Quality of Service Management

for Voice Over IP Networks

Team Members:
Prashant Anantha Krishnan
Sunil Kumar Derasriya
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Objectives and Goals

To analyze and study the parameters

that affect the Quality of Service in

Voice Over IP.
Using a testing tool to estimate these
parameters under all possible
network connections.
Calculate a Mean Opinion
Score(MOS) based on an algorithm.
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Challenges and Difficulties

VoIP is a very new and sophisticated

concept so a lot of study had to be

done to understand these concepts
Finding a testing tool or a simulation
tool that would help us estimating
the parameters, which are required
to calculate the Mean Opinion
Score(MOS).

What is VoIP
It is a technology for transmitting voice

calls over the Internet using packet linked

routes. Also known as IP telephony.
It enables the people to use the Internet
as a transmission medium for sending
voice data in packets using IP rather than
using traditional circuit transmission of
the PSTN.
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Contd
Public Switched
Telephone Network
Initially, PC to PC
voice calls over the
Internet

Gateways allow PCs

to also reach phones

PSTN
(NY)
Gateway

Multimedia
PC
Gateway

IP Network
Multimedia
PC

PSTN
(DC)

or phones to reach
phones

Advantages of VoIP

Greater Efficiency
Lower Cost
Higher Reliability
Supporting Innovation

Quality of Service
The ultimate objective of VoIP is reliable,

high-quality voice service, the kind that

users expect from the PSTN.
It is hard to achieve the same level of QoS
as in PSTN. The main QoS issues are speech
quality, service availability and usability.
Voice requires lower delay, jitter and packet
loss where as Ordinary Data transfer can be
delayed without affecting much to the
clients requirement.
To withstand to such needs a minimum level
of QoS mechanism must be maintained.
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Quality of Service
Parameters

Delay : The amount of time taken

by a packet to reach from the
source to the destination.

Issues with Delay

Echo
Talk Overlap

Types of Delay in a VoIP Call:

Processing Delay
Network Delay
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Jitter
Jitter is the variation in the time

between packets arriving, caused by

network
congestion
or
route
changes.
Removing jitter requires collecting
packets and holding them long
enough to allow the slowest packets
to arrive in time to be played in the
correct sequence which causes
additional delay.
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Packet Loss
Packet Loss is losing packets along

the data path which further degrades

the VoIP Applications.
Voice packets are time-sensitive
unlike Data packets. Therefore,
retransmission is not a solution to
this problem.

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Latency
When the packet is being sent, there is a

latent time till the computer that sent the

packet waits for a confirmation that the
packet has been received.
Latency causes packets to be lost. If a
packet does not arrive in time to be
replayed at the receiving end, the packet is
dropped.
Latency does not distort the voice signal
but delay can be very annoying, making
normal conversation difficult for the
speakers. The parties may start to talk at
the same time or interrupt each other. As a
result, the conversational quality is
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perceived as being poor.

Solutions for QoS Issues

For Voice communications over IP to

become acceptable to the user, the delay

needs to be less than a threshold value.
To ensure good quality of service, we can
use Echo Cancellation and Packet
Prioritization.
Use of service quality models that gives
an estimate of perceptual quality rating
using the networking parameters. Mean
Opinion Score (MOS) is one of the quality
rating on a scale of 1 (bad) to 5
(excellent)
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Perceptual Assessment Model

PESQ is a model for perceptual evaluation of speech quality.
One novel feature of PESQ is the identification of transmission

delays.
First PESQ adjusts the degraded version to be time aligned. Then it
assesses the distortion between original and degraded sample.
Constant delays are not considered in the calculation of the MOS
value, but delay variations change the rating of the speech quality.
One should note that PESQ can only be applied for distortions
which have been known before its development.
In PESQ the original and the degraded signals are mapped onto
an
internal representation using a perceptual model. The
difference in this representation is used by a cognitive model to
predict the perceived speech quality of the degraded signal. This
perceived listening quality is expressed in terms of a mean opinion
score (MOS), an average quality score over a large set of subjects.

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Perceptual Evaluation Of Speech

Quality Model

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E-Model
Mouth to ear transmission

quality measurement
Produces an R factor typically in
the range 50 (bad) -95 (good)
R factor can be related to MOS
score, Terminate Early (TME) etc.
ITU G.107/ G.108 and ETSI ETR250

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Contd
R = Ro - Is - Id - Ie + A
Base R value
- Noise level

Impairments that
occur simultaneously
with speech
- received speech level
- sidetone level
- quantization noise

Advantage factor

Impairments that
are delayed with
respect to speech
- talker echo
- listener echo
- round trip delay

Equipment Impairment
Factor
- CODEC
- multiplexing effects

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Contd..
Packet
Loss

Loss
Model

Jitter

Jitter
Model

Codec
type

Codec
Model

Ie

R Factor
E Model

Delay, measured
using RTCP

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R Factor vs MOS
R Factor

MOS
4.5

4.0

3.0

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How does Rider Works?

Rider is an entry-level network performance measure
program that measures the network response time,
bandwidth, and Voice Over IP parameters between
any two computers on your network.
There are four basic tests performed by Rider when
sending test data between pairs of computers:

Bandwidth testing. How long will it take to copy a big file

across the network?
Response time testing. How long will it take for a packet of
data to travel from one end of the network and back?
Voice over IP testing. If you were to use a new VoIP phone,
how good or bad would the packet loss and jitter be?
Dropped packets hurt the sound quality. Jitter refers to the
variation in packet arrival time. Packets that arrive too late
or out of order (yes, this happens) can't be used.
Stream testing. This is just the general case of Voice over IP
testing. If we wanted to run a movie stream, or some other
application, would it work?
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How to Calculate MOS?

For the codec used pick a corresponding default R

value. The R-values for the most popular codec
has been used. These areR=93 for the G.711
codec, R=80 for the G.729a codec and R=86 for
iLBC codec.
From the Rider streaming test we can calculate the
Jitter and Packet Loss.
From the Rider response time test, we can measure
the network latency between the control and
remote locations.
Add 10 ms if you are using G.729a and 5 ms for
iLBC codec to account for computation time.
Add step no. 2, 3 and 4 to calculate the effective
latency. (Latency plus jitter plus computation time.)
Adjust the R value down based on effective
latency. Deduct 5 for a delay of 150 ms, 20 for a
delay of 250 ms, 30 for a delay of 350 ms.
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Contd..

From the packet loss test in step no. 2, deduct

the R value from consecutive packet losses
using the table given below.
Consecutive
Frames Lost

R-value Deduction

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38

57

66

78

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DEMONSTRATION

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Conclusion
At last, we conclude that Mean Opinion

Score is one of the better and reliable

ways to estimate the quality of service for
a VoIP Network.

The future Implementations of our

project is to bind our application with a
simulation tool.

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