Luis Casadesus

First person shooters: can a smarter network save bandwidth without annoying the players?

Real-time services are very challenging for current network infrastructures. One of these services is online gaming, which has acquired more importance in the last years. The quality experienced by users could be improved by the use of a smarter network, which includes some proxies near the players in order to transfer intelligence from the game server to network borders. Proxies have been largely used for other services, such as web browsing and VoIP. First Person Shooters are a popular genre of online games that generate high rates of small packets. This issue becomes important in some scenarios where many flows of a game share the same path, as would happen between a proxy and the central server of a game. These flows could be multiplexed to improve efficiency by reducing packet overhead, thus allowing a bigger number of players to share the same link. A method named Tunneling, Compressing, and Multiplexing (TCM) has been proposed to multiplex these flows. In this article, this method has been tested using the traffic of eight popular First Person Shooters. The method has shown its ability to achieve bandwidth savings of about 30 percent for IPv4 and above 50 percent for IPv6 for all the games. The added delay and jitter are small enough to not annoy players.

Influence of online games traffic multiplexing and router buffer on subjective quality

This work presents a simulation study of the influence of a multiplexing method on the parameters that define the subjective quality for online games, mainly delay, jitter and packet loss. The results for an available subjective quality estimator from the literature are also shown. Two buffer implementations, each one with two buffer sizes, are tested in order to study the mutual influences of the router buffer and multiplexing on subjective quality. The results show that small buffers are more adequate to maintain delay and jitter in acceptable levels, but they increase packet loss. Multiplexing shows a clear advantage when using a buffer which size is measured in packets. A buffer with a limit in packets per second is also tested.

Online Games Traffic Multiplexing: Analysisand Effect in Access Networks

In this paper, a rerouting-controlled ISL (Inter-Satellite link) handover protocol for LEO satellite networks (RCIHP) is proposed. Through topological dynamics and periodic characterization of LEO satellite constellation, the protocol firstly derives the ISL related information such as the moments of ISL handovers and the intervals during which ISLs are closed and cannot be used to forward packet. The information, combined with satellite link load status, is then been utilized during packet forwarding process. The protocol makes a forwarding decision on a per packet basis and only routes packets to living and non-congested satellite links. Thus RCIHP avoids periodic rerouting that occurs in traditional routing protocols and makes it totally unnecessary. Simulation studies show that RCIHP has a good performance in terms of packet dropped possibility and end-to-end delay.

Online FPS games: effect of router buffer and multiplexing techniques on subjective quality estimators

First Person Shooters are a genre of online games in which users demand a high interactivity, because the actions and the movements are very fast. They usually generate high rates of small packets which have to be delivered to the server within a deadline. When the traffic of a number of players shares the same link, these flows can be aggregated in order to save bandwidth. Certain multiplexing techniques are able to merge a number of packets, in a similar way to voice trunking, creating a bundle which is transmitted using a tunnel. In addition, the headers of the original packets can be compressed by means of standard algorithms. The characteristics of the buffers of the routers which deliver these bundled packets may have a strong influence on the network impairments (mainly delay, jitter and packet loss) which determine the quality of the game. A subjective quality estimator has been used in order to study the mutual influence of the buffer and multiplexing techniques. Taking into account that there exist buffers which size is measured in terms of bytes, and others measured in packets, both kinds of buffers have been tested, using different sizes. Traces from real game parties have been merged in order to obtain the traffic of 20 simultaneous players sharing the same Internet access. The delay and jitter produced by the buffer of the access router have been obtained using simulations. In general, the quality is expected to be reduced as the background traffic grows, but the results show an anomalous region in which the quality rises with the background traffic amount. Small buffers present better subjective quality results than bigger ones. When the total traffic amount gets above the available bandwidth, the buffers measured in bytes add to the packets a fixed delay, which grows with buffer size. They present a jitter peak when the offered traffic is roughly the link capacity. On the other hand, buffers which size is measured in packets add a smaller delay, but they increase packet loss for gaming traffic. The obtained results illustrate the need of knowing the characteristics of the buffer in order to make the correct decision about traffic multiplexing. As a conclusion, it would be interesting for game developers to identify the behaviour of the router buffer so as to adapt the traffic to it.

