IMPROVING QOS AWARE ACTIVE QUEUE MANAGEMENT SCHEME FOR MULTIMEDIA SERVICES

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1 I J I T E ISSN: (1-2), 2012, pp IMPROVING QOS AWARE ACTIVE QUEUE MANAGEMENT SCHEME FOR MULTIMEDIA SERVICES BOOBALAN P. 1, SREENATH N. 2, NANDINI S., KAVITHA U. 2 AND RAJASEKARI D. 2 1 Assistant Professor, Dept of IT, Pondicherry Engineering College, Puducherry, India, boobalanp@pec.edu 2 Profeesor,Dept.of CSE,Final Year IT, Pondicherry Engineering College, Puduchery, India nsreenath@pec.edu, nandini.itpec@gmail.com, kaavitha.u@gmail.com, rajasekari1990@gmail.com Abstract: QoS provision is essential in order to provide multimedia services. In this paper we have considered realtime multimedia service as IPTV Service because real time multimedia service are very much sensitive to delay and packet loss. In the existing system, they proposed Either Side Threshold(EST) scheme as QoS-aware AQM scheme, which can consider the level of quality that each hop can support in DiffServ environment. However, existing Either side threshold algorithm can t guarantee the required QoS especially for long service path because EST reduces the queue threshold when long service path increases. So, in this paper we proposed the Border Gateway Protocol(BGP) in order to improve video transmission rate even when long service path increases and thereby we can provide the reliable quality of service. Using the proposed scheme, we allow the network equipments such as routers and switches to recognize the quality of service information to each hop, and can provide the reliable multimedia service through the QoS-aware AQM and congestion control. Keywords: QoS, IPTV, AQM, BGP 1. INTRODUCTION The video and audio applications such as IPTV(Internet Protocol TV) and Voice over IP based on transmission control protocol (TCP) applications emphasizes the importance of quality of service (QoS) provisioning. QoS has been addressed through Active Queue Management Scheme and Differentiated Service environment. Active Queue Management Scheme has been mainly used to control the congestion which takes place during packet transmission. Differentiated Service environment is a simpler scheme which is used for IP networks to provide reliable quality of service. In this paper we have introduced Active Queue Management Schemes considering both audio and video quality level that users request service in Differentiated service network environment. This scheme used to reduce endto-end network delay and packet loss rate which leads to derive the QoS level range for multimedia service. In order to provide the reliable quality of service, we must provide the quality of service for both short service path and long service path. In the existing Active Queue Management Scheme, they don t consider the maximum end to end delay and loss rate. So they have implemented Either side threshold algorithm[1] for reducing delay and loss rate in order to provide reliable multimedia service. This algorithm is well suited for short service path. But when long service path increases, EST reduces the queue threshold. It leads to increase in end to end delay and packet loss rate, thereby, the reliable service cannot be provided. In order to overcome this problem, we have implemented Border Gateway Protocol (BGP). It is suitable for both short service path and long service path. It helps to improve the video transmission rate as well as it helps to provide the reliable quality of service. In this system we have considered the parameters such as jitter, delay, throughput, loss rate, forwarding node selection, buffer size and number of end to end hops. This parameter mainly used to provide the level of traffic for each hop. This will help to reduce the traffic rate during packet transmission. Using this proposed scheme we can reduce the

2 262 Boobalan, P., Sreenath, N., Nandini S., Kavitha U. and Rajasekari D. delay and loss rate for both short service and long service path. II. DESIGN OF ACTIVE QUEUE MANAGEMENT SCHEME IN DIFFSERV NETWORK The QoS-aware AQM scheme which can recognize the network quality parameters such as delay, jitter and packet loss rate, and the number of end-to-end hops. In our paper, we have introduced as Active Queue Management Schemes such as Drop tail, Random Early Detection, Dynamic Random Early Detection, Dynamic Slope Random Early Detection etc. These schemes can also act as congestion avoidance mechanism. Drop tail is the simple queue management scheme. In drop tail mechanism, arrival packets are dropped when queue overflow occurs. Drop-tail incurs large queue length and high packet loss rate at congested links. In Random Early Detection mechanism, a RED gateway drops incoming packets with a dynamically computed probability when the average number of packets queued exceeds a threshold. This probability increases with the average queue length and the number of packets accepted This approach controls the queue length more effectively and also it is simple to implement. In Double Slope Random Early Detection is a queue management scheme which achieves higher throughput and it act as early congestion notification mechanism. The Active Queue Management scheme has been used for setting for the queue. Once the arriving packet has been classified it has to be distributed over the queue. In this paper we have used three technique to set the queue. Three techniques are priority queueing, Self Clocked Fair Queueing (SCFQ), Start-Time Fair Queueing (SFQ). In priority queueing, traffic is prioritized with a priority-list, applied to an interface with a priority-group command. This can be done based on priority scheduling (i.e). When the router is