PSTN (Public Switched Telephone Network) The Public Switched Telephone Network (PSTN) is the network of the global public circuit-switched telephone networks, in much the same way that the Internet is the network of the world's public IPbased packet-switched networks. Originally a network of fixed-line analog telephone systems, the PSTN is now almost entirely digital and includes mobile as well as fixed telephones. It is sometimes referred to as the Plain Old Telephone System (POTS). Figure 1 PSTN Network Architecture PSTN is the aggregate of the world's circuit-switched telephone networks that are operated by national, regional, or local telephony operators, providing infrastructure and services for public telecommunication. The PSTN consists of telephone lines, fiber optic cables, microwave transmission links, cellular networks, communications satellites, and undersea telephone cables, all interconnected by switching centers, thus allowing most telephones to communicate with each other. Originally a network of fixed-line analog telephone systems, the PSTN is now almost entirely digital in its core network and includes mobile and other networks, as well as fixed telephones. Plain Old Telephone Service, which refers to the standard telephone service that most homes use. In contrast, telephone services based on high-speed, digital communications lines, such as ISDN and FDDI, are not POTS. The main distinctions between POTS and non-POTS services are speed and bandwidth. POTS is generally restricted to about 52 Kbps (52,000 bitsper second). The technical operation of the PSTN adheres to the standards created by the ITU-T. These standards allow different networks in different countries to interconnect seamlessly. The E.163 and E.164 standards provide a single global address space for telephone numbers. The combination of the interconnected networks and the single numbering plan allow telephones around the world to dial each other. PSTN (Public Switched Telephone Network) 1. History The first telephones had no network but were in private use, wired together in pairs. Users who wanted to talk to different people had as many telephones as necessary for the purpose. A user who wished to speak whistled loudly into the transmitter until the other party heard. However, a bell was added soon for signaling, so an attendant no longer need wait for the whistle, and then a switch hook. Later telephones took advantage of the exchange principle already employed in telegraph networks. Each telephone was wired to a local telephone exchange, and the exchanges were wired together with trunks. Networks were connected in a hierarchical manner until they spanned cities, countries, continents and oceans. This was the beginning of the PSTN, though the term was not used for many decades. Automation introduced pulse dialing between the phone and the exchange, and then among exchanges, followed by more sophisticated address signaling including multi-frequency, culminating in the SS7 network that connected most exchanges by the end of the 20th century. The growth of the PSTN meant that teletraffic engineering techniques needed to be deployed to deliver quality of service (QoS) guarantees for the users. The work of A. K. Erlang established the mathematical foundations of methods required to determine the capacity requirements and configuration of equipment and the number of personnel required to deliver a specific level of service. In the 1970s the telecommunications industry began implementing packet switched network data services using the X.25 protocol transported over much of the end-to-end equipment as was already in use in the PSTN. In the 1980s the industry began planning for digital services assuming they would follow much the same pattern as voice services, and conceived a vision of end-to-end circuit switched services, known as the Broadband Integrated Services Digital Network (B-ISDN). The B-ISDN vision has been overtaken by the disruptive technology of the Internet. Beginning in the 1960s, voice calls began to be digitized and manual switching was replaced by automated electronic switching. Digital voice signals can share the same wire with many other phone calls. The advent of fiber-optic cables now allows thousands of calls to share the same line. But fiberoptic and other high-bandwidth cables haven't changed the basic nature of circuit switching, which still requires a connection -- or circuit -- to remain open for the length of the phone call. At the turn of the 21st century, the oldest parts of the telephone network still use analog technology for the last mile loop to the end user. Digital services have been increasingly rolled out to end users using services such as DSL, ISDN, FTTx, cable modem, and online fax systems. for example. In other words. The first three digits are the area code or national destination code (NDC). which is tied to your specific address and phone lines. Routing calls requires multiple switching offices. The first company to be incorporated to provide PSTN services was the Bell Telephone Company in the United States. you have to dial two separate numbers. Within a company or larger organization.the call doesn't have to be routed onto another switch. when you make a call to another user in your same exchange -. which represents the smallest amount of circuits that can be bundled on the same switch. 