Analyzing the effect of tcp and server population on massively multiplayer games

Many Massively Multiplayer Online Role-Playing Games (MMORPGs) use TCP flows for communication between the server and the game clients. The utilization of TCP, which was not initially designed for (soft) real-time services, has many implications for the competing traffic flows. In this paper we present a series of studies which explore the competition between MMORPG and other traffic flows. For that aim, we first extend a source-based traffic model, based on player’s activities during the day, to also incorporate the impact of the number of players sharing a server (server population) on network traffic. Based on real traffic traces, we statistically model the influence of the variation of the server’s player population on the network traffic, depending on the action categories (i.e., types of in-game player behaviour). Using the developed traffic model we prove that while server population only modifies specific action categories, this effect is significant enough to be observed on the overall traffic. We find that TCP Vegas is a good option for competing flows in order not to throttle the MMORPG flows and that TCP SACK is more respectful with game flows than other TCP variants, namely, Tahoe, Reno, and New Reno. Other tests show that MMORPG flows do not significantly reduce their sending window size when competing against UDP flows. Additionally, we study the effect of RTT unfairness between MMORPG flows, showing that it is less important than in the case of network-limited TCP flows

A network resource availability model for IEEE802.11a/b-based WLAN carrying different service types

The electronic version of this article is the complete one and can be found online at: http://jwcn.eurasipjournals.com/content/2011/1/103. This is an Open Access article distributed under the terms of the Creative Commons Attribution License (http://creativecommons.org/licenses/by/2.0), which permits unrestricted use, distribution, and reproduction in any medium, provided the original work is properly cited.Operators of integrated wireless systems need to have knowledge of the resource availability in their different access networks to perform efficient admission control and maintain good quality of experience to users. Network availability depends on the access technology and the service types. Resource availability in a WLAN is complex to gather when UDP and TCP services co-exist. Previous study on IEEE802.11a/b derived the achievable throughput under the assumption of inelastic and uniformly distributed traffic. Further study investigated TCP connections and derived a model to calculate the effective transmission rate of packets under the assumption of saturated traffic flows. The assumptions are too stringent; therefore, we developed a model for evaluating WLAN resource availability that tries to narrow the gap to more realistic scenarios. It provides an indication of WLAN resource availability for admitting UDP/TCP requests. This article presents the assumptions, the mathematical formulations, and the effectiveness of our model

KISS: Stochastic Packet Inspection Classifier for UDP Traffic

This paper proposes KISS, a novel Internet classifica- tion engine. Motivated by the expected raise of UDP traffic, which stems from the momentum of Peer-to-Peer (P2P) streaming appli- cations, we propose a novel classification framework that leverages on statistical characterization of payload. Statistical signatures are derived by the means of a Chi-Square-like test, which extracts the protocol "format," but ignores the protocol "semantic" and "synchronization" rules. The signatures feed a decision process based either on the geometric distance among samples, or on Sup- port Vector Machines. KISS is very accurate, and its signatures are intrinsically robust to packet sampling, reordering, and flow asym- metry, so that it can be used on almost any network. KISS is tested in different scenarios, considering traditional client-server proto- cols, VoIP, and both traditional and new P2P Internet applications. Results are astonishing. The average True Positive percentage is 99.6%, with the worst case equal to 98.1,% while results are al- most perfect when dealing with new P2P streaming applications

Performance evaluation of an open distributed platform for realistic traffic generation

Network researchers have dedicated a notable part of their efforts to the area of modeling traffic and to the implementation of efficient traffic generators. We feel that there is a strong demand for traffic generators capable to reproduce realistic traffic patterns according to theoretical models and at the same time with high performance. This work presents an open distributed platform for traffic generation that we called distributed internet traffic generator (D-ITG), capable of producing traffic (network, transport and application layer) at packet level and of accurately replicating appropriate stochastic processes for both inter departure time (IDT) and packet size (PS) random variables. We implemented two different versions of our distributed generator. In the first one, a log server is in charge of recording the information transmitted by senders and receivers and these communications are based either on TCP or UDP. In the other one, senders and receivers make use of the MPI library. In this work a complete performance comparison among the centralized version and the two distributed versions of D-ITG is presented

