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

Fase 6 evaluación de la red NGN y Qos

Este trabajo describe la configuración de servicios de VoIP sobre un servidor Elastix con cliente Linphone sobre una red WAN. Se documenta sobre el servicio de TVIP y servicos QoSThe voice service requires protocols for signaling (in the control plane) and protocols for data transfer (in the userplane). Protocols used for signaling of the QoS-enabled VoIP in NGN are SIP and Diameter, which work on application layer regarding the OSI (Open System for Interconnection) protocol layering model (both protocols are covered in Chapter 7). For the voice data transfer the QoS-enabled VOIP uses RTP/UDP/IP protocol stack. Above the RTP [[2]], which belongs to transport layer together with the UDP, is the VoIP application that provides voice codec functionalities. RTP is designed to provide end-to-end transport functions for provision of real-time services in Internet, such as audio (e.g., VoIP), video (e.g., IPTV - Internet Protocol Television), and multimedia data, over unicast (e.g., for VoIP) or multicast networks (e.g., for IPTV or multimedia streaming services). However, RTP is typically present at end hosts that have an established session, and hence it does provide QoS guarantees and resource reservation for the real-time services (they have to be implemented by the network, such as NGN, that is used to transfer real-time data). RTP is used together with RTP Control Protocol (RTCP) protocol, where RTP is used to carry media streams (e.g., audio, video, multimedia, etc.) while RTCP is used to provide control feedback between two end points of an RTP session regarding the transmission statistics, QoS, and synchronization of multiple streams. So, RTP standard, defines a pair of protocols, namely RTP and RTCP. Bot

Quality of service for video stream over IP networks

Multimedia networking is one of the most exciting developments in today's Internet. Streaming technology; which allows the player to start playing audio/video data immediately instead of waiting for the entire file to be downloaded, presents an attractive vehicle for the distribution of multimedia content over the Internet. However, transmitting real-time streaming audio/video across a network is a very demanding task in that it mandates significant bandwidth and quality-of-service (QoS). Due to the underlying protocols, today's Internet technologies support real-time services only in a best-effort manner. The qualities of the audio/video are very sensitive to the network impairments, such as packet losses, bit-errors, delays and jitters. To understand the challenges that currently face streaming video delivery, this project investigates the impact of network conditions to video QoS. Specifically, a network traffic model to simulate the network communication problems has been established based on a 2-state Markov model - Gilbert model. Integrated with Windows Media technologies, a network impairment emulator was implemented. These software tools provide a way to better understand the relationship between transmission parameters vs. video quality of service in an emulation environment, which may not be easily analyzed over the real IP networks. These relationships would give us insights for deriving optimal criteria to improve the QoS of video streaming.Keywords and Phrases: Internet, IP networks, streaming video, quality of service, Windows Media, packet capture, network impairment emulator, Internet traffic mode

The QoSxLabel: a quality of service cross layer label

A quality of service cross layer label

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