Slides 3.1 Introducing QOS

stagetofuAI and Robotics

Oct 29, 2013 (3 years and 5 months ago)


BSc Systems 4

Quality of Service in Converged Networks

Notes on Slides 3.1 & 3.2


Slides 3.1

Introducing QOS

Before converged networks were common, network engineering focused on connectivity.
The rates at which data came onto the network resulted in bursty

data flows. In a traditional
network, data, arriving in packets, tries to acquire and use as much bandwidth as possible at
any given time. Access to bandwidth is on a first
come, first
served (FIFO) basis. The data
rate available to any one user varies de
pending on the number of users accessing the
network at that time.

Protocols in nonconverged traditional networks handle the bursty nature of data networks.
Data networks can survive brief outages. For example, when you retrieve e
mail, a delay of a
few se
conds is generally not noticeable. A delay of minutes is annoying, but not serious.

Traditional networks also had requirements for applications such as data, video, and systems
network architecture (SNA). Since each application has different traffic charac
teristics and
requirements, network designers deployed nonintegrated networks. These nonintegrated
networks carried specific types of traffic: data network, SNA network, voice network, and
video network.

A converged network carries voice, video, and data t
raffic. These flows use the same
network facilities. Merging these different traffic streams with dramatically differing
requirements can lead to a number of problems. Key among these problems is that voice
and video traffic is very time
sensitive and must

get priority.

In a converged network, constant, small
packet voice flows compete with bursty data flows.
Although the packets carrying voice traffic on a converged network are typically very small,
the packets cannot tolerate delay and delay variation as

they traverse the network. When
delay and delay variations occur, voices break up and words become incomprehensible.

Conversely, packets carrying file transfer data are typically large and the nature of IP lets the
packets survive delays and drops. It is

possible to retransmit part of a dropped data file, but
it is not feasible to retransmit part of a voice conversation. Critical voice and video traffic
must have priority over data traffic. Mechanisms must be in place to provide this priority.

The key rea
lity in converged networks is that service providers cannot accept failure. While a
file transfer or an e
mail packet can wait until a down network recovers and delays are
almost transparent, voice and video packets cannot wait. Converged networks must pro
secure, predictable, measurable, and, sometimes, guaranteed services. Even a brief network
outage on a converged network seriously disrupts business operations.

Network administrators and architects achieve required performance from the network by
naging delay, delay variation (jitter), bandwidth provisioning, and packet loss parameters
with quality of service (QoS) techniques.

Multimedia streams, such as those used in IP telephony or videoconferencing, are very sensitive to
delivery delays and cre
ate unique QoS demands. If service providers rely on a best
effort network
model, packets may not arrive in order, in a timely manner, or maybe not at all. The result is unclear
pictures, jerky and slow movement, and sound that is not synchronized with ima

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With inadequate network configuration, voice transmission is irregular or unintelligible.
Gaps in speech where pieces of speech are interspersed with silence are particularly

Delay causes poor caller interactivity. Poor caller interactiv
ity can cause echo and talker
overlap. Echo is the effect of the signal reflecting the speaker voice from the far
telephone equipment back into the speaker ear. Talker overlap is caused when one
delay becomes greater than 250 ms. When this long del
ay occurs; one talker steps in on the
speech of the other talker.

The worst
case result of delay is a disconnected call. If there are long gaps in speech, the
parties will hang up. If there are signaling problems, calls are disconnected. Such events are
acceptable in voice communications, yet are quite common for an inadequately prepared
data network that is attempting to carry voice.