ready to transmit a packet, it searches the high queue for a packet. The SCFQ technique is mainly used for fairness property by introducing a new virtual time function and it can used to reduce complexity. Likewise SFQ also a fairness property and it can be used to accomplish low delay and low loss requirements. III. IMPLEMENTATION OF BORDER GATEWAY PROTOCOL (BGP) In the existing system, they have implemented Either side threshold algorithm[1] for reducing delay and loss rate in order to provide reliable multimedia service. There is no problem while implementing this algorithm for short service path. But when long service path increases, EST reduces the queue threshold. That is, it is impossible to set the dynamic threshold to every node. It leads to increase in end to end delay and packet loss rate, thereby, the reliable service cannot be provided. In order to overcome this problem, we have implemented Border Gateway Protocol(BGP) by considering the parameters such as delay, loss rate, jitter, throughput, increasing the buffer size and no.of end to end hops. This parameter mainly used to provide the level of traffic for each hop. It helps to improve the video transmission rate as well as it helps to provide the reliable quality of service. Using this proposed scheme we can reduce the delay and loss rate for both short service and long service path. Border Gateway Protocol(BGP) is used for exchanging routing information between gateway hosts (each with its own router) in a network of autonomous systems. BGP is often the protocol used between gateway hosts on the Internet. The routing table contains a list of known routers, the addresses they can reach, and a cost metric associated with the path to each router so that the best available route is chosen. Hosts using BGP communicate using the Transmission Control Protocol (TCP) and send updated router table information only when one host has detected a change. Only the affected part of the routing table is sent.

3 Improving QOS Aware Active Queue Management Scheme for Multimedia Services 263 The goal of the simulation test was to validate the behavior of multiple reflectors inside a BGP cluster[2]. AS 0 contains two clusters. The first cluster contains two reflectors: nodes 0 and 1. Reflection clients of nodes 0 and 1 are nodes 2, 3, and 4. The second cluster has one reflector node (5), with nodes 6 and 7 as its clients. The three reflectors (nodes 0, 1, and 5) are fully connected via ibgp sessions. External BGP (ebgp) peer sessions exist between nodes 2 and 8, as well as between nodes 7 and 9. Using standard Internal Border Gateway Protocol (IBGP) configurations, all BGP systems within an Autonomous System (AS) must peer with all other BGP systems, forming a full-mesh configuration. This presents scaling concerns, as all external information must be propagated/ distributed to all BGP systems within the AS, resulting in far more information being shared between the IBGP peers then is necessary. BGP Route Reflectors[3] provide a mechanism for both minimizing the number of update messages transmitted within the AS, and reducing the amount of data that is propagated in each message. The deployment of BGP Route Reflectors lead to much higher levels of network scalability and throughput. IV. HIGH DATA TRANSMISSION Data transmission is the amount of data that can be transmitted from the source node to the destination node in the communication network. The communication may be through cables or the wireless systems. High data transmission is the matter of considering the time factor that is how long it takes the time to send the data to the destination or how fast it sends the data to the destination. So we can able to measure the data transmission rate. In general the greater the bandwidth of the given path, the higher the data transfer rate. So data transfer depends upon the bandwidth that we fixing for the transmission. In this paper since we mainly concentrating on long service path, more hops are required to deliver the reliable quality of service. That is, more than two hops. Based on this, We send the data in different traffic flows in the network. The types of different traffic flows are 1. network control,2.video, 3.audio, 4.best effort. In this paper, we are considering audio and video traffic since it is IPTV service. AUDIO Here the audio is transmitted to the destination from the source. In audio transmission the packets are sent as the data and the audio traffic is different from the current internet traffic. The audio data can be sent by UDP, TCP and HTTP, but the majority of the audio data takes place through the User Datagram Protocol. The real audio data can be sent at consistent bit rates at medium time scales like 10s. Real audio sessions employ one or two flows and utilize multiple protocols. The two flows use a UDP for audio data and use TCP for the control. Those using one flow use the TCP alone. The audio flows are very long with a mean duration. The audio data flow which contains the encoded audio information. The data flow is easily identifiable by the fact that its bandwidth is several orders of magnitude larger than the control flow. Audio traffic is highly unidirectional with the bulk of the traffic from the server. UDP traffic does not include transport level congestion control which implies that much audio traffic depends on application level congestion control. Development and deployment of TCP friendly congestion control for audio data is important for network stability. Two approaches to TCP friendly audio are new audio protocols and a general congestion manager. Here real audio traffic is dominated by specific packet sizes. In audio traffic type the audio flow durations are longer than internet web flows. Since many of the audio flow uses UDP it requires application specific congestion control. Audio flows exhibit a significant amount of regularity in packet length, bit rate, and inter