The utility of characterizing packet loss as a function of packet size in commercial routers

This work presents a test called “sizogram” which characterizes the packet loss vs. packet size behavior of a router. This graph can help us to take some decisions about traffic, e.g. multiplexing or not, the number of samples or frames to be included in a packet, etc. In addition, it can be useful to have a better idea of the internal architecture of the router. Simulations have been carried out using different buffer implementations and sizes, and the results show the usefulness of this test.

Comparative of Multiplexing Policies for Online Gaming in Terms of QoS Parameters

This letter compares different policies for multiplexing the traffic of online games. In order to achieve bandwidth savings and to alleviate the high packet rate, headers are compressed and a number of native packets are included into a bigger one, using PPPMux and an L2TP tunnel. Small and controlled delays and jitter are added due to retention at the queue of the multiplexer. The policies are compared using real traffic traces of a popular game, and the results show that the savings are significant, while the impairments are not severe.

Video Conferences through the Internet: How to Survive in a Hostile Environment

This paper analyzes and compares two different video conference solutions, widely used in corporate and home environments, with a special focus on the mechanisms used for adapting the traffic to the network status. The results show how these mechanisms are able to provide a good quality in the hostile environment of the public Internet, a best effort network without delay or delivery guarantees. Both solutions are evaluated in a laboratory, where different network impairments (bandwidth limit, delay, and packet loss) are set, in both the uplink and the downlink, and the reaction of the applications is measured. The tests show how these solutions modify their packet size and interpacket time, in order to increase or reduce the sent data. One of the solutions also uses a scalable video codec, able to adapt the traffic to the network status and to the end devices.

IPTV quality assessment system

Due to the Increasing deployment of real-time multimedia services like IPTV and videoconferencing, the Internet has new Challenges. These new real-time Applications require a reliable performance of the network so as to provide a good Quality of Service (QoS) so it is important for the services providers to estimate the quality offered; and regardless of the transport network to know the quality perceived by the user. For this it is important to have tools to evaluate the quality of service provided. This paper presents a system for IPTV quality assessment. This will allow us to study the users perceived quality for different codecs, bit rates, frame rates and video resolutions, and the impact of the network packet loss rate, in order to determine the objective and subjective quality. We propose an application simulating packet loss as a function of network parameters, which can be used to obtain the received video with different network impairments, without the need for transmitting it. It has two main advantages: first, it avoids the need of transmitting the video a number of times; second, it allows test repeatability.

The effect of router buffer size on subjective gaming quality estimators based on delay and jitter

This work presents a study of the effect of router buffer size on the subjective quality experienced by players of online games, which is a service with very tight real-time requirements. Quality estimators for other real-time services like e.g. VoIP were developed years ago, and they are commonly used when planning a new telephony system. They mainly use delay and loss as KPI (Key Performance Indicators). Subjective quality estimators, analogous to VoIP ones, have also been developed for online games, and some of them are based on delay and jitter, but they do not consider packet loss because many games have highly effective loss handling algorithms. This fact has some implications related to the size of router buffers. If it is too big it may add delay and jitter which are not acceptable for gamers. So a study has been conducted, showing that tiny buffers (some tens of kB) are more adequate in order to maintain game quality in acceptable levels.

Optimization of P2P-TV traffic by means of header compression and multiplexing

This paper studies the optimization of the traffic of a P2P-TV application (SOPCast). First, a traffic characterization is deployed, and it is observed that the service generates a high rate of small UDP packet bursts between peers. Then, an optimization method based on header compression and multiplexing is used for sending together the packets with the same destination. Two multiplexing policies are defined and tested. The first one is based on a fixed multiplexing period, and the other one defines an inter-packet time threshold, with the aim of multiplexing together a whole traffic burst. Simulations using real traffic traces of SOPCast are performed in order to estimate the expected savings for both policies. The results show that the efficiency is improved, achieving uplink bandwidth savings between 26% and 33% for the period-based policy, and roughly 35% for the policy based on a threshold. The amount of packets per second is also reduced by a factor of 10 in both cases. As a counterpart, the addition of a small retention delay is necessary, but the tests show that it does not impair users experience.