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  • 1. 分成兩個部分實驗前的設定跟實驗結果分析
  • 614_1_qos.ppt

    1. 1. Qos Management for VOIP Networks with Edge-to-Edge Admission Control 報告者 : R93922106 莊萬慶 R93922073 王文廷 R93922127 張尚斌 R93525018 田謹維
    2. 2. Reference <ul><li>K. Mase, Y. Toyama, A.A. Bilhaj, Y. Suda, &quot;QoS management for VoIP networks with edge-to-edge admission control&quot;, in: Proceedings GLOBECOM 2001, vol. 4, 2001, pp. 2556 -2560. </li></ul>
    3. 3. Outline <ul><li>Motivation </li></ul><ul><li>Introduction </li></ul><ul><li>VoIP Network Management </li></ul><ul><li>Voice Quality Evaluation </li></ul><ul><li>Edge-to-Edge Admission Control </li></ul><ul><li>Network Dimensioning </li></ul><ul><li>Performance Evaluation </li></ul><ul><li>Conclusion </li></ul>
    4. 4. Motivation <ul><li>VoIP 目前是在 Internet 上最吸引人的一項 Service 。 </li></ul><ul><li>VoIP 在 Internet 上有許多的 Application 。 </li></ul><ul><li>因為 VoIP 通訊便宜,所以現在許多人開始使用 VoIP 的服務了,相對的 VoIP 就 必須達到和一般 PSTN 電話 一樣的品質 。 </li></ul>
    5. 5. Introduction <ul><li>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 . </li></ul>
    6. 6. Introduction(Cont.) <ul><li>A mechanism called call admission control is necessary to reject a new call when enough network spare capacity is not available . </li></ul>
    7. 7. Introduction(Cont.) <ul><li>Traditionally, the Internet has provided the best effort services , and has not supported call admission control. </li></ul><ul><li>However, admission control is necessary for guaranteeing QoS for real-time applications ( 如 : telephone service in the Internet). </li></ul>
    8. 8. Introduction(Cont.) <ul><li>Edge-to-edge measurement based admission control (EMBAC), 它使用了 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. </li></ul>
    9. 9. Introduction(Cont.) <ul><li>EMBAC 在 various network conditions 下,使用 call admission control 來讓 both directional voice flows 的 packet loss rates 維持在 a pre-determined value 之內 . </li></ul><ul><li>The results of voice quality evaluation is used to analyze possible problems, and if necessary to change parameters for admission control. </li></ul>
    10. 10. Outline <ul><li>Motivation </li></ul><ul><li>Introduction </li></ul><ul><li>VoIP Network Management </li></ul><ul><li>Voice Quality Evaluation </li></ul><ul><li>Edge-to-Edge Admission Control </li></ul><ul><li>Network Dimensioning </li></ul><ul><li>Performance Evaluation </li></ul><ul><li>Conclusion </li></ul>
    13. 13. VOIP NETWORK MANAGEMENT(3) <ul><li>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. </li></ul>
    14. 14. VOIP NETWORK MANAGEMENT(4) <ul><li>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. </li></ul>
    15. 15. VOIP NETWORK MANAGEMENT(5) <ul><li>The typical admission control parameters include: </li></ul><ul><li>(1) average packet lost rate for VoIP flows. (2) admission threshold. </li></ul>
    16. 16. Outline <ul><li>Motivation </li></ul><ul><li>Introduction </li></ul><ul><li>VoIP Network Management </li></ul><ul><li>Voice Quality Evaluation </li></ul><ul><li>Edge-to-Edge Admission Control </li></ul><ul><li>Network Dimensioning </li></ul><ul><li>Performance Evaluation </li></ul><ul><li>Conclusion </li></ul>
    17. 17. Voice Quality Evaluation(1/6) <ul><li>Measurement(1/2) </li></ul><ul><ul><li>Generate test call periodically </li></ul></ul><ul><ul><li>Each test call undergoes the same admission control as ordinary calls do </li></ul></ul><ul><ul><li>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. </li></ul></ul>
    18. 18. Voice Quality Evaluation(2/6) <ul><li>Measurement(2/2) </li></ul><ul><ul><li>Device B calculates instantaneous MOS values as well as the average MOS by comparing the original artificial voice signal and the received voice signal. </li></ul></ul><ul><ul><li>The VoIP gateway at the call-terminating PBX monitors and measures packet loss rate for the test call. </li></ul></ul>
    19. 19. Voice Quality Evaluation(3/6) <ul><li>Holding time for a test call is an important design parameter </li></ul><ul><li>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 </li></ul><ul><li>G723.1 coding and enhanced PSQM algorithm are used </li></ul>
    20. 20. Voice Quality Evaluation(4/6)
    21. 21. Voice Quality Evaluation(5/6)
