José I. Aznar

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.

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.

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.

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.

Influence of the distribution of TCRTP multiplexed flows on VoIP conversation quality

TCRTP is a multiplexing scheme for RTP, defined by IETF, which reduces the overhead of real-time flows. This work presents a preliminary study of how the number of multiplexed flows in TCRTP can affect the Quality of Service (QoS) of VoIP in terms of R-factor. Two different buffer policies are considered. The conclusion is that multiplexing all the calls in a single TCRTP flow is not always the best solution for certain buffer policies. The reason is that higher packet sizes can increase loss probability.