Title: Qos Management for VOIP Networks with Edge-to-Edge Admission Control
1Qos Management for VOIP Networks with
Edge-to-Edge Admission Control
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2Reference
- 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.
3Outline
- Motivation
- Introduction
- VoIP Network Management
- Voice Quality Evaluation
- Edge-to-Edge Admission Control
- Network Dimensioning
- Performance Evaluation
- Conclusion
4Motivation
- VoIP????Internet????????Service?
- VoIP?Internet?????Application?
- ??VoIP????,???????????VoIP????,???VoIP????????PSTN
????????
5Introduction
- 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 .
6Introduction(Cont.)
- A mechanism called call admission control is
necessary to reject a new call when enough
network spare capacity is not available.
7Introduction(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).
8Introduction(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.
9Introduction(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.
10Outline
- Motivation
- Introduction
- VoIP Network Management
- Voice Quality Evaluation
- Edge-to-Edge Admission Control
- Network Dimensioning
- Performance Evaluation
- Conclusion
11VOIP NETWORK MANAGEMENT(1)
12VOIP NETWORK MANAGEMENT(2)
13VOIP 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.
14VOIP 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.
15VOIP NETWORK MANAGEMENT(5)
- The typical admission control parameters include
- (1) average packet lost rate for VoIP flows. (2)
admission threshold.
16Outline
- Motivation
- Introduction
- VoIP Network Management
- Voice Quality Evaluation
- Edge-to-Edge Admission Control
- Network Dimensioning
- Performance Evaluation
- Conclusion
17Voice 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.
18Voice 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.
19Voice 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
20Voice Quality Evaluation(4/6)
21Voice Quality Evaluation(5/6)
22Voice 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.
23Outline
- Motivation
- Introduction
- VoIP Network Management
- Voice Quality Evaluation
- Edge-to-Edge Admission Control
- Network Dimensioning
- Performance Evaluation
- Conclusion
24Edge-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.
25Edge-to-Edge Admission Control(2/6)
Endpoint(Node O)
Endpoint(Node T)
Probe Request
Probe
Connect
Voice Exchange
Release Complete
26Edge-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.
27Edge-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.
28Edge-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.
29Edge-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.
30Outline
- Motivation
- Introduction
- VoIP Network Management
- Voice Quality Evaluation
- Edge-to-Edge Admission Control
- Network Dimensioning
- Performance Evaluation
- Conclusion
31V.Network Dimensioning
- For dimensioning the network we need assume some
parameters - Between source and destination ,There have d
links
Destination
Source
d links
32Traffic Matrix
- Location-to-Location VoIP traffic demands are
represented by traffic matrix ai,j - i represents the source
- j represents the destination
33Parameter 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
34The 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
35Outline
- Motivation
- Introduction
- VoIP Network Management
- Voice Quality Evaluation
- Edge-to-Edge Admission Control
- Network Dimensioning
- Performance Evaluation
- Conclusion
36PERFORMANCE EVALUATION
37Simulation 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
38Simulation 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.
39Simulation 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.
40 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.
41Scenario (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
42RESULT
43(No Transcript)
44with 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.
45without 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.
46Outline
- Motivation
- Introduction
- VoIP Network Management
- Voice Quality Evaluation
- Edge-to-Edge Admission Control
- Network Dimensioning
- Performance Evaluation
- Conclusion
47CONCLUSION(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
48CONCLUSION(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.
49REFERENCES
- 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.
50- 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.
51- 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 .
52- 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 .