Performance of TCP/UDP under Ad Hoc IEEE802.11

TCP is the De facto standard for connection oriented transport layer protocol, while UDP is the De facto standard for transport layer protocol, which is used with real time traffic for audio and video. Although there have been many attempts to measure and analyze the performance of the TCP protocol in wireless networks, very few research was done on the UDP or the interaction between TCP and UDP traffic over the wireless link. In this paper, we tudy the performance of TCP and UDP over IEEE802.11 ad hoc network. We used two topologies, a string and a mesh topology. Our work indicates that IEEE802.11 as a ad-hoc network is not very suitable for bulk transfer using TCP. It also indicates that it is much better for real-time audio. Although one has to be careful here since real-time audio does require much less bandwidth than the wireless link bandwidth. Careful and detailed studies are needed to further clarify that issue.Comment: 9 pages, 5 figures, ICT 2003 (10th International Conference on Telecommunication

Delay aspects in Internet telephony

In this work, we address the transport of high quality voice over the Internet with a particular concern for delays. Transport of interactive audio over IP networks often suffers from packet loss and variations in the network delay (jitter). Forward Error Correction (FEC) mitigates the impact of packet loss at the expense of an increase of the end-to-end delay and the bit rate requirement of an audio source. Furthermore, adaptive playout buffer algorithms at the receiver compensate for jitter, but again this may come at the expense of additional delay. As a consequence, existing error control and playout adjustment schemes often have end-to-end delays exceeding 150 ms, which significantly impairs the perceived quality, while it would be more important to keep delay low and accept some small loss. We develop a joint playout buffer and FEC adjustment scheme for Internet Telephony that incorporates the impact of end-to-end delay on perceived audio quality. To this end, we take a utility function approach. We represent the perceived audio quality as a function of both the end-to-end delay and the distortion of the voice signal. We develop a joint rate/error/playout delay control algorithm which optimizes this measure of quality and is TCP-Friendly. It uses a channel model for both loss and delay. We validate our approach by simulation and show that (1) our scheme allows a source to increase its utility by avoiding increasing the playout delay when it is not really necessary and (2) it provides better quality than the adjustment schemes for playout and FEC that were previously published. We use this scheme in the framework of non-elevated services which allow applications to select a service class with reduced end-to-end delay at the expense of a higher loss rate. The tradeoff between delay and loss is not straightforward since audio sources may be forced to compensate the additional losses by more FEC and hence more delay. We show that the use of non-elevated services can lead to quality improvements, but that the choice of service depends on network conditions and on the importance that users attach to delay. Based on this observation, we propose an adaptive service choosing algorithm that allows audio sources to choose in real-time the service providing the highest audio quality. In addition, when used over the standard IP best effort service, an audio source should also control its rate in order to react to network congestion and to share the bandwidth in a fair way. Current congestion control mechanisms are based on packets (i.e., they aim to reduce or increase the number of packets sent per time interval to adjust to the current level of congestion in the network). However, voice is an inelastic traffic where packets are generated at regular intervals but packet size varies with the codec that is used. Therefore, standard congestion control is not directly applicable to this type of traffic. We present three alternative modifications to equation based congestion control protocols and evaluate them through mathematical analysis and network simulation

Congestion Control using FEC for Conversational Multimedia Communication

In this paper, we propose a new rate control algorithm for conversational multimedia flows. In our approach, along with Real-time Transport Protocol (RTP) media packets, we propose sending redundant packets to probe for available bandwidth. These redundant packets are Forward Error Correction (FEC) encoded RTP packets. A straightforward interpretation is that if no losses occur, the sender can increase the sending rate to include the FEC bit rate, and in the case of losses due to congestion the redundant packets help in recovering the lost packets. We also show that by varying the FEC bit rate, the sender is able to conservatively or aggressively probe for available bandwidth. We evaluate our FEC-based Rate Adaptation (FBRA) algorithm in a network simulator and in the real-world and compare it to other congestion control algorithms

Management of Digital Video Broadcasting Services in Open Delivery Platforms

The future of Digital Video Broadcasting (DVB) is moving towards solutions offering an efficient way of carrying interactive IP multimedia services over digital terrestrial broadcasting networks to handheld terminals. One of the most promising technologies is Digital Video Broadcasting-Handheld (DVB-H), at present under standardisation. Services deployed via this type of DVB technologies should enjoy reliability comparable to TV services and high quality standards. However, the market at present does not provide effective and economical solutions for the deployment of such services over multi-domain IP networks, due to their high level of unreliability. This paper focuses on service management, service level agreement (SLA) and network performance requirements of DVB-H services. Experimental results are presented concerning QoS sensitivity to network performance of DVB-H services delivered over a multi-domain IP network. Moreover, a solution for efficient and cost effective service management via QoS monitoring and control and network SLA design is proposed. The solution gives DVB-H operators the possibility of fully managing service QoS without being tied to third party operators

