End-to-end congestion control protocols for remote programming of robots, using heterogeneous networks: A comparative analysis

There are many interesting aspects of Internet Telerobotics within the network robotics context, such as variable bandwidth and time-delays. Some of these aspects have been treated in the literature from the control point of view. Moreover, only a little work is related to the way Internet protocols can help to minimize the effect of delay and bandwidth fluctuation on network robotics. In this paper, we present the capabilities of TCP, UDP, TCP Las Vegas, TEAR, and Trinomial protocols, when performing a remote experiment within a network robotics application, the UJI Industrial Telelaboratory. Comparative analysis is presented through simulations within the NS2 platform. Results show how these protocols perform in two significant situations within the network robotics context, using heterogeneous wired networks: (1) an asymmetric network when controlling the system through a ADSL connection, and (2) a symmetric network using the system on Campus. Conclusions show a set of characteristics the authors of this paper consider very important when designing an End-to-End Congestion Control transport protocol for Internet Telerobotics

Slight-Delay Shaped Variable Bit Rate (SD-SVBR) Technique for Video Transmission

The aim of this thesis is to present a new shaped Variable Bit Rate (VBR) for video transmission, which plays a crucial role in delivering video traffic over the Internet. This is due to the surge of video media applications over the Internet and the video typically has the characteristic of a highly bursty traffic, which leads to the Internet bandwidth fluctuation. This new shaped algorithm, referred to as Slight Delay - Shaped Variable Bit Rate (SD-SVBR), is aimed at controlling the video rate for video application transmission. It is designed based on the Shaped VBR (SVBR) algorithm and was implemented in the Network Simulator 2 (ns-2). SVBR algorithm is devised for real-time video applications and it has several limitations and weaknesses due to its embedded estimation or prediction processes. SVBR faces several problems, such as the occurrence of unwanted sharp decrease in data rate, buffer overflow, the existence of a low data rate, and the generation of a cyclical negative fluctuation. The new algorithm is capable of producing a high data rate and at the same time a better quantization parameter (QP) stability video sequence. In addition, the data rate is shaped efficiently to prevent unwanted sharp increment or decrement, and to avoid buffer overflow. To achieve the aim, SD-SVBR has three strategies, which are processing the next Group of Picture (GoP) video sequence and obtaining the QP-to-data rate list, dimensioning the data rate to a higher utilization of the leaky-bucket, and implementing a QP smoothing method by carefully measuring the effects of following the previous QP value. However, this algorithm has to be combined with a network feedback algorithm to produce a better overall video rate control. A combination of several video clips, which consisted of a varied video rate, has been used for the purpose of evaluating SD-SVBR performance. The results showed that SD-SVBR gains an impressive overall Peak Signal-to-Noise Ratio (PSNR) value. In addition, in almost all cases, it gains a high video rate but without buffer overflow, utilizes the buffer well, and interestingly, it is still able to obtain smoother QP fluctuation

Performance evaluation of the TFD-capable dynamic QoS assurance of HD video streaming in well-dimensioned network

The Traffic Flow Description (TFD) option of the IP protocol is an experimental option, designed by the Authors and described by the IETF’s Internet Draft. This option was intended for signalling for QoS purposes. Knowledge about forthcoming traffic (such as the amount of data that will be transferred in a given period of time) is conveyed in the fields of the option between end-systems. TFD-capable routers on a path (or a multicast tree) between the sender and receiver(s) are able to read this information, process it and use it for bandwidth allocation. If the time horizons are short enough, bandwidth allocation will be performed dynamically. In the paper a performance evaluation of an HD video transmission QoS assured with the use of the TFD option is presented. The analysis was made for a variable number of video streams and a variable number of TCP flows that compete with the videos for the bandwidth of the shared link. Results show that the dynamic bandwidth allocation using the TFD option better assures the QoS of HD video than the classic solution, based on the RSVP protocol

Analysis domain model for shared virtual environments

The field of shared virtual environments, which also encompasses online games and social 3D environments, has a system landscape consisting of multiple solutions that share great functional overlap. However, there is little system interoperability between the different solutions. A shared virtual environment has an associated problem domain that is highly complex raising difficult challenges to the development process, starting with the architectural design of the underlying system. This paper has two main contributions. The first contribution is a broad domain analysis of shared virtual environments, which enables developers to have a better understanding of the whole rather than the part(s). The second contribution is a reference domain model for discussing and describing solutions - the Analysis Domain Model

Improving the Quality of Real Time Media Applications through Sending the Best Packet Next

Real time media applications such as video conferencing are increasing in usage. These bandwidth intensive applications put high demands on a network and often the quality experienced by the user is sub-optimal. In a traditional network stack, data from an application is transmitted in the order that it is received. This thesis proposes a scheme called "Send the Best Packet Next (SBPN)" where the most important data is transmitted first and data that will not reach the receiver before an expiry time is not transmitted. In SBPN the packet priority and expiry time are added to a packet and used in conjunction with the Round Trip Time (RTT) to determine whether packets are sent, and in which order that they are sent. For example, it has been shown that audio is more important to users than video in video conferencing. SBPN could be considered to be Quality of Service (QoS) that is within an application data stream. This is in comparison to network routers that provide QoS to whole streams such as Voice over IP (VoIP), but do not differentiate between data items within the stream or which data gets transmitted by the end nodes. Implementation of SBPN can be done on the server only, so that much of the benefit for one way transmission (e.g. live television) can be gained without requiring existing clients to be changed. SBPN was implemented in a Linux kernel on top of Datagram Congestion Control Protocol (DCCP) and compared to existing solutions. This showed real improvement in the measured quality of audio with a maximum improvement of 15% in selected test scenarios

Broadband Video Streaming with Built-in Resiliency

Mobile TV services are being actively developed for a variety of last hop, broadband wireless technologies. Application layer error control mechanisms such as Broadband Video Streaming seek to reduce packet loss from raw UDP transport. This paper goes further than existing streaming protocols by integrating source-coded error resilience through data-partitioning and intra-refresh macroblocks with the error control mechanism. Results show that for a temporally complex sequence, up to 6.23 dB gain in video quality (PSNR) can result, depending on burst error lengths across an IEEE 802.16e link

Implementing Real-Time Transport Services over an Ossified Network

Real-time applications require a set of transport services not currently provided by widely-deployed transport protocols. Ossification prevents the deployment of novel protocols, restricting solutions to protocols using either TCP or UDP as a substrate. We describe the transport services required by real-time applications. We show that, in the short-term (i.e., while UDP is blocked at current levels), TCP offers a feasible substrate for providing these services. Over the longer term, protocols using UDP may reduce the number of networks blocking UDP, enabling a shift towards its use as a demultiplexing layer for novel transport protocols