Quality of service for VoIP - Dissertation Example

Running Head: QUALITY OF SERVICE FOR VOICE OVER IP (VoIP) Quality of Service for Voice over IP (VoIP) [The [The of the Institution] Quality of Service for Voice over IP (VoIP) TABLE OF CONTENTS Chapter One: INTRODUCTION pages 1.1 Background………………………………………….……………………………………3 1.2 Purpose of the study……………………………………………………………………...4 1.2.1 Aim/Hypothesis……………………………………………………………………..7 1.2.2 Objectives……………………………………………………………………………7 1.3 Dissertation structures……………………………………………………………………7 1.3.1 The Literature Review………………….…………………………………………....7 1.3.2 The Research Design Quality of Service ………………………………………........8 1.3.3 Perceived Quality of Service Analysis………………………………..………….….8 1.3.4 Conclusion………………………………………………………………….………..8 Chapter Two: LITERATURE REVIEW (Voice Over IP (VoIP)) 2.1 Introduction…….……………………..……………….…………………………………9 2.2 Intrinsic Quality of Service……………………..…………………………………..........10 2.2.1 Principle QOS Measurement…….. …..…………………………………………...11 2.2.2 Analyzing system Perceived QoS... ……………….………………………………14 2.2.3 Particular System (Intrinsic QoS) Technology..……………………………..……15 2.2.4 Implementing Voice Over IP …………..…………………………………………16 2.3 Perceived Quality of Service………………….. ………………………………….…….17 2.4 Preferential Quality of Service.……….. .…………………..………………………..….18 2.5 Optimizing VoIP Performance with an Extreme Networks Infrastructure……………...20 2.6 Conclusion…………………………………………………..……………………..…….22 Chapter One INTRODUCTION 1.1 Background No doubt businesses have usually maintained divide voice and data networks. A key reason for this is that heritage network technologies could not meet the varied performance supplies of voice and data. According to the expert analysis new advances in networking technology, counting fast Ethernet, wire-speed switching and Policy-Based Quality of Service management, have complete it probable to build converge voice and data networks (F. Robles). According to the expert report that converged networks decrease costs by get rid of superfluous hardware, communications facilities and sustain staffs. Converged networks also allow a latest generation of incorporated voice/data applications. For instance, users of web-based e-commerce applications can sight product information as talking with customer service agents in a call center. by join networks this can be done throughout a single network link. The focal point of the majority converged network strategies is voice over IP (VoIP). VoIP refers to the transmission of telephone discussion over a packet-switched IP network. This IP network might be as small as a single subnet private LAN, or as huge as the public Internet. Voice over IP on the LAN (National Institute of Standards and Technology, 2003). By VoIP on the LAN, phone dialogue are rehabilitated to a stream of IP packet and sent above an Ethernet network. This network is typically incomplete to a building or campus. As VoIP technology matures, new exchange technique may appear while offered ones become outdated. In spite of the process that is used to adapt VoIP traffic for LANs, one basic process will linger the same: VoIP traffic will forever negotiate the LAN as a stream of IP packets (W.C. Hardy, 2001). Figure 1 (Anonymous, 2002) 1.2 Purpose of the study For the purpose of this research will be take place Voice over IP and QoS. There details are: Voice over IP the broadcast of voice over packet-switched IP networks is one of the mainly significant up-and-coming trends in telecommunications. As with lots of new technologies, VOIP introduces together security risks and opportunities. VOIP has a extremely dissimilar architecture than established circuit-based telephony, and these differences effect in important security issues. Lower cost and better elasticity are between the promises of VOIP for the venture, but VOIP should not be installed devoid of cautious deliberation of the security troubles introduced (W.C. Hardy, 2003). Many administrators may incorrectly presume that since digitized voice travels in packets, they can just plug VOIP components into their already protected networks and stay secure. Though, the procedure is not that simple. This publication give details the onfront of VOIP security for agency and commercial users of VOIP, and sketch steps wanted to assist secure an organization’s VOIP network. VOIP security deliberation for the public switched telephone network (PSTN) is mostly outside the extent of this document (International Telecommunications Union , 2002). VOIP Systems Take a Broad Diversity of Forms, counting traditional telephone handsets, conferencing units, and mobile units. In adding to end-user equipment, VOIP systems comprise a diversity of further components, counting call processors/ call managers, gateways, routers, firewalls, and protocols. The majority of these components have complement used in data networks, but the routine demands of VOIP mean that commonplace network software and hardware must be supplemented by special VOIP components. Not only does VOIP need higher recital than mainly data systems, serious services. One of the major sources of bewilderment for those new to VOIP is the