Jose Saldana

Evaluation of multiplexing and buffer policies influence on VoIP conversation quality

This work presents a study of RTP multiplexing schemes, which are compared with the normal use of RTP, in terms of ITU R-factor quality estimator. The bandwidth saving of the different schemes is studied, and some tests with VoIP traffic are carried out in order to compare R-factor using three different router buffer policies. Network delays are added using an adequate statistical distribution. The tests show the bandwidth savings of multiplexing, and also the importance of the packet size with time-limited buffer policies. The customer experience improvement which can be achieved is measured in terms of R-factor, showing that the use of multiplexing can be interesting in some scenarios, like an enterprise with different offices connected via Internet.

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.

Evaluating the influence of multiplexing schemes and buffer implementation on perceived VoIP conversation quality

This work presents a study of RTP multiplexing schemes, which are compared with the normal use of RTP, in terms of experienced quality. Bandwidth saving, latency and packet loss for different options are studied, and some tests of Voice over IP (VoIP) traffic are carried out in order to compare the quality obtained using different implementations of the router buffer. Voice quality is calculated using ITU R-factor, which is a widely accepted quality estimator. The tests show the bandwidth savings of multiplexing, and also the importance of packet size for certain buffers, as latency and packet loss may be affected. The customers experience improvement is measured, showing that the use of multiplexing can be interesting in some scenarios, like an enterprise with different offices connected via the Internet. The system is also tested using different numbers of samples per packet, and the distribution of the flows into different tunnels is found to be an important factor in order to achieve an optimal perceived quality for each kind of buffer. Grouping all the flows into a single tunnel will not always be the best solution, as the increase of the number of flows does not improve bandwidth efficiency indefinitely. If the buffer penalizes big packets, it will be better to group the flows into a number of tunnels. The router processing capacity has to be taken into account too, as the limit of packets per second it can manage must not be exceeded. The obtained results show that multiplexing is a good way to improve customers experience of VoIP in scenarios where many RTP flows share the same path.

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 players 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 servers 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.

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.

QoS Measurement-Based CAC for an IP Telephony System

This work presents a Call Admission Control (CAC) system for a SIP-based IP Telephony platform. Configured for a multi-branch enterprise environment, the system enables international calls to be established in two steps: one step using Voice over IP (VoIP) through the Internet between the local office and a VoIP-PSTN gateway placed at destination country, and a second step by means of PSTN, from the gateway to the end-user, accounted with local tariffs. CAC decisions are based on Quality of Service (QoS) measurements, call tariffs and also on the number of available lines in the gateway. The CAC has been implemented within a test platform based on virtualization. Measurements to evaluate and validate CAC’s impairment on call establishment delays have been obtained.

First person shooters on the road: Leveraging on APs and VANETs for a quality gaming experience

This work studies the feasibility of using on the road 802.11 Access Points (APs) for the support of First Person Shooter (FPS) games, which are among the online services that exhibit the tightest real-time requirements nowadays. After demonstrating that an acceptable quality of service can be guaranteed, we propose and assess the extension of the AP coverage area by means of vehicular ad-hoc (VANET) communications. The time for which a game can be played in a satisfactory manner can be, in fact, increased exploiting neighboring vehicles, thus improving the general gaming experience. To this aim, we adopted a subjective quality estimator, which reveals whether a game could, or could not, be properly run in a given scenario. Our simulation results show that the AP coverage area can be significantly increased when using other cars as packet forwarders, while maintaining the subjective quality at acceptable levels. All this is performed assessing the feasibility of FPSs on realistic vehicular traffic traces, obtained recording over 100 traversals of an urban road under varying traffic conditions.

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.

Hybrid testbed for network scenarios

In this paper we explain the deployment of a testbed that emulates a network by means of virtualization, allowing us to implement a set of virtual nodes in one physical machine or a small LAN. Nodes participating in the communication are virtual machines adequately connected. The system is a hybrid testbed, as it includes simulation and emulation. But simulations are done in a previous offline stage, in order to avoid computer load caused by the need of real-time simulations, which could limit the maximum number of hosts to be included into the scenario. Finally, two uses of the testbed are presented.

Optimization of low-efficiency traffic in OpenFlow Software Defined Networks

This paper proposes a method for optimizing bandwidth usage in Software Defined Networks (SDNs) based on OpenFlow. Flows of small packets presenting a high overhead, as the ones generated by emerging services, can be identified by the SDN controller, in order to remove header fields that are common to any packet in the flow, only during their way through the SDN. At the same time, several packets can be multiplexed together in the same frame, thus reducing the number of sent frames. Four kinds of small-packet traffic flows are considered (VoIP, UDP and TCP-based online games, and ACKs from TCP flows). Both IPv4 and IPv6 are tested, and significant bandwidth savings (up to 68 % for IPv4 and 78 % for IPv6) can be obtained for the considered kinds of traffic.