    22. 22. Voice Quality Evaluation(6/6) <ul><li>MOS 2 is a critical value for users to notice voice quality degradation. </li></ul><ul><li>packet loss rate 2% is tolerable based on the measurement results of 20 or 60 sec measurement time </li></ul><ul><li>From these observations, 20 sec is a good candidate to obtain reliable voice quality evaluation. </li></ul>
    23. 23. Outline <ul><li>Motivation </li></ul><ul><li>Introduction </li></ul><ul><li>VoIP Network Management </li></ul><ul><li>Voice Quality Evaluation </li></ul><ul><li>Edge-to-Edge Admission Control </li></ul><ul><li>Network Dimensioning </li></ul><ul><li>Performance Evaluation </li></ul><ul><li>Conclusion </li></ul>
    24. 24. Edge-to-Edge Admission Control(1/6) <ul><li>End node A (source) to end node B (destination) through a selected path </li></ul><ul><li>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. </li></ul>
    25. 25. Edge-to-Edge Admission Control(2/6) Endpoint(Node O) Endpoint(Node T) Probe Request Probe Connect Voice Exchange Release Complete
    26. 26. Edge-to-Edge Admission Control(3/6) <ul><li>The probe request is sent from the call-originating node (node O) to the call-terminating node (node T) </li></ul><ul><li>Node O and node T may become a source or destination of the probe packet flows, as mentioned before </li></ul><ul><li>The probe request activates node T to initiate the probing and measurement operation. </li></ul>
    27. 27. Edge-to-Edge Admission Control(4/6) <ul><li>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. </li></ul><ul><li>Node O measures the packet loss rate for the probe flow from node T to node O, and conducts admission test. </li></ul>
    28. 28. Edge-to-Edge Admission Control(5/6) <ul><li>If the result of the admission test is success, node O transmit setup signal to node T. </li></ul><ul><li>If it is failure, node O terminates the call setup. </li></ul><ul><li>In parallel, node T measures the packet loss rate for the probe flow from node O to node T, and conducts admission test. </li></ul>
    29. 29. Edge-to-Edge Admission Control(6/6) <ul><li>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. </li></ul><ul><li>If it is failure, node T will reject setup request from node O. </li></ul>
    30. 30. Outline <ul><li>Motivation </li></ul><ul><li>Introduction </li></ul><ul><li>VoIP Network Management </li></ul><ul><li>Voice Quality Evaluation </li></ul><ul><li>Edge-to-Edge Admission Control </li></ul><ul><li>Network Dimensioning </li></ul><ul><li>Performance Evaluation </li></ul><ul><li>Conclusion </li></ul>
    31. 31. V.Network Dimensioning <ul><li>For dimensioning the network we need assume some parameters </li></ul><ul><li>Between source and destination ,There have “d“ links </li></ul>d links Source Destination
    32. 32. Traffic Matrix <ul><li>Location-to-Location VoIP traffic demands are represented by traffic matrix a[i,j] </li></ul><ul><li>“ i” represents the source </li></ul><ul><li>“ j” represents the destination </li></ul>
    33. 33. Parameter Assuming Explain Parameter The number of the VoIP calls k The maximum transfer rate for a VoIP call w (average number of frozen out calls) / (average number of active calls) Freezeout fraction Represents the percentage of time during which speech is present P Edge-to-edge peak pocket loss rate is no more than a pre-determined value “F” F the edge-to-edge blocking probability is no more than this pre-determined value “B” B
    34. 34. The peak values <ul><li>Blocking probability no more than B/d </li></ul><ul><li>Freezeout fraction no more than F/d </li></ul><ul><li>The Freezeout fraction is the “upperbound” for the packet loss rate </li></ul><ul><li>The capacity of the link is given as “kw” </li></ul>
    35. 35. Outline <ul><li>Motivation </li></ul><ul><li>Introduction </li></ul><ul><li>VoIP Network Management </li></ul><ul><li>Voice Quality Evaluation </li></ul><ul><li>Edge-to-Edge Admission Control </li></ul><ul><li>Network Dimensioning </li></ul><ul><li>Performance Evaluation </li></ul><ul><li>Conclusion </li></ul>
    37. 37. Simulation Model and Assumptions(1) <ul><li>For simplicity ,we ignore the amount of signaling flows ,because it is not significant compared with that of voice flows. </li></ul><ul><li>A PBX is modeled as a switch accommodating </li></ul><ul><ul><li>infinite number of subscriber lines </li></ul></ul><ul><ul><li>infinite number of outgoing and incoming trunks </li></ul></ul><ul><ul><li>each link has 5ms propagation delay </li></ul></ul>
