Delivering Quality End User Experience against Diminished Network Resources - Research Paper Example

Delivering Quality end User Experience against Diminished Network Resources Organization affiliation Country Email The popularity of video conferencing, text messaging, instant messaging, and audio conferencing is on the increase. With the increase in demand and use of this technology, service providers are finding it increasingly challenging to deliver quality service to end users. Quality service of Voice data requires that data be sent out using a high bit rate. However, this is a challenge because it results in wasting of network resources and congestion within the network. Consequently, the main objective, to deliver a quality end user experience is compromised. In this paper three studies that look at Skype as a model have been discussed. Skype has been the most successful service provider and studying it may provide insights into possible solutions. A model has been proposed for further research that is not only user centered, but can also benefit from some of Skype’s attributes such as ability to interact with NATs and firewalls seamlessly. I. Introduction Video conferencing is rapidly gaining momentum and many end users are adopting this service. For instance, skype is the fastest growing audio conferencing service provider with its annual revenue exceeding $ 136 million (Huang, Chen, & Huang, 2009). With the rapid growth and expansion witnessed in this industry, information technology experts are concerned with the impact of video calls, and the quality of experience under a variety of network conditions that are available to them. Skype offers its end users voice calls services, text messaging services, audio conferencing services, and instant messaging (Baset & Schulzrine, 2006). Delivering delay sensitive voice data from one user to user effectively has been a multimedia networking problem. Voice data has a high data rate. Failures in networks for example, impairments in network capacity significantly affects the voice traffic data rates. The rapid increase in multimedia traffic seen over the internet presents a major hurdle in delivery of quality service to end users. In order to deliver quality service, voice data has to be sent using a high bit rate. Using a higher bit rate could however result in the waste of network resources. Using a higher bit rate mat also result in congestion within the network. the aim is to guarantee quality of experience (QoE) the end users receive is high. However, using high bit rates would result in network congestion which would in turn compromise end user QoE. In order to address this problem, studies have been conducted on skype to establish how it manages to deliver to its end users a high QoE. Skype offers better voice quality services than other VoIP clients. In one such study by Baset and Schulzrinne (2006), the investigators studied Skype’s protocol under three different types of network setups. The investigators specifically looked at different Skype functions such as NAT, Login, call establishment, codecs, conferencing, media transfer, and login. The investigators carefully studied Skype’s traffic by intercepting Skype’s system calls and shared library. In another study by Zhang et al., (2013), the investigators studied Skype’s sending rate, frame rate, FEC redundancy, and video rate in order to develop an in-depth understanding of how Skype is able to adjust different network conditions. The investigators varied the available bandwidth, propagation delay, and packet loss rate in to understand the behavior of Skype in different networks. In another study by Huang, Chen, and Huang (2009), the investigators studied Skype in order to understand how it was able to adapt its voice data rate to changing network conditions. They further studied how public domain codecs have been implemented in order to determine whether Skype’s protocol is consumer-centric. Using a set of systematic experiments, the investigators were able to develop a model that has a voice data rate that is user-centric. II. Solutions In one of the studies by Huang, Chen, and Huang (2009), a model that was user-centric was developed. This model can easily be integrated into Skype and other general VoIP services. This model, unlike Skype’s model, is user-centric. To develop the model, the investigators used a number of systematic experiments where differing levels of burstiness and packet loss were used to quantify the level of consumer experience. The investigators adjusted burstiness and packet loss to the point where the consumer experience was optimal. Baset and Schulzrinne (2006), established that Skype uses the STUN protocol to identify firewalls or NATs and this makes it possible for Skype to work seamlessly behind firewalls and NATs. Specifically, Skype randomly selects listener and sender ports, it uses TCP as its voice streaming protocol to traverse NATs and firewalls without necessarily having any firewall or NAT transversal server. The investigators also noted that Skype secures the best network and CPU resources during a call. For example, Skype changes its application priority to high to make sure that it secures the best CPU resources. Skype further uses SNs (super nodes) to evade blocking. Further still, Skype denies a user the ability of preventing a