VoIP

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

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

Delay
Overall or absolute delay can affect VoIP. You might have experienced delay in a telephone conversation with someone on a different continent. The delays can cause entire words in the conversation to be cut off, and can therefore be very frustrating.

When you design a network that transports voice over packet, frame, or cell infrastructures, it is important to understand and account for the predictable delay components in the network. You must also correctly account for all potential delays to ensure that overall network performance is acceptable. Overall voice quality is a function of many factors, including the compression algorithm, errors and frame loss, echo cancellation, and delay.

Figure 7-2 shows various sources and types of delay. Notice that there are two distinct types of delay:

• Fixed delay components are predictable and add directly to overall delay on the connection. Fixed delay components include the following:
  
  
• Coding—The time it takes to translate the audio signal into a digital signal
  
  
• Packetization—The time it takes to put digital voice information into packets and remove the information from packets
  
  
• Serialization—The insertion of bits onto a link
  
  
• Propagation—The time it takes a packet to traverse a link
  
  
• Variable delays arise from queuing delays in the egress trunk buffers that are located on the serial port connected to the WAN. These buffers create variable delays (that is, jitter) across the network.

Figure 7-2 Sources of Delay

Acceptable Delay
The ITU specifies network delay for voice applications in Recommendation G.114. This recommendation defines three bands of one-way delay, as shown in Table 7-1.

Table 7-1 Components and Services

Range in Milliseconds	Description
0 to 150	Acceptable for most user applications.
150 to 400	Acceptable, provided that administrators are aware of the transmission time and its impact on the transmission quality of user applications.
Above 400	Unacceptable for general network planning purposes; however it is recognized that in some exceptional cases, this limit will be exceeded.

NOTE
This recommendation is for connections where echo is adequately controlled, implying that echo cancellers are used. Echo cancellers are required when one-way delay exceeds 25 ms (G.131).

This G.114 recommendation is oriented toward national telecommunications administrations and therefore is more stringent than recommendations that would normally be applied in private voice networks. When the location and business needs of end users are well known to a network designer, more delay might prove acceptable. For private networks, a 200 ms delay is a reasonable goal and a 250 ms delay is a limit. This goal is what Cisco proposes as reasonable, as long as excessive jitter does not impact voice quality. However, all networks must be engineered so that the maximum expected voice connection delay is known and minimized.

The G.114 recommendation is for one-way delay only and does not account for round-trip delay. Network design engineers must consider both variable and fixed delays in their design. Variable delays include queuing and network delays, while fixed delays include coding, packetization, serialization, and dejitter buffer delays. Table 7-2 provides an example of a delay budget calculation.

Table 7-2 Sample Delay Budget

Delay Type	Fixed (ms)	Variable (ms)
Coder delay	18	N/A
Packetization delay	30	N/A
Queuing and buffering	N/A	8
Serialization (64 kbps)	5	N/A
Network delay (through public network)	40	25
Dejitter buffer	45	N/A
Totals	138	33

Packet Loss
Lost data packets, as depicted in Figure 7-3 are recoverable if the endpoints can request retransmission. However, lost voice packets are not recoverable, because the audio must be played out in real time and retransmission is not an option.

Figure 7-3 Packet Loss

Voice packets might be dropped under the following conditions:

• The network is unstable (flapping links).
• The network is congested.
• There is too much variable delay in the network.

Packet loss causes voice clipping and skips. As a result, the listener hears gaps in the conversation. The industry-standard coder-decoder (CODEC) algorithms used in Cisco DSPs correct for 20 to 50 ms of lost voice through the use of Packet Loss Concealment (PLC) algorithms. PLC intelligently analyzes missing packets and generates a reasonable replacement packet to improve the voice quality. Cisco VoIP technology uses 20 ms samples of voice payload per VoIP packet by default. Effective CODEC correction algorithms require that only a single packet can be lost at any given time. If more packets are lost, the listener experiences gaps.If a conversation experiences packet loss, the effect is immediately heard. If the talker says, "Watson, come here. I want you," the listener might hear, "Wat...., come here, ......you."

—End

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.

Visit www.ciscopress.com for a detailed description and to learn how to purchase this title.

< Back to page one

Go to part two: Metrics >