hypothesis that because digitized voice travels in packets just like further data, accessible network architectures and tools can be used with no change. Though, VOIP adds a number of complications to obtainable network technology, and these troubles are exaggerated by security thought (P. Mehta and S. Udani, 2004). Quality of Service (QoS) is basic to the process of a VOIP network that assembles users’ quality prospect. Though, the completion of a variety of security measures can reason a noticeable worsening in QoS. These complications range from firewalls delaying or blocking call setups to encryption-produced latency and holdup variation. Since of the time serious nature of VOIP, and its low acceptance for disturbance and packet loss, lots of security measures apply in traditional data networks are just not applicable to VOIP in their present form; firewalls, interruption detection systems, and further components must be particular for VOIP. Current VOIP systems use either a proprietary protocol, or one of two standards, H.323 and the Session Initiation Protocol (SIP) (B. Goode, 2002). Voice over IP on the WAN VoIP traffic can be sent more than a diversity of IP-based broad area networks (WANs). An IP-based WAN can be one of the subsequent: • A private venture WAN made up of leased lines, a frame spread service or an ATM service. • A public IP carrier such as Qwest or Level 3 • The public Internet For a business that is in view of voice and data convergence, putting VoIP on the WAN is as significant as VoIP on the LAN. The reasons for this are chiefly economical. Cost savings can be instant when long coldness phone calls are unfocused from the public switched telephone network (PSTN) and sent over an obtainable IP-based WAN. Running VoIP traffic on the WAN can be done in several ways: • If the voice traffic has previously been rehabilitated to VoIP traffic on the LAN, then the VoIP traffic will be transmitted above the IP-based WAN like any other IP data traffic (R. Barbieri, 2002). Figure 2 (Telcordia Technologies, 2004) APPROACH TO THE DISSERTATION: 1.2.1 Aim/Hypothesis The aim of this project is to examine how Voice IP perceive the QoS for users and how Voice IP influences the network. Moreover, during the past years, networking in multinational organizations has suffered some decreases (C-N. Chuah, 2000). This project has also the purpose to find what measures the QoS has taken in relation to the users, in order to achieve customer satisfaction and to creation of customer loyalty. The hypothesis is that some problems exist between Customer Service and Voice IP. There are not all employees satisfied with the measures that QoS has taken. 1.2.2 Objectives The main objectives are: 1. Subject of Evaluation (Perceived QoS). 2. User of the System (Perceived QoS). 3. Research of The Technology Being Used In That Particular System (Intrinsic QoS) 4. The Mechanisms Determining System Performance Those Concerns (Intrinsic QoS). 5. Formalize relations between system performance and user perception of quality. 6. Quality of Service 7. Intrinsic Quality of Service 8. Perceived Quality of Service 9. Preferential Quality of Service 1.3 Dissertation structures Here is a brief summary of what the following chapters are going to include, which will help the readers to have an idea of what they are going to read in each chapter. The dissertation will be divided into four sections: 1.3.1 The Literature Review In this chapter existing information from secondary research is going to be given like books, journal articles, reports and electronic databases. The purpose of this chapter is to give the knowledge for the overview of Voice IP and how Voice IP is working in the organizations. This chapter is going to help the author to improve the knowledge from the research subject and to create the appropriate questions in order to give the focus on the research objectives. 1.3.2 The Research Design Quality of Service After gathering the existing information (Literature Review) the next chapter has the purpose to offer a concise description of the research design and methodology. The author is being used both qualitative and quantitative research for more complete image of the research situation. The methods that are going to be used for the purpose of this research are questionnaires with close Case Study Questions for the customer of the IT organizations and interviews with open questions to the two Network Administrators of the organizations. 1.3.3 Perceived Quality of Service Analysis After the research is being completed, with the help of some electronic methods like ‘record’ they will give the analysis of the questionnaires and also the interviews. The data will be presented in table charts. 