A two-level Markov model for packet loss in UDP/IP-based real-time video applications targeting residential users

The packet loss characteristics of Internet paths that include residential broadband links are not well understood, and there are no good models for their behaviour. This compli- cates the design of real-time video applications targeting home users, since it is difficult to choose appropriate error correction and concealment algorithms without a good model for the types of loss observed. Using measurements of residential broadband networks in the UK and Finland, we show that existing models for packet loss, such as the Gilbert model and simple hidden Markov models, do not effectively model the loss patterns seen in this environment. We present a new two-level Markov model for packet loss that can more accurately describe the characteristics of these links, and quantify the effectiveness of this model. We demonstrate that our new packet loss model allows for improved application design, by using it to model the performance of forward error correction on such links

QoS multicast for DiffServ on MPLS and IP platforms

Multicasting has become increasingly important with the emergence of Internet-based applications such Internet protocol (IP) telephony, audio/video conferencing, distributed databases and software upgrading. IP Multicasting is an efficient way to distribute information from a single source to multiple destinations at different locations. One of the challenges the Internet is facing today is to keep the packet forwarding performance up with the skyrocketing demand for bandwidth. On the other hand, the MultiProtocol Label Switching (MPLS), which is an Internet Engineering Task Force (IETF) framework, combines the flexibility of layer 3 routing and layer 2 switching, which enhances network performance in terms of scalability, computational complexity, latency and control message overhead. Besides, MPLS offers a vehicle for enhanced network services such as Quality of Services (QoS)/Class of Service (CoS), Traffic Engineering and Virtual Private Networks (VPNs). In this thesis, we present a new Fair Share Policy (FSP), which is a traffic policing mechanism that utilizes Differentiated Services (DiffServ) to solve the problems of QoS and congestion control. We compare the QoS performance of IP and MPLS multicasting, given their particular constraints. In order to achieve the required QoS, different techniques of reliable multicasting are adapted, such as Forward Error Correction (FEC), Automatic Repeat Request (ARQ) or Hybrid FEC/ARQ with multicast or unicast repairs mechanisms so as to mitigate the effect of errors as well as packet loss. This reliable multicast is for both IP and MPLS platforms with Diffserv. Analytical and simulation models are suggested and employed. The results provide insights into the comparisons between IP multicast in MPLS networks using FSP and plain IP multicasting using the same policy when DiffServ is adopted and when reliable multicast is considered. This comparison will be based on the following QoS measures: total packet delay, delay jitter and residual packet loss probability. Analysis and simulation tools are used to evaluate our fair share policy (FSP) for different homogeneous (when all routers are identical in their capabilities) and heterogeneous (when routers have different capabilities) network scenarios

Multicast Services for Multimedia Collaborative Applications

This work aims at providing multicast services for multimedia collaborative applications over large inter-networks such as the Internet. Multimedia collaborative applications are typically of small group size, slow group membership dynamics, and awareness of participants\u27 identities and locations. Moreover, they usually consist of several components such as audio, video, shared whiteboard, and single user application sharing engines that collectively help make the collaboration session successful. Each of these components has its demands from the communication layer that may differ from one component to another. This dissertation identifies the overall characteristics of multimedia collaborative applications and their individual components. It also determines the service requirements of the various components from the communication layer. Based on the analysis done in the thesis, new techniques of multicast services that are more suitable for multimedia collaborative applications are introduced. In particular, the focus will be on multicast address management and connection control, routing, congestion and flow control, and error control. First, we investigate multicast address management and connection control and provide a new technique for address management based on address space partitioning. Second, we study the problem of multicast routing and introduce a new approach that fits the real time nature of multimedia applications. Third, we explore the problem of congestion and flow control and introduce a new mechanism that takes into consideration the heterogeneity within the network and within the processing capabilities of the end systems. Last, we exploit the problem of error control and present a solution that supports various levels of error control to the different components within the collaboration session. We present analytic as well as simulation studies to evaluate our work, which show that our techniques outperform previous ones