The four major issues that face converged enterprise networks:


Lack of Bandwidth capacity:

Large graphics files, multimed
ia uses, and increasing
use of voice and video cause bandwidth capacity problems over data networks.


end delay (both fixed and variable):
Delay is the time it takes for a packet to
reach the receiving endpoint after being transmitted from the sendi
ng endpoint.
This period of time is called the “end
end delay” and consists of two components:

Fixed network delay:
Two types of fixed network delay are serialization and
propagation delays. Serialization is the process of placing bits on the circuit.
The higher the circuit speed, the less time it takes to place the bits on the
circuit. Therefore, the higher the speed of the link, the less serialization
delay is incurred. Propagation delay is the time it takes frames to transit the
physical media.

able network delay:
Processing delay is a type of variable delay and is
the time required by a networking device to look up the route, change the
header, and complete other switching tasks. In some cases, the packet must
also be manipulated, as, for exampl
e, when the encapsulation type or the
hop count must be changed. Each of these steps can contribute to
processing delay.


Variation of delay (also called jitter):
Jitter is the delta, or difference, in the
total end
end delay values of two voice packets

in the voice flow.


Packet loss:
WAN congestion is the usual cause for packet loss and results in
speech dropouts or a stutter effect if the play out side tries to accommodate
for the loss by retransmitting previously sent packets.

The example in slide
6 shows a network with four hops between a server and a client. Each
hop uses different media with different bandwidths. The maximum available bandwidth is
equal to the bandwidth of the slowest link.

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The calculation of the available bandwidth, however, is
much more complex in cases where
multiple flows are traversing the network. In such cases, you must calculate average
bandwidth available per flow.

Inadequate bandwidth can have performance impacts on network applications, especially
those that are time
nsitive (such as voice) or consume a lot of bandwidth (such as
videoconferencing). These performance impacts result in poor voice and video quality. In
addition, interactive network services, such as terminal services and remote desktops, may
also suffer f
rom lower bandwidth, which results in slow application responses.

Bandwidth is one of the key factors that affects QoS in a network; the more bandwidth there
is, the better the QoS will be.

The best way to increase bandwidth is to increase the link capaci
ty of the network to
accommodate all applications and users, allowing extra, spare bandwidth. Although this
solution sounds simple, increasing bandwidth is expensive and takes time to implement.
There are often technological limitations in upgrading to a h
igher bandwidth.

The better option is to classify traffic into QoS classes and prioritize each class according to
its relative importance. The basic queuing mechanism is First In First Out (FIFO). Other
queuing mechanisms provide additional granularity to
serve voice and business
traffic. Such traffic types should receive sufficient bandwidth to support their application
requirements. Voice traffic should receive prioritized forwarding, and the least important
traffic should receive the unallocated

bandwidth that remains after prioritized traffic is

Cisco IOS QoS software provides a variety of mechanisms to assign bandwidth priority to
specific classes of traffic:

Priority queuing (PQ) or custom queuing (CQ)

Modified deficit round rob
in (MDRR) (on Cisco 12000 Series Routers)

Distributed type of service (ToS)
based and QoS group
based weighted fair
queuing (WFQ) (on Cisco 7x00 Series Routers)

based weighted fair queuing (CBWFQ)

latency queuing (LLQ)

A way to increase the avail
able link bandwidth is to optimize link usage by compressing the
payload of frames (virtually). Compression, however, also increases delay because of the
complexity of compression algorithms. Using hardware compression can accelerate packet
payload compres
sions. Stacker and Predictor are two compression algorithms that are
available in Cisco IOS software.

Another mechanism that is used for link bandwidth efficiency is header compression. Header
compression is especially effective in networks where most pack
ets carry small amounts of
data (that is, where the payload
header ratio is small). Typical examples of header
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compression are TCP header compression and Real
Time Transport Protocol (RTP) header

Example: Using Available Bandwidth More Effi
ciently (Slide 8)

In a network with remote sites that use interactive traffic and voice for daily business,
bandwidth availability is an issue. In some regions, broadband bandwidth services are
difficult to obtain or, in the worst case, are not available.

This situation means that available
bandwidth resources must be used efficiently. Advanced queuing techniques, such as
CBWFQ or LLQ, and header compression mechanisms, such as TCP and RTP header
compression, are needed to use the bandwidth much more effic

In this example, a low
speed WAN link connects two office sites. Both sites are equipped
with IP phones, PCs, and servers that run interactive applications, such as terminal services.
Because the available bandwidth is limited, an appropriate strat
egy for efficient bandwidth
use must be determined and implemented.