4 264 Boobalan, P., Sreenath, N., Nandini S., Kavitha U. and Rajasekari D. packet arrivals and it does not exhibit steady state characteristics. VIDEO One of the most important traffic in packet switched network is video. Since it is delay sensitive media the network must allocate resources to maintain a Quality of service. Traffic characterization of video sources determines the resources required to support video connections. Here the video data is transmitted as the data in the network. In video traffic type we send the video data in the form of frames only. V. PERFORMANCE ANALYZATION In this paper, we have considered the parameters such as delay, jitter, loss rate, throughput and no.of end to end hops. These parameters mainly helps to improve the video transmission rate. This section presents a discrete-time queueing model for the performance analysis of proposed improving QoS-aware AQM method by make use of Border Gateway Protocol(BGP), which includes the arrival model, system model description and performance measures. In the discrete-time queueing system for the proposed improving QoSaware AQM by make use of Border Gateway Protocol (BGP), we will assume that departures always take place before arrivals in any unit time (slot). There is a finite waiting room of K packets, including any in service.the queueing discipline is first-come firstserved. The drop tail mechanism used to discards the packets when it is overflow. The mean queue length ad mean arrival rate are measured over each time window k and the information is used to compute the value of the threshold for the next time window k+1 to bound the delay at the required value. Border Gateway Protocol guarantees the requested QoS level with lesser delay and loss rate. To analyse the AQM performance, we use M/M/1/K model because in each state of the source model, the probability that a time-slot contains an arrival is a Poisson process with a parameter that varies according to a Markove process. In M/M/1K queueing model, the delay between end hops and packet loss rate can be calculated by using following formulas. In order to perform the steady state analysis of the system, we use following formulas (1) ~ (6). In steady state, the queue state probability is calculated through equation (1). k 1 λ 1 λ / µ λ o K + 1 i= 0 1( /) p = for 0 k < K p k = µ λ µ µ 0 for k K k (1) where pk is the stationary state probabilities in queue, λ is packet arrival rate, µ is packet service rate. The aggregate mean queue length (MQL) can be expressed from the equilibrium state probabilities pk: MQL N = ipi (2) i= 0 The IP Network delay is generally the round trip delay for an IP packet within the IP network. This may be measured using BGP since it is a real time traffic protocol. It is important to note that some IP networks treat the RTP packets used to transport both audio and video packets with higher priority than other packet types and hence the delay measured by RTCP may be different to that actually experienced by both audio and video packets. IP network delay comprises the sum of transmission delays and queuing delays experienced by a packet traveling through the collection of routers. The queueing delay can be obtained from Little s result for this finite capacity queue as: L W = (3) S where S is the mean throughput of the discretetime finite capacity queue given by the fraction of time the server is busy. The network throughput is the average rate of successful packets delivery over a communication channel. The throughput is measured in data packets per second. S = (1 P 0 ) β (4) The blocking probability for an M/M/1/K system with ρ < 1 is well-known. (1) ρ ρ 1 ρ K p k = K+ 1 (5)

5 Improving QOS Aware Active Queue Management Scheme for Multimedia Services 265 where ρ(= λ/µ) is utilization. Jitter is the variation in the time between packets arriving, caused by network congestion, timing drift, or route changes. T jitter µ jitter Tmax (6) hop _ counts Using the proposed QoS-aware AQM scheme with Border Gateway Protocol, it provides mechanism for both minimizing the number of update messages transmitted within the AS, and reducing the amount of data that is propagated in each message in terms of long service path. The deployment of BGP Route Reflectors lead to much higher levels of network scalability and throughput. It helps to improve the video transmission rate with lesser delay and loss rate, thereby, we can provide the reliable quality of service. Figure 1c: Loss Rate Comparison Figure 1d: Jitter Comparison Figure 1a Figure 1a: Delay Comparison The result shows the performance analysis of Border Gateway protocol over the existing Either side threshold algorithm. The figure 1(a) shows the throughput comparison of BGP with EST. This shows that performance is good when proposed algorithm is compared with existing system. The figure 1(b) shows the delay comparison. This shows that the delay has been reduced while implementing in BGP when compared to existing EST algorithm. Since we are considering wired network, there must be some minimum delay. The figure 1(c) shows the loss rate comparison. This shows that the packet loss rate also has been reduced icase of BGP when compared to existing system. The figure 1(d) shows the jitter comparison. This shows that packet processing delay variation can be stabilized in terms of BGP. This means that it is possible to provide multimedia service with stable QoS level.

