Transcript Quality of Service Management Over VoIP Networks

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).
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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
IP Network
Gateway
Multimedia
PC
PSTN
(DC)
…or phones to reach
phones
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Advantages of VoIP
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Greater Efficiency
Lower Cost
Higher Reliability
Supporting Innovation
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Quality of Service
• The ultimate objective of VoIP is reliable, high•
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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 client’s 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”
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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 perceived as being poor.
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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
100
90
4.0
80
70
3.0
60
50
0.1
1
10
Percentage of users that terminate calls early
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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?
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For the codec used pick a corresponding default R
value. The R-values for the most popular codec has
been used. These are R=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
1
13
2
38
3
57
4
66
5
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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