1.4 Conclusion Based on previous analysis and with the combination of the literature review the conclusion will be throwing. In the final chapter the author is going to compare the research questions he had at the beginning of the research with the final results. Moreover he will give some recommendations from the results and the analysis, which will help the hotels to be improved. Chapter Two LITERATURE REVIEW (Voice Over IP (VoIP)) 2.1 Introduction This sections highlight the attention in Voice over IP (VoIP) has augmented slowly over the past few years. Enterprises, ISPs, ITSPs (Internet Telephony Service Providers), and carriers sight VoIP as a feasible way to apply packet voice. There are several reasons for implementing VoIP classically comprise toll-bypass, network consolidation, and service meeting. No doubt, toll-bypass permits long-distance calls to be placed with no incurring the usual toll charges. Throughout network consolidation, voice, video, and data can be carried over a single network communications, thus make simpler network management and plummeting cost during the use of ordinary equipment (B. Goode, 2002). By means of service convergence, improved functionality can be implemented during the combination of multimedia services. This full addition permits new applications, such as incorporated messaging and web call center. Though, scheming a VoIP network requires cautious preparation to make sure that voice quality can be correctly uphold. This chapter examines the issues that influence voice quality and the test and study policy for a VoIP network (Zwicker, 2001, pp. 115−126) A figure 2 shows the main mechanism of a VoIP network. The gateway change signals from the customary telephony interfaces (POTS, T1/E1, ISDN, E&M trunks) to VoIP. An IP phone is a terminal that has resident VoIP hold up and can connect straight to an IP network. In this research, the term terminal will be used to pass on to moreover a gateway, an IP phone, or a PC with a VoIP border (Irii et al, 2001, pp. 79−87). If you analyze then you come to know that the server give management and administrative functions to hold up the routing of calls crossways the network. In a system based on H.323, the server is recognized as a gatekeeper. Beyond the shadow in SIP/SDP, the server is a SIP server. In a system support on MGCP or MEGACO, the server is a call agent. lastly, the IP network provides connectivity among all the terminals. The IP network can be a private network, an Intranet, or the Internet (Dimolitsas, 2002, pp. 317−324). Figure 3. VoIP Components (G. Egeland, 2000) On one occasion a call has been set up, tongue will be digitized and then broadcast crossways the network as IP frames. Voice samples are first sum up in RTP (Real-time Transport Protocol) and UDP earlier than being broadcast in an IP frame (Herre et al, 2002). Figure 3 shows an instance of a VoIP frame in both LAN and WAN Figure 3. Encapsulation of VoIP Frame (R. Straka, 2002) 2.2 Intrinsic Quality of Service Beyond the shadow the further haziness in descriptions of packet-switched voice services that have to be elucidating at the outset is what is destined by quality of service. There are at slightest three definitely dissimilar referents for the term QoS that come into view in technical dialogue on the topic. 1. Abilities for, or the classes distinct to attain, preferential treatment of dissimilar types of traffic in packet-switched networks. In a great deal of the data networking literature, chiefly that commerce with the Internet protocol, the term QoS is unspoken to mean a preferential class of service to which a exacting transmission may be allocate (Kalittsev, 2002, pp. 11−15). The class is shaped by requirement of exacting handling or routing ability that can be employed to have enough money specified types of traffic main concern use of the obtainable bandwidth. In this research desertion, QoS refers to together class of service (CoS) and type of service (ToS). The essential objective of CoS and ToS is to attain the bandwidth and latency wanted for a exacting application (Beerends et al, 2001, pp. 1−8). A CoS enables a network administrator to group dissimilar packet flows, each have separate latency and bandwidth requirements. A ToS is a meadow in an Internet Protocol (IP) slogan that enables CoS (Moore et al, 2003, pp. 750−753). 2. Inherent Quality Of Service. When traffic is approved via a packet switched network, with or with no application of QoS capability, the treatment of the traffic will attain positive operational recital levels beneath a variety of levels of demand (Anonymous, 2001). Those individuality that can be measured by the supplier devoid of reference to user insight of quality but that will, nevertheless, influence user insight of quality are referred to companies as important intrinsic QoS. It is usually decided that for packet-switched services such inherent QoS is typify by Latency. The time it takes a packet to get crossways the packet switched network to its target (R. Sinden, 2002). The unpredictability in packet latency Dropped packet rate. The rate with which packets do not get to their purpose in time to be used For some class of traffic these uniqueness will, in general, depend on the size of the insist and the amount of bandwidth owed to that traffic (K. Percy, 2003). 