Telephony specialists approach communications technology from a background shaped by the traditional telephone network, the Public Switched Telephone Network (PSTN). The telephone service provided by the PSTN is called plain old telephone service (POTS). This plain type of telephone network that everyone takes for granted uses circuit-switched connections, which means that when you make a call, you receive a dedicated circuit from one telephone to the other, through everything that is in the middle. The typical dedicated circuit through the PSTN has evolved from a physical connection to a logical connection that involves many switches. When you speak into a phone, a microphone creates an analog transmission that is passed along the circuit.
Decades of knowledge, experience, and innovation have allowed the PSTN to achieve the quality and reliability that it has today. When you pick up a phone, you get a dial tone almost instantly. And when you dial a number, the destination phone starts ringing, usually within a few seconds. Can you even recall the last time your traditional telephone call was dropped by the network? Because the PSTN is so reliable, people are rarely willing to tolerate reduced-quality or dropped calls, and their tolerance usually comes only with additional convenience, such as the convenience provided by mobile phones.
The level of reliability that is expected from the PSTN is sometimes referred to as five nines. This term means that the entire network must be available and functional for 99.999 percent of the time. If you apply this principle over the period of one year:
365 days * 24 hours/day * 60 minutes/hour * 0.00001 = 5.256 minutes
Five nines means that the network can be down for a grand total of less than six minutes over the course of a year!
An international organization that is now a part of the UN, the International Telecommunications Union (ITU) plays a major role in standardizing the technology of the PSTN. The ITU initially provided standards and agreements for connecting telegraph links between countries starting in the nineteenth century and has evolved to oversee many areas of standards development within the global telecom industry.
The ITU includes a specific division known as the Telecommunications Standardization Sector or ITU-T. This division comprises many companies and organizations with interests in telecommunications standards. After the ITU-T standards have been grouped into similar functional areas, they are called recommendations, and they share an assigned letter of the alphabet. The following ITU-T recommendations are most relevant to this discussion:
G Transmission systems and media, and digital systems and networks
H Audiovisual and multimedia systems
P Telephone transmission quality, telephone installations, and local line networks
The recommendation category letter is typically followed by a period and a number, such as G.711 or H.323. An ITU-T standard recommendation is said to be "In Force" when the standard has been approved by ITU-T membership.
Standards are crucial to the success of technologies like VoIP. Without standards, your phone call would very likely be dropped when it passed from vendor A's network to vendor B's network. Accordingly, many VoIP vendors have drawn on the expertise of the ITU-T and built VoIP products based on well-known standards.
How the PSTN Works
To talk about VoIP technology, it helps to understand a little about how the PSTN works today. The following is what has to happen when the caller makes a telephone call to the callee over the PSTN:
- 1. The caller picks up the telephone handset and hears a dial tone.
- 2. The caller enters a telephone number, specifying the address of the callee.
- 3. Signals are sent through the PSTN to set up a circuit for the call. The resources necessary to carry the call are reserved for it.
- 4. The destination phone rings, indicating to the callee that a call has arrived.
- 5. The callee picks up the telephone handset and begins a conversation. The audio, voice conversation is translated to digital format in the center of the network, and then back to analog at the edge.
- 6. The conversation ends, call billing occurs, the circuit is taken down, and resources are released.
Figure 1-1 shows the steps in a typical telephone call.
Figure 1-1. Six Steps in Typical Telephone Call
These steps must happen correctly and quickly for a telephone call to succeed with high quality. When telephony professionals consider providing the same functionality and reliability on relatively new and unreliable IP networks, you can see where some doubts and skepticism can arise.
Five components provide the infrastructure needed for fast and reliable calls on the PSTN. A brief introduction to these components will help you to understand what must be duplicated by VoIP technology to provide the same performance and reliability. Each of these five components is discussed in detail in the following sections:
When you speak into the mouthpiece of a telephone headset, your audio input is initially sent as an analog transmission over the telephone wiring. When the analog transmission reaches the entry point of the PSTN, it is digitized, or converted into digital formata series of 0s and 1s. After it has been digitized, the encoded voice transmission is transported across the PSTN to the far edge, where it is converted back again to an analog signal, and finally to sound.
The method for converting audio into digital form has been standardized. The name of this standard is G.711, and it uses an encoding technique called pulse code modulation (PCM). Within the G.711 standard, however, there are two varieties:
G.711u Also known as µ-law encoding (the Greek letter "mu"), this is used primarily in North America.
G.711a Also known as a-law encoding, this is used primarily outside North America.
G.711 converts analog audio input into digital output at an output rate of 64,000 bits per second, which is commonly referred to as 64 kilobits per second (kbps). A single G.711 voice channel is referred to as digital signal, level 0, or DS0. The fact that a DS0 takes up 64 kbps has been used in building links of the PSTN. Building a phone network link with a capacity for 24 voice channels takes 24 * 64 kbps = 1.536 megabits per second (Mbps). An additional 8 kbps is needed for framing overhead, which gives a total of 1.544 Mbps. A link with this capacity is known as a trunk level 1, or T1, link. Figure 1-2 shows the voice channels in a T1 link.
Figure 1-2. Voice Channels in the PSTN
You will encounter the G.711 standard again in the discussion of VoIP networks later in the chapter.
Switches are the core component of the PSTN. Switches of various types move call traffic from link to link and provide the circuits and dedicated connections necessary for PSTN calls. The links between switches are usually called trunk lines, and the capacity of trunk lines is usually stated in terms of the number of DS0 channels. Trunk lines use a technology called multiplexing to send multiple voice conversations over the same link.