Administrators must chose suitable queuing and compression mechanisms for the network
based on the kind of traffic that is traversing the network. The example uses LLQ and RTP
header compr
ession to provide the optimal quality for voice traffic. CBWFQ and TCP header
compression are effective for managing interactive data traffic.

Four types of delay:

Processing delay:
Processing delay is the

time that it takes for a router (or Layer 3 switc
h) to
take the packet from an input interface and put the packet into the output queue of the
output interface. The processing delay depends on various factors:

CPU speed

CPU use

IP switching mode

Router architecture

Configured features on both the input
and output interfaces

Queuing delay:
Queuing delay is the time that a packet resides in the output queue of a
router. Queuing delay depends on the number of packets that are already in the queue and
packet sizes. Queuing delay also depends on the bandwidth

of the interface and the queuing

Serialization delay:
Serialization delay is the time that it takes to place a frame on the
physical medium for transport. This delay is typically inversely proportional to the link

Propagation delay
Propagation delay is the time that it takes for the packet to cross the link
from one end to the other. This time usually depends on the type of media that is being
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transmitted, be it data, voice or video. For example, satellite links produce the longest

propagation delay because of the high altitudes of communications satellites.

end delay and jitter have a severe quality impact on the network:

end delay is the sum of all types of delays.

Each hop in the network has its own set of variable
processing and queuing delays,
which can result in jitter.

Internet Control Message Protocol (ICMP) echo (ping) is one way to measure the round
time of IP packets in a network.

When considering solutions to the delay problem, there are
two things to

Processing and queuing delays are related to devices and are bound to the
behavior of the operating system.

Propagation and serialization delays are related to the media.

There are many ways to reduce the delay at a router. Assuming that the router
has enough
power to make forwarding decisions rapidly, these factors influence most queuing and
serialization delays:

Average length of the queue

Average length of packets in the queue

Link bandwidth

Network administrators can accelerate the packet dispatc
hing for delay
sensitive flows:

Increase link capacity:

Sufficient bandwidth causes queues to shrink so that
packets do not wait long before transmittal. Increasing bandwidth reduces
serialization time. This approach can be unrealistic because of the cost
s that
are associated with the upgrade.

Prioritize delay
sensitive packets:

This approach can be more cost
than increasing link capacity. WFQ, CBWFQ, and LLQ can each serve certain
queues first (this is a pre
emptive way of servicing queues).

eprioritize packets:
In some cases, important packets need to be
reprioritized when they are entering or exiting a device. For example, when
packets leave a private network to transit an Internet service provider (ISP)
network, the ISP may require that the

packets be reprioritized.

Compress payload:

Payload compression reduces the size of packets, which
virtually increases link bandwidth. Compressed packets are smaller and take
less time to transmit. Compression uses complex algorithms that add delay.
If y
ou are using payload compression to reduce delay, make sure that the
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time that is needed to compress the payload does not negate the benefits of
having less data to transfer over the link.

Use header compression:

Header compression is not as CPU

payload compression. Header compression reduces delay when used with
other mechanisms. Header compression is especially useful for voice packets
that have a bad payload
header ratio (relative large header in comparison
to the payload), which is impr
oved by reducing the header of the packet
(RTP header compression).

By minimizing delay, network administrators can also reduce jitter (delay is more predictable
than jitter and easier to reduce
In the example (Slide 12), an ISP providing QoS connects the
ffices of the customer to each other. A low
speed link (512 kbps) connects the branch
office while a higher
speed link (1024 kbps) connects the main office. The customer uses
both IP phones and TCP/IP
based applications to conduct daily business. Because t
he branch
office only has a bandwidth of 512 kbps, the customer needs an appropriate QoS strategy to
provide the highest possible quality for voice and data traffic.

