Robust header compression over IEEE 802 networks

Tese de mestrado. Redes e Serviços de Comunicação. Faculdade de Engenharia. Universidade do Porto, INESC Porto. 200

Performance Analysis of VoIP in Multi-Hop Wireless Network

This paper presents the performance evaluation of Voice over Internet Protocol (VoIP) in Multi-hop Wireless Network (MWN) developed using Multi-radio Access Relay (MAR). The MWN is deployed using 3 MARs in Universiti Teknikal Malaysia campus. The performance of VoIP are investigated using Real Time Protocol (RTP) and Compress Real Time Protocol (CRTP) header techniques. RTP and CRTP are used to transport voice packets using G711.1, G723.1 and G729.2 codec. The performance of VoIP is analyzed based on three important elements which are delay, jitter and packet loss

An innovative approach for enhancing capacity utilization in point-to-point voice over internet protocol calls

Voice over internet protocol (VoIP) calls are increasingly transported over computer-based networking due to several factors, such as low call rates. However, point-to-point (P-P) calls, as a division of VoIP, are encountering a capacity utilization issue. The main reason for that is the giant packet header, especially when compared to the runt P-P calls packet payload. Therefore, this research article introduced a method to solve the liability of the giant packet header of the P-P calls. The introduced method is named voice segment compaction (VSC). The VSC method employs the unneeded P-P calls packet header elements to carry the voice packet payload. This, in turn, reduces the size of the voice payload and improves network capacity utilization. The preliminary results demonstrated the importance of the introduced VSC method, while network capacity improved by up to 38.33%

Performance estimation of wireless networks using traffic generation and monitoring on a mobile device.

Masters of ScienceIn this study, a traffic generator software package namely MTGawn was developed to run packet generation and evaluation on a mobile device. The call generating software system is able to: simulate voice over Internet protocol calls as well as user datagram protocol and transmission control protocol between mobile phones over a wireless network and analyse network data similar to computer-based network monitoring tools such as Iperf and D-ITG but is self-contained on a mobile device. This entailed porting a ‘stripped down’ version of a packet generation and monitoring system with functionality as found in open source tools for a mobile platform. This mobile system is able to generate and monitor traffic over any network interface on a mobile device, and calculate the standard quality of service metrics. The tool was compared to a computer–based tool namely distributed Internet traffic generator (D-ITG) in the same environment and, in most cases, MTGawn reported comparable results to D-ITG. The important motivation for this software was to ease feasibility testing and monitoring in the field by using an affordable and rechargeable technology such as a mobile device. The system was tested in a testbed and can be used in rural areas where a mobile device is more suitable than a PC or laptop. The main challenge was to port and adapt an open source packet generator to an Android platform and to provide a suitable touchscreen interface for the tool. ACM Categories and Subject Descriptors B.8 [PERFORMANCE AND RELIABILITY] B.8.2 [Performance Analysis and Design Aids] C.4 [PERFORMANCE OF SYSTEMS] Measurement techniques, Performance attribute

ETSI reconfigurable radio systems: status and future directions on software defined radio and cognitive radio standards

This article details the current work status of the ETSI Reconfigurable Radio Systems Technical Committee, positions the ETSI work with respect to other standards efforts (IEEE 802, IEEE SCC41) as well as the European Regulatory Framework, and gives an outlook on the future evolution. In particular, software defined radio related study results are presented with a focus on SDR architectures for mobile devices such as mobile phones. For MDs, a novel architecture and inherent interfaces are presented enabling the usage of SDR principles in a mass market context. Cognitive radio principles within ETSI RRS are concentrated on two topics, a cognitive pilot channel proposal and a Functional Architecture for Management and control of reconfigurable radio systems, including dynamic self-organizing planning and management, dynamic spectrum management, joint radio resource management. Finally, study results are indicated that are targeting a SDR/CR security framework.Postprint (published version

Prediction of RoHCv1 and RoHCv2 compressor utilities for VoIP

Modern cellular networks utilising the long–term evolution (LTE) and the coming 5G set of standards face an ever–increasing demand for low–latency mobile data from connected devices. Header compression is employed to minimise the overhead for IP–based cellular network traffic, thereby decreasing the overall bandwidth usage and, subsequently, transmission delays. Here, we employ machine learning approaches for the prediction of Robust Header Compression version 1’s and version 2’s compression utility for VoIP transmissions, which allows the compression to dynamically adapt to varying channel conditions. We evaluate various regression models employing r2 and mean square error scores next to complexity (number of coefficients) based on an RTP specific training data set and separately captured live VoIP audio calls. We find that the proposed weighted Ridge regression model explains about at least 50 % of the observed results and the accuracy score may be as high as 94 % for some of the VoIP transmissions

Improved multi-point communication for data and voice over IEEE 802.11b

There is a growing demand for faster, improved data and voice services in rural areas without modern telecom infrastructure. A wireless network is often the only feasible solution for providing network access in this environment, due to the sparse populations and difficult natural conditions. A system solution that incorporates the Multipoint Communication System (MCS) algorithm created by TRLabs into the available IEEE 802.11b Wireless Local Area Network (WLAN) devices was proposed and studied in this thesis. It combines the advantages of both systems, that is, the MCS’ capability of integrating Voice over Internet Protocol (VoIP) and data services and the IEEE 802.11b standard, currently the most widely used in WLAN products. A system test bed was set up inside Network Simulator-2 (NS-2). The data and VoIP performance was tested. Modifications to the original MCS algorithm to improve system performance were made throughout this thesis. In a constant rate radio channel, data performance (throughput and transmission efficiency) was measured using the original MCS algorithm, which was comparable to the standard Distribution Coordination Function (DCF) operation of IEEE 802.11b when both were simulated at similar conditions. On an 802.11b platform, the Automatic Rate Fallback (ARF) feature was incorporated into the original MCS algorithm. However, when clients with different data rates were present in the same channel, all the clients involved received unacceptably low and equal data throughput, dragged down by the low rate clients. A modified MCS data polling algorithm was proposed with the capability of repeated polling, which eliminated the negative effect of low rate clients in a multi-rate channel. In addition, the original MCS algorithm was modified to be more efficient in the voice polling process. The voice performance and data throughput were tested at various conditions. However, the one-by-one polling still resulted in very low voice transmission efficiency. The time wasted became more severe with increasing relay distance and channel rate (only 8.5% in an 11 Mbps channel at 30 km). A new voice handling process similar to Time Division Multiple Access (TDMA) mode was implemented and simulated. Its voice efficiency can be kept at 25% at any setting of relay distance and channel rate. Data transmission in the same channel can also benefit from using the new voice scheme. The normalized saturation throughput could be improved by 13.5% if there were 40 voice clients involved in an 11 Mbps channel at the relay distance of 15 km, compared to the original MCS algorithm. More improvement in voice efficiency, voice capacity, and data throughput can be achieved at longer relay distance, or with more voice calls set up