Ringing, Dial Tone, and Other Bells and Whistles

Earlier in this chapter, you considered an end-to-end telephone call between you and a friend. In passing, you learned that various types of "signaling" occurred during that phone call. The following sections examine the various types of signaling:

  • Supervisory signaling

  • Address signaling

  • Information signaling

Supervisory Signaling

Supervisory signaling indicates to the phone switch whether a connected phone is currently on-hook or off-hook and also when a phone receives an incoming call. Supervisor signaling includes:

  • Loop Start Signaling

  • Ground Start Signaling

  • Ringing

Loop Start Signaling

In a home environment, the phone switch in the local CO can determine whether a phone is on-hook or off-hook based on whether current is flowing over the local loop connecting back to that phone. Because an on-hook phone mechanically has its tip and ring circuit open, the -48 volts of DC applied across the tip and ring wires isn't doing anything. The voltage is just sitting there, waiting for the circuit to close. After the handset goes off-hook, however, the tip and ring circuit is closed, and current can begin to flow through that circuit. When the telephone switch at the CO sees this current begin to flow, it knows that the phone has gone off-hook, and the telephone switch sends a dial tone to the caller, indicating that they can begin dialing digits. This type of supervisory signaling is called loop start signaling.

Loop start signaling has an issue with glare. Did you ever pick up the phone to call someone, but you didn't hear any dial tone and instead discovered that someone was on the other end of the line? If so, you experienced glare. Glare occurs when you beat the signaling and pick up your handset before your phone rings. What's really spooky is when the person you were about to call is the person on the other end of the line!

Glare may not be a major concern in a home environment, but what about a line connecting to a company's PBX system? Because the lines connected to a PBX experience a significantly higher call volume than you do on your home phone (unless you have teenagers), the probability of glare occurring with a PBX using loop start signaling is much higher than the probability of glare occurring on your home phone. Therefore, you often find another type of signaling used on PBX systems, and also on pay phones. That other type of signaling is ground start, and the good news is that ground start signaling prevents glare.

Ground Start Signaling

With ground start signaling, the phone switch monitors the voltage potential on the "ring" lead of a line, and when the ring lead has a ground potential, the line is seized. If you watched the 1983 movie WarGames, you witnessed an example of ground start signaling. Do you remember the scene? Matthew Broderick's character rides his bicycle up to a pay phone, but he doesn't have any money. So, he takes the pay phone handset and bangs it against the pay phone's chassis, which loosens the transmitter cover. He opens up the transmitter portion of the handset, pulls out one of the leads (the ring lead), and touches the lead to the chassis of the pay phone (which had a ground potential). By creating this off-hook ground start signal, he can place a call.


Circuits in today's pay phones prevent the falsification of ground start signals, as Matthew Broderick did in WarGames.


Ringing is also considered to be supervisory signaling. Ringing voltage is sent from the telephone switch to alert the destination phone that it is receiving an incoming call. Here is a fun experiment: the next time your home phone rings, start counting (one thousand one, one thousand two, …) to see how many seconds the ringing lasts and how many seconds of silence there are before the ringing begins again. In the United States, the pattern of ringing, called the ring cadence, is two seconds on and four seconds off, as Figure 1-4 illustrates. However, it seems that Hollywood isn't that patient. If a phone rings on a TV show or a movie, most of the time, they use a much shorter ring pattern (typically, one second on and two seconds off).

Figure 1-4. Ringing Pattern Examples

Pop quiz time! Who invented the telephone ringer? This is a question that I ask students in my classes, and the most popular answer is Alexander Graham Bell. However, it was Thomas Watson, Mr. Bell's assistant, who invented the mechanical ringer. Back in those days, the copper wiring over which the voice traffic and ringing voltage was sent wasn't manufactured to the quality standards that today's transmission is. As a result, Thomas Watson used significant voltage to go across this lower quality medium to cause a mechanical ringer to ring. Specifically, the original ringer required 75 volts of AC. Because today's signaling might be communicated over fiber optic cable, and the ringers are rarely the mechanical kind, you might assume that such high voltage levels would not be necessary. However, the tradition of using higher voltages for ringing is still observed today in the PSTN, which is why you shouldn't touch bare telephone wires.