2. your country's exit code (or international access code) and the corresponding country code of the place you're calling. we have 10-digit phone numbers. . In the India. such as a large private branch exchange (PBX). To signal such a switch. There are also private networks run by large companies which are linked to the PSTN only through limited gateways. usually for military purposes. Operators The task of building the networks and selling services to customers fell to the network operators.maybe a neighbor around the corner -.PSTN (Public Switched Telephone Network) Several large private telephone networks are not linked to the PSTN. The last four digits of the phone number represent the subscriber number. To make an international call requires further instructions. Extensions from the main phone number are routed through something called a private branch exchange (PBX) that operates on the premises. which helps route the call to the right regional switching station. The next three digits are the exchange. The call needs to be routed through your long-distance phone carrier to another country's long-distance phone carrier. The phone number itself is a coded map for routing the call. each employee or department might have its own extension. PSTN nodes are sometimes referred to by different names. Transit switches are generally used to aggregate traffic that is carried across long geographical distances. The Central Office 3.1.PSTN (Public Switched Telephone Network) 3. monopolistic markets are generally an interconnection of switches owned by the same operator. While topologies in competitive markets represent an interconnection of networks owned by different service providers. providing an aggregation point for traffic between them. Working This chapter provides a fundamental view of how the PSTN works. Tandem: Connects EOs together. so you should understand the PSTN infrastructure to fully appreciate how it affects signaling and switching. You must connect each new node to every existing node.3. The first approach is a mesh topology.2. 3. It is located at the bottom of the network hierarchy. The End Office provides network access for the subscriber. In some cases. SS7 provides control signaling for the PSTN. PSTN Hierarchy 3. Network Timing 3. the Tandem node provides the EO access to the next hierarchical level of the network. This approach does have its merits. This approach does not scale well when you must connect a large number of nodes.6. Regulatory policies play a major role in exactly how voice network topologies are defined in each country. Access and Transmission Facilities 3. in which all nodes are interconnected.5. but general similarities exist.7. Integration of SS7 into the PSTN 3. There are two primary methods of connecting switching nodes. Depending on geographical region. The second approach is a hierarchical .4. Evolving the PSTN to the Next Generation 4. it simplifies routing traffic between nodes and avoids bottlenecks by involving only those switches that are in direct communication with each other. Transit: Provides an interface to another hierarchical network level. particularly in the areas of signaling and digital switching. Network Topology 3. The three node types we discuss in this chapter include: End Office (EO): Also called a Local Exchange. however. Network Topology The topology of a network describes the various network nodes and how they interconnect. While Class 2 and Class 3 offices are seldom used in today's system. The original telephone system used five numbered levels. Therefore. Transit switches provide further aggregation points for connecting multiple tandems between different networks. connects to multiple Class 5 . 5. in turn. each top-level Class 1 office usually connects to multiple Class 4 offices. most follow some variation of this basic pattern. While actual network topologies vary. as shown on the Original Telecommunications Hierarchy Diagram. skipping the old Classes 2 and 3. Figure 2 Generic PSTN Hierarchies Figure shows a generic PSTN hierarchy. PSTN Hierarchy The public-switched telephone network (PSTN) is organized as a multilevel hierarchy. the original numbering system has survived. Each Class 4 office.PSTN (Public Switched Telephone Network) tree in which nodes are aggregated as the hierarchy traverses from the subscriber access points to the top of the tree. in which End Offices are connected locally and through tandem switches. which are largely driven by cost and the traffic patterns between exchanges. PSTN networks use a combination of these two methods. that office makes the connection directly. . Hierarchy Telecommunications Hierarchy Class 5 Central Office: The Local Exchange The Class 5 CO is also called the end office or local office. If a subscriber places a call to another subscriber connected to the same Class 5 office. However. The Class 5 offices. Offices higher in this hierarchy have only lower level COs as their subscribers. When you pick up your telephone at home. as well as its "parent" Class 4 office one level up in the hierarchy. or end offices. as shown on the Call within the Same Exchange Diagram. It is the local workhorse for the telephone and data communications traffic in one local exchange.PSTN (Public Switched Telephone Network) offices. connect to individual subscribers.500 Class 5 local exchanges in the United States. The Class 5 office is the only office that connects to individual or business subscribers. Figure 3 Original Telecommunications Figure 4 Current 5. each Class 5 CO also connects to other nearby Class 5 offices.1. as shown on the Current Telecommunications Hierarchy Diagram. you receive dial tone from a Class 5 CO. There are currently about 1. as shown on the Tandem Switching Diagram. as shown on the Calling Handoff Diagram. the calling CO passes the call directly to the destination CO. the caller's Class 5 CO passes the call up the hierarchy to its parent Class 4 office. or if that connection is too busy.PSTN (Public Switched Telephone Network) Figure 5 Call Within the Same Exchange If the caller's Class 5 CO is directly connected to the destination Class 5 CO. Figure 6 Calling Handoff However. if the destination CO is not directly connected to the calling CO. . which completes the call to the destination subscriber. which often use a Private Branch eXchange (PBX)—or in the case of some very large businesses. while trunks are used to interconnect PSTN switches. Trunks also provide access to corporate phone environments. their own digital switch. Individual telephone lines connect subscribers to the Central Office (CO) by wire pairs. Access and Transmission Facilities Connections to PSTN switches can be divided into two basic categories: lines and trunks. Figure 8 End Office Facility Interfaces . Figure illustrates a number of common interfaces to the Central Office.PSTN (Public Switched Telephone Network) Figure 7 Tandem Switching 6. remote switching centers are used instead of remote concentrators. or other telephony device. are typically used for subscribers who are located far away from the CO. 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. The analog signal carries two components that comprise the communication between the phone and the CO: the voice component. fax. Remotes. remotes transport the digitized voice stream back to the CO over a trunk circuit.1. Lines Lines are used to connect the subscriber to the CO.1.1. and the signaling component. The signaling that takes place between the analog phone and the CO is called in-band signaling. The terms tip and ring are vestiges of the manual switchboards that were used a number of years ago. between the phone and the CO. 6. The Local Loop Analog Line Signaling Dialing Ringing and Answer Voice Encoding ISDN BRI 6.PSTN (Public Switched Telephone Network) 6.1. While terminating the physical loop.2. and the access signaling between the subscriber and the CO. as they are often generically referred to. Remote line concentrators. In-band signaling is primitive when compared to the out-of-band signaling used in access methods such as ISDN. The local loop allows a subscriber to access the PSTN through its connection to the CO. modem. 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. The wire pair consists of a tip wire and a ring wire. in digital form. They are referred to as analog lines because they use an analog signal over the local loop. also referred to as Subscriber Line Multiplexers or Subscriber Line Concentrators. Remote switching centers provide local switching between subtending lines without using the resources of the CO. or on a remote line concentrator. The local loop terminates on the Main Distribution Frame (MDF) at the CO. The following sections describe the facilities used for lines. extend the line interface from the CO toward the subscribers. see the "ISDN BRI" section in this chapter for more . most phone lines are analog phone lines. Analog Line Signaling Currently. providing the subscriber access into the PSTN. they refer to the tip and ring of the actual switchboard plug operators used to connect calls. In some cases. The DTMF signal is a combination of two tones that are generated at different frequencies. 6. 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 voltage levels vary between different countries. The conversion is completed using a codec (coder/decoder).1. Voice Encoding An analog voice signal must be encoded into digital information for transmission over the digital switching network.5.1. This procedure is commonly referred to as ring trip.1. Dialing When a subscriber dials a number. Figure 9 Voice Encoding Process . 6. The quantization value is then encoded into a binary number to represent the individual data point of the sample. such as billing. The ITU G. 6. or by Dual Tone Multi-Frequency (DTMF) signals. can be initiated. When the destination phone is taken off-hook. DC current from the CO powers the local loop between the phone and the CO. the dialing plan of the CO determines when all digits have been collected. 