Golam Sarwar

DAPS: Intelligent delay-aware packet scheduling for multipath transport

The increasing heterogeneity and asymmetry in wireless network environments makes QoS guarantees in terms of delays and throughput a challenging task. In this paper, we study a novel scheduling algorithm for multipath transport called Delay Aware Packet Scheduling (DAPS) which aims to reduce the receivers buffer blocking time considered as a main parameter to enhance the QoS in wireless environments. We develop an analytical model of maximum receivers buffer blocking time and extend the DAPS algorithm considering implementation issues. Performance evaluations based on ns-2 simulations highlight the enhanced QoS that DAPS can provide. With reference to the classical multipath transport protocol CMT-SCTP, we observe a significant reductions of the receivers buffer occupancy, down by 77%, and the application delay, down by 63%.

Mitigating Receiver's Buffer Blocking by Delay Aware Packet Scheduling in Multipath Data Transfer

Reliable in order multipath data transfer under asymmetric heterogeneous network conditions has known problems related to receivers buffer blocking, caused by out of order packet arrival. Several mitigation techniques have been proposed to address this issue mostly by using various packet retransmission schemes, load-balancing and bandwidth-estimation based mechanisms. In comparison to the existing reactive techniques for buffer block mitigation, we propose a novel and yet simpler to implement, delay aware packet scheduling scheme for multipath data transfer over asymmetric network paths, that proactively minimizes the blocking inside receivers buffer. Our initial simulation results show that, in comparison to the default round robin packet scheduler, by using our proposed delay aware packet scheduling scheme, we can significantly improve overall performance while notably minimizing the receivers buffer usage, which is also beneficial for multi-homed hand-held mobile devices with limited buffering capacity, which, due to their multi-homing and heterogeneous wireless network features (i.e. availability of 3G and Wi-Fi) are also one of the most common use cases for multi-path transport.

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.

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.

Understanding the impact of TFRC feedbacks frequency over long delay links

TFRC is a transport protocol specifically designed to carry multimedia streams. TFRC does not enable a reliable and in order data delivery services. However TFRC implements a congestion control algorithm which is friendly with TCP. This congestion control relies in a feedback mechanism allowing receivers to communicate to the senders an experienced drop rate. Although the current TFRC RFC states that there is little gain from sending a large number of feedback messages per RTT, recent studies have shown that in long-delay contexts, such as satellite-based networks, the performance of TFRC can be improved by increasing the feedback frequency. Nevertheless, currently it is not clear how and why this increase may improve the performance of TFRC. Therefore, in this paper, we aim at understanding the impact that multiple feedback per RTT may have (i) on the key parameters of TFRC (RTT and error rate) and (ii) on the network parameters (reactiveness, fairness and link utilization). We also provide a detailed description of the micromechanisms at the origin of the improvements of the TFRC behavior when multiple feedback per RTT are delivered, and determine the context where such feedback frequencies should be applied.

eCMT-SCTP: Improving performance of multipath SCTP with erasure coding over lossy links

Performance of transport protocols on lossy links is a well researched topic, however there are only a few proposals which make use of the opportunities presented by the use of erasure coding within the transport layer, particularly in the multipath transport protocol context. In this paper, we investigate the improvements in the performance of multipath SCTP transport protocol brought by the novel integration of the on-the-fly erasure code into the congestion control and reliability mechanisms of CMT-SCTP. Our contributions include: integration of transport protocol and erasure codes with regards to congestion control and reliability mechanisms; proposal for a variable retransmission delay parameter in the sender (aRTX) adjustment; evaluation of the performance of CMT-SCTP with erasure coding with simulations. Our results show that we can achieve from 10% to 80% improvements in application goodput under lossy multipath network conditions without a significant penalty i.e. with a minimal (10%) overhead due to the encoding-decoding process. We further evaluate the performance of video streaming using an equivalent of partially reliable CMT-SCTP with erasure coding and again demonstrate a solid performance improvement for our proposal, compared to CMT-SCTP. Finally, we provide an analytical evaluation of CMT-SCTP with erasure codes and validate this with simulation results.

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.

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.

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 for on-line video quality evaluation experiments. 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 utilized 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).

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 applications delay limits, therefore resulting in an increased QoE of multimedia applications.