    38. 38. Simulation Model and Assumptions(2) <ul><li>Calls originate </li></ul><ul><ul><li>according to Poisson distribution between a pair of call-originating and terminating locations </li></ul></ul><ul><li>Call holding time </li></ul><ul><ul><li>base on exponential distribution with the average three minutes. </li></ul></ul>
    39. 39. Simulation Model and Assumptions(3) <ul><li>We assume that </li></ul><ul><ul><li>blocking probability target B is 3% </li></ul></ul><ul><ul><li>freezeout target F is 1.5%. </li></ul></ul><ul><ul><li>For an established call, voice activity p is 30%. </li></ul></ul>
    40. 40. Scenario (1) <ul><li>A bottleneck may occur in the network due to traffic forecast error. </li></ul><ul><li>We select a link in the middle of the network as the bottleneck link, and decrease the capacity from the initial size. </li></ul>
    41. 41. Scenario (2) <ul><li>We use blocking probability and the peak packet loss rate for two seconds interval as the performance parameter. </li></ul><ul><li>We consider three cases </li></ul><ul><ul><li>No Admission Control </li></ul></ul><ul><ul><li>Admission threshold 2% </li></ul></ul><ul><ul><li>admission thresholds 10% </li></ul></ul>
    42. 42. RESULT
    43. 44. with admission control <ul><li>the peak packet loss rate is remarkably improved at the cost of acceptable increase in blocking probability, depending on the given admission thresholds. </li></ul>
    44. 45. without admission control <ul><li>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. </li></ul>
    45. 46. Outline <ul><li>Motivation </li></ul><ul><li>Introduction </li></ul><ul><li>VoIP Network Management </li></ul><ul><li>Voice Quality Evaluation </li></ul><ul><li>Edge-to-Edge Admission Control </li></ul><ul><li>Network Dimensioning </li></ul><ul><li>Performance Evaluation </li></ul><ul><li>Conclusion </li></ul>
    46. 47. CONCLUSION(1) <ul><li>Admission control works well to control packet loss rate under given network conditions. </li></ul><ul><li>We need to properly set the admission thresholds for each end node pair </li></ul>
    47. 48. CONCLUSION(2) <ul><li>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. </li></ul><ul><li>Feedback control based on voice quality and packet loss measurements may be used to dynamically adjust the admission threshold. </li></ul>
    48. 49. REFERENCES <ul><li>[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. </li></ul><ul><li>[2] L. Breslau, E. Q. Knightly, S. Shenker, I. Stoica, and H. Zhang, “Endpoint admission control: architectural issues and performance,” pp. 57-69, SIGCOMM’00, 2000. </li></ul><ul><li>[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. </li></ul><ul><li>[4] G. Bianchi, A. Capone, C. Petrioli, “Throughput analysis of end-to-end measurement-based admission control in IP,” INFOCOM 2000, 2000. </li></ul><ul><li>[5] V. E.lek, G. Karlsson, and R. Ronngren, “Admission control based on end-to-end measurement,” INFOCOM 2000, 2000. </li></ul><ul><li>[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. </li></ul><ul><li>[7] http://www.radcom-inc.com/products/internetsim.htm. </li></ul><ul><li>[8] http://www.genista.co.jp. </li></ul>
    49. 50. <ul><li>Each packet has 40 bytes overhead </li></ul><ul><ul><li>20 bytes IP packet header </li></ul></ul><ul><ul><li>8 bytes UDP header, </li></ul></ul><ul><ul><li>and 12bytes RTP header </li></ul></ul><ul><li>The maximum length of a packet is 60 bytes. </li></ul>
    50. 51. <ul><li>A VoIP gateway and a router are modeled as a queuing system. </li></ul><ul><li>Voice flows and probe flows are given individual classes and their own queues. </li></ul><ul><li>As mentioned in Ⅳ,voice flow is given high priority in packet scheduling than probe flows. </li></ul><ul><li>Specifically, non-preemptive priority scheduling is used. </li></ul><ul><li>Buffer size is 40 packets for voice flows and 20 packets for probe flows for each output link . </li></ul>
    51. 52. <ul><li>One packet is generated every 20 ms during active periods for each call. </li></ul><ul><li>Thus, the maximum rate for a VoIP call, w, mentioned in Sec.Ⅴ, is 24 kbps. </li></ul><ul><li>We assume probe calls have one second duration. </li></ul><ul><li>A size of a probe packet is always 60 bytes. </li></ul><ul><li>The admission threshold is set to 10 %. </li></ul>