Another mystery of ringing voltage goes back to an earlier statement. When discussing how the CO knows when a phone goes off-hook, you learned that the tip and ring circuit is open when the phone is on-hook. That leads to the question, "If the tip and ring circuit is open, how can ringing current flow through this open circuit?" Actually, the circuit is considered open to DC. However, the phone's internal circuitry has an electrical component called a capacitor between the tip and ring wires, and even though direct current does not flow through a capacitor, AC does. Ringing voltage uses AC. Therefore, this ringing current can flow through a phone in the on-hook condition, causing the ringer to ring, as shown in Figure 1-5.

Figure 1-5. Ringing Circuit

Address Signaling

Address signaling allows a phone to specify the "address" (phone number) of the destination phone by dialing digits. Most phones support two types of dialing. The older method, used by rotary phones, is pulse dialing. Pulse dialing opens and closes the tip and ring circuit very rapidly. This series of open and closed circuit conditions within specific timing parameters indicates a dialed digit to the telephone switch, as shown in Figure 1-6.

Figure 1-6. Pulse Dialing

When a phone's handset is lifted off its cradle, the hook switch closes the tip and ring circuit. Similarly, when a phone's handset is placed back on its cradle, the hook switch opens the tip and ring circuit. In fact, as a child, I tried dialing phone numbers by rapidly tapping the hook switch, attempting to simulate pulse dialing. Admittedly, my timing wasn't perfect, and I didn't always dial the correct number, but I dialed someone!

A more efficient approach to address signaling is dual tone multifrequency (DTMF), also known as "touch tone" dialing. With DTMF, two simultaneous frequencies are generated, and a phone switch interprets this combination of frequencies as a dialed digit. For example, the combination of a 697 Hz tone and a 1209 Hz tone indicates a dialed digit of 1, as shown in Table 1-1. You might be curious as to why "dual" tones are used instead of just a single tone; the answer is background noise. The phone company doesn't want the radio or your kids playing in the background to make a sound that may be interpreted as a dialed digit. So, specific combinations of two simultaneous frequencies are used to represent a dialed digit.

Table 1-1. Dual Tone Multifrequency (DTMF) Frequencies


1209 Hz

1336 Hz

1477 Hz

697 Hz




770 Hz




852 Hz




941 Hz




Although pulse dialing served callers well for decades, DTMF dialing offers some significant advantages. Foremost of these advantages is speed. If you remember using an old rotary phone years ago (or even more recentlymy mother still has one), think about how long it took to dial a 0. You positioned your finger in the 0 position, made a clockwise motion to dial the digit, and released the dial. The dial then very slowly rotated counter-clockwise back to its original position. Due to the mechanical inertia built into those rotary phones, it took a full second to dial that 0. DTMF enables you to dial digits much more rapidly. If your fingers are nimble enough, you can dial several digits in a second. The tones generated by a DTMF keypad also enable a caller to interact with devices on the other side of the link. For example, suppose that you are away from home, and you want to check your messages, either on your home answering machine or on your voice mail. You dial your home number and then enter a series of DTMF tones to retrieve your messages. You would not be able to do that using pulse dialing.

Information Signaling

Similar to DTMF, information signaling uses combinations of frequencies, in this case to indicate the status of a call (that is, to provide information to the caller). For example, a busy signal is a combination of a 480-Hz tone and a 620-Hz tone, with on/off times of .5 sec/.5 sec. Another type of information signaling that you are probably familiar with is ring back. Ring back is the ringing sound heard by the caller to indicate that the dialed phone is ringing. Realize, however, that the ring back heard by the caller doesn't occur at exactly the same time as the ringing on the destination phone. Try it. The next time you call someone, ask them how many times their phone rang before they picked up. Compare that number with the number of times you heard ring back. In some instances, the numbers will differ. Table 1-2 lists several other types of information signaling used in North America that you might be familiar with, along with the corresponding frequencies used for each signal.