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. but an on-hook voltage of –48 to –54 volts is common in North America and a number of other geographic regions.PSTN (Public Switched Telephone Network) information. An analog-to-digital converter samples the analog voice 8000 times per second and then assigns a quantization value based on 256 decision levels.3. which converts between analog and digital data. the CO detects the change in loop current and stops generating the ringing voltage. if necessary. Ringing and Answer To notify the called party of an incoming call. Figure illustrates the process of sampling and encoding the analog voice data. 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.711 standard specifies the Pulse Coded Modulation (PCM) method used throughout most of the PSTN. the number is signaled to the CO as either a series of pulses based on the number dialed. The off-hook signals the CO that the call has been answered. Usually.4. including the United Kingdom. the conversation path is then completed between the two parties and other actions. but requires more processing power and results in lower quality speech.PSTN (Public Switched Telephone Network) Two variations of encoding schemes are used for the actual quantization values: A-law and m-Law encoding. G. which are the focus of this section. North America uses m-Law encoding. Within the context of ISDN reference points. BRI multiplexes two bearer (2B) channels and one signaling (D) channel over the local loop between the subscriber and the CO. the digital voice data is decoded and converted back into an analog signal before transmitting over the loop. The two B channels each operate at 64 kb/s and can be used for voice or data communication.6. Digital trunks may be either four-wire .711 encoding/decoding requires little processing and produces high quality speech. 6. Combining ISDN on the access portion of the network with digital trunks on the core network provides total end-to-end digital connectivity. if an analog phone is used) and sent to a local bus: the S/T bus. the local loop is referred to as the U-loop. The emergence of voice over IP (VoIP) has prompted the use of other voice-encoding standards. Trunks Trunks carry traffic between telephony switching nodes.729.1 consumes little bandwidth. but consumes more bandwidth. It uses different electrical characteristics than those of an analog loop. ISDN moves the point of digital encoding to the customer premises. The D channel operates at 16 kb/s and is used for call control signaling for the two B channels. and ITU G.723. ISDN access signaling coupled with SS7 signaling in the core network achieves end-to-end out-of-band signaling.726. making them more suitable for use in packet networks. and voice quality. G.1. For example. The NT1 provides the interface between the Customer Premises Equipment (CPE) and the U-loop.723. 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. In contrast. The D channel can also be used for very low speed data transmission. When voice is transmitted from the digital switch over the analog loop. most trunks in use today are digital trunks. it has been a relatively slow-moving technology in terms of number of installations.2. While analog trunks still exist. There are two ISDN interface types: Basic Rate Interface (BRI) for lines. Each encoding method involves trade-offs between bandwidth. such as ITU G. These encoding methods use algorithms that produce more efficient and compressed data. ISDN also provides out-of-band signaling over the local loop. ISDN access signaling is designed to complement SS7 signaling in the core network. G. ISDN BRI Although Integrated Services Digital Network (ISDN) deployment began in the 1980s. this is commonly referred to as 2B+D. and Primary Rate Interface (PRI) for trunks. processing power required for the encoding/decoding function. Voice quantization is performed within the ISDN phone (or a Terminal Adapter. 6. and European countries use A-law encoding. TDM allocates one timeslot from each digital data stream's frame to transmit a voice sample from a conversation. voice channels are multiplexed into digital bit streams using Time Division Multiplexing (TDM). Transmission rates are calculated in multiples of DS0 rates. representing a single 64 kb/s channel that occupies one timeslot of a Time Division Multiplex (TDM) trunk. A/B/C/D bits are used to indicate trunk supervision signals. Figure 10 T1/E1 Framing Formats The E1 format also contains a channel dedicated to signaling when using in-band signaling. Each frame carries a total of 24 multiplexed voice channels for T1 and 30 channels for E1. T1 and E1 are the most common trunk types for connecting to End Offices. SONET defines the physical interface. Trunks are multiplexed onto higher capacity transport facilities as traffic is aggregated toward tandems and transit switches. The term "robbed bit" comes from the fact that bits are taken from the PCM data to convey trunk supervisory signals. In North America. This is also referred to as A/B bit signaling. In countries outside of North America. making it the medium of choice for high-density trunking. Figure shows the formats for T1 and E1 framing. In the case of Extended Superframe trunks (ESF). and European networks use E1. The basic unit of transmission is Digital Signal 0 (DS0). the more likely