In this example, the customer needs to communicate with HTTP, FTP, e
mail, and voice
ces in the main office. Because the available bandwidth at the customer site is only 512
kbps, most traffic, but especially voice traffic, would suffer from end
end delays.

In this example, the customer performs TCP and RTP header compression, LLQ, and
prioritization of the various types of traffic. These mechanisms give voice traffic a higher
priority than HTTP or e
mail traffic. In addition to these measures, the customer h
as chosen
an ISP that supports QoS in the backbone.

The ISP performs reprioritization for customer traffic according to the QoS

policy for the
customer so that the traffic streams arrive on time at the main office of the customer. This
design guarantees that voice traffic has high priority and a guaranteed bandwidth of 128
kbps, FTP and e
mail traffic receive medium priority and a

bandwidth of 256 kbps, and HTTP
traffic receives low priority and a bandwidth of 64 kbps. Signaling and other management
traffic uses the remaining 64 kbps.

After delay, the next most serious concern for networks
is packet loss. Usually, packet loss occur
s when routers run out of buffer space for a
particular interface (output queue).

This graphic
(Slide 13)
shows examples of the results of packet loss in a converged network.

The graphic (slide 14) illustrates a full interface output queue, which causes n
ewly arriving
packets to be dropped. The term that is used for such drops is “output drop” or “tail drop”
(packets are dropped at the tail of the queue).

Routers might also drop packets for other less common reasons:

Input queue drop:

The main CPU is busy

and cannot process packets (the
input queue is full).

The router runs out of buffer space.

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The CPU is busy and cannot assign a free buffer to the new packet.

Frame errors:

The hardware detects an error in a frame; for example, cyclic
edundancy checks (CRCs), runt, and giant.

Packet loss is usually the result of congestion on an interface. Most applications that use TCP
experience slowdown because TCP automatically adjusts to network congestion. Dropped
TCP segments cause TCP sessions t
o reduce their window sizes. Some applications do not
use TCP and cannot handle drops (fragile flows).

These approaches prevent drops in sensitive applications:

Increase link capacity to ease or prevent congestion.

Guarantee enough bandwidth and increase b
uffer space to accommodate
bursts of traffic from fragile flows. There are several mechanisms available
in Cisco IOS QoS software that can guarantee bandwidth and provide
prioritized forwarding to drop
sensitive applications.

Prevent congestion

by dropping

priority packets be
fore congestion occurs. Cisco
QoS provides queuing mechanisms that start dropping lower
priority packets before
congestion occurs.

Cisco IOS QoS software provides the following mechanisms to prevent congestion:

Traffic polici

Traffic policing propagates bursts. When the traffic rate
reaches the configured maximum rate, excess traffic is dropped (or
remarked). The result is an output rate that appears as a saw
tooth with
crests and troughs.

Traffic shaping
In contrast to pol
icing, traffic shaping retains excess packets
in a queue and then schedules the excess for later transmission over
increments of time. The result of traffic shaping is a smoothed packet output

Shaping implies the existence of a queue and of sufficien
t memory to buffer delayed
packets, while policing does not. Queuing is an outbound concept; packets going out an
interface get queued and can be shaped. Only policing can be applied to inbound traffic on
an interface. Ensure that you have sufficient memor
y when enabling shaping. In addition,
shaping requires a scheduling function for later transmission of any delayed packets. This
scheduling function allows you to organize the shaping queue into different queues.
Examples of scheduling functions are CBWFQ
and LLQ.

Example: Packet Loss Solution (Slide 17)

This graphic shows a customer connected to the network via the WAN who is suffering from
packet loss that is caused by interface congestion. The packet loss results in poor voice
quality and slow data
traffic. Upgrading the WAN link is not an option to increase quality and
speed. Other options must be considered to solve the problem and restore network quality.

