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Week 8: Quality of Service Part 1

QoS Overview
Objectives
After completing this course, students will be able to
 Explain the purpose and characteristics of QoS.
 Explain how network transmission characteristics impact quality.
 Describe minimum network requirements for voice, video, and data traffic.
 Describe the queuing algorithms used by networking devices.
 Explain how networking devices implement QoS.
 Describe the different QoS models.
 Explain how QoS uses mechanisms to ensure transmission quality.

Introduction

In today's networks, users expect content to be immediately available. But if the traffic
exceeds the bandwidth of the links between the source of the content and the user, how do
network administrators ensure a quality experience? Quality of Service (QoS) tools can be
designed into the network to guarantee that certain traffic types, such as voice and video, are
prioritized over traffic that is not as time-sensitive, such as email and web browsing.

This chapter describes network transmission quality, traffic characteristics, queueing

algorithms, QoS models, and QoS implementation techniques.

Network transmission Quality

Prioritizing Traffic
Quality of Service (QoS) is an ever increasing requirement of networks today. New
applications available to users, such as voice and live video transmissions, create higher
expectations for quality delivery.

Congestion occurs when multiple communication lines aggregate onto a single device such
as a router, and then much of that data is placed on fewer outbound interfaces, or onto a
slower interface. Congestion can also occur when large data packets prevent smaller packets
from being transmitted in a timely manner.

When the volume of traffic is greater than what can be transported across the network,
devices queue, or hold, the packets in memory until resources become available to transmit
them. Queuing packets causes delay because new packets cannot be transmitted until
previous packets have been processed. If the number of packets to be queued continues to
increase, the memory within the device fills up and packets are dropped. One QoS technique
that can help with this problem is to classify data into multiple queues, as shown in the figure.

Note: A device implements QoS only when it is experiencing some type of congestion.

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Bandwidth, Congestion, Delay, and Jitter

Network bandwidth is measured in the number of bits that can be transmitted in a single
second, or bits per second (bps). For example, a network device may be described as having
the capability to perform at 10 gigabits per second (Gbps).
Network congestion causes delay. An interface experiences congestion when it is presented
with more traffic than it can handle. Network congestion points are strong candidates for
QoS mechanisms. Figure 8.1 shows three examples of typical congestion points.

Figure 8.1 Examples of Congestion points

Delay or latency refers to the time it takes for a packet to travel from the source to the
destination. Two types of delays are fixed and variable. A fixed delay is a specific amount of
time a specific process takes, such as how long it takes to place a bit on the transmission
media. A variable delay take an unspecified amount of time and is affected by factors such as
how much traffic is being processed.
The sources of delay are summarized in table in Figure 8.2.

Figure 8.2 Source of Delay

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Jitter is the variation in the delay of received packets. At the sending side, packets are sent in
a continuous stream with the packets spaced evenly apart. Due to network congestion,
improper queuing, or configuration errors, the delay between each packet can vary instead
of remaining constant. Both delay and jitter need to be controlled and minimized to support
real-time and interactive traffic.

Packet Loss
Without any QoS mechanisms in place, packets are processed in the order in which they are
received. When congestion occurs, network devices such as routers and switches can drop
packets. This means that time-sensitive packets, such as real-time video and voice, will be
dropped with the same frequency as data that is not time-sensitive, such as email and web
browsing.

For example, when a router receives a Real-Time Protocol (RTP) digital audio stream for
Voice over IP (VoIP), it must compensate for the jitter that is encountered. The mechanism
that handles this function is the playout delay buffer. The playout delay buffer must buffer
these packets and then play them out in a steady stream as shown in Figure 8.3. The digital
packets are later converted back to an analog audio stream.

Figure 8.3 playout Delay Buffer Compensate for jitter

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If the jitter is so large that it causes packets to be received out of the range of this buffer, the
out-of-range packets are discarded and dropouts are heard in the audio, as shown in Figure
8.4.

Figure 8.4 Packet Dropped Due to Excessive jitter

For losses as small as one packet, the digital signal processor (DSP) interpolates what it
thinks the audio should be and no problem is audible to the user. However, when jitter
exceeds what the DSP can do to make up for the missing packets, audio problems are heard.

Packet loss is a very common cause of voice quality problems on an IP network. In a properly
designed network, packet loss should be near zero. The voice codecs used by the DSP can
tolerate some degree of packet loss without a dramatic effect on voice quality. Network
engineers use QoS mechanisms to classify voice packets for zero packet loss. Bandwidth is
guaranteed for the voice calls by giving priority to voice traffic over traffic that is not time-
sensitive.

Traffic Characteristics
Network Traffic Trends
In the early 2000s, the predominant types of IP traffic were voice and data. Voice traffic has
a predictable bandwidth need and known packet arrival times. Data traffic is not real-time
and has unpredictable bandwidth need. Data traffic can temporarily burst, as when a large
file is being downloaded. This bursting can consume the entire bandwidth of a link.