optical fiber will be used for trunk facilities for its increased bandwidth capacity. Synchronous Optical Network (SONET) is the standard specification for transmission over optical fiber. but it still exists in some areas. such as on/off-hook status and winks. For . while E1 uses a byte. frame format. and an OAM&P protocol. optical line rates. the least significant bits from each PCM sample are used as signaling bits. The higher up in the switching hierarchy. Fiber can accommodate a much higher bandwidth than copper transmission facilities. Synchronous Digital Hierarchy (SDH) is the equivalent optical standard. In every sixth frame. On the T1/E1 facility. The T1 format uses "robbed bit" signaling when using in-band signaling. A/B bit signaling has been widely replaced by SS7 signaling. Standard designations describe trunk bandwidth in terms of its capacity in bits/second.PSTN (Public Switched Telephone Network) (twisted pairs) or fiber optic medium for higher capacity. North American networks use T1. The T1 frame uses a single bit for framing. a T1 uses 24 voice channels at 64 kb/s per channel to produce a DS1 transmission rate of 1.096 39.544 mb/s The optical transmission rates in the SONET transport hierarchy are designated in Optical Carrier (OC) units.080 OC-48 STS-48 STM-16 32.256 2488.840 OC-3 STS-3 STM-1 2016 155.736 Table_2: SONET/SDH Transmission Rates SONET Optical Level SONET Electrical Level SDH Level Voice Channels Transmission Rate Mb/s OC-1 STS-1 — 672 51. the electrical equivalent signals are designated as Synchronous Transport Signal (STS) levels.048 E3 (Europe) 480 34.813. calculated as follows: 24 x 64 kb/s = 1.280 OC-768 STS-768 STM-256 516.024 9953.520 OC-12 STS-12 STM-4 8064 622.536 kb/s + 8000 b/s framing bits = 1.368 T3 (North America) 672 44. Table_1: Electrical Transmission Rates Designation Voice Channels Transmission Rate Mb/s T1 (North America) 24 1.512 4976. In North America.PSTN (Public Switched Telephone Network) example. OC-3 is simply three times the rate of OC-1. OC-1 is equivalent to T3.120 . Table_1 summarizes the electrical transmission rates.544 E1 (Europe) 30 2.544 mb/s. and Table_2 summarizes the SONET/SDH transmission rates. for example.64 OC-192 STS-192 STM-64 129. Higher OC units are multiples of OC-1. The ITU SDH standard uses the STM to designate the hierarchical level of transmission.320 OC-96 STS-96 STM-32 64. They are clocked into the receiving switch at the other end of the facility. The ULoop terminates to an NT1. In the United States. The single signaling channel handles the signaling for calls on the other 23 channels. microwave stations and satellites are also used to communicate using radio signals between offices. In the United States. which is typically integrated into the PBX at the customer premises.048 Mb/s. Digital facility interfaces use buffering techniques to store the incoming frame and accommodate slight variation in . Figure 5-8 illustrates a PBX connected to the CO through a PRI trunk. The sending switch clocks the bits in each frame onto the transmission facility. In Europe. U-Loop for PRI is a four-wire interface that operates at 1. You can also designate a channel as a backup D channel to provide redundancy in case of a primary D channel failure. PRI is based on 32 channels at a transmission rate of 2. which are referred to as 30B+2D. Network Timing Digital trunks between two connecting nodes require clock synchronization in order to ensure proper framing of the voice channels. There are 30 bearer channels and two signaling channels.3. PRI converts all data at the customer premises into digital format before transmitting it over the PRI interface. PRI uses 23 bearer channels for voice/data and one signaling channel for call control. This is particularly useful where it is geographically difficult to install copper and fiber into the ground or across rivers. 6. Each channel operates at a rate of 64 kb/s.544 mb/s. As with BRI. This scheme is commonly referred to as 23B+D. ISDN PRI Primary Rate Interface (PRI) provides ISDN access signaling over trunks and is primarily used to connect PBXs to the CO.PSTN (Public Switched Telephone Network) In addition to copper and fiber transmission mediums. Figure 11 ISDN Primary Rate Interface Other variations of this scheme use a single D channel to control more than 23 bearer channels. 7. Because digital transmission facilities connect switches throughout the network. A problem arises if the other digital switch that is connected to the facility has a clock signal that is out of phase with the first switch. Buffer underrun occurs if the frequency of the sending clock is less than the frequency of the receiving clock. discarding an entire frame of data. repeating a frame of data. Occasional slips do not present a real problem for voice calls. One method involves a single master clock source. A stratum 4 clock provides an accuracy of ±32 x 10-6. This condition is known as slip. where the synchronization of many switches is required. Since the deployment of Global Positioning System (GPS) satellites. By using a flattened hierarchy based on GPS receivers. from which other nodes derive timing in a master/slave arrangement.PSTN (Public Switched Telephone Network) the timing of the data sent between the two ends. and it results in buffer overrun or buffer underrun. they are more detrimental to the data transfer. synchronization of time sources between the digital switches is important. There are various methods of synchronizing nodes. this requirement escalates to a network level. The clocks' accuracy is rated in terms of stratum levels. 