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avoidance techniques monitor network traffic loads in an effort to anticipate
avoid congestion at common network and internetwork bottlenecks before congestion
becomes a problem. These techniques provide preferential treatment for premium (priority)
traffic when there is congestion while concurrently maximizing network throughput an
capacity use and minimizing packet loss and delay. For example, Cisco IOS QoS congestion
avoidance features include Weighted Random Early Detection (WRED) and low latency
queuing (LLQ) as possible solutions.

The WRED algorithm allows for congestion avoid
ance on network interfaces by providing
buffer management and allowing TCP traffic to decrease, or throttle back, before buffers are
exhausted. The use of WRED helps avoid tail drops and global synchronization issues,
maximizing network use and TCP
based a
pplication performance. There is no such
congestion avoidance for User Datagram Protocol (UDP)
based traffic, such as voice traffic.
In case of UDP
based traffic, methods such as queuing and compression techniques help to
reduce and even prevent UDP packet

loss. As this example indicates, congestion avoidance
combined with queuing can be a very powerful tool for avoiding packet drops.

Slides 3.2

Implementing QOS


is a generic term that refers to algorithms that provide different levels of quality to
different types of network traffic. The most common implementation uses some sort of
advanced queuing algorithm. Quality of Service (QoS) has become mission
critical a
organizations move to reduce cost of operation, manage expensive WAN bandwidth, or
deploy applications such as Voice over IP (VoIP) to the desktop.

The goal of QoS is to provide better and more predictable network service by providing
dedicated bandwidt
h, controlled jitter and latency, and improved loss characteristics. QoS
achieves these goals by providing tools for managing network congestion, shaping network
traffic, using expensive wide
area links more efficiently, and setting traffic policies across

network. QoS offers intelligent network services that, when correctly applied, help to
provide consistent and predictable performance.

Simple networks process traffic with a FIFO queue. Network administrators need QoS when
some packets need different
treatments than others. For example, e
mail packets can be
delayed for several minutes with no one noticing, while VoIP packets cannot be delayed for
more than a tenth of a second before users notice the delay.

QoS is the ability of the network to provide
better or “special” services to selected users and
applications to the detriment of other users and applications. In any bandwidth
network, QoS reduces jitter, delay, and packet loss for time
sensitive and mission


itter: Variability of delay

Packet loss: Packets not forwarded (dropped)

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One way network elements handle an overflow of arriving traffic is to use a queuing
algorithm to sort the traffic and then determine some method of prioritizing it onto an
output li
nk. Cisco IOS software includes the following queuing tools:

in, first
out (FIFO) queuing

Priority queuing (PQ)

Custom queuing (CQ)

based weighted fair queuing (WFQ)

based weighted fair queuing (CBWFQ)

Each queuing algorithm was desi
gned to solve a specific network traffic problem and has a
particular effect on network performance, as described in the following sections.

Traffic policing propagates bursts. When the traffic rate reaches the configured maximum
rate, excess traffic is dr
opped (or remarked). The result is an output rate that appears as a
tooth with crests and troughs. In contrast to policing, traffic shaping retains excess
packets in a queue and then schedules the excess for later transmission over increments of
The result of traffic shaping is a smoothed packet output rate.

Traffic Shaping
: Traffic shaping monitors traffic from each source for bandwidth use.
When traffic from a specific source is too high, packets from that source are then
queued (delayed).

ffic Policing
: Traffic policing is also called “rate limiting.” Traffic policing is an
improvement on traffic shaping. In traffic policing, the packets are not simply
queued; they also have their IP priority levels altered or they are dropped.

FIFO: Basic

Forward Capability

In its simplest form, FIFO queuing involves storing packets when the network is congested
and forwarding them in order of arrival when the network is no longer congested. FIFO is the
default queuing algorithm in some instance
s, thus requiring no configuration, but it has
several shortcomings. Most important, FIFO queuing makes no decision about packet
priority; the order of arrival determines bandwidth, promptness, and buffer allocation. Nor
does it provide protection against
behaved applications (sources). Bursty sources can
cause long delays in delivering time
sensitive application traffic, and potentially to network
control and signaling messages. FIFO queuing was a necessary first step in controlling
network traffic, bu
t today's intelligent networks need more sophisticated algorithms. In
addition, a full queue causes tail drops. This is undesirable because the dropped packet
could be a high
priority packet. The router can’t prevent this packet from being dropped
there is no room in the queue for it (in addition to the fact that FIFO cannot tell a
priority packet from a low
priority packet). Cisco IOS software implements queuing
algorithms that avoid the shortcomings of FIFO queuing.

