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The PSTN is the global public voice network that provides voice services. The PSTN is a variety of networks and services that are in place worldwide; it provides a circuit-switched service using Signaling System 7 (SS7) for out-of-band call provisioning through the network. Central office (CO) switches exchange SS7 messages to place and route voice calls throughout the network. The PSTN uses Time-Division Multiplexing (TDM) facilities for calls within the network. From the CO to the customer premises, the call can be analog, ISDN, or TDM digital. Each call consumes 64 Kbps of bandwidth, called digital service zero (DS0).
PBX and PSTN Switches
Traditional switches and PBXs route voice using TDM technology and use 64-kbps circuits. The CCDA must understand some of the differences between these devices. The PBX, as its name states, is used in a private network and uses proprietary protocols. The PBX is located in the enterprise's data center. Each PBX may scale up to 1000 phones. Companies deploy PBX networks to obtain enterprise features and to prevent PSTN long-distance charges.
PBXs are customer-owned voice switches. Enterprise companies install and configure their own PBXs to provide telephony service, four-digit dialing, remote-office extensions, voice mail, and private-line routing within other features. Organizations can reduce toll charges by using private tie-lines between their switches. Calls that are placed between offices through the private voice network are called on-net. If a user needs to place a call outside the private network, the call is routed to the local PSTN. If the call is forwarded to the PSTN, it is called off-net.
Figure 15-1 shows a PBX network for an enterprise. Callers use the PBX network when they place calls from San Diego to Chicago, Atlanta, or Houston. The enterprise reduces toll charges by using its private voice network. A separate private network is in place for data traffic. If a user places a call from San Diego to Los Angeles, it is routed to the PSTN from the San Diego PBX. Then, toll charges are incurred for the call.
Another issue in the design is the limitation on the number of calls per private line. If the private lines are T1s, they are each limited to carrying 24 concurrent calls at a time. This is because each call takes 64 kbps of bandwidth with the g.711 codec, and 24 calls times 64 kbps/call equals 1.536 Mbps, the bandwidth of a T1.
PSTN switches are not private. They scale up to 100,000 phones and use open standards because they have to communicate with other switches, PBXs, fax machines, and home telephones. PSTN switches normally are located at the CO of the local or interexchange carrier.
Local Loop and Trunks
Depending on the dialed digits, a call routes through the local loop, one or more trunks, and the destination local loop to reach the destination phone. The local loop is the pair of wires that runs from the CO to the home or business office.
Trunks connect two switches. The type of trunk depends on the function of the switches the trunk is connecting. The term tie-line is frequently used instead of trunk to describe a dedicated line connecting two telephone switches within a single organization. The following is a list of trunk types:
- Interoffice trunk connects two CO switches. Also called a PSTN switch trunk.
- Tandem trunk connects central offices within a geographic area.
- Toll-connecting trunk connects the CO to the long-distance office.
- Intertoll trunk connects two long-distance offices.
- Tie trunk connects two PBXs. Also called a private trunk.
- PBX-to-CO trunk or CO-to-PBX business line connects the CO switch to the enterprise PBX.
Figure 15-2 shows an example of the PSTN. All phones connect to their local CO via the local loop. Calls between Phones 1 and 2 and between Phones 4 and 5 go through interoffice trunks. Calls between Phones 2 and 3 go through tandem trunks within a region. When you place calls between Texas and Massachusetts, they are forwarded to the long-distance toll provider via a toll-connecting trunk and are routed through intertoll trunks.
Ports
- Foreign Exchange Office (FXO) connects to the PSTN. It is an RJ-11 connector that allows an analog connection to be directed to the PSTN's central office or to a station interface on a PBX. The FXO sits on the switch end of the connection. It plugs directly into the line side of the switch, so the switch thinks the FXO interface is a telephone.
- Channelized T1 (or E1) is commonly used as a digital trunk line to connect to a phone switch where each DS0 supports an active phone call connection. Provides 24 (for T1) or 32 (for E1) channels or DS0 for voice calls. The total bandwidth for a T1 is 1.536 Mbps, and the total bandwidth for an E1 is 2.048 Mbps.
Major Analog and Digital Signaling Types
Signaling is needed to provide the state of telephones, digit dialing, and other information. For a call to be placed, managed, and closed, all of the following signaling categories have to occur:
- Addressing provides dialed digits.
- CO to phone (loop and ground start signaling)
- PBX to PBX (E&M)
- T1/E1 Channel Associated Signaling (CAS)
- ISDN PRI Common Channel Signaling (CCS)
- SS7 interswitch PSTN signaling
Loop-Start Signaling
Loop-start signaling is an analog signaling technique used to indicate on-hook and off-hook conditions in the network. It is commonly used between the telephone set and the CO, PBX, or FXS module. As shown in Figure 15-3, with loop-start the local loop is open when the phone is on-hook. When the phone is taken off-hook, a –48 direct current (DC) voltage loops from the CO through the phone and back. Loop-start signaling is used for residential lines.