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. . However. The variation in clock signals eventually causes errors in identifying the beginning of a frame. Another method uses a plesiochronous arrangement. and also shows an example that uses a GPS satellite clock receiver at each office. meaning only one error can occur in 1011 parts. Buffer overrun occurs if the frequency of the sending clock is greater than the frequency of the receiving clock. where each node contains an independent clock whose accuracy is so great that it remains independently synchronized with other nodes. Figure shows an example that uses a stratum 1 clock at a digital switching office to distribute timing to subtending nodes. in which each bit is important. providing timing to subtending nodes in the network. the receiver typically runs free at stratum 2 or less. A stratum 1 clock provides the most accurate clock source with a freerunning accuracy of ±1 x 10 -11. Stratums 1 through 4 denote timing sources in order of descending accuracy. If the GPS receiver loses the satellite signal. although excessive slips result in degraded speech quality. each with a number of atomic clocks on-board. you remove the need to distribute the clock signal and provide a highly accurate reference source for each node. 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. Therefore. The performance requirement 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. The PSTN existed long before SS7. If they produce unrecoverable errors. it describes an existing switching node. and it represented a substantial investment. During SS7's initial deployment. but used timeslots on existing trunk facilities. the examples and information are brief and discussed only in the context of the network nodes presented in this section. you must always consider network timing when establishing SS7 links between nodes in the PSTN. Since SS7 has not been presented in great detail. to preserve the investment and provide minimal disruption to the network. all of the switching nodes and facilities that . 8. SS7 did not introduce new facilities for signaling links. slips on the transmission facilities might affect SS7 messages. cost savings. Therefore. When looking at the SS7 network topologies in later chapters. Instead. In the SS7 network. Federal regulation. it is important to realize that the SSP is not a new node in the network. Similarly. Integration of SS7 into the PSTN This section provides a brief overview of how the SS7 architecture is applied to the PSTN. The network's general structure was already in place. to which SS7 capabilities have been added. 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.PSTN (Public Switched Telephone Network) Figure 12 Network Timing for Digital Transmission SS7 links are subject to the same timing constraints as the trunk facilities that carry voice/data information because they use digital trunk transmission facilities for transport. IXCs embraced SS7 early to cut down on post-dial delay which translated into significant savings on access/egress charges. such as the STP and SCP. however. additional hardware was added and digital switches received software upgrades to add SS7 capability to existing PSTN nodes. SS7 was designed to integrate easily into the existing PSTN. The introduction of SS7 added new nodes. a digital switch with SS7 capabilities is referred to as a Service Switching Point (SSP). and the opportunity to provide new revenue generating services created a need to deploy SS7 into the existing PSTN. can extract each of the timeslots from the digital stream. The channel bank. The individual SS7 link provides the SS7 messages to the digital switch for processing. 9. At each node. such as a T1 or E1. or a Digital Access and Cross-Connect (DAC).and service-related information is passed on to the central processor. SS7 Link Interface The most common method for deploying SS7 links is for each link to occupy a timeslot. which demultiplexes the TDM timeslot from the digital trunk. Figure shows a simple view of the PSTN.1.PSTN (Public Switched Telephone Network) existed before SS7 was introduced are still in place. on a digital trunk. Level 2. As shown in Figure 5-12. View b shows the SS7 topology using simple associated signaling for all nodes. While implementations vary. 8. Dial-up Internet services use data connections . View c shows the actual SS7-enabled PSTN topology. allowing them to be processed individually. the SS7 interface equipment must extract the link timeslot from the digital trunk for processing. Evolving the PSTN to the Next Generation The expansion of the Internet continues to drive multiple changes in the PSTN environment. and possibly a portion of Level 3). The existing switching nodes and facilities are enhanced to