3. Perceived Quality Of Service. No doubt, a perceived quality of service is illustrious from intrinsic QoS as being what fallout when the service is in fact used. Perceived QoS is, then, strong-minded by what user’s knowledge as the belongings of intrinsic QoS on their infrastructure activities, in their environment, in treatment their demand, and how they respond to that knowledge in light of their individual expectations. It is apparent, quite than intrinsic, QoS that eventually determines whether a user will be content with the service delivered (J. Larson, T. Dawson, 2001). 2.2.1 Principle QOS Measurement No doubt, whether VoIP is implemented by devoted or communal facilities, QoS is an significant thought. A QoS-enabled network will distinguish among dissimilar types of traffic and present dissimilar treatments. Using standard-based methods, this can be attain using either the TOS (Type Of Service) bits field in the IP header, or from side to side the use of signaling protocols such as RSVP (Resource reSerVation Protocol) plus MPLS (Multi-Protocol Label Switching) (K. Siddiqui, M. Kamran, S. 2001). If we analyze then we come to know that routers and switches too support prioritization stand on physical port, protocol, IP addresses, transport addresses, or still frame length (J. Thalhammer). In examination whether an IP network can sustain VoIP, the efficiency of its QoS must be assess. In exacting, questions that are of attention comprise the following (H. Schulzrinne, 2004): How can the network distinguish VoIP traffic from further types of traffic? In the occasion of network overcrowding, what is the frame defeat rate of voice vis-à-vis data? What is the average stoppage skilled by voice frames? What is the standard jitter? Can the network level if there are a big number of flows? The subsequent series of tests were intended to examine the performance of ordinary prioritization schemes in routers. All mention figures in this research desertion shows the test pattern. Using the SmartBits, a number of traffic flows were distinct and insert into the IP network (M. Marjalaakso, 2001). These traffic flows symbolize dissimilar types of traffic. They different from one a further in terms of IP addresses, IP precedence, port numbers, size of packets, or a mixture of these issue (Jablon, 2003). By observing the production of the IP network, we can decide how the network luxury the traffic in a different way. To see the result of QoS, there have to be several reserve contention. So, the bandwidth of the sequential link was configured to be 500 kbps (Anonymous, 2001). Since traffic inwards from a 10 Mbps Ethernet port, overcrowding started to happen when the input load exceed regarding 6%.6 this number was quite random. If the WAN bandwidth was distorted, the congestion point would also change (J. Rosenberg, 2001). Figure 4. QoS Test Configuration (O. Arkin, 2002) Priority Queuing In main concern queuing, traffic is secret into divide queues for instance, high, medium, normal, and low. The queues are repair in the severe order of main concern. In other words, the lofty priority queue must be unfilled before the average priority queue is serviced (J. Halpern, 2002), and both the high and average priority queues must be unfilled before the usual queue is serviced and so on. In this instance, when the input load exceed the WAN bandwidth (at around 6%), frames were surplus. The low priority traffic was surplus until there was none left. Then the usual main concern traffic was discarded go after by medium priority traffic (Anonymous, 2001). The results exemplify a possible problem with this policy. If there is a high volume of elevated priority traffic, then it will press out all traffic of lesser priority (A. Romeo, 2003). Figure 6. Classical Priority Queuing (L. Phifer,2001) 2.2.2 Analyzing system Perceived QoS If we analyze then we come to know that the global standard for packet-based multimedia communication like VoIP, give telephone functionality that is similar to the public switched telephone network. But the other key obligation for winning VoIP communications is quality of service (QoS) (A. Conry-Murray, 2002). Voice infrastructure needs networks with extremely low latency, low jitter, and negligible packet loss (P. Hochmuth and T. Greene, 2002). Two issues drive these QoS requirements: • Extremely elevated user expectations • The technical necessities of real-time voice communications No doubt, telephone users have extremely lofty prospect since they are familiar to the QoS give by the PSTN and private PBXbased networks. These connection-oriented, circuit-switched networks give each user with devoted bandwidth for the period of each call. The consequence is very low latency and jitter, and minimal disturbance due to "noise" on the connections (Anonymous, 2002). Low latency permit users to take on natural conversations. Users be different in their holdup lenience, but a good rule of thumb is to limit one-way holdup to about 150 milliseconds. This stoppage budget comprises the processing stoppage initiate by the end systems benefit the latency of the network (Telcordia Technologies, 2004). When coders/decoders in VoIP terminals squeeze voice signals they pioneer three types of holdup: • Dispensation, or algorithmic, holdup the time necessary for the codec to encode a single voice frame • Look in front delay – the time necessary for a codec to inspect part of the after that frame while encoding the present frame • Frame delay – the time necessary for the distribution system to transmit one frame The subsequent are several commonly used ITU-T standard codecs and the amount of one-way holdup that they initiate: • G.711 uncompressed 64 Kbps speech adds insignificant holdup • G.729 encodes speech at 8 Kbps and adds a one-way holdup of concerning 25 ms. • G.723.1 encodes speech at 6.4 Kbps or 5.3 Kbps and adds a one-way delay of concerning 67.5 ms. In wide-ranging, better levels of density pioneer more holdups and need lower network latency to preserve good voice quality. When using G.723.1 compression, for instance, the network component of one-way stoppage should not exceed: 150ms – 67.5 ms = 82.5 ms (G. Egeland, 2000). 