PQ: Prioritizing Traffic

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PQ ensures that important traffic gets the fastest handling at each point where it is used. It
was designed to give strict priority to important traffic. Priority queuing can flexibly prioritize
according to network protocol (for example IP, IPX, or Apple
Talk), incoming interface, packet
size, source/destination address, and so on. In PQ, each packet is placed in one of four

high, medium, normal, or low

based on an assigned priority. Packets that are not
classified by this priority list mechanism fa
ll into the normal queue. During transmission, the
algorithm gives higher
priority queues absolute preferential treatment over low

CQ: Guaranteeing Bandwidth

CQ allows various applications or organizations to share the network among applications
with specific minimum bandwidth or latency requirements. In these environments,
bandwidth must be shared proportionally between applications and users. You can use the
Cisco CQ feature to provide guaranteed bandwidth at a potential congestion point, ensuring
the specified traffic a fixed portion of available bandwidth and leaving the remaining
bandwidth to other traffic. Custom queuing handles traffic by assigning a spec
ified amount
of queue space to each class of packets and then servicing the queues in a round

As shown here, CQ handles traffic by assigning a specified amount of queue space to each
class of packet and then servicing up to 17 queues in a
robin fashion.

The queuing algorithm places the messages in one of 17 queues (queue 0 holds system
messages such as keepalives, signaling, and so on) and is emptied with weighted priority.
The router services queues 1 through 16 in round
robin order
, dequeuing a configured byte
count from each queue in each cycle. This feature ensures that no application (or specified
group of applications) achieves more than a predetermined proportion of overall capacity
when the line is under stress. Like PQ, CQ is

statically configured and does not automatically
adapt to changing network conditions.

Based WFQ: Creating Fairness Among Flows

For situations in which it is desirable to provide consistent response time to heavy and light
network users alike witho
ut adding excessive bandwidth, the solution is flow
based WFQ
(commonly referred to as just WFQ). It is a flow
based queuing algorithm that creates bit
wise fairness by allowing each queue to be serviced fairly in terms of byte count.

For example, if queu
e 1 has 100
byte packets and queue 2 has 50
byte packets, the WFQ
algorithm will take two packets from queue 2 for every one packet from queue 1. This makes
service fair for each queue: 100 bytes each time the queue is serviced.

WFQ ensures that queues do

not starve for bandwidth and that traffic gets predictable
service. Low
volume traffic streams that comprise the majority of traffic, receive increased
service, transmitting the same number of bytes as high
volume streams. This behavior
results in what ap
pears to be preferential treatment for low
volume traffic, when in actuality
it is creating fairness.

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Based WFQ: Ensuring Network Bandwidth

CBWFQ is one of Cisco's newest congestion
management tools for providing greater
flexibility. It will provide a minimum amount of bandwidth to a class as opposed to providing
a maximum amount of bandwidth as with traffic shaping.

CBWFQ allows a network ad
ministrator to create minimum guaranteed bandwidth classes.
Instead of providing a queue for each individual flow, the administrator defines a class that
consists of one or more flows, each class with a guaranteed minimum amount of bandwidth.

CBWFQ preven
ts multiple low
priority flows from swamping out a single high
priority flow.

To contrast the behavior of CBWFQ with WFQ, for example, WFQ will provide a video stream
that needs half the bandwidth of T1 if there are two flows. But, if more flows are added
, the
video stream gets less of the bandwidth because WFQ's mechanism creates fairness. If there
are 10 flows, the video stream will get only 1/10th of the bandwidth, which is not enough.