6 266 Boobalan, P., Sreenath, N., Nandini S., Kavitha U. and Rajasekari D. VI. CONCLUSION QoS provision is essential in order to provide multimedia services. In this paper we have considered real-time multimedia service as IPTV Service because real time multimedia service are very much sensitive to delay and packet loss. We must provide reliable quality of service for both short service path and long service path. So, we proposed the Border Gateway Protocol (BGP) in order to improve video transmission rate even when long service path increases and thereby we can provide the reliable quality of service. Using the proposed scheme, we allow the network equipments such as routers and switches to recognize the quality of service information to each hop, and can provide the reliable multimedia service through the QoS-aware AQM and congestion control. References [1] Hyun Jong Kim, Pyung-Koo Park, Ho Sun Yoon, Seong Gon Choi, QoS-aware Active Queue Management Scheme for Multimedia Services, IEEE Publications, February [2] C. Huitema, Routing in the Internet. Upper Saddle River, NJ: Prentice Hall, [3] T. Bates, R. Chandra, and E. Chen, BGP Route Reflection an Alternative to Full Mesh IBGP, RFC2796, April [4] Pablo J. Argibay-Losada, Andrés Suárez-González, Cándido López- García, Manuel Fernández-Veiga, A New Design for End-to-end Proportional Loss Differentiation in IP Networks, Computer Networks, Available online, Dec [5] Xiaolong Jin, Geyong Min, Performance Analysis of Priority Scheduling Mechanisms under Heterogeneous Network Traffic, Journal of Computer and System Sciences, 73(8), 2007, [6] Hussein Abdel-Jaber, Mike Woodward, Fadi Thabtah, Amer Abu-Ali, Performance Evaluation for DRED Discrete-time Queueing Network Analytical Model. Journal of Network and Computer Applications, September [7] L. Guan, M.E. Woodward, I.U. Awan, Control of Queueing Delay in a Buffer with Time-varying arrival Rate, Journal of Computer and System Sciences, 72(7), , [8] ITU-T Std. Recommendation Y.1541, Network Performance Objectives for IP-based Services, [9] Yang Xiao, Moon Ho Lee, Nonlinear Control of Active Queue Management for Multiple Bottleneck Network, IEICE Trans. Commun., Vol. E89-B, No. 11, , [10] A. A. Akintola, G. A. Aderounmu, L. A. Akanbi, and M. O. Adigun, Modeling and Performance Analysis of Dynamic Random Early Detection (DRED) Gateway for Congestion Avoidance, InSITE2005, June [11] Changhee Joo, Jaesung Hong and Saewoong Bahk, Assuring Drop Probability for Delay-insensitive Traffic in a Differentiated Service Network, CCNC 2005, January 2005, [12] Alhussein A. Abouzeid, Sumit Roy, Modeling Random Early Detection in a Differentiated Services Network, Computer Networks, 40(4), 2002, [13] Hideyuki Shimonishi, Ichinoshin Maki, Tutomu Murase and Masayuki Murata, Dynamic Fair Bandwidth Allocation for DiffServ Classes, ICC 2002, Vol. 4.