machine on Skype from preventing the machine from becoming a SN. The study by Zhang et al., (2013) established that Skype’s protocol is able to adapt its rates to propagation delay. There is a special interaction between Skype’s rate and TCP rate. TCP responds well to increases in propagation delay that Skype. Also, when the propagation delay is small, TCP has a higher rate than Skype. Increases in propagation delay result in a decrease in the TCP rate while the Skype rate remains constant. The investigators established that when, Skype shares a wireless link with TCP flows, Skype operates normally with no congestion and the quality of the video call is good when there is abundant bandwidth. When the scale up factor increases to over 20, congestion increases together with packet losses and queuing delays. The quality of the of Skype’s video calls deteriorates. Skype has built in rate control schemes and user back offs that help it handle traffic when there is congestion, in the network. III. Comparison of solutions The need to find ways of delivering delay sensitive voice data from one user to user effectively has been a multimedia networking problem. Several studies have been developed that look at Skype as a model and use it to find solutions to the problem. In one such study by Huang, Chen, and Huang (2009), the investigators established that the protocol used by Skype is not user centric. The investigators develop a model that is user centric using experimental approaches. The actual findings of this investigationiffer from those by Zhang et al., (2013) and Baset and Schulzrinne (2006). The goal is to come up with a platform that maximizes the QoE. Huang, Chen, and Huang (2009), were able to develop a model that maximizes the end user. This model they note is better than Skype. However, the other two studies had established that Skype is robust VoIP service provide with good end user quality of experience. Baset and Schulzrinne (2006), note that Skype uses super nodes to escape blocking while at the same time uses super nodes to acquire the best CPU and network resources. While Skype may have gained advantage over other VoIP service providers, its success is only limited to those networks where the bandwidth is not restricted. In those organizations where bandwidth has been restricted, Skype would not be able to convert a user’s computer into a super node. Consequently, Skype’s quality of experience would be severely impaired. Baset and Schulzrinne (2006) revealed a mechanism currently being used by Skype to deliver quality experiences to end users that is not sustainable. Huang, Chen, and Huang (2009) have developed a model that is able to deliver quality experiences to its end users. The model does not rely on super nodes that need enormous bandwidths in order to give quality experiences. The study by Zhang et al., (2013) established that Skype’s protocol is able to adapt its rates to propagation delay. Skype has a robust framework that makes it a very responsive to delays. This is one of Skype’s positive attributes. Despite this finding, the best solution is one that maximizes the end user quality of experience. Even though, Zhang et al., (2013) discusses a number of attributes that make Skype a robust platform, Skype needs to refocus its platform so that the end user experience is the ultimate goal. IV. Future research direction More research ought to be conducted to establish how the model developed by Huang, Chen, and Huang (2009) can be integrated into Skype and other VoIP service providers. The model developed by Huang, Chen, and Huang (2009) can also be modified in its implementation phase to be able to interact seamlessly with firewalls and NATs. There is need to conduct more research  so that thefinal version of the model is not only integrated into VoIP service providers’ platforms, but is able to interact seamlessly with NATs and firewalls. V. Conclusion The need to find ways of delivering delay sensitive voice data from one user to user effectively continues to be a multimedia networking problems. Various solutions have actually been developed by studying Skype’s VoIP platform. The studies yielded differing solutions. However, with the ultimate goal being to deliver a quality end user experience, it is critical that a model be developed that fulfills the main objective. One group of investigators found that Skype is not user centered. This group went further to develop a model that is not only user friendly, but can easily be integrated into VoIP services. The studies have indicated a number of weaknesses in skype such as relying on super nodes that need enormous band widths in order to deliver quality experience to end users, and not being user centered. Better protocols need to be developed that deliver quality experience to end users and are not affected by any changes that may be made to network conditions. References S. Baset, H. Schulzrinne, “An analysis of the Skype peer-to-peer internet telephony protocol.”New York, NY: Columbia University, 2006. T. Huang, K. Chen, P. Huang, “Tuning Skype’s redundancy control algorithm for user satisfaction,” Taiwan, National Taiwan University, 2009. X. Zhang, Y. Xu, H. Hu, Y. Liu, Z.Guo “Modeling and analysis of Skype video calls rate control and video quality,” IEEE Trans. on Multim. Vol.2, No. 6, pp 1446-1457. Read More