2.2.3 Particular System (Intrinsic QoS) Technology The query of voice quality is a leading, but by no means elite, anxiety that users state when they ask for reassurances that the quality of their voice services will be satisfactory. A complete explanation of what determines users’ awareness of quality of their telephonic voice services comprises no less than six self-governing service individuality besides the voice quality that has been the subject of this literature review. The main characteristics are explain in references as being Accessibility, the aptitude to start a call when desired Routing speed, the speed with which calls are set up Connection dependability, the consistency of the procedure of setting up a call (R. Straka, 2002) Routing dependability, the consistency of the process of routing the connection to the demanded destination Connection continuity, the aptitude to preserve a connection with satisfactory quality until it is no longer wanted Disconnection reliability, the dependability of the system reply to the instructions to cease, and so stop billing for, the connection Internet zealots will right away object to the names known these feature of perceived QoS, since Internet voice services are connectionless. Though, the imagery of each that follows ought to obviously describe precisely what is concerned, even for “connectionless” connections. 2.2.4 Implementing Voice Over IP No doubt, One of the key confrontation in implementing VoIP is to design and build an IP-based network that get together stringent QoS requirements and is similar in performance to conservative circuit-switched telephone networks. The high latency onward and best-effort release offered by customary software based routers is usually not satisfactory for streaming traffic like VoIP since it does not give utmost latency guarantees or least amount bandwidth guarantees (O. Arkin, 2002). From the viewpoint of an IP-based Ethernet network, a VoIP packet enclose part of a telephone discussion is no dissimilar than a data packet containing part of an e-mail. Both packets are conventional on an entrance port of an Ethernet switch and need to be onward out the way out port of an Ethernet switch. From the viewpoint of the end points, dissimilar types of traffic have very unlike requirements. For instance, e-mail traffic is classically grip using the store-and-forward procedure. It does not have to occur in real-time. An e-mail broadcast does not have to be streamed from one end-point to one more end-point to be winning. Equally, VoIP traffic is a real-time procedure. To total a successful VoIP session, the network has to be able to hold up the streaming of VoIP packets among the two end-points for the period of the phone discussion. VoIP traffic requires a network to assurance bandwidth and ability for VoIP traffic (L. Phifer,2001). To carry VoIP traffic every time and dependably, a network must be able to offer three things: • Packet-forwarding latency that does not go beyond the utmost bearable level for a VoIP discussion • Packet-forwarding jitter, which is the difference in latency over time, which does not go beyond the utmost bearable level to maintain a VoIP session • Guaranteed network bandwidth and ability for VoIP sessions throughout periods of network overcrowding In other words, a network wants to give presentation low latency and low jitter – and defense quality of service. The majority VoIP sessions need one-way latency of not more than about 150 milliseconds. This holdup budget is abridged by any delays bring in by codecs in the end systems. When round-trip holdup go beyond about 300 ms., natural human discussion becomes hard. Anyone who has tried to carry on a discussion on a satellite link has skilled the influence of long delays (Internet Draft, 2004). Depending on the type of voice-compression technique used, each one-way VoIP transmission need flanked by 32 Kbps to 64 Kbps of bandwidth. Some density methods such as G.729 take the bandwidth necessary below 8 Kbps. As you can see, the bandwidth that is requisite for each VoIP session is relatively low. The confront is to make that bandwidth obtainable in spite of network utilization. 