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More recently, video traffic has become the increasingly important to business
communications and operations. According to the Cisco Visual Networking Index (VNI),
video traffic represented 67% of all traffic in 2014. By 2019, video will represent 80% of all
traffic. In addition, mobile video traffic will increase over 600% from 113,672 TB to 768,334
TB.

The type of demands voice, video, and data traffic place on the network are very different.

Voice
Voice traffic is predictable and smooth, as shown in the figure 8.5. However, voice is very
sensitive to delays and dropped packets; there is no reason to re-transmit voice if packets
are lost. Therefore, voice packets must receive a higher priority than other types of traffic.
For example, Cisco products use the RTP port range 16384 to 32767 to prioritize voice
traffic. Voice can tolerate a certain amount of latency, jitter, and loss without any noticeable
effects. Latency should be no more than 150 milliseconds (ms). Jitter should be no more than
30 ms, and voice packet loss should be no more than 1%. Voice traffic requires at least 30
Kbps of bandwidth.

Figure 8.5 Voice Traffic Characteristics

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Video
Without QoS and a significant amount of extra bandwidth capacity, video quality typically
degrades. The picture appears blurry, jagged, or in slow motion. The audio portion of the
feed may become unsynchronized with the video.

Video traffic tends to be unpredictable, inconsistent, and bursty compared to voice traffic.
Compared to voice, video is less resilient to loss and has a higher volume of data per packet,
as shown in Figure 8.6. Notice how voice packets arrive every 20 ms and are a predictable
200 bytes each. In contrast, the number and size of video packets varies every 33 ms based
on the content of the video. For example, if the video stream consists of content that is not
changing much from frame to frame, then the video packets will be small and fewer are
required to maintain acceptable user experience. However, if the video steam consists of
content that is rapidly changing, such as in an action sequence in a movie, then the video
packets will be larger and more are required per 33 ms time slot to maintain an acceptable
user experience.

Figure 8.6 Voice and Video Sampling Comparison

Figure 8.7 summarizes the characteristics of video traffic. UDP ports, such as 554 used for
the Real-Time Streaming Protocol (RSTP), should be given priority over other, less time-
sensitive, network traffic. Similar to voice, video can tolerate a certain amount of latency,
jitter, and loss without any noticeable affects. Latency should be no more than 400
milliseconds (ms). Jitter should be no more than 50 ms, and video packet loss should be no
more than 1%. Video traffic requires at least 384 Kbps of bandwidth.

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Figure 8.7 Video traffic Characteristics

Data
Most applications use either TCP or UDP. Unlike UDP, TCP performs error recovery. Data
applications that have no tolerance for data loss, such as email and web pages, use TCP to
ensure that, if packets are lost in transit, they will be resent. Data traffic can be smooth or
bursty. Network control traffic is usually smooth and predictable. When there is a topology
change, the network control traffic may burst for a few seconds. But the capacity of today’s
networks can easily handle the increase in network control traffic as the network converges.

However, some TCP applications can be very greedy, consuming a large portion of network
capacity. FTP will consume as much bandwidth as it can get when you download a large file,
such as a movie or game. Figure 8.8 summarizes data traffic characteristics.

Figure 8.8 Data traffic Characteristics

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Although data traffic is relatively insensitive to drops and delays compared to voice and
video, a network administrator still needs to consider the quality of the user experience,
sometimes referred to as Quality of Experience or QoE. The two main factors a network
administrator needs to ask about the flow of data traffic are the following:

Does the data come from an interactive application?

Is the data mission critical?
Figure 8.9 is compares these two factors.

Figure 8.9 Factors to Consider for data entry

Queuing Algorithms
Queuing Overview
The QoS policy implemented by the network administrator becomes active when congestion
occurs on the link. Queuing is a congestion management tool that can buffer, prioritize, and,
if required, reorder packets before being transmitted to the destination. A number of
queuing algorithms are available. For the purposes of this course, we will focus on the
following:

First-In, First-Out (FIFO)

Weighted Fair Queuing (WFQ)
Class-Based Weighted Fair Queuing (CBWFQ)
Low Latency Queuing (LLQ)
First In First Out (FIFO)
In its simplest form, FIFO queuing, also known as first-come, first-served (FCFS) queuing,
involves buffering and forwarding of packets in the order of arrival.

FIFO has no concept of priority or classes of traffic and consequently, makes no decision
about packet priority. There is only one queue, and all packets are treated equally. Packets
are sent out an interface in the order in which they arrive, as shown in the figure 8.10.
Although some traffic is more important or time-sensitive based on the priority
classification, notice that the traffic is sent out in the order it is received.

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.
Figure 8.10 FIFO Queuing Example

When FIFO is used, important or time-sensitive traffic can be dropped when congestion
occurs on the router or switch interface. When no other queuing strategies are configured,
all interfaces except serial interfaces at E1 (2.048 Mbps) and below use FIFO by default.
(Serial interfaces at E1 and below use WFQ by default.)