CBWFQ provides the mechanism needed to provide the half of the band
width that video
needs. The network administrator defines a class, places the video stream in the class, and
tells the router to provide 768 kbps (half of a T1) service for the class. Video therefore gets
the bandwidth that it needs. The remaining flows re
ceive a default class. The default class
uses flow
based WFQ schemes fairly allocating the remainder of the bandwidth (half of the
T1, in this example).

Congestion avoidance is a form of queue management. Congestion
avoidance techniques
monitor network tr
affic loads in an effort to anticipate and avoid congestion at common
network bottlenecks, as opposed to congestion
management techniques that operate to
control congestion after it occurs. The primary Cisco IOS congestion avoidance tool is WRED.

The rand
om early detection (RED) algorithms avoid congestion in internetworks before it
becomes a problem. RED works by monitoring traffic load at points in the network and
stochastically discarding packets if the congestion begins to increase. The result of the d
is that the source detects the dropped traffic and slows its transmission. RED is primarily
designed to work with TCP in IP internetwork environments.

WRED combines the capabilities of the RED algorithm with IP precedence. This combination
provides fo
r preferential traffic handling for higher
priority packets. It can selectively discard
priority traffic when the interface starts to get congested and can provide
differentiated performance characteristics for different classes of service.

A full qu
eue causes tail drops. Tail drops are dropped packets that could not fit into the
queue because the queue was full. This is undesirable because the dropped packet may have
been a high
priority packet and the router did not have a chance to queue it. If the

queue is
not full, the router can look at the priority of all arriving packets and drop the lower
packets, allowing high
priority packets into the queue. By managing the depth of the queue
(the number of packets in the queue) by dropping specifie
d packets, the router does its best
to make sure that the queue does not fill and that tail drops do not happen.

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A communications network forms the backbone of any successful organization. These
networks transport a multitude of applications and data, inc
luding high
quality video and
sensitive data such as real
time voice. The bandwidth
intensive applications stretch
network capabilities and resources, but also complement, add value, and enhance every
business process. Networks must provide secure, p
redictable, measurable, and sometimes
guaranteed services. Achieving the required Quality of Service (QoS) by managing the delay,
delay variation (jitter), bandwidth, and packet loss parameters on a network becomes the
secret to a successful end
end bus
iness solution. Thus, QoS is the set of techniques to
manage network resources.

Cisco IOS QoS features enable network administrators to control and predictably service a
variety of networked applications and traffic types, allowing network managers to tak
advantage of a new generation of media
rich and mission
critical applications.

There are three basic steps involved in implementing QoS on a network:

Identify types of traffic and their requirements
: Study the network to
determine the type of traffic th
at is running on the network and then
determine the QoS requirements needed for the different types of traffic.

Define traffic classes
: This activity groups the traffic with similar QoS
requirements into classes. For example, three classes of traffic migh
t be
defined as voice, mission
critical, and best effort.

Define QoS policies
: QoS policies meet QoS requirements for each traffic

The first step in implementing QoS is to identify the traffic on the network and then
determine the QoS

requirements and the importance of the various traffic types. This step
provides some high
level guidelines for implementing QoS in networks that support for
multiple applications, including delay
sensitive and bandwidth
intensive applications. These
ications may enhance business processes, but stretch network resources. QoS can
provide secure, predictable, measurable, and guaranteed services to these applications by
managing delay, delay variation (jitter), bandwidth, and packet loss in a network.

ermine the QoS problems of users
. Measure the traffic on the network
during congested periods. Conduct CPU use assessment on each of the
network devices during busy periods to determine where problems might be

Determine the business model and g
oals and obtain a list of business
. This activity helps define the number of classes that are
needed and allows you to determine the business requirements for each
traffic class.