Table 1-2. Information Signaling

Information Signal


Frequencies (Hz)

Dial tone

Heard by the caller after picking up the telephone handset

350 and 440

Ring back

Heard by the caller, indicating that the called phone is ringing

440 and 480

Busy signal

Heard by the caller, indicating that the called phone is off-hook

480 and 620

Reorder tone

Heard by the caller, indicating that the call cannot be completed successfullyperhaps due to all trunks being busy

480 and 620

A set of rules that determines how information is exchanged is called a protocol. You can think of a protocol as a "language of love" between two devices. The protocol that runs over the Internet, as you might guess, is called the Internet Protocol (IP). For decades, IP has transmitted data, not just across the public Internet, but also across private networks.

Because voice can be digitized (converted to binary 1s and 0s), as described in Chapter 2, binary digits representing the voice can be transmitted across existing IP-based data networks. The process of sending voice traffic across an IP network is called Voice over IP (VoIP).

As Figure 1-7 illustrates, a VoIP network has its own pieces and parts, just as a traditional telephony network does.

Figure 1-7. VoIP Components

The list that follows defines the VoIP network components illustrated in Figure 1-7:

  • IP phones Have an Ethernet network connection used to send and receive voice calls.

  • Call agents Replace many of the features previously provided by PBXs. For example, a call agent can be configured with rules that determine how calls are forwarded. The Cisco CallManager (CCM) product is an example of a call agent.

  • Gateways Can forward calls between different types of networks. For example, you could place a call from an IP phone in your office, through a gateway to the PSTN, to call your home.

  • Gatekeepers Can be thought of as the traffic cops of the wide-area network (WAN). For example, because bandwidth on a WAN is typically somewhat limited, a gatekeeper can monitor the available bandwidth on the WAN. Then, when there is not enough bandwidth to support another voice call, the gatekeeper can deny future call attempts.

  • Multipoint Control Units (MCUs) Are useful for conference calling. On a conference call, multiple people can be talking at the same time, and everyone on that conference call can hear them. It takes processing power to mix these audio streams together. MCUs provide that processing power. MCUs may contain digital signal processors (DSPs), which are dedicated pieces of computer circuitry that can mix those audio streams together.

  • Voice-enabled Ethernet switches Add quality of service features to traditional Ethernet switches, allowing voice packets to be stored in a separate area from data packets. Voice-enabled Ethernet switches can recognize an attached IP phone, provide the attached IP phone with subnet information, and optionally supply power to the IP phone.


The term Ethernet switch should not be confused with the term phone switch, discussed earlier in this chapter. Ethernet switches forward data, whereas phone switches forward phone calls.

With the maturity of PBX-centric phone systems, why might you consider migrating your existing tried and true PBX to a VoIP network? One of the first responses that comes to most people's minds is cost, and that is certainly a valid reason. Actually, you can achieve cost savings from more than one source. Perhaps your company's headquarters has a PBX, and that PBX connects to other PBXs or key systems at remote office locations. You might be paying a recurring monthly cost for the circuits interconnecting these privately owned phone switches. In addition to the circuits you have for your voice traffic, you might also have separate circuits for your data traffic between these offices. With VoIP, you could potentially eliminate the voice circuits, along with their monthly charges, and send your voice and data traffic over a single circuit.

As an example, consider the university where I used to work. We had a PBX at the main campus, and a key system at each of three remote campuses. These remote campuses were small, just some office space in strip malls located in surrounding communities. In fact, each remote campus only had about four telephones each. Still, we had a dedicated T1 circuit from the main campus's PBX to each of the three remote campuses' key systems. There were also separate T1 circuits connecting the main campus's data network to the data network at each remote campus. With a VoIP solution, the university could send both voice and data traffic over the existing data T1 circuits and eliminate the dedicated voice T1s.


