called a class 4 office. Before divestiture, each layer of the network hierarchy was assigned a class number.

Prior to divestiture, offices were categorized by class number, with class 1 being the highest office category and class 5 being the lowest (nearest to subscriber access). Aggregation of transit phone traffic moved from the class 5 office up through the class 1 office. Each class of traffic aggregation points contained a smaller number of offices.

Depending on geographical region, PSTN nodes are sometimes referred to by different names. The three node types we discuss in this chapter include:

• End Office (EO)— Also called a Local Exchange. The End Office provides network access for the subscriber. It is located at the bottom of the network hierarchy.

• Tandem— Connects EOs together, providing an aggregation point for traffic between them. In some cases, the Tandem node provides the EO access to the next hierarchical level of the network.

• Transit— Provides an interface to another hierarchical network level. Transit switches are generally used to aggregate traffic that is carried across long geographical distances.

There are two primary methods of connecting switching nodes. The first approach is a mesh topology, in which all nodes are interconnected. This approach does not scale well when you must connect a large number of nodes. You must connect each new node to every existing node. This approach does have its merits, however; it simplifies routing traffic between nodes and avoids bottlenecks by involving only those switches that are in direct communication with each other. The second approach is a hierarchical tree in which nodes are aggregated as the hierarchy traverses from the subscriber access points to the top of the tree. PSTN networks use a combination of these two methods, which are largely driven by cost and the traffic patterns between exchanges.

Posted in Chapter 5 Tagged: C7, network, signaling, SS7, telecommunication ]]> http://ss7bible.wordpress.com/2009/12/12/network-topology/feed/ 0 The Author http://ss7bible.wordpress.com/2009/11/28/the-public-switched-telephone-network-pstn/ http://ss7bible.wordpress.com/2009/11/28/the-public-switched-telephone-network-pstn/#comments Sat, 28 Nov 2009 16:05:39 +0000 The Author http://ss7bible.wordpress.com/?p=48 ]]> The term Public Switched Telephone Network (PSTN) describes the various equipment and interconnecting facilities that provide phone service to the public. The network continues to evolve with the introduction of new technologies. The PSTN began in the United States in 1878 with a manual mechanical switchboard that connected different parties and allowed them to carry on a conversation. Today, the PSTN is a network of computers and other electronic equipment that converts speech into digital data and provides a multitude of sophisticated phone features, data services, and mobile wireless access.

At the core of the PSTN are digital switches. The term “switch” describes the ability to cross-connect a phone line with many other phone lines and switching from one connection to another. The PSTN is well known for providing reliable communications to its subscribers. The phrase “five nines reliability,” representing network availability of 99.999 percent for PSTN equipment, has become ubiquitous within the telecommunications industry.

This chapter provides a fundamental view of how the PSTN works, particularly in the areas of signaling and digital switching. SS7 provides control signaling for the PSTN, so you should understand the PSTN infrastructure to fully appreciate how it affects signaling and switching. This chapter is divided into the following sections:

• Network Topology

• PSTN Hierarchy

• Access and Transmission Facilities

• Network Timing

• The Central Office

• Integration of SS7 into the PSTN

• Evolving the PSTN to the Next Generation

We conclude with a summary of the PTSN infrastructure and its continuing evolution.

• Service Switching Point (SSP)

• Service Control Point (SCP)

• Signal Transfer Point (STP)

SSPs provide the SS7 functionality of a switch. STPs may be either standalone or integrated STPs (SSP and STP) and are used to transfer signaling messages. SCPs interface the SS7 network to query telecommunication databases, allowing service logic and additional routing information to be obtained to execute services.

SPs are connected to each other using signaling links. Signaling links are logically grouped into a linkset. Links may be referenced as A through F links, depending on where they are in the network.

Signaling is transferred using the packet-switching facilities afforded by SS7. These packets are called signal units (SUs). The Message Transfer Part (MTP) and the Signaling Connection Control Part (SCCP) provide the transfer protocols. MTP is used to reliably transport messages between nodes, and SCCP is used for noncircuit-related signaling (typically, transactions with SCPs). The ISDN User Part (ISUP) is used to set up and tear down both ordinary (analog subscriber) and ISDN calls. The Transaction Capabilities Application Part (TCAP) allows applications to communicate with each other using agreed-upon data components and manages transactions.