Unidirectional and bidirectional optimistic modes IP header compression for real-time video streaming

Communication over Internet Protocol (IP) networks, has become crucial component of day everyday activities. They are utilized over the Internet to support a wide range of services. The flexibility of this kind of transmission relies on the IP User Datagram Protocol (UDP), IP/UDP/Real-time Transport Protocol (RTP) and IP/Transmission Control Protocol (TCP). Unfortunately, the weight of encapsulated protocol headers affects the transmission efficiency. This research aims at improving a technique that reduce the packets header size by compression. Performance analysis of the enhanced efficient techniques in both unidirectional and bidirectional optimistic modes applied to real-time video streaming traffic for UDP/IP and HTTP/TCP flows over free error channel has been conducted. The finding shows that the header compression ratio in each case is good and better than the previous studies. The technique achieved a reduction up to 90% for RTP/UDP/IP, 89% for UDP /IP and 77.5 to 86.5 % for TCP/IP profile. This research contribution is restricted to compression gain and saving for 0x0000, 0x0001, 0x0002 and 0x0006 profiles in the unidirectional and bidirectional optimistic mode

Energy efficient mobile video streaming using mobility

Undeniably the support of data services over the wireless Internet is becoming increasingly challenging with the plethora of different characteristic requirements of each service type. Evidently, about half of the data traffic shifted across the Internet to date consists of multimedia content such as video clips or music files that necessitate stringent real-time constraints in playback and for which increasing volumes of data should be shifted with the introduction of higher quality content. This work recasts the problem of multimedia content delivery in the mobile Internet. We propose an optimization framework with the major tenet being that real-time playback constraints can be satisfied while at the same time enabling controlled delay tolerance in packet transmission by capitalizing on pre-fetching and data buffering. More specifically two strategies are proposed amenable for real time implementation that utilize the inherent delay tolerance of popular applications based on different flavors of HTTP streaming. The proposed mechanisms have the potential of achieving many-fold energy efficiency gains at no cost on the perceived user experience

The VPQ scheduler in access point for VoIP over WLAN

The Voice over Internet Protocol (VoIP) application has observed the fastest growth in the world of telecommunication.VoIP is seen as a short-term and long-trem transmission for voice and audio traffic. Meanwhile, VoIP is moving on Wireless Local Area Networks (WLANs) based on IEEE 802.11 standards.Currently, there are many packet scheduling algorithms for real-time transmission over network.Unfortunately, the current scheduling will not be able to handle the VoIP packets with the proper manner and they have some drawbacks over real-time applications.The objective of this research is to propose a new Voice Priority Queue (VPQ) packet scheduling and algorithm to ensure more throughput, fairness and efficient packet scheduling for VoIP performance of queues and traffics.A new scheduler flexible which is capable of satisfying the VoIP traffic flows.Experimental topologies on NS-2 network simulator were analyzed for voice traffic. Preliminary results show that this can achieve maximum and more accurate VoIP quality throughput and fairness index in access point for VoIP over WLANs.We verified and validated VPQ an extensive experimental simulation study under various traffic flows over WLANs

Audio Streaming System Using Real-Time Transport Protocol Based on Java Media Framework

Audio streaming is an important component of multimedia networking applications.Today’s Internet, however, offers only poor support for such streams due to the lack of the bandwidth and network traffic problems. The work presented in this thesis discusses the problems of real-time audio streaming and investigates solutions for improving the audio data transmitting over the network.To achieve audio media data transmitting over the network in an efficient manner (realtime), the following issues: Initial delay of playing time (downloading time); current streaming protocols which can not cope well with network congestion; compression algorithms efficiency; network bandwidth utilization (network infrastructure); and security concerns of content owners, need to be considered.In this thesis, the implementation method of a real-time audio streaming service system is discussed. The performance of the system implementation both in terms of resulting packet loss, initial delay and delay jitter is presented. This thesis describes audio streaming transmission protocols that are used to implement the system, the system architecture and how the system investigates and addresses the previous issues. A design proposal was outlined to provide an adaptive client/server approach to stream audio contents using Real-Time Transport Protocol (RTP) involving architecture based on the Java Media Framework (JMF) Application Programmable Interfaces (API).RTP protocol is the Internet-standard protocol for the transport of real-time data, including audio and video and can be implemented by using Java Media Framework (JMF). Java Media Framework library and the RTP protocol for audio transmission were used as development tools.The developed system designed in this thesis together with experimental results proved that the system could be implemented successfully. A prototype of the developed system has been implemented and experiments over the Laboratory Local Area Network (LAN)and UPM campus LAN to investigate the issues mentioned before