Comparing P2PTV Traffic Classifiers

Peer-to-Peer IP Television (P2PTV) applications represent one of the fastest growing application classes on the Internet, both in terms of their popularity and in terms of the amount of traffic they generate. While network operators require monitoring tools that can effectively analyze the traffic produced by these systems, few techniques have been tested on these mostly closed-source, proprietary applications. In this paper we examine the properties of three traffic classifiers applied to the problem of identifying P2PTV traffic. We report on extensive experiments conducted on traffic traces with reliable ground truth information, highlighting the benefits and shortcomings of each approach. The results show that not only their performance in terms of accuracy can vary significantly, but also that their usability features suggest different effective aspects that can be integrate

Evidences Behind Skype Outage

Skype is one of the most successful VoIP application in the current Internet spectrum. One of the most peculiar characteristics of Skype is that it relies on a P2P infrastructure for the exchange of signaling information amongst active peers. During August 2007, an unexpected outage hit the Skype overlay, yielding to a service blackout that lasted for more than two days: this paper aims at throwing light to this event. Leveraging on the use of an accurate Skype classification engine, we carry on an experimental study of Skype signaling during the outage. In particular, we focus on the signaling traffic before, during and after the outage, in the attempt to quantify interesting properties of the event. While it is very difficult to gather clear insights concerning the root causes of the breakdown itself, the collected measurement allow nevertheless to quantify several interesting aspects of the outage: for instance, measurements show that the outage caused, on average, a 3-fold increase of signaling traffic and a 10-fold increase of number of contacted peers, topping to more than 11 million connections for the most active node in our network - which immediately gives the feeling of the extent of the phenomeno

Insights into the Design of Congestion Control Protocols for Multi-Hop Wireless Mesh Networks

The widespread deployment of multi-hop wireless mesh networks will depend on the performance seen by the user. Unfortunately, the most predominant transport protocol, TCP, performs poorly over such networks, even leading to starvation in some topologies. In this work, we characterize the root causes of starvation in 802.11 scheduled multi-hop wireless networks via simulations. We analyze the performance of three categories of transport protocols. (1) end-to-end protocols that require implicit feedback (TCP SACK), (2) Explicit feedback based protocols (XCP and VCP) and (3) Open-loop protocol (UDP). We ask and answer the following questions in relation to these protocols: (a) Why does starvation occur in different topologies? Is it intrinsic to TCP or, in general, to feedback-based protocols? or does it also occur in the case of open-loop transfers such as CBR over UDP? (a) What is the role of application behavior on transport layer performance in multi-hop wireless mesh networks? (b) Is sharing congestion in the wireless neighborhood essential for avoiding starvation? (c) For explicit feedback based transport protocols, such as XCP and VCP, what performance can be expected when their capacity estimate is inaccurate? Based on the insights derived from the above analysis, we design a rate-based protocol called VRate that uses the two ECN bits for conveying load feedback information. VRate achieves near optimal rates when configured with the correct capacity estimate

Performance of VoIP with DCCP for satellite links

We present experimental results for the performance of selected voice codecs using the Datagram Congestion Control Protocol (DCCP) with TCP-Friendly Rate Control (TFRC) congestion control mechanism over a satellite link. We evaluate the performance of both constant and variable data rate speech codecs (G.729, G.711 and Speex) for a number of simultaneous calls, using the ITU E-model and identify problem areas and potential for improvement. Our experiments are done on a commercial satellite service using a data stream generated by a VoIP application, configured with selected voice codecs and using the DCCP/CCID4 Linux implementation. We analyse the sources of packet losses which are a main contributor to reduced voice quality when using CCID4 and additionally analyse the effect of jitter which is one of the crucial parameters contributing to VoIP quality and has, to the best of our knowledge, not been considered previously in the published DCCP performance results. We propose modifications to the CCID4 algorithm and demonstrate how these improve the VoIP performance, without the need for additional link information other than what is already monitored by CCID4 (which is the case for Quick-Start). We also demonstrate the fairness of the proposed modifications to other flows. We identify the additional benefit of DCCP when used in VoIP admission control mechanisms and draw conclusions about the advantages and disadvantages of the proposed DCCP/ CCID4 congestion control mechanism for use with VoIP applications