FIFO, which is the fastest method of queuing, is effective for large links that have little delay
and minimal congestion. If your link has very little congestion, FIFO queuing may be the only
queuing you need to use.

Weighted Fair Queuing (WFQ)

Figure 8.11 Weighted Fair Queuing Example

WFQ is an automated scheduling method that provides fair bandwidth allocation to all
network traffic. WFQ applies priority, or weights, to identified traffic and classifies it into
conversations or flows, as shown in the figure 8.11.

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WFQ then determines how much bandwidth each flow is allowed relative to other flows. The
flow-based algorithm used by WFQ simultaneously schedules interactive traffic to the front
of a queue to reduce response time. It then fairly shares the remaining bandwidth among
high-bandwidth flows. WFQ allows you to give low-volume, interactive traffic, such as Telnet
sessions and voice, priority over high-volume traffic, such as FTP sessions. When multiple
file transfers flows are occurring simultaneously, the transfers are given comparable
bandwidth.

WFQ classifies traffic into different flows based on packet header addressing, including such
characteristics as source and destination IP addresses, MAC addresses, port numbers,
protocol, and Type of Service (ToS) value. The ToS value in the IP header can be used to
classify traffic. ToS will be discussed later in the chapter.

Low-bandwidth traffic streams, which comprise the majority of traffic, receive preferential
service, allowing their entire offered loads to be sent in a timely fashion. High-volume traffic
streams share the remaining capacity proportionally among themselves.

Limitations

WFQ is not supported with tunneling and encryption because these features modify the
packet content information required by WFQ for classification.

Although WFQ automatically adapts to changing network traffic conditions, it does not offer
the degree of precision control over bandwidth allocation that CBWFQ offers.

Class-Based Weighted Fair Queuing (CBWFQ)

CBWFQ extends the standard WFQ functionality to provide support for user-defined traffic
classes. For CBWFQ, you define traffic classes based on match criteria including protocols,
access control lists (ACLs), and input interfaces. Packets satisfying the match criteria for a
class constitute the traffic for that class. A FIFO queue is reserved for each class, and traffic
belonging to a class is directed to the queue for that class, as shown in the figure 8.12.

Figure 8.12 CBWFQ Example

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When a class has been defined according to its match criteria, you can assign it
characteristics. To characterize a class, you assign it bandwidth, weight, and maximum
packet limit. The bandwidth assigned to a class is the guaranteed bandwidth delivered to the
class during congestion.

To characterize a class, you also specify the queue limit for that class, which is the maximum
number of packets allowed to accumulate in the queue for the class. Packets belonging to a
class are subject to the bandwidth and queue limits that characterize the class.

After a queue has reached its configured queue limit, adding more packets to the class causes
tail drop or packet drop to take effect, depending on how class policy is configured. Tail drop
means a router simply discards any packet that arrives at the tail end of a queue that has
completely used up its packet-holding resources. This is the default queuing response to
congestion. Tail drop treats all traffic equally and does not differentiate between classes of
service.
Low Latency Queuing (LLQ)
The LLQ feature brings strict priority queuing (PQ) to CBWFQ. Strict PQ allows delay-
sensitive data such as voice to be sent before packets in other queues. LLQ provides strict
priority queuing for CBWFQ, reducing jitter in voice conversations, as shown in the figure
8.13.

Figure 8.12 LLQ Example

Without LLQ, CBWFQ provides WFQ based on defined classes with no strict priority queue
available for real-time traffic. The weight for a packet belonging to a specific class is derived
from the bandwidth you assigned to the class when you configured it. Therefore, the
bandwidth assigned to the packets of a class determines the order in which packets are sent.
All packets are serviced fairly based on weight; no class of packets may be granted strict
priority. This scheme poses problems for voice traffic that is largely intolerant of delay,
especially variation in delay. For voice traffic, variations in delay introduce irregularities of
transmission manifesting as jitter in the heard conversation.

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With LLQ, delay-sensitive data is sent first, before packets in other queues are treated. LLQ
allows delay-sensitive data such as voice to be sent first (before packets in other queues),
giving delay-sensitive data preferential treatment over other traffic. Although it is possible
to enqueue various types of real-time traffic to the strict priority queue, Cisco recommends
that only voice traffic be directed to the priority queue.

References and Supplementary Materials

Books and Journals
1. Bob Vachon and Allan Johnson; 2018; Connecting Networks v6 Companion Guide; 800
East 96th Street Indianapolis, IN 46240 USA; Cisco Press.
2. Rick Graziani and Allan Johnson; 2017; Introduction to Networks v6 Companion
Guide; 800 East 96th Street Indianapolis, IN 46240 USA; Cisco Press.

Online Supplementary Reading Materials

1. CCNA Routing and Switching: Connecting Networks; www.netacad.com; Oct 14, 2019

Online Instructional Videos

1. CISCO CCNA 4 CONNECTING NETWORKS;
https://www.youtube.com/watch?v=weOirQq27xE&list=PL452256E1D4CDA875;
Oct 14, 2019

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