VoIP for Dummies 2005

Note that the transport layer of the packet uses the familiar TCP protocol to construct the packet exchanges between the source computer and the destination computer (in this case, the Web server).

Figure 1-2 illustrates the layered protocol stack shown in Figure 1-1, but this time applied to a VoIP call. Note the header at each layer’s version of the packet (except the application layer). The various headers identify what layer the packet is on during its travel from caller to receiver.

If you compare Figures 1-1 and 1-2, you notice two protocol differences, at the application and transport layers of the TCP/IP model. Other than these differences, a great deal of symmetry exists between voice and data using TCP/IP.

That’s why VoIP can behave like a telephone while delivering many com- puter-related functions.

Application layer differences

The first difference between the VoIP implementation of TCP/IP and the traditional data implementation is in the application layer. In a VoIP call, the application layer utilizes the following three protocols:

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NTP: Network time protocol. This protocol enables timing, which helps ensure that the signals are transmitted and received within the proper timeframe to assure quality.

RTP: Real-time transport protocol. This protocol provides end-to-end network transport functions for digital voice signals encapsulated in the VoIP packet.

RTCP: Real-time transport control protocol. This protocol monitors voice signal delivery and provides minimal control functions to ensure the delivery of packets.

All three of the application layer protocols combine, at nanosecond speeds, to deliver VoIP voice packets.

Transport layer differences

The second difference between the traditional data implementation of TCP/IP and the VoIP implementation is in the transport layer. The lion’s share of computer data networking uses the TCP protocol at the transport layer. For VoIP, the transport layer uses UDP, user datagram protocol. (UDP is used also for real-time videoconferencing networks.)

VoIP Network

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TCP is slower than UDP, but it provides guaranteed delivery of its computer data packets. Keep in mind that we are measuring speed here in nanoseconds. Even if it takes a long time for the packets to reach their destination computer, eventually TCP ensures delivery.

Because voice is a real-time application, it is more important that the voice packets get to the receiver as quickly as possible. That is why UDP is by far the hands-down favorite to provide the transport layer for VoIP networks.

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could be carried electronically over a pair of wires. He also demonstrated that the endpoints for these wires had to be connected to the right equipment — hardware that he invented. Mr. Bell’s inventions ushered in an age of communication that made the world much smaller than it had ever been before.

When Mr. Bell invented the telephone and thereby gave birth to the telephone network, VoIP was not even a consideration. (Truth be told, the idea of a network wasn’t yet a consideration either.) Other inventions would be required before VoIP could become a reality.

The first telephone equipment was analog. Historians and technicians alike have labeled the first phone service POTS, or plain old telephone service. VoIP won’t function very well over a POTS system; it requires a digital network.

Digital networking for telephones was invented in the 1920s, but the first digital networks would not leave the laboratory until much later, in 1964. Today, most phone companies in the United States have updated their equipment to include digital service.

Over time, the POTS network gave way to the PSTN, or public switched telephone network. (The terms PSTN, public switched telephone network, public telephone network, and phone network are used synonymously.)

Although it occurred in what seems like the ancient past, Alexander Graham Bell’s work is important in understanding VoIP. The POTS network that began with his invention has grown into the largest circuit-switched network in the world. It also has become an expensive network, with individuals and companies spending hundreds of billions of dollars each year for communication services.

VoIP, which was developed in 1995, is gradually replacing the PSTN. Some view the PSTN as the antithesis of VoIP, but it still remains the standard of quality by which VoIP is measured. For instance, people often ask whether VoIP provides voice quality as good as what is delivered through the PSTN. Most of the factors used to evaluate the quality of VoIP are based in some way on the PSTN, so understanding a bit about the older networks is important.

Analog Telephone Circuits

As mentioned, phone technology originally was analog, from start to finish. Analog modulation is the technique used to convert sounds (such as your voice) into an electromagnetic form. The analog circuitry of the POTS telephone transmitter converts the voice patterns coming from the caller’s mouth into continuous electromagnetic signal patterns. These patterns are carried on a telephone line circuit, sometimes called a trunk line, where they are

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Combining Analog and Digital

When digital networks were introduced, the phone companies wanted to use them right away because they provided a more efficient means of transmitting signals all over the place. (Digital networks could carry data much faster than analog networks.) The phone companies were presented with a problem, however: how to make existing analog phones work with a digital network.

The answer was to use a codec to convert the analog signal to digital. But where should the conversion take place? At the phone company’s facilities or at their customer’s location? In the early years of the digital revolution, the conversion took place at the phone company’s facilities, which allowed the phone company to utilize the existing wiring between their facilities and the customer’s location. This wiring between a phone company facility and a customer is often called a local loop.

Over the years, the codec has been pushed closer and closer to the customer, all in an effort to make the phone network as close to 100 percent digital as possible. Most parts of the PSTN remain a combination of analog and digital. Customers pick up a phone, which converts audible sound into analog signals. These signals are carried over the local loop to the phone carrier’s facilities, where they are converted to a digital signal. The signals are forwarded to the receiver’s end. After the signals are received by the last piece of carrier equipment (closest to the end customer), they are converted back to analog form. The analog signals go into the receiver side of the POTS telephone and are heard as a replica of the caller’s voice. Figure 2-1 illustrates how a phone call is transmitted over the PSTN.

Calling Pennsylvania 6-5000

Today, a customer can have either an analog or a digital phone. Depending on customer location and end-user equipment, the POTS call can be 100 percent digital.

For more than a century, POTS remained the dominant form of two-way telecommunications. During that time, however, POTS-based telephone systems changed dramatically in the number, length, diameter, and type of wire or cables used and in the types of telephone equipment both at the customer end and at the carrier’s facilities.

Digital Telephony Invades PSTN Territory

When digital networks were implemented back in the 1960s, the telephone carrier companies began using a technique that permitted them to accept analog telephone calls coming into their switching facilities and convert those signals into digital form for transmission on their shiny new networks. They had not yet made the leap into packetizing telephone calls, which is what we have today with VoIP. At the time, they thought it best to keep the circuit-switched telephone carrier network physically separate from the evolving packet-switched computer network.

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The phone companies were able to make other improvements to circuitswitched telephone services. After their circuit-switched carrier network received the caller’s telephony signals, they were able to convert the signals into digital form, as necessary. They discovered that digital signals allowed them to aggregate many more calls onto a given circuit and through a given switch than they could before. This enabled them to streamline how circuitswitched telephone calls could be made.

One innovation was the addition of area codes, which help to process calls over a circuit-switched network. The entire telephone number, including area code, identifies the number of circuits and the location of the switching devices for a given phone on the PSTN network. For example, consider a call originating in Mountain View, California (area code 415) to a person in Pittsburgh, Pennsylvania (area code 412). The call is switched out over three physically distinct circuit switches — 415 to 412 — to set up and carry the call. Figure 2-2 illustrates the routing of such a circuit-switched call.

If these two locations are on the same computer network, and both are using VoIP, the call could be carried over the company’s computer network in packet form. This process is known as on-net VoIP telephony. None of the packets would touch the PSTN. There would be no toll, regulatory, or metered charges for this long-distance telephone call. Figure 2-3 illustrates the routing of such a packet-switched VoIP call.