Qos Management for VOIP Networks with Edge-to-Edge Admission Control

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Qos Management for VOIP Networks with
Edge-to-Edge Admission Control

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Reference

• K. Mase, Y. Toyama, A.A. Bilhaj, Y. Suda, "QoS
  management for VoIP networks with edge-to-edge
  admission control", in Proceedings GLOBECOM
  2001, vol. 4, 2001, pp. 2556 -2560.

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Outline

• Motivation
• Introduction
• VoIP Network Management
• Voice Quality Evaluation
• Edge-to-Edge Admission Control
• Network Dimensioning
• Performance Evaluation
• Conclusion

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Motivation

• VoIP????Internet????????Service?
• VoIP?Internet?????Application?
• ??VoIP????,???????????VoIP????,???VoIP????????PSTN
  ????????

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Introduction

• If a new call is accepted without a particular
  limit, QoS for calls in progress may be degraded
  below an acceptable level, because total
  bandwidth required for the calls exceeds the
  network capacity .

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Introduction(Cont.)

• A mechanism called call admission control is
  necessary to reject a new call when enough
  network spare capacity is not available.

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Introduction(Cont.)

• Traditionally, the Internet has provided the best
  effort services, and has not supported call
  admission control.
• However, admission control is necessary for
  guaranteeing QoS for real-time applications (?
  telephone service in the Internet).

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Introduction(Cont.)

• Edge-to-edge measurement based admission control
  (EMBAC), ???? edge-to-edge probe flow and QoS
  measurement to ensure spare capacity for the new
  flow. This method neither uses hop-by-hop
  signaling, nor requires any additional
  functionality for routers in the backbone
  network.

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Introduction(Cont.)

• EMBAC ?various network conditions?,?? call
  admission control ?? both directional voice flows
  ? packet loss rates ??? a pre-determined value
  ??.
• The results of voice quality evaluation is used
  to analyze possible problems, and if necessary to
  change parameters for admission control.

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Outline

• Motivation
• Introduction
• VoIP Network Management
• Voice Quality Evaluation
• Edge-to-Edge Admission Control
• Network Dimensioning
• Performance Evaluation
• Conclusion

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VOIP NETWORK MANAGEMENT(1)
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VOIP NETWORK MANAGEMENT(2)
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VOIP NETWORK MANAGEMENT(3)

• A VoIP network is designed to satisfy
  requirements such as allowed budget and voice
  quality objectives. While VoIP network is used,
  test calls are periodically generated between a
  set of PBX pairs, and voice quality and
  network-level QoS such as packet loss rates are
  measured for the test calls.

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VOIP NETWORK MANAGEMENT(4)

• As the results, problems in voice level QoS as
  well as network level QoS are identified. These
  problems are, then, analyzed and fixed through
  admission control optimization, network
  optimization, or fault and error recovery,
  depending on the specific causes.

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VOIP NETWORK MANAGEMENT(5)

• The typical admission control parameters include
• (1) average packet lost rate for VoIP flows. (2)
  admission threshold.

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Outline

• Motivation
• Introduction
• VoIP Network Management
• Voice Quality Evaluation
• Edge-to-Edge Admission Control
• Network Dimensioning
• Performance Evaluation
• Conclusion

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Voice Quality Evaluation(1/6)

• Measurement(1/2)
• Generate test call periodically
• Each test call undergoes the same admission
  control as ordinary calls do
• Once a test call is established, the artificial
  voice generation device attached to the
  call-originating PBX (Device A) sends artificial
  voice to the voice quality evaluation device
  attached to the call-terminating PBX (Device B)
  through the forward VoIP path.

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Voice Quality Evaluation(2/6)

• Measurement(2/2)
• Device B calculates instantaneous MOS values as
  well as the average MOS by comparing the original
  artificial voice signal and the received voice
  signal.
• The VoIP gateway at the call-terminating PBX
  monitors and measures packet loss rate for the
  test call.

