Performance Evaluation of MPEG-4 Video Transmission over IP-Networks: Best-Effort and Quality-of-Service

The demand for video communication over internet has been growing rapidly in recent years and the quality of video has become a challenging issue for video transmission. Different types of video coding standards like MPEG-2 and MPEG-4 have been developed to support application like video transmission. MPEG-2 which requires high bit rate transmission has been successful video standard for DVD and satellite digital broadcasting. On the other hand, MPEG-4 supports low bit rate and is suitable for transmitting video over IP networks. In this paper, MPEG-4 Video standard has been used for evaluating the performance of video transmission over two IP networks:- Best-effort and Quality of Service (QoS). For both of the best-effort and QoS IP networks, peak signal noise ratio (PSNR), throughput, frame and packet statistics have been considered as performance metrics. The calculated values of these performance metrics reflect that video transmission over QoS IP network is better than that of the best-effort network. Keywords: video transmission, mpeg, ip networks, best-effort, quality of service, ns-

Efficient Video Transport over Lossy Networks

Nowadays, packet video is an important application of the Internet. Unfortunately the capacity of the Internet is still very heterogeneous because it connects high bandwidth ATM networks as well as low bandwidth ISDN dial in lines. The MPEG-2 and MPEG-4 video compression standards provide efficient video encoding for high and low bandwidth media streams. In particular they include two paradigms which make those standards suitable for the transmission of video via heterogeneous networks. Both support layered video streams and MPEG-4 additionally allows the independent coding of video objects. In this paper we discuss those two paradigms, give an overview of the MPEG video compression standards and describe transport protocols for Real Time Media transport over lossy networks. Furthermore, we propose a real-time segmentation approach for extracting video objects in teleteaching scenarios

Performance evaluation of MPEG-4 video streaming over UMTS networks using an integrated tool environment

Universal Mobile Telecommunications System (UMTS) is a third-generation mobile communications system that supports wireless wideband multimedia applications. This paper investigates the video quality attained in streaming MPEG-4 video over UMTS networks using an integrated tool environment, which comprises an MPEG-4 encoder/decoder, a network simulator and video quality evaluation tools. The benefit of such an integrated tool environment is that it allows the evaluation of real video sources compressed using an MPEG-4 encoder. Simulation results show that UMTS Radio Link Control (RLC) outperforms the unacknowledged mode. The latter mode provides timely delivery but no error recovery. The acknowledged mode can deliver excellent perceived video quality for RLC block error rates up to 30% utilizing a playback buffer at the streaming client. Based on the analysis of the performance results, a self-adaptive RLC acknowledged mode protocol is proposed

Multimedia delivery in the future internet

The term “Networked Media” implies that all kinds of media including text, image, 3D graphics, audio and video are produced, distributed, shared, managed and consumed on-line through various networks, like the Internet, Fiber, WiFi, WiMAX, GPRS, 3G and so on, in a convergent manner [1]. This white paper is the contribution of the Media Delivery Platform (MDP) cluster and aims to cover the Networked challenges of the Networked Media in the transition to the Future of the Internet. Internet has evolved and changed the way we work and live. End users of the Internet have been confronted with a bewildering range of media, services and applications and of technological innovations concerning media formats, wireless networks, terminal types and capabilities. And there is little evidence that the pace of this innovation is slowing. Today, over one billion of users access the Internet on regular basis, more than 100 million users have downloaded at least one (multi)media file and over 47 millions of them do so regularly, searching in more than 160 Exabytes1 of content. In the near future these numbers are expected to exponentially rise. It is expected that the Internet content will be increased by at least a factor of 6, rising to more than 990 Exabytes before 2012, fuelled mainly by the users themselves. Moreover, it is envisaged that in a near- to mid-term future, the Internet will provide the means to share and distribute (new) multimedia content and services with superior quality and striking flexibility, in a trusted and personalized way, improving citizens’ quality of life, working conditions, edutainment and safety. In this evolving environment, new transport protocols, new multimedia encoding schemes, cross-layer inthe network adaptation, machine-to-machine communication (including RFIDs), rich 3D content as well as community networks and the use of peer-to-peer (P2P) overlays are expected to generate new models of interaction and cooperation, and be able to support enhanced perceived quality-of-experience (PQoE) and innovative applications “on the move”, like virtual collaboration environments, personalised services/ media, virtual sport groups, on-line gaming, edutainment. In this context, the interaction with content combined with interactive/multimedia search capabilities across distributed repositories, opportunistic P2P networks and the dynamic adaptation to the characteristics of diverse mobile terminals are expected to contribute towards such a vision. Based on work that has taken place in a number of EC co-funded projects, in Framework Program 6 (FP6) and Framework Program 7 (FP7), a group of experts and technology visionaries have voluntarily contributed in this white paper aiming to describe the status, the state-of-the art, the challenges and the way ahead in the area of Content Aware media delivery platforms

