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.
From the audio perspective, many researchers focus on the audio flows forwarding process. In more details, not all participants are interested in receiving audio from all transmitters. For this reason, participants should declare their interests and forwarding is based on this profiling type. This method is implemented by Selecting Forwarding Unit (SFU) @cite_30 @cite_18 @cite_5 @cite_19 @cite_9 . On the other hand, collecting, mixing and forwarding all audio flows is proposed using another entity called Multipoint Conferencing Unit (MCU) @cite_15 @cite_1 . Additionally, a common trend in NMP projects is that Session Initiation Protocol (SIP) is used to support control messages among transmitter and receiver side. SIP is a protocol widely used in parallel with Real Time Protocol (RTP) for initial handshaking and dynamic transmission modifications during runtime @cite_25 @cite_24 @cite_4 @cite_26 @cite_0 .
@cite_30 @cite_18 @cite_26 @cite_4 @cite_9 @cite_1 @cite_24 @cite_19 @cite_0 @cite_5 @cite_15 @cite_25
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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. A multi-channel parallel transmission protocol is proposed for the medium access control in cognitive radio networks (CRNs). This protocol contains two key elements: multi-channel assignment and multi-channel contention. For an incoming flow-based connection request, the minimum number of parallel channels are assigned to satisfy the rate and interference mask constraints. For the contention of the assigned channels, our protocol provides an extension of the single-channel RTS- CTS-DATA-ACK handshaking of the IEEE 802.11 scheme. The proposed MAC coherently integrates optimization results into a practical implementation. Through numerical examples, we verify that our protocol provides lower connection blocking probability and higher system throughput for CRNs than its single-channel counterpart. 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. In this paper, we present a reservation-based medium access control (MAC) protocol with multicast support for wavelength-division multiplexing networks. Our system is based on the single-hop, passive optical star architecture. Of the available wavelengths (channels), one channel is designated as a control channel, and the remaining channels are used for data transmission. Each node is equipped with a pair of fixed transceiver to access the control channel, and a fixed transmitter and a tunable receiver to access data channels. For easy implementation of the protocol in hardware and for precisely computing the protocol's processing overhead, we give a register-transfer model of the protocol. We simulate the protocol to study its throughput behavior, and present its analytic model. For a node to be able to send data packets in successive data slots with no time gap between them, in spite of the situation that the protocol's execution time may be longer than data transmission time, we propose the idea of multiple MAC units at each node. Unicast throughput of our protocol reaches the theoretically possible maximum throughput for MAC protocols with distributed control, and the multicast throughput is at least as good as, and even better than, those delivered by existing MAC protocols with distributed control.