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Voice Quality Evaluation(3/6)

• Holding time for a test call is an important
  design parameter
• The shorter holding time is desirable to minimize
  increase in network traffic load, while it should
  be long enough to assure reliability in MOS
  evaluation
• G723.1 coding and enhanced PSQM algorithm are used

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Voice Quality Evaluation(4/6)
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Voice Quality Evaluation(5/6)
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Voice Quality Evaluation(6/6)

• MOS 2 is a critical value for users to notice
  voice quality degradation.
• packet loss rate 2 is tolerable based on the
  measurement results of 20 or 60 sec measurement
  time
• From these observations, 20 sec is a good
  candidate to obtain reliable voice quality
  evaluation.

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Outline

• Motivation
• Introduction
• VoIP Network Management
• Voice Quality Evaluation
• Edge-to-Edge Admission Control
• Network Dimensioning
• Performance Evaluation
• Conclusion

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Edge-to-Edge Admission Control(1/6)

• End node A (source) to end node B (destination)
  through a selected path
• Node B is in charge of the admission test and
  judges whether to be able to accept the flow from
  node A to node B or not.

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Edge-to-Edge Admission Control(2/6)
Endpoint(Node O)
Endpoint(Node T)
Probe Request
Probe
Connect
Voice Exchange
Release Complete
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Edge-to-Edge Admission Control(3/6)

• The probe request is sent from the
  call-originating node (node O) to the
  call-terminating node (node T)
• Node O and node T may become a source or
  destination of the probe packet flows, as
  mentioned before
• The probe request activates node T to initiate
  the probing and measurement operation.

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Edge-to-Edge Admission Control(4/6)

• Following the probe request transmission and
  reception, probe packet flows are carried in both
  direction and packet loss rate measurements are
  conducted at the both end nodes.
• Node O measures the packet loss rate for the
  probe flow from node T to node O, and conducts
  admission test.

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Edge-to-Edge Admission Control(5/6)

• If the result of the admission test is success,
  node O transmit setup signal to node T.
• If it is failure, node O terminates the call
  setup.
• In parallel, node T measures the packet loss rate
  for the probe flow from node O to node T, and
  conducts admission test.

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Edge-to-Edge Admission Control(6/6)

• If the result of the admission test is success,
  node T proceeds to transmit connect signal to
  node O, responding the setup signal sent from
  node O.
• If it is failure, node T will reject setup
  request from node O.

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Outline

• Motivation
• Introduction
• VoIP Network Management
• Voice Quality Evaluation
• Edge-to-Edge Admission Control
• Network Dimensioning
• Performance Evaluation
• Conclusion

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V.Network Dimensioning

• For dimensioning the network we need assume some
  parameters
• Between source and destination ,There have d
  links

Destination
Source
d links
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Traffic Matrix

• Location-to-Location VoIP traffic demands are
  represented by traffic matrix ai,j
• i represents the source
• j represents the destination

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Parameter Assuming
Parameter Explain
B the edge-to-edge blocking probability is no more than this pre-determined value B
F Edge-to-edge peak pocket loss rate is no more than a pre-determined value F
P Represents the percentage of time during which speech is present
Freezeout fraction (average number of frozen out calls)/(average number of active calls)
k The number of the VoIP calls
w The maximum transfer rate for a VoIP call
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The peak values

• Blocking probability no more than B/d
• Freezeout fraction no more than F/d
• The Freezeout fraction is the upperbound for
  the packet loss rate
• The capacity of the link is given as kw

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Outline

• Motivation
• Introduction
• VoIP Network Management
• Voice Quality Evaluation
• Edge-to-Edge Admission Control
• Network Dimensioning
• Performance Evaluation
• Conclusion

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PERFORMANCE EVALUATION
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Simulation Model and Assumptions(1)

• For simplicity ,we ignore the amount of signaling
  flows ,because it is not significant compared
  with that of voice flows.
• A PBX is modeled as a switch accommodating
• infinite number of subscriber lines
• infinite number of outgoing and incoming trunks
• each link has 5ms propagation delay

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Simulation Model and Assumptions(2)

• Calls originate
• according to Poisson distribution between a pair
  of call-originating and terminating locations
• Call holding time
• base on exponential distribution with the average
  three minutes.