Define the service levels required by different traffic classes

in terms of
response time and availability
. Questions to consider when defining service
levels include what is the impact on business if the network delays a
BSc Systems 4

Quality of Service in Converged Networks

Notes on Slides 3.1 & 3.2


transaction by two or three seconds. A service level assignment will include
the priority and the

treatment a packet will receive. For example, you would
assign voice applications a high service level (high priority, LLQ and RTP
compression). You would assign low priority data a lower service level
(lower priority, WFQ, TCP header compression).

identifying and measuring network traffic, use business requirements to perform the
second step: define the traffic classes.

Because of its stringent QoS requirements, voice traffic is usually in a class by itself. Cisco has
developed specific QoS mechanis
ms, such as LLQ, to ensure that voice always receives
priority treatment over all other traffic.

After the applications with the most critical requirements have been defined, the remaining
traffic classes are defined using business requirements.

A typical
enterprise might define five traffic classes:


Absolute priority for VoIP traffic.

Small set of locally defined critical business applications.
For example, a mission
critical application might be an order
entry database
that need
s to run 24 hours a day.

Database access, transaction services, interactive traffic, and
preferred data services. Depending on the importance of the database
application to the enterprise, you might give the database a large amount of
idth and a high priority. For example, your payroll department
performs critical or sensitive work. Their importance to the organization
determines the priority and amount of bandwidth you would give their
network traffic.

Best effort:
Popular applications

such as e
mail and FTP could each
constitute a class. Your QoS policy might guarantee employees using these
applications a smaller amount of bandwidth and a lower priority then other
applications. Incoming HTTP queries to your company's external website
ight be a class that gets a moderate amount of bandwidth and runs at low



The unspecified traffic is considered as less than best effort. Scavenger
applications, such as BitTorrent and other point
point applications,

are served by this

In the third step, define a QoS policy for each traffic class. Defining a QoS policy
involves one or more of these activities:

Setting a minimum bandwidth guarantee

Setting a maximum bandwidth limit

Assigning priorities to each c

BSc Systems 4

Quality of Service in Converged Networks

Notes on Slides 3.1 & 3.2


Using QoS technologies, such as advanced queuing, to manage congestion

Using the traffic classes, previously defined QoS policies can be mandated based on the
following priorities (with Priority 5 being the highest and Priority 1 being the lowest):

iority 5

Voice: Minimum bandwidth of 1 Mbps. Use LLQ to give voice priority always.

Priority 4

critical: Minimum bandwidth of 1 Mbps. Use CBWFQ to prioritize critical
class traffic flows.

Priority 3

Transactional: Minimum bandwidth of 1 Mbps. Use C
BWFQ to prioritize
transactional traffic flows.

Priority 2

effort: Maximum bandwidth of 500 kbps. Use CBWFQ to prioritize best
effort traffic flows that are below mission
critical and voice.

Priority 1

Scavenger (less
effort): Maximum bandwi
dth of 100 kbps. Use WRED
to drop these packets whenever the network has a tendency toward congestion.

In this illustration (Slide 21), the packages on the conveyer belt represent data packets
moving through the network.

As packets move through each
phase, they are identified and prioritized, then managed and
sorted, and finally processed and sent.

As the packets are identified, they are sorted into separate queues. Notice how some
packets receive priority (more of these packets are processed over tim
e), and some packets
are selectively dropped.

time applications are especially sensitive to QoS: interactive voice and

Weighted Fair Queuing (WFQ)
: FIFO systems store all packets in one queue. WFQ stores
each type of packet in a se
parate queue and assigns each queue a different priority level.

There are three basic steps involved in implementing QoS on a network:

Identify types of traffic and their requirements
: Study the network to
determine the type of traffic that is running o
n the network and then
determine the QoS requirements needed for the different types of traffic.

Define traffic classes
: This activity groups the traffic with similar QoS
requirements into classes. For example, three classes of traffic might be
defined a
s voice, mission
critical, and best effort.

Define QoS policies
: QoS policies meet QoS requirements for each traffic


The unspecified traffic is considered as less than best effort.