provide basic SS7 call processing functionality. the signaling links actually travel on the digital trunk transmission medium throughout the network. overlaid with SS7-associated signaling capabilities. or to other peripheral processors that are designed for handling call processing–related messages. Of course. call. Figure 13 SS7 Overlaid onto the PSTN View a in the previous figure shows that trunk facilities provide the path for voice and in-band signaling. Although this associated signaling architecture is still quite common in Europe. this process varies based on the actual equipment vendor. the United States primarily uses a quasi-associated signaling architecture. This process is typically performed using a channel bank. more network capacity is used to transport data over the PSTN. dedicated peripheral processors usually process the lower levels of the SS7 protocol (Level 1. or DAC. First. " discusses these VoIP technologies in more detail. Figure 14 VoIP Gateways to the PSTN . which was the average length used for calculations when engineering the voice network. thereby providing an alternative to sending calls over the PSTN. These gateways provide access points for interconnecting the two networks. Among the current leading VoIP technologies are: Soft switches H. Several different architectures and protocols are competing in the VoIP market to establish alternatives to the traditional circuit-switched network presented in this chapter. Circuit-switched connections are dedicated connections. thereby creating a migration path from PSTN-based phone service to VoIP phone service. The technologies are not necessarily exclusive. in which many data transactions share the same facilities. reach much further than simply an increase in network traffic. however. Internet connections tend to be much more lengthy. Data networks use packet switching. Chapter 14. Figure shows the interconnection of VoIP architectures to the PSTN using signaling gateways and trunking gateways. which occupy a circuit for the duration of a call. PRI is also commonly used for business to network access.PSTN (Public Switched Telephone Network) that are set up over the PSTN to carry voice-band data over circuit-switched connections. Phone traffic is being moved to both private packet-based networks and the public Internet.323 Session Initiation Protocol (SIP) Each of these VoIP architectures uses VoIP-PSTN gateways to provide some means of communication between the traditional PSTN networks and VoIP networks. "SS7 in the Converged World. 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. Of course. The phone networks were originally engineered for the three-minute call. This is a much different situation than sending data over a data network. The changes driven by the Internet. some solutions combine the various technologies. meaning that more network capacity is needed. 000 Hz. This process is known as quantization. it then matches each sample to a voltage scale. and so on that comprises the sound. Here are a few key facts: The average human ear is able to hear frequencies from 20–20. Human speech uses frequencies from 200–9. Telephone channels typically transmit frequencies from 300–3. Figure 15 Analog Signal Sample The three basic step of Analog conversion to VoIP: Sampling: It is the reduction of a continuous signal to a discrete signal.000 times (2 * 4000) every second. A sample refers to a value or set of values at a point in time and/or space. Because he was after audio frequencies from 300–4. VoIP Gateways to the PSTN Dr. Here is how it breaks down. It assigns a value from the voltage range based on the amplitude of each audio sample.000 Hz.000 Hz Nyquist believed that you can accurately reproduce an audio signal by sampling at twice the highest frequency. .PSTN (Public Switched Telephone Network) 10. The Nyquist theorem is able to reproduce frequencies from 300–4.000 Hz. during the process of sampling.400 Hz. Audio frequencies vary based on the volume. the sampling device puts an Analog waveform against a Y-axis lined with numeric values. it would mean sampling 8. The quantization process divides the voltage range into 16 total segments (0 to 7 positive and 0 to 7 negative). Quantization: After the digitizing device has taken thousands of samples of the Analog audio. Harry Nyquist (and many others) created a process that allows equipment to convert Analog signals (flowing waveforms) into digital format (1s and 0s). He found that he could accurately reconstruct audio streams by taking samples that numbered twice the highest audio frequency used in the audio. As Figure 1-12 illustrates. After plenty of research. pitch. The second step of the process then matches the PAM sample to a specific voltage value. Moving into the realm of VoIP. DSPs offload the processing responsibility for voice-related tasks from the processor of the router. Compression (optional): Some voice systems allow you to save bandwidth by compressing the audio before sending it to the remote device. but many of them can save a significant amount of bandwidth with little quality degradation. the network now requires the router to convert loads of voice into digitized. having 256 MB of RAM is quite a bit. This final conversion is known as pulse-code modulation (PCM). the digitizing equipment converts the sample value into an 8-bit.PSTN (Public Switched Telephone Network) Encoding: In the first step of the digitization process. Figure 16 RAM used for Packet Processing . packetized transmissions. The compression methods vary in overhead and audio quality. 