A T1 is a digital circuit that sends traffic at a rate of 1.544 Mbps. T1s are often used to carry voice, video, or data traffic. A T1 circuit can be sub-divided into 24 separate channels, and PBXs require a full channel to support a voice path. Not all 24 channels need to be used for a single application (for example, voice, video, or data), however. A T1 can connect into a channel bank, which is a device that takes a T1 connection and breaks it out into 24 separate connections. If a full T1 is not required for a connection, many service providers sell fractional T1s, which provide a specific number of channels (less than 24).

Another cost savings offered by VoIP technologies can come in the form of cable plant expenses. Consider a company with separate infrastructures (that is, fiber optic cabling) for the voice, data, and video networks, as illustrated in Figure 1-8.

Figure 1-8. Voice/Video/Data Before Convergence

With the high-speed data networking technologies available in today's campus environment, voice, data, and even video traffic can peacefully coexist on the same high-speed network, as shown in Figure 1-9. This converged network approach requires less hardware because multiple traffic types use the same hardware.

Figure 1-9. Voice/Video/Data After Convergence

At this point, you can see that cost savings can be a major driving force in the migration to a VoIP network, and rightfully so. However, adopting a converged networking approach offers a number of other advantages. For example, a VoIP network not only mirrors the features offered by a PBX, but also offers a suite of new features.

Later chapters in this book describe many of these features, such as the ability to support converged messaging (that is, having a single repository for voice mail, fax messages, and e-mail), virtual call centers, and extension mobility (that is, the ability for a user to log into a phone and receive their personal telephone settings). You can even play video games on many of the Cisco IP Phones, although that's not one of the reasons you want to give management.

To be fair, however, consider a couple of common concerns that many people have with the idea of VoIP. One of the biggest concerns that often surfaces is the reliability of a VoIP telephony system. After all, PBXs have a reputation for being up and operational most of the time. In fact, many people in the PBX industry boast about the five nines of reliability that they enjoy with their PBX. The five nines of reliability means their PBX is up 99.999 percent of the time. If you considered that level of reliability over the period of a year, that would only be five minutes of downtime during the entire year! This is a measure of reliability not just during working hours, but rather 24 hours a day, 7 days a week, 365 days a year (and 366 in leap year!) of uptime. Sad to say, many corporate decision makers have a dimmer view of network reliability.


I experienced this skepticism personally. While working at a university, I approached my director and suggested that we might want to start considering the adoption of VoIP. My director became quite upset, pointing out that the PBX system "never" went down, whereas I was "always" doing something to the data network, bringing it to its knees. Many in management share this sentiment, and although it might have had some validity at one time, today's VoIP networks can offer comparable levels of reliability to PBXs, as you will learn in Chapter 3.

Another major concern is the quality of the voice calls. Most people have experienced a poor cell phone connection, and many fear a VoIP call might offer a similar level of quality. Voice quality is so critical, this book dedicates an entire chapter (Chapter 6, "Why Quality Matters") to addressing the variety of tools available to VoIP network designers to help them achieve fantastic voice quality.

These tools fall under a technology called quality of service (QoS). So, while there are definite concerns with VoIP, these concerns can certainly be addressed through proper VoIP network design, and later chapters explore many of these design best practices.

As discussed in this section, a VoIP network can help businesses today literally do more with less, enjoying more features and more flexibility with less recurring cost. Even though you haven't yet learned all the specific components that interconnect to form a VoIP network, you do have enough information at this point to start making some fundamental design decisions for a voice network. The next section puts your knowledge to the test.

Voice over IP First-Step
Voice over IP First-Step
ISBN: 1587201569
EAN: 2147483647
Year: 2005
Pages: 138
Authors: Kevin Wallace

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