Moncef Elaoud

Voice performance in WLAN networks - an experimental study

In this work, we measure wireless local area network (WLAN) voice performance and capacity. While most WLAN applications today are data centric, the growing popularity of Voice over IP (VoIP) applications and the escalating trend towards convergence with cellular networks will catalyze an increased mix of voice traffic. Since voice applications must compete with each other and with simultaneous data applications for WLAN bandwidth, quantifying voice performance and capacity in the presence of background data traffic is an important issue. We offer a practical investigation into the ability of 802.11b MAC layer to support simultaneous voice and data. We quantify VoIP capacity for standard WLAN networks, indicative of those already in the field, as well as evaluate the practical benefits of implementing backoff control and priority queuing at the access point. Conclusions are drawn based on an extensive set of real-world measurements conducted using off-the-shelf equipment in a commercial testbed.

Implementing a testbed for mobile multimedia

In an effort to realize wireless Internet telephony and multimedia streaming in a highly mobile environment a testbed emulating a wireless Internet has been built. This allows the setting up of multimedia calls between IP mobiles and integration between IP and PSTN end-points in a wireless environment. Different functionalities and components involved with the wireless Internet streaming multimedia have been prototyped and experimented in the testbed. These include signaling, registration, dynamic binding, location management as well as supporting the QoS features for the mobile users. This paper describes some of the components of the testbed and highlights the experiences while building this testbed which could be beneficial to some who plan to build a similar testbed to realize several features and capabilities of Mobile Wireless Internet, before actually bringing to the market.

Securing ad-hoc networks using IPsec

The use of IPSec for securing communication between nodes of wireless and mobile ad hoc networks has traditionally been considered difficult. We describe an IPSec-based architecture and implementation for ad hoc networks that can seamlessly handle node mobility and IP address change. The approach can he used for securing application traffic as well as configuration and mobility management protocol traffic. A certificate-based approach that aids dynamic key generation and distribution is used for creating security associations between nodes. Simple and backward compatible extensions to the IPSec and PKIX protocols that do not violate existing and proposed standards are described, and an existing implementation is discussed. Initial experimental evaluation reveals that the per-packet latency overhead at the end-host for using our proposed mechanisms is tolerable.

Realizing mobile wireless Internet telephony and streaming multimedia testbed

Streaming real-time multimedia content over the Internet is gaining momentum in the communications, entertainment, music and interactive game industries as well as in the military. In general, streaming applications include IP telephony, multimedia broadcasts and various interactive applications such as multi-party conferences, collaborations and multiplayer games. Successfully realizing such applications in a highly mobile environment, however, presents many research challenges. In order to investigate such challenges and demonstrate viable solutions, we have developed an experimental indoor and outdoor testbed laboratory. By implementing standard IETF protocols into this testbed, we have demonstrated the basic functionalities required of the mobile wireless Internet to successfully support mobile multimedia access. These requirements include signaling, registration, dynamic configuration, mobility binding, location management, Authentication Authorization and Accounting (AAA), and quality of service over a variety of radio access network (RAN) technologies (e.g. 802.11b, CDMA/GPRS). In this paper, we describe this testbed and discuss important design issues and tradeoffs. We detail the incorporation and inter-relation of a wide catalog of IETF protocols-such as SIP, SAP, SDP, RTP/RTCP/RTSP, MGCP, variants of Mobile-IP, DRCP, HMMP, PANA, and DSNP-to achieve our goals. We believe that the results and experiences obtained from this experimental testbed will advance the understanding of the pertinent deployment issues for a Mobile Wireless Internet.

A delay-based admission control mechanism for multimedia support in IEEE 802.11e wireless LANs

Multimedia over IEEE 802.11 wireless local area networks (WLANs) has recently been the focus of many researchers due to its rapidly increasing popularity. Unlike their best-effort counterparts, multimedia applications have quality of service (QoS) needs typically expressed in terms of the maximum allowed delay and/or the minimum required throughput. Therefore, prior to accepting a multimedia application, the network must assure the satisfaction of its QoS requirements. In this paper, we develop a mechanism that can be used to control the admissibility of multimedia applications into WLANs. To develop the proposed mechanism, we first derive an analytical approximation of the delay experienced by packets when travelled through these networks. The analytical approximation of the delay is then used to propose an admission control mechanism for the enhanced distributed channel access (EDCA) method used by the hybrid coordination function (HCF) of IEEE 802.11e. The proposed delay-based admission control mechanism is validated via simulations of voice traffic.

