What Is Voice Transport?

A converged network is one in which data, voice, and video traffic coexist on a single network. Transporting voice and video across the network means that they become applications, just like file transfers, to the network. It also means that the capabilities of the network can be used to provide even more functionality and features than were previously available.

Key Point

To transport voice across the network, it must first be digitizedconverted from analog to digital signalsand then packetized, that is, put into IP packets. These voice IP packets can then be sent over the data network, just like any other IP packets.

Digitization

Figure 7-1 illustrates the digitization process for analog speech, as described in this section.

Figure 7-1. Converting Analog Speech into Digital Signals

Analog speech contains components at many different frequencies, representing the range of sounds that we hear. Most analog speech components are in the frequency range of 300 hertz (Hz) to 3400 Hz. The first step in the digitization process is to filter out anything higher than 4000 Hz, thus isolating the speech component.

This filtered signal is then sampled by a process called pulse amplitude modulation (PAM), which uses the original analog signal to change the amplitude of a pulse signal. The rate at which the sampling is done is driven by the Nyquist theorem, which states that the sampling rate must be at least twice the highest frequency, so it is possible to reconstruct the analog signal from the digital signal. If too few samples are taken, not enough information would be available to re-create the original speech. Thus, for speech filtered at 4000 Hz, sampling must be done at 8000 Hz, or 8000 times per second.

Using a pulse code modulation (PCM) process, each of these samples is then quantized, which means that it is assigned a discrete binary value and then digitized. Eight bits are used for each sample, resulting in 2⁸ = 256 possible values.

Key Point

Because the original analog signal is sampled at 8000 times per second and each sample is represented by 8 bits, voice digitized using PCM is sent at a rate of 8000 * 8 = 64,000 bits per second (bps), or 64 kilobits per second (kbps).

Packetization and Call Processing

The packetization of voice is implemented in the following two ways, as illustrated in Figure 7-2:

• Using traditional phones and a PBX (or digital phones attached to a PBX) to digitize the voice, and then connecting the PBX to a voice-enabled router to perform the packetization. The result is voice encapsulated inside IP packets, or VoIP. These packets are carried across the converged network, therefore replacing the traditional tie trunks between PBXs. This scenario is illustrated at the top of Figure 7-2, and the devices required are detailed in the "VoIP Components" section, later in this chapter.

  Note

  A PBX is a telephone switch used within an organization to provide features such as call holding, call forwarding, conference calling, and voice mail.

  
  

• Using IP phones to digitize and packetize the voice. The call-processing function previously performed by the PBX is now handled by a call-processing managerfor example, Cisco CallManager (CCM) is a software-based system that provides functions such as setting up and terminating calls, routing to voice mail, and so forth. Similar to the previous scenario, this results in VoIP packets traversing the network. VoIP, together with the enhanced features provided by CCM and other applications that are now possible, are collectively known as IP telephony. This scenario is illustrated in the lower portion of Figure 7-2. IP telephony also supports devices other than IP phones, to provide even more flexibility and functionality within the converged network. The devices used in IP telephony are described in the "IP Telephony Components" section, later in this chapter.

Figure 7-2. VoIP and IP Telephony

Note

Figure 7-2 does not show connectivity to the PSTN. IP telephony scenarios with PSTN connections are described in the "IP Telephony Design" section, later in this chapter.

Conversation and Control Traffic

Two categories of voice traffic exist: conversation traffic (the audio, also called bearer traffic) and control (or signaling) traffic.

Key Point

Within VoIP, conversation packets are sent using the User Datagram Protocol (UDP), which provides connectionless transmission.

Because it does not have the overhead that the sequencing, acknowledging, and error-checking features of the Transmission Control Protocol (TCP) require, UDP provides a more efficient, lower-delay service. Voice conversation is susceptible to delayif a voice packet is delayed too much, it could lose its relevance. On the other hand, the loss of a single voice packet is not detrimental to the quality of voice at the receiving end because it can be interpolated from other voice samples.

Conversation packets are sent using the Real-Time Transport Protocol (RTP), which runs on top of UDP. RTP was designed to be used for real-time traffic such as voice. RTP adds another header to the UDP segment that includes some sequencing information and time-stamping information to ensure that the received data is processed in the correct order and that the variation in the delay is within acceptable limits.

Call control traffic is sent using TCP, because the signals must be received in the order in which they were sent, and loss of these packets cannot be tolerated.