256 MB will barely help you survive the Microsoft Windows boot process. the Analog voice is sampled using PAM. From a PC’s perspective. This is where DSPs come into play. binary number. Moving packets between one location and another is not a processor-intensive task. 11. thus Cisco routers are not equipped with the kind of memory and processing resources typical PCs are equipped with. Role of Digital Signal Processors Cisco designed its routers with one primary purpose in mind: routing. This task would easily overwhelm the resources you have on the router. from a router’s perspective. For example. In this step. allowing it to connect to the PSTN or Analog devices. Keep in mind that RTP streams are one way. Having two transport layer protocols is odd. Figure 17 Protocol Packet Header The Payload Type field in the RTP header is used to designate what type of RTP is in use.) At the time the devices establish the call. Once two devices attempt to establish an audio session. You can use RTP for audio or video purposes. you encounter a whole new host of protocol standards. its primary job is statistics reporting. RTP engages and chooses a random. one in each direction. RTP adds time stamps and sequence numbers to the header information. This allows the remote device to put the packets back in order when it receives them at the remote end (function of the sequence number) and use a buffer to remove jitter (slight delays) between the packets to give a smooth audio playout (function of the time stamp). It delivers statistics between the two devices participating in the call. which include: . If you are having a two-way conversation. The interfaces would be able to actively connect to the legacy voice networks.384 to 32. The audio stream stays on the initially chosen port for the duration of the audio session. Understanding RTP and RTCP When you walk into the VoIP world. the devices establish dual RTP streams. the interfaces would be worthless. but did not equip your router with DSPs. but that’s exactly what is happening here. Think of the Real-time Transport Protocol (RTP) and Real-time Transport Control Protocol (RTCP) as the protocols of voice. Figure represents the RTP header information contained in a packet. (The devices do not dynamically change ports during a phone call. session multiplexing) and header checksums (which ensure that the header information does not become corrupted). UDP provides the services it always does: port numbers (that is.767 for each RTP stream. even UDP port number from 16. but would not have the power to convert any voice into packetized form.PSTN (Public Switched Telephone Network) Specifically. encoding. 12. If you were to equip your router with voice interface cards (VIC). a DSP is a chip that performs all the sampling. and compression functions on audio coming into your router. RTP operates at the transport layer of the OSI model on top of UDP. Although this protocol sounds important. RTCP also engages. which allows you to determine the issues that are causing call problems (such as poor audio. and so on) on the network. call disconnects.384 to 32. 13. Not Needed No No No No Dynamic On every packet No Yes No Per Packet .767. Keep this in mind when you configure QoS settings. AS the devices establish the call. as previously discussed. the devices send RTCP packets at least once every 5 seconds. Comparison between Circuit-Switched & Packet-Switched Item Call Setup Dedicated Physical Path Each packet follows the same route Packets arrive in order Is a switch crash fatal Bandwidth available When can congestion occur Potentially wasted bandwidth Store-and-forward transmission Transparency Charging Circuit-Switched Packet-Switched Required Yes Yes Yes Yes Fixed At setup Time Yes No Yes Per Min. the RTP audio streams use an even UDP port from 16. it is not nearly as critical as the actual RTP audio streams. RTCP creates a separate session over UDP between the two devices by using an odd-numbered port from the same range. The Cisco Unified Communication Manager Express (CME) router can log and report this information.PSTN (Public Switched Telephone Network) Packet count Packet delay Packet loss Jitter (delay variations) Although this information is useful. Throughout the call duration. 25 IP FrameRelay ATM Connectionless Broadcasting Modest error rates often accepted Fast data in good channels IP UDP . Connection Mode & Connection Summarized Transfer Modes Circuit Switching Developed for voice Nowadays also for data Well-specified delays Echo problems Packet Switching Developed for data Nowadays also for voice Statistical multiplexing Variable delays Protocols Connection Types Protocols PSTN ISDN PCM Connection Oriented Hand-shaking strict error requirements for fast data transfer ATM FrameRelay x.PSTN (Public Switched Telephone Network) 14.