The Octopus switch

This chapter1 discusses the interconnection architecture of the Mobile Digital Companion. The approach to build a low-power handheld multimedia computer presented here is to have autonomous, reconfigurable modules such as network, video and audio devices, interconnected by a switch rather than by a bus, and to offload as much as work as possible from the CPU to programmable modules placed in the data streams. Thus, communication between components is not broadcast over a bus but delivered exactly where it is needed, work is carried out where the data passes through, bypassing the memory. The amount of buffering is minimised, and if it is required at all, it is placed right on the data path, where it is needed. A reconfigurable internal communication network switch called Octopus exploits locality of reference and eliminates wasteful data copies. The switch is implemented as a simplified ATM switch and provides Quality of Service guarantees and enough bandwidth for multimedia applications. We have built a testbed of the architecture, of which we will present performance and energy consumption characteristics

Survey of unified approaches to integrated-service networks

The increasing demand for communication services, coupled with recent technological advances in communication media and switching techniques, has resulted in a proliferation of new and expanded services. Currently, networks are needed which can transmit voice, data, and video services in an application-independent fashion. Unified approaches employ a single switching technique across the entire network bandwidth, thus, allowing services to be switched in an application-independent manner. This paper presents a taxonomy of integrated-service networks including a look at N-ISDN, while focusing on unified approaches to integrated-service networks.The two most promising unified approaches are burst and fast packet switching. Burst switching is a circuit switching-based approach which allocates channel bandwidth to a connection only during the transmission of "bursts" of information. Fast packet switching is a packet switching-based approach which can be characterized by very high transmission rates on network links and simple, hardwired protocols which match the rapid channel speed of the network. Both approaches are being proposed as possible implementations for integrated-service networks. We survey these two approaches, and also examine the key performance issues found in fast packet switching. We then present the results of a simulation study of a fast packet switching network

Towards zero packet loss with LISP Mobile Node

Host mobility protocols such as Locator-Identifier Separation Protocol Mobile Node (LISP-MN) are known to experience packet loss at the point of handover. For the duration of the handover, packets sent to the MN via the old access link are dropped by the router since it has no way of knowing where the device has moved to. This affects the performance of transport layer protocols of the TCP/IP stack, which results in degradation of network performance. Buffering these packets close to the MN's new location and forwarding them to the MN on handover completion is one way of improving the overall performance of the mobility protocol. Hence, we introduce a novel network node, loc-server, to buffer these packets in order to mitigate the packet loss and reduce the service disruption time (SDT) experienced by MNs during handovers. Using a laboratory testbed implementation, LISP-MN with loc-server support shows significant reduction in packet loss and reduced SDT in comparison to vanilla LISP-MN. Similarly, performance analysis of DASH video player also shows the new architecture helps in improving the average video quality downloaded by the MN and reduces the player's instability

PERFORMANCE ANALYSIS AND PLAYOUT TIME ESTIMATION FOR MULTIMEDIA OVER INTERNET PROTOCOL (MOIP)

This thesis presents the algorithms to estimate the minimum buffering delay and playout delay. The possibility to use the buffering delay estimation in Multimedia Application at the receiver site will reduce the effect of jitter and will also optimize the packet loss. The increasing of Traffic of loaded generator will influence the network behaviour and affect to transmission data using video conferencing netmeeting application over the network connectio

The QUIC Fix for Optimal Video Streaming

Within a few years of its introduction, QUIC has gained traction: a significant chunk of traffic is now delivered over QUIC. The networking community is actively engaged in debating the fairness, performance, and applicability of QUIC for various use cases, but these debates are centered around a narrow, common theme: how does the new reliable transport built on top of UDP fare in different scenarios? Support for unreliable delivery in QUIC remains largely unexplored. The option for delivering content unreliably, as in a best-effort model, deserves the QUIC designers' and community's attention. We propose extending QUIC to support unreliable streams and present a simple approach for implementation. We discuss a simple use case of video streaming---an application that dominates the overall Internet traffic---that can leverage the unreliable streams and potentially bring immense benefits to network operators and content providers. To this end, we present a prototype implementation that, by using both the reliable and unreliable streams in QUIC, outperforms both TCP and QUIC in our evaluations.Comment: Published to ACM CoNEXT Workshop on the Evolution, Performance, and Interoperability of QUIC (EPIQ

The voice activity detection (VAD) recorder and VAD network recorder : a thesis presented in partial fulfilment of the requirements for the degree of Master of Science in Computer Science at Massey University

The project is to provide a feasibility study for the AudioGraph tool, focusing on two application areas: the VAD (voice activity detector) recorder and the VAD network recorder. The first one achieves a low bit-rate speech recording on the fly, using a GSM compression coder with a simple VAD algorithm; and the second one provides two-way speech over IP, fulfilling echo cancellation with a simplex channel. The latter is required for implementing a synchronous AudioGraph. In the first chapter we introduce the background of this project, specifically, the VoIP technology, the AudioGraph tool, and the VAD algorithms. We also discuss the problems set for this project. The second chapter presents all the relevant techniques in detail, including sound representation, speech-coding schemes, sound file formats, PowerPlant and Macintosh programming issues, and the simple VAD algorithm we have developed. The third chapter discusses the implementation issues, including the systems' objective, architecture, the problems encountered and solutions used. The fourth chapter illustrates the results of the two applications. The user documentations for the applications are given, and after that, we analyse the parameters based on the results. We also present the default settings of the parameters, which could be used in the AudioGraph system. The last chapter provides conclusions and future work

Performance of active multicast congestion control

This paper aims to provide insight into the behavior of congestion control mechanisms for reliable multicast protocols. A multicast congestion control based on active networks has been proposed and simulated using ns-2 over a network topology obtained using the Tiers tool. The congestion control mechanism has been simulated under different network conditions and with different settings of its configuration parameters. The objective is to analyze its performance and the impact of the different configuration parameters on its behavior. The simulation results show that the performance of the protocol is good in terms of delay and bandwidth utilization. The compatibility of the protocol with TCP flows has not been demonstrated, but the simulations performed show that by altering the parameter settings, the proportion of total bandwidth taken up by the two types of flow, multicast and TCP, may be modified.Publicad