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Simulation Model and Assumptions(3)

• We assume that
• blocking probability target B is 3
• freezeout target F is 1.5.
• For an established call, voice activity p is 30.

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Scenario (1)

• A bottleneck may occur in the network due to
  traffic forecast error.
• We select a link in the middle of the network as
  the bottleneck link, and decrease the capacity
  from the initial size.

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Scenario (2)

• We use blocking probability and the peak packet
  loss rate for two seconds interval as the
  performance parameter.
• We consider three cases
• No Admission Control
• Admission threshold 2
• admission thresholds 10

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RESULT
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with admission control

• the peak packet loss rate is remarkably improved
  at the cost of acceptable increase in blocking
  probability, depending on the given admission
  thresholds.

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without admission control

• blocking probability is always 0, and peak packet
  loss rate is beyond acceptable level even without
  capacity reduction, and increases as the capacity
  reduction increases.

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Outline

• Motivation
• Introduction
• VoIP Network Management
• Voice Quality Evaluation
• Edge-to-Edge Admission Control
• Network Dimensioning
• Performance Evaluation
• Conclusion

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CONCLUSION(1)

• Admission control works well to control packet
  loss rate under given network conditions.
• We need to properly set the admission thresholds
  for each end node pair

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CONCLUSION(2)

• The relation of packet loss rate and the
  admission threshold is not obvious and it is not
  easy to analytically find the optimal admission
  threshold.
• Feedback control based on voice quality and
  packet loss measurements may be used to
  dynamically adjust the admission threshold.

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REFERENCES

• 1 B. Li, M. Hamdi, D. Jiang. Y. T. Hou, and X.
  Cao, QoS-enabled voice support in the
  next-generation Internet Issues,, existing
  approaches and challenges, IEEE Communications
  Magazine, Vol.38, No.4, April, 2000.
• 2 L. Breslau, E. Q. Knightly, S. Shenker, I.
  Stoica, and H. Zhang, Endpoint admission
  control architectural issues and performance,
  pp. 57-69, SIGCOMM00, 2000.
• 3 F. Borgonovo, A. Capone, L. Fratta, M.
  Marchese, and C. Petrioli, PCP A bandwidth
  guaranteed transport services for IP networks,
  ICC 99, pp. 1999.
• 4 G. Bianchi, A. Capone, C. Petrioli,
  Throughput analysis of end-to-end
  measurement-based admission control in IP,
  INFOCOM 2000, 2000.
• 5 V. E.lek, G. Karlsson, and R. Ronngren,
  Admission control based on end-to-end
  measurement, INFOCOM 2000, 2000.
• 6 M. Schwartz, K. Mase, and D. R. Smith,
  Priority channel assignment in tandem DSI, IEEE
  Trans. on Communications, Vol.Com-28. No.10,
  1980.
• 7 http//www.radcom-inc.com/products/internetsim
  .htm.
• 8 http//www.genista.co.jp.

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• Each packet has 40 bytes overhead
• 20 bytes IP packet header
• 8 bytes UDP header,
• and 12bytes RTP header
• The maximum length of a packet is 60 bytes.

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• A VoIP gateway and a router are modeled as a
  queuing system.
• Voice flows and probe flows are given individual
  classes and their own queues.
• As mentioned in ?,voice flow is given high
  priority in packet scheduling than probe flows.
• Specifically, non-preemptive priority scheduling
  is used.
• Buffer size is 40 packets for voice flows and 20
  packets for probe flows for each output link .

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• One packet is generated every 20 ms during active
  periods for each call.
• Thus, the maximum rate for a VoIP call, w,
  mentioned in Sec.?, is 24 kbps.
• We assume probe calls have one second duration.
• A size of a probe packet is always 60 bytes.
• The admission threshold is set to 10 .