Providing Dynamic TXOP for QoS Support of Video Transmission in IEEE 802.11e WLANs

The IEEE 802.11e standard introduced by IEEE 802.11 Task Group E (TGe) enhances the Quality of Service (QoS) by means of HCF Controlled Channel Access (HCCA). The scheduler of HCCA allocates Transmission Opportunities (TXOPs) to QoS-enabled Station (QSTA) based on their TS Specifications (TSPECs) negotiated at the traffic setup time so that it is only efficient for Constant Bit Rate (CBR) applications. However, Variable Bit Rate (VBR) traffics are not efficiently supported as they exhibit nondeterministic profile during the time. In this paper, we present a dynamic TXOP assignment Scheduling Algorithm for supporting the video traffics transmission over IEEE 802.11e wireless networks. This algorithm uses a piggybacked information about the size of the subsequent video frames of the uplink traffic to assist the Hybrid Coordinator accurately assign the TXOP according to the fast changes in the VBR profile. The proposed scheduling algorithm has been evaluated using simulation with different variability level video streams. The simulation results show that the proposed algorithm reduces the delay experienced by VBR traffic streams comparable to HCCA scheduler due to the accurate assignment of the TXOP which preserve the channel time for transmission.Comment: arXiv admin note: substantial text overlap with arXiv:1602.0369

Optimal packetisation of MPEG-4 using RTP over mobile networks

The introduction of third-generation wireless networks should result in real-time mobile video communications becoming a reality. Delivery of such video is likely to be facilitated by the realtime transport protocol (RTP). Careful packetisation of the video data is necessary to ensure the optimal trade-off between channel utilisation and error robustness. Theoretical analyses for two basic schemes of MPEG-4 data encapsulation within RTP packets are presented. Simulations over a GPRS (general packet radio service) network are used to validate the analysis of the most efficient scheme. Finally, a motion adaptive system for deriving MPEG-4 video packet sizes is presented. Further simulations demonstrate the benefits of the adaptive system

A QoS monitoring system in a heterogeneous multi-domain DVB-H platform

The MobileTV, IPTV, and DVB standards (DVB-H/T) have been defined to offer mobile users interactive multimedia services with quality of service (QoS) consistency analogous to TV services. However, the market has yet to provide effective and economical solutions for the real-time delivery of such services to the corresponding transmitters over multi-domain IP networks. The monitoring system proposed in this paper enables the QoS in the IP networks involved in the delivery of real-time multimedia content to the transmitters to be ascertained. The system utilizes the QoS parameters defined in MPEG-2 Transport Streams to detect problems occurring in the heterogeneous multi-domain IP networks. The ability to detect problems having an adverse effect on QoS allows appropriate control actions to be determined to recover the QoS across the composite IP network. The design and implementation of the proposed QoS-Monitoring system (QoS-MS) is presented, followed by analysis of experimental results that demonstrate the feasibility of the system

Building self-optimized communication systems based on applicative cross-layer information

This article proposes the Implicit Packet Meta Header(IPMH) as a standard method to compute and represent common QoS properties of the Application Data Units (ADU) of multimedia streams using legacy and proprietary streams’ headers (e.g. Real-time Transport Protocol headers). The use of IPMH by mechanisms located at different layers of the communication architecture will allow implementing fine per-packet selfoptimization of communication services regarding the actual application requirements. A case study showing how IPMH is used by error control mechanisms in the context of wireless networks is presented in order to demonstrate the feasibility and advantages of this approach

3D video coding and transmission

The capture, transmission, and display of 3D content has gained a lot of attention in the last few years. 3D multimedia content is no longer con fined to cinema theatres but is being transmitted using stereoscopic video over satellite, shared on Blu-RayTMdisks, or sent over Internet technologies. Stereoscopic displays are needed at the receiving end and the viewer needs to wear special glasses to present the two versions of the video to the human vision system that then generates the 3D illusion. To be more e ffective and improve the immersive experience, more views are acquired from a larger number of cameras and presented on di fferent displays, such as autostereoscopic and light field displays. These multiple views, combined with depth data, also allow enhanced user experiences and new forms of interaction with the 3D content from virtual viewpoints. This type of audiovisual information is represented by a huge amount of data that needs to be compressed and transmitted over bandwidth-limited channels. Part of the COST Action IC1105 \3D Content Creation, Coding and Transmission over Future Media Networks" (3DConTourNet) focuses on this research challenge.peer-reviewe