Search

This letter proposes an algorithm for the dynamic tuning of the maximum size of aggregated frames in 802.11 WLANs. Traffic flows with opposed requirements may coexist in these networks: traditional services as web browsing or file download that need high throughput, and services with realtime requirements that need low latency. The proposed algorithm allows the network manager to find an optimal balance (i.e. the "sweet spot" between throughput and latency: a "delay budget" can be assigned to real-time flows, with the objective of keeping the latency as close as possible to that budget, while penalizing the throughput of traditional services as little as possible.

This work presents an analysis of the effect of the access router buffer size on packet loss rate and how it can affect the QoS of multimedia services when bursty traffic is present. VoIP traffic, real traces of viedoconferencing and videosurvellance are used in two different scenarios with medium link utilization. The study shows that the bursty nature of some applications may impair the MOS of voice calls especially when a certain number of bursts overlap. When link utilization is above 70% good values of VoIP QoS cannot be obtained.

All the routers include a buffer in order to enqueue packets waiting to be transmitted. The behaviour of the routers' buffer is of primary importance when studying network traffic, since it may modify some characteristics, as delay or jitter, and may also drop packets. As a consequence, the characterization of this buffer is interesting, especially when real-time flows are being transmitted: if the buffer characteristics are known, then different techniques can be used so as to adapt the traffic: multiplexing a number of small packets into a big one, fragmentation, etc. This work presents a preliminary study of how to determine the technical and functional characteristics of the buffer of a certain device (as e.g. behaviour, size, limits, input and output rate), or even in a remote Internet network node. Two different methodologies are considered, and tested on two real scenarios which have been implemented; real measurements permit the estimation of the buffer size, and the input and output rates, when there is physical or remote access to the "System Under Test". In case of having physical access, the maximum number of packets in the queue can be determined by counting. In contrast, if the node is remote, its buffer size has to be estimated. We have obtained accurate results in wired and wireless networks.

In this letter the development and testing of an open enterprise Wi-Fi solution based on virtual APs, managed by a central WLAN controller is presented. It allows seamless handovers between APs in different channels, maintaining the QoS of real-time services. The potential scalability issues associated to the beacon generation and channel assignment have been addressed. A battery of tests has been run in a real environment, and the results are reported in terms of packet loss and delay.

In corporate and commercial environments, the deployment of a set of coordinated Wi-Fi APs is becoming a common solution to provide Internet coverage to moving users. In these scenarios, real-time services as online games can also be present. This paper presents a set of experiments developed in a test scenario where an end device moves between different APs while generating game traffic. A WLAN solution based on virtual APs is used, in order to make the handoffs transparent for Layer 3. The results show that it is possible to maintain an acceptable level of subjective quality during the handoff. At the same time, it is set clear that the fact of having a gamer in an AP could be taken into account by radio resource management algorithms, in order to provide a better quality.

We study TCP flows coexistence between Massive Multiplayer Online Role Playing Games (MMORPGs) and other TCP applications, by taking World of Warcraft (WoW) and a file transfer application based on File Transfer Protocol (FTP) as an example. Our focus is on the effects of the sender buffer size and FTP cross-traffic on the queuing delay experienced by the (MMORPG) game traffic. A network scenario corresponding to a real life situation in an ADSL access network has been simulated by using NS2. Three TCP variants, namely TCP SACK, TCP New Reno, and TCP Vegas, have been considered for cross-traffic. The results show that TCP Vegas is able to maintain a constant rate while competing with the game traffic, since it prevents packet loss and high queuing delays by not increasing the sender window size. TCP SACK and TCP New Reno, on the other hand, tend to continuously increase the sender window size, thus potentially allowing higher packet loss and causing undesired delays for the game traffic. In terms of buffer size, we have established that smaller buffers are better for MMORPG applications, while larger buffers contribute to a higher overall delay.

This paper studies the use of a traffic optimization technique named TCM (Tunneling, Compressing and Multiplexing) to reduce the bandwidth of MMORPGs (Massively Multiplayer Online Role-Playing Games), which employ TCP to provide a soft real-time service. In order to optimize the traffic and to improve bandwidth efficiency, TCM can be applied when the packets of a number of players share the same link, which occurs in some scenarios, as e.g. the traffic between proxies and servers of game-supporting infrastructures. First, TCP/IP headers are compressed using standard algorithms that avoid sending repeated fields; next, a number of packets are blended into a bigger one and finally, they are sent using a tunnel. The expected compressed header size has been obtained using traffic traces of a real game. Next, simulations using a traffic model of a popular MMORPG have been performed in order to estimate the expected bandwidth savings and the reduction in packets per second. The obtained bandwidth saving is about 60 percent. Packets per second are also significantly reduced. In addition, the added delays are shown to be small enough so as not to impair layers' experienced quality.

This work presents a study of the behaviour of the router buffer when managing the traffic of P2P-TV applications, where a number of peers exchange video content. First, a summary of the characteristics of SOPCast is presented. Then, the results obtained in simulation tests using different buffer policies are presented. Real traces of the application, obtained from a research project, have been used for the tests, sharing the Internet access with different amounts of background traffic. The results show that a similar buffer behaviour for all the access technologies. In addition, the big amount of small packets generated may impair the video traffic, thus avoiding the retransmission of the contents by the application., Comment: in Spanish

This paper compares two policies which can be used for multiplexing the traffic of a number of players of a First Person Shooter game. A network scenario in which a number of players share an access network has been simulated, in order to compare the policies in terms of a subjective quality estimator. The first policy, namely timeout, achieves higher bandwidth savings, while the second one, period, introduces less delay and jitter. The results show that the difference in terms of QoE is only significant when the number of players is small. Thus, in order to make the correct decision, the concrete network scenario and the characteristics of the router would have to be considered in each case, taking into account the estimation of the subjective quality that can be expected.

This paper presents a graphical interface for the management of a wireless local area network that integrates a set of coordinated Wi-Fi access points. The interface interacts with a network application that is responsible for load balancing and mobility management. The graphical application is able to obtain and display the information stored in a system that includes a number of access points, and displays it in a friendly way for the network manager. Finally, the application allows remote management of the system, adjusting its parameters or making transfers between access points., Comment: in Spanish