PSTN switches are often categorized based on their function. However, switches that perform the same kinds of functions are often known by multiple names. If you think of connecting a phone in your house or in your company to the PSTN, the first point of entry is a switch called a local switch or local office. This type of switch is also known as a Class 5 switch. The local switch is usually operated by a local telephone company, which is often called a local exchange carrier (LEC). The local switch takes analog input from the phone connection and digitizes it for transmission through the center of the PSTN. The digitized conversation is sent over trunk lines to the next switch in the network.
The next type of switch the digital signal encounters is a tandem switch or tandem office. Tandem switches are usually operated by a long-distance company, or interexchange carrier (IXC). Tandem switches are connected to local switches or other tandem switches to provide a logical, circuit-switched path through the PSTN, and are sometimes called Class 1, 2, 3, or 4 switches. They carry massive call volumes and are designed to be very scalable and very reliable.
In VoIP systems, the IP router is analogous to the switches of the PSTN. Figure 1-3 shows different types of PSTN switches.
Figure 1-3. PSTN, Showing Local and Tandem Switches
A private branch exchange (PBX) is the foundation for most corporate voice networks. Typically, a corporate telephone network is different from a residential phone system. In a corporate environment, the network has to serve multiple users who need some advanced features, such as caller ID, call transfer, and call forwarding. In addition, the typical corporation wants its phone system to act as a single network, even if it serves offices in New York, Raleigh, and London.
Whereas residential telephone systems must allocate a separate external phone line for every user, the PBX enables corporate users to share a limited number of external telephone lines, providing cost savings to the company. The PBX also supports traditional telephone features like call waiting, call conferencing, and call forwarding. Many larger corporations connect PBXs together with "tie lines," which enable corporate users to make calls to co-workers without placing the calls on the PSTN at all. To dial another user over a tie line, you typically dial a different phone number, based on the tie line extension.
PBX systems come in all shapes and sizes. Smaller PBX systems, sometimes referred to as key systems, support limited numbers of users in smaller offices. Larger PBX systems may provide phone services for hundreds or thousands of users.
In VoIP systems, an IP PBX is analogous to the PBX of the PSTN, providing many of the same functions and features as a traditional PBX. Figure 1-4 shows an enterprise PBX connected to the PSTN.
Figure 1-4. PSTN, with an Enterprise Connected via a PBX
In some cases a business may outsource the functionality of the PBX and engage a service provider for a solution known as Centrex. With a Centrex solution, the service provider owns and operates all the equipment that is needed to provide call control and phone services. The provider maintains and manages the equipment on its premises, freeing the customer from the cost and management of PBX equipment. For VoIP systems, service providers are now offering IP Centrex, which extends the traditional Centrex model to include IP-based phone calls.
Like IP Centrex solutions, many traditional PBX systems are being retrofitted in other ways to support IP telephony. Service providers have offerings that add a gateway router to your network so that PBX-based calls can be transported on your IP network. A solution like this can provide an easy first step toward a VoIP implementation, as well as a cost savings, because the PSTN is bypassed for long-distance calls. To the user, the service appears transparent, because the PBX-connected phones have all the features and functions that they had before. There is a large investment in traditional PBX systems, so don't expect them to just go away overnight and be totally replaced by VoIP.
Establishing a telephone call requires several different types of signaling: to inform network devices that a telephone is off the hook, to supply destination information so that the call may be routed properly, and to notify both caller and callee that a call has been placed. A relatively new signaling technology, known as Signaling System 7 (SS7), is the ITU standard that provides for signaling, call setup, and management of PSTN calls. Typically, a separate network is used for SS7 flows. Because the data transfer for SS7 does not occur on the same path as the call, it is sometimes referred to as an out-of-band signal.
Two key components make up an SS7 network. The signal transfer point (STP) provides routing through the SS7 network. You could think of STPs as the IP routers of the SS7 network. The session control point (SCP) provides 800-number lookup and other management features. Similarly, Domain Name System (DNS) and Dymanic Host Control Protocol (DHCP) provide address lookup and management for IP networks.
When a phone call is made, the signaling protocols find the route to the callee, establish the connections between switches, and tear down these connections after the call ends. The STPs communicate with the local and tandem switches to reserve capacity between the switches in the path between caller and callee. After the call is completed, the STPs communicate with the switches to release the reserved connections, making them available for other calls. Figure 1-5 shows an SS7 network and signaling paths.
Figure 1-5. PSTN Uses SS7 Networks for Signaling
VoIP systems have a corresponding set of rules for call signaling. The section "Call Setup Protocols" later in this chapter introduces these.
Telephones that connect to the PSTN traditionally come in two flavors: analog and digital:
Analog telephone The type of phone that most people have in their homes today, it connects to the PSTN via traditional phone lines and sends an analog transmission (a waveform that varies over time).
Digital telephone The type of phone that many corporations use, it connects directly to a PBX and sends formatted digital signals.
Nowadays, specialized IP phones can connect to the PSTN as well. IP phones are discussed later in the chapter in the section "IP Phones and Softphones," which includes an explanation of how IP phone technology differs from traditional telephones.
This chapter has not mentioned cellular or mobile telephony technology. You can think of the mobile-phone network as an extension of the PSTNmost mobile calls are carried at least partially over the PSTN. The technology and components of the mobile-phone system are beyond the scope of this book.