Networked Music Performance (NMP) constitutes a class of ultra-low delay sensitive applications, allowing geographically separate musicians to perform seamlessly as a tele-orchestra. For this application type, the QoS indicator is the mouth-to-ear delay, which should be kept under 25 milliseconds. The mouth-to-ear delay comprises signal processing latency and network delay. We propose a strong collaboration between the network and NMP applications to actively keep the to mouth-to-ear delay minimal, using direct state notifications. Related approaches can be characterized as passive, since they try to estimate the network state indirectly, based on the end application performance. Our solution employs Software Defined Networking (SDN) to implement the network-to-application collaboration, being facilitated by the well-defined network interface that SDN offers. Emulation results show that the proposed scheme achieves an improvement of up to 59 in mouth-to-ear delay over the existing passive solutions.
Low latency is essential for videoconferencing applications such as presence and collaboration between remote participants. In modern videoconferencing systems, the Selective Forwarding Unit (SFU) has the role of transparently duplicating and forwarding media streams between participants, hence it must be able to process large volumes of incoming packets at very high rates. SFU performance is heavily affected by the network I O mechanisms employed to move packets from the Network Interface Card (NIC) to user space, and then move the copies back to the NIC. Traditional mechanisms, such as POSIX sockets, are not designed for high performance networking and prove to be a major bottleneck in such scenarios, by increasing packet latency and undermining the SFU’s scalability. In this paper, we present a novel SFU platform which was designed to handle the ultra-low latency requirements of Networked Music Performance (NMP) applications, that is, the collaboration of musicians in real time. We implement a prototype SFU based on POSIX sockets and outline its performance bottlenecks. To overcome them, we turn to the netmap framework for fast packet I O, which provides direct but safe access to the NIC buffers. We argue that ultralow latency videoconferencing is a natural application for netmap and thus design and implement a netmap-based SFU. With the current Internet bandwidth capacities and machine processing performance the personal computer has become an affordable and flexible multimedia platform for high quality audio and video content. Besides the delivery of services such as TV, telephone and radio, the Internet can also be used for the exchange of musical information. Due to the variety and complexity of already existing remote music and communication approaches, an elaboration on this topic is mandatory, which covers any relevant musical, technical or interdisciplinary aspect of remote musical interaction. Therefore, this paper gives an overview of current applied technologies and possibilities with their theoretical background. The Session Initialion Protocol has received tremendous market acceptance since it became an official IETF Internet communications standard in 1999. SIP is the technology that makes possible multimedia communications sessions on the Web—ones that allow voice, video, chat, interactive games, and others to run all at the same time. Now that the deployment of real SIP networks is about to take off, two leaders of the commercial rollout deliver complete guidance on this exciting new technology. Geared to IT and networking professionals and decision-makers at Internet service providers (ISPs), as well as networking (NSPs) and application (ASPs) service providers, this book helps readers sort through the available vendor offerings and services to discover how to integrate and maximize SIP's power across their networks.  Advances in information technology and the great proliferation of the Internet have changed nearly every aspect of the work and life of human beings. Despite progress in networked entertainment, many music professionals and enthusiasts are still sticking to the traditional way of carrying out rehearsals and concerts. Music performance in this way requires physical presence of the participants and has a number of inherent limitations. We introduce a novel system called network-centric music performance (NMP) that enables multiparty music performance through cyberspace. Our target is to support real-time multichannel natural audio streaming over the network, using audio compression schemes that can provide acceptable audio quality. A system like this is bandwidth-demanding and highly delay-sensitive, and requires synchronization of the audio streams. Hence, support from the underlying end systems and networks is critical. However, the current source coding mechanisms and the best effort nature of the internet pose many challenges to achieve the desired quality of service. We have implemented a prototype of NMP, and exploited end system and network influences on NMP. The work was done in a LAN environment using Linux PCs. The system enables two different application scenarios: real-time rehearsal and rehearsal on demand. Real-time multichannel audio transport and different audio compression schemes are supported. Our evaluation results based on both subjective and objective measurements show that the system provides sufficient audio quality level for the target application in such an environment. The scalability test also revealed that the system scales well with increase of clientele. In the future, we extend our system for networks spanning larger distances and experiment with more realistic network conditions in the Internet.    SCTP is a newly developed transport protocol tailored for signaling transport. Whereas in theory SCTP is supposed to achieve a much better performance than TCP and UDP, at present there are no experimental results showing SCTP's real benefits. This article analyzes SCTP's strengths and weaknesses and provides simulation results. We implemented SIP on top of UDP, TCP, and SCTP in the network simulator and compared the three transport protocols under different network conditions.  The MusiNet research project aims to provide a comprehensive architecture and a prototype implementation of a complete Networked Music Performance (NMP) system. In this paper we describe the current status of the project, focusing on critical decisions regarding the system’s architecture and specifications, the low delay audio and video coding techniques to be employed, the media relay design, and the synchronous and asynchronous collaboration algorithms to be adopted. Today's over the top (OTT) video service providers take advantage of content distribution networks (CDNs) and adaptive bitrate (ABR) streaming where a video player adjusts resolutions based on end-to-end network conditions. Although the mechanisms are useful to improve user-perceived video quality, they do not resolve the root causes of congestion problems. To pinpoint a bottleneck and improve video quality-of-experience (QoE), we leverage a software-defined networking (SDN) platform from OTT video service provider's point of view. Our proposed SDN application is designed to monitor network conditions of streaming flow in real time and dynamically change routing paths using multi-protocol label switching (MPLS) traffic engineering (TE) to provide reliable video watching experience. We use an off-the-shelf SDN platform to show the feasibility of our approaches.