153 research outputs found
Xstream-x264: Real-time H.264 streaming with cross-layer integration
We present Xstream-x264: a real-time cross-layer video streaming technique implemented within a well known open-source H.264 video encoder tool x264. Xstream-x264 uses the transport protocol provided indication of the available data rate for corresponding adjustments in the video encoder.We discuss the design, implementation and the quality evaluation methodology utilised with our tool.We demonstrate via experimental results that the streaming video quality greatly improves with the presented cross-layer approach both in terms of lost frame count and the objective video quality metrics Peak Signal to Noise Ratio (PSNR)
Experimental performance of DCCP over live satellite and long range wireless links
We present experimental results for the performance over satellite and long range wireless (WiMax) links of the new TCP-Friendly Rate Control (TFRC) congestion control mechanism from the Datagram Congestion Control Protocol (DCCP) proposed for use with real-time traffic. We evaluate
the performance of the standard DCCP/CCID3 algorithm and
identify two problem areas: the measured DCCP/CCID3 rate
is inferior to the rate achievable with standard TCP and
a significant rate oscillation continuously occurs making the resulting rate variable even in the short term. We analyse the links and identify the potential causes, i.e. long and variable delay and link errors. As a second contribution, we propose a change in the DCCP/CCID3 algorithm in which the number of feedback messages is increased from the currently standard of at least one per return trip time. Although it is recognised that the increase in control traffic may decrease the overall efficiency, we demonstrate that the change results in higher data rates which are closer to what is achievable with TCP on those networks and that the overhead introduced remains acceptable
On the quality of VoIP with DCCP for satellite communications
We present experimental results for the performance of selected voice codecs using DCCP with CCID4 congestion control over a satellite link. We evaluate the performance of both constant and variable data rate speech codecs for a number of simultaneous calls using the ITU E-model. We analyse the sources of packet losses and additionally analyse the effect of jitter which is one of the crucial parameters contributing to VoIP quality and has, to the best of our knowledge, not been considered previously in the published DCCP performance results. We propose modifications to the CCID4 algorithm and demonstrate how these improve the VoIP performance, without the need for additional link information other than what is already monitored by CCID4. We also demonstrate the fairness of the proposed modifications to other flows. Although the recently adopted changes to TFRC specification alleviate some of the performance issues for VoIP on satellite links, we argue that the characteristics of commercial satellite links necessitate consideration of further improvements. We identify the additional benefit of DCCP when used in VoIP admission control mechanisms and draw conclusions about the advantages and disadvantages of the proposed DCCP/CCID4 congestion control mechanism for use with VoIP applications
Performance of VoIP with DCCP for satellite links
We present experimental results for the performance of selected voice codecs using the Datagram Congestion Control Protocol (DCCP) with TCP-Friendly Rate Control (TFRC) congestion control mechanism over a satellite link. We evaluate the performance of both constant and variable data rate speech codecs (G.729, G.711 and Speex) for a number of simultaneous calls, using the ITU E-model and identify problem areas and potential for improvement. Our experiments are done on a commercial satellite service using a data stream generated by a VoIP application,
configured with selected voice codecs and using the DCCP/CCID4 Linux implementation. We analyse the sources of packet losses which are a main contributor to reduced
voice quality when using CCID4 and additionally analyse the
effect of jitter which is one of the crucial parameters contributing to VoIP quality and has, to the best of our knowledge, not been considered previously in the published DCCP performance results. We propose modifications to the CCID4 algorithm and demonstrate how these improve the VoIP performance, without the need for additional link information other than what is already monitored by CCID4 (which is the case for Quick-Start). We also demonstrate the fairness of the proposed modifications to other flows. We identify the additional benefit of DCCP when used in VoIP admission control mechanisms and draw conclusions
about the advantages and disadvantages of the proposed
DCCP/ CCID4 congestion control mechanism for use with VoIP
applications
Mitigating the impact of packet reordering to maximize performance of multimedia applications
We propose a solution to mitigate the performance degradation and corresponding Quality of Experience (QoE) reduction caused by packet reordering for multimedia applications which utilise unreliable transport protocols like the Datagram Congestion Control Protocol (DCCP). We analytically derive the optimum buffer size based on the applications data rate and the maximum delay tolerated by the multimedia application. We propose a dynamically adjustable buffer in the transport protocol receiver which uses this optimum buffer size. We demonstrate, via simulation results, that our solution reduces the packet loss rate, increases the perceived bandwidth and does not increase jitter in the received applications packets while still being within the application's delay limits, therefore resulting in an increased QoE of multimedia applications
Memory and Complexity Analysis of On-the-Fly Coding Schemes for Multimedia Multicast Communications
A new class of erasure codes for delay-constraint applications, called on-the-fly coding, have recently been introduced for their improvements in terms of recovery delay and achievable capacity. Despite their promising characteristics, little is known about the complexity of the systematic and non-systematic variants of this code, notably for live multicast transmission of multimedia content which is their ideal use case. Our paper aims to fill this gap and targets specifically the metrics relevant to mobile receivers with limited resources: buffer size requirements and computation complexity of the receiver. As our contribution, we evaluate both code variants on uniform and bursty erasure channels. Results obtained are unequivocal and demonstrate that the systematic codes outperform the nonsystematic ones, in terms of both the buffer occupancy and computation overhead
Performance evaluation of multipath transport protocol in heterogeneous network environments
Performance of multipath transport protocols is known to be sensitive to path asymmetry. The difference between each path in terms of bandwidth, delay and packet loss has a potential to significantly decrease the overall performance of a data flow carried over multiple asymmetric paths. In this paper, we evaluate and analyse reliable data transfer in Concurrent Multipath Transfer extension of Stream Control Transport Protocol (CMT-SCTP) under various conditions of network asymmetry, with a focus on the use case where 3G and Wi-Fi networks are simultaneously available. We identify various causes of performance degradation, review the impact of CMT-SACK extension under path asymmetry and show that the total achievable goodput of a reliable in-order data flow over multiple heterogeneous paths is ruled by the characteristics of the worst path as perceived by the transport protocol. To support our study, we derive a simple analytical model of the receiver window blocking and validate it via simulation
Improvements in DCCP congestion control for satellite links
We propose modifications in the TCP-Friendly Rate Control (TFRC) congestion control mechanism from the Datagram Congestion Control Protocol (DCCP) intended for use with real-time traffic, which are aimed at improving its performance for long delay (primarily satellite) links. Firstly, we propose an algorithm to optimise the number of feedback messages per round trip time (RTT) rather than use the currently standard of at least one per RTT, based on the observed link delay. We analyse the improvements achievable with proposed modification in different phases of congestion control and present results from simulations with modified ns-2 DCCP and live experiments using the modified DCCP Linux kernel implementation. We demonstrate that the changes results in improved slow start performance and a reduced data loss compared to standard DCCP, while the introduced overhead remains acceptable
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