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.

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.

Multimedia applications over IEEE 802.11 wireless LANs (WLANs) such as wVoIP have recently attracted the focus of many researchers. Unlike best-effort applications, multimedia applications are delay- and/or bandwidth-sensitive. In order for these applications to achieve an acceptable QoS, the network must provide them with certain QoS guarantees. This paper proposes an admission control heuristic for applications with delay requirements in IEEE 802.11e EDCA WLANs. To develop the proposed heuristic, we first derive an analytical approximation of delays experienced by packets when delivered via IEEE 802.11e EDCA WLANs. We validate the heuristic through simulations of voice traffic

The integrated networking technology prototype demonstrates the capability of several emerging networking technologies to operate together seamlessly to enhance network services targeted for dynamic mobile environments such as those found in the battlefield. The prototype consists of four technologies: (1) autoconfiguration technology supports autonomous and rapid network deployment and configuration; (2) self-managed virtual network (SMVN) technology provides virtual networking capabilities for networks which do not natively support these functions; (3) integrated mobility management technology supports session continuity in the presence of node mobility; (4) assured IP QoS technology supports guarantees for mission-critical applications. The prototype is capable of operating over the various types of equipment and protocols to be utilized in battlefield networks. The prototype is designed such that network services and functions are survivable and reconfigurable. In this paper, we describe the four technologies as well as the integrated prototype in the laboratory environment.

We are currently witnessing a boom in the deployment and usage of wireless local area networks (WLANs). Once only seen within the enterprise, WLANs are increasingly making their way into residential, commercial, industrial, and public areas. The recent efforts of carriers to integrate WLANs into their wide-area service offerings are testimony to their growing role in the future of wireless networking. Voice continues to be the most predominant wireless application, and, in order to integrate fully with existing and future cellular systems, WLANs must deliver a high quality voice service. We present an experimental evaluation of VoIP capacity in an IEEE 802.11b network. Moreover, we identify and quantify the contributing factors that limit the capacity to less than 10 voice calls per access point.

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.

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.

We describe a scheme called TCP-SMART to improve the end-to-end throughput of TCP connections in the spotty wide band environment. In this scheme, a TCP-SMART agent at the base station uses local retransmissions, acknowledgment filtering, and selective generation of duplicate acknowledgments to increase the TCP throughput despite extended periods of poor channel conditions and short disconnections. When compared to other schemes, TCP-SMART achieves higher end-to-end TCP throughput.

In a Code Division Multiple Access (CDMA) network, multiple mobile hosts (MHs) can simultaneously transmit over the wireless channel by using different codes. To assure an acceptable quality of service for all users' flows, the network usually tunes the transmit powers of all MHs to achieve a certain level of signal strength as compared to the noise and the interference (SINR) for each user. The traditional assumption in power control schemes is that the SINR requirement is statically determined for each user flow.

The growing demand for mobility has increased the need to develop more efficient, reliable and cost effective services for data transmission over the air interface. The air interface, as compared to the wired domain, is characterized by a high bit-error-rate, limited bandwidth, and intermittent connectivity. In addition, power in the hand-held devices and laptops is limited by the battery technology. We present AFEC, an adaptive forward error-correction scheme, which makes effective use of the air interface by minimizing the number of data bits transmitted to convey message packets in a real-time stream over an air interface. Through simulations, we show that AFEC outperforms the traditional single forward error-correction scheme.