Experimental VoIP capacity measurements for 802.11b WLANs

There is an increasing interest in supporting voice over IP (VoIP) applications over wireless local area networks (WLANs). To provide quality of service guarantee to voice traffic, an admission control mechanism must be administered to avoid over loading the network. In this paper, we present an experimental evaluation of VoIP capacity in WLANs under various data and voice loads. The obtained capacity estimates can be used in admission control engine to make an admission or a rejection decision. In this paper, we identify a suitable evaluation metric to quantify the performance of a voice call. The metric accounts for packet losses and packet delays for each voice flow. We then define voice capacity in a WLAN and present experimental capacity measurements under various background data traffic loads. The experimental capacity measurements are immediately useful for WLAN hotspot providers and enterprise WLAN architects.

The effect of packetization on voice capacity in IEEE 802.11b networks

This paper reports on simulation studies of voice services in a wireless local area network (WLAN). We focus on the widespread IEEE 802.11b standard and present both analytical models and simulation results. Using quantifiable metrics based on packet losses and delays, we advance a definition of access point (AP) capacity as the maximum number of simultaneous voice calls that an AP can support with a desired level of service quality. We examine the role of codec packetization on WLAN channel efficiency, and through simulation show its significant impact on voice capacity. We further report on the dynamics of the collision probability, packet drop rate and medium activity time as more active voice users enter the system.

Performance Evaluation of a Unicast Routing Control Agent for Proactive Diverse Link Selection

The goals of URCA are two-fold: 1) to ensure that links are evenly loaded. URCA attempts to set link costs that ensure that most links are utilized for carrying traffic. This has the result of maximizing the effective capacity or increasing the number of users that can be supported by a given network infrastructure and 2) to ensure that the effect of soft link failures is minimized. These goals must be balanced against frequent network routing re-configuration which can be disruptive and oscillations where traffic moves back and forth between a set of links due to the reconfiguration. This paper describes the functionality of the unicast routing control agent. We discuss the challenges that must be addressed in designing an effective heuristic. We report results from a simulation study to evaluate the performance gains, such as the increase in aggregate throughput, from deploying URCA compared to the case where conventional routing is used. Our simulation experiments show that using URCA can greatly enhance throughput, and reduce delay and packet loss when deployed in a wireless tactical network

Self-initiated and self-maintained overlay networks (SIMONS) for enhancing military network capabilities

Future military networks such as FCS and WIN-T will exploit commercial protocols, such as dynamic IP routing (e.g., OSPF1 PIM and BGP) and DiffServ for QoS. This raises the question of how to provide additional network features needed for many military applications. For example, how can military networks add functions such as multipath routing, efficient network processing, and late binding IPSec tunnels that are vital to providing enhanced robustness, efficiency, security and QoS? In many cases the best approach to provide these enhanced features is to use overlay networks. Examples include application-layer multicasting, Web distribution networks, resilient overlay networks, IPv6 over IPv4 tunnels, and HAIPE. One significant challenge with all these overlay networks is the configuration overhead and resultant management complexity. To address this issue, this paper proposes the use of a new tool that creates self-initiated and self-maintained overlay network (SIMON). Using plug-and-play modules, the tool supports not only a particular need or mission, but supports a wide set of enhanced capabilities. SIMONs are independent of IP addressing and offer the enhanced features without any network intervention. We have designed three main protocols to support the desired capabilities for SIMONs: (a) a dynamic overlay network configuration protocol (DONCP), (b) an overlay network routing protocol (ONRP) and (c) an overlay network multipath routing protocol (ONMRP). In this paper we describe the main features and basic operations of the SIMONs and outline the design and functionalities of DONCP

Cross-layer optimized unicast and multicast routing on overlay networks

Future military networks require capabilities beyond current commercial IP protocols for battlefield situations. These features include enhanced routing, network processing and information-centric networks. We developed a self-initiated and self-maintained overlay network (SIMON) to provide these capabilities. SIMONs aim at providing enhanced features using a methodology similar to IP-based virtual private networks. SIMON is comprised of a suite of protocols that constructs and configures overlay networks as well as routes packets in the virtual network. In this paper, we describe the overlay network routing protocol (ONRP) to provide efficient unicast and multicast routing in SIMONs. ONRP is specifically designed for an overlay network and exploits information available from SIMON and layer 3 routing protocol ONRP uses. In ONRP, each node in the SIMON independently tracks local virtual topology changes. The changes are efficiently distributed to the entire SIMON without the use of flooding, hence reducing ONRPs overhead. ONRP also uses very small update packets. For example, we show that the ONRP overhead is approximately 8 times less than OSPF in a dense 5 connected overlay network. The underlying structure of the topology distribution method is also used to route packets for multicast applications. The ONRP operations, however, are transparent to and independent of the underlying IP routing protocols. We have implemented ONRP on our wireless SIMON test bed. Most of the code runs in application space, with some small modifications to the forwarding engine in the Linux kernel