2.3 Perceived Quality of Service The purpose of go through of this research is perceived quality of packet switched voice services. The function is to explain and propose applications for techniques by which impartially deliberate individuality of those packet-switched voice services can be analyze to forecast user pleasure. Still the thing of study can be explain unmistakably only with the help of detailed definitions and distinctions, and the feasibility of dissimilar evaluative perception and models can be respected only in light of essential sympathetic of packet-switched voice systems. So, we begin here with a appearance of rudiments, covering such fundamentals as notions of voice services, dimension and evaluation of quality of voice service, and dissimilarity in completion and performance among packet- and circuit-switched voice services. Though these topics may look recognizable to the well-informed reader, it is significant for everybody to become recognizable with this part of the book. Because the basics laid here are necessary perspectives, rather than recapitulation of conservative material, on these topics, much of what is obtainable in this research (Internet Draft, 2003). 2.4 Preferential Quality of Service The query of voice quality is a dominant, but by no way elite, anxiety that users state when they ask for reassurances that the quality of their voice services will be satisfactory. A complete account of what decide users’ awareness of quality of their telephonic voice services comprise no fewer than six self-governing service individuality besides the voice quality that has been the subject of this section. The main characteristics are explain in this research as being convenience, the aptitude to begin a call when desired Routing speed, the speed with which calls are set up Connection reliability, the consistency of the process of setting up a call Routing reliability, the dependability of the procedure of routing the connection to the requested purpose Connection continuity, the aptitude to uphold a link with satisfactory excellence until it is no longer wanted Disconnection reliability, the dependability of the system responses to the orders to cease, and so stop billing for, the connection Internet zealots will right away object to the names given these features of perceived QoS, since Internet voice services are connectionless. Though, the images of each that follow ought to obviously define accurately what is involved, even for “connectionless” connections. Accessibility Accessibility is the measurement of perceived QoS that reproduce the dependability of the system as long as the service as obvious to its users. The essential user concerns with accessibility are habitually uttered by questions such as Will I be able to place a call when I want to? If not, how long will it be before I can? The generic gauge of convenience is, then, defined by 1. Approving that an operational service break is amazing that prevents use of the voice service throughout periods that users anticipate to be talented to contact the system 2. Important the accessibility to be AC, the likelihood that a user effort to create a call will come crossways an operational service disruption permanent t time units or longer, where the value AC is unspoken to symbolize the likelihood that no service disruption will be encountered. Routing Speed Specified right of entry to the service, the next step by the user is to identify the call that is to be set up. Routing speed refers to the time it takes for the system to react to that condition. User concerns with routing speed are spoken in questions like How long does it take previous to I know that the call has been extensive to a station that might be respond by the party I have called? Is that time steady and predictable? Person who reads well-known with sources. They may right away note that the first question here is spoken more usually than its complement in that research. The cause is that packet-switched voice systems may support assignment of calls to persons, somewhat than stations, and be routed by flow follow me protocols in which option are tried devoid of the users’ redialing. The essential quantifier for routing speed as apparent by the user is the postdial delay, distinct in the majority general terms as the time failed among the user’s input of the last piece of information wanted to describe what connection is preferred and the receipt of the first suggestion from the system of the nature of the request. In today’s circuit-switched services, for instance, the post dial holdup is the time among the last digit dialed and welcome of the first perceptible answer from the network, which may be Ring back. Representative that the call has been linked to a distant station answer. Representative that the call was connected Slow busy signal. representing the called station was contacted but found to be busy Fast busy signal. representing that the call setup effort was deserted Special information tone and/or record voice statement (Internet Draft, 2003). Indicating that the requested call could not be set up and/or why As will be seen, there is a real question as to what will be the counterpart of these responses in a packet-switched voice service. Protecting VoIP Services using Extreme’s Policy-Based Quality of Service Great Networks’ Policy-Based Quality of Service