Rate-Based End-to-End Congestion Control of Multimedia Traffic in Packet Switched Networks

This paper proposes an explicit rate-based end-to-end congestion control mechanism to alleviate congestion of multimedia traffic in packet switched networks such as the Internet. The congestion is controlled by adjusting the transmission rates of the sources in response to the feedback information from destination such as the buffer occupancy, packet arrival rate and service rate at the outgoing link, so that a desired quality of service (QoS) can be met. The QoS is defined in terms of packet loss ratio, transmission delay, power, and network utilization. Comparison studies demonstrate the effectiveness of the proposed scheme over New-Reno TCP (a variant of AIMD: additive increase multiplicative decrease) technique during simulated congestion. Since it is end-to-end, no router support is necessary, the proposed methodology can be readily applied to today\u27s Internet, as well as for real-time video and voice data transfer in unicast networks

Self-adaptive clock synchronisation based on clock precision difference

This paper presents an innovative strategy to synchronize all virtual clocks in asynchronous Internet environments. Our model is based on the architecture of one reference clock and many slave clocks communicating with each other over the Internet. The paper makes three major contributions to this research area. Firstly, one-way information transmission is applied to reduce traffic overhead on the Internet for the purpose of clock synchronization. Secondly, the slave nodes use local virtual time and the arrival timestamp, from the reference node, to create linear mathematical trend models and to retrieve the clock precision differences between reference clock and slave clocks. Finally, a fault-tolerant and self-adaptive model executed by each slave node based on the above linear trend model is created in order to ensure that the virtual clock is running normally, even when the link between the reference node and this slave node has crashed. We also present detailed simulations of this strategy and mathematical analysis on real Internet environments.<br /

An Investigation into the Effect of Security on Performance in a VoIP Network

Voice over Internet Protocol (VoIP) is a communications technology that transmits voice over packet switched networks such as the Internet. VoIP has been widely adopted by home and business customers. When adding security to a VoIP system, the quality of service and performance of the system are at risk. This study has two main objectives, firstly it illustrates suitable methods to secure the signalling and voice traffic within a VoIP system, secondly it evaluates the performance of a VoIP system after implementing different security methods. This study is carried out on a pilot system using an asterisk based SIP (Session initiation Protocol) server (Asterisk, 2009). Since VoIP is intended for use over the Internet, VPNs (Virtual Private Networks) have been used in a tunnel configuration to provide the service. Additionally the performance of networks level IPSec (Internet Protocol Security) and application level ZRTP (Zimmerman Real Time Transport Protocol) security have been compared with no security. Registration, call setup and voice transmission packets have been captured and analysed. The results have then been extrapolated to the Internet

VOIP Model for ICT Rural Communities Telecentre in Sintok

Transmission of Voice over Internet Protocol (VoIP) on packet switching networks is one of the rapidly emerging real-time applications. VoIP is a formation of audio and voice communication. It receive voice signal activities then encoded in digital form and divided into small parts of information as like voice data network packets. These data network packets are decoded and transmitted voice in signals then sender and receiver having a voice conversion. In a voice conversion, the clients send and receive packets in a bidirectional method. Each client work as a sender and as a receiver depends on the direction of traffic flow over network. The aim of this proposal is to propose a VOIP model for ICT rural community’s telecaster in Sintok