VOIP Quality of Service

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VOIPQuality of Service

• Dr. M Yaseen
• Pakistan Telecom Authority
• Pakistan

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Transition from Cct. Switched to packet switched
Services
Telecom Convergence Fast speed Integrated and
Interactive services -Interactive multimedia,
triple play, Gaming, etc -VOIP

• Business Transformation
• Single Subscription and Authentication
• Bundling of services
• -VOIP, IPTV, High Speed Internet

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NGN Architectural Concept
FTTN/FTTU
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Circuit Switched Voice to Packet Switched voice
64 K voice Channel Connection oriented
Guaranteed BW Good Quality of voice
VOIP Voice compressed and converted into digital
packets and traveled over Internet or private
network (Packet switched Network)
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Packet Switched Network QOS

• Connection less (No Established path unless VP/C)
  
• No Guaranteed BW unless VP/C

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Packet switched Network QOS
Quality of Service Guaranteed different levels
of service to different traffic (Voice, Data,
Video)

• Ensuring the BW to each traffic for consistent
  flow

• Prioritizing the traffic flow

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VOIP degradation in Network Element (NE)
Packet Loss due to large traffic in the network
/ NE Latency Delay for Packet delivery Jitter
Variation in delay of packet delivery
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• VOIP Packets QOS
• Voice packets should not be delayed or dropped
  due to interference from
• other lower priority traffic
• Guaranteed BW
• VOIP Packet Prioritization

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QOS Requirement
Latency Delay for Packet delivery Jitter
Variation in delay of packet delivery Packet
Loss due to large traffic in the network / NE
Latency (Basic Delay ) The delay gained by
voice packet to move across the network until
reaches at endpoint (Mouth- to - Ear)

• Reasons of latency
• Signal processing time at transmitting time
  (Encoding, Compression, Packetization)
• Reverse at Receiving end
• Multi-protocol processing Network elements
  (Routers) in the path
• Congestion in-route of NEs
• No control over number of NEs in the network

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QOS Requirement
How Latency affect Large delays -Can
cause bad echoes. -hard to understand the
conversation -Distorted flow of
conversation (Speaking at same time and
Interrupting each other)

• ITU-T Recommendation
• Callers usually notice roundtrip voice delays of
  250ms or more.
• ITU-T G.114 recommends a maximum of a 150 ms
  one-way latency. Since this includes the entire
  voice path, part of which may be on the public
  Internet.

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QOS Requirement
Jitter Irregular delay in arrival of voice
packets called jitter and causes strange sound
effects

• Reasons of Jitter
• No guaranteed specific path to the
  destination
• - intervals between packet arrival times vary
  since one packet may take more hops
  than the others
• No guaranteed specific bandwidth available
  for the duration of the communication.
• Parts of an IP network can also be
  momentarily overwhelmed with other
• processing duties in, and cause variable delay

Remedy To lessen the effects of jitter, packets
are gathered in a jitter buffer at the receive
end to normalize these delays. If a packet
arrives too late it will be discarded to avoid
the gap between the callers from growing. If it
arrives too quickly then it will be held for a
small duration of time so that its played to the
other party at a constant rate
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QOS Requirement
If jitter buffer is too small too many packets
will be discarded and the call quality will
suffer. If it is too large there will be
additional delay added to the call which can lead
to problems. It is generally bad to have a jitter
buffer larger than 100ms for this reason
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QOS Requirement

• Packet-loss / Drop
• Large number of Voice Packets on the network at
  the same time generate higher delays, and causes
  a backlog of data to be delivered.
• In order to reduce this delay the jitter buffer
  drop some of the packets which results a poor
  call quality.

• ITUT Recommendation
• Ideally 0 Packet loss for VOIP
• VOIP call using a G.711 codec significantly
  degrade with 1 packet loss
• The G.729 codec requires packet loss far less
  than 1 percent to avoid audible errors
• More compressing codecs can tolerate even
  less packet loss

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How to measure VOIP QOS

• Mean Opinion Score (MOS)
• The traditional measure of a users
  perception of quality
• Is the defined in (ITU-T P.800)
• An expert panel of listeners rated
  pre-selected voice samples of voice encoding and
  compression algorithms under controlled
  conditions.
• An MOS score can range from 1 (bad) to 5
  (excellent), and a MOS of 4 is considered toll
  quality.
• The Pulse Code Modulation (PCM) algorithm
  (ITU-T G.711) has a MOS score of 4.4.

• E-Model
• ITU-T G.107 presents a mathematical model,
  known as the E-Model,
• It predict QoS scores using more objective
  impairment factors.

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How to measure QOS
E Model R Ro - Is - Id - Ie
A R Transmission rating factor, which combines
all transmission parameters relevant for the
considered connection
Ro Represents basic signal-to-noise ratio,
including noise sources such as circuit noise
and room noise.
Is Represents combination of all impairments
which occur more or less simultaneously with the
voice signal.
Id represents the impairments caused by delay
and the equipment impairment
Ie represents impairments caused by low bit rate
codecs.
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How to measure QOS
A The advantage factor allows for compensation
of impairment factors
The term Ro and the Is and Id values are
subdivided into further specific impairment
values.
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E-Model Rating Values (R) and MOS scores

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P.862 ITU specifications are used to analyze the
distortion on test voice signals over a VoIP
network, and to produce an estimated MOS score.
This algorithms is used in most of test
equipment
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QOS Optimization

• Dedicated BW
• Shaping Network traffic
• Setting Traffic Priorities
• Managing Traffic congestion

• H323
• SIP
• Resource Reservation Protocol (RSVP)
• Differentiated Services (DiffServ)
• Multi Protocol Labeling Switching (MPLS)
• Subnet Bandwidth Management (SBM)

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Conclusion
Impairments factors Latency, Jitter, Packet Loss
are critical to achieve good quality Voice
over IP Network
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