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

Performance evaluation of TCP, UDP and DCCP for video traffics over 4G network

Fourth Generation (4G) system has been used more widely than the older generations 3G and 2G. Among the reasons are that the 4G’s transfer rate is higher and it supports all multimedia functions. Besides, its’ supports for wide geographical locus makes wireless technology gets more advanced. The essential goal of 4G is to enable voice-based communication being implemented endlessly. To achieve the goal, this study tries to answer the following research questions: (1), are the old protocols suit with this new technology; (2), which one has the best performance and, (3) which one has the greatest effect on throughput, delay, packet delivery ratio and packet loss. The aforementioned questions are crucial in the performance evaluation of the most famous protocols (particularly User Datagram Protocol (UDP), Transmission Control Protocol (TCP), and Datagram Congestion Control Protocol (DCCP)) within the 4G environment. Through the Network Simulator-3 (NS-3), the performance of transporting MPEG-4 video stream including throughput, delay, packet loss, and packet delivery ratio are analyzed at the base station through UDP, TCP, and DCCP protocols over 4G’s Long Term Evolution (LTE) technology. The results show that DCCP has better throughput, and lesser delay, but at the same time it has more packet loss than UDP and TCP. Based on the results, DCCP is recommended as a transport protocol for real time vide

Performance evaluation of TCP, UDP and DCCP traffic over 4G network

Fourth Generation (4G) mobile systems has been used more widely than the older generations 3G and 2G. Among the reasons are that the 4G’s transfer rate is higher and it supports all multimedia functions.Besides, its’ supports for wide geographical locus makes wireless technology gets more advanced.The essential goal of 4G is to enable voice-based communication being implemented endlessly.This study tries to evaluate if the old protocols suit with this new technology.And which one has the best performance and which one has the greatest effect on throughput, delay and packet loss.The aforementioned questions are crucial in the performance evaluation of the most famous protocols (particularly User Datagram Protocol (UDP), Transmission Control Protocol (TCP) and Datagram Congestion Control Protocol (DCCP)) within the 4G environment.Through the Network Simulation-3 (NS3), the performance of transporting video stream including throughput, delay, packet loss and packet delivery ratio are analyzed at the base station through UDP, TCP and DCCP protocols over 4G’s Long Term Evaluation (LTE) technology.The results show that DCCP has better throughput and lesser delay, but at the same time it has more packet loss than UDP and TCP. Based on the results, DCCP is recommended as a transport protocol for real time video

Cross layer techniques for flexible transport protocol using UDP-Lite over a satellite network

Traditional real-time multimedia and streaming services have utilised UDP over RTP. Wireless transmission, by its nature, may introduce a variable, sometimes high bit error ratio. Current transport layer protocols drop all corrupted packets, in contrast, protocols such as UDP-Lite allow error-resilient applications to be supported in the networking stack. This paper presents experimental quantitative performance metrics using H.264 and UDP Lite for the next generation transport of IP multimedia, and discusses the architectural implications for enhancing performance of a wireless and/or satellite environment

TCP-Friendly Rate Control with Token Bucket for VoIP Congestion Control

TCP Friendly Rate Control (TFRC) is a congestion control algorithm that provides a smooth transmission rate for real-time network applications. TFRC refrains from halving the sending rate on every packet drop, instead it is adjusted as a function of the loss rate during a single round trip time. TFRC has been proven to be fair when competing with TCP flows over congested links, but it lacks quality-of-service parameters to improve the performance of real-time traffic. A problem with TFRC is that it uses additive increase to adjust the sending rate during periods with no congestion. This leads to short term congestion that can degrade the quality of voice applications. We propose two changes to TFRC that improve the performance of VoIP applications. Our implementation, TFRC with Token Bucket (TFRC-TB), uses discrete calculated bit rates based on audio codec bandwidth usage to increase the sending rate. Also, it uses a token bucket to control the sending rate during congestion periods. We have used ns2, the network simulator, to compare our implementation to TFRC in a wide range of network conditions. Our results suggest that TFRC-TB can provide a quality of service (QoS) mechanism to voice applications while competing fairly with other traffic over congested links

Building self-optimized communication systems based on applicative cross-layer information

This article proposes the Implicit Packet Meta Header(IPMH) as a standard method to compute and represent common QoS properties of the Application Data Units (ADU) of multimedia streams using legacy and proprietary streams’ headers (e.g. Real-time Transport Protocol headers). The use of IPMH by mechanisms located at different layers of the communication architecture will allow implementing fine per-packet selfoptimization of communication services regarding the actual application requirements. A case study showing how IPMH is used by error control mechanisms in the context of wireless networks is presented in order to demonstrate the feasibility and advantages of this approach

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