VoIP

Book Excerpt:
Authorized Self-Study Guide: Cisco Voice over IP (CVoice), Second Edition

In this book excerpt, the author describes the factors that determine voice quality on the internet.

by Kevin Wallace in Authorized Self-Study Guide, Cisco Voice over IP (CVoice), Second Edition, published by Cisco Press [October 23, 2006]

This is excerpt is from Chapter 7: Improving and Maintaining Voice Quality, pp. 373-381 of Authorized Self-Study Guide, Cisco Voice over IP (CVoice), Second Edition, by Cisco Press

When human speech is converted to analog electrical signals and then digitized and compressed, some of the qualitative components are lost. This chapter explores the components of voice quality that you must maintain, the methods that you can use to measure voice quality, and quality of service (QoS) tools that you can implement in a network to improve voice quality.

Optimizing Voice Quality

Because of the inherent characteristics of a converged voice and data IP network, administrators face certain challenges in delivering voice traffic correctly. This section describes these challenges and offers solutions for avoiding and overcoming them when designing a VoIP network for optimal voice quality.

Factors that Affect Voice Quality

Because of the nature of IP networking, voice packets sent via IP are subject to certain transmission problems. Conditions present in the network might introduce problems such as echo, jitter, or delay. These problems must be addressed with QoS mechanisms.

The clarity, or cleanliness and crispness, of the audio signal is of utmost importance. The listener must be able to recognize the speaker's identity and sense the mood of the speaker. These factors can affect clarity:

• Fidelity—The degree to which a system, or a portion of a system, accurately reproduces, at its output, the essential characteristics of the signal impressed upon its input or the result of a prescribed operation on the signal impressed upon its input (definition from the Alliance for Telecommunications Industry Solutions [ATIS]). The bandwidth of the transmission medium almost always limits the total bandwidth of the spoken voice. Human speech typically requires a bandwidth from 100 to 10,000 Hz, although 90 percent of speech intelligence is contained between 100 and 3000 Hz.
  
  
• Echo—A result of electrical impedance mismatches in the transmission path. Echo is always present, even in traditional telephony networks, but at a level that cannot be detected by the human ear. The two components that affect echo are amplitude (that is, loudness of the echo) and delay (that is, the time between the spoken voice and the echoed sound). You can control echo using echo suppressors or echo cancellers.
  
  
• Jitter—Variation in the arrival of coded speech packets at the far end of a VoIP network. The varying arrival time of the packets can cause gaps in the re-creation and playback of the voice signal. These gaps are undesirable and annoy the listener. Delay is induced in the network by variation in the routes of individual packets, contention, or congestion. You can often resolve variable delay by using dejitter buffers.
  
  
• Packet drops—The discarding of voice packets. Typically, when a VoIP packet is dropped from a network, 20 ms of audio is lost.
  
  
• Delay—The time between the spoken voice and the arrival of the electronically delivered voice at the far end. Delay results from multiple factors, including distance (that is, propagation delay), coding, compression, serialization, and buffering.
  
  
• Sidetone—The purposeful design of the telephone that allows the speaker to hear the spoken audio in the earpiece. Without sidetone, the speaker is left with the impression that the telephone instrument is not working.
  
  
• Background noise—The low-volume audio that is heard from the far-end connection. Certain bandwidth-saving technologies can eliminate background noise altogether, such as voice activity detection (VAD). When this technology is implemented, the speaker audio path is open to the listener, while the listener audio path is closed to the speaker. The effect of VAD is often that speakers think that the connection is broken, because they hear nothing from the other end.

Although each of the preceding factors affects audio clarity, factors that present the greatest challenges to VoIP networks include jitter, delay, and packet drops. A lack of network bandwidth is usually the underlying cause for these issues, which are addressed in the following sections.

Jitter
Jitter is defined as a variation in the delay of received packets, as illustrated in Figure 7-1. On the sending side, packets are sent in a continuous stream with the packets spaced evenly. Because of network congestion, improper queuing, or configuration errors, this steady stream can become uneven, because the delay between each packet varies instead of remaining constant.

When a router receives a VoIP audio stream, it must compensate for the jitter that is encountered. The mechanism that handles this function is the playout delay buffer, or dejitter buffer. The playout delay buffer must buffer these packets and then play them out in a steady stream to the digital signal processors (DSPs) to be converted back to an analog audio stream. The playout delay buffer, however, affects the overall absolute delay.When a conversation is subjected to jitter, the results can be clearly heard. If the talker says, "Watson, come here. I want you," the listener might hear "Wat....s...on.......come here, I......wa......nt........y......ou." The variable arrival of the packets at the receiving end causes the speech to be delayed and garbled.

Figure 7-1 Jitter in IP Networks

Reproduced from the book Authorized Self-Study Guide, Cisco Voice over IP (CVoice), Second Edition. Copyright 2006, Cisco Systems, Inc.. Reproduced by permission of Pearson Education, Inc., 800 East 96th Street, Indianapolis, IN 46240.

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