ability offered a vital and essential component to helpful VoIP on the LAN defense. Policy-Based QoS permit network architects to produce policies that guarantee network accessibility for VoIP traffic all through times of network blocking. For instance, real-time voice traffic can be certain a precise amount of bandwidth to reduce latency and jitter while professionals call signaling is simply assigned a high precedence to make certain quick call set-up times (Request for Comments, 2003). Great Networks’ recital capabilities make sure that the interior switch architecture will never be a point of overcrowding. Though, network congestion, or over-subscription, is strong-minded by traffic prototype. If two wire-speed 100 Mbps traffic streams create from dissimilar entrance ports are ordained for the same 100 Mbps egress port, the way out port will be over-subscribed by a ratio of 2:1. In a non-QoS scenario, each 100 Mbps stream would be gifted to get half of its traffic throughout while the other half will be dropped. With Policy-Based QoS, you can make a policy that favoritism one stream over the other throughout periods of congestion or over-subscription. End-to-End Voice Analysis The purpose is to test the aptitude of the VoIP network to broadcast voice and other linked signals from end to end. The majority significant part of this test is to review speech quality. Figure 6 shows the archetypal test configuration. This configuration allows a number of dissimilar tests, as explain below. Figure 7 (Internet Draft, 2004) 2.6 Optimizing VoIP Performance with an Extreme Networks Infrastructure Extreme Networks’ Layer 3 switches, joint with Policy-Based QoS, are perfect for behind VoIP traffic, and have been intended to accommodate VoIP traffic from the opening. Our high-performance Summit stackable and Black Diamond chassis switches deliver VoIP traffic at wire-speed with low latency and low jitter. At the similar time, our Extreme Ware software suite delivers Policy-Based Quality of Service to defend the release and presentation of mission-critical VoIP traffic, making sure it gets throughout even during periods of network congestion. In further words, Extreme Networks’ switches offer the recital and protection essential for running VoIP traffic on a switched Ethernet network (Telcordia Technologies, 2004). Moreover Extreme Networks delivers optimal performance for VoIP traffic by leveraging influential switching and QoS abilities that are ordinary to all Summit and BlackDiamond switches: • Wire-speed Performance No doubt, Wire-speed switching at Layer 2 and Layer 3 make sure that the packet forwarding speed, whether it is for a data packet containing part of an e-mail or a VoIP packet enclose part of a telephone discussion, will never be an obstruction to deploying VoIP on the LAN. • Non-Blocking Architecture Each great Networks switch has a non-blocking switch fabric, which means that the interior switch backplane will never be a source of overcrowding or packet loss for VoIP traffic. • Low Latency A side-effect of wire-speed switching is low onward latency. Extreme Networks switches bring in 8-12 microseconds (msec) of latency, or stoppage, when forwarding a 64-byte Ethernet packet at Layer 2 or Layer 3. This is an order of enormity less than the 150-300 milliseconds (ms) of holdup that VoIP packets can stand. As a result, the packet-forwarding delay initiate by Extreme Networks switches is over 1,000 times less than the utmost that VoIP traffic will tolerate (G. Egeland, 2000). • Low Jitter Jitter is a gauge of the difference in latency over time. In a perfect situation with no jitter, each packet arrives at its purpose with the exact same amount of latency over an endless period of time. The realism with all networks is that several amount of jitter will survive. The confrontation is to reduce that jitter. Extreme Networks switches demonstrate an average jitter of 10 microseconds (msec). Jitter of less than 1 millisecond (ms) is outstanding for VoIP traffic. So, the great jitter rate is a thousand times closer to 0 than what is necessary for a victorious VoIP session (R. Straka, 2002). 2.6 Conclusion If we analyze then we come to know that VoIP is an IP application that has severe performance necessities. The performance of the IP network has a direct collision on voice quality. This document identifies the broadcast injury factors that should be deliberate. These comprise frame loss rate, delay, and jitter. In exacting, QoS is an significant component of the IP network. When there is resource argument, such as network congestion, it is significant for the network to give better service to real-time traffic such as VoIP at the expenditure of data traffic. This research desertion also examines different quality measures counting MOS, PSQM, PAMS, PESQ, and the E-Model. All of these measures are helpful and can be used in mixture. When testing VoIP, there are dissimilar tests that ought to be performed. These comprise IP network study, end-to-end voice testing, DTMF, fax, and modem testing, destruction simulation, and signaling constant worry testing. 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