Ground-Start Signaling
Ground-start signaling is an analog signaling technique used to indicate on-hook and off-hook conditions. Ground-start is commonly used in switch-to-switch connections. The difference between ground-start and loop-start is that ground-start requires the closing of the loop at both locations. Ground-start is commonly used by PBXs.
The standard way to transport voice between two telephone sets is to use tip and ring lines. Tip and ring lines are the twisted pair of wires that connect to your phone via an RJ-11 connector. As shown in Figure 15-4, the CO switch grounds the tip line. The PBX detects that the tip line is grounded and closes the loop by removing ground from the ring line.
E&M Signaling
E&M is an analog signaling technique often used in PBX-to-PBX tie-lines. E&M is receive and transmit, or more commonly called ear and mouth. Cisco routers support four E&M signal types: Type I, Type II, Type III, and Type V. Types I and II are most popular on the American continents. Type V is used in the United States and Europe.
CAS and CCS Signaling
Digital signaling has two major forms: Channel Associated Signaling (CAS) and Common Channel Signaling (CCS). The major difference is that with CAS the signaling is included in the same channel as the voice call. With CCS the signaling is provided in a separate channel. Table 15-2 shows the common types of CAS and CCS. They are covered in the following sections.
| From | Signaling Type |
|---|---|
| CAS | T1 or E1 signaling DTMF |
| CCS | ISDN PRI or BRI QSIG SS7 |
T1/E1 CAS
Digital T1 CAS uses selected bits within a selected channel to transmit signaling information. CAS is also called robbed-bit signaling or in-band signaling in the T1 implementation. Robbed-bit CAS works with digital voice because losing an occasional voice sample does not affect the voice quality. The disadvantage of robbed-bit CAS is that it cannot be used on channels that might carry voice or data without reducing the data rate to 56 Kbps to ensure that signaling changes do not damage the data stream.
E1 CAS uses a separate channel in the shared medium for CAS, so it does not have this disadvantage. The E1 signaling bits are channel-associated, but they are not in-band.
ISDN PRI/BRI
ISDN T1 PRI provides 23 64-kbps B (bearer) channels for voice, with a separate 64-kbps D (data signaling) channel for signaling. The ISDN E1 PRI provides 30 B channels. The use of messages in a separate channel, rather than preassigned bits, is also called common-channel signaling. Any bit in the signaling channel is common to all the channels sharing the medium rather than dedicated to a particular single channel. ISDN provides the advantage of not changing bits in the channels and thus is useful for data traffic in addition to voice traffic.
The ISDN BRI interface includes two 64-kbps B channels for voice or data and a separate 16-kbps D channel that provides signaling for the interface.
Q.SIG
Q.SIG is the preferred signaling protocol used between PBX switches. It is a standards-based protocol, based on ISDN, that provides services. It is feature-transparent between PBXs. It is interoperable with public and private ISDN networks and provides no restrictions to private dial plans. QSIG is also used between Cisco's Unified CallManager and enterprise PBX in hybrid implementations.
SS7
SS7 is a global ITU standard for telecommunications control that allows voice-network calls to be routed and controlled by call-control centers. SS7 is used between PSTN switches. SS7 implements call setup, routing, and control, ensuring that intermediate and far-end switches are available when a call is placed. With SS7, telephone companies can implement modern consumer-telephone services such as caller ID, toll-free numbers, call forwarding, and so on.
SS7 provides mechanisms for exchanging control, status, and routing messages on public telephone networks. SS7 messages pass over a separate channel than that used for voice communication. You use Common Channel Signaling 7 (CCS7) when speaking about SS7 signaling. CCS7 controls call signaling, routing, and connections between CO, interexchange carrier, and competitive local exchange carrier switches. Figure 15-5 shows the connectivity between SS7 components.
As shown in Figure 15-5, SS7 has the following system components:
Addressing Digit Signaling
- Pulse or rotary dialing
- Dual-tone multifrequency (DTMF) dialing
Pulse dialing uses the opening and closing of a switch at the telephone set. A rotary register at the CO detects the opening and closing of the loop. When the number 5 is dialed on a rotary phone, the dial mechanism opens and closes five times, each one-tenth of a second apart.
