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Quality of Service over Wireless NetworkAbstract— As in the innovation in the 4.0 era, wireless networks as a key solution rapidly growth in both flexible and economical to provide broadband and mobile wireless connectivity. A Wireless Network is a transmission network made up of mixture of wireless nodes accomplished in a mesh topology. Quality of Service (QoS) is particularly important consideration in networking, but for Voice over IP (VoIP) ity faces many technique challenge. This paper will present and discuss the challenges and solutions involved in providing QoS for VoIP in wireless data network.Index Terms—Keywords should be taken from the taxonomy Quality-Of-Service, Wireless Network, Voice over IP.——————————   ?   ——————————1 IntroductionA wireless network is a computer network that uses wireless data connections between network nodes. Wireless networking is a method by which homes, telecommunications networks and business installations avoid the costly process of introducing cables into a building, or as a connection between various equipment locations. Wireless networks are becoming more and more ubiquitous in recent years, ranging from mobile analog and digital cellular telephony up to satellite broadcasting. VoIP is a new Internet application and has been receiving an ever increasing attention. Prime advantages of the VoIP are lower cost for the customers and higher profits for the providers, as compared to traditional telephone networks. Quality of Service (QoS) is an important part of the networking world. It prioritize certains activities on our network over others and dictates the quality of the services provided by the Internet Service Provider(ISP). In this report I will be discussing the QoS in wireless network for VoIP.2 Procedure for Paper Submission2.1 Review Stage IIn the early 2000, tech savvy countries like Korea showed rapid growth in Voice Over Internet Protocol(VOIP). Domestic carriers, formally reluctant to invest in VOIP, have now started to invest into VOIP in recognition to the popular current telecommunication industry as well as the competition in today’s market.  Internet telephony is becoming the norm and it offers powerful technology that enable many things beyond straight voice calling. For example, VOIP services supports variety of option such as pc-to-pc, pc-to-smartphone and smartphone-to-smartphone. This enables inbounds and outbounds calls where the calls can suffer from noticeable quality issues. Compared to the regular landline and mobile phone plans who charge us higher rates mostly by making international calls, VOIP has changed the way in which we communicate with people with bypassing of the regular telephone companies thus reducing in cost of 50% – 95%. It has considerable influences on VOIP call demands. Due to the government regulation, some countries don’t support the VOIP services.  Nowadays, many service providers require proper Quality of Service(QoS) transmission of audio and sometimes telecommunication over packet-switching. Packet-switching is used as a primary data communication worldwide. The very first thing you need to guarantee in order to guarantee quality for VoIP is adequate bandwidth. Problems that can impair the quality can include audio compression and decompression algorithm, network-traffic delays, loss, noise and many other types of distortion. In fig-1 it shows us the most parameters that have affected the QoS in the recent years. http://ieeexplore.ieee.org/document/7820167/?part=undefined%7Cfig2#fig2 Delay transmission in the network seems to be the main problem in contrast to fig-1 which can be avoided by minimizing the delay and maximizing the packet delivery ratio to aid in better quality of service. Figure 2. Voice quality as a function of packet loss rateSpeaking of measuring the network, the parameters as bandwidth, delay, jitter and packet loss, were obtained as results of simulations performed. M.Hassan 1 show that delay, jitter and packet loss can significantly degrade the voice quality. It has been shown that an end-to-end delay of over 150 milliseconds (ms) is intolerable to VoIP users, and the delay between packets must be lower than 20 ms for uninterrupted and smooth hearing.The Symbol element is advertised by the access point. This helps a Symbol phone to make an association decision if there are multiple access points serving the area. The current packet rate is the calculation of average means of number of packets transmitted per second for the past 8 seconds.After the normal 802.11 association process, a Symbol phone sends a proprietary Symbol 802.11 phone registration message (WNMP) to the access point to complete the association.The Symbol phone does not associate to an access point if the advertised packet rate is above the threshold of the access point. The Symbol phone uses its Symbol element as optional information. Basic operation does not require an access point to send Symbol elements.http://www.callingtoronto.com/calling-cards-vs-voip-comparison https://www.therealpbx.com/blog/pbx-voip/factors-affect-voice-quality-in-voip-calls http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=&arnumber=6026220 http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=&arnumber=1620285 http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=&arnumber=53422372.2 Review Stage IIAs shown in figure 2, Watson and Sasse 2 described how the quality of voice degrades as the loss rate increase. As long as there is no space left in the the queue, the arriving packet is called lost. The more people use the Internet, routers as affected getting congested more often,  resulting in packet loss. Packet loss can cause severe damage to voice quality for IP telephony. Each IP packet contains 40–80 ms of speech information, matching the duration of critical units of speech called phonemes. When a packet is lost, a phoneme is lost in the continuous speech. (A. Watson and M. A. Sasse, “Multimedia Conferencing via Multicast: Determining the Quality of Service Required by the End User,” Proc. Int’l. Wksp. AudioVisual Svcs. over Packet Networks, Aberdeen, Scotland, Sept. 1997.)Fig 3. Skype annual growth from 2005-2013According to T.Gara, TeleGeography research show that in 2013, Skype carried an estimated 214 billion minutes of international ‘on-net’ calls, which growth up 36% compare to 2012. Surprisingly, Skype’s traffic was about 40% the size of the entire conventional international telecom market. As shown in fig 3, the international calling minutes has constantly increase from 5 billions minutes call to 54 billion minutes call in 9 years from 2005 to 2013. Skype growth is leading in the international calling industry, its growth in 2013 of 54 billion minutes is about 50% higher than the growth of the rest industry in the market.