DTMF uses two tones simultaneously to indicate the dialed number. Table 15-3 shows the phone keypad and the frequencies used. For example, when the number 5 is dialed, the frequencies 770 Hz and 1336 Hz are sent to the CO.
| Frequency | 1209 Hz | 1336 Hz | 1477 Hz |
|---|---|---|---|
| 697 Hz | 1 | ABC 2 | DEF 3 |
| 770 Hz | GHI 4 | JKL 5 | MNO 6 |
| 852 Hz | PRS 7 | TUV 8 | WXY 9 |
| 941 Hz | * | OPER 0 | # |
PSTN Numbering Plan
The PSTN uses the ITU E.164 standard for public network addressing. The E.164 standard uses a maximum of 15 digits and makes each phone unique in the PSTN. Examples of E.164 addresses are the residential, business, IP phones, and cell phones that you use every day. Each country is assigned a country code to identify it. The country codes can be one to three digits in length. Table 15-4 shows some examples of country codes.
| Country Code | Country |
|---|---|
| 1 | United States, Canada |
| 1-787, 1-939 | Puerto Rico |
| 55 | Brazil |
| 39 | Italy |
| 86 | China |
| 20 | Egypt |
| 91 | India |
| 49 | Germany |
| 380 | Ukraine |
| 44 | United Kingdom |
| 81 | Japan |
| 52 | Mexico |
| 966 | Saudi Arabia |
The ITU website that lists country codes is located at http://www.itu.int/itudoc/itu-t/ob-lists/icc/e164_763.html.
Each country divides its network into area codes that identify a geographic region or city. The United States uses the North American Numbering Plan (NANP). NANP has the address format of NXX-NXX-XXXX, where N is any number from 2 to 9 and X is any number from 0 to 9. The first three digits are the area code. The address is further divided into the office code (also known as prefix) and line number. The prefix is three digits, and the line number is four digits. The line number identifies the phone.
An example of a PSTN address in the United States is 1-713-781-0300. The 1 identifies the United States; the 713 identifies an area code in the Houston, Texas, geographical region. The 781 identifies a CO in west Houston. The 0300 identifies the phone.
Another example of a PSTN address is 52-55-8452-1110. The country code 52 identifies the country of Mexico. The area code 55 identifies the geographic area of Mexico City. The office code 8452 and line number 1110 follows.
Other PSTN Services
Centrex Services
Companies can use the local phone company to handle all their internal and external calls from the CO. In this voice model, the CO acts as the company's voice switch, with PBX features such as four-digit extension dialing, voice mail, and call holds and transfers. The Centrex service gives the company the appearance of having its own PBX network.
Voice Mail
PSTN service providers can enable voice messaging for customers that request the service. Voice mail provides automated call answering and message recording. Users can then retrieve the message and forward it to other extensions.
Database Services
The PSTN must keep call detail records (CDR) in the database systems. CDR information includes all types of call information, such as called party, caller, time, duration, locations, and user service plans. This information is used for billing and reporting.
IVR
IVR systems connect incoming calls to an audio playback system. IVR queues the calls, provides prerecorded announcements, prompts the caller for key options, provides the caller with information, and transfers the call to another switch extension or agent. IVR is used in customer call centers run by companies in all industries to gather and provide information to the customers before transferring them to agents.
ACD
ACD routes calls to a group of agents. ACD keeps statistics on each agent, such as the number of calls and their duration. Based on the statistics, the ACD system then can evenly distribute the calls to the agents or to the appropriate agent skill group. ACD is used by airline reservation systems, customer service departments, and other call centers.
Voice Terminology
You must consider voice traffic requirements when designing a network. The CCDA must be familiar with the following voice engineering terms.
Grade of Service
Grade of service (GoS) is the probability that a call will be blocked when attempting to seize a circuit. If it is determined that a network has a P.02 GoS, the probability is that 2 percent of all attempted calls will be blocked.
Erlangs
An Erlang is a telecommunications traffic unit of measurement representing the continuous use of one voice path for one hour. This means the use of a single voice resource for one hour (3600 seconds). It describes the total traffic volume of one hour. Erlangs determine voice-call usage for bandwidth requirements for voice network designs, including VoIP. It helps determine if a system has been provisioned with enough resources.
If a group of users makes 20 calls in an hour and each call lasts 10 minutes, the Erlangs are calculated as follows:
20 calls per hour * 10 minutes per call = 200 minutes per hour
traffic volume = (200 minutes per hour) / (60 minutes per hour) = 3.33 Erlangs
There are three common Erlang models:
Centum Call Second (CCS)
A Centum Call Second (CCS) represents one call occupying a channel for 100 seconds. It is the equivalent of 1/36th of an Erlang. In other words, 360 CCS equals 1 Erlang (3600 seconds).
Busy Hour
The busy hour is the specific hour within a 24-hour period in which the highest traffic load occurs. Most calls are placed and are of longer durations during this hour. It is also called peak hour.
Busy Hour Traffic (BHT)
BHT is the amount of voice traffic that occurs in the busy hour, expressed in Erlangs. It is calculated by multiplying the average call duration by the number of calls in the hour and then dividing that by 3600.
For example, if 300 calls occurred during the busy hour, with an average duration of 150 seconds, the BHT is calculated as follows:
Blocking Probability
The blocking probability is the probability that a call will be blocked. A blocking probability of 0.02 means that 2 percent of the calls will be blocked.
Call Detail Records
Call detail records include statistical and other information related to all calls placed. Information included in CDRs includes call time, call duration, source phone number, dialed phone number, and the amount billed. For VoIP networks, the CDR may also include source and destination IP addresses.
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